1.

Introduction to Electronic
Communication
Electromagnetic spectrum:
Block diagram of Analog communication system:

The elements of basic analog communication system are input signal or information, input
transducer, transmitter, channel, Noise, Receiver, Output transducer.

1.Information or Input signal:

● The information is transmitted from one place to another.
● This information can be in the form of a sound signal like speech, or it can be in the form of
pictures or it can be in the form of data information.
2.Input transducer:
● The information in the form of sound, picture or data signals cannot be transmitted as it is.
● First it has to be converted into a suitable electrical signal.
● The input transducer block does this job.
● The input transducer commonly used are microphones, TV etc.
3.Transmitter:
● The function of the transmitter is to convert the electrical equivalent of the information to
a suitable form so that it can transfer over long distance.
● Functionality of a transmitter is decided by the type of channel being used.
● Basic block in transmitter are: Amplifier, Oscillator, Mixer.
4.Channel:
● The communication channel is the medium used for transmission of electrical signal from
one place to other.
● The communication medium can be conducting wires, cables, optical fibres or free space.
● Depending on the type of communication medium, two types of communication system
exists.
● Line communication: The line communication systems use the communication medium like
the simple wires or cables or optical fibres. Eg: Telephone, Cable TV.
● Radio communication: The radio communication systems use the free space as their
communication medium. The transmitted signal is in the form of electromagnetic waves.
E.g. Mobile communication, satellite communication.
5.Noise:
● Noise is an unwanted electrical signal which gets added to the transmitted signal when it is
travelling towards the receiver.
● Due to noise quality of information gets degrade.
● Once added the noise cannot be separated out from the information
6.Receiver:

● Receiver processes the received signal to recreate the original form of message signal.
● Construction of a receiver depends on the transmitter type.
● Most of the receivers conform to the super heterodyne type.
● The receiver consists of Amplifier, Oscillator, Mixer.

7.Output transducer:
● Output transducer converts electrical signal into the original form i.e. sound or TV pictures
etc.
● E.g. Loudspeaker, data and image convertor.
Modulation:
Introduction:

● A message carrying a signal has to get transmitted over a distance and for it to establish a
reliable communication, it needs to take the help of a high frequency signal which should
not affect the original characteristics of the message signal.
● The characteristics of the message signal, if changed, the message contained in it also
alters. Hence, it is a must to take care of the message signal.
● A high frequency signal can travel up to a longer distance, without getting affected by
external disturbances. We take the help of such high frequency signal which is called as
a carrier signal to transmit our message signal. Such a process is simply called as
Modulation.
● Modulation is the process of changing the parameters of the carrier signal, in accordance
with the instantaneous values of the modulating signal.

Need for Modulation:

● 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.
● Transmitting and receiving antenna lengths are comparable to a quarter-wavelength of the
frequency used
● Modulation is needed to avoid mixing of signals.
● Because of modulation more number of message signals can be accommodated at higher
frequencies.

Advantages of Modulation

● Reduction of antenna size

● No signal mixing
● Increased communication range
● Multiplexing of signals
● Possibility of bandwidth adjustments
● Improved reception quality

Signals in the Modulation Process:

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

1. Message or Modulating Signal:

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

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

2. Carrier Signal:

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

3. Modulated Signal:

The resultant signal after the process of modulation is called as a modulated signal. This
signal is a combination of modulating signal and carrier signal.

Types of Modulation:

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

In continuous-wave modulation, a high frequency sine wave is used as a carrier wave. This is
further divided into amplitude and angle modulation.
● If the amplitude of the high frequency carrier wave is varied in accordance with the
instantaneous amplitude of the modulating signal, then such a technique is called
as Amplitude Modulation.
● If the angle of the carrier wave is varied, in accordance with the instantaneous value of the
modulating signal, then such a technique is called as Angle Modulation. Angle modulation
is further divided into frequency modulation and phase modulation.
o 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.
o If the phase of the high frequency carrier wave is varied in accordance with the
instantaneous value of the modulating signal, then such a technique is called
as Phase Modulation.

Pulse Modulation:

In Pulse modulation, a periodic sequence of rectangular pulses, is used as a carrier wave. This is
further divided into analog and digital modulation.
In analog modulation technique, if the amplitude or duration or position of a pulse is varied in
accordance with the instantaneous values of the baseband modulating signal, then such a
technique is called as Pulse Amplitude Modulation (PAM) or Pulse Duration/Width Modulation
(PDM/PWM), or Pulse Position Modulation (PPM).
In digital modulation, the modulation technique used is Pulse Code Modulation (PCM) where the
analog signal is converted into digital form of 1s and 0s. As the resultant is a coded pulse train,
this is called as PCM. This is further developed as Delta Modulation (DM). These digital
modulation techniques are discussed in our Digital Communications tutorial

Noise:
 ● Any undesirable electrical energy that falls within the pass band of the signal and interferes
with the proper reception and reproduction of the transmitted signals.
● E.g. In audio recording, any unwanted electrical signals that fall within audio frequency
band of 0Hz to 15kHz will interfere with the music.
● Noise contains a multitude of frequencies and amplitudes that can interfere with the
quality of signal.
● Noise affects the sensitivity of receivers.
● Limits the range of systems for a given transmitted power and sometimes bandwidth

Classification of Noise:
Noise is broadly classified into 2 types:
1. External Noise
2. Internal Noise

External noise: Noise generated outside the device or circuit is called External noise.
Types of External noise:
1. Atmospheric noise (static electricity)
2. Extra-terrestrial noise
3. Man-made noise (Industrial noise)

Atmospheric noise:
– Naturally occurring electrical disturbances that originate within Earth’s atmosphere.
– E.g. Sputtering, cracking etc heard from the speaker when there is no signal present
– Often in the form of impulses that spread energy throughout a wide range of frequencies
– Source: lightning
– At f >30MHz atmospheric noise is insignificant

Extra-terrestrial noise: (deep-space noise)

– Consists of electrical signals that originate from outside Earth’s atmosphere.
– Sources: Milky way, other galaxies and the sun.
– Extra-terrestrial noise is further divided into 2 types:
1. Solar noise:
✔ Generated directly from sun’s heat
✔ Quiet condition
✔ High intensity, sporadic disturbances: sunspot activity and solar flare-ups
2. Cosmic noise (black body noise):

✔ Sources continuously distribute throughout the galaxies.

✔ Since the sources are far away than our sun, their intensity is relatively small.

Man-made noise: (industrial noise)

– Produced by mankind
– Sources: spark producing mechanisms such as commutators in electric motors,
automobile ignition systems, ac power-generating and switching equipment,
fluorescent lights
– Impulsive in nature
– Intense in densely populated metropolitan and industrial areas
Internal noise: The electrical interference generated within a device or circuit
Types of Internal noise:
– Shot noise
– Transit time
– Thermal noise

Shot noise:
– Caused by the random arrival of carriers at the output element of an electronic device such
as diode, transistor etc.
– Sound similar to the metal pellets falling on a tin roof.
– Also called transistor noise and is additive with thermal noise

Transit-time noise:
– Any modification to a stream of carriers as they pass from input to the output of a device
produces an irregular, random variation categorized as transit-time noise.
– When the time it takes for the carriers to propagate through a device is an appreciable part
of the time of one cycle of signal, the noise becomes noticeable.

Thermal noise:
– Associated with the rapid and random movement of electrons within a device due to
thermal agitation.
– Also called Random noise, Johnson noise, Brownian noise
– Present in all electronic communication systems
– Uniformly distributed across the electromagnetic spectrum, hence called white noise.
– Additive in nature- cannot be eliminated
– Puts an upper bound on the performance of a communication system.
Terms:

Thermal noise power:

N= KTB
Where,
N=noise power (Watts)
B= bandwidth (Hz)
K= Boltzmann’s proportionality constant (1.38x10-23 J/K)
T= absolute temperature (Kelvin)
 NOTE: NUMERICALS IN BOOK
2. Analog Modulation
Modulation Index:
✔ Modulation index describes the extent to which modulation is done on a carrier signal.
✔ In an amplitude modulation, it is defined as the ratio of the amplitude of modulating signal
to that of the carrier signal.
✔ Mathematically, this is represented as:

✔ The value of the modulation index is kept less than 1 to avoid distortions in the modulated
signal because it is very hard to demodulate the signal.
✔ Modulation index is mostly represented in terms of percentage.
Amplitude Modulation:
– Amplitude modulation (AM) is one of the earliest modulation methods that is used in
transmitting information over the radio.
– In amplitude modulation, the amplitude of the carrier wave is varied in some proportion
with respect to the modulating data or the signal. Here, the voltage or the power level of
the information signal changes the amplitude of the carrier.
– In AM, the carrier does not vary in amplitude. However, the modulating data is in the form
of signal components consisting of frequencies either higher or lower than that of the
carrier.
– The signal components are known as sidebands and the sideband power is responsible for
the variations in the overall amplitude of the signal.
Types: -
1) Double sideband-suppressed carrier modulation: (DSB-SC)

● The transmitted wave consists of only the upper and lower sidebands
● But the channel bandwidth requirement is the same as before.

2) Single Sideband Modulation: (SSB)

● The modulation wave consists only of the upper sideband or the lower sideband.
● To translate the spectrum of the modulating signal to a new location in the frequency
domain.

3) Vestigial Sideband Modulation: (VSB)

● One sideband is passed almost completely and just a trace of the other sideband is
retained.
● The required channel bandwidth is slightly in excess of the message bandwidth by an
amount equal to the width of the vestigial sideband.

Advantages:
● Amplitude modulation is economical as well as easily obtainable
● It is so simple to implement, and by using a circuit with fewer components it can be
demodulated.
● The receivers of AM are inexpensive because it doesn’t require any specialized components.

Disadvantages:
● The efficiency of this modulation is very low because it uses a lot of power
● It uses amplitude frequency several times to modulate the signal by a carrier signal.
● This declines the original signal quality on the receiving end & causes troubles in the signal
quality.
● AM systems are susceptible toward the generation of noise generation.
● The applications of amplitude modulation limits to VHF, radios, & applicable one to one
communication only

Double sideband-suppressed carrier modulation:

(DSBSC):
● Double Sideband Suppressed Carrier (DSBSC) is an amplitude modulation technique in
which the modulated wave contains both the sidebands along with the suppressed carrier.
● Conventional AM consists of the two sidebands and a carrier where the major transmitted
power is concentrated in the carrier which contains no information. Thus to increase the
efficiency and to save power, the carrier is suppressed in DSBSC system.
● The DSBSC generation using balanced modulator based on nonlinear resistance
characteristics of diode.
● The diode in the balanced modulator use the nonlinear resistance property for producing
modulated signals.
● Carrier voltage is applied in phase at both the diodes, while modulating voltage appears
180° out of phase at the diode inputs as they are at opposite ends of a centre tapped
transformer.
● The modulated output currents of the two diodes are combined in the centre tapped
primary of the output transformer, which then gets subtracted.
● The output of the balanced modulator contains two sidebands and sum of the harmonic
components.
● The input voltage at diode D1 is (vc+vm) and input voltage at diode D2 is (vc−vm).
 (Derivation in book)
Single sideband modulation: (SSB):
The phasing method of SSB generation uses a phase shift technique that causes one of the side
bands to be concealed out. This system uses two balanced modulators M1 and M2 and two
90o phase shifting networks.

Working:
The message signal x(t) is applied to the product modulator M1 and through a 90o phase shifter to
the product modulator M2.
Hence, we get the Hilbert transform

at the output of the wideband 90o phase shifter.

The output of carrier oscillator is applied as it is to modulator M1 whereas it is passed through a
90o phase shifter and applied to the modulator M2.

The outputs of M1 and M2 are applied to an adder.

Therefore,

This expression represents the SSB signal with only LSB i.e. it rejects the USB.

Suppression of the upper sideband:

Here, the modulating and the carrier signals are applied to the upper balanced modulator
directly. Whereas, both these signals are 90o phase shifted and then applied to the lower
balanced modulator.
Advantages:
1. It can generate the SSB signal at any frequency, so the frequency up converter stage is not
required.
2. It can use the low audio frequencies as modulating signal. (In filter method, this is not
possible).
3. It is easy to switch from one sideband to other.

Disadvantages:
1. The main drawback is that the design of the 90o phase shifting network for the modulating
signal is extremely critical.
2. This network has to provide a correct phase shift of 90o at all the modulating frequencies
which is practically difficult to achieve.

(Derivation in book)
Bessel Functions:

Sum:
Difference between narrow band & wide band FM:

Noise Triangle in FM:

● Noise Triangle is unwanted deviation of the carrier frequency. Noise signal produces
amplitude and phase modulation in FM.
● The magnitude of this unwanted frequency deviation depends on the relative amplitude of
the noise with respect to the carrier.
● Let us consider the noise signal and carrier signal vectorially. When noise signal is
superimposed on carrier, the amplitude of noise is added with carrier amplitude.
● It produces amplitude modulation in FM. The maximum deviation of amplitude carrier =
Vn, as shown in below fig.

Fig1. Vector effect of noise on carrier

● At the same time, noise vector is constantly changing phase angle w.r.t carrier signal Vc
which will change the phase deviation (θ) of FM wave i.e. phase deviation θ of FM changes
due to noise. Thus, noise modulates the carrier in terms of amplitude as well as phase.
 ● From above figure, it is seen that noise in AM and PM remains constant for entire audio
range because the modulation index due to signal are independent on modulating
frequency but in FM the effect of noise will be increased with the increase in modulating
frequency.
● Thus, noise has more effect on higher frequencies in FM. The triangular distribution of
noise in FM is called as FM noise triangle.

Noise in FM before limiting:

Noise in FM after limiting:

Frequency Modulation Due to Noise:
Pre-Emphasis and De-Emphasis:
● Pre-emphasis: The noise suppression ability of FM decreases with the increase in the
frequencies. Thus increasing the relative strength or amplitude of the high frequency
components of the message signal before modulation is termed as Pre-emphasis.
 Pre-emphasis circuit
● De-emphasis: In the de-emphasis circuit, by reducing the amplitude level of the received
high frequency signal by the same amount as the increase in pre-emphasis is termed as
De-emphasis.

De-emphasis circuit
● The pre-emphasis process is done at the transmitter side, while the de-emphasis process is
done at the receiver side.
● Thus a high frequency modulating signal is emphasized or boosted in amplitude in
transmitter before modulation. To compensate for this boost, the high frequencies are
attenuated or de-emphasized in the receiver after the demodulation has been performed.
Due to pre-emphasis and de-emphasis, the S/N ratio at the output of receiver is maintained
constant.
● The de-emphasis process ensures that the high frequencies are returned to their original
relative level before amplification.
● Pre-emphasis circuit is a high pass filter or differentiator which allows high frequencies to
pass, whereas de-emphasis circuit is a low pass filter or integrator which allows only low
frequencies to pass.
Difference Between AM and FM
(For 10 marks)
Sr
. Parameter
AM FM
N s
o

1. Full form Amplitude modulation Frequency modulation

AM method of audio transmission FM radio was developed in the

2. Origin was successfully carried out in the United States in the 1930s by
mid-1870s. Edwin Armstrong.
In AM, a radio wave is known as the In FM, a radio wave is known as
Modulating "carrier" or "carrier wave" is the "carrier" or "carrier wave" is
3.
differences modulated in amplitude by the signal modulated in frequency by the
that is to be transmitted. signal that is to be transmitted.
The frequency and phase remain the
Constant same. The amplitude and phase remain
4.
parameters the same.

AM has poorer sound quality, and a

lower bandwidth but is cheaper and FM is less affected by interference,
can be transmitted over long but FM signals are impacted by
5. Quality distances as it has a lower physical barriers. They have a
bandwidth that is why it can hold better sound quality due to higher
more stations available in any bandwidth.
frequency range.

FM radio ranges in a higher

Frequency spectrum from 88.1 to 108.1MHz.
6. AM radio ranges from 535 to 1700
range or up to 1200 to 2400 bits per
kHz or up to 1200 bits per second.
second.

BW is large. Hence a wide channel

Bandwidth BW is much less than FM.
7. is required.
BW BW = 2 fm
BW = 2 x (δ + fm)
Bandwidth is less than FM or PM
Bandwidth requirement is greater
and doesn’t depend upon the
and depends upon the modulating.
modulation index.
Bandwidth requirement is twice the
Bandwidth requirement is twice the
sum of the modulating signal
Bandwidth highest modulating frequency.
8. frequency and the frequency
requirements In AM radio broadcasting, if the
deviation.
modulating signal has a bandwidth
of 15 kHz, then the bandwidth of an
Let’s say, if the frequency deviation
amplitude-modulated signal is 30
is 75kHz and the modulating signal
kHz.
 frequency is 15kHz, the bandwidth
required is 180kHz.

the number of sidebands having

No of The number of sidebands are
9. significant amplitude depends upon
Sidebands constant and equal to 2.
the modulation index
Zero
crossings in
10. Equidistant Not equidistant
modulating
signal
FM (or PM) transmitters are more
AM transmitters and receivers are complex than AM because the
less complex than FM and PM, but variation of modulating signal has
11. Complexity
synchronization is needed in the to be converted and detected from
case of SSBSC carriers. the corresponding variation in
frequencies.
AM receivers are very less
FM receivers are better immune to
susceptible to noise because noise
12. Noise noise and it is possible to decrease
affects the amplitude, which is where
noise by further deviation.
information is stored in AM signals.
Power is wasted in transmitting the All transmitted power is useful so
13. Efficiency
carrier. that’s why FM is very efficient.
MW (Medium wave), SW (short
Broadcasting FM, audio
14. Application wave) band broadcasting, video
transmission in T.V.
transmission in T.V.

(For 5 marks)
Difference Between AM and FM

Amplitude Modulation (AM) Frequency Modulation (FM)

The first successful audio transmission was Developed in 1930 by Edwin Armstrong, in the
carried out in the mid-1870s United States

The radio wave is called a carrier wave and the The radio wave is called a carrier wave, but the
frequency and phase remain the same amplitude and phase remain the same

Has poor sound quality, but can transmit longer

Has higher bandwidth with better sound quality
distance

The frequency range of AM radio varies from The frequency range of FM is 88 to 108 MHz in
535 to 1705 kHz the higher spectrum

More susceptible to noise Less susceptible to noise

Difference Between AM, FM, and PM:
Sr Parameter
FM AM PM
No s

The amplitude The phase modulation

Frequency modulation
modulation is a is a technique of
is a technique of
technique of modulation modulation in which
modulation, in which
in which the amplitude phase of the carrier
frequency of carrier
of the carrier wave wave varies in
1. Definition varies in accordance
varies in accordance accordance with the
with the amplitude of
with the amplitude of amplitude of the
modulating signal.
the modulating signal. modulating signal.
Keeping amplitude and
Keeping frequency and Keeping amplitude and
phase constant.
phase constant. frequency constant.

Noise immunity of FM
AM receivers are very Noise immunity better
2. Noise is superior to AM and
susceptible to noise than AM but not FM
PM

The frequency of the The amplitude of a

A phase of the carrier
carrier wave deviates carrier wave in AM
wave varies as per the
3. Function as per the voltage of diverges as per
voltage of modulating
the modulating signal amplitude or voltage of
signal input.
input. modulating signal input

Constant The amplitude of the The frequency of the The amplitude of the
4. parameter carrier wave is kept carrier wave is kept carrier wave is kept
changeless. invariable. changeless.

Digital PM types:
QPSK, BPSK, QAM
Digital FM types: FSK, AM types: DSB-SC,
5. Types (the combination of
GFSK, offset PSK SSB, VSB, etc.
amplitude and phase,
modulation).

6. Waveforms
 3. Radio Transmitters and Receivers
AM Transmitters:
● Transmitters that transmit AM signals are known as AM transmitters. These transmitters
are used in medium wave (MW) and short wave (SW) frequency bands for AM broadcast.

● The two types of AM transmitters that are used based on their transmitting powers are:
1. High Level
2. Low Level

● High level transmitters use high level modulation, and low level transmitters use low level
modulation. The choice between the two modulation schemes depends on the
transmitting power of the AM transmitter.

● In broadcast transmitters, where the transmitting power may be of the order of kilowatts,
high level modulation is employed.

● In low power transmitters, where only a few watts of transmitting power are required, low
level modulation is used.

● The basic difference between the two transmitters is the power amplification of the carrier
and modulating signals.
High-Level and Low-Level Transmitters:
● Both transmitters have a stable RF source and buffer amplifiers followed by RF power
amplifiers.

● In both types of transmitters, the audio voltage is processed, or filtered, so as to occupy the
correct bandwidth, and compressed somewhat to reduce the ratio of maximum to
minimum amplitude.

● In both modulation systems audio and power audio frequency (AF) amplifiers are present,
culminating in the modulator amplifier, which is the highest-power audio amplifier.

● The only difference is that the amplifier after the modulated RF amplifier is for low-level
modulation, and it must be a linear RF amplifier (class B).

● Remember that this would also have been called low-level modulation if the modulated
amplifier had been the final one, modulated at any electrode other than the collector.

● It follows that the higher the level of modulation, the larger the audio power required to
produce modulation.

● All these stages must be capable of handling amplitude variations caused by the
modulation. Such stages must be class A and consequently are less efficient than class C
amplifiers.
FM Transmitters:
● Frequency modulation can be generated at any point including the radio frequency source.
Accordingly, we can either use direct or indirect method of generation of FM.
● FM transmitters are classified as low-level and high-level transmitters, depending on where
the FM modulation is performed.

● The crystal oscillator generates the stable carrier signal. The modulating signal and the
carrier signal are applied to the phase modulator operating at low power level to generate
a narrowband FM (NBFM) wave.
 ● The narrowband FM wave is then passed through several stages of frequency multipliers to
increase the frequency deviation and the carrier signal frequency to the required level.
● Several stages of frequency multiplication help in choosing a suitable combination for
achieving the required level of multiplication factors needed for deviation and carrier signal
frequency. The output of the FM multiplier stage is a wideband FM (WBFM), but at low
power level.
● The WBFM is then passed through one more stages of power amplifiers to add required
power levels. The WBFM with high power s then transmitted via the antenna towards the
receiver.

FM receiver:
The RF amplifier amplifies the received signal intercepted by the antenna. The amplified signal is
then applied to the mixer stage. The second input of the mixer comes from the local oscillator.
The two input frequencies of the mixer generate an IF signal of 10.7 MHz. This signal is then
amplified by the IF amplifier. Figure (a) shows the block diagram of an FM receiver.

The output of the IF amplifier is applied to the limiter circuit. The limiter removes the noise in the
received signal and gives a constant amplitude signal. This circuit is required when a phase
discriminator is used to demodulate an FM signal.

The output of the limiter is now applied to the FM discriminator, which recovers the modulating
signal. However, this signal is still not the original modulating signal. Before applying it to the
audio amplifier stages, it is de-emphasized. De-emphasizing attenuates the higher frequencies to
bring them back to their original amplitudes as these are boosted or emphasized before
transmission. The output of the de-emphasized stage is the audio signal, which is then applied to
the audio stages and finally to the speaker.
It should be noted that a limiter circuit is required with the FM discriminators. If the demodulator
stage uses a ratio detector instead of the discriminator, then a limiter is not required. This is
because the ratio detector limits the amplitude of the received signal.

In FM receivers, generally, AGC is not required because the amplitude of the carrier is kept
constant by the limiter circuit. Therefore, the input to the audio stages controls amplitudes and
there are no erratic changes the volume level. However, AGC may be provided using an AGC
detector. This generates a dc voltage to control the gains of the RF and IF amplifier.
Superheterodyne Receiver:
● The RF stage is normally a wideband RF amplifier tunable from approximately 540 kHz to
1650 kHz. It is mechanically tied to the local oscillator to ensure precise tuning
characteristics.
● The local oscillator is a variable oscillator capable of generating a signal from 0.995 MHz to
2.105 MHz. The incoming signal from the transmitter is selected and amplified by the RF
stage. It is then combined with a predetermined local oscillator signal in the mixer stage.
● The signal from the mixer is then supplied to the IF (intermediate-frequency) amplifier. This
amplifier is a very-narrow-bandwidth class A device capable of selecting a frequency of
0.455 kHz ± 3 kHz and rejecting all others.
● The IF signal output is an amplified composite of the modulated RF from the transmitter in
combination with RF from the local oscillator.
● The next process is in the detector stage, which eliminates one of the sidebands still
present and separates the RF from the audio components of the other sideband.
● The RF is filtered to ground, and audio is supplied or fed to the audio stages for
amplification and then to the speakers, etc.
Advantages of Superheterodyning:

● It reduces the signal from very high frequency sources where ordinary components
wouldn't work (like in a radar receiver).
● It allows many components to operate at a fixed frequency (IF section) and therefore they
can be optimized or made more inexpensively.
● It can be used to improve signal isolation by arithmetic selectivity

Parts of Receiver:

RF Section:
• A tunable circuit connected to antenna terminals.
• Selects wanted frequency and rejects unwanted frequencies
• Advantages:
– Greater gain: better selectivity
– Improved image-frequency rejection
 – Improved SNR
– Improved rejection of unwanted signals i.e. better selectivity
– Better coupling of receiver to the antenna
– Prevents spurious frequencies from entering the mixer and heterodyning there to
produce an interference frequency equal to the IF.
– Prevents radiation from local oscillator through the antenna

Mixer:
• A nonlinear device having two sets of input terminals and one set of output terminals
• The output contains several frequencies
• The output is tuned to the difference frequency

Local Oscillator:
• Most common types are Armstrong, Hartley, Colpitts, Clapp
• All are LC oscillators
• f0=fs + fIF
• For signal frequency range of 540-1650kHz, the LO frequency range is 995-2105kHz.
• Ratio of max frequency to min frequency is 2.2:1
• If f0 had been below the signal frequency, the LO frequency range would have been
95-1195kHz.
• Ratio of max frequency to min frequency would frequency been 14:1
• Normal tunable capacitor has a capacitance ratio of 10:1, giving frequency ratio of 3.2:1
• 2.2: 1 ratio is well within this range.
• Ratio of the LO frequency to the signal frequency is 995/540= 1.84 and 2105/1650=1.28
• For LO below signal frequency these ratios would be 6.35 and 1.38, respectively.

Intermediate Frequency Amplifier:

Choice of frequency:

● The IF Amplifier of a receiving system is usually a compromise, since there are reasons why
it should be neither low nor high, nor in a certain range between the two.
● The following are the major factors influencing the choice of the IF Amplifier in any system:
1. If the IF Amplifier is too high, poor selectivity and poor adjacent-channel rejection
result unless sharp cutoff filters are used in the IF stages.
2. A high value of intermediate frequency increases tracking difficulties.
3. As the intermediate frequency is lowered, image-frequency rejection becomes poor.
4. A very low IF Amplifier can make the selectivity too sharp, cutting off the sidebands.
 5. If the IF is very low, the frequency stability of the local oscillator must be made
correspondingly higher because any frequency drift is now a larger proportion of the
low IF than of a high IF.
6. The IF Amplifier must not fall within the tuning range of the receiver, or else
instability will occur and heterodyne whistles will be heard, making it impossible to
tune to the frequency band immediately adjacent to the intermediate

Frequencies used:

As a result of many years of experience, the previous requirements have been translated into
specific frequencies, whose use is fairly well standardized throughout the world. These are as
follows:

1. Standard broadcast AM receivers use an IF within the 438- to 465-kHz range, with 455 kHz
by far the most popular
2. AM, SSB and other receivers employed for shortwave or VHF reception have a first IF often
in the range from about 1.6 to 2.3 MHz, or else above 30 MHz.
3. FM receivers using the standard 88- to 108-MHz band have an IF which is almost always
10.7 MHz.
4. Television receivers in the VHF band and in the UHF band use an IF between 26 and 46
MHz, with approximately 36 and 46 MHz the two most popular values.
5. Microwave and radar receivers, operating on frequencies in the 1- to 10-GHz range, use
intermediate frequencies depending on the application, with 30, 60 and 70 MHz among the
most popular.

Diode detector (Envelope detector):

● The envelope detector consists of a diode D and the resistance-capacitor RC combination.
● It is assumed that the amplitude-modulated carrier is supplied by a voltage source with
zero internal resistance, and the diode is an ideal diode.
● When the diode is forward biased, the capacitor charges to the peal of the carrier. When
the carrier voltage decreases after the peak of the carrier, the diode becomes reverse
biased and the capacitor discharges through the resistor.
● Thus the capacitor charges to the peak of each carrier cycle and decays slightly between
the cycles. The time constant RC is selected so that the change in Vc between cycles is at
least equal to the decrease in the carrier amplitude between cycles.
● It is seen that the voltage Vc follows the carrier envelope except that Vc also has
superimposed on it a saw tooth waveform of the carrier frequency.
● Because the carrier frequency is ordinarily much higher than the highest modulating
frequency, the saw tooth distortion can be easily removed by a filter. Thus, the output of
the envelope detector contains:
 i) The modulating signal,
ii) The RF ripples at the carrier frequency,
iii) A DC component which signifies the strength of the carrier.

Advantages:
● Low cost: The diode detector only requires the use of a few low cost components. This made
it ideal for use in transistor (and valve / vacuum tube) radios using discrete components.

● Simplicity: Using very few components, the Diode AM detector was easy to implement. It was
reliable and did not require any setup.

Disadvantages:
● Distortion: As the diode detector is non-linear it introduces distortion onto the detected
audio signal.

● Selective fading: One of the issues often experienced on the short and medium wavebands
where the AM transmissions are located is that of selective fading. The diode envelope
detector is not able to combat the effects of this in the way that some other detectors are able,
and as a result, distortion occurs when selective fading occurs.
● Sensitivity: The diode detector is not as sensitive as some other types. If silicon diodes are
used, these have a turn on voltage of around 0.6 volts as a result, germanium or Schottky
diodes are used which have a lower turn on voltage of around 0.2 to 0.3 volts. Even with the
use of the Schottky diode, the diode envelope detector still suffers from a poor level of
sensitivity

Practical diode detector:

In the practical diode detector, the diode is reversed, so that now the negative envelope is
demodulated to ensure that a negative AGC voltage will be available. The resistor R of the basic
circuit has been split into two parts, R1 and R2 to ensure that there is a series dc path to ground
for the diode, but at the same time, a low-pass filter has been added, in the form of R1-C1, which
removes any RF that might still be present. Capacitor C2 is coupling capacitor, whose main
function is to prevent diode dc output from reaching the volume control R4. The combination
R3-C3 is a low-pass filter designed to remove AF components, providing a dc voltage whose
amplitude is proportional to the carrier strength, and which may be used for automatic gain
control.
From below figure, the diode load is equal to R1+R2, whereas the audio load impedance Zm is
equal to R1 in series with the parallel combination of R2, R3 and R4, assuming that the capacitors
have reactance which may be ignored. This will be true at medium frequencies, but at high and
low audio frequencies Zm may have a reactive component.
Distortion in diode detector:
Two types of distortion may arise in diode detectors. One is caused by the ac and dc diode load
impedance being unequal, and the other by the fact that the ac load impedance acquires a
reactive component at the highest audio frequencies.
Diagonal Clipping:
Diagonal clipping is the name given to the other form of trouble that may arise with diode
detectors. At the higher modulating frequencies, Zm may no longer be purely resistive; it can have
a reactive component due to C and C1. At high modulation depths current will be changing so
quickly that the time constant of the load may be too slow to follow the change. As a result, the
current will decay exponentially, as shown in Figure 11, instead of following the waveform. This is
called diagonal clipping. It does not normally occur when percentage modulation (at the highest
modulation frequency) is below about 60 percent, so that it is possible to design a diode detector
that is free from this type of distortion. The student should be aware of its existence as a limiting
factor on the size of the RF filter capacitors.

Receiver Parameters: (Receiver Characteristics)

1. Sensitivity:
– Ability to amplify weak signals.
 – Often defined in terms of voltage that must be applied to the receiver input terminals to give
a standard output power at the output terminals.
– Standard: 30% modulation by a 400 Hz sine wave, signal applied to the receiver through a
std coupling network called dummy antenna.
– The std output is 50mW, loudspeaker is replaced by a resistance of same value.
– Often expressed in terms of microvolts or decibels below 1V.
– Factors determining the sensitivity of a receiver: Gain of the IF amplifier and the RF amplifier.

2. Selectivity:
– Ability to reject unwanted signals.
– Measured at the end of the sensitivity test with same conditions as for sensitivity
– The frequency of the generator is varied to either side of the frequency to which the
receiver is tuned.
– The ratio of the voltage required of resonance to the voltage required when the generator
is tuned to the receiver’s frequency is calculated and plotted.
– Selectivity is determined by the response of the IF section, with the mixer and the RF
amplifier playing small but important part.
– Selectivity determines the adjacent-channel rejection of the receiver.
3. Image frequency and its rejection:

4. Adjacent channel selectivity (Double Spotting):

– Double spotting is a condition where the same desired signal is detected at two nearby
points on the receiver tuning dial.

– One point is the desired point while the other is called the spurious or image point.

– It can be used to determine the IF of an unknown receiver.

– Poor front-end selectivity and inadequate image frequency rejection leads to double
spotting.

– Double spotting is undesirable since the strong signal might mask and overpower the weak
signal at the spurious point in the frequency spectrum.

– Double spotting can be counter acted by improving the selectivity of RF amplifier and
increasing the value of IF.
5. Fidelity
● Fidelity of a receiver is its ability to reproduce the exact replica of the transmitted signals at
the receiver output.

● For better fidelity, the amplifier must pass high bandwidth signals to amplify the
frequencies of the outermost sidebands, while for better selectivity the signal should have
narrow bandwidth. Thus a trade-off is made between selectivity and fidelity.

● Low frequency response of IF amplifier determines fidelity at the lower modulating

frequencies while high frequency response of the IF amplifier determines fidelity at the
higher modulating frequencies.
Automatic Gain Control: (AGC)
● Automatic Gain Control is a system by means of which the overall gain of a radio receiver is
varied automatically with the changing strength of the received signal, to keep the output
substantially constant.
● A dc bias voltage, derived from the detector is applied to a selected number of the RF, IF
and mixer stages. The devices used in those stages are ones whose trans conductance and
hence gain depends on the applied bias voltage or current.
● For correct AGC operation, this relationship between applied bias and trans conductance
need not be strictly linear, as long as trans conductance drops significantly with increased
bias. The overall result on the receiver output is below.
● All modern receivers are furnished with Working Principle of Automatic Gain Control, which
enables tuning to stations of varying signal strengths without appreciable change in the
volume of the output signal.
● Thus AGC “irons out” input signal amplitude variations, and the gain control does not have
to be readjusted every time the receiver is tuned from one station to another, except when
the change in signal strengths is enormous.
● Working Principle of Automatic Gain Control helps to smooth out the rapid fading which
may occur with long-distance shortwave reception and prevents overloading of the last IF
amplifier which might otherwise have occurred.

(a) simple AGC characteristics (b) various AGC characteristics

Delayed Automatic Gain Control:
Simple AGC is clearly an improvement on no AGC at all, in that the gain of the receiver is reduced
for strong signals. Unfortunately, as below Figure (a) and (b) both show, even weak signals do not
escape this reduction. Figure (b) also shows two other AGC curves. The first is an “ideal” AGC
curve. Here, no Delayed Automatic Gain Control would be applied until signal strength was
considered adequate, and after this point a constant average output would be obtained no
matter how much more the signal strength rose. The second is the Delayed Automatic Gain
Control curve. This shows that AGC bias is not applied until the signal strength has reached a
predetermined level, after which bias is applied as with normal AGC, but more strongly. As the
signal strength then rises, receiver output also rises, but relatively slightly. The problem of
reducing the gain of the receiver for weak signals has thus been avoided, as with “ideal” AGC.

(a) simple AGC characteristics (b) various AGC characteristics

Automatic Frequency Control: (AFC)
● The radar receiver requires a limited tuning range to compensate for transmitter and local
oscillator frequency changes because of variations in temperature and loading. Microwave
radar receivers usually use automatic frequency control (AFC) for this purpose.

AFC in Radio Receivers:

● AFC circuits are used in situations where you must accurately control the frequency of an
oscillator by some external signal. The AFC circuit senses the difference between the actual
oscillator frequency and the frequency that is desired and produces a control voltage
proportional to the difference.
● A varicap is used to keep the IF stable. The varicap application here produces an apparent
reactance, which is included in the oscillator frequency control circuitry.
● This variant of AFC circuits are used in radio receivers, fm transmitters, and frequency
synthesizers to maintain frequency stability. It requires a relatively constant amplitude of
the (received) input-signal. For pulse-radar sets this form isn't practicable.

AFC in Radar Sets:

● Automatic frequency control circuits in a non-coherent or pseudo-coherent radar set uses

two different but similar systems.

1. The transmitters frequency re-adjust the receiver

2. The receiver’s frequency readjusts the transmitter

● Both systems retain a sample of the transmitted signal using a Directional Coupler fitted
between the transmitter and the Duplexer. This RF-signal will be mixed with the local
oscillator frequency to form an AFC-IF-signal. This signal is applied to a frequency-sensitive
discriminator that produces an output voltage proportional in amplitude and polarity to
any change in AFC-IF frequency. If the IF signal is at the discriminator centre frequency, no
discriminator output occurs. The centre frequency of the discriminator is essentially a
reference frequency for the IF signal. The output of the discriminator provides a control
voltage to maintain the local oscillator at the correct frequency.
● The Local Oscillator is adapted to the actual line frequency in this wiring. As a second
variant the control circuit can control the transmitters frequency instead of the LO
frequency! In this case the transmitter-frequency would regulated to the more stable
LO-frequency.
5. Pulse Shaping for Optimum Transmission

Line Code:
A line code is the code used for data transmission of a digital signal over a transmission line. This
process of coding is chosen so as to avoid overlap and distortion of signal such as inter-symbol
interference.

Properties of Line Coding:

● As the coding is done to make more bits transmit on a single signal, the bandwidth used is
much reduced.
● For a given bandwidth, the power is efficiently used.
● The probability of error is much reduced.
● Error detection is done and the bipolar too has a correction capability.
● Power density is much favourable.
● The timing content is adequate.
● Long strings of 1s and 0s is avoided to maintain transparency.

Types of Line Coding:

● Unipolar
● Polar
● Bi-polar

PCM waveforms: (Line codes)

1. Non-return-to zero (NRZ)
2. Return-to-zero (RZ)
3. Phase encoded
4. Multilevel binary

Desirable properties of Line codes:

• DC Component: Eliminating the dc energy from the single power spectrum enables the
transmitter to be ac coupled. Magnetic recording system or system using transformer
coupling are less sensitive to low frequency signal components. Low frequency component
 may have lost, if the presence of dc or near dc spectral component is significant in the code
itself.
• Self-synchronization: Any digital communication system requires bit synchronization.
Coherent detector requires carrier synchronization. For example, Manchester code has a
transition at the middle of every bit interval irrespective of whether a 1 or 0 is being sent
This guaranteed transmitter provide a clocking signal at the bit level.
• Error detection: Some codes such as duo binary provide the means of detecting data error
without introducing additional error detection bits into the data sequence.
• Band width compression: Some codes such as multilevel codes increase the efficiency of
the bandwidth utilization by allowing a reduction in required bandwidth for a given data
rate, thus more information transmitted per unit band width.
• Noise Immunity: For same transmitted energy some codes produces lesser bit detection
error than other in the presence of noise. For ex. The NRZ waveforms have better noise
performance than the RZ type.
• Spectral Compatibility with Channel: Also transmission bandwidth of the code musts is
sufficient small compared to channel bandwidth so that ISI is not problem.
• Transparency: if data are so coded that for any sequence of data, the coded signal is
received faithfully, the code is called transparent. A line code should be so designed that
the receiver does not go out of synchronization for any line sequence of data symbol. A
code is not transparent if for some sequence of symbol, the clock is lost.

Power Spectral Density: (PSD)

The function which describes how the power of a signal got distributed at various frequencies, in
the frequency domain is called as Power Spectral Density (PSD).
PSD is the Fourier Transform of Auto Correlation Similarity between observations Similarity
between observations. It is in the form of a rectangular pulse.

PSD Derivation:
According to the Einstein-Wiener-Khintchine theorem, if the auto correlation function or power
spectral density of a random process is known, the other can be found exactly.
Hence, to derive the power spectral density, we shall use the time auto-correlation (Rx(τ)) of a
power signal x(t) as shown below.
Hence, we get the equation for Power Spectral Density. Using this, we can find the PSD of various
line codes.

Inter Symbol Interference: (ISI)

This is a form of distortion of a signal, in which one or more symbols interfere with subsequent
signals, causing noise or delivering a poor output.
Causes of ISI
The main causes of ISI are −
 ● Multi-path Propagation
● Non-linear frequency in channels
The ISI is unwanted and should be completely eliminated to get a clean output. The causes of ISI
should also be resolved in order to lessen its effect. To view ISI in a mathematical form present in
the receiver output, we can consider the receiver output.
The receiving filter output y(t) is sampled at time ti = iTb (with i taking on integer values),
yielding –

In the above equation, the first term μai is produced by the ith transmitted bit.
The second term represents the residual effect of all other transmitted bits on the decoding of
the ith bit. This residual effect is called as Inter Symbol Interference.
In the absence of ISI, the output will be −
y(ti)=μai
This equation shows that the ith bit transmitted is correctly reproduced. However, the presence
of ISI introduces bit errors and distortions in the output.
While designing the transmitter or a receiver, it is important that you minimize the effects of ISI,
so as to receive the output with the least possible error rate.

Eye Diagram:
● The quality of digital transmission systems is evaluated using the bit error rate. Degradation
of quality occurs in each process modulation, transmission, and detection.
● The eye pattern is experimental method that contains all the information concerning the
degradation of quality. Therefore, careful analysis of the eye pattern is important in
analyzing the degradation mechanism. Eye patterns can be observed using an oscilloscope.
● The interior region of eye pattern is called eye opening
 ● We get superposition of successive symbol intervals to produce eye pattern as shown
below.

• The width of the eye opening defines the time interval over which the received wave can
be sampled without error from ISI
• The optimum sampling time corresponds to the maximum eye opening.
• The height of the eye opening at a specified sampling time is a measure of the margin over
channel noise.
• The sensitivity of the system to timing error is determined by the rate of closure of the eye
as the sampling time is varied. Any nonlinear transmission distortion would reveal itself in
an asymmetric or squinted eye. When the effected of ISI is excessive, traces from the upper
portion of the eye pattern cross traces from lower portion with the result that the eye is
completely closed.
Example of eye pattern:
Binary-PAM Perfect channel (no noise and no ISI)
Q-Factor and BER:
Q factor:

● A Q-factor is measured in the time domain by analysing the statistics of the pulse shape of
the optical signal.
 ● A Q-factor is a comprehensive measure for the signal quality of an optical channel taking
into account the effects of noise, filtering, and linear/non-linear distortions on the pulse
shape, which is not possible with simple optical parameters alone.

Definition:

● The Q-factor, a function of the OSNR, provides a qualitative description of the receiver
performance. The Q-factor suggests the minimum signal-to-noise ratio (SNR) required to
obtain a specific Bit-error-rate (BER) for a given signal. OSNR is measured in decibels. The
higher the bit rate, the higher the OSNR ratio required.

● The Quality factor is a measure of how noisy a pulse is for diagnostic purposes. The eye
pattern oscilloscope will typically generate a report that shows what the Q factor number
is. A larger number means that the pulse is relatively free from noise.

● Q is defined as follows: The ratio between the sums of the distance from the decision point
within the eye (D) to each edge of the eye, and the sum of the RMS noise on each edge of
the eye.

• The Q factor is defined as shown in the figure: the difference of the mean values of the two signal
levels divided by the sum of the noise standard deviations at the two signal levels.

• As Q is a ratio it is reported as a unit-less positive value greater than 1 (Q>1). A Q of 1

represents complete closure of the received optical eye. To give some idea of the associated raw
BER a Q of 6 corresponds to a raw BER of 10-9.

• The mathematic relations to BER when the threshold is set to the optimum value are:
• The Q factor can be written in terms of decibels rather than in linear values:
Duobinary Transfer Function:

Duobinary Impulse Response:

Modified Duobinary Signaling technique:

Impulse Response:

Equalizer:
▪ The equalizer is a device that attempts to reverse the distortion incurred
by a signal transmitted through a channel.
 ▪ In digital communication its purpose is to reduce inter symbol
interference to allow recovery of the transmit symbols. It can be a
simple linear filter or a complex algorithm.
▪ Types: Adaptive or non-adaptive.

Non-Adaptive equalizer:

Linear Equalizer:
It processes the incoming signal with a linear filter.
Zero Forcing Equalizer:
MSE Equalizer:

4. Pulse Modulation
Sampling Theorem:
 ● The sampling theorem can be defined as the conversion of an analog signal into a discrete
form by taking the sampling frequency as twice the input analog signal frequency.
● Input signal frequency denoted by Fm and sampling signal frequency denoted by Fs.
● The output sample signal is represented by the samples. These samples are maintained
with a gap; these gaps are termed as sample period or sampling interval (Ts) and the
reciprocal of the sampling period is known as “sampling frequency” or “sampling rate”.
● The number of samples is represented in the sampled signal is indicated by the sampling
rate.
● The sampling process requires two input signals. The first input signal is an analog signal
and another input is sampling pulse or equidistance pulse train signal. And the output
which is then sampled signal comes from the multiplier block.

sampling-block-diagram

Types of Sampling:

1. Instantaneous Sampling
2. Natural Sampling
3. Flat-topped Sampling

Applications:
● To maintain sound quality in music recordings.
● Sampling process applicable in the conversion of analog to discrete form.
● Speech recognition systems and pattern recognition systems.
● Modulation and demodulation systems
● In sensor data evaluation systems
● Radar and radio navigation system sampling is applicable.
● Digital watermarking and biometric identification systems, surveillance systems.

Sampling Rate:
The sampling rate is the number of samples taken in the duration of one second. it is measured
in hertz or sample per second. The continuously varying amplitude of an analog signal is also
continuous in time. So it needs to be sampled at a fixed rate. This rate is called sampling
rate or sampling frequency. Example of sampling:

This signal is sampled at a sampling rate of 2 samples per second or 2 Hz.

Sampling rate plays important role in the perfect conversion from analog to digital
and reconstruction of an analog signal from the digital signal.
Sampling rate should not be very low or very high. In both cases, the converted signal is not what
we want to achieve. If the sampling rate is low than the original signal is destroyed and if the
sampling rate is very high then it’s not economically beneficial.

Aliasing:
If the analog signal is sampled at a frequency lower than the required rate then the sampled
signal does not appear to be anything like the original signal. And the reconstruction of the
original signal becomes impossible. Such case is called aliasing as shown in the figure below.
In this example, a sinusoidal signal is sampled at a rate of 3/4 of its frequency. which is very lower
than its required rate. The reconstructed signal (red signal) is recovered from the sample which
does not look anything like the original signal.

Nyquist Theorem:
The sampling rate or sampling frequency should be greater than twice the input signal’s
frequency. Nyquist theorem suggests the minimum sampling rate for a signal which can
be perfectly reconstructed from its samples.

Sample and Hold Circuit:

Sample & Hold Circuit is used to sample the given input signal and to hold the sampled value.
Sample and hold circuit is used to sample an analog signal for a short interval of time in the range
of 1 to 10µS and to hold on its last sampled value until the input signal is sampled again. The
holding period may be from a few milliseconds to several seconds.

Need:
digital communication is advantageous when compared with analog communication. However, to
utilize a digital system, the applied signal at the input must also be in digitized form. While
originally, a signal is analog in nature. So, in order to change the analog signal into digital form, a
sample and hold circuit is used. It is usually placed before an analog to digital converter.

Basic Block diagram:

In the sample mode of the circuit, the switch present is closed, and so this charges capacitor C,
with the instantaneous value of the applied input signal. However, in the hold mode of the circuit,
the switch now gets open, and so no further charging is possible.

But now at the hold mode, the capacitor holds the charge that was initially being stored at the
time of sample mode.

While the question arises that why the stored charge is held by capacitor rather being dissipated.
So, this is because the circuit has no path for the dissipation of the stored charge through it.
Advantages:
● It provides synchronization to all the channel present in multi-channel analog to digital
converter.
● The possibility of cross-talk can also be reduced.

Applications:
1. It is used in analog signal processing.
2. Also finds applications in digital voltmeter.
3. In the data conversion system as well as in sampling oscilloscopes.
4. In data distribution systems and analog to digital converters.
5. In filters used for signal construction.
Quantization:
● Quantization is a process that converts sampled signal into an approximate quantized signal
which consists of only a finite number of predecided voltage levels. It is a process of rounding off
or approximation.
● Each sampled value at the input of the quantizer is approximated or rounded off to the nearest
standard predecided voltage level. These standard levels are known as “quantization level”

The quantization process takes place as follows:

● The input signal x (t) is assumed to have a peak to peak swing of VL to VH volts. This entire
voltage range has been divided into “Q” equal intervals each of size “s”
● “s” is called as the step size and its value is given as,

● In the fig, the value of Q =8

● q0, q1……. q7 are the quantization levels also called as decision threshold.
-xq(t) represents the quantized version of x (t). We obtain xq(t) at the output of the quantizer
When x(t) is in the range Δ0∆0 then corresponding to any value of x(t) the quantizer output will be
equal to “q0”. Similarly, for all the values of x (t) in the range ∆1 quantizer output will be equal to
“q1”
● Thus in each range from Δ1 to Δ7 the signal x(t) is rounded off to the nearest quantization level
and the quantized signal is produced.
● The quantized signal xq(t) is thus an approximation of x (t).
● The difference between them is called as quantization error or quantization noise.
● This error should be as small as possible.
● To minimize the quantization error step size “s” should be small as possible which is done by
increasing the number of quantization level Q.

Quantization Error:
● 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.
● 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.
● The following figure illustrates an example for a quantization error, indicating the difference
between the original signal and the quantized signal.
Pulse Amplitude Modulation (PAM):
● A modulation technique in which the amplitude of the pulsed carrier signal is changed
according to the amplitude of the message signal is known as Pulse Amplitude Modulation
(PAM).

● In case of PAM, that flat top sampling is widely used and is more popular than natural
sampling. This is so because, at the time of signal transmission the channel noise
introduces some form of distortion in it, that can be easily eliminated in case of flat tops.

● As against when natural sampling is done in case of PAM signals, then the top of the pulses
varies according to the modulating signal. So, in this case, the elimination of noise
component from the sampled signal becomes somewhat difficult at the time of detection.

● At the same time, natural sampling is a bit complex process as compared to flat top
sampling.

● Hence, flat top sampling is preferred in pulse amplitude modulation.

Block Diagram:

● it consists of a low pass filter, a modulator along with a train generator and a pulse
reshaping circuit.

● Here the modulating signal is given to the low pass filter in order to band limit the message
signal.

● The LPF at the beginning is placed in order to avoid aliasing of the samples. The LPF passes
only the low-frequency component of the signal and eliminates the high-frequency signal
component. The output of LPF is then provided to a modulator, where it gets mixed with
the rectangular pulse train.
 ● Basically, the pulsed carrier gets modulated by the message signal here. The rectangular
carrier pulse is generated by the pulse generator circuit.

● The modulator generates a PAM signal. The sampled pulses can be achieved either by
natural or flat top sampling. The output of the modulator is provided to the pulse reshaping
circuit. This basically shapes the pulses so that it can be easily detected at the receiver.

PAM signal generation:

● Here Rb, Re and Rc denote base resistor, emitter resistor and collector resistor and Vcc is the
supply voltage. The transistor employed in the circuit is responsible for the desired pulsed
signal.

● The circuit is also known as emitter follower as the output is received from the emitter
terminal of the transistor.

● The rectangular pulse is applied to the transistor along with the message signal. When
the peak of the pulse is high, it causes the transistor to operate in the saturation region.
Thus, it behaves as a short circuit, allowing the analog signal to reach the output.

● However, in case of the low peak of the rectangular pulse, the transistor operates in the
cut-off region. Resultantly, it starts behaving as an open circuit and blocks the signal
transmission.

● In this way, a pulse amplitude modulated signal is transmitted.

Demodulation of PAM:
● For Demodulation of the Pulse Amplitude Modulated signal, PAM is fed to the low pass
filter.

● The low pass filter eliminates high frequency ripples and generates the demodulated signal
which has its amplitude proportional to PAM signal at all time instant.
● This signal is then applied to an inverting amplifier to amplify its signal level to have the
demodulated output with almost equal amplitude with the modulating signal.

Advantages:
● PAM is the simplest form of pulse modulation.
● Its implementation is quite easy.
Disadvantages:
● The transmission bandwidth required is very large.
● Due to the variation in amplitude, the power required by the generating unit also varies.
● Less immune to noise due to amplitude variation.
Applications: It is used in LED lighting, in microcontrollers in order to produce control signals
and in the Ethernet communication system.
Pulse Code Modulation: (PCM)
● A technique by which analog signal gets converted into digital form in order to have signal
transmission through a digital network is known as Pulse Code Modulation (PCM).

● PCM systems are basically signal coders also known as waveform coders. PCM allows the
representation of the continuous time message signal as a sequence of binary coded
pulses. The binary form permits only 2 probable states i.e., 0 and 1.

● The major steps involved in PCM is sampling, quantizing and encoding.

PCM Block Diagram:

PCM Transmitter and Receiver

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: 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.
Decoder: The decoder circuit decodes the pulse coded waveform to reproduce the original signal.
This circuit acts as the demodulator.
Reconstruction Filter:
● After the digital-to-analog conversion is done by the regenerative circuit and the decoder, a
low-pass filter is employed, called as the reconstruction filter to get back the original signal.
● Hence, the Pulse Code Modulator circuit digitizes the given analog signal, codes it and
samples it, and then transmits it in an analog form. This whole process is repeated in a
reverse pattern to obtain the original signal.

Bandwidth:

Advantages:
1. Immune to channel induced noise and distortion.
2. Repeaters can be employed along the transmitting channel.
3. Encoders allow secured data transmission.
4. It ensures uniform transmission quality.
Disadvantages:
1. Pulse code modulation increases the transmission bandwidth.
2. A PCM system is somewhat more complex than another system.

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.

● Companding is done in order to improve SNR of weak signals.

● There are two types of Companding techniques −

A-law Companding Technique:

● Uniform quantization is achieved at A = 1, where the characteristic curve is
linear and no compression is done.
● A-law has mid-rise at the origin. Hence, it contains a non-zero value.
● A-law companding is used for PCM telephone systems.

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.

Characteristic of Compander:
Compressor characteristic:

The graph clearly represents that the compressor provides high gain to weak signal and low gain
to high input signal.

Expander characteristic:
expander performs reverse operation of the compander. So, the artificially boosted signals is
attenuated to have the originally transmitted signal.

The figure below shows the companding curve for PCM system:

The compressor and expander performs inverse operations thus in the above figure the dotted
line represents the linear characteristic of the compander indicating that the originally
transmitted signal is recovered at the receiver.
Delta Modulation: (DM)
● A modulation technique that converts or encodes message signal into a binary bit
stream is known as Delta Modulation. Here only 1 bit is used to encode 1 voltage level
thus, the technique allows transmission of only 1 bit per sample.
● As PCM has the property of converting message signal directly into a sequence of a binary
coded pulse, this resultantly increases the bandwidth requirement of the system. So, in
order to remove the drawbacks of PCM, delta modulation is used.

Transmitter

Delta Modulator

● It consists of a 1-bit quantizer and a delay circuit along with two summer circuits.
● The summer in the accumulator adds quantizer output (±Δ) with the previous sample
approximation. This gives present sample approximation. i.e.,

● The previous sample approximation u[(n-1)Ts ] is restored by delaying one sample period Ts .
● The samples input signal x(nTs ) and staircase approximated signal xˆ(nTs ) are subtracted to get
error signal e(nTs ).
● Thus, depending on the sign of e(nTs ), one bit quantizer generates an output of +Δ or -Δ .
● If the step size is +Δ, then binary ‘1’ is transmitted and if it is -Δ, then binary ‘0’ is transmitted .
Receiver
● At the receiver end also known as delta demodulator, as shown in Fig.2, it comprises of a low
pass filter(LPF), a summer, and a delay circuit. The predictor circuit is eliminated here and hence
no assumed input is given to the demodulator.

● The accumulator generates the staircase approximated signal output and is delayed by one
sampling period Ts.
● It is then added to the input signal.
● If the input is binary ‘1’ then it adds +Δ step to the previous output (which is delayed).
● If the input is binary ‘0’ then one step ‘Δ’ is subtracted from the delayed signal.
● Also, the low pass filter smoothens the staircase signal to reconstruct the original message signal
x(t).
● The delta modulation has two major drawbacks as under: (a)Slope Overload Distortion (b)
Granular Noise
Advantages:
● Due to transmission of 1 bit per sample, it permits low channel bandwidth as well as
signaling rate.
● ADC is not required. Thus permits easy generation and detection.

Disadvantages: Delta modulation leads to drawbacks such as slope overload distortion

and granular noise.

Applications: It is widely used in radio communication devices and digital voice storage
and voice information transmission where signal quality is less important.

Slope Overload Distortion:

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

Quantization Errors in Delta Modulation

● the rate of rise of input signal x(t) is so high that the staircase signal cannot 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.
Time Division Multiplexing: (TDM)
● Time division multiplexing is a technique of separating the signals in time domain.

● In TDM the transmission from multiple sources take place on the same medium but not at
the same time.

● The transmissions from various sources are interleaved in time domain. In other words, the
data from the various sources is arranged in non-contiguous manner by dividing the data
into small chunks, which also makes the system efficient.

● Pulse code modulation is the most common encoding technique used for TDM digital
signals.

● PCM system used in North America is a 24-channel system with the sampling rate of 8000
samples per second, 8 bits per sample and a pulse width of 0.625 μs.

● We can calculate that sampling interval is 1/8000 = 125 μs, and period required for each
pulse group is 8 x 0.625 = 5 μs.

● If we transmit only one channel without using the multiplexing technique, then the
transmission will contain 8000 frames per second, which will consist of the activity only
during the first 5 μs and nothing at all during the rest 120 μs.

● Thus will be wasteful and employs complicated method for encoding single channel.
Therefore TDM technique is used so that each 125 μs frame is used to provide 24 adjacent
channel time slots with the twenty-fifth time slot for synchronization.

Advantages:

● Simple circuit design.

● It uses entire channel bandwidth for the transmission of the signal.
● The problem of Intermodulation distortion is not present in TDM.
● Pulse overlapping can sometimes cause crosstalk but it can be reduced by utilizing guard
time. Thus, is not much serious.
Disadvantages:

● The transmitting and receiving section must be properly synchronized in order to have
proper signal transmission and reception.
● Slow narrowband fading can wipe out all the TDM channels.

Applications:

● TDM finds its application mainly in a digital communication system, in cellular radio, in
satellite communication system, in the transmission of SDH and SONET system, GSM
telephone system etc.
T1 Digital Carrier System: (5 marks)

● Figure above shows the basic time division multiplexed scheme, called T1 digital system,
which is used to convey multiple signals over telephone lines using wideband coaxial cable.
● It accommodates 24 analog signals referred to as s1 through s24. Each signal is bandlimited
to approx. 3.3kHz and is sampled at a rate 8kHz. Each of the time division multiplexed
signal is next A/D converted and companded.
● The commutators sweep continually from s1 to s24 and back to s1, etc. at the rate of 8000
revolutions per second thereby providing 8000 samples per second of each signal. Each
sample is encoded into 8 bits. The digital signal generated during the course of one
complete sweep of the commentator is therefore 24x8=192 bits.
● To provide synchronization an extra single bit is made available immediately preceding the
192 bits that carry the encoded signals. Thus, the total of 193 bits is called frame. The time
slots for the 24 signals in a frame are shown below.
T1 Digital Carrier System: (10 marks)

● Figure above shows the basic time division multiplexed scheme, called T1 digital system,
which is used to convey multiple signals over telephone lines using wideband coaxial cable.
● It accommodates 24 analog signals referred to as s1 through s24. Each signal is bandlimited
to approx. 3.3kHz and is sampled at a rate 8kHz. Each of the time division multiplexed
signal is next A/D converted and companded.
● The commutators sweep continually from s1 to s24 and back to s1, etc. at the rate of 8000
revolutions per second thereby providing 8000 samples per second of each signal. Each
sample is encoded into 8 bits. The digital signal generated during the course of one
complete sweep of the commentator is therefore 24x8=192 bits.
● To provide synchronization an extra single bit is made available immediately preceding the
192 bits that carry the encoded signals. Thus, the total of 193 bits is called frame. The time
slots for the 24 signals in a frame are shown below.
● 12 successive F slots are used to transmit a 12-bit code, 110111001000. This code is
transmitted repetitively once every 12 frames and is used at the receiver to establish
synchronization.
● Each signal is sampled 8000 times per second so that a complete frame occupies
1/8000=125μs. This time accommodates 193 bits so that the bit rate on a T1 channel is
193/125 Mbps= 1.544Mbps.
● The eighth bit of each encoded sample is used for both voice transmission and for
signalling. During five successive frames each sample is encoded into eight bits. But during
the sixth frame the samples are encoded into only seven bits, the eighth bit being used for
signalling. This pattern is repeated every six frames. Thus in six frames the number of bits
used for quantization encoding is 5 x 8+1 x 7=47 bits so that samples are encoded on an
5
average into 47/6= 7 6 bits.

● 4 T1 lines are multiplexed to generate a T2 transmission system with a bit rate of

6.312Mbps, 7 T2 lines convert to T3 line with bit rate of 44.736 Mbps after multiplexing, six
T3 lines convert to T4 line with bit rate of 274.176 Mbps. At each stage additional frame
synchronization bits must be added so that at each multiplexer output it would be possible
to distinguish which bits belong to which input.
Frequency Division Multiplexing (FDM): (5 marks)
● In FDM, standardized groupings of channels are used and several steps of frequency
translation take place before all the channels have been placed in their location in the
frequency spectrum that is transmitted in a particular link.
● Basic group generally consists of 12 adjacent 4kHz channels, occupying the frequency
range from 60 to 108kHz. A low level pilot is transmitted at 104.08kHz for regulating and
modulating purpose.
● Narrower channels are used in submarine cables and so here a basic group consists of 16
3kHz channels, occupying the same 48kHz range as the basic group with pilot inserted at
84kHz.

● Figure above shows the channel arrangement for the basic group B. All the channels in the
basic group B are inverted. All 12 channels in the basic group are LSBs. the basic 16 channel
group is a mix of erect and inverted channels.
● The next step up from the basic group is the supergroup, consisting of five groups and
occupying the frequency range of 312 to 552kHz i.e. a bandwidth of 240kHz.
● The figure below shows the location of channels and groups in the basic supergroup. The
basic supergroup is erect. The basic supergroup is formed in the group translating
equipment, in a process similar to basic group formation.
● The supergroup pilot is inserted at 547.94 kHz.

● Five supergroups are assembled to form a mastergroup and then a supermastergroup. The
supermastergroup, 15-supergroup assembly, thus consists of 900 channels.
● FDM is used for simultaneous telephone, telex and data transmission.
Frequency Division Multiplexing (FDM): (10 marks)
● In FDM, standardized groupings of channels are used and several steps of frequency
translation take place before all the channels have been placed in their location in the
frequency spectrum that is transmitted in a particular link.
● Basic group generally consists of 12 adjacent 4kHz channels, occupying the frequency
range from 60 to 108kHz. A low level pilot is transmitted at 104.08kHz for regulating and
modulating purpose.
● Narrower channels are used in submarine cables and so here a basic group consists of 16
3kHz channels, occupying the same 48kHz range as the basic group with pilot inserted at
84kHz.

● Figure above shows the channel arrangement for the basic group B. All the channels in the
basic group B are inverted. All 12 channels in the basic group are LSBs. the basic 16 channel
group is a mix of erect and inverted channels.
● Figure below shows the basic arrangement for channel translating equipment.
● The next step up from the basic group is the supergroup, consisting of five groups and
occupying the frequency range of 312 to 552kHz i.e. a bandwidth of 240kHz.
● The figure below shows the location of channels and groups in the basic supergroup. The
basic supergroup is erect. The basic supergroup is formed in the group translating
equipment, in a process similar to basic group formation.
● The supergroup pilot is inserted at 547.94 kHz.

● Five supergroups are assembled to form a mastergroup and then a supermastergroup (3

mastergroups). The supermastergroup, 15-supergroup assembly, thus consists of 900
channels.
● FDM is used for simultaneous telephone, telex and data transmission.
NOTE:
• 12 channels= 1 basic group
• 5 basic group= 1 supergroup (60 channels)
• 5 supergroups= 1 mastergroup (300 channels)
• 3 mastergroups= 1 supermastergroup (300 x3= 900 channels)

Difference Between TDM and FDM:

TDM FDM

The signals which are to be multiplexed can Signals which are to be multiplexed are added in
occupy the entire bandwidth but are isolated in the time domain. But they occupy diff slots in the
the time domain frequency. domain

TDM is preferred for the digital signals. FDM is usually preferred for the analog signals.

Synchronization required Synchronization not required

TDM circuiting is not very complex. The FDM requires a complex circuiting at the
transmitter and receiver.

In TDM the problem of crosstalk is not severe. FDM suffers from the problem of crosstalk due to
imperfect band filters.
TDM FDM

Due to fading only a few TDM channels will be Due to wideband fading in transmission medium, all
affected FDM channels are affected

Due to slow narrow band fading all the TDM Due to slow narrowband fading taking place in the
channels may get wiped out. transmission channel only a single channel may be
affected in FDM

6. Digital Modulation Techniques

Digital Modulation:
● 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.
1. 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.
2. FSK – Frequency Shift Keying
The frequency of the output signal will be either high or low, depending upon the input
data applied.
3. 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 BPSKBPSK and Quadrature Phase Shift
Keying QPSKQPSK, according to the number of phase shifts. The other one is Differential
Phase Shift Keying which changes the phase according to the previous value.
 4. 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

Binary Phase Shift Keying: (BPSK)

● BPSK is the known as the simplest form of Phase Shift Keying (PSK).

● In a BPSK modulation process, the phase of the sinusoidal carrier signal is changed
according to the message level (“0” or “1”) while keeping the amplitude and frequency
constant.

● When CINR is poor base station choose BPSK modulation technique in most of the
adaptive modulation technique adopted in cellular communication.

● It is less immune to the interference.

BPSK Spectrum
 NOTE: derivation in notebook
Advantages:
● It is considered to be more robust among all the modulation types due to difference of
180 degree between two constellation points. Hence it can withstand severe amount of
channel conditions or channel fading.

● Due to above, BPSK modulation is used by most of the cellular towers for long distance
communication or transmission of the data.

● BPSK demodulator requires to make only two decisions in order to recover original binary
information. Hence BPSK receiver is very simple compare to other modulation types.

● BPSK is power efficient modulation technique as less power is needed to transmit the
carrier with less number of bits.

Disadvantages:
● In BPSK modulation, one bit is carried by one single analogue carrier. Hence data rate in bits
per second is same as the symbol rate. This is half in comparison to the QPSK modulation
technique and many times less compare to other higher modulation techniques such as
16QAM, 64QAM etc.

● Due to above reason, BPSK is not bandwidth efficient modulation technique compare to
other modulation types.

Applications:
● The BPSK modulation is a very basic technique used in various wireless standards such
as CDMA, WiMAX (16d, 16e), WLAN 11a, 11b, 11g, 11n, Satellite, DVB, Cable modem
etc.

● It is used in OFDM and OFDMA to modulate the pilot subcarriers used for channel
estimation and equalization.

● It is used for long distance wireless communication.

● As we know different channels are used for specific data transmission in cellular
systems. The channels used to transmit system related information which are very
essential are modulated using BPSK modulation.
BPSK transmitter:

The BPSK signal can be generated by applying carrier signal to the balanced modulator.
● The baseband signal b(t) is applied as a modulating signal to the balanced
● The NRZ level encoder converts the binary data sequence into bipolar NRZ signal.

BPSK receiver:
The transmitted BPSK signal is,
 scheme to recover baseband signal from BPSK signal

Operation of Receiver:
1) Phase shift in received signal:
● This signal undergoes the phase change depending upon the time delay from the
transmitter to receiver.
● This phase change is normally fixed phase shift in the transmitted signal. Let the phase shift
be θ. Therefore, the signal at the input of the receiver is,

s(t)=b(t) 2𝑃cos(2πf0t+θ) ..(7)

2) Square law device
● Now from this received signal, carrier is separated since this is coherent detection.
● As shown in the above figure, the received signal is passed through a square law device.
● At the output of the square law device the signal will be,
cos2(2πf0t+θ)
Note here that we have neglected the amplitude, because we are only interested in the carrier of
the signal.
We know that,

1 1
∴ Cos2 (2πf0t+θ) = 2
+ 2
cos2(2πf0t+θ)
1
Here, 2
represents D.C.level.
3)Bandpass filter
● This signal is then passed through a bandpass filter whose passband is centred around 2f0.
● Bandpass filter removes the D.C level of 1212 and at its output we get,
● cos2(2πf0t+θ) this signal has a frequency of 2f02f0.

4) Frequency divider
● The above signal is passed through a frequency divider by two.
● Therefore, at the output of frequency divider we get a carrier signal whose frequency is
f0 i.e., cos(2πf0t+θ).

5) Synchronous demodulator
● The synchronous (coherent) demodulator multiplies the input signal and the recovered
carrier.
● Therefore, at the output of multiplier we get,

6)Bit synchronizer and integrator:

● The above signal is then applied to the bit synchronizer and integrator. The integrator
integrates the signal over one-bit period.
● The bit synchronizer takes care of starting and ending times of a bit.
● At the end of bit duration Tb, the bit synchronizer closes switch S2 temporarily. This
connects the output of an integrator to the decision device. It is equivalent to sampling the
output of integrator.
Thus BPSK waveform has full cycles of sinusoidal carrier.

Quadrature Phase Shift Keying: (QPSK)

● 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 bigits.
● 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.
 ● Application: Satellite communication and mobile communication.

QPSK Phasor Diagram:

Binary QPSK
Input Output
Phase

bo be

-1 -1 -3π/4

-1 1 -π/4

1 -1 3π/4

1 1 π/4

Constellation diagram of QPSK: (Signal Space representation / Geometric

Representation)
 NOTE: derivation in notebook

QPSK Transmitter:
QPSK Receiver:
M-ary PSK:
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.
● 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.

NOTE: derivation in notebook

Constellation diagram M-ary PSK:

M-ary PSK transmitter:

M-ary PSK receiver:

Binary Amplitude Shift Keying (BASK):

 ● The digital data to be transmitted is the binary number 1011. Two amplitudes are used to
directly represent the data, either 0 or 1. In this case, the modulation is called binary
amplitude shift keying or BASK.

● The signal is divided into 4 pulses of equal duration which represent the bits in the digital
data.

● The number of bits used for each character is a function of the system, but is typically
eight, seven of which represent the 128 possible characters, the last bit is used to check for
errors.

● It is simplest form and it has been used for radio telegraphy transmission in Morse Code.
● In BASK a sinusoidal carrier is simply gated on and off by the bit sequence to be
transmitted.

● Advantage: simplicity
● Disadvantage:

● Application:

Derivation:
Spectrum of BASK:

BASK Transmitter & Receiver:

Quadrature amplitude modulation (QAM):
 ● Quadrature Amplitude Modulation combines amplitude & phase changes to give additional
capacity & is widely used for data communications

● Quadrature Amplitude Modulation, QAM is a signal in which two carriers shifted in phase
by 90 degrees are modulated and combined. As a result of their 90° phase difference they
are in quadrature and this gives rise to the name. Often one signal is called the In-phase or
“I” signal, and the other is the quadrature or “Q” signal.

● The resultant overall signal consisting of the combination of both I and Q carriers contains
of both amplitude and phase variations. In view of the fact that both amplitude and phase
variations are present it may also be considered as a mixture of amplitude and phase
modulation.

Constellation diagram:

Advantages: increases the efficiency of transmission for radio communications systems by

utilising both amplitude and phase variations.

Disadvantages:
it is more susceptible to noise because the states are closer together so that a lower level of noise
is needed to move the signal to a different decision point. Receivers for use with phase or
frequency modulation are both able to use limiting amplifiers that are able to remove any
amplitude noise and thereby improve the noise reliance. This is not the case with QAM.

When a phase or frequency modulated signal is amplified in a radio transmitter, there is no need
to use linear amplifiers, whereas when using QAM that contains an amplitude component,
linearity must be maintained. Unfortunately, linear amplifiers are less efficient and consume more
power, and this makes them less attractive for mobile applications.

Applications:
 • 64-QAM and 256-QAM are often used in digital cable television and cable
modem applications.
• current Homeplug AV2 500-Mbit/s powerline Ethernet devices use 1024-QAM and
4096-QAM
• ADSL technology for copper twisted pairs, whose constellation size goes up to
32768-QAM

QASK Transmitter:

QASK Receiver:
Binary Frequency Shift Keying (BFSK):
In a BFSK (Binary Frequency-Shift Keying) modulation process by keeping the amplitude and
phase constant, the frequency of the sinusoidal carrier signal is changed according to the
message level (“0” or “1”). A block diagram of the BFSK modulation techniques and its related
signal waveforms are shown below.

BFSK Spectrum:
Constellation Diagram:

BFSK Transmitter:
BFSK Receiver: