Chapter 1

Amplitude Modulation
1. Introduction

Communication is the process; convey the message from one place to the
other place with the help of Information source, Transmitter, Channel,
Receiver and Destination.

The block diagram of a communication system will have five blocks,

including the information source, input transducer, transmitter, channel
and the noise, receiver and destination blocks.

Figure 1.1: Analog communication block diagram

Information Source

As we know, a communication system serves to communicate a message

or information. This information originates in the information source. In
general, there can be various messages in the form of words, group of
words, code, symbols, sound signal etc. However, out of these messages,
only the desired message is selected and communicated. Therefore, we can
say that the function of information source is to produce required message
which has to be transmitted.

Input Transducer

A transducer is a device which converts one form of energy into another

form. The message from the information source may or may not be

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Chapter 1: Amplitude Modulation

electrical in nature. In a case when the message produced by the

information source is not electrical in nature, an input transducer is used to
convert it into a time-varying electrical signal.

For example, in case of radio-broadcasting, a microphone converts the

information or massage which is in the form of sound waves into
corresponding electrical signal.

Transmitter

The function of the transmitter is to process the electrical signal from

different aspects. For example in radio broadcasting the electrical signal
obtained from sound signal, is processed to restrict its range of audio
frequencies (upto 5 kHz in amplitude modulation radio broadcast) and is
often amplified. In wire telephony, no real processing is needed. However,
in long-distance radio communication, signal amplification is necessary
before modulation. Modulation is the main function of the transmitter. In
modulation, the message signal is superimposed upon the high-frequency
carrier signal. In short, we can say that inside the transmitter, signal
processing such as restriction of range of audio frequencies, amplification
and modulation of signal are achieved. All these processing of the message
signal are done just to ease the transmission of the signal through the
channel.

The Channel and the Noise

The term channel means the medium through which the message travels
from the transmitter to the receiver. In other words, we can say that the
function of the channel is to provide a physical connection between the
transmitter and the receiver.

There are two types of channels, namely point-to-point channels and

broadcast channels.

Example of point-to-point channels is wire lines, microwave links and

optical fibers. Wire-lines operate by guided electromagnetic waves and
they are used for local telephone transmission.

In case of microwave links, the transmitted signal is radiated as an

electromagnetic wave in free space. Microwave links are used in long

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Chapter 1: Amplitude Modulation

distance telephone transmission. An optical fiber is a low-loss, well-

controlled, guided optical medium. Optical fibers are used in optical
communications.

During the process of transmission and reception the signal gets distorted
due to noise introduced in the system. Noise is an unwanted signal which
tends to interfere with the required signal. Noise signal is always random
in character. Noise may interfere with signal at any point in a
communication system. However, the noise has its greatest effect on the
signal in the channel.

Receiver

The main function of the receiver is to reproduce the message signal in

electrical form from the distorted received signal. This reproduction of the
original signal is accomplished by a process known as the demodulation or
detection. Demodulation is the reverse process of modulation carried out
in transmitter.

Destination

Destination is the final stage which is used to convert an electrical

message signal into its original form For example in radio broadcasting,
the destination is a loudspeaker which works as a transducer i.e. converts
the electrical signal in the form of original sound signal.

1.1 Modulation

In the modulation process, two signals are used namely the modulating
signal and the carrier.

The modulating signal is nothing but the baseband signal or information

signal while the carrier is a high frequency sinusoidal signal.

In the modulation process, some parameter of the carrier wave (such as

amplitude, frequency or phase) is varied in accordance with the
modulating signal. This modulated signal is then transmitted by the
transmitter .Modulation is the process of changing the parameters of the
carrier signal, in accordance with the instantaneous values of the
modulating signal.

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Chapter 1: Amplitude Modulation

1.1.1 Need for Modulation

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.

1.1.2 Advantages of Modulation

1. Reduction in the height of antenna

2. Avoids mixing of signals
3. Increases the range of communication
4. Multiplexing is possible
5. Improves quality of reception

We will discuss each of these advantages in detail below.

1. Reduction in the height of antenna

For the transmission of radio signals, the antenna height must be multiple
of λ/4, where λ is the wavelength.

λ = c /f

Where c: is the velocity of light and f: is the frequency of the signal to be

transmitted

The minimum antenna height required to transmit a baseband signal of f =

10 kHz is calculated as follows:

The antenna of this height is practically impossible to install. Now, let us

consider a modulated signal at f = 1 MHz. The minimum antenna height is
given by,

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Chapter 1: Amplitude Modulation

This antenna can be easily installed practically. Thus, modulation reduces

the height of the antenna.

2. Avoids mixing of signals

If the baseband sound signals are transmitted without using the modulation
by more than one transmitter, then all the signals will be in the same
frequency range i.e. 0 to 20 kHz. Therefore, all the signals get mixed
together and a receiver cannot separate them from each other.

Hence, if each baseband sound signal is used to modulate a different

carrier then they will occupy different slots in the frequency domain
(different channels). Thus, modulation avoids mixing of signals.

3. Increase the range of communication

The frequency of baseband signal is low, and the low frequency signals
cannot travel long distance when they are transmitted. They get heavily
attenuated. The attenuation reduces with increase in frequency of the
transmitted signal, and they travel longer distance. The modulation process
increases the frequency of the signal to be transmitted. Therefore, it
increases the range of communication.

4. Multiplexing is possible

Multiplexing is a process in which two or more signals can be transmitted

over the same communication channel simultaneously. This is possible
only with modulation.

The multiplexing allows the same channel to be used by many signals.

Hence, many TV channels can use the same frequency range, without
getting mixed with each other or different frequency signals can be
transmitted at the same time.

5. Improves quality of reception

With frequency modulation (FM) and the digital communication

techniques such as PCM, the effect of noise is reduced to a great extent.
This improves quality of reception.

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Chapter 1: Amplitude Modulation

1.1.3 Types of Modulation

There are many types of modulations. Depending upon the modulation

techniques used, they are classified as shown in the following figure.

Figure 1.2: Block diagram of modulation types

The types of modulations are broadly classified into continuous-wave

modulation and pulse modulation.

1.1.3.1 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.

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Chapter 1: Amplitude 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.

 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.

1.1.3.2 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

1.2 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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Chapter 1: Amplitude Modulation

Figure 1.3: Amplitude modulation wave forms

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 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 Expression for AM wave

Following are the mathematical expressions for these waves.

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Chapter 1: Amplitude Modulation

Time-domain Representation of the AM Waves

Let the modulating signal be, m (t) = Am cos(2πfmt) and the carrier signal be,
c (t) = Ac cos(2πfct)

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+Amcos(2πfmt)]cos(2πfct) Equation 1

Modulation index (μ): The ratio of the message amplitude (Am) of the
message signal into the carrier amplitude (Ac) carrier signal is called
Modulation Index or Modulation Depth

Using AM Definition of AM we can get Am = μ Ac and μ =ka Am then

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. We
know that from definition of AM, Modulation index (μ) is equal to the
ratio of Am and Ac. Mathematically, we can write it as μ=Am / Ac

Rearrange the Equation 1 as Using AM Definition of AM we can get Am

= μ Ac and μ =ka Am then

S(t) = [Ac+ μ Ac cos(2πfmt)] cos(2πfct) Equation 2

S(t) = [Ac+ ka Am Ac cos(2πfmt)] cos(2πfct)

S(t) = [Ac+ ka Ac Am cos(2πfmt)] cos(2πfct)

Substitute m(t) in above equation the we get

S(t) = [Ac+ ka Ac m(t)] cos(2πfct)

S(t) = Ac [1+ ka m(t)] cos(2πfct) Equation 3

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Chapter 1: Amplitude Modulation

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.

1.2.1 Modulation Index (μ) from wave form

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

We will get the maximum amplitude of the modulated wave,

Figure 1.4: AM Modulation index graphical representation

⇒Amax = Ac+Am Equation 4

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

⇒Amin = Ac−Am Equation 5

Add Equation 4 and Equation 5.

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

Equation 6

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Chapter 1: Amplitude Modulation

Subtract Equation 5 from Equation 4.

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

Equation 7

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

⸫ Modulation Index ( μ) = Am / Ac Equation 8

= [(Amax−Amin)/2)] / [(Amax+Amin)/2],
Therefore

Equation 9

Therefore, Equation 3 and Equation 9 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. For a perfect modulation, the value of modulation index
should be 1, which implies the percentage of modulation should be 100%.

Figure 1.5: AM Modulation index for perfect modulation graphical

representation

For Under modulation, 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.

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Chapter 1: Amplitude Modulation

Figure 1.6: AM Modulation index for under modulation graphical

representation

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.

Figure 1.7: AM Modulation index for over modulation graphical

representation

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.

1.2.2 Frequency Spectrum and Bandwidth

The modulated carrier has new signals at different frequencies, called side
frequencies or sidebands. They occur above and below the carrier
frequency.

f USB = f C + fm
f LSB = f C - fm

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Chapter 1: Amplitude Modulation

Here f C is Carrier frequency, fm is modulating frequency,

fUSB is upper sideband frequency and fLSB is lower sideband

frequency

Consider the expression of AM wave given by equation 3

S(t) = Ac [1+ ka m(t)] cos(2πfct)

S(t) = Ac [1+ ka Am cos(2πfmt)] cos(2πfct) since μ =kaAm
S(t) = Ac [1+ μ cos(2πfmt)] cos(2πfct)
S(t) = Ac cos(2πfct) + μ Ac cos(2πfmt)] cos(2πfct)
S(t) = Ac cos(2πfct) + μ Ac cos(2πfmt)] cos(2πfct)
Using cos (A) cos(B) relation we get I/2[cos(A+B) + cos (A-B) ]
Then S(t) = Ac cos(2πfct) + μ Ac /2[cos(fc - fm) + cos ( fc + fm) / 2]
Therefore S(t) = Ac cos(2πfct) + μ Ac /2cos f LSB + μ Ac /2 cos f USB]

Figure 1.8: Frequency Spectrum of the AM Wave

This contains full carrier and both the sidebands, hence it is also called
Double Sideband Full Carrier (DSBFC) system.

We will be discussing this system, its modulation circuits and transmitters

next, in this section. We know that bandwidth of the signal can be
obtained by taking the difference between highest and lowest frequencies.
From above figure we can obtain bandwidth of AM wave as,

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Chapter 1: Amplitude Modulation

= (fc + fm) - (fc - fm)

= fc + fm -fc + fm

⸫ Bandwidth = 2 fm

Thus, it is clear that the bandwidth of the amplitude modulated wave is

twice the highest frequency present in the baseband or modulating signal

1.2.3 Power Calculations of AM Wave

Consider the following equation of amplitude modulated wave.

S (t) =Ac cos (2πfct) +μAc / 2cos [2π (fc+fm) t] +μAc / 2cos [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 + PUSB + PLSB ------------1

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

P = V2 / R= (V cos 450 )2 / R i.e. P = V2/ 2R (Since P = V rms / 2R)

Where, 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 Pc=Ac2 / 2R -----------2

Upper sideband power PUSB=(μAc/2) cos 450)2/ 2R i.e. PUSB=(μAc)2/8R

-----------3

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

PLSB=(μAc/2)cos 450)2 / 2R i.e. PUSB=(μAc)2/8R ----------4

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Chapter 1: Amplitude Modulation

⇒ Pt=Ac2 / 2R+ (μAc)2/8R + (μAc)2/8R ----------5

Pt =Ac2 / 2R [1+μ2/2] (since Pc=Ac2 / 2R) ----------6
Therefore, Pt = Pc [ 1 + μ2 / 2 ] ---------- 7

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=1μ=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

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 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)=10cos(2π×103t) is amplitude modulated with a

carrier signal c(t)=50cos(2π×105t). Find the modulation index, the carrier
power, and the power required for transmitting AM wave.

Solution

Given, the equation of modulating signal as m(t)=10cos(2π×103t)

We know the standard equation of modulating signal as
m(t)=Amcos(2πfmt)
By comparing the above two equations, we will get
Amplitude of modulating signal as Am=10voltsand Frequency of
modulating signal as fm=103Hz=1KHzGiven, the equation of carrier
signal is c(t)=50cos(2π×105t)
The standard equation of carrier signal is c(t)=Accos(2πfct)
By comparing these two equations, we will get Amplitude of carrier signal
as Ac=50volts

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Chapter 1: Amplitude Modulation

and Frequency of carrier signal as fc=105Hz=100KHz

We know the formula for modulation index as μ=Am / Ac
Substitute, Am and Ac values in the above formula. μ=0.2
Therefore, the value of modulation index is 0.2 and percentage of
modulation is 20%.
The formula for Carrier power, Pc= is Pc=Ac2 / 2R
Assume R=1ΩR=1Ω and substitute Ac value in the above formula.
Pc = (50)2 / 2(1) =1250W
Therefore, the Carrier power, Pc is 1250 watts.
We know the formula for power required for transmitting AM wave is
Pt = Pc [1+μ2 /2]
Substitute Pc and μ values in the above formula.
Pt=1250(1+(0.2)2 / 2) =1275W
Therefore, the power required for transmitting AM wave is 1275 watts.

Problem 2

The equation of amplitude wave is given by

3 5
s(t)=20[1+0.8cos(2π×10 t)]cos(4π×10 t). Find the carrier power, the total
sideband power, and the band width of AM wave.

Solution

Given, the equation of Amplitude modulated wave is

s(t) = 20[1+0.8cos(2π×103t)] cos(2π×2×105t)
Re-write the above equation as
s(t) = 20[1+0.8cos(2π×103t)] cos(2π×2×105t)
We know the equation of Amplitude modulated wave is
s(t)=Ac[1+μcos(2πfmt)]cos(2πfct)
By comparing the above two equations, we will get
Amplitude of carrier signal as Ac = 20volts

Modulation index as μ = 0.8

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Chapter 1: Amplitude Modulation

Frequency of modulating signal as fm=103Hz or 1KHz

Frequency of carrier signal as fc =2×105Hz = 200KHz

The formula for Carrier power, Pc is

Pc = Ac2 / 2R

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

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

Therefore, the Carrier power, Pc is 200watts.

We know the formula for total side band power is PSB = Pc(μ2) / 2
Substitute Pc and μ values in the above formula.PSB = 200×(0.8)2/2 = 64W

Therefore, the total side band power is 64 watts.

We know the formula for bandwidth of AM wave is BW=2fmBW=2fm
Substitute fm value in the above formula.BW=2(1K)=2 KHz
Therefore, the bandwidth of AM wave is 2 KHz.

1.2.4 AM Modulators

In this chapter, let us discuss about the modulators, which generate

amplitude modulated wave. The following two modulators generate AM
wave.

1) Square law modulator 2) Switching modulator

1.2.4.1 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 Acos(2πfct)
respectively. These two signals are applied as inputs to the summer
(adder) block.

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Chapter 1: Amplitude Modulation

Figure 1.9: Generation of AM Square law Modulator

This summer block produces an output, which is the addition of the

modulating and the carrier signal. Mathematically, we can write it as

V1t = m(t)+Ac cos(2πfct)

This signal V1t is applied as an input to a nonlinear device like diode. The
characteristics of the diode are closely related to square law.

V2t = k1V1(t)+k2V12(t) (Equation 1)

Where, k1 and k2 are constants. Substitute V1(t) in Equation 1

V2(t) = k1[m(t)+Ac cos(2πfct)]+k2[m(t)+Ac cos(2πfct)]2

V2(t) = k1m(t)+k1Accos(2πfct)+k2m2(t)+k2A2ccos2(2πfct)+2k2m(t)Accos(2πfct)+2k2m(t)

V2(t) = k1m(t)+k2m2(t)+k2A2ccos2(2πfct)+k1Ac[1+(2k2k1)m(t)]cos(2πfct)

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

s(t)=k1Ac[1+(2k2/k1) m(t)] cos(2πfct)⇒s(t) = 2(k2/k1)Ac m(t)] cos(2πfct)

The standard equation of AM wave is s(t)=Ac[1+kam(t)]cos(2πfct)

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Chapter 1: Amplitude Modulation

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 k1 and the amplitude sensitivity ka as 2k2 / k1.

1.2.4.2 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. Following is the block diagram of switching
modulator.

Figure 1.10: Generation of AM switching modulator

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

c(t)=Accos(2πfct) 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
V1(t)=m(t)+c(t)=m(t)+Ac cos(2πfct)

These 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)c(t).

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Chapter 1: Amplitude Modulation

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

V2(t) ={V1(t)if c(t)>0 and 0 if c(t)<0
We can approximate this as V2(t)=V1(t)x(t) (Equation 2)
Where, x(t) is a periodic pulse train with time period T=1fc
The Fourier series representation of this periodic pulse train is
x(t)=1 / 2+2 / π ∑n=1∞[(−1)n−1/2n−1]cos(2π(2n−1)fc(t)
⇒x(t)=1 / 2+2πcos(2πfct)−2 / 3πcos(6πfct)+....
Substitute, V1(t) and x(t) values in Equation 2.

V2(t)=[m(t)+Ac cos(2πfct)][12+2πcos(2πfct)−23πcos(6πfct)+.....]

V2(t)=m(t)2+Ac2cos(2πfct)+2m(t)πcos(2πfct)+2Acπcos2(2πfct)−2m(t)
3πcos(6πfct)−2Ac3πcos(2πfct)cos(6πfct)+..

V2(t)=Ac2(1+(4πAc)m(t))cos(2πfct)+m(t)2+2Acπcos2(2πfct)−2m(t)3π
cos(6πfct)−2Ac3πcos(2πfct)cos(6πfct)+....

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

s(t)=Ac2(1+(4πAc) m(t)) cos(2πfct)

We know the standard equation of AM wave is s(t)=Ac [1+Kam(t)]

cos(2πfct)

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 4πAc

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Chapter 1: Amplitude Modulation

1.2.5 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.

1. Square Law Demodulator 2. Envelope Detector

Figure 1.11: Detection of AM Wave or Square law Demodulator

1.2.5.1 Square Law Demodulator

Square law demodulator is used to demodulate low level AM wave. The

above block diagram of the square law demodulator.

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+kam(t)] cos(2πfct)

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

V2(t)=k1V1(t)+k2V21(t) (Equation 1)

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

V2(t)V2(t) is the output of the square law device k1 and k2 are constants
Substitute V1(t)V1(t) in Equation 1

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Chapter 1: Amplitude Modulation

V2(t)=k1(Ac[1+kam(t)] cos(2πfct)) +k2(Ac[1+kam(t)] cos(2πfct))2

⇒V2(t)=k1Accos(2πfct)+k1Ackam(t)cos(2πfct)
+k2Ac2[1+Ka2m2(t)+2kam(t)](1+cos(4πfct)2)

⇒V2(t)=k1Accos(2πfct)+k1Ackam(t)cos(2πfct)+K2Ac22+K2Ac22cos(4π
fct)+k2Ac2ka2m2(t)2+k2Ac2ka2m2(t)2cos(4πfct)+k2Ac2kam(t)+k2Ac2k
am(t)cos(4πfct)

In the above equation, the term k2Ac2kam(t) is the scaled version of the
message signal. It can be extracted by passing the above signal through a
low pass filter and the DC component k2Ac22can be eliminated with the
help of a coupling capacitor.

1.2.5.2 Envelope Detector

Envelope detector is used to detect (demodulate) high level AM wave.

Following is the block diagram of the envelope detector.

Figure 1.12: Detection of AM wave or envelop 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)s(t) is applied as an input to this detector. We
know the standard form of AM wave is

s(t)=Ac[1+kam(t)]cos(2πfct)

22
Chapter 1: Amplitude Modulation

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.

1.3 DSB-SC 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 as shown in the above figure.

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

power is being wasted in the carrier, which carries no information.

23
Chapter 1: Amplitude Modulation

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 above
figure.

1.3.1 Mathematical Expressions for DSB-SC

Let us consider the same mathematical expressions for modulating and

carrier signals as we have considered in the earlier chapters.

i.e., Modulating signal m(t) = Am cos(2πfmt)

Carrier signal c(t) = Ac cos(2πfct)

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

product of modulating signal and carrier signals.
s(t) = m(t)c(t)
s(t) = Am Ac cos(2πfmt)cos(2πfct)

1.3.2 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 Accos(2πfmt)cos(2πfct)
⇒s(t)=AmAc2cos[2π(fc+fm)t]+AmAc2cos[2π(fc−fm)t]

24
Chapter 1: Amplitude Modulation

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.

1.3.3 Power Calculations of DSB-SC Wave

Consider the following equation of DSBSC modulated wave.

s(t) = AmAc2cos[2π(fc+fm)t]+AmAc2cos[2π(fc−fm)t]

Power of DSBSC wave is equal to the sum of powers of upper sideband

and lower sideband frequency components.
P t= PUSB+PLSB

We know the standard formula for power of carrier signal is Pc=Ac2 / 2R

First, let us find the powers of upper sideband and lower sideband one by
one. Upper sideband power
PUSB = Am2 Ac2 / 8R

Similarly, we will get the lower sideband power same as that of upper
sideband power.
PLSB=Am2 Ac2 / 8R

Now, let us add these two sideband powers in order to get the power of
DSBSC wave.
Pt = 2(Am2 Ac2) / 8R
⸫ Pt = Am2 Ac2) / 4R

Therefore, the power required for transmitting DSBSC wave is equal to

the power of both the sidebands.

25
Chapter 1: Amplitude Modulation

1.3.4 DSB-SC 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

1.3.4.1 Balanced Modulator

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.

Figure 1.13: Block diagram of the balanced modulator

The same carrier signal c(t)=Accos(2πfct) 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 path AM modulator is

S1(t) = Ac[1+kam(t)] cos(2πfct)
Output of the lower path AM modulator is
S2(t) = Ac[1−kam(t) cos(2πfct)

26
Chapter 1: Amplitude Modulation

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+kam(t)]cos(2πfct)−Ac[1−kam(t)]cos(2πfct)
⇒s(t)=Accos(2πfct)+Ackam(t)cos(2πfct)−Accos(2πfct)+Ackam(t)cos(2πfct)
⸫ s (t) = 2Ackam(t)cos(2πfct)

We know the standard equation of DSBSC wave is

s(t) = Ac cos(2πfct) m(t)

By comparing the output of summer block with the standard equation of

DSBSC wave, we will get the scaling factor as 2ka

1.3.4.2 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)m(t) is
applied to the input transformer. Whereas, the carrier signals c(t)c(t) is
applied between the two center tapped transformers

Figure 1.14: Circuit diagram of the ring modulator

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.

27
Chapter 1: Amplitude Modulation

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 signal is multiplied by -1. This results in 1800 phase
shift in the resulting DSBSC 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

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

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

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

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.

1.3.5 DSB-SC 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

1.3.5.1 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.

28
Chapter 1: Amplitude Modulation

Let the DSB-SC wave be

s(t) = Accos(2πfct)m(t)

The output of the local oscillator is c(t) = Accos(2πfct+ϕ)

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

Figure 1.15: Block diagram of the coherent detector

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

V(t) = S(t)XC(t)

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

V(t)=Accos(2πfct)m(t) X Accos(2πfct+ϕ)
=Ac2cos(2πfct)cos(2πfct+ϕ)m(t)
=Ac2 / 2[cos(4πfct+ϕ)+cosϕ]m(t)
V(t)=Ac2 / 2[ cosϕm(t)]+Ac2 / 2[cos(4πfct+ϕ)m(t)]

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

V0t=Ac2/2 cosϕm(t)

The demodulated signal amplitude will be maximum, when ϕ=00 That’s

why the local oscillator signal and the carrier signal should be in phase,

29
Chapter 1: Amplitude Modulation

i.e., there should not be any phase difference between these two signals.
The demodulated signal amplitude will be zero, when ϕ=±900. This effect
is called as quadrature null effect.

1.3.5.2 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.

Figure 1.16: Block diagram of the COSTAS loop for DSB-SC

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 −900 phase shift to
one of the product modulator as shown in figure.

We know that the equation of DSB-SC wave is

S(t) = Accos(2πfct)m(t)

Let the output of VCO be

C1(t) = cos(2πfct+ϕ)

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

30
Chapter 1: Amplitude Modulation

Hence, the output of the upper product modulator is

V1(t) = S(t)c1(t)

Substitute, s(t)s(t) and c1(t)c1(t) values in the above equation.

V1(t) = Accos(2πfct)m(t)cos(2πfct+ϕ)
After simplifying, we will get
v1(t) asv1(t)=Ac2cosϕm(t)+Ac2cos(4πfct+ϕ)m(t)

This signal is applied as an input of the upper low pass filter. The output
of this low pass filter is
V01(t)=Ac2cosϕm(t)

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

The output of −900 phase shifter is

c2(t)=cos(2πfct+ϕ−900) =sin(2πfct+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)c2(t)

Substitute, s(t) and c2(t) values in the above equation.

⇒V2(t)=Accos(2πfct)m(t)sin(2πfct+ϕ)

After simplifying, we will get

V2(t)=c2sinϕm(t)+Ac2sin(4πfct+ϕ)m(t)

This signal is applied as an input of the lower low pass filter. The output
of this low pass filter is
V02(t)=Ac2sinϕm(t)

The output of this Low pass filter has −900 phase difference with the
output of the upper low pass filter.

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Chapter 1: Amplitude Modulation

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.

1.4 SSB-SC 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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Chapter 1: Amplitude Modulation

1.4.1 Mathematical Expressions for SSB-SC

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(2πfmt)
Carrier signal c(t)=Ac cos(2πfct)
Mathematically, we can represent the equation of SSBSC wave as
S (t) = AmAc2cos[2π(fc+fm)t]for the upper sideband or
S (t) = AmAc2cos[2π(fc−fm)t]for the lower sideband

1.4.2 Bandwidth of SSB-SC 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.
i.e., Bandwidth of SSBSC modulated wave = 2fm / 2 = fm

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

to the frequency of the modulating signal.

1.4.3 Power Calculations of SSB-SC Wave

Consider the following equation of SSBSC modulated wave.

S (t) = AmAc / 2 Cos [2π (fc+fm) t] for the upper sideband
Or
S (t) = AmAc / 2 Cos [2π (fc-fm) t] for the lower sideband

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

frequency components.
PSSB = PUSB = PLSB

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

Upper sideband power PUSB=(μAc/2) cos 450)2/ 2R
i.e. PUSB=(μAc)2/8R

33
Chapter 1: Amplitude Modulation

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

PLSB = (μAc/2)cos 450)2 / R [ Since cos 450 = (1 / )]

PUSB = (μAc)2/8R

Therefore, the power of SSBSC wave is

PSSB=PUSB=PLSB=μ2Ac2 / 8R

1.4.4 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.

1.4.5 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 excellent frequency stability.

1.4.6 Applications

 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.

34
Chapter 1: Amplitude Modulation

1.4.7 SSB-SC 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

1.4.7.1 Frequency Discrimination Method

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

Figure 1.17: Block diagram of the frequency discrimination method for

SSB-SC signal

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.

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.

1.4.7.2 Phase Discrimination Method

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

35
Chapter 1: Amplitude Modulation

This block diagram consists of two product modulators, two −900 phase
shifters, one local oscillator and one summer block. The product
modulator produces an output, which is the product of two inputs.
The −900 phase shifter produces an output, which has a phase lag
of −900 with respect to the input.

Figure 1.18: Block diagram of the phase discrimination method for

SSB-SC signal

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 Amcos(2πfmt) and the carrier signal Accos (2πfct)
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(2πfmt)cos(2πfct)
⇒ S1(t) =AmAc / 2{cos[2π(fc+fm)t]+cos[2π(fc−fm)t]}

The modulating signal Amcos(2πfmt) and the carrier signal Accos(2πfct)

are phase shifted by −900 before applying as inputs to the lower product
modulator. So, the lower product modulator produces an output, which is
the product of these two inputs.

36
Chapter 1: Amplitude Modulation

The output of lower product modulator is

S2 (t) = Am Ac cos(2πfmt−900) cos(2πfct−900)
S2 (t) = Am Ac sin(2πfmt)sin(2πfct)

S2 (t) = AmAc2{cos[2π(fc−fm)t]−cos[2π(fc+fm)t]}

Add S1(t) and S2 (t)in order to get the SSBSC modulated wave s(t) having
a lower sideband.

S(t)=AmAc / 2{cos[2π(fc+fm)t]+cos[2π(fc−fm)t]}+
AmAc / 2{cos[2π(fc−fm)t]−cos[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)s(t) having a upper sideband.

S(t)=AmAc2{cos[2π(fc+fm)t]+cos[2π(fc−fm)t]}−AmAc2{cos[2π(fc−fm)t
]−cos[2π(fc+fm)t]}

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

1.4.8 SSB-SC 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.

1.4.8.1 Coherent Detector

Here, the same carrier signal (which is used for generating SSBSC wave)
is used to detect the message signal.

37
Chapter 1: Amplitude Modulation

Figure 1.19: Block diagram of the coherent detector for SSB-SC 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.

s(t)=AmAc2cos[2π(fc−fm)t]
The output of the local oscillator is c(t)=Ac cos(2πfct)

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)=AmAc / 2{cos[2π(fc−fm)t]}Accos(2πfct)
=AmAc2/2cos[2π(fc−fm)t]cos(2πfct)

=AmAc2/ 4cos[2π(fc−fm)t]cos(2πfct)

=AmAc2 / 4{cos[2π(2fc−fm)]+cos(2πfm)t}-
AmAc2/4[sin(2πfmt)]+AmAc2 / 4 [sin[2π(2fc−fm)t]

38
Chapter 1: Amplitude Modulation

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 isv0(t)=AmAc2/4cos(2πfmt)

Here, the scaling factor is Ac2/4.

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

Consider the following SSBSC wave having an upper sideband.

s(t)=AmAc / 2cos[2π(fc+fm)t]
The output of the local oscillator is c(t)=Ac cos(2πfct)
We can write the output of the product modulator as v(t)=s(t)c(t))
Substitute s(t)s(t) and c(t)c(t) values in the above equation.

v(t) = AmAc / 2{cos[2π(fc+fm)t]} Accos(2πfct)

v(t) = AmAc2 / 2{cos[2π(fc+fm)t]} Accos(2πfct)

= AmAc2 / 4{cos[2π(fc+fm)t]cos(2πfct) –{sin[2π(2fc+fm)t] sin(2πfmt)}

v(t)=AmAc2 / 4 [cos(2πfmt)]+AmAc2 / 4cos[2π(2fc+fm)t]

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

Vo (t)=AmAc2 / 4[cos(2πfmt)]

Here too the scaling factor is Ac2 / 4.

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

39
Chapter 1: Amplitude Modulation

1.5 VSBSC Modulation

In the previous chapters, we have discussed SSB-SC 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.

Figure 1.20: VSB modulation spectrum

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.

40
Chapter 1: Amplitude Modulation

1.5.1 Bandwidth of VSBSC Modulation

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. Therefore The Bandwidth of VSBSC Modulated
Wave = fm+fv

1.5.2 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.

1.5.3 Disadvantages

Following are the disadvantages of VSBSC modulation.

 Bandwidth is more when compared to SSBSC wave.

 Demodulation is complex.

1.5.4 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.

1.5.5 Generation of VSB-SC

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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Chapter 1: Amplitude Modulation

Figure 1.21: Block diagram for generation of VSB-SC Signal

In this method, first we will generate DSB-SC 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)m(t) and carrier signal Accos(2πfct) 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(2πfct) m(t)

Apply Fourier transform on both sides P(f)=Ac2[M(f−fc) +M(f+fc)]
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(f)=P(f)H(f)
Substitute P(f)P(f) value in the above equation. S(f)=Ac / 2
[M(f−fc)+M(f+fc)]H(f)
The above equation represents the equation of VSBSC frequency
spectrum.

1.5.6 Detection 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

42
Chapter 1: Amplitude Modulation

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.

Figure 1.22: Block diagram of detection of VSB-SC signal

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)s(t) and the carrier signal is Accos(2πfct).
From the figure, we can write the output of the product modulator as
v(t)=Accos(2πfct)s(t)
Apply Fourier transform on both sides
V(f)=Ac / 2[S(f−fc)+S(f+fc)]
We know that S(f)=Ac2[M(f−fc) +M(f+fc)]H(f)

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

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

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

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

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

43
Chapter 1: Amplitude Modulation

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

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

V(f)=Ac / 24M(f)[H(f−fc)+H(f+fc)]+Ac / 24[M(f−2fc)H(f−fc)+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.

V0(f)=Ac / 24M(f)[H(f−fc)+H(f+fc)]

Short Answer Questions

1. Write the expression for AM modulated wave.

2. What are the methods for generating and detecting methods for
AM?
3. Define modulation index and calculate the bandwidth of AM.
4. Write any 4 differences and applications of AM DSB-SC, SSB-SC,
and VSB.
5. What is threshold effect in envelope detector?

Long Answer Questions

1. Define modulation? And explain the need of modulation.

2. A carrier amplitude modulated to a depth of 50% by a sinusoidal,
produces side band frequencies of 5.005MHZ and 4.995mhz.the
amplitude of each side frequency is 40v.Find the frequency and
amplitude of carrier signal.
3. Derive the express for AM signal with time and frequency domain
description.
4. Calculate the power and bandwidth in AM signal.
5. Explain the generation of AM signal with switching modulator.
6. Discuss the any detection of DSB-SC modulated wave with the
help of detector block diagram and explain.
7. Draw the block diagram for the generation and detection of a VSB
signal and explain the principle of operation.
8. Discuss various methods used to generate SSB signals with neat
sketches.

44
Chapter 1: Amplitude Modulation

9. Draw the block diagram for SSB detection using any detection
method and explain its operation.
10. Draw the block diagram of AM detector using envelops detector
method and explain its operation.
11. Compare the AM, DSB-SC, SSB-SC, VSB.

45
Chapter 2
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.
 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.

2.1 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.

46
Chapter 2: Angle Modulation

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.

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.

Figure 2.1: Frequency Modulation Signal representation

2.1.1 Mathematical Representation for FM Wave

The equation for instantaneous frequency variation or deviation is

proportional to message signal, it can be written as a definition of
Frequency Modulation as (fi –fc) m(t)

fi - fc = kfm (t) ------------------- 1

Where, fc is the carrier frequency, kf is the frequency sensitivity, m (t) is

the message signal

47
Chapter 2: Angle Modulation

We know that Angle modulation is S (t) = Ac Cos [θi (t)] -------------- 2

Where θi (t) is denoted by θi (t) = 2πfc +

S (t) = Ac Cos [2πfc + ] ----------------- 3

A complete oscillations occurs whenever θi (t) is changes by 2 radians

θi (t) = 2πfct +

Apply the derivative on both sides with respect to ‘t ‘ then above

equation becomes
d/dt (θi (t)) = 2πfc + d/dt (

We know that d/dt (θi (t)) is a angular velocity i.e. d/dt (θi (t)) = Wi (t),
then
Wi (t) = 2πfc + d/dt ( Since Wi (t) = 2πfi(t)
2πfi (t) = 2πfc + d/dt ( then divide by 2π we get
fi (t) = fc + d/dt ( (1/ 2π)
Therefore fi (t) - fc = d/dt ( (1/ 2π) ---------------- 4
Now equate 1 and 4 Equations we get
d/dt ( (1/ 2π) = Kf m (t) ----------------5
d/dt ( = 2π Kf m (t) ----------------6
Integrating on both sides the we get
= 2πkf ∫m (t) dt
Substitute, value in the above equation 3 then we get
S (t) = Ac Cos [(2πfct+2πkf ∫ m (t) dt)] --------- 7
This is the Standard FM wave equation.

2.1.2 Single Tone Frequency Modulation Signal

If the modulating signal is m (t) = Am Cos (2πfmt), i.e. for single tone

s(t)=Ac cos[(2πfct+2πkf ∫Am Cos2πfm t )dt)] --------- 8

48
Chapter 2: Angle Modulation

s(t)=Ac cos[(2πfct+2πkf Am (Sin 2πfm t) / 2πfm)] ---- 9

s(t)=Ac cos[(2πfct+kf Am (Sin 2πfm t) / fm)] --------- 10
we know that (Kf Am) = and / fm =
Substitute the above values in equation 10 then finally we get for
single tone FM signal
S (t) = Ac cos (2πfct+βsin (2πfmt))
Where, β = modulation index =Δf / fm and β =kfAm / 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.
 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.

49
Chapter 2: Angle Modulation

2.2 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.

Figure 2.2: Message signal

Figure 2.3: Carrier signal

Figure 2.4: FM Signal

50
Chapter 2: Angle Modulation

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.

2.2.1 Mathematical Representation for PM Wave

The equation for instantaneous phase ϕi in phase modulation is ϕi=kpm(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(2πfct+ϕi)
Substitute, ϕi value in the above equation. s(t)=Ac cos(2πfct+kpm(t))
This is the equation of PM wave.
If the modulating signal, m (t) = Am Cos (2πfmt), then the equation of PM
wave will be
S (t) = Ac Cos(2πfct+βcos(2πfmt))
Where, β = modulation index = Δϕ=kpAm, Δϕ is phase deviation
Phase modulation is used in mobile communication systems, while
frequency modulation is used mainly for FM broadcasting.

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.

Generation of NBFM

We know that the standard equation of FM wave is

s(t) = Ac cos(2πfct+2πkf∫m(t)dt)
⇒s(t) = Ac cos(2πfct) cos(2πkf∫m(t)dt) – Ac sin(2πfct) sin(2πkf∫m(t)dt)
For NBFM, The modulation index β is small, i.e., less than 1.
We know that cosθ≈1 and sinθ≈1 when θ is very small.
By using the above relations, we will get the NBFM equation as
s(t)=Ac cos(2πfct)−Ac sin(2πfct)2πkf∫m(t)dt

51
Chapter 2: Angle Modulation

2.3 Narrow Band FM Modulator

Here, the integrator is used to integrate the modulating signal m (t). The
carrier signal Ac Cos (2πfct) is the phase shifted by −900 to get Ac
sin(2πfct) with the help of −900 phase shifter. The product modulator has
two inputs ∫m(t)dt and Ac sin(2πfct). It produces an output, which is the
product of these two inputs.

Figure 2.5: Block diagram of narrow band frequency modulation

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

Here, the integrator is used to integrate the modulating signal m(t)m(t).

The carrier signal Ac Cos (2πfct) is the phase shifted by −900 to get Ac
sin(2πfct) with the help of −900 phase shifter. The product modulator has
two inputs ∫m(t)dt and Ac sin(2πfct). It produces an output, which is the
product of these two inputs.

This is further multiplied with 2πkf by placing a block 2πkf 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.

2.4 Bessel Functions and Properties

Bessel functions are used to solve in 3D or 2D the wave equation at a

given (harmonic) frequency.

52
Chapter 2: Angle Modulation

The solution is generally a sum of spherical Bessel functions that gives the
acoustic (which deals with sound) pressure at a given location of the 3D or
2D space.

Figure 2.6: Bessel function graph for frequency modulation signal

Elative sideband amplitude

Modulation index 0 1 2 3 4 5
0.00 1.00
0.25 0.98 0.12
0.5 0.94 0.24 0.03
1.0 0.77 0.44 0.11 0.02
2.0 0.22 0.58 0.35 0.13 0.03
2.41 0.00 0.52 0.43 0.20 0.06 0.02
Table2.1: Bessel function graph values for Frequency Modulation Signal

2.5 Wide Band Frequency Modulation Equation

Frequency modulation uses the instantaneous amplitude of a modulating

signal (voice, music, data, etc.) to directly vary the frequency of a carrier
signal.

53
Chapter 2: Angle Modulation

Modulation index, β, is used to describe the ratio of maximum frequency

deviation of the carrier to the maximum frequency deviation of the
modulating signal. The concept was pioneered by Edwin H. Armstrong in
the late 1920s and patented in the early 1930s.

Depending on the modulation index chosen, the carrier and certain

sideband frequencies may actually be suppressed. Zero crossings of the
Bessel functions, Jn(β), occur where the corresponding sideband, n,
disappears for a given modulation index, β.

Spectrum analysis of sinusoidal FM wave using Bessel Function

The composite spectrum for a single tone consists of lines at the carrier
and upper and lower sidebands (of opposite phase), with amplitudes
determined by the Bessel function values at those frequencies.

In this method β value is greater than 1 (β ≥ 1)

S(t) = Ac Cos (2 fct + Sin 2 fmt) ---- 1 θ 2 fct + Sin 2 fmt

Cos = Ac Re ( e j ) ----- 2 Since e j = Cos + j Sin
Since Wc = 2 fc & Wm = 2 fm

In equation 4 let f (t) = e Sin Wm t, This FM wave is produced by a

sinusoidal modulating wave m(t) that is a periodic function of time ‘t’, its
only fc = nfm

We know that exponential Fourier series

Here we substitute f(t) value then we get from below equation 7

54
Chapter 2: Angle Modulation

We know that Cn value with limits

Now T = 1/fm then 1/T = fm and limits are -T/2 = - 1/ 2fm, T/2 = 1 /
2fm

We know that Bessel function is defined as

Assume Wm t = and dt = d / Wm then Limits are – to

Now the expression of Cn becomes

Now substitute Cn value in equation 7 then we get

55
Chapter 2: Angle Modulation

And once again substitute above expression in equation 4 then we get

Then finally Wide Band Frequency Modulation (WBFM) equation

2.6 Carson's Rule for Frequency Modulation Bandwidth

The bandwidth of an FM signal is not as straightforward to calculate as

that of an AM signal.

A very useful rule of thumb used by many engineers to determine the

bandwidth of an FM signal for radio broadcast and radio communications
systems is known as Carson's Rule. This rule states that 98% of the signal
power is contained within a bandwidth equal to the deviation frequency,
plus the modulation frequency doubled. Carson's Rule can be expressed
simply as a formula:

By using Carson’s rule

Band Width for FM = 2nfm here n =
Therefore BW for FM = 2( )fm
We know that
Then using above value BW = 2[( )+1] fm
Therefore Bandwidth for Frequency Modulation (BW) = 2( )

2.6.1 Power Calculation in FM signal

56
Chapter 2: Angle Modulation

From above equation we consider amplitude only for power calculation

Therefore The Power of the FM signal from above equation is

We know that for maximum Bessal function value is one i.e.

Now the Power of the FM Signal by using above equation

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=2KHz
Frequency sensitivity, kf=40Hz/volt
We know the formula for Frequency deviation as
Δf=kfAm
Substitute kf and Am values in the above formula.
Δf=40×5=200Hz
Therefore, frequency deviation, Δf is 200Hz

57
Chapter 2: Angle Modulation

The formula for modulation index is

β=Δf /fm
Substitute Δf and fmfm values in the above formula.
β=200 / 2000=0.1
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= 2×2 KHz
BW = 4 KHz
Therefore, the bandwidth of Narrow Band FM wave is 4 KHz.

Problem 2

An FM wave is given by s(t)=20cos(8π×106t+9sin(2π×103t)). Calculate

the frequency deviation, bandwidth, and power of FM wave.

Solution

Given, the equation of an FM wave as

s(t)=20cos(8π×106t+9sin(2π×103t))
We know the standard equation of an FM wave as
s(t)=Accos(2πfct+βsin(2πfmt))
We will get the following values by comparing the above two equations.
Amplitude of the carrier signal, Ac=20V
Frequency of the carrier signal, fc=4×106 Hz or fc=4MHz
Frequency of the message signal, fm=1×103Hz or fm=1KHz
Modulation index, β=9

58
Chapter 2: Angle Modulation

Here, the value of modulation index is greater than one. Hence, it is Wide
Band FM.
We know the formula for modulation index as
β=Δf / fm
Rearrange the above equation as follows.
Δ=βfm
Substitute ββ and fm values in the above equation.
Δ = 9 × 1 KHz
Δ = 9 KHz
Therefore, frequency deviation, Δf is 9KHz.
The formula for Bandwidth of Wide Band FM wave is
BW=2(β+1) fm
Substitute ββ and fm values in the above formula.
BW=2(9+1)1KHz
BW = 20KHz
Therefore, the bandwidth of Wide Band FM wave is 20KHz
Formula for power of FM wave is
Pc = (Ac)2 / 2R
Assume, R=1Ω and substitute Ac value in the above equation.
P = (20)2/ 2(1)
P = 200 Watts
Therefore, the power of FM wave is 200 Watts.

2.7 Generation of WBFM

The following two methods generate WBFM wave.

 Direct method
 Indirect method

59
Chapter 2: Angle Modulation

2.7.1 Direct Method for FM Generation

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.

2.7.2 Hartley Oscillator

The Hartley oscillator is circuit in which the oscillation frequency is

determined by a tuned circuit consisting of capacitors and inductors, that
is, an LC oscillator. The circuit was invented in 1915 by American
engineer Ralph Hartley.

Figure 2.7: Hartley oscillator circuit for generation of direct FM signal

60
Chapter 2: Angle Modulation

The distinguishing feature of the Hartley oscillator is that the tuned circuit
consists of a single capacitor in parallel with two inductors in series (or a
single tapped inductor), and the feedback signal needed for oscillation is
taken from the center connection of the two inductors.

Oscillators are often characterized by the Frequency of their Output

Signal

 A low-frequency oscillator (LFO) is an electronic oscillator that

generates a frequency below approximately 20 Hz. This term is
typically used in the field of audio synthesizers, to distinguish it
from an audio frequency oscillator.
 An audio oscillator produces frequencies in the audio range, about
16 Hz to 20 kHz.[2]
 An RF oscillator produces signals in the radio frequency (RF) range
of about 100 kHz to 100 GHz.
From Hartley oscillator circuit

fi (t) = -------- 1

c(t) = Co + Cos 2 mt

where Co = the total capacitance in absence of modulation and

= Maximum change in capacitance value

Substitute c(t) value in above equation 1 then we get

fi (t) = -------- 2

fi (t) = -------- 3

fi (t) = -------- 4

fo =

61
Chapter 2: Angle Modulation

Now we can write the total frequency fo value in above equation 4

fi (t) = fo -------- 5

--------- 6

Now above equation 6 can be using binomial theorem then it becomes

--------- 7

We know that according to FM definition gives as fi (t) α m(t)

fi(t) = fo +kf m(t) -------- 8

Since m(t) = Am Cos 2 mt

fi (t) = fo(1 + kfAm Cos 2 mt) --------- 9

fi (t) = fo(1 + Cos 2 mt ) --------- 10

fi (t) = fo(1+ Cos 2 mt) --------- 11

Now equate above two equations’ 7 and 11 then we get =

fi (t) = fo+ Cos 2 mt -------- 12

Where, fi (t) is the instantaneous frequency of WBFM wave

This is the desired relation for instantaneous frequency of WBFM wave.

2.8 Indirect Method FM Generation

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.

62
Chapter 2: Angle Modulation

Figure 2.8: Block diagram for generation of NBFM Signal

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 ββ 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.

2.8.1 Explain the Indirect Method of FM Generation or Armstrong

FM Method

The direct methods cannot be used for the broadcast applications. Thus,
the alternative method i.e. indirect method called as the Armstrong method
of FM generation is used.

 In this method the FM is obtained through phase modulation. A

crystal oscillator can be used hence the frequency stability is very
high.

63
Chapter 2: Angle Modulation

The block diagram of the Armstrong method is shown below:

Figure 2.9: Block diagram for generation of WBFM Signal

Figure: Armstrong method (Indirect method of FM generation)

The system specifications are as follows:

The oscillator frequency of the narrow band phase modulator is 0.1 MHz
The frequency deviation of the NBFM signal is =
10 Hz

The mixer uses high – side tuning, the oscillator frequency of the mixer is
8.5 MHz

A low pass filter is used to get the mixer output

The final carrier frequency at the output is 100 MHz
The output frequency deviation = 75 KHz
What are the values of multipliers n1 and n2?

n = n1 n2 = =

At the mixer (It is used for frequency translation)

f 2 –( n 1 f1) =

n 2 f2 – n1 n2 f1 = fc
8.5 n2 – 7500 0.1 = 100

n1 n2 = 7500
n1 = 7500

64
Chapter 2: Angle Modulation

Operation

 The crystal oscillator generates the carrier at low frequency typically

at 1MHz. This is applied to the combining network and a 90° phase
shifter.
 The modulating signal is passed through an audio equalizer to boost
the low modulating frequencies. The modulating signal is then
applied to a balanced modulator.
 The balanced modulator produced two side bands such that their
resultant is 90° phase shifted with respect to the unmodulated
carrier.
 The unmodulated carrier and 90° phase shifted sidebands are added
in the combining network.
 At the output of the combining network we get FM wave. This wave
has a low carrier frequency fc and low value of the modulation
index mf .
 The carrier frequency and the modulation index are then raised by
passing the FM wave through the first group of multipliers. The
carrier frequency is then raised by using a mixer and then the fc and
mf both are raised to required high values using the second group of
multipliers.
 The FM signal with high fc and high mf is then passed through a
class C power amplifier to raise the power level of the FM signal.

2.9 Demodulation of Frequency Modulation Signal

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

 Frequency discrimination method

 Phase discrimination method

65
Chapter 2: Angle Modulation

2.9.1 Frequency Discrimination Method

In the above equation, the amplitude term resembles the envelope of AM

wave and the angle term resembles the angle of FM wave. So, in order to
get the modulating signal m(t), as per the requirement we can recover it
from the envelope of AM wave.

The following figure shows the block diagram of FM demodulator using

frequency discrimination method

As you see the above block diagram consists of the differentiator and the
envelope detector. Differentiator is used to convert the FM wave into a
combination of both AM wave and FM wave. This means, it converts the
frequency variations of FM wave into the corresponding voltage
(amplitude) variations of AM wave

Figure 2.10: Block diagram for detection of FM signal (Frequency

discrimination method)

66
Chapter 2: Angle Modulation

As per the operation of the envelope detector, it produces the demodulated

output of AM wave, which is called the modulating signal.

2.9.2 Simple FM Slope Detector

The circuit diagram of a simple slope detector is as shown in figure.

Figure 2.11: Simple slope detector

The output voltage of the tank circuit is then applied to a simple diode
detector of an RC load with proper time constant. This detector is identical
to the AM diode detector. Even though the slope detector circuit is simple
it has the following drawbacks.

Drawbacks of Slope Detector

i. It is inefficient.
ii. It is linear only over a limited frequency range.
iii. It is difficult to adjust as the primary and secondary winding of the
transformer must be tuned to slightly different frequencies.

Advantages of Slope Detector

The only advantages of the basic slope detector circuit are its simplicity.
To overcome the drawbacks of the simple slope detector, a balanced slope
detector is used.

2.9.3 Balanced FM Slope Detector (Balanced Frequency Discriminator)

The circuit diagram of the balanced slope detector is shown in Figure. 2.

67
Chapter 2: Angle Modulation

Figure 2.12: Balanced slope detector

As shown in the circuit diagram, the balanced slope detector consists of

two slope detector circuits.

The input transformer has a center tapped secondary. Hence, the input
voltages to the two slope detectors are 180° out of phase. There are three
tuned circuits.

Out of them, the primary is tuned to IF i.e., fc .

The upper tuned circuit of the secondary (T1) is tuned above fc by Δf i.e.,
its resonant frequency is (fc+ Δf).
The lower tuned circuit of the secondary is tuned below fc by Δf i.e., at
(fc – Δf).
R1C1 and R2C2 are the filters used to bypass the RF ripple.
Vo1 and Vo2 are the output voltages of the two slope detectors.
The final output voltage Vo is obtained by taking the subtraction of the
individual output voltages, Vo1 and Vo2, i.e.,

Working Operation of the Circuit

The circuit operation can be explained by dividing the input frequency

into three ranges as follows:

68
Chapter 2: Angle Modulation

Figure 2.13: Characteristics of the balanced slope detector

i. fin = fc: When the input frequency is instantaneously equal to fc, the
induced voltage in the T1 winding of secondary is exactly equal to that
induced in the winding T2.
Thus, the input voltages to both the diodes D1 and D2 will be the same.

Therefore, their dc output voltages Vo1 and Vo2 will also be identical but
they have opposite polarities. Hence, the net output voltage Vo = 0.

ii. (fc < fin < (fc + Δf): In this range of input frequency, the induced
voltage in the winding T1 is higher than that induced in T2.
Therefore, the input to D1 is higher than D2.
Hence, the positive output Vo1 of D1 is higher than the negative output
Vo2 of D2.
Therefore, the output voltage Vo is positive.
As the input frequency increases towards (fc + Δf), the positive output
voltage increases as shown in 3.

69
Chapter 2: Angle Modulation

If the output frequency goes outside the range of (fc – Δf) to (fc + Δf),
the output voltage will fall due to the reduction in tuned circuit
response.

Advantages
i. This circuit is more efficient than simple slope detector.
ii. It has better linearity than the simple slope detector.

Drawbacks

i. Even though linearity is good, it is not good enough.

ii. This circuit is difficult to tune since the three tuned circuits are to be
tuned at different frequencies i.e., fc, (fc+Δf) and (fc – Δf).
iii. Amplitude limiting is not provided.

2.9.4 Phase Discrimination Method

The following figure shows the block diagram of FM demodulator using

phase discrimination method.

Figure 2.14: Phase discrimination method for demodulation of FM signal

This block diagram consists of the multiplier, the low pass filter, and the
Voltage Controlled Oscillator (VCO). VCO generates an output signal
v(t), when the frequency is proportional to the input signal voltage d(t).
Initially, when the signal becomes d(t) zero, we need to adjust the VCO in
order to produce an output signal v(t), that has a carrier frequency and -
900 phase shift with respect to the carrier signal.

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Chapter 2: Angle Modulation

FM wave s(t) and the VCO output v(t) which is the resultant of frequency
proportional to the input signal voltage are applied as inputs of the
multiplier. With the high and low frequency components the multiplier
produces the resultant output. Low pass filter recognizes and eliminates
the high frequency component thus producing only the low frequency
component as its output.

 This low frequency component consists of only the term-related

phase difference. Hence, we get the required modulating signal m(t)
from this output of the low pass filter.

2.9.5 Phase Locked Loops (PLL)

Introduction to PLL

The concept of Phase Locked Loops (PLL) first emerged in the early
1930’s.But the technology was not developed as it now, the cost factor for
developing this technology was very high. Since the advancement in the
field of integrated circuits, PLL has become one of the main building
blocks in the electronics technology. In present, the PLL is available as a
single IC in the SE/NE560 series (560, 561, 562, 564, 565 and 567) to
further reduce the buying cost, the discrete IC’s are used to construct a
PLL.

Figure 2.15: Block diagram of Phase Locked Loops (PLL)

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Chapter 2: Angle Modulation

PLL Applications

 Frequency Modulation (FM) stereo decoders, FM Demodulation

networks for FM operation.
 Frequency synthesis that provides multiple of a reference signal
frequency.
 Used in motor speed controls, tracking filters.
 Used in frequency shift keying (FSK) decodes for demodulation
carrier frequencies.

PLL Block Diagram

The block diagram of a basic PLL is shown in the figure below. It is

basically a flip flop consisting of a phase detector, a low pass filter (LPF),
and a Voltage Controlled Oscillator (VCO).

The input signal Vi with an input frequency fi is passed through a phase

detector. A phase detector basically a comparator which compares the
input frequency fiwith the feedback frequency fo .The phase detector
provides an output error voltage Ver (=fi+fo), which is a DC voltage. This
DC voltage is then passed on to an LPF. The LPF removes the high
frequency noise and produces a steady DC level, Vf (=Fi-Fo). Vf also
represents the dynamic characteristics of the PLL.
The DC level is then passed on to a VCO. The output frequency of the
VCO (fo) is directly proportional to the input signal. Both the input
frequency and output frequency are compared and adjusted through
feedback loops until the output frequency equals the input frequency. Thus
the PLL works in these stages – free-running, capture and phase lock.
As the name suggests, the free running stage refer to the stage when there
is no input voltage applied. As soon as the input frequency is applied the
VCO starts to change and begin producing an output frequency for
comparison this stage is called the capture stage. The frequency
comparison stops as soon as the output frequency is adjusted to become
equal to the input frequency. This stage is called the phase locked state.

Now let us study in detail about the various parts of a PLL – The phase
detector, Low Pass Filter and Voltage Controlled Oscillator.

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Chapter 2: Angle Modulation

1. Phase detector
This comparator circuit compares the input frequency and the VCO output
frequency and produces a dc voltage that is proportional to the phase
difference between the two frequencies. The phase detector used in PLL
may be of analog or digital type

2. Low Pass Filter (LPF)

A Low Pass Filter (LPF) is used in Phase Locked Loops (PLL) to get rid
of the high frequency components in the output of the phase detector. It
also removes the high frequency noise.
All these features make the LPF a critical part in PLL and helps control
the dynamic characteristics of the whole circuit.
The dynamic characteristics include capture and lock ranges, bandwidth,
and transient response.
The lock range is the tracking range where the range of frequencies of the
PLL system follows the changes in the input frequency.

The capture range is the range in which the Phase Locker Loops attains
the Phase Lock.

When the filter bandwidth is reduced, the response time increases. But this
reduces the capture range. But it also helps in reducing noise and in
maintaining the locked loop through momentary losses of signal.

Two types of passive filter are used for the LPF circuit in a PLL. An
amplifier is used also with LPF to obtain gain. The active filter used in
PLL is shown below.

3. Voltage Controlled Oscillator (VCO)

The main function of the VCO is to generate an output frequency that is

directly proportional to the input voltage.

The connection diagram of a SE/NE 566 VCO is shown in the figure

below. The maximum frequency of the VCO is 500 KHz.

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Chapter 2: Angle Modulation

This VCO provides simultaneous square wave and triangular wave outputs
as a function of the input voltage. The frequency of oscillation is
determined by the resistor R and capacitor C along with the voltage Vc
applied to the control terminal.
2.10 Compare Amplitude Modulation and Frequency Modulation
Difference Between AM and FM
Amplitude Modulation (AM) Frequency Modulation (FM)
The first successful audio Developed in 1930 by Edwin
transmission was carried out in the Armstrong, in the United States
mid-1870s
The radio wave is called a carrier The radio wave is called a carrier
wave and the frequency and phase wave, but the amplitude and phase
remain the same remain the same
Has poor sound quality, but can Has higher bandwidth with better
transmit longer distance sound quality
The frequency range of AM radio The frequency range of FM is 88 to
varies from 535 to 1705 kHz 108 MHz in the higher spectrum
More susceptible to noise Less susceptible to noise
Table 2.2: Compare amplitude modulation and frequency modulation
2.11 Pre-emphasis and De-emphasis
As we already know that in FM, the noise has a greater effect on the
higher modulating frequencies. This effect can be reduced by increasing
the value of modulation index (mf) for higher modulating frequencies (fm).
This can be done by increasing the deviation Δf and Δf can be increased
by increasing the amplitude of modulating signal at higher modulating
frequencies.
Thus, if we boost the amplitude of higher frequency modulating signals
artificially then it will be possible to improve the noise immunity at higher
modulating frequencies.

The artificial boosting of higher modulating frequencies is called as

pre-emphasis.

Boosting of higher frequency modulating signal is achieved by using the

pre-emphasis circuit as shown in fig.1(a).

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Chapter 2: Angle Modulation

Figure 2.16: Pre-emphasis Circuit and its characteristics

As shown in the fig.1, the modulating AF signal is passed through a high
pass RC filter, before applying it to the FM modulator.
As fm increases, reactance of C decreases and modulating voltage applied
to FM modulator goes on increasing.
The frequency response characteristics of the RC high pass network is
shown in fig.1(b).
The boosting is done according to this pre arranged curve.
The amount of pre-emphasis in US FM transmission and sound
transmission in TV has been standardized at 75 μsec.

The pre-emphasis circuit is basically a high pass filter. The pre-emphasis

is carried out at the transmitter. The frequency for the RC high pass
network is 2122 Hz as shown in fig.1 (b). Hence, the pre-emphasis circuit
is used at the transmitter as shown in fig.2.

Figure 2.17: FM transmitter including the pre-emphasis

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Chapter 2: Angle Modulation

De-emphasis

The process that is used at the receiver end to nullify or compensate the
artificial boosting given to the higher modulating frequencies in the
process of pre-emphasis is called De-emphasis.

That means, the artificially boosted high frequency signals are brought to
their original amplitude using the de-emphasis circuit.

The 75 μsec de-emphasis circuit is standard and it is as shown in fig. 3.

Figure 2.18: De-emphasis circuit and its characteristics

It shows that it is a low pass filter. 75 μsec de-emphasis corresponds to a

frequency response curve that is 3 dB down at a frequency whose RC time
constant is 75 μsec.i.e.,

The demodulated FM is applied to the De-emphasis circuit. With increase

in fm the reactance of C goes on decreasing and the output of de-emphasis
circuit will also reduce as shown in fig.3.

Short Answer Questions

1. Define terms frequency deviation and modulation index in FM

wave.
2. Differentiate AM and FM.
3. What are the types of FM signals and calculate its bandwidth.

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Chapter 2: Angle Modulation

4. Write the time domain equation for FM wave and write the single
tone FM equation.
5. What are the applications of PLL?
6. Define angle modulation and mention its types.

Long Answer Questions

1. Define frequency modulation? Derive the expression of FM signal

and give with single tone equation.
2. Draw the spectrum of FM signal using Bessel function and
calculate bandwidth.
3. Write NBFM and WBFM equations.
4. Compare direct and indirect methods of generating FM signals.
Explain the generation technique of FM using Armstrong method
with neat diagram.
5. Explain the balanced slop detector technique in FM using neat
sketch.
6. Draw the block diagram of FM demodulator and explain the effect
of noise in detail and compare the performance of AM and FM in
the presence of noise.
7. With a neat block diagram explain the pe-emphasis and de-
emphasis in FM.
8. With neat block diagram explain the generation of narrow band FM
and wide band FM.
9. Explain the phase locked loop (PLL).
10. Write the applications of FM and write the comparisons of AM and
FM.

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Chapter 3
Transmitters
3.1 Classification of Transmitter

1. According to the type of modulation used.

2. According to the service involved.
3. According to the frequency range used.
4. According to the power used.

Classification of radio transmitters

1. According to the type of modulation used.
2. According to the service involved.
3. According to the frequency range used.
4. According to the power used.

1. According to the type of modulation used

1. Amplitude Modulation Transmitters
2. Frequency Modulation Transmitters
3. Pulse Modulation Transmitters

2. According to the service involved

1. Radio Broadcast Transmitters
2. Radio Telephony Transmitters
3. Radio Telegraph Transmitters
4. Television Transmitters
5. Radar Transmitters
6. Navigational Transmitters

3. According to the frequency range used.

Frequency Range Band Designation
300-3000 kHz MF
3-30 MHz HF
30-300 MHz VHF
300-3000 MHz UHF

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Chapter 3: Transmitters

4. According to the power used.

 In high power transmitters, a power supply circuit to transform the

input electrical power to the higher voltages needed to produce the
required power output.
 A radio frequency (RF) amplifier to increase the power of the signal,
to increase the range of the radio waves
 A radio transmitter is an electronic circuit which transforms
electric power from a power source, a battery or mains power, into
a radio frequency alternating current to apply to the antenna, and the
antenna radiates the energy from this current as radio waves.

3.1.1 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.

Figure 3.1: The block diagram of AM Transmitter

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.
 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.

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Chapter 3: Transmitters

3.1.2 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.

Figure 3.2: The block diagram of FM transmitter

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.

Requirements of a Receiver

AM receiver receiving 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.

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Chapter 3: Transmitters

 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.

Function of Radio Receiver

1. Intercept the incoming modulated signal by receiving antenna

2. Select the desired signal and reject unwanted signal
3. Amplify the selected RF signal
4. Detect the modulated signal to get back original message signal
5. It amplifies the modulating signal and finally receive sound signal
through loud speaker

3.2 Classification of Radio Receivers

Radio receivers are into two types

1. Tuned Radio Frequency Receiver

2. Super heterodyne Receiver

3.2.1 Tuned Radio Frequency (TRF) Receiver

A tuned radio frequency receiver (or TRF receiver) is a type of radio

receiver that is composed of one or more tuned radio frequency (RF)
amplifier stages followed by a detector (demodulator) circuit to extract the
audio signal and usually an audio frequency amplifier

The classic TRF receivers of the 1920s and 30s usually consisted of three
sections:

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Chapter 3: Transmitters

Figure 3.3: The block diagram of TRF receiver

 One or more tuned RF amplifier stages. These amplify the signal of

the desired station to a level sufficient to drive the detector, while
rejecting all other signals picked up by the antenna.
 A detector, which extracts the audio (modulation) signal from the
radio carrier signal by rectifying it.
 Optionally, but almost always included, one or more audio
amplifier stages which increase the power of the audio signal.
 TRF's disadvantages as "poor selectivity and low sensitivity
 The major problem with the TRF receiver, particularly as a
consumer product, was its complicated tuning.
 All the tuned circuits need to track to keep the narrow bandwidth
tuning. Keeping multiple tuned circuits aligned while tuning over a
wide frequency range is difficult.
 A super heterodyne receiver only needs to track the RF and LO
stages;
 The IF amplifier which is fixed-tuned.

3.2.2 Super Heterodyne Receiver

Signals enter the receiver from the antenna and are applied to the RF
amplifier where they are tuned to remove the image signal and also reduce
the general level of unwanted signals on other frequencies that are not
required.

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Chapter 3: Transmitters

Figure 3.4: Super heterodyne Receiver and Waveforms

The signals are then applied to the mixer along with the local oscillator
where the wanted signal is converted down to the intermediate frequency.
Here significant levels of amplification are applied and the signals are
filtered. This filtering selects signals on one channel against those on the
next. It is much larger than that employed in the front end. The advantage
of the IF filter as opposed to RF filtering is that the filter can be designed
for a fixed frequency. This allows for much better tuning. Variable filters
are never able to provide the same level of selectivity that can be provided
by fixed frequency ones.

Once filtered the next block in the super heterodyne receiver is the
demodulator. This could be for amplitude modulation, single sideband,
frequency modulation, or indeed any form of modulation. It is also
possible to switch different demodulators in according to the mode being
received.

The final element in the super heterodyne receiver block diagram is shown
as an audio amplifier, although this could be any form of circuit block that
is used to process or amplified the demodulated signal.

3.2.3 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.

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Chapter 3: Transmitters

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.

Figure 3.5: An AM Receiver

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: 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.

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Chapter 3: Transmitters

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.

3.3 RF Section and Characteristics

The amplitude modulated wave received by the antenna is first passed to

the tuner circuit through a transformer. ... It selects the frequency, desired
by the AM receiver. It also tunes the local oscillator and the RF filter at
the same time.

Frequency changing and tracking

 The receiver has a number of tunable circuits such as the antenna or

mixer or a local oscillator tuned circuit
 All the circuits must be tuned correctly if any station is to be tuned
 For this reason the capacitors in the various tuned circuits are
ganged

Figure 3.6: Mixer equivalent at fo

Intermediate frequency

Mixers generate signals that are the sum or difference of incoming signal
or carrier frequency

(fRF ) and the frequency of local oscillator ( fLO )

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Chapter 3: Transmitters

The difference between RF and Local Oscillator is always constant (IF)

If fLO fRF high – side injection fLO = fRF + fIF
If fLO fRF low – side injection fLO = fRF + fIF
The difference between RF and Local Oscillator is always constant (IF)
fIF = fLO - fRFz
Common AM receivers using fIF = 455 kHz and FM receivers using
fIF =10.7 MHz

Image frequency Image frequency

Image frequency

In a standard broadcast receiver, the local oscillator frequency is made

higher than incoming signal frequency. It is made equal to the signal
frequency plus the intermediate frequency.

fLO = fRF + fIF and fRF = fLO - fIF

When and are mixed the difference frequency called intermediate

frequency is resulted. As such, it is the any one passed and amplified by
the IF stage.

fSI = fLO + fIF (since fLO = fRF + fIF )

fSI = fRF + fIF + fIF
fSI = fRF + 2fIF

Unfortunately this spurious intermediate signal is also be amplified by the

IF stage and will provide interference. This has the effect of two station

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Chapter 3: Transmitters

being received simultaneously and is naturally undesirable. The term “fSI”

is called the “Image frequency” and is defined as the signal
frequency(fRF) plus twice the intermediate frequency (fIF)

fSI = fRF + 2fIF

Where
fSI = Image frequency, fRF = Signal frequency and fIF = Intermediate frequency

Image Frequency Rejection Ratio

It is defined as the ratio of the gain at the signal frequency (fRF) to the gain
at the image frequency(fSI). This is also designated as the rejection of
image frequency by a signal tuned frequency. The rejection of image
frequency is given by

Where
(fSI / fRF ) - ( fRF / fSI ) and Q is the loaded quality factor of tuned circuit

Problem 1. A super heterodyne radio receiver with an intermediate

frequency of 455kHz is tuned to a station operating at 1200kHz. The
associated image frequency is _______ kHz.

Solution:

We know that fSI = fRF + 2fIF

Given values are Intermediate frequency (fIF ) = 455 kHz,
Signal frequency(fRF ) = 1200 kHz
Now Image frequency (fSI ) = 1200 kHz + 2 (455kHz)
= 1200+900
Therefore Image frequency = 2100kHZ

Problem 2. For a broad cast super heterodyne AM receiver having no RF

amplifier, the loaded quality factor Q of the antenna coupling circuit is
100.Now if the intermediate frequency is 455kHz then determine.

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Chapter 3: Transmitters

i. The image frequency and its rejection ratio at the incoming

frequency of 1MHz
ii. The image frequency and its rejection ratio at the incoming
frequency of 25MHz

Solution:

For Q = 100, fIF = 455kHz and fRF = 1MHz

fSI = fRF + 2fIF then fSI = 1MHz+2(455 Khz )
fSI = 1.91MHz

(fSI / fRF ) - ( fRF / fSI ) =1.386

= 1.37603
For Q = 100 , fIF = 455kHz and fRF = 1MHz
fSI = fRF + 2fIF then fSI = 25MHz+2(455 Khz )
fSI = 25.91MHz

(fSI / fRF ) - ( fRF / fSI ) =0.07

= 7.071

Problem 3. For a receiver width IF and RF frequencies 455kHz, 900 Khz

respectively determine (i) Local oscillator frequency (ii) Image frequency
(iii) IFRR

Solution:
fIF = 455kHz, fRF =900kHz and what is fLO
fLO = fRF + fIF = fLO=900kHz + 455kHz therefore fLO =1355kHz
fSI = fRF + 2fIF =900kHz + 2(455kHz) therefore fSI =1810kHz
For (fSI / fRF ) - ( fRF / fSI ) = 1.513

= 121.04

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Chapter 3: Transmitters

3.4 Automatic Gain Control (AGC) or Automatic Volume

Control (AVC)

1. Simple AGC: Simple AGC is a system by means of which overall gain

of a radio receiver is varied, automatically with the changing strength of
the receiver signal to keep the output substantially constant. Hence the
receiver gain is automatically reduced as the input signal becomes more
& more strong There is a reduction in gain for weak signals. It is used
in domestic radio receiver.

Figure 3.7: Block diagram for AM radio receiver with AGC

2. Delayed AGC: As shown in the diagram, AGC biased is not applied

until the input signal strength reaches the predetermined level of point
A After this level, the point A AGC bias is applied just like simple
AGC but more strongly. There is no reduction in gain for weak signals.
The problem of reducing the receiver gain for weak signal is avoided
.the delayed AGC is not used in low cost radio receiver. It is used in
high quality receiver like communication receiver.

Figure 3.8: Delayed AGC circuit

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Chapter 3: Transmitters

Figure 3.9: AGC graph for simple and delayed

AGC circuits are used in a wide range of applications including AM

receivers for controlling weak and strong signals received by the receiver

Amplitude limiting

An amplitude limiting circuit is used to limit the amplitude of an output

signal within a predetermined range and is referred to as a limiter
circuit or a slice circuit, depending on its applications.

3.5 Amplitude Limiter in FM Receiver

In order to make full use of the advantages offered by FM, a demodulator

must be preceded by an Amplitude Limiter in FM Receiver

FM demodulators react to amplitude changes as well as frequency

changes. The limiter is a form of clipping device, a circuit whose output
tends to remain constant despite changes in the input signal.

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Chapter 3: Transmitters

Figure 3.10: Amplitude limiter circuit

Figure shows a typical FET Amplitude Limiter in FM Receiver.

Examination of the dc conditions shows that the drain supply voltage has
been dropped through resistor RD. Also, the bias on the gate is leak-type
bias supplied by the parallel Rg – Cg combination. Finally, the FET is
shown neutralized by means of capacitor CN, in consideration of the high
frequency of operation.
It is seen that the bias on the FET is increased in proportion to the size of
the input voltage. As a result, the gain of the amplifier is lowered, and the
output voltage tends to remain constant.

3.6 FM Receiver

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

Figure 3.11: Block diagram of FM receiver

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Chapter 3: Transmitters

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.

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

3.7 Comparison of AM and FM Receivers.

Amplitude Modulation (AM) Frequency Modulation (FM)

The first successful audio Developed in 1930 by Edwin
transmission was carried out in Armstrong, in the United States
the mid-1870s
The radio wave is called a carrier The radio wave is called a carrier
wave and the frequency and phase wave, but the amplitude and phase
remain the same remain the same
Has poor sound quality, but can Has higher bandwidth with better
transmit longer distance sound quality
The frequency range of FM is 88
The frequency range of AM radio
to 108 MHz in the higher
varies from 535 to 1705 kHz
spectrum
More susceptible to noise Less susceptible to noise
Table 3.1: for comparison AM and FM

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Chapter 3: Transmitters

Short Answer Questions

1. Define transmitters and receivers.

2. Write the main requirements of AM broadcast transmitters.
3. Define the term fidelity.
4. Explain the need of amplitude limiter in FM receiver.
5. Mention the advantages of RF amplifier.
6. Define intermediate frequency and AGC.

Long Answer Questions

1. Draw a neat block diagram of AM transmitter.

2. Draw the block diagram of super heterodyne receiver and explain its
operation? what are the advantages of this receiver?
3. What is Automatic gain controlling radio receiver? And explain in
detail.
4. Explain the tuned radio frequency receiver with the help of block
diagram.
5. Compare AM and FM receiver.
6. List and discuss the characteristics of receiver.
7. What is an amplitude limiter? Explain its operation with neat sketch.
8. Distinguish between simple AGC and delayed AGC.
9. A super heterodyne radio receiver is tuned to receive 1550 KHz
carrier amplitude modulated by a 6 KHz sinusoidal tone. Assuming
the IF to be 455 KHz, identify the input and output frequency
components for the IF amplifier. The IF band width is 15 KHz.

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Chapter 4
Pulse Modulation
Analog to Digital

The communication that occurs in our day-to-day life is in the form of

signals. These signals, such as sound signals, generally, are analog in
nature. When the communication needs to be established over a distance,
then the analog signals are sent through wire, using different techniques
for effective transmission.

The Necessity of Digitization

The conventional methods of communication used analog signals for long

distance communications, which suffer from many losses such as
distortion, interference, and other losses including security breach.

In order to overcome these problems, the signals are digitized using

different techniques. The digitized signals allow the communication to be
more clear and accurate without losses.

The following figure indicates the difference between analog and digital
signals. The digital signals consist of 1s and 0s which indicate High and
Low values respectively.

Figure 4.1: Representation of signals

Advantages of Digital Communication

As the signals are digitized, there are many advantages of digital

communication over analog communication, such as −

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Chapter 4: Pulse Modulation

 The effect of distortion, noise, and interference is much less in

digital signals as they are less affected.
 Digital circuits are more reliable.
 Digital circuits are easy to design and cheaper than analog circuits.
 The hardware implementation in digital circuits, is more flexible
than analog.
 The occurrence of cross-talk is very rare in digital communication.
 The signal is un-altered as the pulse needs a high disturbance to alter
its properties, which is very difficult.
 Signal processing functions such as encryption and compression are
employed in digital circuits to maintain the secrecy of the
information.
 The probability of error occurrence is reduced by employing error
detecting and error correcting codes.
 Spread spectrum technique is used to avoid signal jamming.
 Combining digital signals using Time Division Multiplexing TDM is
easier than combining analog signals using Frequency Division
Multiplexing FDM.
 The configuring process of digital signals is easier than analog
signals.
 Digital signals can be saved and retrieved more conveniently than
analog signals.
 Many of the digital circuits have almost common encoding
techniques and hence similar devices can be used for a number of
purposes.
 The capacity of the channel is effectively utilized by digital signals.

4.1 Elements of Digital Communication

The elements which form a digital communication system are represented

by the following block diagram for the ease of understanding.

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Figure 4.2: block diagram of digital communication

Following are the sections of the digital communication system.

Source: The source can be an analog signal. Example: A Sound signal

A digital signal is generally represented by a binary sequence.

Source encoder: The source encoder compresses the data into minimum
number of bits. This process helps in effective utilization of the bandwidth.
It removes the redundant bits unnecessary excess bits, i.e., zeroes.

Channel encoder: The channel encoder does the coding for error
correction. During the transmission of the signal, due to the noise in the
channel, the signal may get altered and hence to avoid this, the channel
encoder adds some redundant bits to the transmitted data. These are the
error correcting bits.

Digital modulator: The signal to be transmitted is modulated here by a

carrier. The signal is also converted to analog from the digital sequence, in
order to make it travel through the channel or medium.

Channel: The channel or a medium, allows the analog signal to transmit

from the transmitter end to the receiver end.

Digital demodulator: This is the first step at the receiver end. The
received signal is demodulated as well as converted again from analog to
digital. The signal gets reconstructed here.

Channel decoder: The channel decoder, after detecting the sequence,

does some error corrections. The distortions which might occur during the

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transmission, are corrected by adding some redundant bits. This addition

of bits helps in the complete recovery of the original signal.

Source decoder: The resultant signal is once again digitized by sampling

and quantizing so that the pure digital output is obtained without the loss
of information. The source decoder recreates the source output.

Destination: This is the output which is produced after the whole

process. Example − The sound signal received.

4.2 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

4.2.1 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.

Figure 4.3: Generation of PAM block diagram

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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.

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.

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Figure 4.4: Pulse amplitude modulation natural and flat – top wave forms

4.2.2.1 Demodulation of PAM Signals

For pulse amplitude modulated (PAM) signals, the demodulation is done

using a Holding circuit. Fig.1 shows the block diagram of a PAM
demodulator

Figure 4.5: Detection of PAM block diagram

In this method, the received PAM signal is allowed to pass through a

Holding circuit and a low pass filter (LPF) as shown in fig 4.6.

Now, Here the switch ‘S’ is closed after the arrival of the pulse and it is
opened at the end of the pulse. In this way, the capacitor C is charged to
the pulse amplitude value and it holds this value during the interval
between the two pulses.

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Figure 4.6: Simple holding circuit and PAM Signal

Hence, the sampled values are held as shown in fig.3.After this the holding
circuit output is smoothened in Low Pass filter as shown in fig.3.

It may be observed that some kind of distortion is introduced due to the

holding circuit. In fact the circuit of fig.4 is known as zero-order Holding
circuit. This zero-order Holding circuit considers only the previous sample
to decide the value between the two pulses.

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

Figure 4.7: Generation of PWM 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.

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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.

Figure 4.8: Pulse Width Modulation Wave form

To generate PWM signals in analog circuits, a saw tooth triangle

waveform is used. It can be easily generated using a simple oscillator. A
comparator can also be included. The logic of generating PWM signals is
very simple. If the value of the reference signal (saw tooth triangle signal)
is more than the modulation waveform, the resultant PWM signal is a high
signal, otherwise, the resultant waveform is in the low state. This method
is also called as a carrier-based generation where the reference signals act
as the carrier waveform.

4.2.2.1 Pulse Width Demodulation

Figure 4.9: Block diagram of pulse width demodulation

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The PWM and the carrier signals are connected to the inputs of a product
detector, and then a sequence of pulses having the width inversely
proportional to the width of PAM pulse presents at output. When the Va
signal passes through the low-pass filter, a demodulated signal is obtained.

4.2.3 Pulse Position Modulation

In PPM, the amplitude and width of the pulses is kept constant but the
position of each pulse is varied in accordance with the amplitudes of the
sampled values of the modulating signal.

The position of the pulses is changed with respect to the position of

reference pulses.

The PPM pulses can be derived from the PWM pulses as shown in fig.1.

Here, it may be noted that with increase in the modulating voltage the
PPM pulses shift further with respect to reference.

Figure 4.10: Generation of pulse position modulation signal

The PWM pulses obtained at the comparator output are applied to a

monostable multivibrator. The monostable is negative edge triggered.

Hence, corresponding to each trailing edge of PWM signal, the

monostable output goes high.

It remains high for a fixed time decided by its own RC components.

Thus, as the trailing edges of the PWM signal keep shifting in proportion
with the modulating signal x(t), the PPM pulses also keep shifting, as
shown in fig.3.

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Figure 4.11: Pulse Position Modulation Wave form

4.2.3.1 Demodulation of PPM Signal

The PPM demodulator block diagram has been shown in fig.4.

Figure 4.12: Block diagram of demodulation of PPM signal

The operation of the demodulator circuit may be explained as under:

 The noise corrupted PPM waveform is received by the PPM

demodulator circuit.
 The pulse generator develops a pulsed waveform at its output of
fixed duration and applies these pulses to the reset pin (R) of a SR
flip-flop.
 A fixed period reference pulse is generated from the incoming PPM
waveform and the SR flip-flop is set by the reference pulses.
 Due to the set and reset signals applied to the flip-flop, we get a
PWM signal at its output.
 The PWM signal can be demodulated using the PWM demodulator.

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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.

4.2.3.2 Difference between PAM, PWM, and PPM

S. No. Parameter PAM PWM PPM
Variable of Amplitude Width Position
1
Carrier Pulse
Bandwidth Low High High
2
Requirement
Noise High Minimum Minimum
3
Interference
Information Amplitude Width Position
4
Contained in Variations Variations Variations
Power Low Moderate High
5 efficiency
(SNR)
Transmitted Varies with an Varies with Remains
6 Power amplitude of variation in Constant
pulses width
Need to transmit Not needed Not needed Necessary
7 synchronizing
pulses
Bandwidth BW depends on BW depends on BW depends on
8 (BW) the width of the the rise time of the rise time of
depends on pulse the pulse the pulse
Transmitter varies with the varies with the remains
power amplitude of the amplitude and constant with
9
pulses width of the the width of the
pulses pulses
The complexity
10 of generation Complex Easy Complex
and detection
11 Similarity Similar to AM Similar to FM Similar to PM

Table 4.1: Comparisons of pulse modulation with carrier

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4.3 Pulse Code Modulation

Definition In this modulation there is no carrier and pulse train but we can
send the digital data with the help of sampler, quantizer and encoder,
Which converts the analog signal to digital signal is called a pulse code
modulation..

4.3.1 Pulse Code Modulation (PCM)

It is that the technique used for reworking analog signal into digital
signal. PCM has good or sensible signal to noise ration. For
transmission, Pulse Code Modulation wants high transmitter bandwidth.
PCM technique is split into three elements, initial is that the transmission
at the provision end, second regeneration at the transmission path and
conjointly the receiving end.

Figure 1.13: Block diagram of PCM signal generation

The message signal is the signal which is being transmitted for

communication and the carrier signal is a high frequency signal which has
no data, but is used for long distance transmission.

There are many modulation techniques, which are classified according to

the type of modulation employed. Of them all, the digital modulation
technique used is Pulse Code Modulation PCM.

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A signal is pulse code modulated to convert its analog information into a

binary sequence, i.e., 1s and 0s. The output of a PCM will resemble a
binary sequence. The following figure shows an example of PCM output
with respect to instantaneous values of a given sine wave.

Figure 1.14: Representation of PCM signal

Instead of a pulse train, PCM produces a series of numbers or digits, and

hence this process is called as digital. Each one of these digits, though in
binary code, represent the approximate amplitude of the signal sample at
that instant.

In Pulse Code Modulation, the message signal is represented by a

sequence of coded pulses. This message signal is achieved by representing
the signal in discrete form in both time and amplitude.

Basic Elements of PCM

The transmitter section of a Pulse Code Modulator circuit consists

of Sampling, Quantizing and Encoding, which are performed in the
analog-to-digital converter section. The low pass filter prior to sampling
prevents aliasing of the message signal.

The basic operations in the receiver section are regeneration of impaired

signals, decoding, and reconstruction of the quantized pulse train.
Following is the block diagram of PCM which represents the basic
elements of both the transmitter and the receiver sections.

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Low pass filter: This filter eliminates the high frequency components
present in the input analog signal which is greater than the highest
frequency of the message signal, to avoid aliasing of the message signal.

Sampler: This is the technique which helps to collect the sample data at
instantaneous values of message signal, so as to reconstruct the original
signal. The sampling rate must be greater than twice the highest frequency
component W of the message signal, in accordance with the sampling
theorem.

Quantizer: Quantizing is a process of reducing the excessive bits and

confining the data. The sampled output when given to Quantizer reduces
the redundant bits and compresses the value.

Encoder: The digitization of analog signal is done by the encoder. It

designates each quantized level by a binary code. The sampling done here
is the sample-and-hold process. These three sections LPF, Sampler, and
Quantizer will act as an analog to digital converter. Encoding minimizes
the bandwidth used.

Regenerative Repeater

The most important feature of PCM system lies in its ability to control the
effects of distortion and noise when the PCM wave travels on the channel.
This is accomplished by means of using a chain of regenerative repeaters
as shown in fig.2.

Such repeaters are spaced close enough to each other on the transmission
path.

This section increases the signal strength. The output of the channel also
has one regenerative repeater circuit, to compensate the signal loss and
reconstruct the signal, and also to increase its strength.

Figure 4.15: Regenerative repeater block

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4.3.2 PCM Receiver

Fig. 3 shows the block diagram of a PCM receiver.

Figure 4.16: Pulse code modulation signal receiver

The regenerator at the start of PCM receiver reshapes the pulse and
removes the noise.

This signal is then converted to parallel digital words for each sample.

Now, the digital word is converted to its analog value denoted as xq(t) with
the help of a sample and hold circuit.

This signal, at the output of sample and hold circuit is allowed to pass
through a low-pass reconstruction filter to get the original message signal
x(t) .

Sampling

Sampling is defined as, “The process of measuring the instantaneous

values of continuous-time signal in a discrete form.”

Sample is a piece of data taken from the whole data which is continuous
in the time domain.

When a source generates an analog signal and if that has to be digitized,

having 1s and 0s i.e., High or Low, the signal has to be discretized in time.
This discretization of analog signal is called as Sampling.

The following figure indicates a continuous-time signal X(t) and a

sampled When Xs(t) is multiplied by a periodic impulse train, the
sampled signal Xs(t) is obtained.

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Figure 4.17: Sampling signal

Sampling Rate

To discretize the signals, the gap between the samples should be fixed.
That gap can be termed as a sampling period Ts.

Sampling Frequency = 1/Ts

⸫ Sampling Frequency = fs

Where,
 Ts is the sampling time
 fs is the sampling frequency or the sampling rate

Sampling frequency is the reciprocal of the sampling period. This

sampling frequency, can be simply called as Sampling rate. The sampling
rate denotes the number of samples taken per second, or for a finite set of
values.

For an analog signal to be reconstructed from the digitized signal, the

sampling rate should be highly considered. The rate of sampling should be
such that the data in the message signal should neither be lost nor it should
get over-lapped. Hence, a rate was fixed for this, called as Nyquist rate.

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Nyquist Rate

Suppose that a signal is band-limited with no frequency components

higher than W Hertz. That means, W is the highest frequency. For such a
signal, for effective reproduction of the original signal, the sampling rate
should be twice the highest frequency. Which means,

fs = 2W

Where,
 fS is the sampling rate
 W is the highest frequency

This rate of sampling is called as Nyquist rate.

A theorem called, Sampling Theorem, was stated on the theory of this

Nyquist rate.

Sampling Theorem

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 band limited.

The sampling theorem states that, “a signal can be exactly reproduced if it

is sampled at the rate fs which is greater than twice the maximum
frequency W.”

To understand this sampling theorem, let us consider a band-limited

signal, i.e., a signal whose value is non-zero between some –
W and W Hertz.

Such a signal is represented as x (f) =0 for |f|>W For the continuous-time

signal x (t) the band-limited signal in frequency domain, can be
represented as shown in the following figure.

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Figure 4.18: Band limited signal for fs = 2W

We need a sampling frequency, a frequency at which there should be no

loss of information, even after sampling. For this, we have the Nyquist rate
that the sampling frequency should be two times the maximum frequency.
It is the critical rate of sampling.
If the signal xtt is sampled above the Nyquist rate, the original signal can
be recovered, and if it is sampled below the Nyquist rate, the signal cannot
be recovered.

The following figure explains a signal, if sampled at a higher rate

than 2w in the frequency domain.

Figure 4.19: Band limited Signal for fs < 2W

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The above figure shows the Fourier transform of a signal xs(t). Here, the
information is reproduced without any loss. There is no mixing up and
hence recovery is possible.
The Fourier Transform of the signal xs(t) is

∞
Xs (w)=1/Ts∑ X(w−nw0)
n=−∞

Where Ts = Sampling Period and w0 = 2πTs

Let us see what happens if the sampling rate is equal to twice the highest
frequency (2W)

That means,
fs > 2W

Where,
 fs is the sampling frequency
 W is the highest frequency

The result will be as shown in the above figure. The information is

replaced without any loss. Hence, this is also a good sampling rate.

Now, let us look at the condition,

The resultant pattern will look like the following figure.

Figure 4.20: Band limited signal for fs > 2W

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We can observe from the above pattern that the over-lapping of

information is done, which leads to mixing up and loss of information.
This unwanted phenomenon of over-lapping is called as Aliasing.

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.”
The corrective measures taken to reduce the effect of Aliasing are –

 In the transmitter section of PCM, a low pass anti-aliasing filter is

employed, before the sampler, to eliminate the high frequency
components, which are unwanted.
 The signal which is sampled after filtering, is sampled at a rate
slightly higher than the Nyquist rate.
This choice of having the sampling rate higher than Nyquist rate,
also helps in the easier design of the reconstruction filter at the
receiver.

Scope of Fourier Transform

It is generally observed that, we seek the help of Fourier series and Fourier
transforms in analyzing the signals and also in proving theorems. It is
because –

 The Fourier Transform is the extension of Fourier series for non-

periodic signals.
 Fourier transform is a powerful mathematical tool which helps to
view the signals in different domains and helps to analyze the signals
easily.
 Any signal can be decomposed in terms of sum of sines and cosines
using this Fourier transform.
 In the next chapter, let us discuss about the concept of Quantization.

4.4 Quantization

The digitization of analog signals involves the rounding off of the values
which are approximately equal to the analog values. The method of

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sampling chooses a few points on the analog signal and then these points
are joined to round off the value to a near stabilized value. Such a process
is called as Quantization.

Quantizing an Analog Signal

The analog-to-digital converters perform this type of function to create a

series of digital values out of the given analog signal. The following figure
represents an analog signal. This signal to get converted into digital, has to
undergo sampling and quantizing.

The quantizing of an analog signal is done by discretizing the signal with a

number of quantization levels. Quantization is representing the sampled
values of the amplitude by a finite set of levels, which means converting a
continuous-amplitude sample into a discrete-time signal.

The following figure shows how an analog signal gets quantized. The blue
line represents analog signal while the brown one represents the quantized
signal.

Both sampling and quantization result in the loss of information. The

quality of a Quantizer output depends upon the number of quantization
levels used. The discrete amplitudes of the quantized output are called
as representation levels or reconstruction levels. The spacing between
the two adjacent representation levels is called a quantum or step-size.

The following figure shows the resultant quantized signal which is the
digital form for the given analog signal.

This is also called as Stair-case waveform, in accordance with its shape.

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Types of Quantization

There are two types of Quantization - Uniform Quantization and Non-

uniform Quantization.
The type of quantization in which the quantization levels are uniformly
spaced is termed as a Uniform Quantization. The type of quantization in
which the quantization levels are unequal and mostly the relation between
them is logarithmic, is termed as a Non-uniform Quantization.
There are two types of uniform quantization. They are Mid-Rise type and
Mid-Tread type. The following figures represent the two types of uniform
quantization.

Figure 1: Mid-Rise type uniform Figure 2: Mid-tread type uniform

quantization quantization

Figure 1 shows the mid-rise type and figure 2 shows the mid-tread type of
uniform quantization.

 The Mid-Rise type is so called because the origin lies in the middle
of a raising part of the stair-case like graph. The quantization levels
in this type are even in number.
 The Mid-tread type is so called because the origin lies in the middle
of a tread of the stair-case like graph. The quantization levels in this
type are odd in number.
 Both the mid-rise and mid-tread type of uniform quantizers are
symmetric about the origin.
4.4.1 Quantization Error (Noise)
For any system, during its functioning, there is always a difference in the
values of its input and output. The processing of the system results in an
error, which is the difference of those values.

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The difference between an input value and its quantized value is called
a Quantization Error. A Quantizer is a logarithmic function that
performs Quantization rounding off the value. An analog-to-digital
converter (ADC) works as a quantizer.

Figure 4.21: Quantization error in PCM

For uniform quantization, the maximum quantization error is given as

from above fig LSB =

max = Here is step size

 This error is uniformly distributed over the interval - max .

The pdf of this error is given as

f ( = for -

 Mean square value of this quantization error is given as

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E [ 2] = 2
f ( d

2 3
f ( d

E [ 2] = 2
d = [ 3 /3

-
3 3

= [ ]

E [ 2] = 2
/ 12

 Since the quantization error has zero mean value ,Noise power is
equal to mean square value ie

Noise power = E [ 2] = 2
/ 12

 Let the peak to peak normalized signal amplitude be (- 1,1). Then

step size will be i.e.

= =

= since number of levels q = 2v

Here v is the number of bits used in PCM

2
Noise power = / 12 = (2 / 2v)2 / 12
= (4 / 22v) / 12 = 1 / 3 22v

 Let the signal power be P. Then above equation becomes,

= P / 1/(3 22v) = 3 22v

 For normalized power, P=1. Then above equation becomes

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= 3 22v

[ ] dB = 10 log 10 3 22v

[ ] dB = (4.8 + 6v). dB.

This is the signal to noise ratio for linear quantization.

Example 2. A television signal with a band width of 4.2 MHz is

transmitted using binary PCM. The number of quantization levels is
512.Calculate, i) Code word length ii) Transmission bandwidth iii) Final
bit rate iv) Output signal to quantization noise ratio

The bandwidth is 4.2 MHz, means highest frequency component will have
frequency of 4.2 MHz i.e. W = 4.2 MHz and quantization levels q = 512

i) Number of bits and quantization levels are related in binary PCM as

q = 2v
512 = 2v
Log 512 = v log 2
v = Log 512 / log 2 = 9 bits
Thus the codeword length is 9 bits

ii) The transmission channel bandwidth is given as ,

BT vW
BT 9x 4.2 x 105
BT 37.8 MHz

iii) Final bit rate will equal to the signaling rate i.e.
r = v fs
Sampling frequency fs 2W by sampling theorem.
fs 2x4.2 MHz since W =4.2 MHz
fs 8.4 MHz

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Now Signaling rate r = v fs

r = 9x8.4x106 r = 75.6 x 106 bits / sec

iv) The signal to noise ratio

[ ] dB (4.8 + 6v). dB.

[ ] dB (4.8 + 6x9). dB.

[ ] dB 58.8 dB.

4.5 Signal to Noise Ratio (SNR) in PCM

Let the sine wave represented by m (t) = Am Cos Wm t

Signal power calculated from the message amplitude i.e. PS = (Am)2 /2

Error signal is given by error of the quantized signal and message signal
i.e. error (e) = mQ(t) –m(t)

Noise power PN is calculated from error signal using pdf

2
Therefore PN = (pdf ) de
2
PN = (k) de

/2 /2

Using property of pdf function as area under pdf is unity

i.e. Area of pdf =1 xk=1 k= 1/

2
Now noise power PN = (1/ ) de

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= [ e 3 / 3]

3 3

= [ ]

2
Noise power PN = /12
Now signal to noise ratio power (S/N) i.e. PS/PN
PS/PN =[ (Am)2 / 2] / [ 2/12] = 6 (Am)2 / 2

PS/PN = 6 (Am)2 / 2

4.6 Companding in PCM

The word Companding is a combination of Compressing and Expanding,
which means that it does both. This is a non-linear technique used in PCM
which compresses the data at the transmitter and expands the same data at
the receiver. The effects of noise and crosstalk are reduced by using this
technique.SNR)

Figure 4.22: Block diagram of Companding Technique

Figure 4.22: Graph representation for Companding Technique

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There are two types of Companding techniques. They are –

µ-law Companding Technique

 Uniform quantization is achieved at µ = 0, where the characteristic

curve is linear and no compression is done.
 µ-law has mid-tread at the origin. Hence, it contains a zero value.
 µ-law companding is used for speech and music signals.

Figure 4.23: PCM Technique Performances

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4.7 Differential PCM

For the samples that are highly correlated, when encoded by PCM
technique, leave redundant information behind. To process this redundant
information and to have a better output, it is a wise decision to take a
predicted sampled value, assumed from its previous output and summarize
them with the quantized values. Such a process is called as Differential
PCM (DPCM).

4.7.1 DPCM Transmitter

The DPCM Transmitter consists of Quantizer and Predictor with two

summer circuits. Following is the block diagram of DPCM transmitter.

Figure 4.24: Generation of DPCM Signal

The signals at each point are named as –

 x(nTs) is the sampled input

 xˆ(nTs) is the predicted sample
 e(nTs) is the difference of sampled input and predicted output, often
called as prediction error
 v(nTs) is the quantized output
 u(nTs) is the predictor input which is actually the summer output of
the predictor output and the quantizer output

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The predictor produces the assumed samples from the previous outputs of
the transmitter circuit. The input to this predictor is the quantized versions
of the input signal x(nTs).

Quantizer Output is represented as −

v (nTs) =Q [e (nTs)]
=e (nTs) +q (nTs)
Where q (nTs) is the quantization error

Predictor input is the sum of quantizer output and predictor output,

u (nTs) =xˆ (nTs) +v (nTs)
u (nTs) =xˆ(nTs)+e(nTs)+q(nTs)
u (nTs) =x (nTs) +q (nTs)

The same predictor circuit is used in the decoder to reconstruct the original
input.

4.7.2 DPCM Receiver

The block diagram of DPCM Receiver consists of a decoder, a predictor,

and a summer circuit. Following is the diagram of DPCM Receiver.

Figure 4.25: Detection of DPCM signal

The notation of the signals is the same as the previous ones. In the absence
of noise, the encoded receiver input will be the same as the encoded
transmitter output.

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As mentioned before, the predictor assumes a value, based on the previous

outputs. The input given to the decoder is processed and that output is
summed up with the output of the predictor, to obtain a better output.

4.8 Delta Modulation

The sampling rate of a signal should be higher than the Nyquist rate, to
achieve better sampling. If this sampling interval in Differential PCM is
reduced considerably, the sampleto-sample amplitude difference is very
small, as if the difference is 1-bit quantization, then the step-size will be
very small i.e., Δ delta.

Delta Modulation

The type of modulation, where the sampling rate is much higher and in
which the step size after quantization is of a smaller value Δ, such a
modulation is termed as delta modulation.

Features of Delta Modulation

Following are some of the features of delta modulation.

 An over-sampled input is taken to make full use of the signal

correlation.
 The quantization design is simple.
 The input sequence is much higher than the Nyquist rate.
 The quality is moderate.
 The design of the modulator and the demodulator is simple.
 The stair-case approximation of output waveform.
 The step-size is very small, i.e., Δ deltadelta.
 The bit rate can be decided by the user.
 This involves simpler implementation.

Delta Modulation is a simplified form of DPCM technique, also viewed

as 1-bit DPCM scheme. As the sampling interval is reduced, the signal
correlation will be higher.

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4.8.1 Delta Modulator

The Delta Modulator comprises of a 1-bit quantizer and a delay circuit

along with two summer circuits. Following is the block diagram of a delta
modulator.

Figure 4.26: Generation of Delta Modulation Signal

The predictor circuit in DPCM is replaced by a simple delay circuit in

DM.

From the above diagram, we have the notations as –

 x(nTs) = over sampled input

 ep(nTs) = summer output and quantizer input
 eq(nTs) = quantizer output = v(nTs)
 xˆ(nTs) = output of delay circuit
 u(nTs) = input of delay circuit

Using these notations, now we shall try to figure out the process of delta
modulation.

ep (nTs) =x (nTs) −xˆ (nTs) ---------equation 1

x(nTs)−u([n−1]Ts) =x(nTs)−[xˆ[[n−1]Ts]+v[[n−1]Ts]] ---------equation 2

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Further,
v (nTs) =eq (nTs) =S.sig. [ep(nTs)] ---------equation 3
u(nTs)=xˆ(nTs)+eq(nTs)

Where,
 xˆ(nTs) = the previous value of the delay circuit
 eq(nTs) = quantizer output = v(nTs)

Hence,
u(nTs)=u([n−1]Ts)+v(nTs) ---------equation 4

Which means?
The present input of the delay unit
= The previous output of the delay unit + the present quantizer output
Assuming zero condition of Accumulation,
n
u(nTs)=S∑sig[ep(jTs)]
j=1
n
Accumulated version of DM output = ∑ v (jTs) ---------equation 5
J =1
Now, note that

xˆ(nTs)=u([n−1]Ts)

n−1
=∑ v(jTs) ---------equation 6
j=1

Delay unit output is an Accumulator output lagging by one sample.

From equations 5 & 6, we get a possible structure for the demodulator.

A Stair-case approximated waveform will be the output of the delta

modulator with the step-size as delta (Δ). The output quality of the
waveform is moderate.

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4.8.2 Delta Demodulator

The delta demodulator comprises of a low pass filter, a summer, and a

delay circuit. The predictor circuit is eliminated here and hence no
assumed input is given to the demodulator.

Following is the diagram for delta demodulator.

Figure 4.27: Detection of Delta Modulation Signal

From the above diagram, we have the notations as –

 vˆ(nTs) is the input sample

 uˆ(nTs) is the summer output
 x¯(nTs) is the delayed output

A binary sequence will be given as an input to the demodulator. The stair-

case approximated output is given to the LPF.

Low pass filter is used for many reasons, but the prominent reason is noise
elimination for out-of-band signals. The step-size error that may occur at
the transmitter is called granular noise, which is eliminated here. If there
is no noise present, then the modulator output equals the demodulator
input.

The delta modulation has two major drawbacks as under:

1. Slope overload distortion

2. Granular or idle noise

Now, we will discuss these two drawbacks in detail.

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1. Slope Overload Distortion

This distortion arises because of large dynamic range of the input signal.

Figure 4.28: Quantization Errors in Delta Modulation

We can observe from fig.1 , the rate of rise of input signal x(t) is so high
that the staircase signal ca not approximate it, the step size ‘Δ’ becomes
too small for staircase signal u(t) to follow the step segment of x(t).

Hence, there is a large error between the staircase approximated signal and
the original input signal x(t).

This error or noise is known as slope overload distortion.

To reduce this error, the step size must be increased when slope of signal x
(t) is high.

2. Granular or Idle Noise

Granular or Idle noise occurs when the step size is too large compared to
small variation in the input signal.

This means that for very small variations in the input signal, the staircase
signal is changed by large amount (Δ) because of large step size.

Fig.1 shows that when the input signal is almost flat, the staircase signal
u(t) keeps on oscillating by ±Δ around the signal.

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The error between the input and approximated signal is called granular
noise.

The solution to this problem is to make the step size small.

Solution

In order to overcome the quantization errors due to slope overload and

granular noise, the step size (Δ) is made adaptive to variations in the input
signal x(t).

Particularly in the steep segment of the signal x(t), the step size is
increased. And the step is decreased when the input is varying slowly.

This method is known as Adaptive Delta Modulation (ADM).

The adaptive delta modulators can take continuous changes in step size or
discrete changes in step size.

Advantages of DM Over DPCM

 1-bit quantizer
 Very easy design of the modulator and the demodulator

However, there exists some noise in DM.

 Slope Over load distortion (when Δ is small)

 Granular noise (when Δ is large)

4.9 Signal to Noise Ratio (SNR) in DM

Let the sine wave represented by m (t) = Am Cos Wm t

Slope of m(t) Derivation of m(t)
Maximum slope of Delta Modulation (DM) is the ratio of step size( ) and
sampling period
Max. Slope of DM = / Ts
Slope overload distortion occurs
If slope of sine wave slope of DM

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Max m(t)

Max Am sin 2 fm t

Max m 2 fm cos 2 fm t since cos 2 fm t for maximum value is 1

Then we get above equation

Am2 fm from this equation Am value is

Am

Here slope overload distortion will not occur then it is less than or equal
condition we can write

Am

Where Am is Amplitude of sine signal

is step size ,fm is signal frequency and is sampling period
Now Am value also we can write as

Am =
Now signal power from Am we know that PS=V2/R when R=1

PS = Am2/2 = ( )2 /2
2 2
Signal power PS = /8 fm2Ts2
Now calculate the noise power from quantization error in DM
Maximum quantization error is

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fe(e) =

Now noise power is PN = V2 / R

Noise is given by ‘e’ and its pdf is also given by fe(e) then

Mean square value is E(e2) it is also known as PN

Now E(e2) = = 2
fe(e) de

From pdf graph limits ( ) and fe(e) value is

2
PN = ( ) de

= [e 3 / 3]

3 3
= [ ]
2
Therefore noise power PN =

Noise power is extended from –fs to fs the receiver signal passed through
low pass filter whose cutoff frequency is W

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2 2
Output noise power = x ==

Now signal to noise ratio power (S/N) i.e. PS/PN

2 2
We know that Signal power PS =[ /8 fm2Ts2]
2 2
PS/PN = [ /8 fm2Ts2] / [(wTs 2
)/3]

PS/PN = 3fs3 / 8 2
fm3

4.10 Adaptive Delta Modulation ADM

In digital modulation, we have come across certain problem of

determining the step-size, which influences the quality of the output wave.
A larger step-size is needed in the steep slope of modulating signal and a
smaller stepsize is needed where the message has a small slope. The
minute details get missed in the process. So, it would be better if we can
control the adjustment of step-size, according to our requirement in order
to obtain the sampling in a desired fashion. This is the concept
of Adaptive Delta Modulation.

Following is the block diagram of Adaptive delta modulator.

Figure 4.29: Generation of adaptive delta modulation

The gain of the voltage controlled amplifier is adjusted by the output

signal from the sampler. The amplifier gain determines the step-size and
both are proportional.

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ADM quantizes the difference between the value of the current sample and
the predicted value of the next sample. It uses a variable step height to
predict the next values, for the faithful reproduction of the fast varying

4.11 Comparison between PCM, DM, ADM and DPCM

Basis of PCM DM ADM DPCM

Comparison
Number of It can use 4, 8 It uses one It uses only Bits can be
Bits or 16 bits per bit for one one bit for more than
sample. sample. one sample. one but are
less than
PCM.
Levels And The number of Step size is Step size Number of
Step Size levels depends kept fixed varies levels is
on number of and cannot according to fixed.
bits. Level be varied. the signal
size is fixed. variation.
Quantization Quantization Slope- Quantization Slope
Error & error depends overload noise is overload
Distortion on the number distortion present but distortion and
of levels. is present. no other quantization
errors. noise is
present.
Bandwidth Highest Lowest Lowest Bandwidth
bandwidth is bandwidth bandwidth is required is
required since is required. required. less than
the number of PCM.
bits is high.
Feedback There is no Feedback Feedback Feedback
feedback in exists in exists in the exists in the
transmitter or the transmitter. transmitter.
receiver. transmitter.
Complexity Complex Simple to Simple to Simple to
system to implement implement.
implement.

Table 4.2: Comparison between PCM, DM, ADM and DPCM

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Short Answer Questions

1. State the sampling theorem.

2. What are the applications of FDM and TDM?
3. What are the applications of PAM?
4. Difference between PAM, PWM, PPM.
5. Write short notes on (i) granular noise (ii) slope overloading error.
6. What is quantization error? And what is the difference between
uniform quantization and non uniform quantization.
7. Write any 4 differences between DPCM and adaptive DPCM.

Long Answer Questions

1. Define pulse modulation? Explain the modulation and

demodulation of pulse modulation.
2. With neat sketch explain the generation of PPM from PWM and its
demodulation.
3. With neat diagram explain the generation of PWM.
4. Compare FDM and TDM.
5. Draw the block diagram and explain PCM generation and detection
technique.
6. Derive the expression for quantization error.
7. Explain companding techniques in PCM.
8. What is the difference between pulse code modulation and delta
modulation?
9. Explain in detail about granular noise and slope overload noise in
delta modulation.
10. Explain the operation of the adaptive delta modulation with the help
of block diagram.

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Digital Modulation
Techniques
Digital-to-Analog signals is the next conversion we will discuss in this
chapter. These techniques are also called as Digital Modulation
techniques.

Digital Modulation provides more information capacity, high data

security, quicker system availability with great quality communication.
Hence, digital modulation techniques have a greater demand, for their
capacity to convey larger amounts of data than analog modulation
techniques.

There are many types of digital modulation techniques and also their
combinations, depending upon the need. Of them all, we will discuss the
prominent ones.

ASK – Amplitude Shift Keying

The amplitude of the resultant output depends upon the input data whether
it should be a zero level or a variation of positive and negative, depending
upon the carrier frequency.

FSK – Frequency Shift Keying

The frequency of the output signal will be either high or low, depending
upon the input data applied.

PSK – Phase Shift Keying

The phase of the output signal gets shifted depending upon the input.
These are mainly of two types, namely Binary Phase Shift
Keying BPSK and Quadrature Phase Shift Keying QPSK, according to the

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number of phase shifts. The other one is Differential Phase Shift

Keying DPSK which changes the phase according to the previous value.

M-ary Encoding

M-ary Encoding techniques are the methods where more than two bits are
made to transmit simultaneously on a single signal. This helps in the
reduction of bandwidth.

The types of M-ary techniques are –

 M-ary ASK
 M-ary FSK
 M-ary PSK

All of these are discussed in subsequent chapters.

5.1 Amplitude Shift Keying

Amplitude Shift Keying ASK is a type of Amplitude Modulation which

represents the binary data in the form of variations in the amplitude of a
signal.

Any modulated signal has a high frequency carrier. The binary signal
when ASK modulated, gives a zero value for Low input while it gives
the carrier output for High input.

The following figure represents ASK modulated waveform along with its
input.

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Figure 5.1: Grahical representation of amplitude shift keying signal

To find the process of obtaining this ASK modulated wave, let us learn
about the working of the ASK modulator.

5.1.1 ASK Modulator

The ASK modulator block diagram comprises of the carrier signal

generator, the binary sequence from the message signal and the band-
limited filter. Following is the block diagram of the ASK Modulator.

Figure 5.2: Block diagram of amplitude shift keying signal modulator

The carrier generator sends a continuous high-frequency carrier. The

binary sequence from the message signal makes the unipolar input to be
either High or Low. The high signal closes the switch, allowing a carrier
wave. Hence, the output will be the carrier signal at high input. When

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there is low input, the switch opens, allowing no voltage to appear. Hence,
the output will be low, ASK Wave and Representation is shown below fig
5.3.

Figure 5.3: Amplitude shift keying modulator signal represntation

The band-limiting filter, shapes the pulse depending upon the amplitude
and phase characteristics of the band-limiting filter or the pulse-shaping
filter.

5.1.2 ASK Demodulator

There are two types of ASK Demodulation techniques. They are –

 Asynchronous ASK Demodulation/detection

 Synchronous ASK Demodulation/detection

The clock frequency at the transmitter when matches with the clock
frequency at the receiver, it is known as a Synchronous method, as the
frequency gets synchronized. Otherwise, it is known as Asynchronous.

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Asynchronous Amplitude Shift Keying Demodulator

The Asynchronous ASK detector consists of a half-wave rectifier, a low

pass filter, and a comparator. Following is the block diagram for the same.

Figure 5.4: Asynchronous amplitude shift keying detector

The modulated ASK signal is given to the half-wave rectifier, which

delivers a positive half output. The low pass filter suppresses the higher
frequencies and gives an envelope detected output from which the
comparator delivers a digital output.

Synchronous ASK Demodulator

Synchronous ASK detector consists of a Square law detector, low pass
filter, a comparator, and a voltage limiter. Following is the block diagram
for the same.

Figure 5.5: Synchronous Amplitude Shift Keying Detector

The ASK modulated input signal is given to the Square law detector. A
square law detector is one whose output voltage is proportional to the
square of the amplitude modulated input voltage. The low pass filter
minimizes the higher frequencies. The comparator and the voltage limiter
help to get a clean digital output.

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ASK Applications

Modulation has an important role in communications. And amplitude shift

keying applications are mentioned below. They are:

 Low-frequency RF applications
 Home automation devices
 Industrial networks devices
 Wireless base stations
 Tire pressuring monitoring systems

Thus, Ask (amplitude shift keying) is a digital modulation technique to

increase the amplitude characteristics of the input binary signal. But its
drawbacks make it so limited. And these drawbacks can be overcome by
the other modulation technique which is FSK.

5.2 Frequency Shift Keying

Frequency Shift Keying (FSK) is the digital modulation technique in

which the frequency of the carrier signal varies according to the digital
signal changes. FSK is a scheme of frequency modulation.

The output of a FSK modulated wave is high in frequency for a binary

High input and is low in frequency for a binary Low input. The
binary 1s and 0s are called Mark and Space frequencies.

The following image is the diagrammatic representation of FSK modulated

waveform along with its input.

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Figure 5.6: Frequency Shift Keying Modulated output with

Representation

To find the process of obtaining this FSK modulated wave, let us know
about the working of a FSK modulator.

5.2.1 FSK Modulator

The FSK modulator block diagram comprises of two oscillators with a

clock and the input binary sequence. Following is its block diagram.

Figure 5.7: Generation of frequency shift keying signal

The two oscillators, producing a higher and a lower frequency signals, are
connected to a switch along with an internal clock. To avoid the abrupt
phase discontinuities of the output waveform during the transmission of
the message, a clock is applied to both the oscillators, internally. The
binary input sequence is applied to the transmitter so as to choose the
frequencies according to the binary input.

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5.2.2 FSK Demodulator

There are different methods for demodulating a FSK wave. The main
methods of FSK detection are asynchronous detector and synchronous
detector. The synchronous detector is a coherent one, while asynchronous
detector is a non-coherent one.

Asynchronous Frequency Shift Keying Detector

The block diagram of Asynchronous FSK detector consists of two band

pass filters, two envelope detectors, and a decision circuit. Following is the
diagrammatic representation.

Figure 5.8: Detection of asynchronous frequency shift keying signal

The FSK signal is passed through the two Band Pass Filters BPFs, tuned
to Space and Mark frequencies. The output from these two BPFs look like
ASK signal, which is given to the envelope detector. The signal in each
envelope detector is modulated asynchronously.

The decision circuit chooses which output is more likely and selects it
from any one of the envelope detectors. It also re-shapes the waveform to
a rectangular one.

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Synchronous Frequency Shift Keying Detector

The block diagram of Synchronous FSK detector consists of two mixers

with local oscillator circuits, two band pass filters and a decision circuit.
Following is the diagrammatic representation.

Figure 5.9: Detection of synchronous frequency shift keying signal

The FSK signal input is given to the two mixers with local oscillator
circuits. These two are connected to two band pass filters. These
combinations act as demodulators and the decision circuit chooses which
output is more likely and selects it from any one of the detectors. The two
signals have a minimum frequency separation.

For both of the demodulators, the bandwidth of each of them depends on

their bit rate. This synchronous demodulator is a bit complex than
asynchronous type demodulators.

5.3 Phase Shift Keying

Phase Shift Keying (PSK) is the digital modulation technique in which the
phase of the carrier signal is changed by varying the sine and cosine inputs
at a particular time. PSK technique is widely used for wireless LANs, bio-
metric, contactless operations, along with RFID and Bluetooth
communications.

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PSK is of two types, depending upon the phases the signal gets shifted.
They are −

Binary Phase Shift Keying

This is also called as 2-phase PSK or Phase Reversal Keying. In this

technique, the sine wave carrier takes two phase reversals such as 0° and
180°.

Figure 5.10: Representation of Phase Shift Keying Signal

BPSK is basically a Double Side Band Suppressed Carrier DSBSC

modulation scheme, for message being the digital information.

5.3.1 Binary Phase Shift Keying (BPSK) Modulator

The block diagram of Binary Phase Shift Keying consists of the balance
modulator which has the carrier sine wave as one input and the binary
sequence as the other input. Following is the diagrammatic representation.

Figure 5.11: Generation of Phase Shift Keying Signal

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The modulation of BPSK is done using a balance modulator, which

multiplies the two signals applied at the input. For a zero binary input, the
phase will be 0° and for a high input, the phase reversal is of 180°.

Following is the diagrammatic representation of BPSK Modulated output

wave along with its given input.

Figure 5.12: Binary phase shift keying (BPSK) modulated output wave

The output sine wave of the modulator will be the direct input carrier or
the inverted 180°phaseshifted180°phaseshifted input carrier, which is a
function of the data signal.

5.3.2 Binary Phase Shift Keying (BPSK) Demodulator

The block diagram of BPSK demodulator consists of a mixer with local

oscillator circuit, a band pass filter, a two-input detector circuit. The
diagram is as follows.

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Figure 5.13: Detection of Binary Phase Shift Keying (BPSK) Signal

By recovering the band-limited message signal, with the help of the mixer
circuit and the band pass filter, the first stage of demodulation gets
completed. The base band signal which is band limited is obtained and this
signal is used to regenerate the binary message bit stream.

In the next stage of demodulation, the bit clock rate is needed at the
detector circuit to produce the original binary message signal. If the bit
rate is a sub-multiple of the carrier frequency, then the bit clock
regeneration is simplified. To make the circuit easily understandable, a
decision-making circuit may also be inserted at the 2nd stage of detection.

5.4 Quadrature Phase Shift Keying

The Quadrature Phase Shift Keying QPSK is a variation of BPSK, and it

is also a Double Side Band Suppressed Carrier DSBSC modulation
scheme, which sends two bits of digital information at a time, called
as Binary digits. Instead of the conversion of digital bits into a series of
digital stream, it converts them into bit pairs. This decreases the data bit
rate to half, which allows space for the other users. This is the phase shift
keying technique, in which the sine wave carrier takes four phase reversals
such as 0°, 90°, 180°, and 270°. If this kind of techniques are further
extended, PSK can be done by eight or sixteen values also, depending
upon the requirement.

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5.4.1 QPSK Modulator

The QPSK Modulator uses a bit-splitter, two multipliers with local

oscillator, a 2-bit serial to parallel converter, and a summer circuit.
Following is the block diagram for the same.

Figure 5.14: Generation of Quadrature Phase Shift Keying (QPSK) Signal

At the modulator’s input, the message signal’s even bits (i.e., 2 nd bit,
4th bit, 6th bit, etc.) and odd bits (i.e., 1st bit, 3rd bit, 5th bit, etc.) are
separated by the bits splitter and are multiplied with the same carrier to
generate odd BPSK (called as PSKI) and even BPSK (called as PSKQ).
The PSKQ signal is anyhow phase shifted by 90° before being modulated.
The QPSK waveform for two-bits input is as follows, which shows the
modulated result for different instances of binary inputs.

Figure 5.15: Quadrature Phase Shift Keying (QPSK) Modulated Wave

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5.4.2 Quadrature Phase Shift Keying Demodulator

The QPSK Demodulator uses two product demodulator circuits with local
oscillator, two band pass filters, two integrator circuits, and a 2-bit parallel
to serial converter. Following is the diagram for the same.

Figure 5.16: Detection of Quadrature Phase Shift Keying (QPSK) Signal

The two product detectors at the input of demodulator simultaneously

demodulate the two BPSK signals. The pair of bits are recovered here
from the original data. These signals after processing are passed to the
parallel to serial converter.

5.5 Differential Phase Shift Keying

In Differential Phase Shift Keying DPSK the phase of the modulated

signal is shifted relative to the previous signal element. No reference
signal is considered here. The signal phase follows the high or low state of
the previous element. This DPSK technique doesn’t need a reference
oscillator. The following figure represents the model waveform of DPSK.

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Figure 5.17: Representation of Quadrature Phase Shift Keying (QPSK)

Signal

It is seen from the above figure that, if the data bit is Low i.e., 0, then the
phase of the signal is not reversed, but continued as it was. If the data is a
High i.e., 1, then the phase of the signal is reversed, as with NRZI, invert
on 1 a form of differential encoding.

If we observe the above waveform, we can say that the High state
represents an M in the modulating signal and the Low state represents
a W in the modulating signal.

5.5.1 Differential Phase Shift Keying Modulator

DPSK is a technique of BPSK, in which there is no reference phase signal.

Here, the transmitted signal itself can be used as a reference signal.
Following is the diagram of DPSK Modulator.

Figure 5.18: Generation of Differential Phase Shift Keying (DPSK)

Signal

DPSK encodes two distinct signals, i.e., the carrier and the modulating
signal with 180° phase shift each. The serial data input is given to the
XNOR gate and the output is again fed back to the other input through 1-
bit delay. The output of the XNOR gate along with the carrier signal is
given to the balance modulator, to produce the DPSK modulated signal.

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5.5.2 Differential Phase Shift Keying Demodulator

In DPSK demodulator, the phase of the reversed bit is compared with the
phase of the previous bit. Following is the block diagram of DPSK
demodulator.

Figure 5.19: Detection of Differential Phase Shift Keying (DPSK) Signal

From the above figure, it is evident that the balance modulator is given the
DPSK signal along with 1-bit delay input. That signal is made to confine
to lower frequencies with the help of LPF. Then it is passed to a shaper
circuit, which is a comparator or a Schmitt trigger circuit, to recover the
original binary data as the output

5.6 M-ary Encoding

The word binary represents two bits. M represents a digit that corresponds
to the number of conditions, levels, or combinations possible for a given
number of binary variables.

This is the type of digital modulation technique used for data transmission
in which instead of one bit, two or more bits are transmitted at a time. As a
single signal is used for multiple bit transmission, the channel bandwidth
is reduced.

5.6.1 M-ary Equation

If a digital signal is given under four conditions, such as voltage levels,

frequencies, phases, and amplitude, then M = 4.The number of bits
necessary to produce a given number of conditions is expressed
mathematically as

N = log 2 M

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Where
N is the number of bits necessary
M is the number of conditions, levels, or combinations possible with N
bits.

The above equation can be re-arranged as

2N =M

For example, with two bits, 22 = 4 conditions are possible.ie 00,01,10,11.

5.6.2 Types of M-ary Techniques

In general, Multi-level M−ary modulation techniques are used in digital

communications as the digital inputs with more than two modulation
levels are allowed on the transmitter’s input. Hence, these techniques are
bandwidth efficient.

There are many M-ary modulation techniques. Some of these techniques,

modulate one parameter of the carrier signal, such as amplitude, phase,
and frequency.

M-ary ASK

This is called M-ary Amplitude Shift Keying M−ASK or M-ary Pulse

Amplitude Modulation PAM. The amplitude of the carrier signal, takes
on M different levels.

Representation of M-ary ASK

Sm (t) = Am Cos (2πfct) Amϵ (2m−1−M) Δ, m=1,2....M and 0 ≤ t ≤

Some prominent features of M-ary ASK and This method is also used in
PAM.

 Its implementation is simple.

 CM-ary ASK is susceptible to noise and distortion.

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M-ary FSK

This is called as M-ary Frequency Shift Keying M−ary FSK.

The frequency of the carrier signal, takes on M different levels.

Representation of M-ary FSK

Si (t) = 2EsTs√cos [π Ts (n c + i)t] 0 ≤ t ≤ Ts and i=1,2,3.....M

Where fc=nc2Ts for some fixed integer n.

Some prominent features of M-ary FSK are –

 Not susceptible to noise as much as ASK.

 The transmitted M number of signals are equal in energy and
duration.
 The signals are separated by 12Ts12Ts Hz making the signals
orthogonal to each other.
 Since M signals are orthogonal, there is no crowding in the signal
space.
 The bandwidth efficiency of M-ary FSK decreases and the power
efficiency increases with the increase in M.

M-ary PSK

This is called as M-ary Phase Shift Keying M−ary PSK.

The phase of the carrier signal, takes on M different levels.
Representation of M-ary PSK
Si(t) = 2Et√cos(wot+ϕit) 0 ≤ t ≤ T and i=1,2...M
ϕi (t) = 2πi M where i=1,2,3......M

Some prominent features of M-ary PSK are –

 The envelope is constant with more phase possibilities.

 This method was used during the early days of space
communication.
 Better performance than ASK and FSK.

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 Minimal phase estimation error at the receiver.

 The bandwidth efficiency of M-ary PSK decreases and the power
efficiency increases with the increase in M.

So far, we have discussed different modulation techniques. The output of

all these techniques is a binary sequence, represented as 1s and 0s. This
binary or digital information has many types and forms, which are
discussed further.

5.6.3 Probability of Error and the Distance between Signals

Table 5.1: Probability of error differences

These expressions illustrate the dependence of the error probability on the

distance between two signal points. In general, we can write as,

5.6.4 Quadrature Phase Shift Keying (QPSK)

It is a type linear modulation scheme, The phase of the carrier takes on 1

of 4 equally spaced values, such as 0, /2, , and 3/2 where each value of
phase corresponds to a unique pair of message bits.The QPSK Signal for
this set of symbol states may be defined as:

The above equation gives transmitter design.

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5.6.5 QPSK Waveform and Transmitter Design

Figure 5.20: QPSK Waveform and Transmitter Design

5.6.6 QPSK Constellation Diagram

Quadrature Phase Shift Keying has twice the bandwidth efficiency of

BPSK since 2 bits are transmitted in a single modulation symbol.

Figure 5.21: QPSK Constellation Diagram

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5.6.7 Quadrature Amplitude Modulation

Quadrature amplitude modulation (QAM) is modulation techniques that

we can utilize in analog modulation concept and digital modulation
concept. Depending upon the input signal form we can use it in either
analog or digital modulation schemes. In QAM, we can modulate two
individual signals and transmitted to the receiver level. And by using the
two input signals, the channel bandwidth also increases. QAM can able to
transmit two message signals over the same channel. This QAM technique
also is known as “quadrature carrier multiplexing”.

Quadrature Amplitude Modulation Definition

QAM can be defined as it is s a modulation technique that is used to

combine two amplitude modulated waves into a single channel to increase
the channel bandwidth.

5.6.8 Quadrature Amplitude Modulation Block Diagram

The below diagrams show the transmitter and receiver block diagram of
the QAM scheme.

QAM Modulator

Figure 5.21: Generation of quadrature amplitude modulation block

diagram

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QAM Demodulator

Figure 5.22: Detection of Quadrature Amplitude Modulation Block

Diagram

QAM Working Principle

“In the QAM transmitter, the above section i.e., product modulator1 and
local oscillator are called the in-phase channel and product modulator2 and
local oscillator are called a quadrature channel. Both output signals of the
in-phase channel and quadrature channel are summed so the resultant
output will be QAM.” The below waveforms are indicating the two
different carrier signals of the QAM technique.

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The output waveforms of QAM are shown below.

Figure 5.23: The output waveforms of QAM Signal

At the receiver level, the QAM signal is forwarded from the upper channel
of receiver and lower channel, and the resultant signals of product
modulators are forwarded from LPF1 and LPF2. These LPF’s are fixed to
the cut off frequencies of input 1 and input 2 signals. Then the filtered
outputs are the recovered original signals.

Advantages of QAM

The quadrature amplitude modulation advantages are listed below. They

are

 One of the best advantages of QAM – supports a high data rate. So,
the number of bits can be carried by the carrier signal. Because of
these advantages it preferable in wireless communication networks.
 QAM’s noise immunity is very high. Due to this noise interference
is very less.
 It has a low probability of error value.
 QAM expertly uses channel bandwidth.

Quadrature Amplitude Modulation Applications

The applications of QAM include the following.

 The applications of QAM are mostly observed in radio

communications and data delivery applications systems.

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 QAM technique has wide applications in the radio communications

field because, as the increment of the data rate there is the chance of
noise increment but this QAM technique is not affected by noise
interference hence there is an easy mode of signal transmission can
be possible with this QAM.
 QAM has wide applications in transmitting digital signals like
digital cable television and in internet services.
 In cellular technology, wireless device technology quadrature
amplitude modulation is preferred.

5.7 Baseband Receiver or Integrate & Dump Filter

Figure 5.24: Integrate & Dump Filter

 Here the signal x(t) is distorted by noise and applied to the input of
the integrate and dump filter
 The capacitor is discharged fully at the beginning of the bit interval
(by closing sw1 switch), the integrator then integrates the noisy
input signal over one bit period

(a) Input pulse to the integrator (assuming that thr noise is absent). This pulse represents
binary 1 the width of the pulse is T
(b) Output of Integrator.The initial output is zero.At t=T, the out put
Of Integrator is r(t) = AT

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5.7.1 Signal to Noise Ratio for integrate and Dump filter

i ) Output signal power

The out of the integrator can be written as ,

r(t) = (t) +n(t)]dt

r(t) = (t)dt + ]dt

r(t) = x0 (t)+ n0(t)

x0 (t) =

=
ii) Transfer function of integrator:

A network that performs integration has transfer function of .

There is a delay at t =T. Hence transfer function of integrator.

H(f) =

H(f) = -j

H(f) =

H(f) =

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iii) Output noise power

Input noise has psd of white noise.i.e. sni (f) = . Hence psd of output
noise will be,

Power, P =

Mean value of noise power, (no2)=

2
Sno(f) = sni (f)

Here H(f) is Transfer function of filter,

Sno(f) is psd of output noise and sni (f) is psd of input noise

sni (f) =

2
Sno(f) =

Mean value of noise power, (no2) =

We know that H(f) =

Mean value of noise power, (no2) =

Put = x, then dx = df , df = dx

now we get above equation becomes

Mean value of noise power, (no2) = dx

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iv) Signal to Noise ratio (S/N) =

= = =

= or

5.7.2 Probability of Error and Calculation

In statistics, the term "error" arises in two ways. Firstly, it arises in the
context of decision making, where the probability of error may be
considered as being the probability of making a wrong decision and which
would have a different value for each type of error.

Secondly, it arises in the context of statistical modeling (for example

regression) where the model's predicted value may be in error regarding
the observed outcome and where the term probability of error may refer
to the probabilities of various amounts of error occurring.

Step1: Find signal energy per bit Eb or E =

Step 2: Find “No” form given psd data, as PSD =

Step 3: Apply the formulae to get probability of error
Master formula for all modulation techniques (coherent)

Pe = erfc Where Ed is a bit energy difference

Pe = erfc where = (t)dt

P(t) = x01(t) – x02(t)

Probability of error for ASK

Pe = erfc

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= (t)dt

P(t) = x01(t) – x02(t)

1 A Cos Wc t = x01(t)

0 0 = x02(t)

P(t) = A Cos Wc t – 0 = A Cos Wc t

= (t)dt

= Wct dt = Wct dt

= dt Since Wct =

= dt

= here limits are 0 to T

= Since

Pe = erfc ince A2T/2 = E and Wc is very high

Pe=

Probability of error for FSK

Pe = erfc

= (t)dt

P(t) = x01(t) – x02(t)

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1 A Cos( W + )t = x01(t)

0 A Cos( W- )t = x02(t)

P(t) = A Cos( W + )t – A Cos( W- )t

= -2Cos dt

= , =

= + ,

dt =

dt =

Using above relations value becomes

= - - }

Here carrier frequency is high i.e. w=0

= {T- } Maximum at T = 3 /2

= {2.42}

Pe = erfc{ (2.42)}

Pe = erfc{ } Since A2/2 = E and Wc T

Pe =

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Probability of error for PSK

Pe = erfc

= (t)dt

PSK Signal can be written as

P(t) = x01(t) – x02(t)

1 A Cos Wct = x01(t)

0 - A Cos Wct = x02(t)

P(t) = A Cos Wct – (- A Cos Wct)

P(t) = 2A Cos Wct

= dt

= dt

= dt

= ) dt

= }

Here limits range is 0 and T and Wc is very high

Sin2Wct range [-1,1] then it is zero using value in probability error

equation the we get

Pe = erfc we know that = T

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Pe = erfc (Since = E)

` Pe = erfc

5.8 Optimum Receiver

Optimal Receiver: An optimal receiver is one that is designed to

minimize the probability that a decision error occurs. There exists no
other receiver structure that can provide a lower probability of error.

An optimum filter is such a filter used for acquiring a best estimate of

desired signal from noisy measurement. It is different from the
classic filters like low pass, high pass and band pass filters

It is a generized filter for receiving binary coded signals

Figure 5.25: Optimum Receiver

y (t) = x01(t)+n(t) ; x1(t)

y (t) = x02(t)+n(t) ; x2(t)

Decision boundary gives as

P -

fx [(n0(t)] = /2

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pe =

Pe = dno(t)

= y=

dy = dn0(t) = dy ; = ; and y =

= y=

Pe = dy

Pe = dy

Pe = dy

Now error function (erfc) = dy

Pe = erfc

Here erfc value is decreases as is increases

i.e. erfc

Optimal Receiver: An optimal receiver is one that is designed to

minimize the probability that a decision error occurs. There exists no
other receiver structure that can provide a lower probability of error.

5.9 Coherent Detection

A “coherent” optical transmission system is characterized by its capability

to do “coherent detection,” which means that an optical receiver can track

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the phase of an optical transmitter (and hence “phase coherence”) so as to

extract any phase and frequency information carried by a transmitted
signal.Coherent detection originates from radio communications, where a
local carrier mixes with the received radio frequency (RF) signal to
generate a product term. As a result, the received RF signal can be
frequency translated and demodulated.

A block diagram of coherent detection is shown in Fig. 5.26. In this

circuit, the received RF signal m(t) cos(ωsct) has an information-carrying
amplitude m(t) and an RF carrier at frequency ωsc, whereas the LO has a
single frequency at ωloc. The RF signal multiplies with the LO in an RF
mixer, generating the sum and the difference frequencies between the
signal and the LO. The process can be described by the following
equation:

Figure 5.26: block diagram of coherent detection

A low-pass filter is usually used to eliminate the sum frequency

component and thus the baseband signal can be recovered if the frequency
of the LO is equal to that of the signal (ωloc = ωsc). When the RF signal has
multiple and very closely spaced frequency channels, excellent frequency
selection in coherent detection can be accomplished by fine tuning the
frequency of the LO. This technique has been used in ratio
communications for many years, and high-quality RF components such as
oscillators, RF mixers, and amplifiers are standardized.

5.10 Inter symbol Interference

Generally, digital data is represented by electrical pulse, communication

channel is always band limited. Such a channel disperses or spreads a
pulse carrying digitized samples passing through it. When the channel
bandwidth is greater than bandwidth of pulse, spreading of pulse is very
less.

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Transmitted Waveform Pulse Dispersion

Transmitted Waveform Pulse Dispersion

Figure 5.27: Inter symbol Interference for Transmitted signal

But when channel bandwidth is close to signal bandwidth, i.e. if we

transmit digital data which demands more bandwidth which exceeds
channel bandwidth, spreading will occur and cause signal pulses to
overlap. This overlapping is called Inter Symbol Interference.In short it
is called ISI. Similar to interference caused by other sources, ISI causes
degradations of signal if left uncontrolled. This problem of ISI exists
strongly in Telephone channels like coaxial cables and optical fibers.

The main objective is to study the effect of ISI, when digital data is
transmitted through band limited channel and solution to overcome the
degradation of waveform by properly shaping pulse

The effect of sequence of pulses transmitted through channel is shown in

fig. The Spreading of pulse is greater than symbol duration, as a result
adjacent pulses interfere. i.e. pulses get completely smeared, tail of
smeared pulse enter into adjacent symbol intervals making it difficult to
decide actual transmitted pulse. First let us have look at different formats
of transmitting digital data. In base band transmission best way is to map
digits or symbols into pulse waveform. This waveform is generally termed
as Line codes.

5.11 The Eye Diagram

In telecommunication, an eye pattern, also known as an eye diagram, is

an oscilloscope display in which a digital signal from a receiver is
repetitively sampled and applied to the vertical input, while the data rate is
used to trigger the horizontal sweep. ... An open eye pattern corresponds
to minimal signal distortion.

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Several system performance measures can be derived by analyzing the

display. If the signals are too long, too short, poorly synchronized with the
system clock, too high, too low, too noisy, or too slow to change, or have
too much undershoot or overshoot, this can be observed from the eye
diagram.

Figure 5.28: Graphical representation of eye diagram

An open eye pattern corresponds to minimal signal distortion. Distortion

of the signal waveform due to inter symbol interference and noise appears
as closure of the eye pattern

Short Answer Questions

1. Draw the spectrum of BFSK.

2. What are the applications of ASK.
3. What is correlative coding?
4. What is an eye diagram?
5. Write differences between base band transmissions and pass band
transmissions.
6. Define probability of error?
7. Write the differences between ASK, PSK, FSK, QPSK

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Long Answer Questions

1. Explain ASK modulation and demodulation with neat block

diagram.
2. With neat diagrams and equations explain about PSK system
3. Explain in detail the generation and detection of BPSK system.
Derive the expression for its bit error probability.
4. Explain with neat diagrams coherent BFSK transmitter and
receiver. Explain the signal space diagram for coherent BFSK
system.
5. What is the need of digital modulation in digital communication?
Explain QAM modulation and demodulation.
6. What is the principle of QPSK? Generation and detection of QPSK.
7. Draw the signal space diagram of ASK.
8. What is an eye diagram? Describe how eye pattern is helpful to
obtain the performance of the system in detail with a neat sketch.
9. Explain correlation receiver with block diagram. Also explain why
the correlation receiver is also called an integrated and dump filter.
10. What is the expression for baud rate of BPSK system?

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