CTT (Communication Theory and Telemetry) FULL

Communication Theory and
Telemetry
L1: Introduction
Dr. Jayanta K Rakshit

Previous Knowledge
The students will be know about
working of semiconductor devices.
The student will be know about

signals and systems.
The students will be know about

electromagnetic waves.
Recommended books
• B.P Lathi, “Modern Digital and Analog

Communication Systems” Oxford University Press
• R. P. Singh, S. D. Sapre, “Communication Systems”,
Tata McGraw-Hill Education.
• Taub & Schilling, “Communication system”, TMH.
• Simon Haykin, Michael Moher, “Communication
Systems”, 5th Edition, Wiley,
• D. Patranabis, Telemetry principles, TMH, New
Delhi.
Introduction
• The word communication arises from the Latin
word “commūnicāre”, which means “to share”.
• Communication is the basic step for the exchange
of information.
Communication is the bridge to share.
•
Communication can be defined as the process of
exchange of information through means such as
words, actions, signs, etc., between two or more
individuals
Communication System
• Electrical communication systems are designed
to send messages or information from a source
that generates the messages to one or more
destinations.
• In general, a communication system can be
represented by the functional block diagram
shown in following Figure.
• A transducer is usually required to convert the output of a
source into an electrical signal that is suitable for
transmission.
• For example, a microphone serves as the transducer that

converts an acoustic speech signal into an electrical signal,
and a video camera converts an image into an electrical
signal.
• At the destination, a similar transducer is required to

convert the electrical signals that are received into a form
that is suitable for the user; e.g., acoustic signals, images,
etc.
Signal
• It is a physical quantity which varies with respect to time or space or
independent or dependent variable.
(Or)
It is electrical waveform which carries information.
•
Ex: m(t) = Acos(ωt+ϕ)
Where, A= Amplitude or peak amplitude(Volts)
ω = Frequency ( rad/sec)
ϕ = Phase (rad)
• Hence, a signal can be a source of energy which transmits some

information. This signal helps to establish communication between a
sender and a receiver.
• An electrical impulse or an electromagnetic wave which travels a

distance to convey a message, can be termed as a signal in
communication systems.
Signal
• Signals physically exist in the time domain and are
usually expressed as a function of the time parameter.
• it may even be possible to view the signals on an
oscilloscope. But equally important is the
characterization of the signals in the Frequency
Domain or Spectral Domain. That is, we characterize
the signal in terms of its various frequency components
(or its spectrum). Fourier analysis (Fourier Series and
Fourier Transform) helps us in arriving at the spectral
description of the pertinent signals.
Types of signals
Transmitter
• The transmitter converts the electrical signal into a form that
is suitable for transmission through the physical channel or
transmission medium.
• For example, in radio and TV broadcast, the Federal

Communications Commission (FCC in USA, PTA, FAB, in
Pakistan) specifies the frequency range for each transmitting
station.
• Hence, the transmitter must translate the information signal

to be transmitted into the appropriate frequency range that
matches the frequency allocation assigned to the transmitter.
• Thus, signals transmitted by multiple radio stations do not

interfere with one another. 11
Transmitter
• In general, the transmitter performs the matching of the
message signal to the channel by a process called
modulation.
• Usually, modulation involves the use of the information

signal to systematically vary either the amplitude,
frequency, or phase of a sinusoidal carrier.
12
Transmitter
• For example, in AM radio broadcast, the information signal
that is transmitted is contained in the amplitude variations of
the sinusoidal carrier, which is the center frequency in the
frequency band allocated to the radio transmitting station.
13
AM Transmission
14
Transmitter
• In FM radio broadcast, the information signal that is
transmitted is contained in the frequency variations of
the sinusoidal carrier.
15
Transmitter
• Thus, through the process of modulation, the
information signal is translated in frequency to match
the allocation of the channel.
• The choice of the type of modulation is based on several

factors, such as:
– amount of bandwidth allocated,
– types of noise and interference that the signal encounters in
transmission over the channel
– electronic devices that are available for signal amplification
prior to transmission.
16
Channel
• The communications channel is the physical
medium that is used to send the signal from the
transmitter to the receiver.
• In wireless transmission, the channel is usually

the atmosphere (free space).
• On the other hand, telephone channels usually

employ a variety of physical media, including
copper wires and optical fiber cables.
17
Channel
• Whatever the physical medium for signal transmission,
the essential feature is that the transmitted signal is
corrupted in a random manner by a variety of possible
mechanisms.
• The most common form of signal degradation comes in

the form of additive noise, which is generated at the
front end of the receiver, where signal amplification is
performed.
• This noise is often called thermal noise.
18
Channel
• In wireless transmission, additional additive
disturbances are man-made noise, and atmospheric
noise picked up by a receiving antenna.
• Interference from other users of the channel is

another form of additive noise that often arises in
both wireless and wireline communication systems.
19
Channel
• In the design of a communication system, the system
designer works with mathematical models that
statistically characterize the signal distortion
encountered on physical channels.
• Often, the statistical description that is used in a

mathematical model is a result of actual empirical
measurements obtained from experiments involving
signal transmission over such channels.
20
Receiver
• The function of the receiver is to recover the message
signal contained in the received signal.
• If the message signal is transmitted by carrier modulation,

the receiver performs carrier demodulation in order to
extract the message from the sinusoidal carrier.
21
Receiver
• Besides performing the primary function of signal
demodulation, the receiver also performs a number of
peripheral functions, including signal filtering and
noise suppression.
22
Receiver
• Since the signal demodulation is performed in the
presence of additive noise and possibly other signal
distortion, the demodulated message signal is
generally degraded to some extent by the presence of
these distortions in the received signal.
• The fidelity of the received message signal is a

function of
– Type of modulation,
– Strength of the additive noise
– Type of any non-additive interference.
23
THE END
Thank you
L2# Modulation

Modulation/demodulation
• In the modulation process, two signals are used namely the
modulating signal and the carrier. The modulating signal is nothing
but the base-band signal or information signal while the carrier is
a high frequency sinusoidal signal .
• In the modulation process, some parameter of the carrier wave
(such as amplitude, frequency or phase ) is varied in accordance
with the modulating signal. This modulated signal is then
transmitted by the transmitter.
• The receiver demodulates the received modulated signal and gets
the original information signal back. Thus, demodulation is exactly
opposite to modulation .
• In the process of modulation the carrier wave actually acts as
carrier which carries the information signal from the transmitter to
receiver.
Modulation
Types of Modulation
Benefits or Need of Modulation
 To reduce the length or height of antenna
 To avoids mixing of signals
 To improve the power radiation
 For narrow banding or to use antenna with single
or same length
 For multiplexing
 To reduce noise effect
 To avoid equipment limitation or to reduce the
size of the equipment.
Reduction in the height of antenna
• For the transmission of radio signals, the antenna
height must be multiple of λ/4 ,where λ is the
wavelength.
λ = c /f
example: The minimum antenna height required to
transmit a baseband signal of f = 10 kHz is calculated
as follows :
The antenna of this height is practically impossible to

install.
• Now, let us consider a modulated signal at
f = 100 MHz
The minimum antenna height is given by,
• Minimum antenna height=λ/4=c/4f =750 cm.
This antenna can be

easily installed
practically. Thus,
modulation reduces
the height of the
antenna.
Avoids mixing of signals
• If the baseband sound signals are transmitted without
using the modulation by more than one transmitter,
then all the signals will be in the same frequency range
i.e. 0 to 20 kHz . Therefore, all the signals get mixed
together and a receiver can not separate them from
each other.
• Hence, if each baseband sound signal is used to
modulate a different carrier then they will occupy
different slots in the frequency domain (different
channels). Thus, modulation avoids mixing of signals.
Power Radiation
 Radiated power ∝ (1/𝜆)2
Without modulation, the frequency of signal is
low, wavelength will large and Power will low.
If Power is very low, it will not reach destination
and noise can also be added in the channel.
If we used modulation, the frequency of
modulated signal is high and radiated power will
become high.
The modulation process increases the frequency of the
signal to be transmitted. Therefore, it increases the range
of communication.
Narrow banding
 Narrow banding does not mean that reducing the
bandwidth. Bandwidth will remain same.
 It means that making wavelength same.
 Example: We calculated the height of antenna based
on wavelength. Without modulation, each frequency
will need antenna to communicate.
 If we use modulation, the wavelengths of many
signals will be the same(not exactly).
 By using modulation, antenna size requirement
becomes same.
Multiplexing
Multiplexing is the process of sending several

signals through a common channel.
Most of the television channels use the same
frequency range(bandwidth).
How multiple signals go without overlapping
each other?
Baseband and Carrier Communication
• Baseband signal: is message signal (information

bearing signal) delivered by the information
source or the input transducer .it is usually low
frequency signal.
• Communication that uses modulation to shift

the frequency spectrum of message signal is
known as carrier communication.
What are the Basic Types of
Analogue Modulation Methods ?
Consider the carrier signal below:
sc(t ) = Ac(t) cos( 2πfct + θ )
1. Changing of the carrier amplitude Ac(t) produces

Amplitude Modulation signal (AM)
2. Changing of the carrier frequency fc produces

Frequency Modulation signal (FM)
3. Changing of the carrier phase θ produces

Phase Modulation signal (PM)
What are the Basic Types of
Amplitude Modulation Methods ?
1. Conventional Amplitude Modulation (DSB-FC)

(Alternatively known as Full AM or Double
Sideband with full carrier (DSB-FC) modulation
2. Double Side Band Suppressed Carrier (DSB-SC)
modulation
3. Single Sideband (SSB) modulation
4. Vestigial Sideband (VSB) modulation
Example of AM transmitter
Example of AM (radio) Receiver
Thank you
L#3: Amplitude Modulation

Conventional Amplitude Modulation (Full AM)
AM is the process of varying the instantaneous amplitude
of carrier signal accordingly with instantaneous amplitude
of message signal.
• If sm(t) is the message signal and
carrier signal, then AM signal is:
S (t )  Ac . cos(ct )  sm (t ). cos(ct )
Modulating signal (information signal) can also be expressed as:
The amplitude-modulated wave can be expressed as
By substitution
Therefore The full AM signal may be written as
Where,
- modulation index
Using,
Frequency spectrum of Amplitude Modulation
Fourier transform of a cosine signal cos (ωct) consists of
two impulses at ωc and –ωc as
cos(ct )   [ (  c )   (  c )]
So, Ac cos(ct )  Ac [ (  c )   (  c )]
In general, AM wave :
S (t )  Ac . cos(ct )  sm (t ). cos(ct )
sm (t )  Sm ( )

 1 jct
sm (t ). cos(ct )  sm (t )  e  e 
 jct 

2
sm (t ). cos(c t )  S m (  c )  S m (  c )
1
2
So, the Fourier transform of AM wave is:
S ( )  S m (  c )  S m (  c )  Ac [ (  c )   (  c )
1
2
Considering baseband signal as,
mAc
S ( )  Ac [ (  c )   (  c )   {  (c  m )}   {  (c  m )} 
2
mAc
 {  (c  m )}   {  (c  m )}
2
S(ω)
πAc
mπAc/2 BW=2fm
LSB
USB
-(ωc+ ωm) ω= -ωc -(ωc- ωm) ω=0 ωc- ωm ω= ωc ωc+ ωm
BW
• Frequency spectrum of AM comprises of:
• Carrier frequency ωc .
• A lower side band whose highest frequency component is present at ωc-ωm
• An upper side band whose highest frequency component is present at ωc+ωm
Because of the two side bands in the frequency spectrum it is often called Double
Sideband -full carrier(DSB-FC)
The information in the base band (information) signal is duplicated in the LSB and USB and
the carrier conveys no information.
Modulation index or percentage of modulation
• m is merely defined as a parameter, which determines

the amount of modulation.
• What is the degree of modulation required to establish a
desirable AM communication link?
Answer is to maintain m<1.0 (m<100%).
• This is important for successful retrieval of the original

transmitted information at the receiver end.
Modulation index (m) is defined as: Am

m
Ac
Modulation index
Amax  Amin
Am 
2
Amax  Amin Amax  Amin
Ac  Amax  Am  Amax  
2 2
Amax  Amin
m
Amax  Amin
Effect of Modulation Index
m=0.5, called under modulation
m=1.0, called 100% modulation
If the amplitude of the

modulating signal is
higher than the carrier
m>1, amplitude, severe
Called over distortion to the
modulation modulated signal will
occurs.
Power distribution in AM
2
Acar ( Ac / 2 ) 2 Ac2
Pc   
R R 2R 2
ASB (mAc / 2 2 ) 2 m 2 Ac2
PLSB / USB   
R R 8R
Power distribution in AM
Now, the information are

contained in AM sidebands only.
Thus the fraction of total power
is used to transmit the
information.
 m12 m22 m32 

For multi tone modulation: Pt  Pc 1          
 2 2 2 
The percentage of total power carried by the sidebands is called the transmission efficiency.
For single tone modulation

Observations
1. If m=0 Pt=Pc no modulation occurs
2. As m PSB ( m P )
2
c , But Pc remain same as it is
4
independent of m
3. If m=1, Pt=1.5Pc and efficiency will be 33.33 %
4. Over-modulation, i.e. Am>Ac , should be avoided

because it will create distortions.
In terms of power efficiency, for m=1 modulation, only 33% power

efficiency is achieved which tells us that only one-third of the
transmitted power carries the useful information.
Numerical Problems
P1. An unmodulated AM transmitted power=100W. Find
AM transmitted power with 100% modulation.
Solution:
PC=100 W, m=1, Pt=?
Pt=(3/2)*100=150
Pt=150 W
Numerical Problems
P2. For an AM signals, total side band power=200 W with
50 % of modulation. Find the total transmitted AM power.
Solution:
m=0.5
Pt=PC+PSB
=0.2
200
  0.2
PT
Pt=1000 W
THANK YOU
L#4: Generation of AM

Generation of AM
Methods of AM Generation
1. Low level Amplitude Modulation
2. High level Amplitude Modulation

Low level AM Modulation
• Modulation done at low power.
• At low power level small power is associate with the carrier signal
and the modulating signal.
• Therefore power amplifier is required to boost the amplitude
modulated signals up to the desired output level.
• Wideband power amplifier is used just to preserve the sideband of
modulated signal.
• E.g. Square law diode modulation and switching modulation
method
Block diagram of low level AM modulator

High level AM Modulation
• Modulation done at low power.
• The modulating and carrier signal first power up and then applied
to the AM high level modulator.
• E.g. Collector modulation method
Block diagram of high level AM modulator

Generation of AM signal
(a) Square law modulator
Square law modulator
• A DC battery is connoted across diode to get fix operating
point on V-I characteristic of diode.
• The working of circuit may be explained by considering the
fact when two different frequency are passed through a
diode, the process of AM take place.
• When carrier and modulating frequencies are applied at the
input of diode, then different frequency term appear at the
output of diode.
• These different frequency terms are applied across a tuned
circuit which is tuned to the carrier and has narrow
bandwidth just pass to two side bands along with the carrier
and reject the other frequencies.
• Hence AM wave produced.
Therefore,
Where, a and b are constant.
The LC tuned circuit acts as a band pass filter. The circuit
is tuned to frequency fc and its bandwidth is equal to
2fm.
Hence the output voltage contains only two useful terms:
Let, m(t) be the message signal
Carrier signal, c(t)=A.cos(2πfct) With fc=1000 kHz
Modulated signal
Band pass filter

Switching modulator
• A typical switching modulator can be realised with the help of the
arrangement shown in fig 3.7, where the switching action is provided by
a non-linear device like diode.
• Diode has been assumed to be ideal in the sense that it offers no
resistance in the forward direction and infinite resistance in the reverse
direction.
The input voltage can be written as
Where
The resulting load voltage V2(t) is
Thus the load voltage V2(t) varies periodically between the values V1(t) and zero at
a rate equal to the carrier frequency fc.
Switching modulator
• Thus, assuming the modulating signal to be weak compared with the
carrier wave, we have effectively replaced the non-linear behaviour of
the diode by an approximately equivalent pice-wire linear time
varying operation.
Where gp(t) is a periodic pulse train of duty cycle equal to one half,
and period To = 1/fc as shown in fig
Switching modulator
The function gT0(t) can be expressed in Fourier series as
Switching modulator
The function gp(t) can be expressed in Fourier series as
Therefore,
Observations:
Thank you
L#5: Demodulation of AM signal

Demodulation of Conventional DSB-AM Signals
Coherent (i.e., synchronous) demodulation (or detection) is a method
to recover the message signal from the received modulated signal that
requires a carrier at the receiver. This carrier signal must match in
frequency and phase to the received signal.
Demodulation consists of multiplication of the incoming modulated
signal m(t)cos ωct by a carrier cos ωct followed by a low pass filter
This can be verified in the time domain by observing e(t) as follows:

Finding the Fourier transform of the signal e(t)
Signal e(t) consists of two components (1/2)m(t) and

(1/2)m(t)cos2ωct, with their non-overlapping spectra
The spectrum of the second component, being a modulated signal
with carrier frequency 2fc, is centred at ±2fc
Higher components are suppressed by low-pass filter

• On the other hand, the desired component (1/2)M(f), being a
low-pass spectrum (centred at f = 0) passes through the filter
unharmed, resulting in (½)m(t)
•You can get rid of the inconvenient fraction ½ in the
output by using a carrier 2cosωct instead of cosωct.
• This method of recovering the baseband signal is called
synchronous detection or coherent detection where we use
a carrier of exactly the same frequency(same phase) as the
carrier used for modulation.
But . . . Amplitude Modulation has the advantage of

not requiring coherent detection methods. Non-
coherent methods can be used which are much
simpler to implement.
Demodulation of AM wave
• Two methods
1. Square law detectors
2. Envelope detectors
SQUARE-LAW DEMODULATOR
 The Square law detector circuit is used for detecting modulated
signal of small magnitude, so that the operating region may be
restricted to the non-linear portion of V-I characteristic of the
device.
 It may be observed that the circuit is very similar to the square law
modulator. The only difference lies in the filter circuit.
 In a square law modulator, the filter used is a BPF (Band pass
filter), where as in a square law detector , low pass filter (LPF) is
used.
 In the circuit, the DC supply voltage VAA is used to get the fixed operating point
in the non-linear portion of the diode V-I characteristic, since the operation is
limited to the non-linear region of the diode characteristic, the lower portion of
the modulated wave form is compressed.
• The produce envelope applied distortion. Due to this, the average value of the
diode current is no longer constant.
i=av(t)+bv2(t)
i=av(t)+bv2(t) v(t)=Ac (1+m cosωm t) cosωc t
i=a.Ac (1+mcosωm t) cosωc t +b. Ac 2 (1+m cosωm t) 2 cos 2 ωc t
i=a.Ac (1+mcosωm t) cosωc t +

b.A2c (1+m2 cos2 ωmt+ 2m cosωmt) cos 2 ωc t
i=a.Ac (1+mcosωm t) cosωc t +

bA2c (cos2ωct +m 2cos2ωct .cos2ωm t + 2m.cos2ωct. cosωmt)
We get by above equation:-

2ωc , ωc – ωm , ωc + ωm , ωc – 2ωm and ωm
Hence the diode current i containing all these frequencies

is pass the frequencies below or up to modulating
frequency ωm and reject the other higher frequency
component.
Therefore the modulating signal with ωm frequency is
recovered from the input modulated signal.
Envelope Detector
The major advantage of conventional AM signal transmission is the ease in which
the signal can be demodulated .
There is no need for a synchronous demodulator.
One way to demodulate an AM wave: envelope detector: Consist of a diode and a
resistor-capacitor (RC) filter.
The operation of envelope detector:
 On a positive half-cycle of the input
signal, the diode is forward-biased and
the capacitor C charges up rapidly to the
peak value of the input signal.
When the input signal falls below this value, the diode becomes reverse biased
and the capacitor C discharges slowly through the load resistor Rl.
The discharging process continues until the next positive half-cycle.
When the input signal becomes greater than the voltage across the capacitor, the
diode conducts again and the process is repeated.
Envelope Detector
Choosing the RC Time Constant in Envelope Detector
How the envelope is constructed
Time constant τ = RC too short
Clipping portion of negative cycle
Time constant τ = RC too long

It is desired to keep the time const. R-C very high as compared to

time period of the carrier wave in order to minimize spikes or
fluctuation on the deleted envelop.
On the other hand, if it is kept too high, the discharge curve becomes
approximately horizontal. In that case, negative peaks of the deleted
envelop may be completely or partially missing. The recovered
baseband signal is distorted at negative peaks as shown in fig. This
type of distortion is known as diagonal clipping. An optimum value
of the time const has to be chosen which provides a compromise
between two following facts:
i) The spike or fluctuation in the detected envelope should be
minimum.
ii) Negative peaks of the detected envelop should not be missed even
partially, (i.e., diagonal clipping)
The above two factors can be compromised if we make R-C large

enough so that it follows the entire envelope of rectified
modulated wave and at the same time no portion of the envelop
missed. The negative peaks will be totally maintained in the
detected output and no portion will be clipped, if the discharge
curve follows the modulation envelop.
For this, the rate of discharge of capacitor  rate of decrease of
modulation envelop.
Let us desire a relation which satisfies this condition. Consider a

amplitude modulation signal as
v(t )  vc (1  m. cos wmt ). cos wct

Rate of decay of Envelop:
The envelop of the AM wave is given as
e  vc (1  m. cos wmt )
The rate of charge of this envelope is
d
 (e)  vc .m.wm . sin wm t
dt
The negative slope indicated the decay of the voltage.
The slope at the instant to will be
de

dt  vc .m.wm . sin wmt0
t  t0
This is the decay of the envelope at an instant t=t0.
Rate of discharge of capacitor:

The capacitor C discharges exponentially. Let us assume that the
capacitor starts discharging at t=t0. Before the capacitor starts
discharging at this instant, it has been charged to a voltage equal
to the value of the envelope at t=t0. This initial voltage across the
capacitor is equal to the envelope voltage “e” at t=t0 and is given
by,
e0  (e)t t0  vc (1  m. cos wmt0 )
The capacitor discharges exponentially from the initial voltage ‘e0’.
The capacitor voltage at any instant t is given by
t t0

ec  e0e RC
The rate of change of capacitor voltage (slope of the discharge

curve) is
t t
d d  0
 (ec )   [e0 e RC ]
dt dt
t t
1  0
 e0 .e RC
RC
This rate of change at t=t0 is given by
d e0
 (ec )t t0 
dt RC
Putting the value of e0 from eqn
de
 (ec ) vc
dt  (1  m. cos wm t 0 )
t  t0 RC
To avoid diagonal clipping the slope of the discharge curve at t=t0, the
above eqn must be equal to or greater than the envelope decay rate
eqn
vc
(1  m. cos wm t0 )  vc .m.wm . sin wm t0
RC
1 wm .m. sin wm t 0

RC 1  m. cos wm t 0
At any instant, t 1 wm .m. sin wm t


RC 1  m. cos wm t
The condition for maximizing is:

d wm .m. sin wm t cos wmt  m
[ ]0
dt 1  m. cos wm t
sin wmt  1 m 2
1 wm .m 1 m 2

RC 1 m 2
1 wm .m

RC 1 m 2
1
 wm .m For m<<1
RC
Advantages
It is simple to implement.
Demodulation of AM signals can be done using simple circuits consisting of
diodes
AM receivers are very cheap as no specialized components are needed.
AM waves can travel a long distance.
AM wave have low bandwidth.
Disadvantages
 An AM signal is not efficient in terms of its power usage. Power wastage takes
place in DSB-FC transmission.
It is not efficient in terms of its bandwidth. It requires a bandwidth equal to
twice that of the highest audio frequency. In AM sidebands contain the signal. The
power in sidebands is the only useful power. For 100 % modulation, the power
carried by sidebands is only 33.33 %. The power carried by the AM wave
decreases with the decreases in modulation index.
AM detectors are sensitive to noise hence an amplitude modulated signal is
prone to high levels of noise.
Reproduction is not high fidelity. For high fidelity (stereo) transmission
bandwidth should be 40 kHz. To avoid interference the actual bandwidth used by
AM transmission is 10 kHz.
Thank you
L#6: Double sideband suppressed
carrier (DSB-SC)

Double sideband suppressed carrier (DSB-SC)
From AM Spectrum:
Carrier signal ωc carries no information about ωc.

Carrier signal consumes a lot of power more than 50 %.
Question: Can one suppress the carrier ?

Ans. Yes, just transmit two side bands (i.e DSB-SC)
But what is the penalty?
System complexity at the receiver.

DSB-SC theory
General expression of AM wave:
The modulated signal is simply the product of these two
s (t )  Ac cos(c t ) Am cos(m t )
 Ac Am cos(c t ) cos(m t )
since cos A cos B 

1
cos( A  B)  cos( A  B) 
2
Am Ac Am Ac
 cos(c  m )t  cos(c  m )t
2  2  
USB LSB
DSB-SC theory
sc (t )  Ac cos ct
sm (t )  Am cos  mt X s(t )  Ac cos( ct ) Am cos( mt )
Frequency Spectrum of a DSB-SC AM Signal

Generation and Detection of DSB-SC
• The simplest method of generating a DSB-SC signal is

merely to filter out the carrier portion of a full AM (or
DSB-FC) waveform.
• Given carrier reference, modulation and demodulation

(detection) can be implemented using product devices or
balanced modulators.
BALANCED MODULATOR
Sm(t) S1(t)
AM Modulator 1
Sm(t) Accos(ct)
S(t)
Carrier 
DSB-SC
Accos(ct)
AM Modulator 2
-Sm(t) S2(t)
BALANCED MODULATOR
• The two modulators are identical except for the sign

reversal of the input to one of them. Thus,
s1 (t )  Ac (1  m cos( mt )) cos(ct )
s2 (t )  Ac (1  m cos( mt )) cos(ct )
s(t )  s1 (t )  s2 (t )
 2mAc cos( mt ) cos( ct )
Balanced modulator
A circuit which can produce an output which is the product of two
signals input to it is called product modulator. Such an output
when the inputs are the modulating signals and carrier signal is a
DSB-SC. One such product modulator is a balanced modulator.
Balanced modulator
For diode D1 the modulator v-I relationship becomes:
Similarly for diode D2:

Balanced modulator
Now,
This voltage is input to the band pass filter centre

frequency fc and bandwidth 2fm. Hence it allows the
component corresponding to the second term of vi which
is the desired DSB-SC signal.
Ring modulator
 One switching modulator known as the ring modulator is shown in Fig
below.
 During the positive half cycles of the carrier, diodes D1 and D3 conduct,
and D2 and D4 are open. Hence terminal ‘a ‘ is connected to ‘b’, and ‘c’ is
connected to ‘d’.
 During the negative half cycles of the carrier D1 and D3 are open and D2
and D4 are conducting, thus connecting terminal ‘a ‘ to ‘d’ and terminal ‘b’
to ‘c’.
Ring modulator
• Hence the output is proportional to m(t) during

the positive half cycle and to - m(t) during the
negative half cycle. In effect m(t) is multiplied by
a square pulse train w0(t) .
• The Fourier series representation of this square is
given by
Ring modulator
When this waveform is passed through a band pass filter

tuned to fc , the filter output will be the desired signal (4/π)
m(t)cosωc. In this circuit there are two inputs m(t) and
cos2π fc(t). The input to the band pass filter does not
contain either of these inputs. Consequently, this circuit is
an example of a double balanced modulator.
Demodulation of DSB-SC signal
The process of extracting an original message signal from DSBSC
wave is known as detection or demodulation of DSBSC. The
following demodulators (detectors) are used for demodulating
DSBSC wave.
(1) Coherent/synchronous Detector
(2) Using envelope detector after carrier reinsertion

Coherent/synchronous Detector
• Since the carrier is suppressed the envelope no longer represents

the modulating signal and hence envelope detector which is of
the non-coherent type cannot be used directly.
v(t )  s (t ) cos( c t )  2mAc cos( mt ) cos( c t )cos( c t )
Am
2 Ac cos( mt ) cos 2 ( c t )
Ac
 1  cos 2 c t 
 2 Am cos( mt ) 
 2 
 Am cos( mt )  Am cos( mt ) cos( 2 c t )
since sm (t )  Am cos( mt )
 sm(t)  sm(t ) cos ( 2 c t)
 
Unwanted erm(
t removed by LPF)
• It is necessary to have synchronization in both
frequency and phase between the transmitter
(modulator) & receiver (demodulator), when DSB-SC
modulation ,which is of the coherent type, is used.
Both phase and frequency must be known to
demodulate DSB-SC waveforms.
THANK YOU
L#7: Demodulation of Double
sideband suppress carrier (DSB-SC)

Effect of phase and frequency errors in
synchronous detection
• the frequency and phase of the local oscillator

signal in coherent detection method at the
receiver end, must be identical to the
transmitted carrier signal. Any kind of the
discrepancy in frequency or phase produces a
distortion in the detected output at the
receiver end.
Mathematical Expression
S(t)= x(t ) cos c t
synchronous detector provides

ed (t )  x(t ) cos c .cos[(c   )t   ]
1
ed (t )  x(t ){cos[( )t   ]  cos[(2c   )t   ]}
2
Now, when this signal is allowed to pass through a low
pass filter (LPF) having a cut-off frequency ωm the terms
centered around ±2ωc are filtered out.
So, 1
e0 (t )  x(t ) cos[(  )t   ]
2
The base band signal x(t) is multiplied by a slow-time

varying function cos(∆ωt+φ) which distorts the message
signal.
consider the following special cases:
(i) When the frequency error ∆ω and phase error φ are
both zero, then
1
e0 (t )  x(t )
2
This means there is no distortion in the detected output
signal
(ii) When there is only the phase error i.e.,
∆ω=0 but, φ≠0
1
Then, e0 (t )  x(t ) cos 
2
This shows that the output signal is multiplied by cosφ
(iii) When there is only the frequency error, i.e.
∆ω ≠ 0 and, φ=0
1
e0 (t )  x(t ) cos(  )t
2
the multiplying factor cos(∆ω)t is time-independent and
produce distortion in the detected output signal.
(iv) When both the errors are non-zero, i.e.

∆ω ≠ 0 and, φ ≠ 0
in this case, the constant phase error provides attenuation and
frequency error produced distortion in this detected output signal.
Hence we get an attenuated and distorted output signal at the
receiver end.
Carrier acquisition in DSB-SC System or
synchronization techniques in DBS-SC system
The phase and the frequency of the locally generated

carrier signal in synchronous detector is very critical.
Precision phase and frequency control of the local carrier
requires an expensive and a complex circuitry to the
receiver end. Some important synchronization techniques
are given as under:
1. Pilot Carrier
• A small amount of carrier signal known as pilot carrier is
transmitted along with the modulated signal from the
transmitter. This small amount of carrier signal is called Pilot
Carrier. This pilot carrier, separated at the receiver by an
appropriate filter, is amplifier, and is used to phase lock the
locally generated carrier at the receiver. The phase locking
provides synchronization. This system, where a weak carrier is
transmitted along with the DSB-SC signal is also referred to as
partially suppressed carrier system as the carrier is not totally
suppressed. The process in which a large carrier is transmitted
along with DSB-SC signal is known as amplitude modulation.
The large carrier simplifies the reception system. The DSB-SC
with partially suppressed carrier is equivalent to an over
modulated AM signal.
2. Costa’s Receiver
• This system used for synchronous detection of DSB-SC
signal has been shown in figure (next page). This system
has two synchronous detection-
 one detector is fed with a locally generated carrier signal
which is in phase with the transmitted carrier signal. This
detector circuit is called in-phase coherent detector or I-
channel.
 The other synchronous detector employs a local carrier
which is in phase quadrature with the transmitted carrier
signal and is called Quadrature phase coherent detector or
Q-channel. On combining, the two detectors constitute a
negative feedback system which synchronizes the local
carrier signal with the transmitted carrier signal.
Block diagram of Costa’s Receiver
Operation
• let us assume that the local carrier signal is synchronized with the
transmitted carrier signal and =0, the output of the I-Channel is
the desired modulating signal, but the output of the Q-channel is
zero(Since sin =0) because of the quadrature null effect. Now,
assuming that the local oscillator frequency drifts slightly i.e.,  is
a very small non-zero quantity, I-channel output will be almost
unchanged, but Q-channel output now is not a zero, rather some
signal appear at its output and is proportional to sin . Thus, the
output of the Q-channel,
(i) is proportional to  (since sin=0 for small )
(ii) would have a polarity same as the I-channel for one direction
of phase shift in local oscillator, whereas, the polarity would be
opposite to I-channel for the other direction of phase shift.
Operation
The phase discriminator provides a dc control signal which may be
used to correct local oscillator phase error. The local oscillator is a
voltage controlled oscillator (VCO). Its frequency may be adjusted by
an error control dc signal.
Limitation
The costa’s receiver cases phase control when there is no
modulation i.e., x(t)=0. The phase control re-establishes itself on
the reappearance of modulation. However, the reestablishment is so
fast that distortion is not perceptible in case of voice
communication.
Squaring Loop
the receiver signal is squared by a squaring circuit as shown in figure.
The output of the squarer will be given as
[ A.x(t ) cos ct ]2  A2 x 2 (t ) cos 2ct
For simplicity let us assume that x(t) is a single tone sinusoidal denoted as cosmt i.e.,
x(t )  cos mt
Squaring Loop
Then the output of the squarer becomes
[ A cos ct. cos mt ]2  A2 cos 2mt (t ) cos 2ct
A2
 (1  cos 2m )(1  cos 2ct )
4
A2
 (1  cos 2mt  cos 2ct  cos 2ct cos 2mt )
4
The term cos2c t can be obtained by using a narrow band filter

centered at  2c. This frequency 2c is kept constant by tracking
through a phase locked loop (PLL). The PLL uses negative feedback
techniques to provide a constant frequency signal, cos(2ct).
The VCO output is frequency divided by 2, to yield a synchronized
local carrier of frequency c. This local carrier signal is used in
synchronous detector. The frequency division can be accomplished by
using a bistable multivibrator.
Envelope Detection after Suitable Carrier
Re-insertion
The other possible method of demodulating DSB-SC signal is
by inserting a carrier generated at the receiver and with the
help of a local oscillator. However, the phase and the
frequency of the re-inserted carrier must be properly
synchronized with those at the transmitter end in order to
avoid distortion. We know that if we insert a sufficient carrier
of same frequency and phase to DSB-SC signal, it converts
DSB-SC signal into a conventional AM wave. Now, this AM
wave is demodulated by an envelope detector. However, phase
and frequency errors will result in similar type of distortion
are obtained in coherent detection.
Lets us consider that the receiver DSB-SC signal is
expressed by
s (t )  c(t ).x(t )
Or s(t )  A cos( 2f ct ).x(t )
Let us assume A=1,
s(t )  cos(2f c t ).x(t )

The inserted carrier at the receiver will be
c' (t )  A cos( 2f c t   )
Where = amount of phase discrepancy
The resulting signal will be
r (t )  s (t )  c' (t )
r (t )  s(t )  c' (t )  cos( 2f c t ) x(t )  A cos( 2f c t   )

r (t )  x(t ) cos(2f c t )  A cos(2f c t ) cos   A sin( 2f c t ) sin 
r (t )  [ x(t )  A cos  ] cos( 2f c t )  ( A sin  ) sin( 2f c t )
r (t )  e(t ) cos[( 2f ct )   (t )]

Where
e(t )  [ A  x(t )] 2 A x(t )[1  cos  ]
2
A sin 
 (t )  tan [
1
]
x(t )  A cos 
So, r (t )  e(t ){cos[( 2f ct )   (t )]}

It may be observed that e(t) is the envelope of the
resulting signal r(t).
Also, if we take =0, then envelope will be given by
e(t )  A  x(t )
Hence, modulating signal x(t) can be recovered
from r(t) using an enveloped detector since the r(t)
is basically a conventional AM wave given by
r (t )  [ A  x(t )] cos( 2f ct )
This is however possible only

when [ A + x(t) ]  0 for all values of t.
It is possible only
when the modulation index m is less than unity.
If   0, then the phase error exists between the two
carriers. It is given as
1
2 x(t ) x(t ) 2 2
e(t )  A[1  cos   { } ]
A A
If A|x(t)|, then we have

e(t )  A  x(t ) cos 
The desired signal output will thus be x(t) cos.
If =0 and there is a difference in frequency f between
the two oscillators, then the envelope of the resulting
signal r(t) will be given by
e(t )  A  x(t ) cos[ 2ft] for A>>|x(t)|
Advantages and disadvantages of DSB-SC system
Advantages
It provides 100% modulation efficiency.
Due to suppression of carrier, it consumes less power.
It provides a larger bandwidth.
disadvantages
It involves a complex detection process.
Using this technique it is sometimes difficult to recover
the signal at the receiver.
It is an expensive technique when it comes to
demodulation of the signal.
DSB-SC technique allows us to have a transmission that reduces
overall power consumption rate, thereby ensuring a stronger signal at
the output.
Thank you
QUADRATURE AMPLITUDE
MODULATION (QAM)

WHAT IS QAM?
 A form of modulation which is widely used for
modulating data signals onto a carrier used for
radio communications.
 QAM is a signal in which two carriers shifted in
phase by 900 are modulated.
 The resultant output consists of both amplitude
and phase variations.
 Hence it may also be considered as a mixture of
amplitude and phase modulation.
WHY QAM?
• The main aim is to save the bandwidth.
HOW?
Double sideband(DSB) even with a suppressed carrier
occupies twice the bandwidth of the modulating signal.
This is wasteful of the available frequency spectrum.
QAM places two independent double sideband
suppressed carrier signals in the same spectrum as one
ordinary double sideband suppressed carrier signal.
TYPES OF QAM
 It exists in both analogue and digital formats.
 The analogue versions of QAM are typically used to
allow multiple analogue signals to be carried on a single
carrier.
 It combines phase modulation and amplitude modulation
in a form of modulation known as quadrature amplitude
modulation, QAM.
 Digital formats of QAM are often referred to as
“Quantised QAM”.
 It combines phase shift keying and amplitude keying in a
form of modulation known as quadrature amplitude
modulation, QAM .
QAM THEORY
Quadrature amplitude theory states that both
amplitude and phase change within a QAM signal.
The basic way in which a QAM signal can be
generated is to generate two signals that are 90°
out of phase with each other and then sum them.
This will generate a signal that is the sum of both
waves, which has a certain amplitude resulting from
the sum of both signals and a phase which again is
dependent upon the sum of the signals.
QAM THEORY….
• As there are two RF carrier signals that can be
modulated, these are referred to as the I – (In-)
phase and Q - Quadrature signals.
• The I and Q signals can be represented by the
equations below:
I = A cos(Ψ)
Q = A sin(Ψ)
It can be seen that the I and Q components are
represented as cosine and sine. This is because the
two signals are 90° out of phase with one another.
QAM MODULATOR
• The modulator is used to encode the signal, often
data, onto the radio frequency carrier that is to be
transmitted.
• The QAM modulator essentially follows the idea
that can be seen from the basic QAM theory where
there are two carrier signals with a phase shift of
90° between them.
• These are then amplitude modulated with the two
data streams known as the I or In-phase and the Q
or quadrature data streams.
QAM MODULATOR
u(t) = Acm1(t) cos 2πfct + Acm2(t) sin 2πfct

QAM MODULATOR ……
• The two resultant signals are summed and then
processed as required in the RF signal chain, typically
converting them in frequency to the required final
frequency and amplifying them as required.
• It is worth noting that as the amplitude of the signal
varies any RF amplifiers must be linear to preserve
the integrity of the signal.
• Any non-linearities will alter the relative levels of the
signals and alter the phase difference, thereby distorting
the signal and introducing the possibility
of data errors.
QAM DEMODULATOR
• The QAM demodulator is very much the
reverse of the QAM modulator.
• The signals enter the system, they are split and
each side is applied to a mixer.
• One half has the in-phase local oscillator
applied and the other half has the quadrature
oscillator signal applied.
QAM DEMODULATOR
THANK YOU
L#9:
Single Sideband suppress carrier
(SSB-SC) Modulation

Introduction
• Standard AM and DSB-SC techniques are wasteful of
bandwidth because they both require transmission
bandwidth of 2fm Hz, where fm is the bandwidth of the
baseband modulating signal m(t).
• In both cases the transmission bandwidth is occupied by
the upper sideband (USB) and lower sideband (LSB)
• USB and LSB are uniquely related to each other, as they
are symmetric w.r.t fc.
• Therefore, to transmit information contained within m(t)
we used to transmit only one side band.
•As far as demodulation is concerned, we can coherently
demodulate SSB (as we did the DSB-SC signal) by
multiplying SSB with cos(ωct) followed by LPF.
Frequency domain representation of
SSB signals
Frequency domain representation of SSB
signals for single tone signal
Time domain description of SSB-SC signal
SSB-SC with single tone modulating signal:
x(t )  cos mt
The spectrum of this modulating signal is a pair impulses at
ω=ωm
carrier signal= cosct.
from the spectrum of LSB corresponds to a time

domain signal cos(c-m)t.
cos(c  m )t  cos mt. cos ct  sin mt. sin ct

Similarly, the expression for single tone SSB-SC with USB
may be given as
cos(c  m )t  cos mt. cos ct  sin mt. sin ct
So,
s(t )SSB SC  cos mt. cos ct  sin mt. sin ct
Now, sinct and sinmt may be written as

sin mt  cos( mt  )

sin c t  cos(c t  )
2 2
• Thus the sine terms can be obtained from the
corresponding cosine terms, by giving a phase
shift of (-/2).
• Thus, in a general modulating signal x(t); if all
frequency components are shifted by (-/2), it
may lead to SSB-SC signal.
s(t ) SSBSC  x(t ) cos ct  xh (t ) sin ct
Where xh(t) is a signal obtained by shifting the phase of
every component present in x(t) by (-/2).
+sign for LSB and –sign for USB.
Hilbert Transform
When the phase angles of all frequency components of a
given signal are shifted by 900, the resulting frequency in
the time domain is known as the Hilbert Transform of the
signal.
It should be noted Hilbert Transfrom of a signal does not
change the domain as compared to Forward Transform
which changes the signal from time domain to frequency
domain.
H/W
x(t) (-/2 Phase shifter) xh(t) Here x(t) signal is
x(w) xh(w)
passed through phase
shifter H(W) and the
o/p is xh(t).
Block diagram of Hilbert Transform
The characteristics of the system are given by:
(i) The magnitude of the frequency components present in
x(t) remains unchanged, when it passed through the system
i.e., |h(w)|=1 and
(ii) The phase of the positive frequency components is
shifted by -/2 and Phase of the negative frequency
components is shifted by +/2. |h(w)| and Q() are plated
in fig below by continuous and doted line respectively.
H()
|H()|
+/2
Q()

-/2
Amplitude and Phase response of phase shifter

So, the transformer function of the system is given by
iQ( ) iQ( )
H ( ) | H ( ) |e  1.e
Now, 
Q( )   ;  0
2

 ;  0
2

j
H ( ) e 2  j for   0

j
e 2
j for   0
Hence, H ( )
1 0
J
 1  0
 -sgn( )
H ( )   j sgn(  )
 The response xh(t) or xh() of the system is given by

X h ( )  x( ).H ( )
 -j x( ). sgn(  )
Talking the inverse Fourier Transform on the both side we
get,
X h (t )  F [-j x( ). sgn(  )]
1
Now, the time domain of sgn() is given by
1
  sgn(  )
j t
h(t )  F 1[-j sgn(  )]
1

t
And x(t) x(w)
Then, 1 1
xh (t )  [ x(t )  ]
 t

1 x( ) Which is the Hilbert
  d
  t   Transform of x(t)
Properties if Hilbert Transform
• (i) A signal x(t) and its Hilbert Transform xh(t)
have the same energy density spectrum.
• (ii) A signal x(t) and its Hilbert Transform xh(t)
have the same auto correction function.
• (iii) A signal x(t) and its Hilbert Transform xh(t)
are mutually orthogonal, i.e.


 x(t ) x (t )dt  0
h
• (iv) It xh(t) is Hilbert Transform of x(t), then the
Hilbert Transform of xh(t) is –x(t). That is if
H [ x(t )]  xh (t )
then
H [ xh (t )]  -x(t)
Application of Hilbert Transform
• Hilbert Transform are used:
• (i) Generation of SSB signal
• (ii) Design of minimum phase type filters.
• (iii) Representation of band pass signals.
Power saving in SSB-SC
for the upper sideband
for the lower sideband

Power in carrier  power in one sideband
% of power saving 
Total power
m2 m2
Pc 1  1
4 4
% of power saving  2
 2
 83 .33 %, for m  1
m m
Pc 1  1
2 2
Bandwidth of SSB-SC Wave
• the DSB-SC modulated wave contains two

sidebands and its bandwidth is 2fm.
• Since the SSB-SC modulated wave contains only
one sideband, its bandwidth is half of the
bandwidth of DSBSC modulated wave.
Bandwidth of SSB-SC modulated wave = 2fm/2=fm

Advantages and disadvantages
Advantages
Bandwidth or spectrum space occupied is lesser than
AM and DSB-SC waves.
Transmission of more number of signals is allowed.
Power is saved.
High power signal can be transmitted.
Less amount of noise is present.
Signal fading is less likely to occur.
Disadvantages
The generation and detection of SSB-SC wave is a
complex process.
The quality of the signal gets affected unless the SSB
transmitter and receiver have an excellent frequency stability.
Applications
• For power saving requirements and low

bandwidth requirements.
• In point-to-point communications.
• In telemetry, and radar communications.
• In military communications, etc.
THANK YOU
L#10:Modulation and
demodulation of SSB-SC signal

Generation of SSB-SC signal
We can generate SSBSC wave using the
following two methods.

Frequency discrimination method
Phase discrimination method

Frequency Discrimination Method
First generate DSB-SC wave with the help of the product modulator. Then,
apply this DSB-SC wave as an input of band pass filter. This band pass filter
produces an output, which is SSB-SC wave.
Select the frequency range of band pass filter as the spectrum of the
desired SSB-SC wave. This means the band pass filter can be tuned to either
upper sideband or lower sideband frequencies to get the respective SSB-SC
wave having upper sideband or lower sideband.
Limitations of Frequency Discrimination Method
• The Frequency Discrimination Method is useful

only if the baseband signal is restricted at its lower
edge due to which the upper and lower sidebands
are non-overlapping.
• The system is not useful for video communication
where baseband signal starts from dc.
• The baseband signal must be appropriately related
to carrier frequency. In fact, the design of the
bandpass filter (BPF) becomes difficult if the
carrier frequency is quite higher that the bandwidth
of the baseband signal.
Phase discrimination method
This block diagram consists of two product modulators, two phase

shifters, one local oscillator and one summer block. The product
modulator produces an output, which is the product of two inputs.
The phase shifter produces an output, which has a phase lag of with
respect to the input.
Operation
s1 (t )  x(t ). Ac . cos( 2f ct )
s2 (t )  xh (t ). Ac . sin( 2f ct )
s(t )  x(t ). Ac . cos( 2f ct )  xh (t ). Ac . sin( 2f ct )
Advantages of Phase Discrimination method
• (i) it can generate the SSB signal at any

frequency, so the frequency up converter stage
is nor required.
• (ii) It can use the low audio frequencies as
modulating signal. (In filter method, this is not
possible).
• (iii) It is easy to switch from one sideband to
the other.
Drawbacks
• (i) The disadvantage is that the design of the 900
phase shifting network for the modulating signal
is extremely critical.
• (ii) This network has to provide a correct phase
shift of 900 at all the modulating frequencies
which is partially difficult to achieve.
Why is SSB not used for broadcasting?
• We have seen that there are so many advantages a SSB
system has over the DSB-FC system. Still it is not used
widely in the radio broadcasting applications. There are
two reasons for it as under;
• (i) As the SSB transmitter and receiver require excellent
frequency stability, a small frequency shift in the system
can result in degradation in the quality of the transmitted
signal. Thus, it is not possible to transmit a good quality
music using the SSB system.
• (ii) It is not possible to design a tunable receiver oscillator
with very high frequency stability. Now, with the advent of
the frequency synthesizers, this has becomes possible. But,
such receivers are too expensive.
SSB-SC Demodulator
• The process of extracting an original message
signal from SSBSC wave is known as detection or
demodulation of SSB-SC. Coherent detector is
used for demodulating SSB-SC wave.
Coherent Detector
s(t )  x(t ). Ac . cos( 2f ct )  xh (t ). Ac . sin( 2f ct )
First the received signal is multiplied by a locally
generated replica of the carrier signal. Multiplying the
formulas for upper and lower sideband SSB signals by
cos(ωct) yields
Observation
 0.5Acx(t) is the desired component.
 0.5Acx(t) cos 2ωct and 0.5Acxh (t) sin 2ωct
have spectra centered about 2ωc.
The components around 2ωc are removed by
the lowpass filter with cut-off frequency fm.
Frequency Domain Analysis of Operation
This translates the sidebands around ±ωc down to

baseband and forms M(ω) which is the desired term and
also translates them up to ±2ωc which are the terms
removed by the lowpass filter.
Phase and frequency error in coherent
detection
• In the coherent detection process, it is assumed the
ideal operating conditions in which the locally
generated carrier is in the perfect synchronization.
But in practice a phase error may arise in the
locally generated carrier wave. The detected
output will get modified due to the phase error.
1 1
vo (t )  Vc x(t ) cos( )  Vc xh (t ) sin(  )
4 4
Such a phase distortion does not have serious effects with the
voice communication. But it will have untolerable effects in the
transmission of video.
THANK YOU
L#11:Vestigial Sideband
Transmission

Vestigial sideband transmission
SSB-SC modulated signal has only one sideband frequency. Theoretically, we can get one
sideband frequency component completely by using an ideal band pass filter.
However, practically we may not get the entire sideband frequency component. Due to
this, some information gets lost.
To avoid this loss, a technique is chosen, which is a compromise between DSBSC and SSBSC.
This technique is known as Vestigial Side Band Suppressed Carrier (VSBSC) technique. The
word “vestige” means “a part” from which, the name is derived.
VSBSC Modulation is the process, where a part of the signal called as vestige is modulated
along with one sideband. The frequency spectrum of VSBSC wave is shown in the following
figure.
Along with the upper sideband, a part of
the lower sideband is also being
transmitted in this technique. Similarly,
we can transmit the lower sideband
along with a part of the upper sideband.
A guard band of very small width is laid
on either side of VSB in order to avoid
the interferences. VSB modulation is
mostly used in television transmissions.
Bandwidth of VSBSC Modulation
• the bandwidth of SSBSC modulated wave is fm.
• Since the VSBSC modulated wave contains the
frequency components of one side band along
with the vestige of other sideband, the bandwidth
of it will be the sum of the bandwidth of SSB-SC
modulated wave and vestige frequency fv.
Advantages and disadvantages
Advantages
• Highly efficient.
• Reduction in bandwidth when compared to AM and DSB-SC
waves.
• Filter design is easy, since high accuracy is not needed.
• The transmission of low frequency components is possible,
without any difficulty.
• Possesses good phase characteristics.
Disadvantages
•Bandwidth is more when compared to SSBSC wave.
•Demodulation is complex.
Applications
• The most prominent and standard application
of VSBSC is for the transmission of television
signals.
• Also, this is the most convenient and efficient
technique when bandwidth usage is
considered.
Generation of VSBSC
Generation of VSBSC wave is similar to the generation of SSBSC wave.
In this method, first we will

generate DSBSC wave with the help
of the product modulator. Then,
apply this DSBSC wave as an input
of sideband shaping filter. This filter
produces an output, which is VSB-
SC wave.
The modulating signal m(t) and carrier signal Ac.cos(2πfct) are applied as inputs to the product
modulator. Hence, the product modulator produces an output, which is the product of these
two inputs.
Therefore, the output of the product modulator is
Apply Fourier transform on both sides
DSBSC frequency spectrum

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

Demodulation of VSBSC
Demodulation of VSBSC wave is similar to the demodulation of SSBSC wave. Here, the same
carrier signal (which is used for generating VSBSC wave) is used to detect the message
signal.
Hence, this process of detection is called as coherent or synchronous detection. The VSBSC
demodulator is shown in the following figure
In this process, the message signal can
be extracted from VSB-SC wave by
multiplying it with a carrier, which is
having the same frequency and the
phase of the carrier used in VSB-SC
modulation. The resulting signal is then
passed through a Low Pass Filter. The
output of this filter is the desired
message signal
Let the VSB-SC wave be s(t) and the carrier signal is Ac.cos(2πfct).
the output of the product modulator
Apply Fourier transform on both sides

We know
So,
In the above equation, the first term represents the scaled version of the desired message
signal frequency spectrum. It can be extracted by passing the above signal through a low pass
filter.
P1: A Modulating signal 10 sin(2 x 103t) is used
to modulate a carrier signal 20 sin(2 x 104t).
Determine the modulation index, percentage
modulation, frequencies of the seideband
components and their amplitude. What will be
the band with of the modulated signal?
Solution:
(i) The modulating signal vm= 10 sin(2 x 103t)
Let us compare this with the following expression
vm= Vm 10 sin(2 x fmt)
Then, we get,Vm= 10volt, fm= 1 x 103 Hz = 1 kHz
(ii) The carrier signal vc = 20 sin(2 x 104t)
Comparing this with the expression
vc = Vc sin(2fct), we obtain
Vc  20 volt, f c  110 Hz  10 kHz

4
(a) Modulation index and percentage modulation:

Vm 10
m   0.5 and
Vc 20
% mod ulation  0.5 100  50 %
P2: The antenna current of an AM transmitter is 10
ampere when it is modulated to depth of 30% by an
audio signal. It increases to 11 ampere when
another signal modulates the carrier signal. What
will be the modulation index due to second signal?
Solution:
It is given that It1 = 10 amp, m1 = 0.3, It2 = 11 amp.
2
 I t1  m12
   1
 Ic  2
Therefore
I t1 10
Ic  1/ 2
 1/ 2
 9.78
 m  2
 (0.3) 2
1  2  1  2 
1
   
(ii) After modulating with the second signal, we have

2
m
I  I [1 
2
t2 ]2
c
t
2
It2 2 11 2
Or, mt  2[( ) 1]  2[( ) 1]
Ic 9.78
Therefore mt  0.73
mt  [m  m ]2
1
2 1/ 2
2
m2  [mt2  m12 ]1/ 2  [(0.73) 2 (0.3) 2 ]1/ 2  0.66 or 66%

P3: A given AM broadcast station transmits a total
power of 50kW when the carrier is modulated by a
sinusoidal signal with a modulation index of 0.707.
Calculate:
(i) the carrier power
(ii) the transmission efficiency, and
(iii) the peak amplitude of the carrier assuming the
antenna to be represented by a (50+j0)  load.
Solution:
The total power transmitted by the AM broadcast
station is given by
Pc (1  Px )  50 kW
Where Pc is the carrier power and
1
Px  (0.707 )  0.25
2
2
Thus, Pc (1  0.25)  50 kW
50
Or, Pc   40 kW
1.25
(ii) The transmission efficiency, , is given by
Pc .Px Px 0.25
   0.2 or 20%
Pc  Pc Px 1  Px 1.25
(iii) Also, carrier power, Ac2
Pc   40 10 3W
2  50
Therefore, peak carrier amplitude,
A c  2 X 50 X 40 X 10  2000 volts or 2kV.

3
P4: A carrier signal of 1.0 Volt amplitude and a
sinusoidal modulating signal of 0.5 V, put in series, are
applied to a square law modulator of characteristics,
io  10  kVi  k Vi mA
' 2
Where Vi is input in volts, and
k  2mA / V and k ' 0.2mA /V 2

Considering only the frequency components of the
AM signal corresponding to the carrier frequency, find
the depth of modulation in the resulting AM Signal.
Solution: Vi (t )  cos(c t )  0.5 cos(mt )
io  10  kVi  k Vi ' 2
io  10  2  10 [cos(c t )  0.5 cos(mt )] 

3
0.2  10 [cos(c t )  0.5 cos(mt )]

3 2
considering only the carrier terms, we have

io  2 10 3 cos(ct )  0.2 10 3 cos(ct ) cos(mt )
io  2 10 3 cos(ct )1  0.1cos(mt )
0. 2
m  0. 1 modulation depth will be 10 %
2
THANK YOU
Angle modulation

Basic concept: A non-linear process
Modulated wave does not look like message
wave.
Amplitude of an exponentially modulated
wave is constant.
Therefore, regardless of message signal the
average transmitted power is
 It is less sensitive to noise.

Basic concept: A non-linear process
A sinusoidal carrier signal is defined as:
For un-modulated carrier signal the total

instantaneous angle is:
Thus one can express c(t) as:
Also called exponential modulation

Types of angle modulation
Thus:
• Varying the frequency fc
Frequency modulation
• Varying the phase φc(t)

Phase modulation
Phase modulation
In angle modulation: Amplitude is constant, but
angle varies (increases linearly) with time.
Kp=phase sensitivity of the modulator
The resulting PM wave is
Instantaneous phase
• It is a type of angle modulation in which
the instantaneous frequency varied linearly
in accordance to baseband or modulating
signal.
The instantaneous frequency is
Where Kf is known as the frequency sensitivity

  i t   0
d
 i
dt
i   i dt
i   [c  k f m(t )]dt  i t  k f  m(t )dt

s (t )  A. cos c t  k f  m(t )dt 
s (t )  A. cos c t  k f  m(t )dt 
 t
 0 
FREQUENCY DEVIATION
The maximum change in
instantaneous frequency from
the average frequency ωc is
called frequency deviation.
Δω
Angle modulation viewed as PM or FM
Relationship between PM and FM
Expression for PM wave is:
Expression for FM wave is:
• In PM, the phase angle varies linearly with base band

signal x(t).
• In FM, the phase angle varies linearly with the integral
of baseband signal x(t).
•Thus, FM can be obtained from FM and the converse is

also true.
FM using PM
PM using FM
Summary
In phase modulation m(t) drives the variation of phase .

In frequency modulation m(t) drives the variation of frequency f.
APPLICATIONS OF ANGLE
MODULATION
• Radio Broadcasting
• Two way mobile radio
• Microwave communication
• TV sound transmission
• Cellular radio
• Satellite communication
Comparing Frequency Modulation to
Phase Modulation
FM PM
Frequency deviation is proportional to Phase deviation is proportional to
modulating signal m(t) modulating signal m(t)
Noise immunity is superior to PM (and of
Noise immunity better than AM but not FM
course AM)
Signal-to-noise ratio (SNR) is better than in Signal-to-noise ratio (SNR) is not as good as
PM in FM
FM is widely used for commercial broadcast PM is primarily for some mobile radio
radio (88 MHz to 108 MHz) services
Modulation index is proportional to
Modulation index is proportional to
modulating signal m(t) as well as modulating
modulating signal m(t)
frequency fm
For practical implementation reasons, analog FM is easier to generate than PM, and FM
provides better performance in most common environments. However, analog PM has
been (and continues to be) used for a few, isolated systems.
FM VS AM
1. Reduction to noise: The main advantage of frequency modulation is a reduction in
noise. As most noise is amplitude based, this can be removed by running the received signal
through a limiter so that only frequency variations remain.
FM is considered to be superior to AM.
Does not require linear amplifiers in the transmitter: As only frequency changes
contain the information carried, amplifiers in the transmitter need not be linear.
 AM use linear amplifier to produced the final RF signal.

 FM has constant carrier amplitude so it is not necessary to use linear amplifier.
Resilient to signal strength variations: In the same way that amplitude noise can be
removed, so too can signal variations due to channel degradation because it does not suffer
from amplitude variations as the signal level varies. This makes FM ideal for use in mobile
applications where signal levels constantly vary.
 The stronger signal will be capture and eliminate the weaker.
 In AM, the weaker signal can be heard in the background.
Enables greater efficiency : The use of non-linear amplifiers (e.g., class C and class D/E
amplifiers) means that transmitter efficiency levels can be higher. This results from linear
amplifiers being inherently inefficient.
Disadvantages of FM
Requires more complicated demodulator: One of
the disadvantages is that the demodulator is a more
complicated, and hence more expensive than the very
simple diode detectors used in AM.
Sidebands extend to infinity either side: The

sidebands for an FM transmission theoretically extend
out to infinity. To limit the bandwidth of the
transmission, filters are used, and these introduce
some distortion of the signal.
THANK YOU
Frequency Modulation

SINGLE TONE FREQUENCY MODULATION
• General expression for FM wave:
Equation shows non-linear modulation process.

Hence to avoid complexity, we use single tone frequency
modulation.
where Jn(β) are Bessel functions of the first kind.
Expanding the equation for a few terms we have:
Ideal Frequency Spectrum
of FM
Features of FM
• Maximum amplitude of the carrier wave is kept constant.
• Frequency of carrier wave is varied according to x(t).
• This deviation is called ‘Frequency Deviation’
• Amount of deviation depends upon amplitude of x(t)
The carrier swing=2x frequency deviation=2.∆f

In FM broadcasting, it has been internationally agreed to restrict
maximum deviation to 75 kHz on each side of the centre frequency for
sounds of maximum loudness.
Therefore FM channel width is 2x75=150 kHz.
Allowing 25 kHz guard band on either side, then the channel width
=2(75+25)=200 kHz.
In Television broadcasting a maximum frequency deviation of 25 kHz is
permitted for the sound portion.
Frequency range for angle modulation is 88MHz to 108MHz.
A single tone FM is represented by the following voltage
equation:
v(t)=12.cos(6x108t+5.sin(1250t))
Determine the followings:
(i) Carrier frequency,
(ii) modulating frequency,
(iii) The modulation index,
(iv) Maximum deviation,
(v) What power will this FM wave dissipate in 10 
resistors.
v(t) =A.cos(ωct+βsinωmt) v(t)=12.cos(6x108t+5.sin(1250t))
(i) Carrier frequency, ωc =6x108 rad/sec fc=95.5 MHz
(ii) Modulating frequency, ωm= 1250 rad/sec

fm=199 Hz
(iii) Modulation index, β=5
(iv) For maximum frequency deviation,
f=β.f m =5x199=995 Hz
(v) The power dissipation, 2

Vrms (12 / 2 ) 2 72
P    7.2 Watt
R R 10
Classification of FM
Depending upon the value of the modulation index β, FM

are of two types:
 Narrowband approximation (NBFM)
Wideband approximation (WBFM)

Comparison between Narrowband and
Wideband FM
Sr. Parameter NBFM WBFM
No
.
1. Modulation index Less than or slightly greater Greater than 1
than 1
2. Maximum 5 kHz 75 kHz
deviation
3. Range of 20 Hz to 3 kHz 20 Hz to 15 kHz
modulating
frequency
4. Maximum Slightly greater than 1 5 to 2500
modulation index
5. Bandwidth Small approximately same as Large about 15 times
that of AM BW = 2fm greater than that of NBFM.
BW = 2(+fmmax)
6. Applications FM mobile communication like Entertainment broadcasting
police wireless, ambulance, (can be used for high
short range ship to shore quality music transmission)
communication etc.
Narrowband FM (NBFM)
Narrowband FM (NBFM)
Generation of NBFM
Generation of NBFM
The NBFM obtained by using the above method is not perfect and
has some direction. It also differ from ideal FM in a no. of ways:
(1). The envelope of the narrow hand FM contains residual

amplitude modulation and therefore varies with time.
S (t )  Ac cos ct  Ac sin ct. sin mt

 e(t) cos(ct   )
Where
e(t )  Ac2   2 Ac2 sin 2  m t
and Ac sin mt

  tan (
1
)  tan 1 ( sin mt )
Ac
(2) For a sinusoidal modulating wave, the angle i(t)
contains harmony direction in the form of third and higher
order harmonies of modulating frequency ωm
Expanding the RHS of the previous eq in Power series, We get,
1 3 3 1 5 5
   sin( mt )   sin mt   sin mt  ......
3 5
The instantaneous phase angle is therefore,
1 3 3 1 5 5
 i (t )  c t   sin mt   sin mt   sin mt  .....
3 5
Thus a NBFM consists of residual AM and harmonic PM
which can be reduced to a negligible value by restricting β
to a small value.
Ac
S (t )  Ac cos c t  [cos(c  m )t  cos(c  m )t ]
2
It is interesting to note that this wave is similar to an AM
wave for the same modulating signal. For a single tone
sinusoidal modulation, the corresponding AM signal takes
the form.
mAc
S AM (t )  Ac cos c t  [cos(c  m )t  cos(c  m )t ]
2
we find that in case of sinusoidal modulating wave, the basic
difference between an AM wave and NBFM wave is that the sign of
the lower side frequency in the NBFM is reversed. Thus NBFM
essentially requires the same bandwidth (2fm) as the AM wave.
carrier Phasor has been assumed to be the reference. It

should be noted that the resultant of the Phasor
corresponding to the two side frequency is always
perpendicular to the carrier phasor
Transmission Bandwidth of Frequency modulated Wave
Theoretically, an FM wave contains an infinite no. of side frequency (for

sinusoidal modulation) and consequently the bandwidth required for
transmitting such signal is also of infinite extent.
The bandwidth of FM signal depends upon the value of modulation index B,
with the increase of modulation index B, more and more no. of sidebands
acquire significant amplitudes and thus bandwidth is increased.
In practice, we find that FM wave is effectively limited
to a no. of significant side frequency. The separation
between the two extreme significant side frequency
on the two sides of the carrier is also the effective
bandwidth of FM.
Let us consider that in the spectrum of FM wave the

no. of significant sidebands is n. Since the upper
sidebands are separated by m , therefore, they form a
frequency band of nm.
A similar frequency band of nm is formed by the
lower sidebands.
Thus, for n sidebands the bandwidth of FM wave is given
by:
B.W  2.nm radius/sec
B.W  2.nfm Hz
The no. of significant sideband n produced in FM wave
can be obtained from the plot of Jn(β).
The no. of significant sidebands produced in WBFM may
be consider to be an integer approximately equal to β, ie.,
n  β; β >>1
2 
B.W  2.nm  2  n  .m 
m [   ]
m
 2( ) radiana
 B.W  2(f ) Hz.
Thus, the approximate bandwidth of a wideband FM system
is given as twice the frequency deviation.
Universal Bandwidth curve
Schwartz developed a graph for determining the bandwidth
of an FM signal if the modulation index is known.
any frequency component with a signal strength (voltage) less than 1%
of that of the unmodulated carrier will be considered too small to be
significant. This curve is also called universal curve which shows the
variation of the BW ‘β’ normalized with respect to f against β.
Universal Bandwidth curve

Carson’s Rule
Carson’s rule provides a thumb formula to calculate
the bandwidth of a single tone wideband FM.
Accordingly to this rule the FM bandwidth is given
as twice the sum of the frequency deviation and the
highest modulating frequency.
The FM bandwidth is given by:
B.W  2(  m )  2  2m
But, 

m
 1
B.W  2  2.  2 (1  ) radian / sec
 
Or 1
B.W  2f (1  ) Hz.

two special cases:
(i) When  << m [narrow band FM]

i.e.,  << 1
Then, B.W = 2m; which is equal to AM
(ii) When,
 >> m [wide band FM]
i.e.,  >> 1
Then,  = 2;
For larger value of , the B.W relation in above

equation has very small error and can be assumed
to be true B.W for all practical purpose.
THANK YOU
FM Generation

FM Generation
The Direct Method or Parameter Variation Method

In direct method or parameter variation method, the baseband or
modulating signal directly modulates the carrier.
The carrier signal is generated with the help of an oscillator circuit.
This oscillator circuit uses a parallel tuned L-C circuit.
Thus the frequency of oscillation of the carrier generation is governed
by the expression:
Now, we can make the carrier frequency ωc to vary in accordance with the baseband
or modulating signal x(t) if L or C is varied according to x(t).
An oscillator circuit whose frequency is controlled by a modulating
voltage is called voltage controlled oscillator (VCO).
The frequency of VCO is varied according to the modulating signal
simply by putting a shunt voltage variable capacitor with its tuned
circuit.
This voltage variable capacitor is called varactor or varicap.
This type of property is exhibited by reverse biased semiconductor

diodes. Also the capacitance of bipolar junction transistors (BJT) and
field-effect transistors (FET) is varied by the Miller-effect. This miller
capacitance may be utilized for frequency modulation.
The inductance L of the tuned circuit may also be varied in accordance

with the baseband or modulating signal x(t).
The FM circuit using such inductors is called saturable reactor
modulator.
Reactance Modulator
In direct FM generation, the instantaneous frequency of the carrier is changed
directly in proportion with the message signal.
For this, a device called voltage controlled oscillator (VCO) is used.
A VCO can be implemented by using a sinusoidal oscillator with a tuned circuit
having a high value of Q.
The frequency of this oscillator is changed by changing the reactive components
involved in the tuned circuit. If L or C of a tuned circuit of an oscillator is changed in
accordance with the amplitude of modulating signal then FM can be obtained
across the tuned circuit as shown in figure below.
Fig.: Principle of Reactance Modulator

A two or three terminal device is placed across the tuned circuit. The reactance of
the device is varied proportional to modulating signal voltage. This will vary the
frequency of the oscillator to produce FM. The devices used are FET, transistor or
varactor diode.
Varactor Diode Modulator
A varactor diode is a semiconductor diode whose junction capacitance varies

linearly with the applied bias and the varactor diode must be reverse biased.
Working Operation
The varactor diode is reverse biased by the negative dc source –Vb.
The modulating AF voltage appears in series with the negative supply voltage. Hence, the
voltage applied across the varactor diode varies in proportion with the modulating voltage.
This will vary the junction capacitance of the varactor diode.
The varactor diode appears in parallel with the oscillator tuned circuit.
Hence the oscillator frequency will change with change in varactor diode capacitance and
FM wave is produced.
The RFC will connect the dc and modulating signal to the varactor diode but it offers a very
high impedance at high oscillator frequency. Therefore, the oscillator circuit is isolated
from the dc bias and modulating signal.
Working Operation
The capacitance Cd of the varactor diode is given by the following relation
k
cd   k .(VD ) 1/ 2
VD k is constant of proportionality.
Where VD is the total instantaneous voltage across the varactor diode and is given
by
VD  V0  x(t )
The oscillator frequency is
1
c 
LC
Now, the capacitance of the oscillator tank circuit will be Co+Cd and thus the
instantaneous frequency of oscillation i is given as:
1
i 
L0 (C0  Cd )
substituting the value of Cd ,
1
i 
L0 (C0  k .Vd ) 1/ 2
Thus the frequency i is dependent upon VD which in turn depends

on the modulating signal x(t). Therefore, the oscillator frequency i is
dependent on modulating signal and thus frequency modulation is
generated.
Limitations of Direct Method of FM Generation
1. In this method, it is very difficult to get

high order stability in carrier frequency because in
this method the basic oscillator is not a stable
oscillator, as it is controlled by the modulating
signal.
2. Generally in this method we get distorted

FM, due to non-linearity of the varactor diode.
Frequency stabilization system in FM transmitter
The instability of the carrier frequency can be alleviated using a feedback mechanism as
shown in Figure
Figure: Frequency stabilized FM Signal Generation
Frequency discriminator is a device whose output voltage has an

instantaneous amplitude that is proportional to the instantaneous
frequency of the FM signal applied to its input.
The DC voltage at the output of the low-pass filter is applied to VCO
of the FM transmitter in such a way as to modify the frequency of the
oscillator in a direction that tends to restore the carrier frequency to its
correct value.
INDIRECT METHOD for FM Signal Generation
(ARMSTRONG MODULATOR,
used for WBFM Signal Generation)
In the indirect method, the message signal is first used to produce a

narrow-band FM, which is followed by frequency multiplication to
increase the frequency deviation to the desired level.
In this method, the carrier-frequency stability problem is alleviated

by using a highly stable oscillator (e.g., crystal oscillator) in the
narrowband FM generation.
This modulation scheme is called the Armstrong wide-band

frequency modulator, in recognition of its inventor.
Indirect method of generating a wide-band FM wave
The frequency multiplier block
For the non-linear device we have the input-output relation
where
the output of the non-linear device can be written as

where employing the properties
we obtain
which can be written as

When v(t) passes through a band-pass filter with mid-band frequency
𝑛𝑓𝑐, we obtain the WBFM signal at the output of the BP filter, and the
WBFM signal generated is
Whose instantaneous frequency is given as

Practical Armstrong method for Fm Generator
Base band
Freqn Freqn RF
signal Narrow
Integrator Multiplier Mixer Multiplier Amplier
band PM
x(t) X1 X2
fc=100 MHz
Crystal Crystal f = 75 kHz
oscillator oscillator
f1=0.1 MHz f2=8.5 MHz
For commercial use it is required to transmit audio signals consisting

frequency in the range 50 Hz to 15 kHz and the value of f=75 kHz.
Let the final carrier frequency of the FM required is fc = 100 MHz.
We begin with NBFM with carrier frequency fc1 =100 kHz generated
by a crystal oscillator. In order to limit the harmonic distortion
Produced by the NBPM, we restrict the modulation index  to a
maximum of 0.3 radian.
Let us assume, 1 = 0.2 radian. The lowest modulation frequency
50Hz produces a deviation f1= 0.2 x 50 = 10 Hz at the NBPM o/p
while the largest modulation frequency 15 kHz produces a frequency
deviation of f2= 0.2 x 15 kHz = 3 kHz.
The lowest modulation frequency is therefore of immediate concern.
We select the value of f1= 10Hz. So that at the highest modulating
frequency  becomes even less.
In order to produce a frequency deviation of f = 75 kHz at the o/p, a
frequency multiplication is required.
For Example, f1= 10Hz. And the required deviation is f = 75 kHz.
Therefore, we required a total frequency multiplication factor
75000
  7500
10
A straightforward frequency multiplication equal to this value will
lead to a very high value of carrier frequency than the desired 100
MHz. In order to achieve the desired deviation and carrier frequency,
we take help of a two-stage frequency multiplier. This arrangement
uses two multiplies and a mixer. The mixer enables one to translate
the carrier frequency suitable without altering f. The final stage
multiplier gives the desired carrier frequency and deviation.
Let 1 and 2 are the multiplication factors for the two multipliers, so
that,
f 75000
  1 . 2    7500
f1 10
The carrier frequency at the first multiplier o/p is translated downwards

to frequency (f2-1f1) by mixing another oscillator. The carrier
frequency at the input of the second multiplier is fc/2,
Thus fc
f 2  1 f1 
2
Thus, with, f1 = 0.1 MHz, f2 = 8.5 MHz, we have
100
8.5  0.11 
2
Solving
1 = 100 and 2 = 75
THANK YOU
FM Demodulation

FM Detector
• An electronic circuit in which frequency variations of
modulated signals are converted to amplitude variations
first, with the help of tuned circuit,
• And then original information is extracted with the AM
demodulation techniques say diode detector.
The detector or demodulator circuit should be:
•Insensitive to amplitude changes.

•Not be too critical in its adjustment and operation.
•Converts frequency variations into amplitude.
BASIC FM DEMODULATOR
NOTE: Amplitude Variations are added to wave according

to frequency variations, and frequency variations remain
present in wave.
SLOPE DETECTOR
The most basic circuit employed as FM demodulator is parallel tuned
LC circuit, often known as slope detector.
• The carrier frequency should fall on one side of resonant frequency
• The entire frequencies should fall on linear region of transfer curve
of tuned circuit.
fr
fc
fr>>fc
Detuned Amplifier circuit Envelope detector
The output is then applied to a diode detector with RC load of

suitable time constant.
• The circuit is, in fact, identical to that of AM detector.
Characteristics of slope detector
LIMITATIONS OF SLOPE DETECTOR
• It is inefficient, as it is linear in very limited

frequency range.
• It reacts to all amplitude changes.
•It is relatively difficult to tune, as tuned circuit
must be tuned to different frequency than carrier
frequency.
BALANCED SLOPE DETECTOR
•This circuit uses two slope detectors, connected in back to

back fashion, to opposite ends of center tapped
transformer.
•And hence fed 1800 out of phase.
•The top secondary circuit is tuned above the IF by an
amount δf, and bottom circuit is tuned below IF by δf.
• Each circuit is connected to diode detectors with suitable
RC loads.
• The output is taken across series combination of loads, so
that it is algebraic sum of the individual outputs.
Balanced Slope Detector
The difficulties arising in simple slope detector circuit are
overcome in balanced slope detector.
Continued….
Final output voltage V0 is
Vo=V1-V2
Circuit Operation:
The circuit operation depends on range of frequencies.
(i) For fin = fc:
Voltage at T1 = Voltage at T2
Input voltage at D1=Input voltage at D2
 V01 = V02
 Vo = 0
Continued…
(ii) fc < fin < (fc + f):
Voltage induced in T1 > Voltage induced in T2.
 Input voltage at D1 > Input voltage at D2.
V1 > V2
Output voltage V0 is positive as frequency increase
towards (fc + f).
The positive output voltage increases as shown in Fig
(next slide).
(iii) (fc − f) < fin < fc:
Voltage induced in T2 > Voltage induced in T1.
 Input voltage to D2 > Input voltage to D1.
 V0 is negative. V2 > V1.
Characteristics of the balanced slope detector
Advantages and Limitations
Advantages:
(i) This circuit is more efficient than simple slope detector.
(ii) It has better linearity than the simple slope detector.
Limitations:
(i) Even though linearity is good, it is not good enough.
(ii) This circuit is difficult to tune since the three tuned circuits are
to be tuned at different frequencies, and
(iii) Amplitude limiting is not provided.
Foster-Seeley Discriminator
(Phase Discriminator)
 Primary and secondary windings both are tuned to the center frequency ‘fc’ of the
incoming signal.
Although the individual component voltages will be the same at diode inputs at all
frequencies, but the vector sum will differ with the phase difference between
primary and secondary windings.
As CC & C2 are coupling & RF Bypass capacitors respectively,
therefore VL3≈ VIN So
Voltage across diode= VIN + Secondary voltage/2
Output voltage= V01-V02
Continued…
Circuit Operation:
(i) When fin = fc:
Primary and secondary voltages are exactly 90
out of phase.
As shown in vector diagram,
Input at D1 = Input at D2
V01 = V02
Vo = 0
Continued…
(ii) When fin > fc:
Primary and secondary voltages are less than
90 out of phase.
Input at D1 >Input at D2
 V01 >V02
Vo is positive.
(iii) When fin < fc:
Primary and secondary voltages are more than
90 out of phase.
Input at D2 > Input at D1
V02 > V01
 Vo2 is Positive.
Continued… Frequency response of Phase Discriminator
Advantages:
(i) Tuning procedure is simpler than balanced slope detector, because it
contains only two tuned circuits and both are tuned to the same frequency .
(ii) Better linearity, because the operation of the circuit is dependent more on
the primary to secondary phase relationship which is very much linear.
Limitations:
It does not provide amplitude limiting. So in the presence of noise or any
other spurious amplitude variations, the demodulator output respond to
them and produce errors.
Continued….
Advantages:
•Easy alignment.
•Good linearity.
•Amplitude limiting is provided so that additional
limiter is not required.
Disadvantages:
•Complicated operation.
•More components are required.
THANK YOU
L#16: FM Demodulation_part 2

Ratio Detector
Similar to the Foster-Seeley discriminator .
(i) The direction of diode is reversed.
(ii) A large capacitance Cs is included in the
circuit.
(iii) The output is taken different locations.
Advantages:
 Easy to align.
 Good linearity due to linear phase relationship
between primary and secondary.
 Amplitude limiting is provided inherently. Hence
additional limiter is not required.
Change 1: Diode D2 is reversed so
that sum of Vao & Vbo appears across
points a’ and b’ instead of difference.
Change 2: A capacitor C5 with large

time constant is connected across a’-b’
in order to keep Vao+Vbo constant.
Change 3: Output is taken from O-O’ as

the difference of Vao + Vbo appears
there. Ground is shifted to O’.
Continued…
Circuit Operation:
With diode D2 reversed, O is now positive with respect to
b, so that Va is now sum voltage.
Large capacitor C5 is connected to keep this sum voltage
constant.
Output voltage V0 is equal to half of the difference
between the output voltages from the individual diodes.
 Vo= (Vo2-Vo1)/2
Thus, output voltage is proportional to the difference
between the individual output voltages.
Continued….
Amplitude Limiting Action:
•As FM input voltage tries to increase, the secondary
voltage also increases. So that extra diode current flows
through D1 and D2. Hence, load current increases.
•But voltage across C5 will not change instantaneously.
•Thus, load current has increased but load voltage is almost
constant.
•The ratio detector thus provides the amplitude limiting by
the process called ‘Diode Variable Damping’.
Function of L3:
•L3 is used to match the low impedance secondary to
primary.
•Also L3 gives a voltage step-down to prevent too-great
damping of primary by the ratio detector action.
OPERATION AT RESONANCE
• No phase shift occurs at resonance and both Vao & Vbo
are equal. Hence their difference (output) is zero.
• During negative part of cycle of input signal, polarity
across secondary also changes and both diodes get reverse
biased.
• But C5 with large time constant maintains voltage at
constant level.
OPERATION ABOVE RESONANCE
When a tuned circuit operates at a frequency higher than resonance,
the tank is inductive.
• Secondary voltage V1 is nearer in phase with primary voltage, while
V02 is shifted further out of phase with primary.
So output voltage in this case will be positive as shown in vector

diagram:
OPERATION BELOW RESONANCE
• When a tuned circuit operates below resonance, it is capacitive.
Secondary current leads the primary voltage and
• secondary voltage V02 is nearer in phase with primary voltage and
voltage V01 is shifted away in phase from primary voltage.
•So the output in this case will be negative.
Continued….
Advantages:
•Easy alignment.
•Good linearity.
•Amplitude limiting is provided so that additional
limiter is not required.
Disadvantages:
•Complicated operation.
•More components are required.
Why limiter stage is not used before ratio detector?
In ratio detector a large value capacitor is placed that functions
as amplitude limiter.
Limiter Function:
•If the input voltage fall, the diode current will fall, but the load
voltage will not, at first, because of the presence of the large
capacitor.
•The effect is that of an increased diode load impedance, the
diode current has fallen, but the load voltage remained constant.
•So that, damping is reduced and the gain of the driving amplifier
increases, this time counteracting an initial fall in the input
voltage.
•The ratio detector provides what is known as diode variable
damping.
•This maintains a constant output voltage desire changes in the
amplitude of the input.
•Thus, limiter stage is not used before ratio detector.
Performance Comparison of FM Demodulators
S.No. Parameter of Balanced Slope Foster-Seeley Ratio Detector
Comparison detector (Phase)
discriminator
(i) Alignment/tuning Critical as three Not Critical Not Critical
circuits are to be tuned
at different frequencies
(ii) Output characteristics Primary and secondary Primary and Primary and
depends on frequency relationship secondary phase secondary phase
relation. relation.
(iii) Linearity of output Poor Very good Good
characteristics
(iv ) Amplitude limiting Not providing Not Provided Provided by the
inherently inherently ratio detector.
(v) Amplifications Not used in practice FM radio, TV receiver
satellite station sound section ,
receiver etc. narrow band
FM receivers.
Phase-locked Loops
• It is the best frequency demodulator.
•A phase-locked loop (PLL) is an electronic circuit with a
voltage- or current-driven oscillator that is constantly
adjusted to match in phase (and thus lock on) with the
frequency of an input signal.
•PLL has low cost and superior performance even at low
SNR (signal-to-noise ratio)
PLL Characteristics
Basic operation
e(t)=i(t)-0(t)---phase error

e(t)=i(t)- K v  v2 (t )dt
−
When e(t)=0 , the PLL is said to be phase locked
When e(t)<1 rad , the PLL is said to be near phase locked
So, sin[e(t)] e(t)
The loop filter operates on error signal v1(t) to produce the output v2(t).
So, 
v2 (t ) =  v1 ( )h(t −  )d Where, h(t) is the Impulse response of the LPF
−
t 
 e (t ) =  i (t ) − K vK d   sin[ e ( )]h(t −  )dtd
0 −
Differentiating both sides,


d e (t ) d i (t )
= − K vK d  sin[ e ( )]h(t −  )d
dt dt −
A non-linear equivalent model of PLL

Rearranging,

d e (t ) d i (t )
+ K vK d   e ( )h(t −  )d =
dt −
dt
Taking Fourier transform,

1
e ( f ) =  (f)
H( f ) i
1 + K0
jf
H( f )
L( f ) = K d ----open loop transfer function of the PLL
jf
1
e ( f ) = i ( f )
1 + L( f )
Thus
Under the above mentioned condition, the phase of the VCO becomes equal
to the phase of the incoming signal and the phase lock is therefore
established
From the Fig.
Substituting the value of θe(t)
Or,
Then,
The corresponding time domain representation of the equation can
be obtaining by taking inverse Fourier Transform
Now,
Therefore,
Out put is same as modulating

signal except some scaling factor.
THANK YOU
L#17: Sampling theory and
Pulse modulation
Jayanta K Rakshit
Introduction
• Most of the signals that we use in our daily life are analog in nature
( for eg: speech, weather signals etc).
• Digital system possess many advantages in comparison to analog
system such as they are immune to noise, can be stored, processed
with more efficient algorithms, secure, more robust and cost
effective etc.
• Most of the effective signal processor are digital signal processors
which needs digital information in order to process it.
• Hence there arises a need to convert our analog signal to discrete
time signal in order to process them properly through digital signal
processors and then reconvert them back to analog signals so that
we can understand them.
• Sampling is the answer to this need.
• Sampling is a way to convert a signal from continuous time to
discrete time.
Sampling Theorem
This provides a mechanism for representing a continuous time signal
by a discrete time signal, taking sufficient number of samples of
signal so that original signal is represented in its samples
completely. It can be stated as:
(i) A band-limited signal of finite energy with no frequency
component higher than fm Hz, is completely described by its sample
values which are at uniform intervals less than or equal to 1/2fm
seconds apart. T = 1
s
2f
where Ts is sampling time
m
(ii) Sampling frequency must be equal to or higher than 2fm Hz.

[fs ≥ 2fm]
A continuous time signal may be completely represented in samples

and recovered back, if fs≥2fm, where fs is sampling frequency and fm
is maximum frequency component of message signal.
Proof of sampling theorem
➢Sampling of input signal x(t)
can be obtained by multiplying
x(t) with an impulse train δ(t) of
period Ts.
➢The output of multiplier is a
discrete signal called sampled
signal which is represented with
y(t) in the diagrams,
➢y(t)=x(t).δ(t)......(1)
The Fourier series representation of δ(t) :

T /2
1 1 1
a0 =
Ts   (t )dt =
−T / 2
Ts
 ( 0) =
Ts
T /2
2 2 T /2
Where,
  (t ) cos(ns )dt =
an = 2
Ts −T / 2
Ts
bn =
Ts   (t ) sin(n )dt =0
−T / 2
s
Contd…Proof of sampling theorem
Contd…Proof of sampling theorem
To reconstruct x(t), one has to recover input signal spectrum X(ω)
from sampled signal spectrum Y(ω), which is possible when there is
no overlapping between the cycles of Y(ω) which is possible if
fs≥2fm
For fs=2fm, is known as Nyquist rate.
1
Ts = is known as Nyquist interval
2 fm
Aliasing Effect
The overlapped region in case of under sampling represents Aliasing
effect. It can be termed as “the phenomenon of a high-frequency
component in the spectrum of a signal, taking on the identity of a
lower-frequency component in the spectrum of its sampled version.
This effect can be removed by considering

(i) fs >2fm or
(ii) by using anti aliasing filters which are
low pass filters and eliminate high
frequency components.
- All physically realizable signals are not completely band limited.

– If there is a significant amount of energy in frequencies above half the sampling frequency
(fs/2), aliasing will occur.
– Aliasing can be prevented by first passing the analog signal through an antialiasing filter
(also called a prefilter) before sampling is performed.
– The anti-aliasing filter is simply a LPF with cutoff frequency equal to half the sample rate.
Determine the Nyquist rate and the Nyquist interval
Solution: The angles are stated in radians, so the three frequencies are 25 Hz,
150 Hz and 60 Hz, respectively. The highest frequency is 150 Hz,
so the Nyquist rate is twice of 150 Hz = 300 Hz.
The Nyquist interval is the reciprocal of the Nyquist rate =1/300 sec= 3.333 ms
Solution: We begin by using the identity

2 sin(A) . cos(B) = sin(A + B) + sin(A – B).
Therefore, m(t) = 4. [sin(600πt +200πt) + sin(600πt -200πt)]
= 4.[sin(800πt) +sin(400πt)].
The highest frequency is 400 Hz; Nyquist rate is 2x400 Hz = 800 Hz.
The Nyquist interval is the reciprocal of the Nyquist rate = 1/800 sec =
1.250 ms.
sampling techniques
Three types of sampling techniques: (i) Impulse sampling,
(ii) Natural sampling, (iii) Flat Top sampling
Impulse sampling
Obtained by multiplying input signal x(t) with impulse train of
period Ts.
Also called ideal sampling. Practically not used because pulse width
cannot be zero and the generation of impulse train not possible.
Impulse sampling
Impulse sampling can be performed by multiplying input signal x(t)
with impulse train of period ‘T’. Here, the amplitude
of impulse changes with respect to amplitude of input signal x(t).
The output of sampler is given by
To get the spectrum of sampled signal, consider Fourier transform of

the above equation:
This is called ideal sampling or impulse sampling. You cannot use this practically
because pulse width cannot be zero and the generation of impulse train is not
possible practically.
Natural sampling
➢ This type of sampling similar to ideal sampling
except for the fact that instead of delta function,
now we use rectangular train of period Ts. i.e.
multiply input signal x(t) to pulse train.
➢ An electronic switch is used to periodically shift
between the two contacts at a rate of fs = (1/Ts) Hz,
staying on the input contact for C seconds and on
the grounded contact for the remainder of each
sampling.
➢ The output xs(t) of the sampler consists of segments
of x(t) and hence Xs(t) can be considered as the
product of x(t) and sampling function s(t).
➢ Xs(t)= x(t)×s(t)
The signal Xs(t) has the spectrum which consists of message spectrum and repetition
of message spectrum periodically in the frequency domain with a period of fs. But
the message term is scaled by ‘Co”(sinc function) which is not the case in
instantaneous sampling.
Flat Top sampling
✓ Flat Top sampling: During transmission, noise is introduced at top of the
transmission pulse which can be easily removed if the pulse is in the form of flat
top.
✓Here, the top of the samples are flat i.e. they have constant amplitude and is equal
to the instantaneous value of the baseband signal x(t) at the start of sampling.
Hence, it is called as flat top sampling or practical sampling.
➢ Flat top sampling makes use of sample and hold circuit.
➢ Theoretically, the sampled signal can be obtained by
convolution of rectangular pulse h(t) with ideally sampled
signal ,sδ(t).
g(t)= s(t) ⊗ h(t)
The duration of
each sample is τ
f(t) ⊗ δ(t) = f(t); property of delta function Applying a modified form; s(t) in place of
δ(t)
Sample and hold circuit to generate flat top sample
Sample and hold circuit is used for the generation of the sampled
signal to attain flat top sampling, which is shown in the Fig. below.
1. The switch G1closes at each

sampling instant to sample the
modulating signal.
2. The capacitor C holds the
sampled voltage for period at
the end of which switch G2 is
closed in order to discharge the
capacitor.
3. Thus the signal generated as a
result of sample and hold
process is the flat top sampled
signal.
On convolution of s(t) and h(t), we get a pulse whose duration is
equal to h(t) only but amplitude defined by s(t).
Train of impulses given by:
Signal s(t) obtained by multiplication of message signal x(t) and δTs(t)
Spectrum of flat top samples

Aperture Effect
Spectrum of flat topped sample is given by;
This equation shows that signal g(t) is obtained by passing the signal s(t) through a
filter having transfer function H(f).
Figure(a) shows one pulse of rectangular pulse train and each sample of x(t) i.e.
s(t) is convolved with this pulse.
Figure (b) shows the spectrum of this pulse. Thus, flat top sampling introduces an
amplitude distortion in reconstructed signal x(t) from g(t). There is a high
frequency roll off making H(f) act like a LPF, thus attenuating the upper portion
of message signal spectrum. The high frequencies of x(t) is affected. This is known
as aperture effect.
How to minimize aperture effect
An equalizer at the receiver end is needed to compensate aperture effect. The receiver
contains low pass reconstruction Filter with cut off slightly higher than fm Hz.
Equalizer in cascade with reconstruction filter has the effect of decreasing

the in band loss of reconstruction filter, frequency increases in such away
so as to compensate aperture effect.
where td is time delay introduced by LPF being equal to τ/2

Performance comparison of three sampling techniques
THANK YOU
L#18:Pulse modulation

Pulse Amplitude Modulation (PAM)
The amplitude of the pulses of the carrier pulse train is varied in accordance
with the modulating signal, that is amplitude of the pulses depends on the
value of m(t) during the time of pulse.
In fact the pulses in a PAM signal may of Flat-top type or natural type
or ideal type.
•The Flat-top PAM is most popular and is widely
used. The reason for using Flat-top PAM is that
during the transmission, the noise interferes with
the top of the transmitted pulses and this noise
can be easily removed if the PAM pulse as Flat-
top.
•In natural samples PAM signal, the pulse has
varying top in accordance with the signal
variation. Such type of pulse is received at the
receiver, it is always contaminated by noise.
Then it becomes quite difficult to determine the
shape of the top of the pulse and thus amplitude
detection of the pulse is not exact.
Pulse Amplitude Modulation (PAM)
PAM could be:
(i)Single polarity PAM: A suitable fixed DC bias is added to the signal
to ensure that all the pulses are positive.
(ii) Double polarity PAM: In this the pulses are both positive and
negative.
Generation of PAM
1. There are two operations involved in the generation of PAM signal
Instantaneous sampling of the message signal m(t) every Ts sec,
where the sampling rate fs = 1/Ts is chosen in accordance with the
sampling theorem.
2. Lengthening the duration of each sample so obtained to some
constant value T.
Modulating Low pass

Multiplier
signal Filter
m(t) PAM SIGNAL
Pulse Train
Generator
Figure: PAM signal

Sample and Hold Circuit for
Generating Flat-top sampled PAM
• The sample and hold circuit consists of two Field
Effect Transistor switches and a capacitor.
• The sampling switch is closed for a short duration
by a short pulse applied to the gate G1 of the
transistor. During this period, the capacitor C is
quickly charged up to a voltage equal to the
instantaneous sample value of the incoming signal.
• Now, the sampling switch is opened and the
capacitor holds the charge. The discharge switch is
then closed by a pulse applied to gate G2 of the
other transistor. Due to this, the capacitor is
discharged to zero volts. The discharges switch is
then opened and thus capacitor has no voltage. Figure: (a) Sample and hold
Hence the output of the sample and hold circuit circuit generating flat top
consists of a sequence of flat-top samples as shown sampled PAM (b) Waveforms of
in figure. flat top sampled PAM
Mathematical Representation of PAM
•We may express the PAM signal as
Where Ts = sampling period,

m(nTs) = sample value of m(t) obtained at t = nTs
h(t) = standard rectangular pulse of unit amplitude and duration T.
The spectrum of flat-top PAM signal is

Naturally Sampled PAM signal
• The natural sampling is basically pulse amplitude modulation.
Therefore it is called naturally sampled PAM signal.
•The time-domain representation of a naturally sampled PAM
signal will be given as
• The frequency spectrum of naturally sampled PAM signal will be

given as
Ideally Sampled PAM signal
• The instantaneous sampling is basically PAM. It is called ideally or
instantaneously sampled PAM signal.
• The time-domain representation of a ideally sampled PAM signal
will be given as
•The frequency domain representation i.e., frequency spectrum of a

ideally sampled PAM signal will be given as
Demodulation of PAM
PAM signal sampled at Nyquist rate can be reconstructed at the
receiver end , by passing it through an efficient Low Pass Filter (LPF)
with exact cut off frequency of fs/2. This is
known as Reconstruction or Interpolation Filter.
The low pass filter eliminates the high-frequency ripples and
generates the demodulated signal. This signal is then applied to the
inverting amplifier to amplify its signal level to have the demodulated
output with almost equal amplitude with the modulating signal .
For a flat topped PAM, a holding circuit followed by a LPF gives

demodulated signal.
Demodulation of PAM
Switch S closes after the arrival of pulse
and opens at the end of pulse.
Capacitor C charges to pulse amplitude
value and holds this value during interval
between two pulses.
The sampled values are shown in fig.
Holding circuit o/p smoothened in LPF.
Known as zero order holding circuit,
which considers only the previous sample
to decide value between two pulses
First order holding circuit considers
previous two samples, second order
holding circuit considers previous three
samples
a) Holding Circuit b) output of holding circuit c)output of Lowpass filter

Transmission of PAM signals
For PAM signals to be transmitted through space using antennas, they

must be amplitude/frequency/ phase modulated by a high frequency
carrier and only then they can be transmitted.
Thus the overall system is PAM-AM, PAM-FM or PAM-PM and at
receiving end, AM/FM/PM detection is first employed to get the PAM
signal and then message signal is recovered.
Advantages of PAM :
• It is the simple and simple process for modulation and demodulation

• Transmitter and receiver circuits are simple and easy to construct.
Drawbacks of PAM signal
• The bandwidth required for the transmission of a PAM signal is very
large in comparison to the maximum frequency present in the
modulating signal.
• Since the amplitude of the PAM pulses varies in accordance with the
modulating signal therefore the interference of noise is maximum in a
PAM signal. This noise cannot be removed easily.
• Since the amplitude of the PAM pulses varies, therefore, this also
varies the peak power required by the transmitter with modulating
signal.
Pulse Time Modulation (PTM)
• In pulse time modulation, amplitude of pulse is held constant,
whereas position of pulse or width of pulse is made proportional to
the amplitude of signal at the sampling instant.
• There are two types of pulse time modulation.
(i). Pulse Width Modulation (PWM)
(ii). Pulse Position Modulation (PPM)
In both the cases amplitude constant and does not carry information
so amplitude limiters can be used providing good noise immunity.
Pulse Width Modulation
•In PWM, Width of the pulses of the carrier pulse train is varied in
accordance with the modulating signal. The amplitude and positions
of the pulses are constant in this modulation.
• It is also called Pulse Duration Modulation (PDM), Pulse Length
Modulation (PLM).
Figure: Illustration of PWM (a) Modulating signal (b) Pulse Carrier (c) PWM signal
Pulse Width Modulation
Three types of pulse-width modulation (PWM) are possible:
(a) The leading edge of the pulse
being constant, the trailing edge
varies according to the message
signal.
(b) The trailing edge of the pulse

being constant, the leading edge
varies according to the message
signal
(c) The center of the pulse being

constant, the leading edge and the
trailing edge varies according to
the message signal (Symmetrical
PWM)
Generation of PWM
Direct method
The non inverting input of the comparator is
fed by the input message or modulating signal
x(t) and the other input by a saw-tooth signal
which operates at carrier frequency.
The comparator compares the two signals
together to generate the PWM signal at its output.
Its o/p is high only when the instantaneous value
of x(t) is higher than sawtooth waveform.
The rising edges of the PWM signal occurs at
the fixed time period (kTs) while trailing edge
depends on amplitude of message signal x(t).
When saw-tooth voltage waveform greater than x(t), o/p of comparator is zero, trailing
edge is modulated
If saw-tooth. waveform is reversed, trailing edge is fixed while leading edge is
modulated.
Replacing saw-tooth waveform by triangular, both leading and trailing edge modulated.
(symmetrical PWM)
The amplitude of PWM will be positive saturation of the comparator shown as ‘A’, being
same for all pulses.
Indirect Method
• Modulating signal (A) applied to i/p of PAM circuit [S(t) pulse train] and
PAM signal generated(B).
•S(t) also is i/p to Ramp generator(Integrator circuit), all having equal
slopes, amplitude and generation(D).
•These ramp pulses added to PAM pulses to produce varying height
samples. These varying height ramp gates a Schmitt Trigger circuit to
generate varying width rectangular pulses of PWM.
Generation of PWM using 555 timer
Working
It is basically a monostable
multivibrator with a modulating
signal applied at the control voltage
input.
Internally the control voltage is
Generation of PWM using 555 timer
adjusted to (2/3)Vcc. Externally
applied modulating signal changes
the control voltage and hence the
threshold voltage level.
As a result, the time period required
to charge the capacitor up to the
threshold voltage level changes,
giving pulse modulated signal at the
output. Modulated waveform
PWM Demodulation
• The received PWM
signal is applied to the
schmitt trigger circuit.
This schmitt trigger
circuit removes the noise
in the PWM waveform.
• The regenerated PWM
is then applied to the
ramp generator and the
synchronization pulse
detector. The ramp
generator produces ramps
for the duration of pulses
such that height of ramps
are proportional to the
width of PWM pulses.
PWM Demodulation
• The maximum ramp voltage is retained

till the next pulse.
• Synchronous pulse detector produces
reference pulses with constant amplitude
and pulse width. These pulses are
delayed for specific amount of delay as
shown in figure.
• The delayed reference pulses and the
output of ramp generator is added with
the help of adder. The output of adder is
given to the level shifter. Here, negative
offset shifts the waveform.
• The negative part of the waveform is
clipped by rectifier.
• Finally, the output of rectifier is passed
through low pass filter to recover the
modulating signal.
Advantages of PWM:
• Noise is less, since in PWM, amplitude is held constant.
• Signal and noise separation is very easy
• PWM communication does not required synchronization between
transmitter and receiver.
Disadvantages of PWM:
•In PWM, pulses are varying in width and therefore their power
contents are variable. This requires that the transmitter must be able to
handle the power content of the pulse having maximum pulse width.
•Large bandwidth is required for the PWM as compared to PAM.
Pulse Position Modulation (PPM)
In PPM, the position of the pulse relative to its un-modulated time
occurrence is varied in accordance with the message signal.
The amplitude and width of the pulses are constant in this modulation.
PPM Generation
• PPM generator consists of
differentiator and monostable
multivibrator.
•The differentiator generates
positive and negative spikes
corresponding to leading and
trailing edges of the PWM
waveform.
• Diode D1 is used to bypass the
positive spikes.
•The negative spikes are used to
trigger the multivibrator.
•The monostable multivibrator
then generates the pulses of same
width and amplitude with
reference to the trigger to give
PPM waveform.
Demodulation of PPM
First convert PPM signal to PWM signal.

F/F circuit is set or turn ‘ON’ [high o/p] when the reference pulse generator arrives.
This reference pulse is generated by reference pulse generator of the receiver with
the synchronization signal from the transmitter.
The F/F circuit is reset or turn ‘OFF’ [low o/p] at the leading edge of the position
modulated pulse. This repeats.
At the output of the F/F, PWM pulse will be generated.
The PWM pulses are then demodulated by PWM demodulator to get the modulating
signal.
Demodulation of PPM
PPM
Advantages of PPM:
• Like PWM, in PPM, amplitude is held constant thus less noise

interference.
• Signal and noise separation is very easy
• Because of constant pulse widths and amplitudes, transmission
power for each pulse is same.
Disadvantages of PPM:
• Synchronization between transmitter and receiver is required.
• Large bandwidth is required for the PPM as compared to PAM
Transmission BW of PWM and PPM
• Both PWM and PPM have DC value.

• Both need a sharp rise time and fall time to preserve the
message information
• Rise time be very less than Ts i.e. tr≪ Ts
• Transmission BW: BT ≥ 𝟏/𝟐𝒕𝒓
• BW higher than PAM.
Comparison of PAM,PWM & PPM
Q. For a PAM transmission of voice signal with fm=3kHz,
calculate the transmission BW. Given that fs=8 kHz and the
pulse duration τ=0.1Ts
Soln:
Ts= 1/f𝑠= 125µs
τ=0.1xTs=0.1×125=12.5µs
BW≥ 𝟏/𝟐𝝉≥ 40 kHz
Q. For the above signal if rise time is 1% of pulse width,
find minimum Transmission BW for PWM and PPM?
Soln:
Tr= τX0.01=1.25x10-7
BT≥1/2tr≥4 MHz
Thus BW of PWM/PPM much higher than PAM
THANK YOU
L#19:Pulse Code Modulation

Pulse Code Modulation
 Pulse Code Modulation

 Quantizing
 Encoding
 Analog to Digital Conversion
 Bandwidth of PCM Signals
PULSE CODE MODULATION (PCM)
 DEFINITION: Pulse code modulation (PCM) is essentially analog-
to-digital conversion of a special type where the information
contained in the instantaneous samples of an analog signal is
represented by digital words in a serial bit stream.
 The advantages of PCM are:

• Relatively inexpensive, digital circuitry may be used extensively.
• PCM signals derived from all types of analog sources may be
merged with data signals and transmitted over a common high-
speed digital communication system.
• In long-distance digital telephone systems requiring repeaters, a
clean PCM waveform can be regenerated at the output of each
repeater, where the input consists of a noisy PCM waveform.
• The noise performance of a digital system can be superior to that
of an analog system.
• The probability of error for the system output can be reduced
even further by the use of appropriate coding techniques.
Sampling, Quantizing, and Encoding
 The PCM signal is generated by carrying out three basic

operations:
1. Sampling
2. Quantizing
3. Encoding
1. Sampling operation generates a flat-top PAM signal.
2. Quantizing operation approximates the analog values by using a
finite number of levels. This operation is considered in 3 steps
a) Uniform Quantizer
b) Quantization Error
c) Quantized PAM signal output
3. PCM signal is obtained from the quantized PAM signal by
encoding each quantized sample value into a digital word.
Analog to Digital Conversion
 The Analog-to-digital Converter (ADC)
performs three functions:
Analog – Sampling
Input • Makes the signal discrete in time.
Signal
• If the analog input has a bandwidth
Sample
of W Hz, then the minimum sample
frequency such that the signal can
be reconstructed without distortion.
ADC – Quantization
Quantize • Makes the signal discrete in
111
110 amplitude.
101
100
011
• Round off to one of q discrete levels.
– Encode
010
001
Encode 000
• Maps the quantized values to digital

words that are  bits long.
 If the (Nyquist) Sampling Theorem is
Digital Output satisfied, then only quantization introduces
Signal distortion to the system.
111 111 001 010 011 111 011
Quantization
 The output of a sampler is still continuous in amplitude.
– Each sample can take on any value e.g. 3.752, 0.001, etc.
– The number of possible values is infinite.
 To transmit as a digital signal we must restrict the number of

possible values.
 Quantization is the process of “rounding off” a sample according

to some rule.
– E.g. suppose we must round to the nearest tenth, then:
3.752 --> 3.8 0.001 --> 0
Illustration of the Quantization Error
Quantization Example
Analogue signal
Sampling TIMING
Quantization levels.
Quantized to 5-levels
Quantization levels
Quantized 10-levels
PCM encoding example
Levels are encoded

using this table
Table: Quantization levels with belonging code words
M=8
Chart 2. Process of restoring a signal.

Chart 1. Quantization and digitalization of a signal. PCM encoded signal in binary form:
Signal is quantized in 11 time points & 8 quantization segments. 101 111 110 001 010 100 111 100 011 010 101
Total of 33 bits were used to encode a signal
Types of Quantization
The Mid-Rise type is so called because the origin lies in the middle of a raising part
of the stair-case like graph. The quantization levels in this type are even in number.
The Mid-tread type is so called because the origin lies in the middle of a tread of the
stair-case like graph. The quantization levels in this type are odd in number.
Both the mid-rise and mid-tread type of uniform quantizers are symmetric about the
origin.
Uniform Quantization
Dynamic Range: • Most ADC’s use uniform
(-8, 8)
quantizers.
Output sample
XQ
7
• The quantization levels of a
uniform quantizer are
5
1
equally spaced apart.
-8 -6 -4 -2 -1 2 4 6 8
• Uniform quantizers are
Input sample X
-3
optimal when the input
-5 distribution is uniform.
-7 When all values within the
Quantization Characteristic Dynamic Range of the
Example: Uniform  =3 bit quantizer quantizer are equally likely.
q=8 and XQ = {1,3,5,7}
MIDTREAD QUANTIZER
1. A midtread quantizer assumes
values of the form ∆Hi⋅
where ∆ is the step size
and Hi = 0, ±1, ±2, ±3, ...
2. It is called mid-tread because
the origin lies in the middle of a
tread of a staircase like graph.
MIDRISER QUANTIZER
1. A mid-riser quantizer has output levels
are given by (∆/2)Hi, where ∆ is the step
size and Hi= ±1, ±2, ±3, ....
2. The origin lies in the middle of the
rising part of the staircase-like
characteristic graph.
Nonuniform Quantization
 Many signals such as speech have a non-uniform distribution.
– The amplitude is more likely to be close to zero than to be at higher levels.
 Nonuniform quantizers have unequally spaced levels
– The spacing can be chosen to optimize the SNR for a particular type of signal.
Output sample XQ
6
2 Example: Nonuniform quantizer
-8 -6 -4 -2 2 4 6 8
-2
Input sample X
-4
-6
WHY IT IS NECESSARY TO HAVE NON-
UNIFORM QUANTIZATION
1. Using linear quantization, the quantization error is given by:
2. If q quantization levels of a bipolar signal are used, we can write:
3. Consider a PCM system with 𝜐 = 4 bits and 𝑥𝑚𝑎𝑥 = 1 Volt, then:

𝑞 = 24 = 16
The maximum quantization error is therefore
4. At maximum, the relative error is 1 volt out of 16 volts or 6.25%

5. At lower levels, e.g. 2 volts, the relative error is 1 volt out of 2 volts or 50%.
6. To reduce this high relative error at low levels, PCM systems use non-
uniform quantization.
Companding
1. With uniform sampling, the quantization step is fixed thus resulting
in uniform quantization noise power.
2. However signal power is not constant, it is proportional to the
square of the signal amplitude. This means Quantization Noise is
very significant at low amplitudes.
3. To reduce quantization noise at lower amplitudes, we use
commanding:
Companding = Compressing + Expanding
Companding
Nonuniform quantizer
Discrete Uniform
samples Compressor Quantizer
••••
Channel
••••
received Decoder Expander

digital
signals
COMPANDING IN COMMUNICATION SYSTEMS
1. The loudest sound that can be tolerated (120 dB SPL) is about

one-million times the amplitude of the weakest sound that can be
detected (0 dB SPL).
2. If the quantization levels are equally spaced (uniform
quantization), 12 bits must be used to obtain telephone quality
speech.
3. However, only 8 bits are required if the quantization levels are
made unequal (companding) to match the characteristics of human
hearing.
KEY POINTS ABOUT PCM
1. While PCM is a pulse modulation technique much like PWM,
PAM or PPM.
2. PCM is digital while the others are either analogue in time or
amplitude, i.e PCM pulses are discrete in time and amplitude
unlike PAM, PWM or PPM.
3. Essential aspects of a PCM transmitter are sampling, quantizing
and encoding.
4. PCM is not a modulation in the conventional sense because it
does not rely on varying a characteristic of a carrier (amplitude,
frequency or phase).
PCM TRANSMITTER
PCM REPEATER
PCM RECEIVER
PCM TRANSMISSION BANDWIDTH
1. Assume the a PCM encoder has q levels which are
encoded to 𝜐 bits.
2. We can infer 𝑞 = 2𝜐.
3. The number of bits per second can be expressed as:

𝑓𝑝𝑐𝑚 = 𝜐𝑓𝑠 where 𝑓𝑠 ≥ 2𝑓𝑚 (Nyquist criterion)
4. It therefore follows that the bandwidth, BW of a PCM

channel is bounded by:
𝐵𝑊𝑝𝑐 ≥ 2𝜐𝑓𝑚
EXAMPLE 1
A TV signal with a bandwidth of 4.2 MHz is transmitted
using binary PCM system using 512 quantization levels.
Determine:
(a) Code word-length
(b) The PCM bandwidth/bit rate
SOLUTION
(a) 𝑓𝑚 = 4.2 𝑀𝐻𝑧
𝑞 = 2𝜐 = 512
𝜐 = 𝑙𝑜𝑔2512 = 9 bits
(b) Bandwidth, BW = 2𝜐𝑓𝑚 = 2 x 9 × 4.2 = 75.6 Mb/s
ADVANTAGES AND DISADVANTAGES OF PCM
ADVANTAGES OF PCM
1. PCM provides high noise immunity.
2. Allows regeneration of clean signal by using repeaters placed
between the transmitter and the receiver.
3. PCM signals can be stored for later use or retransmission with high
fidelity
4. PCM signals can be encrypted more easily and to very high
standards.
DISADVANTAGES OF PCM
1. PCM requires complex circuitry to sample, quantize, code and
decode.
2. PCM requires large bandwidth compared with that of the original
analog signal.
THANK YOU
L#20: DELTA MODULATION

DELTA MODULATION
The type of modulation, where the sampling rate is much higher and
in which the step size after quantization is of a smaller value Δ, such
a modulation is termed as delta modulation.
1. Delta modulation seeks to overcome the problem of high

bandwidth requirement in conventional PCM.
2. Instead of generating and transmitting many bits per sample, only
one bit is transmitted.
3. During coding, the present sample is compared with the previous
and a 0 or 1 transmitted depending on whether the sample is
higher or lower than the previous.
Delta Modulator
The Delta Modulator comprises of a 1-bit quantizer and a delay circuit
along with two summer circuits.
From the above diagram
Ts is sampling interval
Using these notations
Further,
Where,
Hence,
Which means,
The present input of the delay unit
Assuming zero condition of Accumulation,
Now, note that
Delay unit output is an Accumulator output lagging by one sample.

Delta Demodulator
The delta demodulator comprises of a low pass filter, a summer, and a
delay circuit. The predictor circuit is eliminated here and hence no
assumed input is given to the demodulator.
A binary sequence will be
given as an input to the
demodulator. The stair-case
approximated output is given
to the LPF.
Low pass filter is used for
many reasons, but the
prominent reason is noise
elimination for out-of band
From the above diagram, we have the notations as – signals. The step-size error that
may occur at the transmitter is
called granular noise, which
is eliminated here. If there is
no noise present, then the
modulator output equals the
demodulator input.
Advantages of DM Over PCM
1. 1-bit quantizer,
2. Very easy design of the modulator and the demodulator
Disadvantages of DM
1. Slope Over load distortion (when Δ is small)
2. Granular noise (when Δ is large)
SLOPE OVERLOAD
Slope-overload occurs when the step size is too small to follow a
steep segment of the input waveform x(t ).
GRANULARITY
Granularity refers to a situation where the staircase function x(t)
hunts around a relatively flat segment of the input function, with a
step size that is too large relative to the local slope characteristic of
the input.
ADAPTIVE DELTA MODULATION
THE PRINCIPLE OF ADAPATIVE DELTA
MODULATION
Adaptive Delta
Modulation seeks to
overcome quantization
errors arising from
slope overload and
granular noise by
varying the step size
in accordance to the
signal amplitude.
TRANSMITTER AND RECEIVER
The logic for step size control is added in the diagram. The step
size increases or decreases according to a specified rule
depending on one bit quantizer output.
RECEIVER
The receiver has two portions. The first portion produces the step size
from each incoming bit. Exactly the same process is followed as that
in transmitter. The previous input and present input decide the step
size. It is then applied to the second portion i.e., an accumulator
which builds up staircase waveform. The low pass filter then
smoothens out the staircase waveform to reconstruct the original
signal.
Advantages
➢The signal to noise ratio of ADM is better than that of DM
because of the reduction in slope overload distortion and
idle noise.
➢Because of the variable step size , the dynamic range of
ADM is wider than DM.
➢Utilization of bandwidth is better in ADM than DM
Disadvantages
Implementation of step size control logic circuit is not
easy.
DIFFERENTIAL PULSE CODE
MODULATION (DPCM)
1. Some signals such as speech have high correlation between
adjacent samples.
2. When such highly correlated samples are encoded using basic
PCM, the resulting code contains a lot of redundant information.
3. In such cases, basic PCM scheme is not the preferred coding
method.
4. By removing this redundancy before encoding an efficient
coded signal can be obtained.
5. One method of removing redundancy is by using the Differential
PCM (DPCM) method.
6. DPCM is based on the principle that by knowing the past
behaviour of a signal up to a certain point in time, it is possible
to predict future values.
DPCM TRANSMITTER
The predictor produces the assumed samples from the previous

outputs of the transmitter circuit. The input to this predictor is the
quantized versions of the input signal. Predictor input is the sum of
quantizer output and predictor output.
DPCM RECEIVER
The predictor assumes a value, based on the previous outputs. The

input given to the decoder is processed and that output is summed
up with the output of the predictor, to obtain a better output.
Advantages of DPCM communication
As the difference between x(nT) and x*(nT) is being encoded and
transmitted by the PCM technique a small difference voltage is to be
quantized and encoded, this will need less number of quantization
levels and hence less number of bits to represent them
o Thus signaling rate and bandwidth of a DPCM will be less than
that of PCM
Disadvantages of DPCM communication
❑ High bit rate

❑ Practical usage is limited
❑ Need the predicator circuit to be used which is very complex
THANK YOU
L#21: Digital modulation
techniques

Categories of Modulation
DIGITAL MODULATION
• If the variation in the parameter of the carrier is discrete
then it is termed as digital modulation technique.
• In digital modulation , an analog carrier signal is
modulated by a discrete signal.
• Digital modulation can be considered as digital to-analog
and the corresponding demodulation as analog-to-digital
conversion.
• In digital communications, the modulating wave consists
of binary data and the carrier is sinusoidal wave.
DIGITAL MODULATION TECHNIQUES
AMPLITUDE SHIFT KEYING
(ASK)
 In ASK, the amplitude of the signal is changed in
response to information and all else is kept fixed.
 Bit 1 is transmitted by a signal of one particular
amplitude. To transmit 0,we change the amplitude
keeping the frequency constant.
AMPLITUDE SHIFT KEYING (ASK)
• As information signal is of binary format, only two

voltage levels occur at input (either +1V or 0V)
• As a result, we get only two voltage levels at output (either
[„Ac cos(ωct)‟] or „O‟V)
• because of these two output voltage levels, the carrier is
either in “ON” or “OFF” state.
• that‟s the reason why ASK is also known as “ON-OFF
Keying (OOK)”
Generation of ASK signal
ASK signal can be generated by simply applying the
incoming binary data (unipolar form) and the sinusoidal
carrier to the two inputs of a product modulator. The
resultant output will be ASK waveform. the bandpass filter
will remove the high frequency signal to
make the ASK signal waveform perfectly.
Synchronous ASK Demodulation
Working operation
Let 𝑋𝐴𝑆𝐾 (𝑡) be the ASK modulated signal, which is
When we input the ASK modulated signal to the two terminals of

balance modulator, then the output signal of balanced modulator can
be expressed as
Working operation
Where K represents the gain of balanced modulator. The first term of

equation is the data signal amplitude and the second term is the 2nd
harmonic of the modulated signal. From the output signal Xout(t), if
the first data signal amplitude receives the demodulated ASK signal,
this means that the data signal can be recovered correctly.
Bandwidth of BASK
Although there is only one carrier frequency, the process of
modulation produces a complex signal that is a combination of many
simple signals, each with a different frequency.
When we decompose an ASK-modulated signal ,we get a spectrum
of many simple frequencies.
However the most significant ones are those between fc- Nbaud/2
and fc+ Nbaud/2 with fc as the carrier frequency and Nbaud as the baud
rate.
The Bandwidth requirements of ASK are calculated using the
formula:
BW=(1+d) Nbaud
where d is a constant factor whose value lies between 0 and 1. The
value of d depends on the modulation and filtration process and the
medium. Bit rate is the number of bits per second.
Baud rate is the number of signal units per second.
Baud rate < Bit rate.
Bandwidth of BASK
** The bandwidth of the ASK signal is approximately 3/Ts Hz

Advantages of ASK modulation
Its generation and detection are easy thus facilitate simple transmitter
and receiver sections.
PROBLEMS WITH ASK

1. ASK transmission is highly susceptible to noise interference.
 The term noise refers to unintentional voltages introduced on to a line by
various phenomena such as heat or electromagnetic induction created by
other sources.
 These unintentional voltages combine with the signal to change the
amplitude.
 A 0 can be changed to 1 and a 1 can be changed to 0.
2. High amount of energy is required.
 A popular ASK technique is called on/off keying(OOK).
 In OOK one of the values is represented by no voltage.
 The advantage is a reduction in the amount of energy required to transmit
information.
3. Poor bandwidth efficiency.
BINARY PHASE SHIFT KEYING
In Phase shift keying, the phase of the carrier is varied to represent
binary 1 or 0.
Both peak amplitude and frequency remain constant as the phase
changes.
For example we start with a phase of 0 degree to represent binary
1 then we can change the phase to 180 degree to represent binary 0.
The phase of the signal during each bit duration is constant and the
value depends on the bit (0 or 1).
This method is often called 2-PSK or binary PSK , because two
different phases(0 degree and 180 degree are used).
Working operation
Carrier signal:
A=peak amplitude.
For symbol „1‟
For symbol „0‟
By Combining
Binary signal and its equivalent signal
Generation of BPSK signal
The block diagram of Binary Phase Shift Keying consists of the
balance modulator which has the carrier sine wave as one input and
the binary sequence (Bipolar NRZ) as the other input.
Fig.: Generation of
BPSK
BPSK modulated output
Coherent detection of BPSK signal
A coherent reference for synchronous detection cannot be obtained by the use of an
ordinary phase-locked tracking loop, since there are no spectral line components at ±
fc. However, since the signal has a spectrum that is symmetric with respect to the
(suppressed) carrier frequency, either a squaring loop or a Costas PLL can be used
to obtain synchronisation.
The diagram for a squaring loop in a coherent detector is shown below:
Bandwidth for BPSK signal
the spectrum of the BPSK signal is centered around the carrier
frequency fc.
If fb = 1/Tb , then for BPSK, the maximum frequency in the
baseband signal will be fb. The main lobe is entered around carrier
frequency fc and extended from fc – fb to fc + fb .
Therefore Bandwidth of BPSK signal will be,
BW = Highest frequency – Lowest frequency in the main lobe
BW = (fc + fb)– (fc - fb )
Or, BW = 2fb
Hence, the minimum bandwidth of BPSK signal is equal to twice
of the highest frequency contained in baseband signal.
Salient Features of BPSK
(i) BPSK has a bandwidth which is lower than that of a BFSK

signal
(ii) BPSK has the best performance of all the three digital
modulation techniques in presence of noise. It yields the minimum
value of probability of error/
(iii) Binary phase shift keying (BPSK) has a very noise immunity.
Drawbacks of BPSK
To generate the carrier in the receiver, we start by squaring
b(t ) 2 P cos( 2f t   )
c
If the receiver signal is  b(t ) 2 P cos( 2f t   )

c
then the squared signal remains same as before. Hence, the recovered
carrier is unchanged even if the input signal has changed its sign.
Therefore, it is not possible to determine whether the receiver signal
is equal to b(t) or – b(t). In fact, this results in ambiguity in the output
signal.
THANK YOU
L#22: Digital Modulation
Techniques_PART 2

Coherent Binary Frequency Shift
Keying (BFSK)
 In FSK, the frequency of the carrier signal is varied to represent
binary 1 or 0.
The frequency of the signal during each bit duration is constant and
its value depends on the bit (0 or 1).
Both peak amplitude and phase
remain constant.
The output of a FSK modulated
wave is high in frequency for a
binary High input and is low in
frequency for a binary Low input.
The binary 1s and 0s are called
Mark and Space frequencies.
Frequency Shift Keying (FSK)
Conversion table for BPSK representation
b(t) Input d(t) PH(t) PL(t)

1 +1V +1V 0V
0 -1V 0V +1V
Hence, if symbol ‘1’ is to be transmitted, the carrier frequency will be
If symbol ‘0’ is to be transmitted, then the carrier frequency will be

Generation of BFSK
Operation
We know that input sequence b(t) is same as PH(t). An inverter is
added after b(t) to get PL(t). The level shifter PH(t) and PL(t) are
unipolar signals. The level shifter converts the ‘+1’ level to PT
s b
Zero level is unaffected.

Thus, the output of the level shifter will be either PT (if ‘+1’) or zero
s b
(if input is zero). In other words, when a binary ‘0’ is to be

transmitted, PL(t) = 1 and PH(t) = 0, and for a binary ‘1’ to be
transmitted, PH(t) = 1 and PL(t) = 0, Hence, the transmitted signal will
have a frequency of either fH or fL. Further, there are product
modulators after level shifter. The two carrier signals 1(t) and 2(t)
are used.
1(t) and2(t) are orthogonal to each other. In one bit period of input
signal, 1(t) and 2(t) have internal number of cycles. Thus, the
modulated signal is having continuous phase.
The BFSK Signals
Thus, the modulated signal is having continuous phase.
Bandwidth of BFSK Signal
it is obvious that the width of one lobe is 2fb. The two main lobes
due to fH and fL are placed such that the total width due to both
main loves is 4fb.
Therefore, we have
Bandwidth of BFSK = 2fb + 2fb
Or, BW = 4fb.
Now, if we compare this bandwidth with that of BPSK, we note
that,
BW (BFSK) = 2 X BW(BPSK)
Coherent Detection of BFSK
Block Diagram of BFSK receiver (detection of BFSK)

Operation
The detector consists of two correlators that are individually tuned to two
different carrier frequencies to represent symbols ‘1’ and ‘0’. A correlator
consists of a multiplier followed by an integrator. Then, the received
binary FSK signal is applied to the multipliers of both the correlators. To
the other input of the multiplied output of each multiplier is fc1 and fc2 are
applied. The multiplied output of each multiplier is subsequently passed
through integrators generating output l1 and l2 in the two paths. The output
of the two integrators are then fed to the decision making device. The
decision making device is essentially a comparator which compares the
output l1 and output l2. If the output l1 produced in the upper path is greater
than the output l2 produced in the lower path, the detector makes a decision
in favour of symbol 1. If the output l1 is less than l2, then the decision
making device decides in favour of symbol 0. This type of digital
communication receivers are also called correlation receivers.
Salient Features of BFSK
(i) BFSK is relatively easy to implement
(ii) It has better noise immunity than ASK. Hence, the probability of
error free reception of data is high.
Drawback of BFSK
The major drawback is its high bandwidth requirement. Therefore,
FSK is extensively used in low speed modems having bit rates
below 1200 bits/sec.
Performance Comparison of Three Basic Digital
Modulation Techniques
S. Parameter of Binary ASK Binary FSK Binary PSK
N comparison
1 Variable characteristic Amplitude Frequency Phase
2 Bandwidth (Hz) 2fb 4fb 2fb
3 Noise Immunity Low high High
4 Probability of error High low Low
5 Performance in Poor Better than ASK Best of three
presence of noise
6 System complexity Simple Moderately Very complex
Complex
7 Bit rate or data rate Suitable upto Suitable upto Suitable for
100 bits/sec. 1200 bits/sec. high bit rates
8 Demodulation method Envelope Envelope Coherent
detection detection detection
Non-Coherent Detection
Non-Coherent Detection
 Requires no reference wave; does not exploit phase reference

information (envelope detection)
Amplitude Shift Keying (ASK)
Frequency Shift Keying (FSK)
Differential Phase Shift Keying (DPSK)
 Non coherent detection is less complex than coherent detection

(easier to implement), but has worse performance.
Non-Coherent Binary Amplitude Shift
Keying (ASK)
In the binary ASK case, the transmitted signal is defined as
s (t )  2 P cos(2f t )
s c
Binary ASK signal can also be demodulated non-coherent using

envelop detector. This greatly simplifies the design consideration
required in synchronous detection. Non-coherent detection schemes do
not require a phase-coherent local oscillator. This method involves
some form of rectification and low pass filtering at the receiver.
non-coherent ASK receiver
Non-coherent detection as used in analog communication does not

require carrier for reconstruction. The simplest form of incoherent
detector is the envelope detector. The output of envelope detector is
the baseband signal. Once the baseband signal is recovered, its
samples are taken at regular intervals and compared with threshold.
If Z(t) is greater than threshold () a decision will be made in favour
of symbol ‘1’
If Z(t) the sampled value is less than threshold () a decision will be
made in favour of symbol ‘0’.
Non- Coherenent FSK Demodulation
The detector consists of two band pass filters one tuned to each of the
two frequencies used to communicate ‘0’s and ‘1’s., The output of
filter is envelope detected and then baseband detected using an
integrate and sum operation. The detector is simply evaluating which
of two possible sinusoids is stronger at the receiver. If we take the
difference of the outputs of the two envelope detectors the result is
bipolar baseband. The resulting envelope detector outputs are sampled
at t = kTb and their values are compared with the threshold and a
decision will be made in favour of symbol 1 or 0.
THANK YOU
L23:
Non-coherent detection of PSK

Differential Phase Shift Keying (DPSK)
In order to eliminate the need for phase synchronisation of coherent
receiver with PSK, a differential encoding system can be used with
PSK. The digital information content of the binary data is encoded in
terms of signal transitions. As an example, the symbol 0 may be used
to represent transition in a given binary sequence (with respect to the
previous encoded bit) and symbol ‘1’ to indicate no transition. This
new signalling technique which combines differential encoding with
phase-shift keying (PSK) is known as differential phase-shift keying
(DPSK).
Generation of DPSK
A DPSK system may be viewed as the non coherent version of the
PSK. It eliminates the need for coherent reference signal at the
receiver by combining two basic operations at the transmitter
(1) Differential encoding of the input binary wave and
(2) Phase shift keying
Hence the name differential phase shift keying [DPSK]. To send
symbol ‘0’ we phase advance the current signal waveform by 1800
and to send symbol 1 we leave the phase of the current signal
waveform unchanged
Operation
The data stream b(t) is applied to the input of the encoder. The
output of the encoder is applied to one input of the product
modulator. To the other input of this product modulator, a sinusoidal
carrier of fixed amplitude and frequency is applied. The relationship
between the binary sequence and its differentially encoded version is
illustrated for a assumed data sequence 00100100111 (see next
page). In this illustration it has been assumed that the encoding has
been done in such a way that transition in the given binary sequence
with respect to the previous encoded bit is represented by a symbol 0
and no transition by symbol ‘1’. It may be noted that an extra bit has
been arbitrarily added as an initial bit. This is essential to determine
the encoded sequence. The phase of the generated DPSK signal has
been shown in the third row of Table (next page).
Differentially encoded sequence with phase
Binary data {b(k)} 0 0 1 0 0 1 0 0 1 1

Differentially encoded data {b(k)} 1* 0 1 1 0 1 1 0 1 1 1
Phase of DPSK 0  0 0  0 0  0 0 0
Shifted differentially encoded data {dk-1} 1 0 1 1 0 1 1 0 1 1
Phase of shifted DPSK 0  0 0  0 0  0 0
Phase comparison output - - + - - + - - + +
Detected binary sequence 0 0 1 0 0 1 0 0 1 1
*Arbitrary starting reference bit
Detection of DPSK
The received DPSK signal is applied to one input of the multiplier.

To the other input of the multiplier, a delayed version of the received
DPSK signal by the time interval Tb is applied. The delayed version
of the receiver DPSK signal has been shown in the 4th row of the
table. The output of the difference is proportional to cos (), here  is
the difference between the carrier phase angle of the received DPSK
signal and its delayed version, measured in the same bit interval.
Detection of DPSK
The phase angle of the DPSK signal and its delayed version have
been shown in 3rd and 5th rows respectively (see Table previous
page). The phase difference between the two sequence for each bit
interval is used to determined the sign of the phase comparator
output. When  = 0, the integrator output is positive whereas when
 = , the integrator output is negative. By comparing the integrator
output with a decision level of zero volt, the decision device can
reconstruct the binary sequence by assigning a symbol ‘0’ for
negative output and a symbol ‘1’ for positive output. The
reconstructed binary data is shown is the last row of the table. It is
thus seen that in the absence of noise, the receiver can reconstruct
the transmitted binary data exactly. DPSK may be viewed as a non-
coherent version of PSK. It may also be noted that the
reconstruction is invariant with the choice of the initial bit in the
encoded data.
Example 14.1: A binary data stream 0010010011 needs to
be transmitted using DPSK technique. Prove that the
reconstruction of the DPSK signal by the technique
discussed in the previous article is independent of the
choice of the extra bit.
Solution:
we have observed that DPSK signal can be detected accurately
without having a local oscillator for generation of synchronous
carrier. The initial bit in the differentially encoded data was assumed
to be ‘1’. In this example, we use the initial bit to be symbol ‘0’ and
verify that the reconstruction is invariant with the choice of the
initial bit. The results obtained for this case are given in Table 14.5.
it can be easily verified that the extra chosen bit 0 changes the phase
of the DPSK sequence but the detected sequence remains invariant.
Table: Differentially encoded sequence with phase
Binary data {b(k)} 0 0 1 0 0 1 0 0 1 1
Differentially encoded data 0* 1 0 0 1 0 0 1 0 0 0
{b(k)}
Phase of DPSK  0   0   0   
Shifted differentially encoded 0 1 0 0 1 0 0 1 0 0
data {dk-1}
Phase of shifted DPSK  0   0   0  
Phase comparison output - - + - - + - - + +
Detected binary sequence 0 0 1 0 0 1 0 0 1 1
*Starting reference bit
Advantages and Disadvantages of DPSK
Salient Features:
a) DPSK does not need carrier at the receiver end. This means that the
complicated circuitry for generation of local carrier is not required.
b) The bandwidth requirement of DPSK is reduced as compared to
that of BPSK.
Drawbacks:
a) The probability of error of DPSK is higher than that of BPSK.
b) Because DPSK uses two successive bits for its reception, error in
the first bit creates error in the second bit. Therefore, error
propagation in DPSK is more. On the other hand, in BPSK single bit
can go in error since detection of each bit is independent.
c) Noise interference in DPSK is more.
Quadrature Phase Shift keying (QPSK)
As a matter of fact, in communication systems, we have two main
resources. These are the transmission power and the channel
bandwidth. The channel bandwidth depends upon the bit rate or
signalling rate fb. In digital bandpass transmission, we use a carrier
for transmission. This carrier is transmitted over a channel. If two
or more bits are combined in some symbols, then the signalling rate
will be reduced. Thus, the frequency of the carrier needed also
reduced. This reduces the transmission channel bandwidth. Hence,
because of grouping of bits in symbols, the transmission channel
bandwidth can be reduced. In quadrature phase shift keying, two
successive bits in the data sequence are grouped together. This
reduces the bits rate or signalling rate and thus reduces the
bandwidth of the channel.
In case of BPSK, we know that when symbol changes the level, the
phase of the carrier is changed by 1800. Because, there were only
two symbols in BPSK, the phase shift occurs in two levels only.
However, in QPSK, two successive bits are combined. In fact, this
combination of two bits forms four distinct symbols. When the
symbol is changed to next symbol, then the phase of the carrier is
changed.
Symbol and corresponding phase shift in QPSK

S. Input successive bits Symbol Phase shift in
No. carrier
1 1(1V) 0(-1V) S1 /4
2 0(-1V) 0(-1V) S2 3/4
3 0(-1V) 1(1V) S3 5/4
4 1(1V) 1(1V) S4 7/4
Representation of QPSK signal
In QPSK system the information carried by the transmitted signal is
contained in the phase. The transmitted signals are given by
Generation of QPSK
Here, the input binary sequence is first converted to a bipolar NRZ

type of signal. This signal is denoted by b(t). It represents binary ‘1’
by + 1 V and binary ‘0’ by – 1 V. The demultiplexer divides b(t) into
two separate bit streams of the odd numbered and even numbered
bits. Here, ae(t) represents even numbered sequence and a0(t)
represents odd numbered sequence. The symbol duration of both of
these odd and even numbered sequences is 2Tb.
It may be observed that the first even bit occurs after the first odd bit.
Hence, even numbered bit sequence be(t) starts with the delay of one
bit period due to first odd bit. Thus, first symbol of ae(t) is delayed by
one bit period ‘Tb’ with respect to first symbol of a0(t). This delay of
Tb is known as offset. This shows that the change in levels of ae(t) and
a0(t) cannot occur at the same time due to offset or staggering.
Also, the bit steam ae(t) modulates carrier 𝑃𝑠 𝑠𝑖𝑛( 2𝜋𝑓𝑐 𝑡)
a0(t) modulates 𝑃𝑠 𝑐𝑜𝑠( 2𝜋𝑓𝑐 𝑡)
The two modulated signals can be written as,
se(t) and s0(t) are basically BPSK signals. The only difference is that
T= 2Tb here. The value of ae(t) and a0(t) would be + 1 V or – 1V.
The output of the adder is QPSK signal and it is given by,
S (t ) = S (t ) + S (t )
0 e
or S (t ) = a0 (t ) Ps cos(2f c t ) + ae (t ) Ps sin( 2f c t )
QPSK waveform
QPSK waveform
Phasor diagram of QPSK signal
S (t ) = a0 (t ) Ps cos(2f c t ) + ae (t ) Ps sin( 2f c t )

Detection of QPSK signal
The power splitter directs the input QPSK signal to the I and Q product
detectors and the carrier recovery circuit. The carrier recovery circuit
reproduces the original transmit carrier oscillator signal. The recovered
carrier must be frequency and phase coherent with the transmit
reference carrier.
The QPSK signal is demodulated in the I and Q product detectors,
which generate the original I and Q data bits. The outputs of the
product detectors are fed to the bit combining circuit, where they are
converted from parallel I and Q data channels to a single binary output
data stream. The incoming QPSK signal may be any one of the four
possible output phases
let the received QPSK signal be (-sinωct + cos ωct).

Again, the receive QPSK signal (-sin ωct + cos ωct) is one of the
inputs to the Q product detector. The other input is the recovered
carrier shifted 90° in phase (cos ωct). The output of the Q product
detector is
Advantages and Disadvantages of QPSK
Advantages of QPSK:
➢ QPSK provide very good noise immunity
➢ It provides low error probability
➢ Bandwidth is twice efficient as compared to BPSK modulation
For the same BER, the bandwidth required by QPSK is reduced to half as
compared to BPSK
➢ It is more efficient utilization of the available bandwidth of the
transmission channel
➢ Carrier power almost remains constant because of OQPSK amplitude is
not much
Disadvantages of QPSK:
✓QPSK is not more power efficient modulation technique compare to other
modulation types as more power is required to transmit two bits
✓ QPSK is more complex compared to BPSK receiver due to four states
needed to recover binary data information.
THANK YOU
L#24: Telemetry Principles

INTRODUCTION
Telemetry is presentation of measured values at location remote from site of measurement.
Greek words ‘Tele’: remote, ‘meter’: measuring. e.g., telemetry is the measurement
of remote (or far-off) physical variables or quantities. A physical variable or quantity under
measurement is called measurand.
In modern instrumentation system components of the system are located at a distance.
• it is necessary to transmit measured data.
• telemetry is the science of measuring data at a distance.
• used in commercial, industrial, military and space operations.
Telemetry can be done by different methods:
• Mechanical,
• Hydraulic,
• Electrical,
• Optical etc.
INTRODUCTION
The mechanical methods, either pneumatic or hydraulic have acceptable results for short
distances and are used in environments that have a high level of electromagnetic interference
and in those situations where, for security reasons, it is not possible to use electrical signals,
for example, in explosive environments. More recently, use of optical fiber systems allows
the measurement of broad bandwidth and high immunity to noise and interference.
Other proposed telemetry systems are based on ultrasound, capacitive or magnetic coupling,
and infrared radiation, although these methods are not routinely used.
The main advantage of electric over mechanical methods is that:
Electrically based telemetry does not have practical limits regarding the distance between
the measurement and the analysis areas, and can be easily adapted and upgraded in already
existing infrastructures.
Electric telemetry methods are further divided depending on the transmission channel that
they use as wire telemetry and wireless (or radio) telemetry.
• Wire telemetry is technologically the simplest solution. The limitations of wire
telemetry are the low bandwidth and low transmission speed that it can support.
However, it is used when the transmission wires can use the already existing
infrastructure, as, for example, in most electric power lines that are also used as wire
telemetry carriers
• Wireless telemetry is more complex than wire telemetry, as it requires a final radio
frequency (RF) stage. Despite its complexity, it is widely used because it can transmit
information over longer distances; thus, it is used in those applications in which the
measurement area is not normally accessible. It can also transmit at higher speeds
and have enough capacity to transmit several channels of information if necessary.
Telemetry using radio waves or wireless offers several distinct advantages over other
transmission methods. Some of these advantages are:
No transmission lines to be cut or broken.
 Faster response time

 Lower cost compared to leased lines
 Ease of use in remote areas where it’s not practical or possible to use wire or coaxial
cables
 Easy relocation
 Functional over a wide range of operating conditions
BLOCK DIAGRAM OF TELEMETRY SYSTEM
Telemetry involves three steps:
a. converting measured quantity to signal
b. Transmission of that signal over proper channel
c. Its reconversion to actual data for recording, displaying for graphical analysis and
further computation.
BLOCK DIAGRAM OF TELEMETRY SYSTEM
It consists of (not all the blocks will be always
present)
1. transducers to convert physical variables
to be measured into electric signals that
can be easily processed;
2. conditioning circuits to amplify the low-
level signal from the transducer, limit its
bandwidth, and adapt impedance levels;
3. signal-processing circuit that sometimes
can be integrated in the previous circuits;
4. subcarrier oscillator whose signal will be
modulated by the output of the different
transducers once processed and adapted;
5. Codifier circuit, which can be a digital encoder, an analog modulator, or a digital
modulator, that adapts the signal to the characteristics of the transmission channel, which is a
wire or an antenna;
6. radio transmitter, in wireless telemetry, modulated by the composite signal;
7. impedance line adapter, in case of wire transmission, to adapt the characteristic
impedance of the line to the output impedance of the circuits connected to the adapter; and
8. a transmitting antenna, for wireless communication,
The receiver end consists of similar modules. For wireless telemetry, these modules are:
1. a receiving antenna designed for maximum efficiency in the RF band used;
2. a radio receiver with a demodulation scheme compatible with the modulation scheme; and
3. Demodulation circuits for each of the transmitted channels.
For wireless telemetry, the antenna and the radio receiver are replaced by a generic front end to
amplify the signal and adapt the line impedance to the input impedance of the circuits that
follow.
TYPES OF TELEMETRY SYSTEMS
 Non-electrical telemetry system: Mechanical type, Pneumatic type.
Landline Telemetry System: Power Lines, Telephone Lines and Electrical

Wires. Distance ranges from 50m to 1 km e.g., labs, industries.
 Radio-Frequency System: Radio links from1 km to 50 km at 4MHz. For

distance >50 Km Microwave links are used 890 MHz to 30GHz. Repeaters are
installed after every 30 to 60 km for long distance transmission.
Non-electrical telemetry system
Mechanical Telemetry System
Levers are systems which shift the
load point to power point with gain or
mechanical advantage. This shifting
can be made long using a number of
levers in cascade.
Using the above transfer mechanism
technique an indication of a load can
be obtained at dial-pointer indication
system.
The A-lever system along with an extension lever and a pendulum type indicating scale
completes the systems. The platform taking the load may be flushed with the ground level and ,
except for the pendulum scale, the rest may be put away from the sight by putting in a housing,
for example. In recent times, the pendulum scale is also dispensed with and better indication
facilities are available with the older lever-type weighing machines.
Pneumatic Telemetry system
The pneumatic telemetry system is a position telemetry system and can be used for any process
variable, such as flow, pressure, level etc.
A typical scheme is shown in Figure for level
telemetering up to a distance of about 100 m.
There are four bellows elements: A and B forming
the transmitter block along with a stroke lever and
the interface disc d; and C and D form the
receiving and display block along with the link and
the pointer-scale arrangement. The two blocks are
connected by pneumatic lines. With the float rising
or falling, the rod via the float arm pivot lowers or
moves up pressing bellows element B or
expanding it and pressing element A, so that
pressure in line 1 or 2 expanding element D or C at
tthe receiving end which is displayed on the scale.
LANDLINE TELEMETRY SYSTEM
Voltage Telemetry System
Electrical systems are also used for short distance telemetering. The distance is around a few hundred metres (300
m) only. Such systems can be classified as (i) voltage telemetering, and (ii) current telemetering. These may be
two-wire type or three-wire type. A typical voltage telemetry scheme is shown in fig.(a).
The receiver side can use a LVDT as a primary sensor is shown in Fig (b). The LVDT secondary coils are directly
transmitting the differential voltage through a three-wire system over a certain distance.
Fig.(b): A telemetry scheme with LVDT as a transmitting element

Fig.(a):A typical voltage telemetry scheme
An important aspect of telemetering system is the signal to noise ration. Noise is of special consideration in
voltage telemetry system as in this current is very low and the signal power is very small. The transmission system
is to be specially designed to keep the interference to a minimum making the ratio S/N >> 2.
current telemetry system
The current telemetry system can develop higher signal power making it more immune to
interference arising mainly due to thermal and induced emf effects. The receiver is a cross-coil
current meter. It must be mentioned that the current must have a non-zero minimum value or
a live-zero for open circuit protection in the system.
It should be mentioned at this stage that such transmitters send the variable in the standard 4 to
20 mA range using a two-wire or three-wire scheme.
Frequency Telemetering
In a frequency telemetering system, the signal processing involves derivation of frequency in
proportion to an electrical signal after it has been obtained from the transducer, by use of an
appropriate unit such as a voltage-to-frequency converter, or a current-to-frequency converter.
For example, a 4 to 20 mA signal can be transformed into frequency ranges of
5 to 15 Hz (10 Hz) 9 to 15 Hz (6 Hz) 5 to 25 Hz (20 Hz) 6 to 27 Hz (21 Hz)
10 to 30 Hz (20 Hz) 7.5 to 15 Hz (7.5 Hz) 18 to 30 Hz (12 Hz)
The choice of these frequency ranges is governed by the commercial availability of low-cost telegraph and teletype
communication channels. A schematic block diagram of such a telemetry system is shown in Fig below
THANK YOU
L#25:Error Control Coding

Purpose
(i) Data can be corrupted during transmission.

(ii) For reliable communication,
(iii) errors must be detected and corrected.
Introduction
• Error control coding:
Extra bits(one or more) are added to the data at the
transmitter (redundancy) to permit error detection or
correction at the receiver.
• Classification of codes:
1) Error detecting codes: capable of only
detecting the errors.
2) Error correcting codes: capable of
detecting as well as correcting the errors.
TYPES OF ERRORS
 Single bit error :-

- Only one bit in the data unit has changed.
 Burst error :-
- It means that two or more bits in the data unit has
changed.
Types of error control
1. Automatic repeat request(ARQ) technique: receiver can
request for the retransmission of the complete or a part of message if
it finds some error in the received message. This requires an additional
channel called feedback channel to send the receiver’s request for
retransmission.
Appropriate for
• Low delay channels
• Channels with a return path
Not appropriate for delay sensitive data, e.g., real time speech and data
2. Forward error correction(FEC) technique: no such
feedback path and there is no request is made for
retransmission.
• Coding designed so that errors can be corrected at the

receiver
• Appropriate for delay sensitive and one-way
transmission (e.g., broadcast TV) of data
• Two main types, namely block codes and convolutional
codes
Classification of Error correcting codes
• Based upon memory:
Block code: does not need memory.
Convolutional code: needs memory.
• Based upon linearity:
Linear code
Nonlinear code
Error correcting code
Block codes Convolution codes

Drawbacks of coding techniques
• Higher transmission bandwidth.
• System complexity.
Error Detection Techniques
1. Parity Checking
2. Check Sum Error Detecting
3. Cyclic Redundancy Check (CRC)

Parity Checking
 Here an additional bit is appended with the existing message bits,
known as the parity bit.
 As a result of addition of this extra bit, the resultant word now will
have either even or odd parity i.e. number of 1s in the code word will
be either even or odd.
 If it is known that the parity of the received message is always
going to be even or odd as the case may be and if the received signal
does not tally with the expected result, the presence of an error is
detected.
 The limitation of this method is that it can only detect odd number
of errors and also it is unable to locate the position of the error.
Data word Parity bit (even) Data word Parity bit (odd)
1001011 0 1001011 1
0010011 1 1000110 0
Important definitions
• Code word: The code word is the n bit encoded block of
bits. It contains message bits and parity or redundant bits.
• Code rate/code efficiency: It is defined as the ratio of the
number of message bits(k) to the total number of bits(n)
in a code word.
Code rate (r) = k/n
• Hamming distance: number of locations in which their
respective elements differ.
e.g., 10011011
11010010 have a Hamming distance = 3
Alternatively, we can compute by adding code words (mod
2) =01001001 (now count up the ones)
• Hamming weight of a code word: It is defined
as the number of nonzero elements in the
code word.
• Minimum distance, dmin: The minimum

distance of a linear block code is defined as
the smallest Hamming distance between any
pair of code vector in the code.
Error Detection and Correction Capabilities
A code scheme has a Hamming distance dmin = 4. What is
the error detection and correction capability of this
scheme?
s  dmin 1
2t  dmin 1
Solution
This code guarantees the detection of up to three
errors (s = 3),
but it can correct up to one error.
Hamming codes
• Consider a family of (n,k) linear block codes
that have the following parameters:
Block length: n=2m -1
No. of message bits: k=2m -m-1
No. of parity bits: n-k=m
Where m≥3
These are so called Hamming codes.
• Hamming codes have the property that the
minimum distance d  3 independent of the
min
value assigned to the no. of parity bits m.
• Thus, Hamming codes are single error

correcting code.
Transmission Model
Error Modulator
Digital Source Line X(w)
Control (Transmit
Source Encoder Coding
Coding Filter, etc)
Hc(w) Channel
Transmitter
N(w) Noise
+
Error Demod
Digital Source Line
Control (Receive
Sink Decoder Decoding Y(w)
Decoding Filter, etc)
Receiver
THANK YOU
L#26: Error Control Coding
(Block coding)

Linear Block Codes
Definition: A code is said to be linear if any two code words in
the code can be added in modulo 2 addition to produce a
third code word in the code.
Code word length= n bits
m0,m1,m2……….mk-1 c0,c1,c2………cn-k-1
k message bits (n-k) parity bits
(n,k) linear block code

• A vector notation is used for the message bits and parity
bits
– message bit m = [m0 m1….mk-1 ]
– Parity bit c = [c0 c1……..cn-k-1 ]
m x
Linear block
encoder
--The code vector can be mathematically represented by

X=[M:C]
M= k message vector
C= (n-k) parity vector
• A block code encoder generates the parity vector or parity
bits required to be added to the message bits to generate the
code word. The code vector x can also be represented as
[X]=[M][G]
X=code vector of (1×n) size
M=message vector of (1×k) size
G=generator matrix of (k×n) size
• The generator matrix depends on the type of linear block
code used and is defined as
G = [ Ik | P]
Where Ik = (k×k) identity matrix
P= k×(n-k) coefficient matrix
1 0 . . 0
0 1 . . 0
 
I .
k . . . .
 
. . . . .

0 0 0 0 1
 k k
 p00 p10 ... pnk 1,0 

 p p11 ... pnk 1,1 
P  01 
 . . ... . 
 
p
 0,k 1 p1,k 1 ... pnk 1,k 1  k ( n  k )
• The parity vector can be obtained as
C=MP
 p00 p10 ... pn  k ,0 
c 0 c .....c
1 n  k 1
  m0 m1...... mk 1  p01 p11 ... pn  k ,1 

 . . ... . 
 
 p0,n  k p1,k 1 ... pn  k ,k 1 
By solving the above matrix equation the parity vectors are:
c0  m0 P00  m1P01  .......  mk P0, k 1

c1  m0 P10  m1P11  .......  mk P1, k 1
c2  m0 P20  m1P21  .......  mk P2, k 1
Parity check matrix(H)
• There is another way of expressing the relationship between
the message bits and the parity bits of a linear block codes.
Let H denote an (n-k)×n matrix defined as
H = [PT | In-k]
Where PT= (n-k)×k matrix representing the transpose of the
coefficient matrix P
In-k = (n-k)×(n-k) identity matrix
The matrix H is called as parity check matrix
* If generator matrix G is known, then parity check matrix can be calculated and vice-versa.
Properties of G and H matrix
 Parity-check matrix H
The parity-check matrix of a canonical generator
matrix is an (n-k)-by-n matrix satisfying
where the columns of H are linearly independent.
 Then, the code words (or error-free receptions) should

satisfy (n-k) parity-check equations.
Problem:1
Consider a (6,3) linear block code defined by the generator matrix
(a) Find the parity check matrix H of the code in systematic form.
(b) Find the encoding table for the linear block code.
(c) What is the minimum distance dmin of the code. How many
errors can the code detect. How many errors can the code correct.
(d) Draw the hardware encoder diagram.
Solution
(a)
(b)
(c) From encoding table, we have
Hence the (6,3) linear block code can detect 2 bit errors and
correct 1 bit error in 6 bit output codeword
(d) The output for general code word is
The hardware encoder implementation is

Problem:2
The parity check bits of a (7,4) Hamming code, are

generated by
c5 = d1 + d3 + d4
c6 = d1 + d2 + d3
c7 = d2 + d3 + d4
i) Find the generator matrix [G] and parity check matrix [H] for this
code
ii) Prove that GHT = 0
iii) Determine the eight code vectors of the dual code for the (7,4)
Hamming code describe above.
iv) Find the minimum Hamming distance of part (iii).
Solution:
(i)
[c5, c6, c7]1x3=[d1, d2, d3, d4]1x4 [P]4x3
 P11 P12 P13 
P P22 P23 
[c5, c6, c7]1x3=[d1, d2, d3, d4]1x4  P21 P32 P33 
31
 
 P41 P42 P43  4 X 3
c5  P11d1  P21d 2  P31d3  P41d 4
c6  P12d1  P22d 2  P32d3  P42d 4 Given

c5 = d1 + d3 + d4
c7  P13d1  P23d 2  P33d3  P43d 4 c6 = d1 + d2 + d3
c7 = d2 + d3 + d4
Comparing,
P11=1 P21=0 P31=1 P41=1
P12=1 P21=1 P31=1 P41=0
P13=0 P23=1 P33=1 P43=1
1 1 0
So, 0 1 1 
P
1 1 1
 
1 0 1 4X 3
n=7 and k=4
Here G is a 4 × 7 matrix in which 4 × 4 identity matrix
The parity check matrix H is

(ii)
(iii)
Given
c5 = d1 + d3 + d4
c6 = d1 + d2 + d3
c7 = d2 + d3 + d4
Let, m=[0 1 0 1] So, c5 =1, c6 =0, c7 =0,
d1 d2 d3 d4 d5 d6 d7
So, code word for m=[0 1 0 1] is
0 1 0 1 1 0 0
Try yourself for other code words

THANK YOU
(Syndrome decoding)

Syndrome: Definition & properties
• The generator matrix G is used in the encoding operation at
the transmitter. On the other hand, the parity check matrix H
is used in the decoding operation at the receiver.
Let x represent the transmitted code word and y represent
the received code word. We express the vector y as the sum
of the original code vector X and a vector E, given by
Y  X E
Where E is called the error vector or error pattern. The ith
element of the E equals 0 if corresponding element of y is the
same as X. On the other hand the ith element of E equals 1 if
there is an error at the ith location.
• The syndrome vector is defined as
S=YHT
• Property: The syndrome depends only on the error pattern

and not on the transmitted code word.
S=(X+E)HT
=XHT+EHT
= EHT
Syndrome decoding
• We have discussed about the encoder for the linear block
code. Now let us learn about the decoder. The two important
functions of the decoder are
– Error detection in the received code
– Error correction
• The above two functions are accomplished by syndrome
decoding.
Detection of Error
• Since we know XHT =0
At the receiver, if S=YHT =0 then Y=X and there is no
error
but if S=YHT ≠0 then Y ≠X and error exist in the
received codeword.
Correction of Error
 Steps:
1. For the given received vector find the syndrome vector as
S=YHT.
2. The syndrome vector will resemble any of the column of H
matrix, which indicates there is an error in the
corresponding bit of the received vector.
3. Now calculate error vector E.
suppose 2nd column of H matrix and syndrome
vector is same that means there is an error at the 2nd bit of
received signal. Then the error vector will be E=[0100000] if
n=7.
4. Finally determine the transmitted vector as X  Y  E
PROBLEM
For a systematic (7,4) linear block code, the parity matrix P is
given by
i) Find all possible code vectors.

ii) Draw the corresponding encoding circuit
iii) Detect and correct the following received data R=[1 0 1 1 1 0 0]
iv) Draw the syndrome calculation circuit.
Solution:
n=7 and k=4
There are 24 = 16 message vectors given by u=[0000, 0001, 0010,
0011, 0100, 0101, 0110, 0111, 1000, 1001, 1010, 1011, 1100, 1101,
1110, 1111 ]
The generator matrix G is in the form of
v6 = u3, v5 = u2, v4 = u1, v3 = u0,

v2 = u0 + u2 + u3, v1 = u0 + u1 + u3, v0 = u0 + u1 + u2
+ Means EXOR operation
The code vectors V
Encoder circuit
Syndrome circuit:
Syndrome vector is obtained from the received sequence

r = [r0 r1 ….. r7] using the parity check matrix of the code:
syndrome digits are:

s0 = r0 + r3 + r4 + r5, s1 = r1 + r3 + r4 + r6 ,
s2 = r2 + r3 + r5 + r6
From decoding table, this syndrome corresponds to error pattern
e = [1 0 0 0 0 0 0]. Hence the corrected code word is
y = r +e = [ 1 0 1 1 1 0 0 ] + [1 0 0 0 0 0 0 ] = [0 0 1 1 1 0 0 ]
Syndrome calculation circuit
syndrome digits are:
s0 = r0 + r3 + r4 + r5, s1 = r1 + r3 + r4 + r6 , s2 = r2 + r3 + r5 + r6
Consider a (6,3) linear block code defined by the generator matrix
(a) Calculate the syndrome vector single bit error.

(b) Suppose c = [1 1 1 0 0 0] is sent and r = [1 1 1 0 0 1] is received.
Show how the code can correct this error.
Solution
(a)
The syndrome for general received word is
Let the message vector is [ 0 0 0 0 0 0 ]

(b)
Given that c = [1 1 1 0 0 0] is sent and r = [1 1 1 0 0 1] is received.
From decoding table, this syndrome corresponds to error pattern

e = [0 0 0 0 0 1]. Hence the corrected code word is
THANK YOU
(Cyclic codes)

Cyclic codes
• Cyclic codes are also linear block codes.
• A binary code is said to be cyclic if it exhibits two fundamental
properties:
– Linearity property: Sum of any two code words in the code is also a
code word.
– Cyclic property: Any cyclic shift of a code word in the code is also a
code word.
(n-k) parity bit K message bit

(n,k) cyclic code
Code word polynomial
• The code word [x0 x1 x2 ….. Xn-1] cab be expressed in the form
of a code word polynomial as
n 1
X ( p)  x 0  x1p  x 2 p  ....... x n1 p
2
• Some important conclusion from the code word polynomial:

– For binary codes, the coefficients of p,p2 ………are 1 or 0.
– Each power of p in the polynomial X(p) represents a one bit
cyclic shift in time. Thus multiplication of the polynomial
X(p) by p is equivalent to a cyclic shift or rotation to right
by one digit.
How do we make such a shift cyclic?
• For this a special type of polynomial multiplication known as
n
modulo ( p  1) is introduced. Thus a single shift can be
obtained by multiplying X(p) by p as
n 1
 x 0 p  x1 p  ....... x n2 p
n 2
pX(p) mod ( p  1) = x n 1
The above polynomial represents the code word

[Xn-1 x0 x1 x2 ….. Xn-2]
Generator polynomial for cyclic code
• It is used for the generation of cyclic code words and is
represented as
X(p)=M(p)G(p)
Where M(p)=message polynomial
G(p)=generator polynomial of degree (n-k)
2 n  k 1 nk
G ( p)  1  g p  g p  ........  g p  p
1 2 n  k 1
n  k 1 n k
g p
i
G ( p)  1   p
i
i 1
NOTE: The degree of generator polynomial is equal to the no. of

parity bits in the code word.
Generation of non-Systematic
code-words
There are two types of Cyclic Codes on basis
of Encoding:
Non-Systematic Encoding:
The output code word is generated using polynomial
multiplication.
Information bits are not packed together in the
codeword. These are rarely used.
X1(p)= M1(p).G(p)
X2(p)= M2(p).G(p)
X3(p)= M3(p).G(p)
Generation of Systematic code-words
• There are three steps involved in the encoding process for an
(n,k) cyclic code. They are
nk
– Multiply the message polynomial M(p) by p
nk
– Divide p M ( p) by the generator polynomial G(p) to obtain the
remainder C(p) nk
p M ( p)
 Q( p) 
C ( p)
G ( p) G ( p)
Where Q(p)=Quotient
nk
--Add the remainder polynomial C(p) and p M ( p) to obtain the code
word polynomial X(p).
nk
i,e, X ( p)  [ p M ( p)]  C ( p)
Consider the generator polynomial for a (7,3) cyclic code
defined by:
g(p) = p4 + p3 + p2 + 1
(a) Find all non-systematic code vector.
Solution
Given that the generator polynomial for a (7,3) cyclic code is
g(p) = p4 + p3 + p2 +1
The output code words are given by
c(p) = M(p)g(p) 2
M(p)=m2p +m1p+m0
For example one multiplication :
Consider the generator polynomial for a (7,4) cyclic code defined by
g(p) = p3 + p2 + 1
(a) Find the systematic output codeword for input c = [ 1 1 1 1 ].
Solution
Given that c = [ 1 1 1 1 ].
Given that the generator polynomial for a (7,4) cyclic code is
g(p) = p3 + p2 +1
The systematic output code word is
Generator and parity check matrix of the cyclic code
The cyclic codes are linear block codes. Therefore generator and parity check
matrix can be defined.
The generator matrix G(p) has a size of k x n, i.e. k no. of rows and n no. of columns
G(p)=pn-k-1+ pn-kgn-k-1+…….+g2p2+g1p+1
We multiply both sides by pi,
So, piG(p)=pn-k-1+i + pn-k+ign-k-1+…….+g2p2+i +g1p1+i+pi.

Where, i=(k-1), (k-2), …..,2,1,0
The above equation represents the polynomial for the rows of the
generating polynomial. It is possible to obtain the generator matrix
from this equation
Problem
For a (7,4) cycle code, determine the generator matrix if
G(p) =1+p+p3.
Solution:
Here, n=7 and k=4, hence, n-k = 3
G(p) =1+p+p3
(i) We multiply both the side of G(p) by pi where i=(k-1)……,1,0.
 piG(p)= pi+ p1+i+ p3+I, i=(k-1)……,1,0.
But, i=4,  i=3,2,1,0.
(ii) By substituting these values of I into equation (i), we get four
different polynomials as under. These polynomials correspond to the
four row of the generator matrix as under.
Row No. 1 : i = 3  p3G(p) = p3 + p4 + p6
Row No. 2 : i = 2  p2G(p) = p2 + p3 + p5
Row No. 3 : i = 1  p G(p) = p + p2 + p4
Row No. 4 : i = 0  G(p) = 1 + p + p3
The generation matrix for (n,k) code is of size k x n. Therefore,
for the (7, 4) cycle code, the generator matrix will be a 4 x 7
matrix.
The polynomials corresponding to the for rows are therefore, as

under:
Row No. 1 : i = 3  p6 + 0p5 + p4 + p3 + 0p2 + 0p + 0
Row No. 2 : i = 2  0 p 6 + p5 + 0p4 + p3 + p2 + 0p + 0
Row No. 3 : i = 1  0p6 + 0p5 + p4 + 0p3 + p2 + p + 0
Row No. 4 : i = 0  0p6 + 0p5 + 0p4 + p3 + 0p2 + p + 1
These polynomials can be converted into generator matrix
G as under:
p p p p p p p
6 5 4 3 2 1 0
1 0 1 1 0 0 0
 
G0 1 0 1 1 0 0
0 0 1 0 1 1 0 

 0 0 0 1 0 1 1  4x7
The cycle codes are basically block codes. Therefore, its
code vectors can be obtained by using the generator matrix
as under:
X= MG
Where, M = 1 X k message vector.

Problem
For the generator matrix of the previous example,
determine all the possible code vectors.
Solution:
All the code vectors can be obtained by using the following
expression:
X = MG
Let M = (m3 m2 m1 m0) = ( 1 0 1 0 )
Therefore,
1 0 1 1 0 0 0
0 1 0 1 1 0 0
X  [1010 ] 
0 0 1 0 1 1 0
 
0 0 0 1 0 1 1
Therefore, X = [ 1  0  0  0 00 00
10 10 10 00 00 10
00 10 00 00]
Therefore, We have X = [1 0 0 1 : 1 1 0]
Similarly, the other code vectors can be obtained.
Systematic Form of Generator Matrix
the generator matrix in the systematic form is given by.
G  [I : Pk kx ( n  k )
]
k x (n - k)
This means that there are k number of rows in the generator matrix.
Let us represent the row number (in general) by i. Then ith row of the
generator matrix is represented by,
ith row of G  p (n-i)  Ri ( p)...Where i  1,2,....k

Now, we divided p(n-i) by he generator matrix G(p). The result of
the division is expressed as,
p ( n i ) Re mainder
 Quotient 
G ( p) G ( p)
Let Remainder = Ri(p)
Quotient = Qi(p)
Substituting this into previous equation, We obtain
( n i )
p Ri ( p )
 Qi ( p ) 
G ( p) G ( p)
So,
p ( ni )  Qi ( p)G( p)  Ri ( p); where i  1,2,....k
In mod-2 additions, the addition and subtraction will yield the
same result.
So, p ( ni )
 Ri ( p)  Qi ( p)G( p)
the above expression represents the ith row of the systematic
generator matrix.
Problem
For systematic (7,4) cycle code, determine the
generator matrix and parity check matrix. Given :
G(p) =p3 + p + 1. Also obtain the code vector for the
message bit: [1 0 1 0]
Solution:
(i) The ith row of the generator matrix is given by equation (16.44)
as under:
p ( ni )
 Ri ( p)  Qi ( p)G( p)
where i=1,2,….k,
(ii) It is given that the cycle code is systematic (7,4) code.
Therefore, n=7, k=4 and (n-k) = 3
Substituting these values into the above expression, we obtain
p ( 7i )  Ri ( p)  Qi ( p).(p3 + p + 1).....i  1,2, ...4.
(iii) With i=1, the above equation is given by,

p 6  Ri ( p)  Qi ( p).(p3 + p + 1)
The quotient Qi(p) can be obtained by dividing p(n-i) by Q(p).

Therefore, to obtain Qi(p), divide p6 by (p3 + p2 + 1).
Here the quotient polynomial Qi(p) = p3 + p + 1

And the remainder polynomial Ri(p) = p2 + 0p + 1
So, we obtain
p 6  Ri ( p)  (p3 + p + 1)(p3 + p + 1)
 p6 + p4 + p3 + p4 + p2 + p + p3 + p + 1
 p6 + 0p5 + (1  1)p4 + (1  1)p3 + p2 + (1  1)p + 1
 p6 + 0p5 + 0p4 + 0p3 + p2 + 0p + 1

1st Row polynomial
 p6 + 0p5 + 0p4 + 0p3 + p2 + 0p + 1
1st Row elements 1000101

Using the same procedure, we can obtain the
polynomial for the other rows of the generator matrix
as under:
2nd Row Polynomial  p5 + p2 + p + 1
3rd Row Polynomial  p4 + p2 + p
4th Row Polynomial  p3 + p + 1
These polynomial can be transformed into the generator
matrix as under:
p p0 
6
p5 p4 p3 p2 p1
Row1  
1 0 0 0 1 0 1
Row2   
G 0 1 0 0 1 1 1
Row3   
 0 0 1 0 1 1 0
Row4 
 0 0 0 1 0 1 1  4 x 7
The parity check matrix is given by:
H= [PT : I3x3]
The transpose matrix PT is given by interchanging
the rows and columns of the P matrix.
1 1 1 0
P T  0 1 1 1 
 
1 1 0 1 3 x 4
Hence, the parity check matrix is given by,
1 1 1 0 1 0 0
H  0 1 1 1 0 1 0
 
1 1 0 1 0 0 1 3 x 7
(ii)
Message bit, M[M3 M2 M1 M0]=[1 0 1 0]
Therefore, X=MG
 
1 0 0 0 1 0 1

G  1 0 1 00 1 0 0 1 1 1
 
0 0 1 0 1 1 0
0 0 0 1 0 1 1 4 x 7
Therefore, X= [1 0 1 0 : 0 1 1]
THANK YOU
L#29: ENCODER AND
DEODER FOR CYCLIC CODE

Working Operation of the encoder
• The flip-flops (F/F) are used to construct a shift register. . All the flip-flop are initialized to
zero state.
• Operation of all these flip-flops is governed by an external clock.
• The feedback switch is closed and the output switch is connected to the message input.
• First k message bits are shifted to the transmitter and also shifted into the shift register.
• After shifting the k message bits the shift register will contain the (n-k) parity bits.
• Therefore after shifting the k message bits, the feedback switch is open circuited and the
output switch is thrown to parity bit position.
• Now with every shift, the parity bits are transmitted over he channel. Thus, this encoder
generates the code words.
Encoder for an (n, k) cyclic code
• The encoder thus

performs the division
operations and
generates the
remainder.
• The remainder is nothing but the parity bits.

• When all the message bits are shifted out, what remains inside the shift register is
remainder. The encoder also consists of module-2 adders.
• The output of the coefficient multipliers i.e., g1, g2, …. etc. are added to the flip-flop
outputs to generate the parity bits.
Example
Draw the encoder for a (7,4) cyclic Hamming code generated by the generator polynomial,
G(p) = 1 + p + p3.
Solution
The generator polynomial is given by
G(p) = p3 + 0p2 +p + 1
The generator polynomial of an (n,k) cyclic code is expressed as under:
n  k 1
G ( p)  1   g i p i  p n  k
i 1
For a (7,4) cyclic Hamming code, n=7 and k=4.

Therefore, 7  4 1
G( p)  1   g p  p  1 g p  g p  p
i
i 74
1 2
2 3
i 1
Hence,
G( p)  p3  g 2 p 2  g1 p  1
Comparing, we get obtain
g1=1 and g2=0
Thus, the encoding for a (7,4) Hamming code is
Encoder for cyclic Hamming Code

Example
A message 10 11 01 is to be transmitted in cyclic code with a generator polynomial
G(p) = p4 + p3 + 1. Obtain the transmitted code word. How many check bits does the encoded
message contain? Draw the encoding arrangement for the same.
Solution
(i) First, we obtain the message polynomial M(p) i.e.,
M ( p)  m0  m1 p  m2 p 2  m3 p3  m4 p 4  m5 p5
(ii) Then, we multiply M(p) by pn-k to obtain pn-k M(p).
(iii) After that, we divided pn-k M(p) by the generator polynomial G(p).
(iv) Lastly, we obtain the code word X as under:
X  [m5 , m4 , m3 , m2 , m1 , m0 : c3c2 c1c0 ]
The given message is 1 0 1 1 0 1. Therefore, there are 6 message bits.
Thus, k=6
(ii) The degree of generator polynomial is four. The degree of generator polynomial is equal to
the number of parity bits in the code.
Therefore, number of parity bits (n-k)=4
So, code word length n=k+4=10
The message polynomial corresponding to the message 1 0 1 1 0 1 = [m5, m4,m3,m2,m1,m0] is

given as,
M ( p)  m0  m1 p  m2 p 2  m3 p3  m4 p 4  m5 p5
Substituting the values of message bits, we obtain

M ( p)  1  0 p  p 2  p 3  0 p  p 5
The generator polynomial can be written in the form as under:
G ( p)  1  0 p  0 p 2  p 3  p 4
Let us multiply the message polynomial M(p) by pn-k
Since (n-k)=4; So pn-k = p4
Therefore, p 4 M ( p)  p 4 [1  0 p  p 2  p3  p 4  p5
Or p 4 M ( p)  p 4  0 p 5  p 6  p 7  0 p 8  p 9
Now, we divide p 4 M ( p) by the generator polynomial

Remainder polynomial  p2
The code word polynomial can be obtained by adding p 4 M ( p) to the remainder polynomial C(p).
Thus,
X ( p)  p M ( p)  C ( p)
4
But, C ( p)  c0  c1 p  c2 p  ....cnk 1 p nk 1

Here, n  k 1  (10  6  1)  3
Therefore, C ( p)  c0  c1 p  c2 p 2  c3 p3
But we have obtained c(p) = p2, so, comparing the two, we obtain c0 = 0, c1 =0, c2 =1, and c3 = 0.
Therefore, Parity bits (c c c c )  (0100)
3 2 1 0
Therefore the code word is given as under:

X  m ,m ,m ,m ,m ,m : c c c c
5 4 3 2 1 0 3 2 1 0
Thus, Code word

X  [101101: 0100]
And the corresponding code word polynomial is given as under:
X ( p)  p 9  0 p 8  p 7  p 6  0 p 5  p 4  0 p 3  p 2  0 p  0
The encoded message consists of 4 check bits or parity bits c , c , c and c
3 2 1 0
Now, let us draw the encoder for the cyclic code.

The generator polynomial is given as
G ( p)  p 4  p 3  1
Or G ( p)  p 4  p 3  0 p 2  0 p  1
The generator polynomial of an (n,k) cyclic is expressed as under

n  k 1
G ( p)  1   g i p i  p n  k
i 1
Substiruting n=10 and k=6 and n-k = 4 in the above equation, we shall have
3
G ( p)  1   gi p i  p 4
i 1
Or G( p)  p  g3 p  g 2 p  g1 p  1
4 3 2
So, g1 = 0 g2 = 0 and g3 = 1
encoding arrangement
Cyclic Code Syndrome Calculation
• (n, k) cyclic code syndrome calculassions circuit:
• The register is initialized to the zero state.

S1 is closed and S2 is opened → the received r(x) is shifted into register
After this, contents of register are s(x)
S1 is opened and S2 is closed → s(x) is shifted out and the register is cleared,
ready for the next cycle
Example
Design a syndrome calculator for a (7,4) Hamming code, generated by the generator polynomial
G(p)=1+p+p3.
Solution
The given generator polynomial is G(p)=1+p+p3
The general form of generator polynomial is
G(p)=p3+g2p2+g1p+1
g1 = 1 g2 = 0
Therefore the required syndrome calculator is:

CYCLIC REDUNDANCY CHECK (CRC)
• Cyclic codes are special linear block codes with one extra property. In a cyclic code, if a
code word is cyclically shifted (rotated), the result is another code word.
• A cyclic redundancy check (CRC) is an error-detecting code commonly used in digital
networks and storage devices to detect accidental changes to raw data.
• Blocks of data entering these systems get a short check value attached, based on the
remainder of a polynomial division of their contents; on retrieval the calculation is repeated,
and corrective action can be taken against presumed data corruption if the check values do not
match.
Basic scheme for Cyclic Redundancy Checking
DIVISION IN CRC ENCODER
DIVISION IN CRC DECODER
CRC USING POLYNOMIAL
• We can use a polynomial to
represent a binary word.
• Each bit from right to left is
mapped onto a power term.
• The rightmost bit represents the
“0” power term. The bit next to it
the “1” power term, etc.
• If the bit is of value zero, the
power term is deleted from the
expression.
ADVANTAGES OF CYCLIC CODES
• Cyclic codes have a very good performance in detecting single-bit errors,

double errors, an odd number of errors.
• They can easily be implemented in hardware and software.
• They are especially fast when implemented in hardware.
•This has made cyclic codes a good candidate for many networks.
THANK YOU
L#30:Convolutional Codes

Convolutional Codes
• Convolutional codes are different from block codes by the existence
of memory in the encoding scheme.
• Though convolutional codes can detect errors, they are good for
error correction.
• These codes can be used for correcting random errors, burst errors
or both.
• Convolutional codes are also known as recurrent codes.
•The fundamental hardware unit for convolutional encoder is a
tapped shift register with (L+1) stages. L is the constraint length of
the convolutional encoder.
Convolutional Codes
The fundamental hardware unit for convolutional encoder
Here g0, g1, g2,…… etc are

tap gains which are nothing
but binary digits 0 or 1. 0
represents no connection and
1 represents connection. So,
each tap gain is a binary digit
representing a short circuit or
open circuit connection.
The message bits enter one by one into the tapped shift register, which are
then combined by mod-2 summers to generate the encoded output.
x = mL g L  ........m1 g1  m0 g 0 The name convolution encoding comes from the fact
x = i =0 mi g i that the equation the form of binary convolution.
L
Practical convolutional encoder
To provide an extra bit needed for error control, a complete
convolutional encoder must generate output bits at a rate greater
than the message bit rate, rb. This is acieved by using two or more
mod-2 adders to the registers.
Fig.: (2, 1, 2)
convolutional
encoder
 x ' j = m j −2  m j −1  m j

 x '' j = m j −2  m j
The previous encoder can be described by two generator sequences:
• Note that g(1) and g(2) are called generator sequences of the encoder. Generator
sequences are nothing but impulse response of the encoder. The encoder output is
obtained by the convolution of the input sequence with the impulse response of
the encoder, hence the name convolutional code. Impulse response of the encoder
is the response of the encoder to a single “1” bit that moves through it.
• Numerous other convolutional codes are obtained by modifying the encoder
shown in figure. If we just change the connections to the mod-2 summers, then the
encoded output will change.
• The message bits in the register are combined by mod-2 addition to form the
encoded output. The input data to the encoder, which is assumed to be binary, is
shifted into and along the shift register k-bits at a time. The number of output bits
for each k-bit input sequence is n bits. The switch samples the all mod-2 adders in
sequence, once during each bit interval.
Some important terms of convolutional codes
Code rate, R
➢ Number of message bits k = 1

➢ Number of encoded bits n = 2
➢ Rate ½ means for each one-input bit, encoder provides 2 output
bits. Encoder operates on one-bit at a time.
➢ Suppose input sequence = 10110, then total number of encoded
output bits = 5 x 2 = 10
➢ Number of message bits k = 1
➢ Number of encoded bits n = 3
➢ Rate 𝟏/𝟑 means for each one-input bit, encoder provides 3 output bits.
Encoder operates on one-bit at a time.
➢ Suppose input sequence = 11110, then total number of encoded output
bits = 5 x 3 = 15
➢ Number of message bits k = 2 (two input bits are processed at a time)

➢ Number of encoded bits n = 3 (each group of 2 inputs are transformed into 3 bits)
➢ Rate 𝟐/𝟑 means for each group of 2 input bits, encoder provides 3 output bits.
Encoder operates on 2 bits at a time.
➢ Suppose input sequence = 1110, then total number of encoded output bits = 3 + 3
=6
For convolutional codes k and n are usually very small integers

Code dimension
A convolutional code is described by 3 integers: n, k, L
n= number of encoded bits per message bit,
k= number of message bit taken at a time by the encoder,
L= encoder’s memory element.
Consider below (2, 1, 2) convolutional encoder. msg box represents

place for current input bit.
Let us assume input sequence = 1
Initial condition
Place input bit 1 into msg box
After applying 1st clock pulse (1st shift)

After applying 2nd clock pulse (2nd shift)
After 3rd clock pulse, the input bit “1” is discarded. So, the input bit 1
influences the output of the encoder for 2 shifts. Hence L (=2) is the
number of shifts over which a single message bit can influence the
encoder output bit.
Constraint length
n(L+1) = 2(2+1) = 2(3) = 6 bits

So, each message bit influences a span of n(L+1) successive output
bits. The quantity n(L+1) is called the constraint length measured in
terms of encoded output bits. Here L is the encoder’s memory or
number of bits used to represent state of the encoder.
L = 3 means single message bit influences encoder output for 3
successive shifts.
Time domain approach of
convoluational code
L
x1 = x
(1)
i =  g l(1) mi −l , i = 0,1
l=0
L
x2= x ( 2)
i =  g l( 2 ) mi −l , i = 0,1
l=0
X = {x01 x02 x11 x12 x12 x22 ......}

x2 = x02 x12 x22 x32 .....
Where, x1 = x01 x11 x12 x31 .....
Problem
The convolutional encoder for which generator sequences are:
{g 01 , g11 , g 12 } = (1, 1, 1) {g 02 , g12 , g 22 } = (1, 0, 1)
Determine the encoder sequence for the following input

sequence: (m , m , m , m , m )=1 0 0 1 1
0 1 2 3 4
Solution
(i) To obtain the bit steam xi(1) L
x1 = x
(1)
i =  g l(1) mi −l , i = 0,1
l=0
Substituting i=0 x01 = g 01 .m0 = 11 = 1
Substituting i=1 x11 = g 01 .m1 + g11.m0 = (1 0) + (11) = 1
Similarly
x12 = g 01 .m2 + g11.m1 + g 12 .m0 = (1 0) + (1 0) + (11) = 1
x31 = g 01 .m3 + g11.m2 + g 12 .m1 = (11) + (1 0) + (1 0) = 1
x14 = g 01 .m4 + g11.m3 + g 12 .m2 = (11) + (11) + (1 0) = 0
x51 = g11.m4 + g 12 .m3 = (11) + (11) = 0

x61 = g 12 .m4 = (11) = 1
Therefore code bits obtained at the output of the top adder:
x61 x51 x14 x31 x12 x11 x01 = (1 0 0 1111)
(ii) To obtain the bit steam xi( 2) L
x2= x
( 2)
i =  g l( 2 ) mi −l , i = 0,1
l=0
Substituting i=0,1….
x02 = g 02 .m0 = 11 = 1
x12 = g 02 .m1 + g12 .m0 = (1 0) + (0 1) = 0
x22 = g 02 .m2 + g12 .m1 + g 22 .m0 = (1 0) + (0  0) + (11) = 1

x32 = g 02 .m3 + g12 .m2 + g 22 .m1 = (11) + (0  0) + (1 0) = 1
x42 = g 02 .m4 + g12 .m3 + g 22 .m2 = (11) + (0 1) + (1 0) = 1
x52 = g12 .m4 + g 22 .m3 = (0 1) + (11) = 1

x62 = g 22 .m4 = (11) = 1
Therefore code bits obtained at the output of the top adder:
x62 x52 x42 x32 x22 x12 x02 = (1 0 11111)
The output encoded sequence:
X = {x x x x x x ......}
1 2
0 0
1 2
1 1
1 2
2 2
Codeword= 11 10 11 11 01 01 11
Transform domain approach of
convoluational code
Convolution in the time domain is transformed into multiplication of
Fourier transforms in frequency domain.
The general expression of the impulse response in the polynomial form:
G ( p ) = g 0 + g1 p + g 2 p 2 + ........
The polynomial corresponding to top and bottom adder are
G (1) ( p ) = g 01 + g11 p + g 12 p 2 + ........
G ( 2 ) ( p ) = g 02 + g12 p + g 22 p 2 + ........
For the top adder: g 01 = 1, g11 = 1, g 12 = 1,
So, G (1) ( p ) = 1 + p + p 2
For the bottom adder: g 1
0 = 1, g1
1
= 0, g 2 = 1,
1
So, G ( 2) ( p) = 1 + p 2
G ( 2 ) and G ( 2 ) are called generator polynomials of the code
The codeword polynomial corresponding to the top adder:
x (1) ( p ) = G (1) ( p ).m( p )

The codeword polynomial corresponding to the bottom adder:
x ( 2 ) ( p ) = G ( 2 ) ( p ).m( p )
Obtain the individual coefficients and multiplex.

Problem
Determine the codeword for the cyclic encoder for the message signal
(10011), using the transform domain approach. The impulse response
of the input top adder output path is (1,1,1) and that of the input
bottom adder output path is (1,0,1).
Solution
First, let us write the generator polynomial G(1)(p).
The impulse response of the input top adder output path of the
convolutional encoder of figure (previous page). Therefore, we have
g0(1) = 1
g1(1) = 1
and g2(1) = 1
Therefore , the generator polynomial G(1)v(p) is given by,
G(1)(p) = g0(1) + g1(1) p + g2(1) p2
Or G(1)(p) = 1 + p + p2
The given message is ( m0 m1 m2 m3 m4 = 1 0 0 1 1). Therefore, the
message polynomial is given by,
M(p) = m0 + m1(p) + m2p2 + m3p3 + m4 p4
Or M(p) = 1 + p3 + p4
Now, we find the code word polynomial for the top adder.
X(1)(p) = G(1)(p). M(p)
X(1)(p) = (1 + p + p2 ) (1 + p3 + p4 )
= 1 + p 3 + p4 + p + p 4 + p 5 + p 2 + p 5 + p 6
Or X(1)(p) = 1 + p + p2 + p3 + (1+1)p4 + (1+1)p5+ p6
Or X(1)(p) = 1 + p + p2 + p3 + p6 (addition is Mod-2)
Now, we obtain the generator polynomial G(2) (p).
The impulse response of the input bottom adder output path of
convolutional encoder of figure 16.47 is (1, 0, 1). Therefore,
g0(2) = 1, g1(1) = 0 and g2(2) = 1
Therefore the generator polynomial G(2) (p) is given by,
G(2) (p) = g0(2) + g1(2) p + g2(2) p2
Or G(2) (p) = 1 + p2
Codeword polynomial for the bottom adder is given by
X(2)(p) = G(2)(p). M(p)
Substituting equations (ii) and (iv) we get,
X(2)(p) = (1 + p2 ) (1 + p3 + p4 )
Or X(2)(p) = 1 + p2 + p3 + p4 + p5+ p6
Next, let us obtain the code sequences.
The output sequence at the output of the top-adder can be obtained
from the corresponding generator polynomial X(1)(p),
X(1)(p) = 1 + p + p2 + p3 + p6 = 1 + p + p2 + p3 + 0p4+ 0p5+ p6
Therefore, the corresponding code sequence is (1 1 1 1 0 0 1)
Similarly the code sequence for the bottom adder can be obtained
from its generator polynomial.
X(2)(p) = 1 + p2 + p3 + p4 + p5+ p6
Or X(2)(p) = 1 + 0p + p2 + p3 + p4 + p5+ p6
Thus, corresponding code sequence is (1 0 1 1 1 1 1)
Hence, the final code word at the output of the encoder is obtained
by multiplexing (interleaving) the two code sequence.
Codeword = 11 10 11 11 01 01 11
THANK YOU
L#31: Graphical representation
of Convolutional codes

Graphical representation of
Convolutional codes
3 different but related graphical representations can be used to study
of convolutional encoding.
• Code tree = Tree diagram
• Code trellis = Trellis diagram
• State diagram
Note that we can easily find output of the encoder from any of the
above diagrams.
Given a sequence of message bits and the initial state, you can use
any of following 3 diagrams to find the resulting output bits.
Code Tree
The convention used to distinguish the input binary symbols is as
follows:
Input 0 ---- specifies upper branch

Input 1 ---- specifies lower branch
Each branch of the tree represents an input symbol, with the

corresponding pair of output binary symbols indicated on the branch.
It is often convenient to represent the codeword of a convolutional
code as paths through a code tree.
A convolutional code is sometimes called a (linear) tree code.
• Code tree: the left most node is called the root. Since the encoder
has 1 binary input, there are 2 branches stemming from each node.
(starting at the root).
• The upper branch leaving each node corresponds to input 0
and the lower branch corresponds to the input digit 1.
• On each branch we have 2 binary code digits viz., the 2 outputs from
the encoder.
• Each branch of the tree represents an input symbol, with the
corresponding pair of output binary symbols indicated on the branch.
Initial state: m2 m1  00

x1  m0  m1  m2
x2  m0  m2
If, m0=0
x1  0  0  0  0 x2  0  0  0
x1 x2  00
If, m0=1
x1  1  0  0  1 x2  1  0  1 x1 x2  11
The message sequence of
length m bits produce an
encoded sequence of length
n(m+L-1) bits.
For the shift register to be

restored to its zero-initial
state, a terminating sequence
of L=3-1=2 zeros is appended
to the last input bit of the
message sequence. The
terminating zeros is called the
TAIL OF THE MESSAGE
Example
Code TRELLIS
The trellis diagram of a convolutional
code is obtained from its code tree. All
state transitions at each time step are
explicitly shown in the diagram to
retain the time dimension, as is present
in the corresponding tree diagram.
Usually, supporting descriptions on
state transitions, corresponding input
and output bits etc. are labeled in the
trellis diagram. It is interesting to note
that the trellis diagram, which
describes the operation of the encoder,
is very convenient for describing the
behavior of the corresponding decoder,
especially
when the famous “Viterbi Algorithm”
(VA) is followed.
State diagram
 Since the encoder is a linear sequential
circuit, its behavior can be described by a
state diagram.
 The encoder consists of the m message
bits stored in the shift register.
There are 2𝑚 possible states. At any
time instant, the encoder must be in one
of these states.
The encoder undergoes a state
transition when a message bit is shifted
into the encoder register as shown below.
The encoder undergoes a state
transition when a message bit is shifted
into the encoder register.
Decoding of convolutional codes
Maximum Likelihood Decoding-Virtebi Algorithm
 The Viterbi decoder examines the entire received sequence
of a given length.
 It works on maximum likelihood decoding rule which tried to
reduce the error between the detected sequence and the original
transmitted sequence.
 Trellis diagram is constructed for a system based on the received
sequence the path is straight and the trellis level by level.
 If a condition raises in such a way that there is no path for the
corresponding sequence then the viterbi decoding helps to detect the
best path based on the subsequent sequence.
The best path is termed as survivor.
Metric: the metric of a path is defined as the Hamming distance
between the each branch of each surviving path and the received
sequence.
Path metric is obtained by summing the branch metrices
Let the received codeword: 11 01 11
a0 represent the initial state,

a1, b1 represent the next possible states
Viterbi algorithm for all possible input bits is shown in fig below:
Choosing the path for smaller metrices
Note: None of the surviving path metrices is equal to zero. This

shows presence of error in the received signal.
Advantages of convoluational codes
 Convolution coding is a popular error-correcting coding method

used in digital communications.
 The convolution operation encodes some redundant information
into the transmitted signal, thereby improving the data capacity of
the channel.
 Convolution Encoding with Viterbi decoding is a powerful FEC
technique that is particularly suited to a channel in which the
transmitted signal is corrupted.
 It is simple and has good performance with low implementation
cost.
THANK YOU
L#32: Line Coding

Line Coding
•Line Coding is the process for converting digital data into digital
signal.
•Digital data is found in binary format.
•It is represented (stored) internally as series of 1s and 0s.
•Digital signal is denoted by discreet signal, which represents digital
data.
Types of Line Coding:
Unipolar
single voltage level to represent data.
•Binary 1, high voltage
•Binary 0, no voltage is transmitted.
•There are two common variations of unipolar signalling:
1. Non-Return to Zero (NRZ)
2. Return to Zero (RZ)
Unipolar Non-Return to Zero (NRZ):
•Duration of the MARK pulse (Ƭ ) is equal to the duration (To) of
the symbol slot.
Advantages:
•Simplicity in implementation
• Doesn’t require a lot of bandwidth for transmission.
Disadvantages:
•Presence of DC level (indicated by spectral line at 0 Hz).
•Contains low frequency components. Causes “Signal Droop”
•Does not have any error correction capability.
•Does not posses any clocking component for ease of synchronisation.
•Is not Transparent. Long string of zeros causes loss of synchronisation.
Unipolar Return to Zero (RZ):
•MARK pulse (Ƭ ) is less than the duration (To) of the symbol slot.
•Fills only the first half of the time slot, returning to zero for the
second half.
Advantages:
•Simplicity in implementation.
•Presence of a spectral line at symbol rate which can be used as symbol timing
clock signal.
Disadvantages:
•Presence of DC level (indicated by spectral line at 0 Hz).
•Continuous part is non-zero at 0 Hz. Causes “Signal Droop”.
•Occupies twice as much bandwidth as Unipolar NRZ.
•Is not Transparent.
Polar Signalling:
•Polar RZ •Polar NRZ
Polar NRZ:
•A binary 1 is represented by a pulse g1(t)
•A binary 0 by the opposite (or antipodal) pulse g0(t) = -g1(t).
Advantages:
•No DC component.
Disadvantages:
•Is not transparent.
Polar RZ:
•A binary 1: A pulse g1(t)
•A binary 0: The opposite (or antipodal) pulse g0(t) = -g1(t).
•Fills only the first half of the time slot, returning to zero for the
second half.
Advantages:
•No DC component.
Disadvantages:
•Occupies twice as much bandwidth as Polar NRZ.
Bipolar Signalling:
Alternate mark inversion (AMI)
•Uses three voltage levels (+V, 0, -V)
•0: Absence of a pulse
•1: Alternating voltage levels of +V and –V
Bipolar NRZ:
Bipolar RZ:
Bipolar Signalling:
Advantages:
•No DC component.
•Occupies less bandwidth than unipolar and polar NRZ schemes.
•Does not suffer from signal droop (suitable for transmission over
AC coupled lines).
•Possesses single error detection capability.
Disadvantages:
•Is not Transparent.
Coded mark inversion (CMI)
➢In telecommunication, coded mark inversion (CMI) is a non-
return-to-zero (NRZ) line code.
➢It encodes zero bits as a half bit time of zero followed by a half bit
time of one, and while one bits are encoded as a full bit time of a
constant level. The level used for one bits alternates each time one
is coded.
Manchester Signalling:
➢In telecommunication and data storage, Manchester code (also known as phase encoding,
or PE) is a line code in which the encoding of each data bit is either low then high, or high
then low, for equal time. It is a self-clocking signal with no DC component. Consequently,
electrical connections using a Manchester code are easily galvanically isolated.
➢Manchester code derives its name from its development at the University of Manchester,
where the coding was used for storing data on the magnetic drums of the Manchester Mark
1 computer.
Note: There is always a transition at the centre of bit duration.

Advantages:
•No DC component.
•Does not suffer from signal droop (suitable for transmission over
AC coupled lines).
•Easy to synchronise.
•Is Transparent.
Disadvantages:
•Because of the greater number of transitions it occupies a
significantly large bandwidth.
•Does not have error detection capability
THANK YOU
L#33: FDM and TDM system for
telemetry

Introduction
The telemetry systems can be classified into two broad types:
(i) The frequency division multiplexed type (FDM), and
(ii) the time division multiplexed type (TDM).
 The frequency division multiplexed type comprises a number of
data channels. each of which modulates a separate sub carrier
oscillator.
 FDM is more of ananalogue type system and the conventional
frequency modulation in association with other analogue
modulation methods is used in transmission of information.
 The outputs from all the above sub-carrier oscillators are
mixed/summed to forma composite signal which modulates a high
frequency carrier and the resultant wave is transmitted by fin
appropriate FM transmitter system through a radio link.
Al the receiving end
This wave is received by an FM receiver which is then FM-
demodulated and amplified. The amplified output is passed through
the same number of channels as the transmitting side data channels
consisting of band-pass filters of frequencies of the sub-carriers.
Output of each such channel is further demodulated or detected to
obtain analogue output that has originally been transmitted. This is
then stored or displayed or monitored as necessary.
Such a system is often known as FM/FM system

FM-FM system
Frequency division actually means separation of individual carrier

channels in the frequency domain. These separate carriers are termed
as sub-carrier frequencies which, in turn, modulate a higher carrier
frequency in the Rf range. The sub- carriers can be modulated in
amplitude. single side band (SSB), phase or frequency for
transmission of information.
FM-FM transmission system
FM-FM receiving system
The selection of’ multiplex baseband is dependent on the given

number of channels and the data bandwidth.
The sub-carrier frequencies are usually equally spread over a
specified frequency range called the overall bandwidth.
Division of channel bandwidths

IRIG STANDARDS
a sinewave modulating signal when frequency-modulates a carrier,
the modulated output is
where
or
For transmitting a sinewave signal by FM, the bandwidth is

extended by twice the peak deviation. Also for avoiding cross talk, a
finite guard band between adjacent channels must be there.
Inter Range Instrumentation Group (IRIG) developed two telemetry
standards of baseband configuration.:
(a)The proportional bandwidth (PBW)
(b)The constant bandwidth (CBW).
Table: IRIG Standard PBW Telemetry Channels
Table: IRIG Standard C BW Telemetry Channels
The choice of centre frequencies of the adjacent channel has been
made so that the difference is approximately 4 times the peak
deviation of theses channels.
The modulation index mβ is also standardized at 5 for ±7.5%

deviation from centre frequency of each channel in PBW—FDM
system so that for channel 1, fm should have a maximum value as
calculated below:
FHBEF=430 Hz, fc+ mβ fm = 400 Hz+5fm
Giving fm =6Hz
Time division multiplex system for telemetry
In frequency division multiplexing, the transmission channel is.

shared by the multiple signals, each being allocated a portion of the
spectrum of the bandwidth. In contrast. in time division multiplexing
(TDM) each signal can get the entire bandwidth and the signals time-
share the channel such that each signal is transmitted for a brief period
of time over a time frame. If there are n signals, the time frame is
divided into n slots and jth signal gets its chance to be transmitted
only at (n+j)th slots in lime.
The five-signal time frame explained

TDM telemetry system
TDM-PAM System
Time division multiplexing may be used with both analogue and
digital input signals. For digital signals, serial data words are
formatted such as sequential bytes—one byte of which may be
transmitted during the time interval allocated to a particular signal
slot. If there are 20 signals to be covered per frame, each frame has
to have at least 20 slots and during the coverage of each slot 1 byte
or 8 bit-word of that signal is transmitted. Thus, per time frame, there
would be 8 x 20 = 160 bits at least. Each channel, however, is
scanned only after other 19 channels have been covered. The
receiving side arrangement should be such that 8 bits after every 152
bits be made to form a single digital bit stream to be interpreted as
the signal from a single source or line.
The analogue signal is sampled at a high rate over the entire period of
transmission. The sampled signal forms a series of pulses of height
proportional to the instantaneous amplitude of the signal. The process
of production of pulses of very short intervals Tp at regular intervals T
basically becomes the pulse amplitude modulation (PAM).
The multiplexing process with sync. control

Hardware for TDM-PAM system
Sketches of waveforms of the clock, counters, one-shot multi and
gate outputs of the TDM-PAM system
PAM-PM System
For transmission purpose, the varying PAM signals are used to
modulate a carrier which is then transmitted over an RF link. Usually,
phase modulation is used and system is known as PAM/PM SYSTEM.
PAM/PM/PM system
Sometimes. a number of PAM signals are used to phase-modulate the sub-carrier
oscillator outputs which are then mixed and the composite signal is again used to
phase-modulate an appropriate RF carrier. This modulated carrier is then amplified
and transmitted through the RF channel. Such a system is called the PAM/PM/PM
system. This is schematically show n in Fig. 5.12. Thus, it becomes a combination
of TDM and FDM systems for obtaining the signal to be transmitted.
TDM-PCM System-Transmitter
PCM is a digital modulation where a binary word that represents
digital data. TDM is used to transmit serial digital data of different
channels with each channel given a time slot for transmitting its
binary word of data. The data from different channels are thus
interleaved and transmitted sequentially.
TDM-PCM System-Receiver
THANK YOU
L#34: Optical fiber telemetry
and Remote control

Introduction
➢ With the arrival of optical fibre, it was proposed that such a fibre cable can be used as a communication
channel/medium. A optical fiber cable accommodate very wide bandwidths of information or send vey high
data rate without causing error. The frequency of the optical spectrum covers a range of 3 x 1011 to 3 x 1016 Hz
with the visible band from 4.3 X 1014 to 7.5 X1014 Hz.
➢ A fibre optic telemetry/communication system, however, requires some additional components for the
capability of the cable to be exploited. A high precision single wavelength source is required which is to be
modulated through some appropriate mechanism by the coded information which, on the receiving side, is
detected, amplified, shaped and decoded.
➢ The information of source output is first converted into digital pulses using, usually, an A/D converter. These
pulses are then used to control the flashing of a light source of high intensity. The light pulses, so produced,
are allowed to pass through a fibre optic cable connecting the receiving station. Light detectors like
photocells or photodiodes receive the photo-pulses and convert them into electrical pulses which are then
amplified, reshaped and fed to the decoder-D/A converter-to obtain the original information back.
➢ If, however, the receiving station is very far away from the transmitting station, light intensity in the cable is
greatly attenuated, and, repeater units are to be installed in between; the number of which depends on the
distance. Repeater unit essentially is a detector-cum-transmitter unit where the feeble light pulses are
converted to electrical pulses, amplified and used to modulate another light beam for transmission.
Optical fiber
• An optical fiber is essentially a waveguide for light
• It consists of a core and cladding that surrounds the core
• The index of refraction of the cladding is less than that of the core, causing rays of
light leaving the core to be refracted back into the core
• A light-emitting diode (LED) or laser diode (LD) can be used for the source
• Advantages of optical fiber include:
• Greater bandwidth than copper
• Lower loss
• Immunity to crosstalk
• No electrical hazard
Optical Fiber & Communications System
Optical Fiber
• Optical fiber (or "fiber optic") refers to the medium and the technology
associated with the transmission of information as light pulses along a glass
or plastic strand or fiber.
• Optical fiber carries much more information than conventional copper wire
and is in general not subject to electromagnetic interference and the need to
retransmit signals.
• Communication system that uses light as the carrier of the information from a
source to a destination through a guided fiber cable (glass or plastic) are called fiber
optic system.
• The information carrying capacity of a communication system is directly proportional
to it’s bandwidth.
• The wider bandwidth the greater is it’s information carrying capacity.
• Because of high information carrying capacity and low attenuation ,now-a-days fiber
are finding wide application in telecommunications ,Local area networks ,sensors,
computer networks ,etc.
Advantage of fiber optic communication compared
to metallic cable communication
1. Extremely Wide (Large) Bandwidth

2. Immunity to electrostatic interference
3. Elimination of cross talk
4. Lighter weight and smaller size
5. Security
6. Greater safety
7. Longer life and easy to maintenance
Total Internal Reflection
• Optical fibers work on the principle of total
internal reflection
• With light, the refractive index is listed
• The angle of refraction at the interface between
two media is governed by Snell’s law:
n1 sin1 = n2 sin 2
Refraction & Total Internal Reflection
Numerical Aperture
• The numerical aperture of the fiber is
closely related to the critical angle and
is often used in the specification for
optical fiber and the components that
work with it
• The numerical aperture is given by the
formula:
N . A. = n12 − n22
• The angle of acceptance is twice that

given by the numerical aperture
Modes and Materials
• Since optical fiber is a waveguide, light can propagate in a
number of modes
• If a fiber is of large diameter, light entering at different angles
will excite different modes while narrow fiber may only excite
one mode
• Multimode propagation will cause dispersion, which results in
the spreading of pulses and limits the usable bandwidth
• Single-mode fiber has much less dispersion but is more
expensive to produce. Its small size, together with the fact that
its numerical aperture is smaller than that of multimode fiber,
makes it more difficult to couple to light sources
Types of Fiber
• Both types of fiber described earlier are known as step-index fibers because
the index of refraction changes radically between the core and the cladding
• Graded-index fiber is a compromise multimode fiber, but the index of
refraction gradually decreases away from the center of the core
• Graded-index fiber has less dispersion than a multimode step-index fiber
Dispersion
• Dispersion in fiber optics results from the fact that in multimode
propagation, the signal travels faster in some modes than it
would in others
• Single-mode fibers are relatively free from dispersion except for
intramodal dispersion
• Graded-index fibers reduce dispersion by taking advantage of
higher-order modes
• One form of intramodal dispersion is called material dispersion
because it depends upon the material of the core
• Another form of dispersion is called waveguide dispersion
• Dispersion increases with the bandwidth of the light source
Losses
• Losses in optical fiber result from attenuation in the material itself and from scattering,
which causes some light to strike the cladding at less than the critical angle
• Bending the optical fiber too sharply can also cause losses by causing some of the light
to meet the cladding at less than the critical angle
• Losses vary greatly depending upon the type of fiber
• Plastic fiber may have losses of several hundred dB per kilometer
• Graded-index multimode glass fiber has a loss of about 2–4 dB per kilometer
• Single-mode fiber has a loss of 0.4 dB/km or less
OPTICAL SOURCES &
DETECTORS
OPTICAL SOURCES
Optical Source find applications in the area of medical, automotive, analytical

equipments, communications and industry.
Types of Optical Source
➢ Tungsten, Deuterium, Mercury, Hollow Cathode Lamp
➢ Optical Source specifically suited to FO systems are:
✓ Light Emitting Diode (SLED, ELED, SLD)

✓ Laser Diode (DFB, DBR)
Optical Source Requirement for Performance (For Fiber Optics)
Physical dimensions to suit the optical fiber

• Narrow radiation pattern (beam width)
• Linearity (output light power proportional to driving current)
• Ability to be directly modulated by varying driving current
• Fast response time
• Adequate output power into the fiber
• Narrow spectral width (or line width)
• Stability and efficiency
• Driving circuit issues
• Reliability and cost
Basic LED operation
A PN junction acts as the active or recombination region.
• When the PN junction is forward biased, electrons and holes recombine either
radiatively (emitting photons) or non-radiatively (emitting heat). This is simple LED
operation.
Emitted wavelength depends on bandgap energy • Transitions can take place from any energy state in
either band to any state in the other band. This results in a range of different wavelengths produced in this
spontaneous emission. This accounts for the fact that LEDs produce a range of wavelengths. Typically the
range is about 80 nm or so.
Light Emitting Semiconductors
LASER: Basic Operation
Laser Transition Processes

(Stimulated and Spontaneous Emission)
Energy absorbed from the Random Coherent release of

incoming photon release of energy energy
Conditions for Large Stimulated Emissions
All three processes occur together with a balance between absorption and emission.
❖ Two conditions to be satisfied for stimulated emissions to overwhelm the

spontaneous emissions are:
The population of excited level should be greater that that at the lower energy
level and
The radiation density in the medium should be very large.
Comparison between LED and LASER
PHOTODETECTORS:
Photodetectors find applications in the area of medical, automotive, safety and analytical
equipments, cameras, communications, astronomy and industry.
Types of Photodetectors
Photodiode, Photodiode Array, Light Dependent Resistor

Avalanche Photodiode
Photomultiplier Tube, Microchannel Plate, Image Intensifier
Position Sensitive Detector
CCD
Photodetector Requirements for Performance
High sensitivity at the operating wavelength of the source
Short response time to obtain a desirable bandwidth
Minimum noise contribution
Compatible size for efficient coupling and packaging
Linear response over a wide range of light intensity
Stability of performance characteristics
Low bias voltage
Low cost
Principle of Photodetection (In Semiconductor)
In semiconductors, conduction band and valence band are separated by a
forbidden band gap.
Electrons at the valance band are bound. The electrons in the conduction band
are free and when small voltage is applied they move and causes current flow
Populating the conduction band with electrons causes the semiconductor to
conduct current.
The value of band gap Eg determines the conductive properties of semiconductor
Photodetection
p–n junction in thermal equilibrium with zero bias voltage
A PN junction can be formed by diffusing either a P-type impurity such as Boron, into a
N-type bulk silicon wafer, or a N-type impurity, such as Phosphorous, into a P-type bulk
silicon wafer.
Photodetection
(a) Photogeneration of e-h pair

(b) Reverse biased p-n junction with carrier drift in depletion region
(c) Energy band diagram showing photogeneration and separation of e-h pair
➢ All semiconductor photodetectors use photon absorption in depletion region to convert
photons into electron hole pairs, and then sense them.
➢ When a semiconductor is illuminated by light having an energy E = hγ greater than its
band gap energy Eg the light is absorbed in the semiconductor and electron hole pairs
are generated. γ is the frequency of light.
➢ Incident photon after passing through p-region will be absorbed in the depletion layer.
The absorbed energy creates EHP, Electron raises to conduction band and hole fall to
valance band. The free electron travel down the barrier and the free hole will travel up
the barrier to constitute current flow.
➢ The photon absorbed in the neutral p or n regions, outside the depletion region create
EHP, but these free charges will not move quickly due to lack of strong electric field.
Most of the free charges will diffuse slowly through the diode and may recombine
before reaching the junction. These charges produce negligible current, thus reducing
the detector’s responsivity.
➢ EHP created close to the depletion layer can diffuse and subsequently be swept across
the junction by the large electric field due to applied reverse biased voltage. An
external current is produced but it is delayed with respect to variations in the incident
optical power
➢ It is desirable that photon be absorbed in the depletion layer so that it can contribute
maximum in generation of photocurrent.
➢ Typical pn photodiodes have a rise time of the order of microseconds making them
unsuitable for high speed optical systems.
➢ The existence of electric field across the junction facilitate the rise of photocurrent
➢ The primary operating wavelength regions for FO communication systems are 850nm,
1310nm and 1550 nm. The photodetectors which are used in these systems are :
- PN junction photodiodes
- PIN photodiodes
- Avalanche photodiode
Remote Control
General Description
In general, remote control system is defined as a closed loop system consisting of (a) sensors,
(b) a signal conditioning system, (c) a transmission system to send the information to a remote
station (control), (d) a control centre or point where automatic decision making is done, (e)
the devices that would transform decision output into appropriate control signals, (f)
channels/links to transmit information to actuators at action points, often remote from the
control points, and (g) the actuators that are located at action points and are operated by the
received control signals to effect desired responses.
there are two points or stations remote to reach other-

(i) the action point or site and (ii) control point
Remote control in space programme
Remote control is extensively used in space programme and also in targeting of moving
objects detrimental to national peace. In the latter group two specific cases are (i) a command
guidance system and (ii) a beam rider system.
The link as usual could be

multi wire type, or two-wire
or even radio link type
Fig: The schematic of a command guidance system.
Area of application for remote control is growing in the industrial organizations as well. Robots, for example, are specially
considered as workhorses in hazardous and hostile envirenments of manufacturing and such workhorses are controlled from a
safe remote distance.
Typical scheme of an industrial remote
control system
Remote stations in a pipe-line
The components in a communication-based supervisory control system are (a) master station, (b) several remote
stations, and (c) the interconnecting data links or channels. The master station is the central station from which
the remote stations are controlled. Basically there are units and equipment at such remote stations that need be
controlled by receiving the status report via telemetered data.
The remote stations in a pipe-line are the pumping stations at different sites where there are
(i) equipment to operate the pumps or

other operational equipment, and (ii)
sensors of pick-ups for sensing the status
of this equipment. The central master
station gets the telemetered pick-up
signals for processing, and then
transmits back the processed signal with
an instruction for controlling the
operational equipment at the remote
service stations.

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Communication Theory and

Dr. Jayanta K Rakshit

The student will be know about

The students will be know about

• B.P Lathi, “Modern Digital and Analog

• For example, a microphone serves as the transducer that

• At the destination, a similar transducer is required to

• Hence, a signal can be a source of energy which transmits some

• An electrical impulse or an electromagnetic wave which travels a

• For example, in radio and TV broadcast, the Federal

• Hence, the transmitter must translate the information signal

• Thus, signals transmitted by multiple radio stations do not

• Usually, modulation involves the use of the information

• The choice of the type of modulation is based on several

• In wireless transmission, the channel is usually

• On the other hand, telephone channels usually

• The most common form of signal degradation comes in

• This noise is often called thermal noise.

• Interference from other users of the channel is

• Often, the statistical description that is used in a

• If the message signal is transmitted by carrier modulation,

• The fidelity of the received message signal is a

Dr. Jayanta K Rakshit

The antenna of this height is practically impossible to

This antenna can be

Multiplexing is the process of sending several

• Baseband signal: is message signal (information

• Communication that uses modulation to shift

1. Changing of the carrier amplitude Ac(t) produces

2. Changing of the carrier frequency fc produces

3. Changing of the carrier phase θ produces

1. Conventional Amplitude Modulation (DSB-FC)

Dr. Jayanta K Rakshit

The amplitude-modulated wave can be expressed as

So, the Fourier transform of AM wave is:

Considering baseband signal as,

-(ωc+ ωm) ω= -ωc -(ωc- ωm) ω=0 ωc- ωm ω= ωc ωc+ ωm

• m is merely defined as a parameter, which determines

• This is important for successful retrieval of the original

Modulation index (m) is defined as: Am

m=0.5, called under modulation

m=1.0, called 100% modulation

If the amplitude of the

Now, the information are

 m12 m22 m32 

For single tone modulation

3. If m=1, Pt=1.5Pc and efficiency will be 33.33 %

4. Over-modulation, i.e. Am>Ac , should be avoided

In terms of power efficiency, for m=1 modulation, only 33% power

Dr. Jayanta K Rakshit

1. Low level Amplitude Modulation

2. High level Amplitude Modulation

Block diagram of low level AM modulator

Block diagram of high level AM modulator

Carrier signal, c(t)=A.cos(2πfct) With fc=1000 kHz

Band pass filter

Dr. Jayanta K Rakshit

This can be verified in the time domain by observing e(t) as follows:

Signal e(t) consists of two components (1/2)m(t) and

Higher components are suppressed by low-pass filter

But . . . Amplitude Modulation has the advantage of

1. Square law detectors

i=a.Ac (1+mcosωm t) cosωc t +b. Ac 2 (1+m cosωm t) 2 cos 2 ωc t