Communication System

A system to deliver a message signal from an

information source to a user destination, with the source
and the user being physically separated from each other

Ravinder Nath
Department of Electrical Engineering
National Institute of Technology Hamirpur

1
 Communication- History
 Earliest form of communication: Vocal chord sounds generated by
animals and human beings, with reception via the human ear. Language
consist of grunts. Then languages were developed with larger
vocabularies.

 Use of Sense of sight : For relatively long distance communication,

sense of sight was used to augment that of sound. In the 2nd century B.C.
Greek telegraphers used torch signals to communicate. Different
combinations and positions of torches were used to represent the letter of
Greek alphabet. (First Digital communication).

 Use of Drum sound: Used for long distances since drum sounds were
easily distinguished from background noises

 Use of semaphore flags: In early 18th century, like the torches of ancient
Greece, semaphore flags used to communicate letters. But relied on the
2
human eye to decide the transmission length.
 Communication- History
 Using one wire electrical transmission system: In 1753, Charles
Morrison, a Scottish surgeon, suggested electrical transmission system
using one wire (plus ground) for each alphabet. A system of pith balls and
paper with letter printed on it was used at the receiver.
 Use of telegraph codes : In 1835, Samuel Morse started his experiment.
After two year Morse (US) and Sir Charles Wheatstone (UK) invented the
telegraph and made public in 1844. Electrical communication was
established and appreciated.
 Use of telephone: Alexander Graham Bell invented the telephone in 1876,
an analog electrical communication
 Use of radio broadcasting: In 1910, Lee De Forest produced a program
from the Metropolitan Opera House in NY city. After five years many
universities opened their radio stations
Use of television: In 1927 and in 1930, public television began in England
and US respectively. Regular broadcast scheduling came into play in 1939
during the opening of the NY Worlds’ Fair.
3
 Communication- History
Use of Satellite : In 1960, satellite Telstar –I was launched and began to
relay TV programs.
 Use of Computer : In 1970 the phase of computer communication
revolution. Data transfer became an integral part of our daily lives. Merging
of other disciplines in communication and computer engineering.
 Use of Personal communication: In 1980 the personal communication
revolution began and before the decades of 90s ended, the average
professional had a cellular phone , a pager, a high speed digital connection
to the internet from home for use in paying bills or accessing daily news,
and a home Fax machine.
 Use of GPS: Global Positioning satellite system (GPS) assisting now in
navigating cars through traffic jams, ships in high seas, aircraft in flight,
rockets and satellite in space, etc.
Unique set of applications and innovations are continue. Access to
wideband universal wireless, many new applications on cell phone
(3G/4G)mobile, direct satellite transmission, etc.
4
Communication- History
Year Event

1838 Telegraphy (Morse)

1876 Telephone (G.Bell)

1902 Radio transmission

1933 FM radio

1936 TV broadcasting

1953 Color TV
5
 Communication- History

Year Event

1962 Satellite communication

1972 Cellular phone

1985 Fax machines

1990s GPS, HDTV, handheld

computers

6
 Department of Electrical Engineering
National Institute of Technology Hamirpur

EE-422: Communication Systems

Ravinder Nath
 Class (4th year (VIII Sem)students)

Time Table

 Lecture
Monday 12.00am ~01.00pm
Wednesday 10.00am ~11.00am
Friday 12.00am ~01.00pm
Communication System

9
 Evaluation

 Continuous Evaluation during the semester 50%

(Mid sem Exam, Assignments, and Class Tests)

 End Semester Exam 50%

 Syllabus
UNIT-1 Introduction to Communication Systems

UNIT-2 Continuous Wave Modulation

UNIT-3 Angle Modulation

UNIT-4 Radio Receiver

UNIT-5 Pulse Modulation

UNIT-6 Digital Modulation Techniques

UNIT-7 Advanced Communication Systems

 UNIT-1:
Introduction to Communication Systems

 Communication process, sources of information,

communication channels, base band and pass
band signals, representation of signals and
systems, switched communication systems.
 UNIT-2:
Continuous-wave Modulation
 Amplitude modulation (AM), frequency
spectrum of the AM wave, representation of
AM, power relations in the AM wave, AM
detector, vestigial side-band modulation.
 UNIT-3:
Angle Modulation
 Frequency spectrum of Frequency
Modulation (FM) and Phase Modulation,
generation of FM (direct and indirect
method), demodulation of FM signal.
 UNIT-4:
Radio Receiver

 Tuned Radio-Frequency (TRF) receiver,

super heterodyne receiver.
 UNIT-5:
Pulse Modulation
 Sampling process, Pulse Amplitude
Modulation (PAM), Time Division Multiplexing
(TDM), Frequency Division Multiplexing (FDM),
Pulse Width Modulation (PWM), Pulse Position
Modulation (PPM).
 .
 UNIT-6:
Digital Modulation Techniques
 Quantization process, Pulse Code Modulation
(PCM), Differential Pulse Code Modulation
(DPCM), Delta Modulation (DM), Adaptive Delta
Modulation, Amplitude –Shift Keying (ASK),
Frequency-Shift Keying (FSK), Phase-Shift
Keying (PSK).
 UNIT-7:
Advanced Communication Systems:

 Computer communication system, satellite

communications, mobile communication.
 Text/Reference Books
1. Communication Systems by Simon Haykin, John Wiley
& Sons Pvt. Ltd.

2. An Introduction to Analog and Digital Communications

by Simon Haykin, Wiley India Pvt. Ltd.

3. Principles of Communication Systems by H. Taub and

D.L. Schilling, McGraw-Hill Education

4. Electronic Communication Systems by George

Kennedy, McGraw-Hill Education

5. Principles of Communication Engineering by Anokh

Singh, S. Chand & Co.
 Communication System
 A communication system conveys information from its source to
a destination.
Examples:
 Telephone
 TV
 Radio
 Cell phone
 Satellite
 A communication system is composed of the following:

User of
Source of Information
Information Transmitter Receiver Or
Destination
Message Recreated
signal Message signal

Three basic elements of communication system: Transmitter, Channel, Receiver.

 Communication System
There are three basic elements to every communication system, namely, transmitter,
channel, and receiver as shown in Figure
Source
of Transmitter Channel Receiver User
Information

Message Transmitted Received Estimate of message

signal signal signal signal

The purpose of a communication system is to deliver a message signal from an

information source in recognizable form to a user destination, with the source and
the user being physically separated from each other. The transmitter is located at
one point in space, the receiver is located at some other point separate from the
transmitter, and the channel is a physical medium that connects them together. The
purpose of the transmitter is to transform the message signal produced by the
source of information into a form suitable for transmission over the channel. The
receiver has the task of operating on the received signal so as to reconstruct a
recognizable form of the original message signal and to deliver it to the user
destination. 21
 Communication System
Communication involves the transmission of information, i.e. speech,
data, command, etc., from one point to another
Source
of Input Output
Transmitter Channel Receiver
Information Transducer Transducer

• Source of Information: Analog or digital

Example: Speech, music, written text

• Input Transducer: Converts the message produced by a source to a form

suitable for the communication system.
Example: Speech waves  Microphone  Voltage

• Transmitter : Couple the message to the channel (Modulation)

Modifies the message signal suitable for transmission over channel
Operations: Amplification, Modulation
Modulation encodes message into amplitude, phase or frequency of
carrier signal (AM, PM, FM) 22
 Communication System
Source
of Input Output
Transmitter Channel Receiver
Information Transducer Transducer

• Channel: Physical medium that does the transmission

Provides the connection between transmitter and receiver
Examples: Air, wires, coaxial cable, radio wave, laser beam, fiber optic cable
Every channel introduces some amount of distortion, noise and interference
• Receiver: Extracts message from the received signal (Demodulation)
Recreates the original message signal from the received signal
Operations: Amplification, Demodulation, Filtering
Goal: The receiver output is a scaled, possibly delayed version of
the message signal (ideal transmission)
Examples: TV set, radio, web client
• Output transducer: Converts electrical signal into the form desired by the system
Examples: Loudspeakers, PC
 Communication System : Information Sources
Source
of Transmitter Channel Receiver User
Information

Message Transmitted Received Estimate of message

signal signal signal signal

•Analog Information Source:

An analog information source produces messages which are defined on a
continuum. (e.g. :Microphone)
•Digital Information Source:
A digital information source produces a finite set of possible messages.
(e.g. :Typewriter)

Design of communication system is dependent on the type of information source

Communication systems Design concerns:
• Selection of the information–bearing waveform
• Bandwidth and power of the waveform
• Effect of system noise on the received information
24
• Cost of the system
 Communication System : Information Sources
Sources of information: Usually base-band signals/low pass signals.
 Speech/voice: Low bandwidth in range of 4kHz.
 Picture signals: (i) TV: Large bandwidth in the range of 4.5 MHz.
(ii) Facsimile signal (Fax): Normally low bandwidth signal.
 Data: Variable bandwidth depends on the data rate.

Speech/Voice Signal : Spectral occupancy ≤ 4kHz . Most of the essence of

speech is occupied in the band 300 Hz to 3300Hz

25
Communication System : Information Sources
Picture signals:(i) TV: Large bandwidth in the range of 4.5 MHz.
(ii) Facsimile signal (Fax): Normally low BW signal.
TV signals
• In TV signals / moving pictures the signal must make sure that the motion essence
is not lost.
• To retain the motion essence the scanning is to be done at a faster rate.
• Scanning at a fast rate will require large Bandwidth (Appox. 4.5MHz) because the
variation is fast.
• During scanning the 2-dimensional picture is converted to 1-dimentional data
stream.
Facsimile/Fax
• In still /stationary pictures in the form of maps, documents can be scanned at
slower rate.
• Scanning at a slower rate will require small Bandwidth , therefore transmission
can be made over telephone lines.
26
• Slower rate of scanning keep the channel occupied for longer time.
Communication System : Information Sources
Data
• The data can be analog or digital form

• Bandwidth requirement depends on the rate at which data is to be transmitted.

• Digital signal has discrete set of values.

• Even in digital communication, it need to be made analog communication,

because physical system (Channel) is not digital rather it is analog in nature.

27
 Communication System : Information Sources
TV Picture: Bandwidth is appox. 4.5MHz
TV picture signal is a moving picture signal which ensures that the motion essence is
not lost, which is achieved by scanning and transmitting the picture at fast rate. In the
scanning process the 2-D picture is converted into 1-D signal i.e the spatial information
is converted into temporal information. A picture shown in figure is composed of bright
and dark spots called picture elements arranged in a particular order/sequence.

w
w 4
If the picture width is „w’ and height is „h’ then the ratio 
h 3 is termed as aspect ratio.
3
Also h w .
4
Since the complete picture is scanned by 625 horizontal scanning lines the distance
h 3w 3w
between the scanning lines or height of a picture element is equal to  
625 4  625 2500
 Communication System : Information Sources
TV Picture: Bandwidth is appox. 4.5MHz

If each picture point is assumed to be of square cross-section, then the number of

picture elements per horizontal line is given as;
w 2500
Number of picture elements per line = 3w 
2500
3
Since each picture frame has 625 horizontal scanning lines and there are 25 frames
scanned every second.
2500
625  25 
Total number of picture elements scanned per second = 3
Considering the case when white and dark elements are placed alternately in a picture
then pair of such white dark elements would make one cycle and there will be number
625  25  2500
of cycles in one second =  6.51106 . Hence 6.51 MHz. Relaxing
3 2
condition of alternate bright and dark elements the maximum video frequency is 5.0 MHz.
 Communication Channels
Medium for propagation of transmitted signal.

Communication Channels

Wire line Fiber-optic Wireless Underwater Storage

Channels Channels Electromagnetic Acoustic Channels
Channels Channels

(Nature of signal: (Nature of signal: (Nature of signal: (Nature of signal: (Nature of signal:
Electrical) Mod. Light wave) Electromagnetic wave) Acoustical) Magnetic, optical)

Twisted pair wire Optical fiber cable Modes of Propagation Water: (VLF) (10 Storage disk
of waves kHz)
BW: several kHz BW: Enormous 10^14Hz
Ground-wave: (MF)
Coaxial cable Low transmission loss
(0.3-3 MHz)
BW: several MHz Small size and weight
Sky-wave: (HF) (3-30
MHz)
30
LOS: (VHF) (Few GHz)
Wired Medium
Communication Channels :
Wired Medium
Communication Channels:
Wired Medium
Communication Channels:
Wireless Medium
 Communication Channels:
Wireless Medium

Mobile Radio Channel Satellite Channel

Communication Channels: Wireless Medium
• The free space or wireless channel can be classified into mobile radio channel and
satellite channel.

• The nature of the signal is an electromagnetic wave and antennas are used at the
transmitter as well as at receiver to achieve coupling with the free space channel.

• Mobile radio channel extends the capability of the public telecommunication

network by introducing mobility into the network by virtue of its ability to
broadcast.

• Radio wave propagation will take place by scattering from the surfaces of
surrounding buildings and by diffraction over and /or around them.

• Thus the signal reaches the receiving antenna via more than one path hence
named as multipath phenomenon.

• In satellite channel the radio wave propagation take place using “line of sight”
path for communication.
 Communication Channels: Wireless Medium
Three modes of propagation of electromagnetic waves in the
atmosphere and in free space
Mainly there are three modes of propagation of electromagnetic waves in the
atmosphere and in free space namely, ground wave propagation, sky wave propagation,
and line of sight propagation.
In ground wave propagation, as illustrated in figure (a) is the dominant mode of
propagation for frequencies in the medium frequency (MF) band (0.3-3 MHz). This
band is used for Amplitude Modulation (AM) broadcasting. The range with ground
wave propagation is limited to 150 km. The earth is acting as a waveguide for the wave
and remain confined to the ground surface that is why named ground wave.
Communication Channels: Wireless Medium
Three modes of propagation of electromagnetic waves in the
atmosphere and in free space
Sky wave propagation as illustrated in figure (b) results from transmitted
signal being reflected from the ionosphere, which consists of several layers of
charged particles ranging in altitude from 50km to 400 km above the surface of the
earth. In AM broadcasting, the range with sky wave propagations limited from
140km to 400km. Sky wave propagation frequency lies in band 30-60 MHz also it
is possible in the range 30-300MHz (Very High Frequency)
Communication Channels: Wireless Medium
Three modes of propagation in the atmosphere and in free space
Line of sight propagation, as illustrated in figure (c), a message signal is
transmitted from an earth station via an uplink to a satellite, amplified in a
transponder and then retransmitted from the satellite via a downlink to another
earth station. The uplink frequency is 6GHz and downlink frequency is 4GHz. The
satellite is situated in the geostationary orbit. The line of sight offers the following
unique system capabilities (i) Broad area coverage (ii) Wide transmission
bandwidth (iii) Reliable communication link.

(c) LOS and satellite communication Three Modes of propagation in free space
 Communication Channels
Characteristics of Communication Channels and their classification

Communication Channels

Linear/ Time invariant/ Power limited/

Nonlinear Time varying Bandwidth Limited
Channels Channels Channels

Telephone channel:
Telephone channel: Optical-Fiber Channel: Band Limited
Linear Time invariant
Satellite Channel:
Satellite Channel: Mobile radio Channel: Power Limited
Non-linear Time varying

39
 Fourier Transform

Fourier Transform of g(t)

Fourier Transform of g(t)

40
 Fourier Transform

where a→0

The FT also get simplified under limiting condition a→0 to

j 1
 
 f j f
Hence
1 1
FT sgn(t )  or sgn(t ) 
j f j f 41
 Fourier Transform
• Recall our expressions for the Fourier Transform and its inverse:
 

 
1
x(t )  X ( j ) e jt d   X ( f ) e j 2 ft df
2 (synthesis)
 
 
X ( j )  X ( )  

x(t ) e  jt dt OR X ( f )  

x(t ) e  j 2 ft dt
(analysis)

• Properties: Linearity
Fax(t )  by(t )  aX ( j)  bY ( j)  ax(t )  by(t )  aX ( j)  bY ( j)

Fax(t )  by (t )   ax(t )  by (t ) e  jt dt
Proof:
1
T 
 
1 1
  ax(t ) e dt   by (t ) e  jt dt
 j t

T  T 
1    1


 a   x(t ) e dt   b  y (t ) e dt 
 jt  jt

T   T  
 aX ( j )  bY ( j )
 Time Shift Properties
Fourier Transform:
• Time Shift:
x(t  t0 )  X ( j)e jt0
Proof: 
Fx(t  t 0 ) 
1

 jt
x (t  t 0 ) e dt
T 

make a change of variables : λ  t-t 0 , which implies t  λ  t 0


Fx(t  t 0 ) 
1

 j (   t 0 )
x (  ) e d
T 

1   jt0

   x ( ) e  j 
d e
T   
 X ( j )e  jt0
• Note that this means time delay is equivalent to a linear phase shift in the
frequency domain (the phase shift is proportional to frequency).
• We refer to a system as an all-pass filter if:
X ( j )  1 X ( j )  0
• Phase shift is an important concept in the development of surround sound.
 Fourier Transform: Properties
• Time Scaling:
1 j
x(at )  X ( )
a a
Proof:


Fx(at )   x(at ) e  jt dt
1
T 
assume a  0, make a change of variables : λ  at , which implies t   / a, and dt  (1 / a)d
 
 j ( ) 1
Fax(t )   x( ) e a ( )d
1
T  a
1 1 

 ( )   x ( ) e  j ( / a ) 
d 
a T  
1 j
 ( )X ( )
a a

44
 Fourier Transform: Properties
• Time Reversal:
x(t )  X ( j)
Proof:

j
Fx(t )  X ( )
1
 X ( j )
a a a 1

We can also note that for real-valued signals:

X ( j )  X ( j ) X ( j )
 X ( j ) X ( j )  X * ( j ) (complex conjugate)

45
Time reversal is equivalent to conjugation in the frequency domain.
 Fourier Transform: Properties
• Multiplication by a complex exponential: (Frequency Translation)

x(t )e j0t  X (  0 ) for any real number 0

Proof:


F x(t)e jω0t
 
1
T  x(t )e jω0t e  jt dt




1
 x(t )e  j ( ω0 )t dt
T

 X (  ω0 )

• Why is this property useful?

• This produces a translation in the frequency domain. How might this be useful in a
communication system?
 Fourier Transform: Properties
• Multiplication by a Sinusoidal: (Modulation Theorem)
1
x(t ) cos(0t )  [ X (  0 )  X (  0 )]
2
Proof:
e j0t  e j0t 1
x(t ) cos(0t )  x(t )[ ]  [ X (  0 )  X (  0 )]
2 2

Oscillators are used for the generation of the sinusoidal signals

 Fourier Transform: Properties
• Differentiation in the Time Domain:

dn
n
x (t )  ( j ) n
X ( j )
dt

• Integration in the Time Domain:

t
1
x( )d  j X ( j )  X (0) ( )
• What are the implications of time-domain differentiation in the frequency domain?
• Why might this be a problem? Hint: additive noise.
 Fourier Transform: Properties
• Convolution in the time domain:
x(t )  h(t )  X ( j) H ( j)
Proof: 
x (t )  h(t )   x( )h(t   )d




  j t
Fx(t )  h(t )     x ( ) h(t   ) d e dt
    



  x( )   h(t   )e  jt
dt  d
   
change of variables :   t    d  dt

 
  x( )   h( )e  j (    )
d  d
   
   

   x ( )e  j 
d    h( )e  j 
d 
   
 X ( j ) H ( j )
 Fourier Transform: Properties
• Multiplication in the time domain:

1 1
x(t )  y(t ) 
2
[ X ( j )  Y ( j )] 
2  X ( )Y (   )d


• Parseval‟s Theorem:
 
1
x (t )dt  2  X ( j ) d
2 2



• Duality:

X (t )  2 x()
 Hilbert Transform
 Fourier Transform is useful for evaluating the frequency content of an
energy signal or in a limiting sense, that of a power signal.
 Fourier Transform provides the mathematical basis for analyzing and
designing frequency selective filters for signals separation on the basis
of their frequency contents.
 Another method of separating signals is based on phase selectivity,
which uses phase shifts between the pertinent signals to achieve the
desired separation. The simplest phase shift is that of 1800, which is
merely a polarity reversal in the case of sinusoidal signal.
 Another phase shift of interest is that of ± 900, when the phase angles of
all components of a given signal are shifted by ± 900 the resulting
function of time is known as the Hilbert Transform (HT) of the signal.
 Hilbert Transform has several applications which includes:
(i) Useful for representation of Band-pass signals.
(ii) Useful for representation of certain kind of modulation schemes
e.g. Single Side Band modulation. 51
 Hilbert Transform
Hilbert Transform is an operation that shifts the phase of the given x(t) by –π/2.
This can be achieved by passing the signal x(t) through a LTI system/filter with
Transfer Function
 j f 0
H  f    j sgn  f   
 j f 0
The output of this LTI system/filter for any input x(t) is xˆ (t )


x( )

1 1
xˆ (t )  x(t )   d
t  t 


xˆ (t ) =HT of x(t) (HT is a time to time transformation.)

Taking the Fourier Transform of xˆ (t ) we can write
Xˆ ( f )   j sgn( f ) X ( f ) 52
Hilbert Transform

53
Hilbert Transform

54
Hilbert Transform

55
 Hilbert Transform
Properties of Hilbert Transform
The Hilbert Transform differs from the Fourier transform in that it operates
exclusively in the time domain

 A signal x(t) has its Hilbert transform xˆ (t ) have the same amplitude spectrum

 If xˆ (t ) is the Hilbert transform of x(t), then the Hilbert transform of xˆ (t ) is - x(t)

 A signal x(t) and its Hilbert transform xˆ (t ) are orthogonal

Proof: Using Parseval‟s Theorem
  

   X ( f )   j sgn( f ) X ( f )  df

x(t ) xˆ (t )dt  X ( f )  Xˆ ( f )  df 
 
  
  

 X ( f )   j sgn( f ) X ( f ) df  j  

sgn( f ) X ( f ) X  ( f )df  j
2
 sgn( f ) X ( f ) df56
  
 Hilbert Transform
Example
Consider the cosine function
x(t )  cos(2 fct )
whose Fourier Transform is
1
X ( f )   ( f  f c )   ( f  fc ) 
2
Using the Fourier transform of xˆ (t ) we get
Xˆ ( f )   j sgn( f ) X ( f )

j
Xˆ ( f )    ( f  f c )   ( f  f c )sgn( f )
2
1
  ( f  fc )   ( f  f c )
2j

which represents the Fourier transform of the sine function sin(2 fc t ) . Hence the
Hilbert transform of the cosine function is equal to sine function
57
Hilbert Transform

58
 Analytic Signals
Motivation:
(i) It is more convenient to work with complex exponential representation
rather than trigonometric sinusoids. If we are working with cos(w0t), it is
better / convenient to work with e jw0t . In that sense e jw0t is analytic
representation of cos(w0t), because the real part of e jw0t is the real signal
cos(w0t)
(ii) The basic idea is that the negative frequency components of the Fourier
transform (or spectrum) of a real-valued function are superfluous, due to
the Hermitian symmetry of such a spectrum. These negative frequency
components can be discarded with no loss of information, provided one is
willing to deal with a complex-valued function instead. That makes certain
attributes of the function more accessible and facilitates the derivation of
modulation and demodulation techniques, such as single-sideband signals.

Therefore it is necessary to extend this concept to any arbitrary signal not

59
necessary only to sinusoidal signals.
 Analytic Signals
Analytic signals
Let x(t) is a real valued signal, its complex analytic representation is
x p (t )  x(t )  jxˆ(t )
Example: Let x(t)=cos(w0t) then its complex analytic representation is
x p (t )  cos(w0t )  j sin(w0t )  e jw0t

The real signal sinusoid has frequency components at -f0 and + f0

but complex exponential has only one frequency component at + f0
In case we want the negative frequency component -f0 then the
analytic function can be written as
xn (t )  cos(w0t )  j sin(w0t )  e jw0t

Consider x(t) is a real valued signal, its Fourier transform X(f), then the
transform has symmetry about the f = 0} axis:
X(-f) = X(f)*
where X(f)* is the complex conjugate of X(f)
 Analytic Signals
Analytic signals
The spectrum of x p (t ) is obtained by taking the Fourier transform of it and
given as
X p ( f )  X ( f )  jXˆ ( f )  X ( f )  j   j sgn( f ) X ( f )

2 X ( f ) f 0

 0 f 0
 Analytic Signals
Complex Envelop Representation of Band-pass signals
Let x(t) is a real valued Band pass signal. The spectrum of the signal x(t) is
given as X(f) and represented as

X ( f )

It can be viewed as a translated version of some low pass/base band signal

x (t ) which may be real or complex
x p (t )  x (t )e j 2 fct and x p (t )  x(t )
 Analytic Signals
Complex Envelop Representation of Band-pass signals

X ( f )

As a general case consider that x (t ) is a complex signal represented as

x(t )  xRe (t )  jxIm (t )  xI (t )  jxQ (t )
Therefore
   
x p (t )  xI (t )  jxQ (t ) e j 2 fct  xI (t )  jxQ (t ) cos(2 fct )  j sin(2 f ct ) 


Since x(t )  Re x p (t ) 
Hence x(t )  xI (t ) cos(2 fct )  xQ (t )sin(2 fct )
where xI (t )  Inphase component and xQ (t )- Quadrature phase component
 Analytic Signals
Complex Envelop Representation of Band-pass signals
X ( f )

x (t )  xI (t )  jxQ (t ) x(t )  xI (t ) cos(2 fct )  xQ (t )sin(2 fct )

where xI (t )  Inphase component and xQ (t )- Quadrature phase component
Communication System: Base-band Communication
• Communication system aims to transmit information signals (Base-
band signals) through a communication channel.
• The term baseband is used to designate the band of frequencies
representing the original signal as delivered by the input transducer
• For example: the voice signal from a microphone is a baseband
signal and contains frequencies in the range 0-4000Hz
 Communication System: Base-band Communication
In the base-band communication system, the nature of the signal is electrical and the
transmitter is simply a amplifier, the channel will be either a wire line or a wireless
channel. The receiver in the simplest form will be an amplifier.

Noise n(t)
m(t) x(t) y(t) Replica of m(t)
Transmitter Channel Hc(f) Receiver & no(t)

We begin with an ideal situation assuming distortion less channel no noise is present. In
such situation the output of the channel is given as y(t )  Kx(t   ) . The output is the
replica of the input with some attenuation factor and delay. The Transfer function of
such a channel is obtained by taking the Fourier Transform and ratio of output to input
Yf 
given by: H c  f    Ke j 2 f  . The impulse response of the channel can be obtained
Xf
by taking the inverse Fourier Transform of H c  f  . Therefore the impulse response is
given as hc  t   K  t   .
 Communication System: Base-band Communication
The magnitude and phase characteristics are obtained as Hc  f   K , Hc  f   2 f 
These are the ideal channel characteristics and can be plotted as shown in Figure

Hc  f   K Hc  f   2 f 

f f
-2πτ
From the channel characteristics it is observed that the signal magnitude characteristics
have constant amplitude characteristics and all frequencies undergoes a same amount of
delay or phase shift. Hence we can conclude that ideal channel have constant amplitude
and linear phase characteristics. This kind of channel characteristics is too much to
expect, in fact it is not even required because we will be transmitting signal with finite
bandwidth. Therefore concerned with the characteristics of the channel to hold good
within the message bandwidth, and beyond the bandwidth of the signal it is not of our
interest.
 Communication System: Base-band Communication
For the message signal bandwidth B the distortion less channel characteristics should
be Hc  f   Ke j 2 f  , f B as shown in figure
Hc  f   K Hc  f   2 f 

B
-B
-B B f f
-2πτ
Real channels do not even satisfy these less stringent characteristics and offer distortion
which is defined as: Anything that a channel does to a signal other than pure delay and
constant multiplication is considered to be distortion which can be classified into two
categories: (i) Linear distortion (ii) Non-linear distortion
Linear distortion is due to the linear characteristics of the channel. The channel can be
modeled as a linear filter. The linear distortion may be either amplitude distortion or
phase distortion. If the amplitude characteristics is Hc  f   K , f  B not a constant
then it is known as amplitude distortion as shown in Figure
 Communication System: Base-band Communication

20log H c  f  (dB)

-B B f

If Δ is small less than 1 dB or so then we can ignore the distortion and can consider the
channel with negligible amplitude distortion.
Similarly if the phase characteristics is Hc  f   2 f   m , f  B not
linear in f then it is known as phase distortion. This means that different frequencies
undergo different delays. In analog signal for speech phase diction is not serious as
ears are not sensitive. But for picture signal transmission delay distortion is serious as
eyes are sensitive to phase distortion. Moreover for data transmission delay distortion
is fatal.
 Communication System: Base-band Communication
Linear distortion is easy to tackle in communication systems by providing equalization
at the receiver. This way the effect of linear distortion in the channel can be equalized
and the message signal can be recovered without distortion. The block diagram for
channel and equalizer has been shown in Figure.

Channel Hc(f) Equalizer Heq(f)

Ke j 2 f  
The Transfer function of the equalizer is given as H eq  f   , f  B. So that
Hc  f 
Heq  f   H c  f   Ke j 2 f   , f  B For implementation of equalizer we assume
that H c  f  is either known or some strategy is to be devised for learning about
characteristics of the channel. The effect of noise is to be considered which has been
ignored in the beginning.
 Communication System: Base-band Communication
The other type of distortion is non-linear distortion which is due to the nonlinear
characteristics of the amplifier, mixer or some other components present in the
transmitter and receiver of the communication systems. The nonlinear characteristics of
the amplifier are shown in Figure
y(t)

x(t)

The input output non-linear characteristic can be modeled by some polynomial as:
y(t )  a1 x(t )  a2 x 2 (t )  a3 x3 (t )  We take a simple case y(t )  a1 x(t )  a2 x (t )
2

in this simple case too the resulting output will generate other frequencies
x(t ) : f1  y(t ) : f1 , 2 f1 For more frequencies in input x(t ) : f1 , f 2  y(t ) : f1 , 2 f1 , f 2 , 2 f 2 , f1  f 2

These extra frequencies components are named as harmonic distortion or inter-

modulation distortion and resulting into cross-talk, co-channel interference. These
additional frequency components may be distortion for our self and also for others.
Communication System: Base-band Communication
Advantages of Base-Band Communication system: In baseband
communication, where the information is transmitted using a signal with
frequencies clustered around dc (zero frequency).

 Simplicity
 Low cost
 Ease of installation and maintenance
 High rates

Limitations of Base-Band Communication system: This form of

communication system has following limitations.

 It is adequate if the distance for transmission is short.

 It is applicable only if one signal is to be sent at any one time.
 It is suitable only if wire line (twisted pair wire or cable) channel is
used as a medium for communication.
 Communication System: Base-band Communication
Limitations of Base-Band Communication system in Free Space
Let us consider the case to communicate with base-band signal in free space.

The efficiency of radiation of antenna is dependent on the wavelength of the signal.

Consider the baseband signal Speech/voice signal with frequency f = 4000Hz. The
wavelength is given as c = f λ, where c is the velocity of light and λ is the
wavelength. The wavelength of the speech/voice signal is given as
c 10 108
   2.5 105 mtr
f 4000
For efficient radiation of the this signal, having wavelength λ , require antenna size
ranging from λ/4 to λ/10 . Therefore the size of the antenna required will be
 2.5  105
  0.625  105  62.5km
4 4
This size of antenna is impractical and not preferred in communication. Therefore
Modulation is required.
 Communication System: Pass-band Communication
 A baseband signal can be transmitted over a pair of wires (like in a telephone)
or coaxial cables.
 But a baseband signal cannot be transmitted over a radio link or a satellite
because this would require a large antenna to radiate the low-frequency
spectrum of the signal.
 To overcome the above limitations it is preferred to have Pass-band
communication, in which the basic function of the transmitter is to match the
characteristics of the signal coming out of the signal source with the
characteristics of the channel/medium.
 Basically the transmitter modifies the message signal into a form suitable for
transmission over the channel. This modification is achieved by means of a
process known as Modulation.
 Hence the baseband signal spectrum must be shifted to a higher frequency by
modulating a carrier with the baseband signal . In general carrier signal is a
high frequency signal
 Transmitter: Need of Modulation
 Baseband signal transmission cannot be used for radio communication.
 To transmit the baseband signal for radio communication, modulation must be used.
 Modulation is necessary because of following advantages:

1. Reduction in height of antenna

Ease of radiation is possible with practical size of transmitting and receiving
antenna. Length of antenna = λ/4. Power radiated α (1/λ) ²
2. Increase the range of communication
3. Frequency assignment /Avoid mixing of signals
For TV and radio broadcasting, each station has a different assigned carrier
4. Improves quality of reception
5. Multiplexing is possible
Combining several signals for simultaneous transmission on one channel by
placing each signal on different carrier frequency
6. Reduce noise and interference
By using proper frequency where noise and interference are at minimum 75
 Transmitter: Modulation Process
 A signal at baseband is often used to modulate a higher
frequency carrier signal in order that it may be transmitted via
radio.

 Modulation results in shifting the signal up to much higher

frequencies (radio frequencies, or RF) than it originally spanned.

 Modulation is a signal processing operation that is basic to the

transmission of an information-bearing signal over a
communication channel, whether in the context of digital or
analog communication

 Modulation is the process of embedding the information-bearing

(message) signal into the carrier signal of high frequency

76
 Transmitter: Modulation Process
 Modulation operation is accomplished by changing some parameters of
a carrier wave in accordance with the information-bearing (message)
signal.
 The carrier wave may take one of two basic forms, depending on the
application of interest:
(i) Sinusoidal carrier wave, whose amplitude, phase, or frequency is the
parameter chosen for modification by the information-bearing signal
(ii) Periodic sequence of pulses, whose amplitude, width, or position is the
parameter chosen for modification by the information-bearing signal
 Transmitter: Continuous Wave Modulation
Modulation: A process whereby certain characteristics of a wave often
called carrier, are varied in accordance with a modulating
signal. The modulating signal is the information-bearing
/baseband signal
Continuous Wave Modulation
A parameter of a sinusoidal carrier wave generally of higher frequency is
varied continuously in accordance with the message signal

(i) Amplitude Modulation (AM)

(ii) Angle Modulation
(a) Frequency Modulation (FM)
(b) Phase Modulation (PM)

Let us define some notations for continuous wave modulation scheme

Message: m(t): message waveform or modulating signal
Carrier : c(t )  Ac cos(2 fct ), where fc is the carrier frequency in Hz
 Continuous Wave Modulation: Linear Modulation
All these Modulation scheme are studied under the general scheme :
Linear Modulation
Frequency translation is the basic operation in modulation: also called mixing,
heterodyning etc.

Linear Amplitude Modulation is achieved by Variation in Multiplier Operation

The amplitude of high frequency carrier signal c(t) is varied
according to the instantaneous amplitude of the message signal m(t)

A. DSB-SC Modulation: Double Sideband with Suppressed Carrier Modulation

T
Ac2
Sc  Lim  cos ( t )dt
2
c
T  2T
T
1  cos 2c t 
T
Ac2 Ac2
 Lim
T  2T 
T
2
dt 
2

2) DSB-SC signal
s(t )  m(t ) c(t )  Ac m(t )cos(2 fct )  Ac m(t )cos(ct )
Continuous Wave Modulation: DSB-SC Modulation

Time domain representation of DSB-SC Modulation scheme

 Continuous Wave Modulation: DSB-SC Modulation
Frequency domain representation of DSB-SC Modulation scheme
Taking the Fourier transform of m(t) and s(t) . Let M(f) is the spectrum of m(t) and
spectrum of DSB-SC modulated wave is given as S(f)

Spectrum of message signal

Ac
S( f )   M  f  fc   M  f  f c 
2

Upper sideband (USB) Lower sideband (LSB) Upper sideband (USB)

Spectrum of DSB-SC Modulated wave

 Continuous Wave Modulation: DSB-SC Modulation
Features of DSB-SC Modulation
1. Bandwidth: Bandwidth of message signal is W
Bandwidth of DSB-SC modulated wave 2W
 For real signal we know that amplitude spectrum is even symmetrical and phase
spectrum is odd symmetrical therefore –ve side can be created from its +ve side.
 So BW conservation point of view much more information is transmitted than it is
required and occupying extra BW because both side bands are present in the
modulated signal but no specific information of carrier exclusively i.e. No carrier
component.
Upper side band (USB)
2. Two sidebands:
Lower side band (LSB)
3. No carrier component: Hence DSB-SC modulation scheme
T T

  c
1 1
4. Power transmitted: ST  Lim s 2 (t )dt  Lim  A m(t ) cos c t  2
dt
T  2T T  2T
T T
T T
1  cos 2c t 
  Ac m(t ) cos ct  
1 2 1
 Lim dt  Lim Ac2 m2 (t )   dt
T  2T
T
T  2T
T
 2 
T
Ac2
 Lim
T  4T

T
m2 (t )dt  Sc Sm
 Continuous Wave Modulation: DSB-SC Modulation
Demodulation of DSB-SC: Recovery of the message signal m(t)
The process to recover the original message signal m(t) from the received
signal s(t)

The output of the product modulator is given as;

v(t )  s(t )  2cos(ct )  Ac m(t ) cos ct  2cos ct  AC m(t ) 1  cos 2ct 
 k m(t )  k m(t ) cos 2ct
Signal v(t) is passed through a LPF to retain only the lower frequencies to
get vo(t)
v0 (t )  m(t )
 Continuous Wave Modulation: DSB-SC Modulation
Demodulation of DSB-SC: Recovery of the message signal m(t)
Frequency domain representation of Demodulation operation
Spectrum of the output signal v(t) can be obtained by taking the Fourier
transform of it.
k
V ( f )  k M ( f )   M  f  2 f c   M  f  2 f c 
2
Since the BW of m(t) is W < fc , therefore LPF filter will reject the high frequencies

To accomplish this operation there is requirement of Local oscillator to have its

output in-phase and frequency like that of the carrier signal used in the transmitter.
So these demodulators are named as coherent demodulator/synchronous
demodulator.
 Continuous Wave Modulation: DSB-SC Modulation
Demodulation of DSB-SC: Recovery of the message signal m(t)
Effect of Loss of Coherence
Let the local oscillator has an output 2cos c    t   
The recovered signal y(t )  k m(t ) cos t   
Let us simplify the cases
Case-I: Assume no frequency perturbation or frequency is exactly same
as the carrier i.e. δw=0 and Ѳ≠0
Then y(t )  k m(t ) cos 

If   , y(t )  0; otherwise also cos is an attenuation factor
2
Therefore there is perfect phase matching requirement

Case-II: Assuming no phase variation or the phase of the local oscillator

is same as the carrier i.e. and Ѳ=0 and δw ≠ 0
Then y(t )  k m(t ) cos t 
Therefore the factor cos(δwt) will result distortion as it is a function of time i.e.
 Continuous Wave Modulation: DSB-SC Modulation
Demodulation of DSB-SC: Recovery of the message signal m(t)
Effect of Loss of Coherence
Case-II:
factor cos(δwt) varies with time . Sometime goes up sometime come
down. The amplitude of the signal y(t) changes with time. This effect is
called Warbling Effect
Conclusions
 There is an importance for the carrier recovery at the receiver. We
require a fairly good carrier recovery circuit at the receiver to produce a
coherent carrier which is synchronism in phase and frequency with the
incoming carrier. The circuit to accomplish it is Phase Locked Loop
(PLL) which is increasing the cost and adding complexity at the
receiver.
 This is one of the disadvantage of the DSB-SC modulation scheme.
 The best way to overcome this complexity is to include carrier in the
transmitted signal.
 Continuous Wave Modulation: AM
B. Amplitude Modulation: Double Sideband with Carrier Modulation: AM

Transmitter
AM” radio band ~ 500 to 1600 kHz

The modulated signal is s(t )  Ac [1  ka m(t ) ] cos (2fct 

 e(t ) cos (2 fct 
where ka is a constant called the amplitude sensitivity of the modulator
responsible for the generation of the modulated signal s(t)
Now this waveform in time domain have interesting properties. If |Kam(t)|<1
scaling factor instantaneous magnitude is within 1 then e(t) is always positive
and follows the shape of m(t).
 Continuous Wave Modulation: AM
B. Amplitude Modulation: Double Sideband with Carrier Modulation: AM

Base band message signal m(t)

emax (t )
emin (t )
AM wave for |ka m(t)|<1

AM wave for |ka m(t)|>1:

Over modulated
 Continuous Wave Modulation: AM
B. Amplitude Modulation: Double Sideband with Carrier Modulation: AM
 B. Amplitude Modulation: Double Sideband with Carrier Modulation: AM
Modulation index (depth of modulation) (µ) : relative variation of waveform
emax (t )  emin (t ) Half the peak to peak excursion
 
emax (t )  emin (t ) mean value
If |m(t)|<1 →µ ≤ 1 Distortion less envelop. When µ=1→100% modulation
 The modulation index (µ) is a value that describes the relationship between the
amplitude of the modulating signal and the amplitude of the carrier signal.
 This index is also known as the modulating factor or coefficient, or the degree of
modulation.
 Multiplying the modulation index by 100 gives the percentage of modulation.

Overmodulation and Distortion

 The modulation index (µ) should be a number between 0 and 1.
 If the amplitude of the modulating voltage is higher than the carrier voltage, µ will be greater
than 1, causing distortion.
 If the distortion is great enough, the intelligence signal becomes unintelligible.
 Distortion of voice transmissions produces garbled, harsh, or unnatural sounds in the speaker.
 Distortion of video signals produces a scrambled and inaccurate picture on a TV screen.
 B. Amplitude Modulation: Double Sideband with Carrier Modulation: AM

Frequency domain representation of Amplitude Modulation scheme

Taking the Fourier transform of m(t) and s(t) . Let M(f) is the spectrum of m(t) and
spectrum of AM modulated wave is given as S(f)

S( f ) 
Ac
 ( f  f c )   ( f  f c )  ka Ac M ( f  f c )  M ( f  f c )
2 2

Spectrum of message signal

Spectrum of AM wave
 Continuous Wave Modulation: AM
Features of Amplitude Modulation
1. Bandwidth: Bandwidth of message signal is W
Bandwidth of AM modulated wave 2W
 Bandwidth is the difference between the upper and lower sideband
frequencies. BW = fUSB − fLSB
 The spectrum is more or less similar to DSB-SC with some added
information
Upper side band (USB)
2. Two sidebands:
Lower side band (LSB)
3. Carrier component present: Transmitting carrier does not require any extra
BW. It does not have any information about the
message signal but it helps in designing the
receiver circuit simple. This all add inefficiency
in the scheme.

4. Power transmitted: ST  Sc  Sc Sm
 Total transmitted power (ST) is the sum of carrier power (Sc ) and power of
the two sidebands (SUSB and SLSB).
 Continuous Wave Modulation: AM
Example:
A standard AM broadcast station is allowed to transmit
modulating frequencies up to 5 kHz. If the AM station is
transmitting on a frequency of 980 kHz, what are sideband
frequencies and total bandwidth?
fUSB = 980 + 5 = 985 kHz
fLSB = 980 – 5 = 975 kHz
BW = fUSB – fLSB = 985 – 975 = 10 kHz
BW = 2 (5 kHz) = 10 kHz
Example
A 1.4 MHz carrier is modulated by a music signal that has frequency
components from 20Hz to 10kHz. Determine the range of
frequencies generated for the upper and lower sidebands.
 AM: Single Tone Modulation
Consider a modulating wave m(t) that consist of a single tone or single
frequency component i.e m(t )  Am cos(2f mt ) where Am is the amplitude of
the sinusoidal modulating wave and fm is its frequency. The corresponding
AM wave is therefore given by.
s(t )  Ac [1  ka Am cos(2f mt ) ] cos (2fct 
 Ac cos(2 fc t )  12  Ac cos 2 ( fc  f m )t   12  Ac cos 2 ( fc  f m )t 
voltage

Modulating signal m(t )  Am cos(2f mt )

The resulting amplitude modulated

carrier

Information contained in the

envelope shape.
94
time
 AM: Single Tone Modulation

Measuring Modulation Index µ =ka Am

Ac 1   

Modulation factor µ Ac 1   
AM: Single Tone Modulation

Varying Modulation Index µ=ka Am

Modulating signal
m(t )  Am cos(2f mt )

µ=0

µ = 0.5

µ = 1.0

µ > 1 Over
modulated

96
 AM: Single Tone Modulation
s(t )  Ac [1  ka m(t ) ] cos (2fct  when m(t )  Am cos(2f mt )

s(t )  Ac [1  ka Am cos(2f mt ) ] cos (2fct  Varying Modulation Index µ=ka Am

S50% (t)

Sm (t)

i = 50%

t
Ac
t
Sc (t) S100% (t)

i = 100%

t t

S(t)
S150% (t)

i = 150%

t
t
 AM: Single Tone Modulation
Frequency domain representation of Single Tone Modulation scheme
Taking the Fourier transform of the given AM wave

s(t )  Ac cos(2f ct )  12  Ac cos 2 ( f c  f m )t   12  Ac cos 2 ( f c  f m )t 

We get

S ( f )  12 Ac  ( f  f c )   ( f  f c ) Carrier component

 14  Ac  ( f  f c  f m )   ( f  f c  f m )
  Ac  ( f  f c  f m )   ( f  f c  f m )
Side band
1
4 components

Upper side-frequency
fm
Carrier
Sum of side frequency phasors
fm
98
Lower side-frequency
 AM: Single Tone Modulation
Time domain and Frequency domain representation
s(t )  Ac [1  ka m(t ) ] cos (2fct  S( f ) 
Ac
 ( f  f c )   ( f  f c )  ka Ac M ( f  f c )  M ( f  f c )
2 2
Carrier component Side bands components
m(t )  Am cos(2f mt )
Time domain Frequency domain
 AM: Single Tone Modulation
Features of Single Tone Modulation

When the percentage modulation is less than 20 %, the power in one side
frequency is less than 1% of the total power in the AM wave
Continuous Wave Modulation: AM wave Demodulation
Demodulation of AM wave: Recovery of the message signal m(t)
Envelope Detector
AM wave Demodulation: Envelope Detector
 Continuous Wave Modulation: AM
Features of Amplitude Modulation Scheme
 Amplitude modulation is the process of varying the amplitude of a carrier wave in
proportion to the amplitude of a message signal. The frequency of the carrier
remains constant
 The function of the carrier in AM is simply to provide a signal to heterodyne (mix) with
the modulated audio, to convert all the AF components to a higher frequency.


The bandwidth of an AM signal is equal to twice the highest frequency.
 The bandwidth does not depend on the power of the modulating signal.
 The spectrum of the AM wave is more or less similar to DSB-SC including
information of carrier. The BW is same

 AM scheme do not require synchronous demodulator, rather the

demodulator circuit in this scheme is very simple
 AM scheme is power inefficient compared to DSB-SC as carrier is included
which do not carry any massage information. Power Inefficiency: η ≤50% for
any message signal. If message signal is sinusoidal signal then η~33.3%
with µ=1
 For the Broadcast application where there is one transmitter and 100’s of
receiver (as the cost of receiver is very low due to simplicity) power
inefficiency can be tolerated due to low cost of receivers.
 Continuous Wave Modulation: AM

Virtues, Limitations, and Modification of Amplitude Modulation Scheme

1. Virtues
(a) Easy: Modulator and demodulators
(b) Relatively cheap

2. Limitations
(a) Wasteful of power : carrier power is significant component
(b) Wasteful of BW : For real signal the LSB and USB are identical
and have even symmetry. Transmitting both is
resulting redundant information transmission

3. Modification of AM:
(a) Double Side Band- Suppressed Carrier (DSB-SC) Modulation
(b) Suppressed Side Band Modulation
(i) Single Side Band (SSB) Modulation
(ii) Vestigial Side Band (VSB) Modulation
 Continuous Wave Modulation: SSB Modulation
Suppressed Side Band Modulation

SSB Modulation VSB Modulation

Single Side Band (SSB) Modulation
 In amplitude modulation (AM), two-thirds of the transmitted power is in
the carrier, which conveys no information.

 Signal information is contained within the sidebands.

 Single-sideband (SSB) is a form of AM where the carrier is suppressed

and one sideband is eliminated.
SSB signals offer four major benefits:
1. Spectrum space is conserved and allows more signals to be transmitted in
the same frequency range.
2. All power is channeled into a single sideband. This produces a stronger
signal that will carry farther and will be more reliably received at greater
distances.
3. Occupied bandwidth space is narrower and noise in the signal is reduced.
4. There is less selective fading over long distances.
 Continuous Wave Modulation: SSB Modulation
Methods of Generating Single Side Band (SSB) Modulated Signal
There are two methods,
i) Filtering method ii) Phasing method
Filtering method for generation of SSB
 A filter removes the undesired sideband and producing SSB
 Quartz crystal filters are the most widely used sideband filters since
they are very selective and inexpensive.
 Continuous Wave Modulation: SSB Modulation
Methods of Generating Single Side Band (SSB) Modulated Signal
Filtering method for generation of SSB

For practical generation of a SSB signal, requires a Band pass filter of ideal
characteristics. To design a filter with such ideal characteristics require high Q
factors, because larger the Q factor offers smaller BW and sharp characteristics.
Component like inductor and capacitor do not meet out the requirement of such high
Q factor but Quartz crystal can achieve such sharp characteristics.
 Continuous Wave Modulation: SSB Modulation
Methods of Generating Single Side Band (SSB) Modulated Signal
Phasing method for generation of SSB
 Another way to produce SSB uses a phase shift method to eliminate
one sideband.

 Two balanced modulators driven by carriers and modulating signals

90º out of phase produce DSB.

 Adding the two DSB signals together results in one sideband being
cancelled out.
Continuous Wave Modulation: SSB Modulation
Phasing method for generation of SSB
 Continuous Wave Modulation: SSB Modulation
Phasing method for generation of SSB
• The SSB signal is obtained by passing the DSB-SC signal through an ideal BPF
• In case LSB is to be retained then either the filter can be Ideal BPF or an ideal LPF
• Similarly if USB is to be retained then either the filter can be Ideal BPF or an ideal
HPF

½ sgn(f+fc)

-fc
-½ sgn(f-fc)

fc

1 1 1
H LP ( f )  sgn  f  fc   sgn  f  fc   sgn  f  fc   sgn  f  fc 
2 2 2
 Continuous Wave Modulation: SSB Modulation
Phasing method for generation of SSB

SSSB ( f )  H LP ( f )  SDSB ( f )
Ac 1
SSSB ( f )   M  f  fc   M  f  fc   sgn  f  fc   sgn  f  fc 
2 2
Ac A
SSSB ( f )   M  f  fc  sgn  f  fc   M  f  fc  sgn  f  f c   c  M  f  f c  sgn  f  f c   M  f  f c  sgn  f  f c 
4 4
Ac A
SSSB ( f )   M  f  fc   M  f  f c   c  M  f  f c  sgn  f  f c   M  f  f c  sgn  f  f c 
4 4

Using the following relations to obtain the inverse FT i.e the time domain representation
Ac A
m(t ) cos 2 fc t  c  M  f  fc   M  f  f c 
2 4
mˆ (t)   j sgn( f )M ( f )
m(t )e j 2 fct  M ( f  fc )

mˆ (t )e j 2 fct   jM ( f  fc )sgn( f  fc )

 Continuous Wave Modulation: SSB Modulation
Phasing method for generation of SSB
Using these relationships we get
1  Ac  Ac Ac
F   M  f  fc  sgn  f  fc   M  f  fc  sgn  f  fc     m(t )e
  ˆ  j 2 fct
 mˆ (t )e j 2 fct
4  4j 4j
Ac
 mˆ (t )sin 2 fc t
2
Hence the time domain SSB signal
Ac Ac
sSSB (t )  m(t ) cos 2 fc t  mˆ (t )sin 2 fc t for LSB
2 2
Ac Ac
sSSB (t )  m(t ) cos 2 fc t  mˆ (t )sin 2 f ct for USB
2 2
 Continuous Wave Modulation: SSB Modulation
Phasing method for generation of SSB
Ac Ac
sSSB (t )  m(t ) cos 2 fc t  mˆ (t )sin 2 fc t for LSB
2 2
This is a second method, but it is also not convenient to generate m҇(t).
High Q factor tuned filter is the first method

Demodulation of SSB signal

Coherent / synchronous demodulator:
Since carrier is not present in the transmitted signal

Loss of phase coherence

 Ac Ac 
2 m(t ) cos 2 f c t  ˆ
m (t )sin 2 f c 
t  4cos  2 f c t   (t ) 
 2 

yD (t )  m(t )cos (t )  mˆ (t )sin  (t )

If phase coherence then Ѳ(t)=0 and yD(t)=m(t)

 Continuous Wave Modulation: VSB Modulation
Motivation:
 SSB saves BW
 Difficult to generate SSB signal as realization of sharp cut-off filter using
filtering method
 Using phasing method for generation of SSB signal require Ideal HT

Further it is found that a filter has best characteristics at the centre but have
non ideal characteristics (Magnitude and phase distortion) near the edge of the
filter. If the signal m(t) has significant low frequency contents then sharp cut-off
will distort these low frequency contents.

The spectrum is rich low frequency contents, therefore there will be both
magnitude and phase distortion due to filter
 Continuous Wave Modulation: VSB Modulation
Certain kind of signals it is not a problem. Speech signal has two special
features
(i) Speech signal as a message signal has spectrum M(f) which have an
energy gap centered at the origin. For which the energy gap is from –300 Hz
to +300 Hz(600Hz wide). Hence any one sideband can be easily isolated with
the help of practical band pass filters.
(ii) Ears are less sensitive or even insensitive to phase distortion. Therefore
speech signal even if undergo phase distortion has no effect on the quality of
signal
M(f)
Spectrum of speech signal
f
-300 300

Hence for speech signal can be applied SSB scheme. But for TV picture signal
is rich in low frequency contents also eyes are sensitive to phase distortion.
Since TV picture signal has large BW therefore it is a strong case for saving of
BW.
 Continuous Wave Modulation: VSB Modulation

Now the BW=W+β, where W~ 4 to 5 MHz and β~500kHz; β<<W

VSB is a special case of SSB if β=0
 Continuous Wave Modulation: VSB Modulation

Comparing with SSB modulation scheme VSB have

(i) Most of the one sideband and some vestige or trace of the other
sideband
(ii) In place of sharp cut-off filter there is a filter with gradual roll off.

A VSB signal can be generated by passing a DSB-SC signal through a

sideband shaping filter [VSB filter]

Here the requirement of sharp cut-off sideband filter is relaxed

 Continuous Wave Modulation: VSB Modulation
General theory of Suppressed Side Band Modulation

Transmitter

Receiver

What should be the nature of filter H(f), or what class of filter will ensure the
recovery of the copy of message signal m(t). We cannot choose any filter we have
to design a suitable roll off filter to satisfy the requirement
 Continuous Wave Modulation: VSB Modulation
General theory of Suppressed Side Band Modulation

Transmitter

A
X ( f )  U ( f )H ( f )   M  f  f c   M  f  f c  H ( f )
2

Receiver

Á
V( f )   X  f  f c   X  f  f c 
2
Substituting X(f) from above we get:

V( f ) 
AÁ
 
 M  f  2 fc  H  f  fc   M  f  H  f  f c    M  f  H  f  f c   M  f  2 f c  H  f  f c 
4 
 Continuous Wave Modulation: VSB Modulation
General theory of Suppressed Side Band Modulation
AÁ
V( f )  M ( f )  H  f  f c   H  f  f c  Low frequency
4 component
AA AA
High frequency  M  f  2 fc  H  f  fc   M  f  2 fc  H  f  fc 
component 4 4

AÁ
V0 ( f )  M ( f )  H  f  f c   H  f  f c 
4
In order to get the output v0 (t )  k m(t ) as a copy of the message signal what should
be the condition on H(f). It is clear that it should be constant and not a function of
frequency
H  f  fc   H  f  fc   2H  f c   constant

Considering H  fc   1 then
2
H  f  fc   H  f  fc   1
What this condition means, to understand it consider H(f) as a HPF
 Continuous Wave Modulation: VSB Modulation
General theory of Suppressed Side Band Modulation

There is a gradual roll off in spite of sharp cut-off. The condition given is satisfied
H  f  fc   H  f  fc   1

For a specific roll off which is satisfying a symmetry about fc and –fc is called
Vestigial symmetry.

Representation of VSB signal

x(t )  xI (t )cos 2 fct  xQ (t )sin 2 fct every Bandpass signal

For DSB-SC case : xI (t )  m(t ) and xQ (t )  0

For SSB case : xI (t )  m(t ) and xQ (t )  mˆ (t )

 Continuous Wave Modulation: VSB Modulation
Representation of VSB signal
x(t )  xI (t )cos 2 fct  xQ (t )sin 2 fct every Bandpass signal
For VSB signal how to get xI (t )  ?and xQ (t )  ?
Multiply both side with in phase and quad. phase component
x(t )  xI (t )cos 2 fct  xQ (t )sin 2 fct
to get
x(t )cos 2 fct  xI (t )cos2 2 fct  xQ (t )sin 2 fct  cos 2 fct

The spectrum after passing through a LPF we get

 X  f  fc  +X  f  f c  f W
XI ( f )  
 0 f W
where A
X( f )   M  f  fc   M  f  fc  H ( f )
2
A
 M ( f )  H  f  f c  +H  f  f c  f W
XI ( f )   2
 0 f W

 Continuous Wave Modulation: VSB Modulation
Representation of VSB signal
A
 M ( f )  H  f  f c  +H  f  f c  f W
If H  f  fc   H  f  fc   1 then XI ( f )   2
 0 f W

Simplifies to X I ( f )  A M ( f ) or xI (t ) 
A
m(t )
2 2

Similarly
 j  X  f  fc  -X  f  f c  f W
XQ ( f )    
 0 f W

Simplifying
jA
XQ ( f )  M ( f )  H  f  fc  -H  f  f c 
2

xQ (t ) is obtained by passing original m(t ) through a filter HQ ( f )  j H ( f  fc )  H ( f  f c )

HQ ( f )
Therefore m(t )  m(t )  xQ (t )

A A
Hence xVSB (t )  m(t ) cos 2 fc t  m(t )sin 2 fc t
2 2
Continuous Wave Modulation: VSB Modulation
 Continuous Wave Modulation: VSB Modulation
Nature of HQ ( f ) for SSB
H(f)

-fc fc
HQ(f)/j

-1

HQ ( f )
= - sgn( f )  H Q ( f )  - j sgn( f )
j
 Continuous Wave Modulation: VSB Modulation
Nature of HQ ( f ) for VSB
H(f)

-fc fc
Continuous Wave Modulation: Comparison
 Radio Receivers: Tuned Radio Frequency Receiver (TRF)
A radio receiver is an electronic equipment that pick ups the desired modulated
signal, reject the unwanted signal, and recover the message signal from it.
Function of Radio Receivers:
 Intercept the incoming modulated signal.
 Select desired signal and reject unwanted signals
 Amplify selected RF signal
 Recover the original message signal
 Amplify the recovered message signal
Design of Receivers:
 The radio receivers has to be cost effective
 Requirements:
 Has to work according to application as for AM and FM signals
 Tune to and amplify desired radio station
 Filter out other stations
 Demodulator has to work with all radio stations regardless of carrier frequency
 Radio Receivers: Tuned Radio Frequency Receiver (TRF)
Classification of Radio Receivers: Depending upon applications
AM Receivers: Receive the broadcast of speech or music from AM
transmitters which operate on long wave, medium wave or
short wave bands.
AM Receiver

TRF Superheterodyne
A tuned radio frequency receiver (or TRF receiver) is a type of radio
receiver that is usually composed of one or more tuned radio frequency
(RF) amplifier stages followed by a detector (demodulator) circuit to extract
the audio (message) signal and an audio frequency amplifier. Popular in
the 1920s, it could be tedious to operate because each stage must be
individually tuned to the station's frequency. By the mid 1930s it was
replaced by the superheterodyne receiver invented by Edwin Armstrong
 Radio Receivers: Tuned Radio Frequency Receiver (TRF)
The TRF receiver is the simplest type of AM radio receiver. The block diagram of a Tuned Radio
Frequency (TRF) receiver is shown in the figure. Infinite number of transmitters installed
throughout the world radiates radio waves in space. In general, these transmitters radiate
different frequencies. Electromagnetic waves surrounding an antenna will induce currents of their
frequency in the antenna. A provision should be there in the receiver to select only the desired
RF signal out of a number of frequencies to the receiver. This function of selecting the desired
RF signal and rejecting the rest is achieved by the tuned voltage amplifiers in the RF amplifier
stage. Tuned RF amplifiers contain a parallel LC tuned circuit. The desired RF signal is selected
by the tuned circuit.
 Radio Receivers: Tuned Radio Frequency Receiver (TRF)
When the RF signal reaches the receiving antenna, a very weak voltage is induced in
it. It is not possible to extract the audio signal from this voltage. It is necessary first to
amplify the RF signal to a required level. This is achieved in a radio receiver with the
help of tuned RF amplifier. Thus RF amplifier serves two purposes.

1. Selection of desired RF signal

2. Amplification of the selected RF signal to a suitable value. Usually two or three
tuned RF amplifier stages are used.
The amplified RF signal is applied to the detector or demodulator stage where the
audio signal is extracted from the audio signal. Diode detectors are the most common
detector used for AM detection.
The demodulated signal amplitude will be very small in amplitude. In order to drive a
loudspeaker, it must be first amplified. The audio amplifier stage includes the audio voltage and
power amplifier. The audio voltage amplifier will be a class A amplifier and the power amplifier will
be a class B push-pull amplifier. The voltage and power level of the audio signal from the output
of the detector is raised in this stage. The signal gets sufficient energy to drive the loudspeaker.
The output of the audio amplifier stage is applied to the loudspeaker. It will reproduce the original
sound by converting the electrical audio frequency waves into sound waves
 Radio Receivers: Tuned Radio Frequency Receiver (TRF)

Features of TRF Receiver

Radio Receivers: Superheterodyne Receiver
Radio Receivers: Superheterodyne Receiver
Radio Receivers: Superheterodyne Receiver
 Radio Receivers: Superheterodyne Receiver

Local Oscillator (LO): The local oscillator is tuned to translate the

desired AM signal to intermediate frequency
(IF) which is fixed at 455kHz
Mixer: The mixer is a nonlinear device which operates on sum and
difference frequencies between the LO and carrier
 Radio Receivers: Superheterodyne Receiver
Frequency Translation & Mixing
 It is desirable in many applications to translate a band-pass signal to a new
centre frequency.
 Going from one frequency to another frequency is the main feature of mixing.
 This is achieved by multiplying and appropriate filtering: this is also named as
mixing, converting, heterodyning.

e(t) m(t) cos(2πf2t)

Input signal Mixer BPF
Centre freq f 2
m(t) cos(2πf1t)

LO 2 cos[2π(f1±f2)t]

e(t )  2m(t ) cos  2 f1t  cos 2  f1  f 2  t   m(t ) cos  2 f 2t   m(t ) cos 2  2 f1  f 2  t 
What is the basic problem that encounter in the mixer
If the input have f1+2f2 which is said to be image frequency of f1 also result in mixing to f2
 Radio Receivers: Superheterodyne Receiver

e(t) m(t) cos(2πf2t)

Input signal Mixer BPF
Centre freq f 2
m(t) cos[2π(f1 ± 2f2) t]

LO 2 cos[2π(f1±f2)t]

If the input of the formm(t) cos[2π(f1 ± 2f2) t] are also translated to f2 i.e

e(t )  2k (t ) cos 2  f1  2 f 2  t  cos 2  f1  f 2  t   k (t ) cos  2 f 2t   k (t ) cos 2  2 f1  3 f 2  t 

f1  f 2 also f1  2 f 2  f 2 when the LO is f1  f 2

Therefore f1  2 f2 and f1  2 f 2 are the image frequencies.
If both the signals f1 and f1 ± 2f2 are present then both will be received and result
cross talk
 Radio Receivers: Superheterodyne Receiver

e(t) m(t) cos(2πfIFt)

Input signal Mixer BPF
Centre freq f IF
m(t) cos2πfct

LO 2 cos[2π(fc ± fIF)t]

AM Radio
Frequency band : 540 kHz to 1.6 MHz
Intermediate Frequency fIF = 455 kHz
Image frequencies 540+910 =1450 kHz
RF filter not required to be highly selective
LO Requirement
Range of LO fc  f IF  85kHz to 1145kHz
Range of LO fc  f IF  995kHz to 2055kHz
Ratio in 1st case ~13 to 14
Ratio in 2nd case ~2 that is why we go for LO f c  f IF superheterodyne
Radio Receivers: Superheterodyne Receiver
Radio Receivers: Superheterodyne Receiver
Radio Receivers: Superheterodyne Receiver
 Fourier Transforms of Periodic Signals
Periodic signals represented in terms of FT provided if  (f) permitted

g (t )
Let g(t) finite energy arbitrary signal.

The signal g(t) can be made periodic with period TO

0 t
gT0 (t )

gT0 (t )   g (t  mT )
m 
0

0 T0 2T0 t
Periodic signal gT0 (t ) can be expressed in complex exponentials Fourier Series
as:
 
gT0 (t )   g (t  mT )   C
m 
0
n 
n e j 2 f0 n t
 (1)
143
 Fourier Transforms of Periodic Signals
T0 /2
1
where Cn 
T0 
T0 /2
gT0 (t ) e j 2 f0 nt dt

Since

 gT0 (t ) T0 / 2  t  T0 / 2
g (t )  

 0 otherwise
FT
g(t) is an a-periodic signal and can have FT as: g (t )  G( f )
Fourier series coefficients can also be written as:

Cn  f 0  g (t ) e  j 2 f0 n t dt


 f 0 G (nf 0 )
Substituting Fourier series Coefficients in Equation (1) we get:
 
gT0 (t )  
m 
g (t  mT0 )  f 0  G (
n 
n f 0 ) e j 2 f0 n t
144
Fourier Transforms of Periodic Signals
 
gT0 (t )  
m 
g (t  mT0 )  f 0  G (
n 
n f 0 ) e j 2 f0 n t
(2)

This is one of the form of Poisson „s sum formula

Fourier Transform of the RHS of Equation (2):

 FT 
gT0 (t )   g (t  mT )  f  G(n f )  ( f  nf )
m 
0 0
n 
0 0 (3)

This states that making a signal periodic in the time domain has the effect of
sampling the spectrum of the signal in the frequency domain

145
Fourier Transforms of Periodic Signals
Now consider that we sample the same signal g(t) in time domain with
sampling frequency f s so the sampling period is Ts  1/ fs

g (t )   g (nT )  (t  nT )
n 
s s  (4)

g (t ) g (t )

Sampler
0 t 0 n
Using the duality property of the Fourier Transform. The Fourier
Transform of the RHS of Equation (4) like Equation (3) we get:
FT 
g (t )  f  G( f  k f )  G ( f )
0
k 
0  (5)

This states that sampling a signal in the time domain has the effect of making
the spectrum of the signal periodic in the frequency domain
146
Fourier Transforms of Periodic Signals
g (t ) G()
Fourier Transform Spectrum

0 t 0 

Making Periodic Sampling Spectrum

Fourier Series Ck
gT0 (t )

T0 2T0 t 0 k
147
Fourier Transforms of Periodic Signals
g (t ) G()
Fourier Transform

0 t 0 

Periodicity in
Sampler

Ts 1/ fs Spectrum

g (t ) DTFT G ( )

0 n  0  
148
 Sampling Process
The sampling process is usually described in the time domain. It is an operation that is
basic to digital communication and DSP
Consider an arbitrary signal m(t) of finite energy, which is specified in time. Suppose that
we sample the signal m(t) instantaneously and at a uniform rate, once every Ts seconds.
Consequently we obtain an infinite sequence of samples spaced Ts second apart and
denoted by m (t ) . This ideal form of sampling is called instantaneous sampling
Let m (t ) denote the signal obtained by individually weighting the elements of a periodic
sequence of Dirac delta functions spaced Ts seconds apart by the sequence of numbers
m[nTs] as shown in figure and expressed as m (t )   m  nTs    t  nTs 
n 

m(t ) m (t )
Ts 1/ fs

0 t 0 n
Sampler 
m (t )   m  nT   t  nT 
n 
s s

149
 Sampling Process
To study this sampling operation in frequency domain we take the Fourier
transform of the sampled signal m (t ) and given as
FT 
m (t )  f  M ( f  k f )  M ( f )
s
k 
s  (1)

where M(f) is the Fourier transform of the original signal m(t), and fs is the
sampling rate. This equation states that the process of uniformly sampling a
continuous-time signal of finite energy results in a periodic spectrum with a
period equal to the sampling rate

Another useful expression for the Fourier transform of the ideal sampled signal m (t )
may be obtained by taking the Fourier transform of the delta function   t  nTs  is
exp(-j2π f Ts) which is DTFT

M ( f )  
n 
m( nTs ) e  j 2 n f Ts
 (2)

Suppose that the signal m(t) is strictly band-limited, with no frequency

components higher than W Hertz that is M(f) is zero for |f|≥W. The spectrum
of the sampled signal given in Equation 1 can be plotted for different cases
Sampling Process
Sampling Process
 Sampling and Reconstruction Process
From the sampled signal m (t ) the original continuous time signal m(t) can be
obtained. This process is called reconstruction process.
The Fourier transform of the sampled signal can be expressed as

M ( f )  fs M ( f )  fs  M( f k f )
k 
s

k 0
Under the following two conditions:
(i) M(f )=0 for |f |≥W
m(t ) m (t )
(ii) fs=2W
We can write the above as
0 t
1 Reconstruction 0 n
M( f )  M ( f ) -W  f  W
2W
Substituting
1 
 n   j n f 
M( f ) 
2W

n 
m
 2W
 exp  
  W
,

-W  f  W

The signal m(t) is obtained from M(f) by taking the inverse Fourier transform
 
 n   j n f 
W
1
m(t )   M ( f ) exp  j 2 ft  df   2W

n 
m  
 2W 
exp 
 W 
 exp  j 2 ft  df
 W
 Sampling and Reconstruction Process
Interchanging the order of summation and integration:
 
 n  1  j nf 
W
m(t )   M ( f ) exp  j 2 ft  df   m    exp   exp  j 2 ft  df
 n   2W  2W W  W 

 n  1   n 
W
  m   exp  j 2 f  t    df
n   2W  2W W   2W  
Integrating the integral term we get

 n  sin  2 Wt  n  
 n 
m(t )   m      2W
m  sin c  2Wt  n  -  t  
n   2W   2 Wt  n  n  

This provides an interpolation formula for

reconstructing the original signal m(t) from
the sequence of sample values m[nTs], with m(t )  hlp  t 
the sinc function sinc(2Wt) playing the role
of an interpolation function. Each sample is
multiplied by a delayed version of the
m ( f )  Hlp  f 
interpolation function, and all the resulting
waveforms are added to obtain m(t)
 Sampling and Reconstruction Process
We can now write the sampling theorem for strictly band-limited signals of finite
energy as follows:
(i) A band–limited signal of finite energy, which has no frequency components
higher than W Hertz, is completely described by specifying the values of the
signal at instants of time separated by Ts=1/2W sec.
(ii) A band–limited signal of finite energy, which has no frequency components
higher than W Hertz, may be completely recovered from a knowledge of its
samples taken at the rate of 2W samples per second.
The sampling rate (fs) of 2W samples per second, for a signal bandwidth of W Hertz is
called the Nyquist rate (fN), its reciprocal 1/2W (in seconds) is called the Nyquist interval.
The phenomenon of the overlapping of the high-frequency components with the fundamental
component in the frequency spectrum of the sampled signal is sometimes referred to as
folding. The frequency fs/2 is often known as the folding frequency (Nyquist frequency).
 Sampling and Reconstruction Process
The derivation of the sampling theorem, as described herein, is based on the
assumption that the signal m(t) is strictly band limited. In practice, however, the
information-bearing signal m(t) is not strictly band limited, with the result that some
degree of under sampling is encountered. Consequently, some aliasing is produced by
the sampling process. Aliasing refers to the phenomenon of a high frequency
component in the spectrum of the signal seemingly taking on the identity of a lower
frequency in the spectrum of its sampled version.

To combat the effects of aliasing in practice we may use two corrective measures:
(i) Prior to sampling, a low pass pre-alias filter is used to attenuate those high
frequency components of the signal that are not essential to the information
being conveyed by the signal.
(ii) The filtered signal is sampled at a rate slightly higher than the Nyquist rate.
The use of a sampling rate higher than the Nyquist rate also has the
beneficial effect of easing the design of the reconstruction filter used to
recover the original analog signal from its sampled version.
 Sampling and Reconstruction Process
When sampling has been done higher than Nyquist rate the reconstruction filter can be
allowed to have transition band, which is practically possible to realize. If the sampling
has been done at Nyquist rate then LPF must have ideal characteristics which is
practically impossible to realize.

An Addendum to the Sampling Theorem

An interesting note (given by LJ Fogel) on the sampling theorem may be
introduced here. A signal can still be defined completely by the sampled data at a
rate less than fN = 2W Hertz provided that the derivatives of the signal are known at
the sampling rate as well as the amplitude information. If all the ‘n‟ derivatives
along with amplitude information are available then the time between samples can
be (n+1)/2W seconds.
Modulation
 Pulse Amplitude Modulation (PAM)
In pulse Amplitude modulation (PAM), the amplitude of regularly spaced pulses are
varied in proportion to the corresponding sample values of a continuous message signal.
Or in PAM, amplitude of pulses is varied in accordance with instantaneous value of
message signal.
Message Signal m(t) Pulse-Amplitude PAM Signal s(t)
Modulator

Carrier Signal p(t)

m(t )

Ts 1/ fs

0 t (p)
Sampler

Pulse-amplitude modulation is somewhat similar to natural sampling, where the message

signal is multiplied by a periodic train of rectangular pulses. In natural sampling the top of
each modulated rectangular pulse varies with the message signal whereas in PAM it is
maintained flat.
 Pulse Amplitude Modulation (PAM)
There are two operations involved in the generation of the PAM signal:
1. Instantaneous sampling of the message signal m(t) every Ts seconds, where the
sampling rate fs=1/Ts is chosen in accordance with the sampling theorem.
2. Lengthening the duration of each sample so obtained to some constant value „p’

m(t )
Sampler
Ts 1/ fs

0 t (p)

In digital circuit technology, these two operations are jointly referred to as “sample and
hold”. One important reason for intentionally lengthening the duration of each sample is
to avoid the use of an excessive channel bandwidth, since bandwidth is inversely
proportional to pulse duration.
The output signal of PAM process can be viewed as the product of the input m(t) and
the carrier p(t) which is a unit pulse train with period Ts. Thus, p(t) can be expressed as

p(t )   u t  kT   u t  kT  p  ,
n 
s s p  Ts
where u(t) is unit-step function
 Pulse Amplitude Modulation (PAM)
The PAM signal will be given as

s(t )  m(t )  p(t )  m(t )   u t  kT   u t  kT
k 
s s  p 

This is the time-domain description of the input-output relationship of the

uniform-rate finite-pulsewidth sampler
m(t )
Sampler
Ts 1/ fs

0 t (p)

It is of interest to investigate the frequency domain characteristics of the sampler

output i.e the PAM signal
Since the unit pulse train p(t) is a periodic function with period Ts, it can be
represented by a Fourier series

p(t )  Ce
n 
n
jns t
, s  2 Ts
where ws is the sampling frequency in rad/sec and is related to the sapling period Ts
 Pulse Amplitude Modulation (PAM)
The Fourier coefficients Cn is given as
Ts
1
Cn 
Ts 
0
p (t )e  jns t dt

Since p(t)=1 for 0 ≤ t≤ p, then above becomes

1  e jns p p sin  ns p / 2   jns p / 2
p
1
e
 jns t
Cn  dt   e
Ts 0
jnsTs Ts ns p / 2

Substituting Cn to get the expression for p(t);

p  sin  ns p / 2   jns p / 2 jns t
p(t )  
Ts n  ns p / 2
e e

The PAM signal s(t) is given as:


s(t )   C m(t )e
n 
n
jns t

The Fourier transform of s(t) is given as:

 
p  sin  ns p / 2 
S ( )   s(t )e  jt
dt   Cn M   ns    M   ns  e  jns p /2
 n  Ts n  ns p / 2
 Pulse Amplitude Modulation (PAM)
The Fourier coefficients Cn is given as

p sin  ns p / 2   jns p / 2 p

C0  lim Cn  lim e 
n 0 n 0 T
s ns p / 2 Ts

Therefore S(w) for n=0 can be written as

p
S ( ) n 0  M ( )
Ts

For n ≠ 0, Cn is a complex quantity, but the magnitude of Cn may be written as

p sin  ns p / 2 
Cn 
Ts ns p / 2

Therefore the magnitude of the spectrum of the PAM signal is given as:
 
S ( )  C
n 
n M   ns   C
n 
n M   ns 

The amplitude spectrum of the PAM signal is plotted for different situations
 Pulse Amplitude Modulation (PAM)
The Fourier coefficients Cn is given as

Amplitude spectrum of unit pulse train p(t)

Amplitude spectrum of message signal m(t)

Amplitude spectrum of PAM signal s(t) for {ws>2wc}

 Pulse Amplitude Modulation (PAM)
The Fourier coefficients Cn is given as

Amplitude spectrum of PAM signal s(t) for {ws<2wc}

Here the original signal cannot be recovered from the spectrum using an ideal low-
pass filter due to overlapping of the complementary components.
The phenomenon of the overlapping of the high-frequency components with the
fundamental component in the frequency spectrum of the sampled signal is referred
to as folding. The frequency fs/2 is often known as the folding frequency (Nyquist
frequency).
 Multiplexing
 Multiplexing is a technique used to combine and send the multiple
data streams/signals over a single medium.

 Multiplexing is the process of simultaneously transmitting two or more

individual signals over a single communication channel. Due to
multiplexing it is possible to increase the number of communication
channels so that more information can be transmitted. The typical
applications of multiplexing are in telemetry and telephony or in the
satellite communications.
 Multiplexing
 In telecommunications and computer networks, multiplexing is a
method by which multiple analog or digital signals are combined into
one signal over a shared medium. The aim is to share a scarce
resource. For example, in telecommunications, several telephone calls
may be carried using one wire.

 The purpose of multiplexing is to enable signals to be transmitted more

efficiently over a given communication channel, thereby decreasing
transmission costs.

 De-multiplexing is achieved by using a device called De-

multiplexer (DEMUX) available at the receiving
end. DEMUX separates a signal into its component signals (one input
and n outputs).
Time-Division Multiplexing (TDM) & Frequency-Division Multiplexing(FDM)

Time-division multiplexing (TDM) is a method of transmitting and receiving

independent signals over a common signal path by means of synchronized
switches at each end of the transmission line so that each signal appears on the
line only a fraction of time in an alternating pattern. This method transmits two or
more digital signals or analog signals over a common channel.
Time-Division Multiplexing (TDM)
Time-Division Multiplexing (TDM)
 Frequency-Division Multiplexing (FDM)
Frequency-division multiplexing (FDM) is a technique by which the
total bandwidth available in a communication medium is divided into a
series of non-overlapping frequency bands, each of which is used to carry
a separate signal.
This allows a single transmission medium such as a cable or optical
fiber to be shared by multiple independent signals. Another use is to carry
separate serial bits or segments of a higher rate signal in parallel.
The most common example of frequency-division multiplexing is radio and
television broadcasting, in which multiple radio signals at different
frequencies pass through the air at the same time.
Another example is cable television, in which many television channels are
carried simultaneously on a single cable.
FDM is also used by telephone systems to transmit multiple telephone
calls through high capacity trunklines, communications satellites to
transmit multiple channels of data on uplink and downlink radio beams,
and broadband DSL modems to transmit large amounts of computer data
through twisted pair telephone lines, among many other uses.
Frequency-Division Multiplexing (FDM)
Frequency-Division Multiplexing (FDM)
Frequency-Division Multiplexing (FDM)
Frequency-Division Multiplexing (FDM)
Frequency-Division Multiplexing (FDM)
Time-Division Multiplexing (TDM) & Frequency-Division Multiplexing(FDM)
 Pulse Duration Modulation (PDM)
Pulse Width Modulation (PWM) or Pulse Duration Modulation
(PDM) or Pulse Length Modulation is an analog modulating scheme in
which the duration or width or time of the pulse carrier varies proportional
to the instantaneous amplitude of the message signal.
In Pulse-duration modulation, the samples of the message signal are used to vary
the duration of the individual pulses.

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

There are three variations of PWM:

 The leading edge of the pulse being constant, the trailing edge varies
according to the message signal. (a)
 The trailing edge of the pulse being constant, the leading edge varies
according to the message signal.(b)
 The center of the pulse being constant, the leading edge and the
trailing edge varies according to the message signal.(c)
 Pulse-Position Modulation (PPM)
In PDM, long pulses expend considerable power during the pulse while
bearing no additional information. If this unused power is subtracted from
the PDM, so that only time transitions are preserved, we obtain a more
efficient type of pulse modulation known as Pulse Position Modulation

In PPM, the position of a pulse relative to its un-modulated time of occurrence is

varied in accordance with the message signal.

Pulse Position Modulation (PPM) is an analog modulating scheme in

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

The transmitter has to send synchronizing pulses (or simply sync pulses) to
keep the transmitter and receiver in synchronism. These sync pulses help
maintain the position of the pulses. The following figures explain the Pulse
Position Modulation.
 Pulse-Position Modulation (PPM)
Pulse position modulation is done in accordance with the pulse width
modulated signal. Each trailing of the pulse width modulated signal
becomes the starting point for pulses in PPM signal. Hence, the position
of these pulses is proportional to the width of the PWM pulses.
Pulse-Position Modulation (PPM)
 Pulse-Position Modulation (PPM)

Generation of PPM signal

 Pulse-Position Modulation (PPM)
Generation of PPM signal
Comparison between PAM, PWM and PPM
 UNIT-6:
Digital Modulation Techniques
 Quantization process, Pulse Code Modulation
(PCM), Differential Pulse Code Modulation
(DPCM), Delta Modulation (DM), Adaptive Delta
Modulation, Amplitude –Shift Keying (ASK),
Frequency-Shift Keying (FSK), Phase-Shift
Keying (PSK).
 Quantization Process
 Quantization, in mathematics and digital signal processing, is the process
of mapping input values from a large set (often a continuous set) to output
values in a (countable) smaller set, often with a finite number of elements.

 Rounding and truncation are typical examples of quantization processes.

 Quantization is involved to some degree in nearly all digital signal

processing, as the process of representing a signal in digital form ordinarily
involves rounding.

 Quantization also forms the core of essentially all lossy

compression algorithms.

 The difference between an input value and its quantized value (such
as round-off error) is referred to as quantization error.

 A device or algorithmic function that performs quantization is called

a quantizer. An analog-to-digital converter is an example of a quantizer.
 Quantization Process

 The digitization of analog signals involves the rounding off of the

values which are approximately equal to the analog values. The
method of sampling chooses a few points on the analog signal and
then these points are joined to round off the value to a near stabilized
value. Such a process is called as Quantization.

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

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

 The quantizing of an analog signal is done by discretizing the signal

with a number of quantization levels. Quantization is representing the
sampled values of the amplitude by a finite set of levels, which means
converting a continuous-amplitude sample into a discrete-time signal.
 Quantization Process
The following figure shows how an analog signal gets quantized. The blue line
represents analog signal while the brown one represents the quantized signal.

Both sampling and quantization result in the loss of information. The quality
of a Quantizer output depends upon the number of quantization levels used.
The discrete amplitudes of the quantized output are called as representation
levels or reconstruction levels. The spacing between the two adjacent
representation levels is called a quantum or step-size.
 Quantization Process
Types of Quantization
There are two types of Quantization - Uniform Quantization and Non-uniform Quantization.
The type of quantization in which the quantization levels are uniformly spaced is
termed as a Uniform Quantization.
The type of quantization in which the quantization levels are unequal and mostly the
relation between them is logarithmic, is termed as a Non-uniform Quantization.

There are two types of uniform quantization. Mid-Rise type and Mid-Tread type. The
following figures represent the two types of uniform quantization.
 Quantization Process
 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.
Quantization Error
During quantization, there is always a difference in the values of its input and output.
The quantization process results in an error, which is the difference of those values.
The difference between an input value and its quantized value is called a Quantization
Error. A Quantizer is a logarithmic function that performs Quantization rounding off
the value. An analog-to-digital converter (ADC) works as a quantizer.
 Quantization Process
Quantization Noise
It is a type of quantization error, which usually occurs in analog audio signal,
while quantizing it to digital. For example, in music, the signals keep
changing continuously, where a regularity is not found in errors. Such errors
create a wideband noise called as Quantization Noise.
 Pulse Modulation

• Message Signal
Classification of Pulse Modulation
 Pulse Code Modulation (PCM)
 So far we have gone through different modulation techniques. The one
remaining is digital modulation, which falls under the classification of pulse
modulation. Digital modulation has Pulse Code Modulation (PCM) as the main
classification. It further gets processed to Delta Modulation and Adaptive Delta
Modulation (ADM).

 In Pulse Code Modulation a message signal is represented by a sequence of

coded pulses, which is accomplished by representing the signal in discrete form
in both time and amplitude. A signal is Pulse Code modulated to convert its
analog information into a binary sequence, i.e., 1s and 0s. The output of a Pulse
Code Modulation will resemble a binary sequence.

 Instead of a pulse train, PCM produces a series of numbers or digits, and hence
this process is called as digital. Each one of these digits, though in binary code,
represent the approximate amplitude of the signal sample at that instant.

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

coded pulses. This message signal is achieved by representing the signal in
discrete form in both time and amplitude.
Pulse Code Modulation (PCM)
Pulse Code Modulation (PCM)
 Pulse Code Modulation (PCM)
Basic Elements of PCM
The transmitter section of a Pulse Code Modulator circuit consists of Sampling,
Quantizing and Encoding, which are performed in the A/D converter section.
The low pass filter prior to sampling prevents aliasing of the message signal.
The basic operations in the receiver section are regeneration of impaired
signals, decoding, and reconstruction of the quantized pulse train. The
following figure is the block diagram of PCM which represents the basic
elements of both the transmitter and the receiver sections.
 Pulse Code Modulation (PCM): Basic Elements
Low Pass Filter (LPF):
This filter eliminates the high frequency components present in the input analog
signal which is greater than the highest frequency of the message signal, to avoid
aliasing of the message signal.
Sampler:
This is the circuit which uses the technique that helps to collect the sample data at
instantaneous values of the message signal, so as to reconstruct the original signal.
The sampling rate must be greater than twice the highest frequency component W of
the message signal, in accordance with the sampling theorem.
Quantizer:
Quantizing is a process of reducing the
excessive bits and confining the data.
The sampled output when given to
Quantizer, reduces the redundant bits
and compresses the value.
Encoder:
To exploit the advantage of sampling and quantizing for the purpose of making it
robust to noise, interference and other channel degradation, we require the use of an
encoding process. It translate the discrete set of sample values to an appropriate
code. Example each quantized pulse translated to a binary code of sequence 0,1.
Encoding minimizes the bandwidth used.
 Pulse Code Modulation (PCM): Basic Elements
Regeneration:
The most important feature of PCM system lies in the ability to control the effect of
distortion and noise produced by transmitting a PCM signal through a channel. This
capability is accomplished by reconstructing the PCM signal by means of a chain of
regenerative repeaters located at sufficiently close spacing along the transmission route.
Three basic functions are performed by a regenerative repeaters: equalization, timing,
and decision making

The equalizer shapes the received pulses so as to compensate for the effects of
amplitude and phase distortions produced by the transmission characteristics of the
channel. The timing circuitry provides a periodic pulse train, derived from the received
pulses, for sampling and equalized pulses at the instant of time where the SNR is a
maximum. The sample so extracted is compared to a predetermined threshold in the
decision-making device. In each bit interval decision is then made whether the
received symbol is a 1 or a 0 on the basis of whether the threshold is exceeded or not.
 Pulse Code Modulation (PCM): Basic Elements
Regeneration:
If the threshold is exceeded, a
clean new pulse representing
symbol ‘1’ is transmitted to the next
repeater. Otherwise, another clean
new pulse representing symbol ‘0’
is transmitted. In this way, the
accumulation of distortion and
noise in a repeater span is
completely removed. Ideally, the
regenerated signal is exactly the
same as the signal originally
transmitted.
In practice, however, the regenerated signal departs from the
original for two main reasons.
1. The unavoidable presence of channel noise and interference causes the
repeater to make wrong decisions occasionally, thereby introducing bit errors
into the regenerated signal.
2. If the spacing between received pulses deviates from its assigned value, a jitter
is introduced into the regenerated pulse position, thereby causing distortion.
 Pulse Code Modulation (PCM): Basic Elements
Decoder:
The decoder circuit decodes the pulse coded waveform to reproduce the original
signal. This circuit acts as the demodulator. Also the decoding process involves
generating a pulse from the code.

Reconstruction Filter:
After the digital-to-analog conversion is done by the regenerative circuit and the
decoder, a Low Pass Filter (LPF) is employed, called as the reconstruction filter to get
back the original signal.

Hence, the Pulse Code Modulator circuit digitizes the analog signal given,
codes it, and samples it. It then transmits in an analog form. This whole
process is repeated in a reverse pattern to obtain the original signal.
 Pulse Code Modulation (PCM)
Electrical Representation of Binary Symbols:
There are several line codes that can be used for the electrical representation
of binary symbols 1 and 0 as:
1. On-Off signaling, in which symbol 1
is represented by transmitting a
pulse of constant amplitude for the
duration of the symbol, and symbol
0 is represented by switching off the
pulse. (a)
2. Nonreturn-to-zero (NRZ)
signaling, in which symbols 1 and
0 are represented by pulses of
equal positive and negative
amplitudes.(b)
3. Return-to-zero (RZ) signaling, in
which symbol 1 represented by
positive rectangular pulses of
half-symbol width, and symbol 0
represented by transmitting no
pulse. (c)
 Pulse Code Modulation (PCM)
Electrical Representation of Binary Symbols:
4. Bipolar return-to-zero (BRZ) signaling,
which uses three amplitude levels as
shown in (d). The positive and
negative pulses of equal amplitude
are used alternately for symbol 1,
and no pulse is used for symbol 0.
Thus the power spectrum of the
transmitted signal has no dc
component as well as low frequency
components.
5. Split-phase (Manchester code) which is
shown in (e). In this method of
signaling, symbol 1 is represented by
a positive pulse followed by a
negative pulse, with both pulses
being of equal amplitude and half-
symbol width. For symbol 0 , the
polarities of these two pulses are
reversed.
Virtues, Limitations, and Modifications of PCM
Virtues, Limitations, and Modifications of PCM
 Bandwidth of PCM

Assume m(t) is band-limited to B hertz.

Minimum sampling rate = 2B samples / second
A/D output = n bits per sample (quantization level M=2n)
Assume a simple PCM without redundancy.
Minimum channel bandwidth = bit rate /2

 Bandwidth of PCM signals:

BWPCM  nB (with sinc functions as orthogonal basis)
BWPCM  2nB (with rectangular pulses as orthogonal basis)

 For any reasonable quantization level M, PCM requires much

higher bandwidth than the original m(t).
 Differential Pulse-Code Modulation (DPCM)
 The samples that are highly correlated, when encoded by PCM technique, leave
redundant information behind.
 To process this redundant information and to have a better output, it is a wise
decision to take predicted sampled values, assumed from its previous outputs
and summarize them with the quantized values. Such a process is named
as Differential PCM (DPCM) technique.
 Often voice and video signals do not change much from one sample to next.
 Such signals has energy concentrated in lower frequency.
 Sampling faster than necessary generates redundant information.
 Can save bandwidth by not sending all samples.

 Send true samples occasionally.

 In between, send only change from previous value.
 Change values can be sent using a fewer number of bits than true
samples.
Examples (CCITT standards) * 32 k bits / s (4-bit quantization and 8 k samples /s) for 3.2kHz
* 64 k bits / s (4-bit quantization and 16 k samples /s) for 7 kHz
 Differential Pulse-Code Modulation (DPCM)

For slowly varying signals, a future sample can predicted from past
samples.

e(nTs) = m(nTs) -ˆm(nTs)

eq(nTs) = e(nTs) + q(nTs)
mq(nTs) = ˆm(nTs) + eq(nTs) DPCM Receiver
mq(nTs) = ˆm(nTs) + e(nTs) +q(nTs)
mq(nTs) = m(nTs)+q(nTs)
DPCM Transmitter

Tapped-delay-line filter used as a prediction filter

Differential Pulse-Code Modulation (DPCM)
Quantization error is accumulated.
Differential Pulse-Code Modulation (DPCM)
Quantization error is not accumulated.
 Delta Modulation (DM)

 The sampling rate of a signal should be higher than the Nyquist

rate, to achieve better sampling.

 If this sampling interval in a DPCM is reduced considerably, as the

sampling interval is reduced, the signal correlation will be higher.
the sample-to-sample amplitude difference is very small, as if the
difference is 1-bit quantization, then the step-size is very small
i.e., Δ (delta).

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

viewed as 1-bit DPCM scheme.

 The type of modulation, where the sampling rate is much higher

and in which the step size after quantization is of smaller value Δ,
such a modulation is termed as delta modulation
 Delta Modulation (DM)
Features of Delta Modulation:
 An over-sampled input is taken to make full use of a signal correlation.
 The quantization design is simple.
 The input sequence is much higher than Nyquist rate.
 The quality is moderate.
 The design of the modulator and the demodulator is simple.
 The stair-case approximation of output waveform.
 The step-size is very small, i.e., Δ (delta).
 The bit rate can be decided by the user.
 It requires simpler implementation.

In its basic form, DM provides a staircase approximation to the

oversampled version of the message signal. The difference between the
input and the staircase approximation is quantized into two levels, namely,
± Δ , corresponding to positive and negative differences, respectively.
 Delta Modulation (DM)
 If the approximation falls below the signal at any sampling epoch, it is increased by Δ.
 If, on the other hand, the approximation lies above the signal, it is diminished by Δ.

Denoting the input signal m(t), and its staircase approximation as mq(t), the basic
principle of delta modulation may be formalized in the following set of discrete-
time relations:
 Delta Modulation (DM)

The modulator consist of a comparator, quantizer, and accumulator . The

output of the accumulator is
n n
mq (nTs )    sgn e iT    e
i 1
s
i 1
q iTs 
Delta Modulation (DM)

Transmitter

Receiver
 Delta Modulation (DM)
Delta modulation is subject to types of quantization error:
1. Δ is small: Slop overload distortion
2. Δ is large: Granular noise
Delta Modulation (DM)
 Delta Modulation (DM)
There is an optimum value for Δ in terms of signal bandwidth,
signal power, and sampling frequency.

Example. Let m(t)  A cos  2πfo t  and the sampling frequency, f s  kf o

where k is an integer, k  2. What is the minimum value of  for no slope overload?

dm(t )
 2 Af o π sin  2πf o t  which has the maximum value of 2 Af o π.
dt
1 1
Ts  
f s kf o
 dm(t ) 2 Aπ
For no slope overload,  max  2 A f o .   
Ts dt k
 Delta Modulation (DM)
Advantages of DM over DPCM
 1-bit quantizer
 Very easy design of modulator & demodulator
However, there exists some noise in DM and following are the types of
noise.
 Slope Over load distortion (when Δ is small)
 Granular noise (when Δ is large)

Adaptive Delta Modulation

In digital modulation, we come across certain problems in determining the
step-size, which influences the quality of the output wave.
The larger step-size is needed in the steep slope of modulating signal and a
smaller step size is needed where the message has a small slope. Hence, it
would be better if we can control the adjustment of step-size, according to
our requirement. This is the concept of Adaptive Delta Modulation (ADM).
 Pass Band Digital Modulation
Digital Modulation provides more information capacity, high data security,
quicker system availability with great quality communication. Hence, digital
modulation techniques have a greater demand, for their capacity to convey
larger amounts of data than analog ones.
Amplitude Shift Keying (ASK)
The amplitude of the resultant output depends upon the input data whether it
should be a zero level or a variation of positive and negative, depending upon
the carrier frequency.
ASK is a type of Amplitude Modulation which represents the binary data in the
form of variations in the amplitude of a signal.
Following is the diagram for ASK modulated waveform along with its input.
 Pass Band Digital Modulation
Frequency Shift Keying (FSK)
The frequency of the output signal will be either high or low, depending upon
the input data applied
FSK is the digital modulation technique in which the frequency of the carrier
signal varies according to the discrete digital changes. FSK is a scheme of
frequency modulation.
Following is the diagram for FSK modulated waveform along with its input.

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.
 Pass Band Digital Modulation
Phase Shift Keying (PSK)
Phase Shift Keying is the digital modulation technique in which the phase
of the carrier signal is changed by varying the sine and cosine inputs at a
particular time. PSK technique is widely used for wireless LANs, bio-metric,
contactless operations, along with RFID and Bluetooth communications.
Binary Phase Shift Keying (BPSK)
This is also called as 2-phase PSK (or) Phase Reversal Keying. In this technique, the
sine wave carrier takes two phase reversals such as 0° and 180°.
Following is the image of BPSK Modulated output wave along with its input.
R Nath 2021, NITH
Some notes about DM
Bit rate = sampling rate n
Reconstructed signal z (nTs )   y(iTs ) where y(iTs) = +1 or -1 and  is
i 1
the step size.
Types of noise
* Quantization noise: step size  takes place of smallest
quantization level.
* Granular noise: z(nTs) is always different from z((n-1)Ts).
* Slope overload noise: maximum slope of output signal is  / Ts.
 
 too small: slope overload noise.
 too large: quantization noise and granular noise.
Amplitude Modulation (AM): Frequency Domain

m(t)
s(t ) Amplitude
s(t )  Ac [1  ka m(t ) ] cos (2fct 
Modulator
c(t )  Ac cos (2fct 
Fourier Transform

S( f ) 
Ac
 ( f  f c )   ( f  f c )  ka Ac M ( f  f c )  M ( f  f c )
2 2
Carrier component
Side bands components

M(f)
kaAc M(f-fc)/2

0 fc
0 fm -fc
fc

227
 Transmitter: Modulation Process
Why Use Modulation ?
• Carrying one signal on the other ~ Using Carrier

• Modulated carrier transmitted

• Problems with transmitting with base band signal:

 Antennas difficult at low frequencies. Antenna length ~ wavelength c/f.
 Noise and interference at low frequencies.
 Can’t share with others.

• Easier to transmit carrier at high frequency

 Can choose convenient frequency: Smaller antenna, avail useful propagation effects

 Fractional bandwidth much smaller: Easier design of antenna and other components,
can have many frequency channels
228
 Transmitter: Modulation Process
Need for Modulation

RF electromagnetic wave
conveys the message
Antenna signal

Transmitter
Message signal
/ Base band
signal (LF)
• Antenna length ~ wavelength c/f
• Problem of signal interference and noise
• Need for division of frequency bands
• Need for RF carrier modulation, i.e.
rendering carrier to covey BB signal
https://www.slideserve.com/mac/ch
apter-3-continuous-wave- 229
modulation
 Transmitter: Modulation Process
Characteristics of carrier signal modified in accordance with message signal
Choice of carrier signal classification of modulation process

Continuous Wave Pulse

Modulation Modulation
c(t) = sinusoidal wave c(t) = pulse train
Modulation Techniques

Base band signal Analog Digital

m(t) Pulse
to be transmitted Pulse
s(t) PCM
low frequency Modulator
Modulated
Sinusoidal carrier c(t) signal
high frequency
PAM PDM PPM
c(t )  A cos (2fct   
Frequency Phase
230
Amplitude Angle
 Amplitude Modulation (AM)

Amplitude Modulation: The amplitude of the carrier is modified in

accordance with the message signal“
AM” radio band ~ 500 to 1600 kHz.

Simplest case of AM is where carrier is just ON and OFF

voltage
This signal controls whether
carrier is turned ON or OFF

The resulting modulated carrier

231
time
 Amplitude Modulation (AM)

Modulation by a sine wave: s(t )  Ac [1  ka Am cos(2f mt ) ] cos (2fct 

voltage

Modulating signal m(t )  Am cos(2f mt )

The resulting amplitude modulated

carrier

Information contained in the

envelope shape.

time

232
Amplitude Modulation (AM)

Varying Modulation Index µ=ka Am

Modulating signal
m(t )  Am cos(2f mt )

µ=0

µ = 0.5

µ = 1.0

µ > 1 Over
modulated

233
Amplitude Modulation (AM)

Measuring Modulation Index µ

Modulation factor µ

234
 Amplitude Modulation (AM)
Modulation by a sine wave

s(t )  Ac cos(2f ct )  12  Ac cos 2 ( f c  f m )t   12  Ac cos 2 ( f c  f m )t 

Fourier Transform

S ( f )  12 Ac  ( f  f c )   ( f  f c ) Carrier component

 14  Ac  ( f  f c  f m )   ( f  f c  f m )
  Ac  ( f  f c  f m )   ( f  f c  f m )
Side band
1
4 components

Upper side-frequency
fm
Carrier
Sum of side frequency phasors
fm
235
Lower side-frequency
Amplitude Modulation (AM)

236
 Amplitude Modulation (AM)

m(t) s(t )  Ac [1  ka m(t ) ] cos (2fct 

Amplitude
Modulator
c(t )  Ac cos (2fct 

| ka m(t ) | 1 | ka m(t ) | 1 Over modulation

237
Correct AM Distortion in envelope
Amplitude Modulation (AM): Frequency Domain

m(t)
s(t ) Amplitude
s(t )  Ac [1  ka m(t ) ] cos (2fct 
Modulator
c(t )  Ac cos (2fct 
Fourier Transform

S( f ) 
Ac
 ( f  f c )   ( f  f c )  ka Ac M ( f  f c )  M ( f  f c )
2 2
Carrier component
Ac/2 Side bands

M(f)
kaAc M(f-fc)/2

0 fc
0 fm -fc
fc

238
 Amplitude Modulation (AM)
Power in AM

-fc 0 fc

 Power in Carrier
 Power in two Side bands
 Base band signal “hidden” in side bands
 One side band redundant

239
 Amplitude Modulation (AM): Variants
Suppressed carrier Double Side Band Modulation (SC-BSB):

To achieve power efficient transmission ….

Balanced Modulator

m(t)
s1 (t )
s(t )

Ac cos (2fct 

s2 (t )
- m(t)

240
Amplitude Modulation (AM): Variants

m(t)

Ac cos (2fct 

241