UNIT 1: INTRODUCTION TO COMMUNICATION SYSTEMS

Historical Notes- Communication Systems

1843 – Samuel Morse builds the first long distance electric telegraph line.
1876 – Alexander G Bell and Thomas A. Watson exhibit an electric telephone in Boston
1877 – Thomas Edison patents the phonograph.
1902 – Guglielmo Marconi transmits radio signals
1925 – John Logie Baird transmits the first television signal.
1934 - Police Radio uses conventional AM mobile communication system.
1935 - Edwin Armstrong demonstrate FM
1946 - First public mobile telephone service - push-to-talk
1958 – Chester Carlson presents the first photocopier for office use.
1960 - Improved Mobile Telephone Service, IMTS - full duplex
1960 - Bell Lab introduce the concept of Cellular mobile system
1963 – First geosynchronous communications satellite is launched
1965 – First email sent
1966 – Charles Kao realizes that silica-based optical waveguides
1968 - AT&T proposes the concept of Cellular mobile system to FCC.
1969 – The first hosts of ARPANET, Internet's ancestor are connected.
1981- Hayes Smart modem introduced.
1981- Nordic Mobile Telephone, automatic mobile phone is put into operation
1983 - Advanced Mobile Phone System (AMPS), FDMA, FM
1983 – Microsoft Word software is launched.
1989- Tim Berners-Lee and Robert Cailliau build World Wide Web
1991 - Global System for Mobile (GSM), TDMA, GMSK
1991 - U.S. Digital Cellular (USDC) IS-54, TDMA, DQPSK
1993 - IS-95, CDMA, QPSK, BPSK
1994 – Internet radio broadcasting is born.
1999- Sirius satellite radio is introduced. Napster peer-to-peer file sharing is launched.
2001 – First digital cinema transmission by satellite in Europe of a feature film
2003 – Myspace is launched.
2003 – Skype video calling software is launched.
2004 – Facebook is launched.
2005 – YouTube, the video sharing site, is launched.
2006 – Twitter is launched.
2007 – iPhone is launched.
2010 – Instagram is launched. iPad is created.
2011 – Snapchat is launched.
Elements of a General Communication System
Communication anywhere, anytime, involving transmission of information from one point to other point
(places).

The basic components of a communication system are information source, input transducer, transmitter,
communication channel, receiver, output transducer, and destination.
• Input transducer converts the message to an electrical signal.
• The transmitter converts the input signal to transmitted signal suited for the transmission channel.
• Transmission cannel is the electric medium that bridges the distance from source to destination.
• The receiver converts the received signal in a form appropriate for the output transducer.
• Output transducer converts the output electrical signal the desired message form.
Communication Channels
1. Telephone Channels (twisted pair of wires)
2. Coaxial Channel (50Ω, 75Ω)
3. Optical Fiber (single mode, multimode)
 Enormous potential bandwidth (70 x 1012 Hz)
 Low transmission losses (0.158 db/km at 1.55μm)
 Immunity to electromagnetic interference
 Small size and weight
 Ruggedness and flexibility
4. Wireless broadcast channels (AM,FM,TV)
 super heterodyne receivers
5. Mobile radio channels
 multipath fading, dispersive
6. Satellite channels (geosynchronous , low orbit)
 Broad-area coverage
 reliable transmission links
 wide transmission bandwidth
Further Classification of channels
 Linear (e.g. telephone) or nonlinear (e.g. satellite)
 Time invariant (e.g. optical fiber)) or time variant (mobile radio channel)
 Bandwidth limited (e.g. telephone channel)
 Power limited (e.g. optical fiber link abs satellite)

Basic operations in the transmitter

1. Modulation (Analog) and Coding (Digital)
Basic operations in the receiver
1. Amplification
2. Filtering
3. Demodulation (Analog) and Decoding (Digital)
Effects of the channel on the transmitted signal
1. Attenuation: decreasing the signal strength;
2. Distortion of the signal waveform: caused by channel characteristics (linearity, frequency response,
etc.)
3. Noise: contamination of random natural signals added to the transmitted signal
 4. Interference: contaminations of extraneous signal of human sources – machinery, power lines,
digital switching circuits, etc.
Simplex- This type of communication is one-way. Examples are: Radio, TV broadcasting, Beeper (personal
receiver)
Full Duplex- Most electronic communication is two-way and is referred to as duplex.
When people can talk and listen simultaneously, it is called full duplex. The telephone is an example of this
type of communication.
Half Duplex- The form of two-way communication in which only one party transmits at a time is known as
half duplex. Examples are: Police, military, etc. radio transmissions, Citizen band (CB), Family radio, and
Amateur radio

Fundamental Limitations of a Communication System

While designing a communication system, an engineer generally faces several limitations
1. Noise limitation
The noise may be defined as an unwanted form of energy which tends to interfere with the transmission and
reception of the desired signals in a communication system. Noise can be classified into two categories
depending upon the source, such as:
 External noise
 Internal noise
External noise is that types of noise whose sources are external to a communication system. Examples of
external noise are atmospheric noise, galactic noise and industrial noise.
Internal noise is that type of noise whose sources are internal to a communication system. Examples of
internal noise are thermal noise (due to random motion of the charged particles like electrons) and shot
noise. This type of noise is unavoidable and it forms a basic limitation on transmission and reception of
signals.
Typical noise variation is measured in micro-volts.
In long distance communication systems, operating with limited amount of signal power, the signal may be
as small as noise or even smaller than the noise. Thus, in such cases, the presence of noise severally limits
the capabilities of a communication system.
2. Bandwidth limitation - band of frequencies allocated for transmission of message signal. Ex: band
limited case: telephone channel, mobile communication
The information theory states that the greater is the transmission bandwidth of a communication system, the
more is the information that can be transmitted.
For example, suppose one is listening to music in an AM radio. The complete amount of information
available to the human ear is contained in a frequency range upto 15 kHz, i.e., musical information extends
upto a frequency of 15 kHz. However, in AM radio the maximum modulating frequency is restricted upto 5
kHz and hence the maximum bandwidth of AM transmission is 10 kHz.
Therefore, an AM radio receiver cannot reproduce all the information contained in the music because this
will require a bandwidth of 30 kHz.
On the other hand, the bandwidth allocated to a FM transmission is about 200 kHz. Thus, on FM receiver
can easily reproduce the transmitted information without any distortion.
This means that a FM system has a better fidelity than an AM system.
Thus, we can conclude that bandwidth is a major fundamental limitation of a communication system.
3. Transmission power: average power of transmitted signal. Ex: power limited case: space
communication, satellite channel
4. Hartley-Shannon law:
In a communication system the efficient transmission depends on bandwidth of the channel, Signal power
and instrumental requirements. Shanon’s shows that the rate of transmission without error is given as
 C = B log2 (1 + SNR) bit/s
Where B = channel bandwidth, SNR = signal to noise ratio
Increase in bandwidth means the increase of frequencies. So the transmission speed also increases.
And decrease in signal power.
4. Equipment limitation
The noise and bandwidth limitation dictate theoretically what can or cannot be achieved in terms of
performance in a communication system. However, this theoretical limit may not be realized in a practical
system due to equipment limitations.
For example, the theory might require a band pass filter with a quality factor of 100 at a centre frequency of
1 kHz. Such a filter cannot be realised in practice. Even if a filter with nearly identical characteristics is
built, the cost may exceed. Thus equipment limitation is another major problem in a communication system.

Bandwidth
Bandwidth (BW) is that portion of the electromagnetic spectrum occupied by a signal. It is also the
frequency range over which a receiver or other electronic circuit operates. More specifically, bandwidth is
the difference between the upper and lower frequency limits of the signal or the equipment operation range.
Suppose, the bandwidth of the voice frequency range from 300 to 3000 Hz. The upper frequency is f 2 and
the lower frequency is f1. The bandwidth, then, is BW = f2 - f1 = 3000-300 =2700 Hz

Analog and Digital Signals and Systems

Signals: "A detectable physical quantity or impulse (as a voltage, current, or magnetic field strength) by
which messages or information can be transmitted." Or "A signal is a source of information, generally a
physical quantity, which varies with respect to time, space, temperature like any independent variable"
Systems: A System is any physical set of components that takes a signal, and produces a signal. In terms of
engineering, the input is generally some electrical signal X, and the output is another electrical signal
(response) Y. However, this may not always be the case. Consider a household thermostat, which takes input
in the form of a knob or a switch, and in turn outputs electrical control signals for the furnace.
Analog Signal
 An analog or analogue signal is any continuous signal.
 Ex: Sound, Light, Temperature, Position or Pressure
Digital Signal
 Digital signals consist of patterns of bits of information. These patterns can be generated in many
ways, each producing a specific code. Modern digital computers store and process all kinds of
information as binary patterns. All the pictures, text, sound and video stored in this computer are
held and manipulated as patterns of binary values.
 Ex: Computers, CDs, DVDs, and other digital electronic devices.
Analog System
 It’s a system that is continuous both in time and magnitude. And also its graph is also like a
waveform going in both positive and negative cycles
 A system designed to operate on continuous values such as voltage, pressure, temperature, RPM, etc
Digital System
 Digital systems are designed to store, process, and communicate information in digital form. They
are found in a wide range of applications, including process control, communication systems, digital
instruments, and consumer products. The digital computer, more commonly called the computer, is
an example of a typical digital system.
…………………….
Review of Signals
In communication system, the frequency domain refers to the analysis of mathematical
functions or signals with respect to frequency, rather than time. Put simply, a time-domain graph shows how
a signal changes over time, whereas a frequency-domain graph shows how much of the signal lies within
each given frequency band over a range of frequencies. A frequency-domain representation can also include
information on the phase shift that must be applied to each sinusoid in order to be able to recombine the
frequency components to recover the original time signal.
The process of (electronic) communication involves the generation, transmission and reception of various
types of signals. The communication process becomes fairly difficult, because:
a) the transmitted signals may have to travel long distances (there by undergoing severe attenuation)
before they can reach the destination i.e., the receiver.
b) of imperfections of the channel over which the signals have to travel
c) of interference due to other signals sharing the same channel and
d) of noise at the receiver input.
A signal can be anything which conveys information Signals are classified into the following categories:
1. Continuous Time and Discrete Time Signals - A signal is said to be continuous when it is defined
for all instants of time.
A signal is said to be discrete when it is defined at only discrete instants of time
2. Deterministic and Non-deterministic Signals - A signal is said to be deterministic if there is no
uncertainty with respect to its value at any instant of time. Or, signals which can be defined exactly
by a mathematical formula are known as deterministic signals.
A signal is said to be non-deterministic if there is uncertainty with respect to its value at some
instant of time. Non-deterministic signals are random in nature hence they are called random signals.
Random signals cannot be described by a mathematical equation. They are modelled in probabilistic
terms.
3. Even and Odd Signals - A signal is said to be even when it satisfies the condition x(t) = x(-t). A
signal is said to be odd when it satisfies the condition x(t) = -x(-t)
4. Periodic and Aperiodic Signals- A signal is said to be periodic if it satisfies the condition x(t) = x(t
+ T) or x(n) = x(n + N). Where T = fundamental time period, 1/T = f = fundamental frequency.
5. Energy and Power Signals- A signal is said to be energy signal when it has finite energy. A signal
is said to be power signal when it has finite power.
T 2
Total energy of a signal g(t) may be defined as E  lim  g (t ) dt
T   T
2
1 T
T  2T  T
Average power of a signal g(t) may be defined as P  lim g ( t ) dt

NOTE: A signal cannot be both (either one) energy and power simultaneously.
a. Power of energy signal = 0
b. Energy of power signal = ∞
[Energy=Ability to work. 8 types of energy. 1) Potential, 2) Kinetic, 3) Gravitational, 4) Chemical,
5) Nuclear, 6) Elastic, 7) Motion and 8) Thermal and temperature
Work= Force x Distance (Joules)
Power= How Fast/Slow doing the work =Work/Time (Watts)]
6. Real and Imaginary Signals- A signal is said to be real when it satisfies the condition x(t) = x*(t).
A signal is said to be odd when it satisfies the condition x(t) = -x*(t)
Note: For a real signal, imaginary part should be zero. Similarly for an imaginary signal, real part
should be zero.
Continuous systems
The type of systems whose input and output both are continuous signals or analog signals are called
continuous systems.
Discrete systems
The type of systems whose input and output both are discrete signals or digital signals are called digital
systems.
Analog versus Digital comparison chart
Analog Digital
Signal Analog signal is a continuous signal which Digital signals are discrete time signals
represents physical measurements. generated by digital modulation.
Waves Denoted by sine waves Denoted by square waves
Representation Uses continuous range of values to Uses discrete or discontinuous values to
represent information represent information
Example Human voice in air, analog electronic Computers, CDs, DVDs, and other digital
devices. electronic devices.
Technology Analog technology records waveforms as Samples analog waveforms into a limited
they are. set of numbers and records them.
Data Subjected to deterioration by noise during Can be noise-immune without
transmissions transmission and write/read cycle. deterioration during transmission and
write/read cycle.
Response to More likely to get affected reducing Less affected since noise response are
Noise accuracy analog in nature
Flexibility Analog hardware is not flexible. Digital hardware is flexible in
implementation.
Uses Can be used in analog devices only. Best Best suited for Computing and digital
suited for audio and video transmission. electronics.
Applications Thermometer PCs, PDAs
Bandwidth Analog signal processing can be done in There is no guarantee that digital signal
real time and consumes less bandwidth. processing can be done in real time and
consumes more bandwidth to carry out the
same information.
Memory Stored in the form of wave signal Stored in the form of binary bit
Power Analog instrument draws large power Digital instrument drawS only negligible
power
Cost Low cost and portable Cost is high and not easily portable
Impedance Low High order of 100 megaohm
Errors Analog instruments usually have a scale Digital instruments are free from
which is cramped at lower end and give observational errors like parallax and
considerable observational errors. approximation errors.

Baseband and Band-pass Communication

Baseband Transmission
The baseband transmission does not use modulator
and demodulator.
Baseband transmission is transmission of the
encoded signal using its own baseband frequencies
i.e. without any shift to higher frequency ranges.
Band-pass Transmission
It is used modulator and demodulator.
Pass-band transmission is the transmission after
shifting the baseband frequencies to some higher
frequency range using modulation.
 If the baseband signal is transmitted directly then it
is known as baseband transmission.
It is prefer at low frequencies.
It is used for short distances.
More noise as signal is original.
E.g. general telephony.
If modulated signal is transmission over the
channel, it is known as band-pass transmission.
It has fixed band of frequencies around carrier
frequency.
It is used for long distances.
Less noise as signals is modulated.
E.g. AM & FM.

Baseband transmission sends the information signal as it is without modulation (without frequency
shifting). Ethernet refers to baseband transmission.
Almost all sources of information generate baseband signals. Baseband signals are those that have
frequencies relatively close to zero such as the human voice (20 Hz – 5 kHz) and the video signal from a TV
camera (0 Hz – 5.5 MHz).
Pass-band (or Band-pass) transmission shifts the signal to be transmitted in frequency to a higher
frequency and then transmits it, where at the receiver the signal is shifted back to its original frequency.
The process of shifting the baseband signal to pass-band range for transmission is known as
MODULATION and the process of shifting the pass-band signal to baseband frequency range at the receiver
is known as DEMODULATION.
Modulation
In electronics and telecommunications, modulation is the process of varying one or more properties of a
periodic waveform, called the carrier signal, with a modulating signal that typically contains information to
be transmitted.
Modulation is defined as the process by which some characteristic of a carrier wave is varied in accordance
with an information-bearing signal.
Modulator is a device that performs modulation
Modulation is a process of mixing a signal with a sinusoid to produce a new signal. This new signal,
conceivably, will have certain benefits over an un-modulated signal. f (t )  A sin(t   ) , this sinusoid has
3 parameters (Amplitude, Phase, and Frequency) that can be altered.
'Mo' in Modem stands for Modulation and 'dem' stands for demodulation. Only the MoDem whose name
itself denotes Modulator-Demodulator
Need for Modulation
Modulation is extremely necessary in communication system because of the following reasons:
1. Avoids mixing of signals
2. Increase the range of communication
3. Wireless communication
4. Reduces the effect of noise & distortions
5. Reduces height of antenna
For the transmission of radio signals, the antenna height must be multiple of λ/4, where λ is the
wavelength. λ = c /f;
Where c: is the velocity of light & f: is the frequency of the signal to be transmitted
6. Reduce band width. Narrow banding the signal
7. Multiplexing is possible. Multiplex more number of signals
8. Improves quality of reception
9. To reduce equipment complexity
Modulation Processes
1. Analog- Continuous-Wave (CW) modulation
 amplitude modulation ( AM )
 frequency modulation ( FM )
  phase modulation ( PM )
2. Analog Pulse modulation
 pulse-amplitude modulation ( PAM )
 pulse-duration modulation ( PDM )
 pulse-position modulation ( PPM )
3. Digital -Pulse modulation
 Pulse code modulation (PCM)
 Differential pulse code modulation (DPCM)
 Delta modulation (DM)
 Adaptive delta modulation (ADM)
4. Digital Multiplexing modulation
 Frequency-division multiplexing (FDM )
 Time-division multiplexing (TDM )
 Code-division multiplexing (CDM )
Analog Modulation Techniques: AM, FM, PM
 Analog modulation refers to the process of transferring an analog baseband (low frequency) signal,
like an audio or TV signal over a higher frequency signal such as a radio frequency band.
1. Amplitude Modulation
 A type of modulation where the amplitude of the carrier signal is modulated (changed) in proportion
to the message signal while the frequency and phase are kept constant.
 Several variants of AM are used as Double Side Band Suppressed Carrier (DSBSC) Modulation,
Single Sideband Suppressed Carrier (SSBSC) Modulation and Vestigial Sideband Amplitude
Modulation (VSBAM).
2. Angle Modulation
 In Angle Modulation, the message signal's amplitude is used to control the frequency or phase of the
carrier signal. This rise to the two methods FM & PM.
 Frequency modulation a type of modulation where the frequency of the carrier signal is modulated
(changed) in proportion to the message signal while the amplitude and phase are kept constant.
 Phase modulation a type of modulation where the phase of the carrier signal is varied accordance to
the low frequency of the message signal is known as phase modulation.

Introduction to Radio Communication

Radio is the technology of using radio waves to carry information by systematic modulating properties of
electromagnetic energy waves transmitted through space, such as their amplitude, frequency, phase, or pulse
width. When radio waves strike an electrical conductor, the oscillating fields induce an alternating current in
the conductor. The information in the waves can be extracted and transformed back into its original form.
Radio systems need a (i) transmitter to modulate, (ii) antenna to convert electric currents into radio waves
and radio waves into an electric current. An antenna can be used for both transmitting and receiving.
The electrical resonance of tuned circuits in radios allow individual frequencies to be selected. The
electromagnetic wave is intercepted by a tuned receiving antenna. A radio receiver receives its input from
an antenna and converts it into a form that is usable for the consumer, such as sound, pictures, digital data,
measurement values, navigational positions, etc. Radio frequencies occupy the range from a 3 kHz to
300 GHz, although commercially important uses of radio use only a small part of this spectrum.
A radio communication system requires a transmitter and a receiver, each having an antenna and
appropriate terminal equipment such as a microphone at the transmitter and a loudspeaker at the receiver in
the case of a voice-communication system.
 Fig. K. Schematic view of radio communication
Fig. K shows the Radio communication. Information such as sound is converted by a transducer such as
a microphone to an electrical signal, which modulates a radio wave sent from a transmitter. A receiver
intercepts the radio wave and extracts the information-bearing electronic signal, which is converted back
using another transducer such as a speaker.
Key-summary
When a high-frequency alternating current (AC) passes through a copper conductor it generates radio waves
which are propagated into the air using an antenna

• radio waves have frequencies between:

– 3 Hz – 300 KHz - low frequency
– 300 KHz – 30 MHz – high frequency
– 30 MHz – 300 MHz – very high frequency
– 300 MHz – 300 GHz – ultra high frequency
• radio waves are generated by an antenna and they propagate in all directions as a straight line
• radio waves travel at a velocity of 186.000 miles per second
• radio waves become weaker as they travel a long distance
Model of a Digital Communication System

Model-1
 Model-3

Elements of a Digital Communication System

In digital communication system, the message signal to be transmitted is digital in nature. This means that
digital communication involves the transmission of information in digital form. The elements of digital
communications system as follows:
1. Discrete information source
2. Source Encoder
3. Channel Encoder
4. Modulator
5. Electrical Communication Channel
6. Noise
7. Demodulator
8. Channel Decoder
9. Source Decoder
10. Destination
The overall purpose of the system is to transmit the message or sequence of symbols coming out of source to
a destination point as a high rate and accuracy as possible. The source and destination point are physically
separated in space and a communication channel connects the source to the destination point. The
communication channel accepts electrical (i.e., electromagnetic) signals and the output of the channel is
usually a smeared of destroyed version of the input due to the non-ideal nature of communication channel.
In addition to this, the information bearing signal is also corrupted by unpredictable electrical signals (i.e.
noise) from both man-made and natural causes. Thus, the smearing and the noise introduce errors in the
information being transmitted and limit the rate at which information can be communicated from the source
to the destination.
Discrete Information Source: Information source may be classified into two categories based upon the
nature of their output i.e. analog Information sources and discrete information sources. In case of analog
communication, the information source is analog. Analog information sources, such as microphone actuated
by speech emit one or more continuous amplitude signals.
In case of digital communication system, the information source produces a message signal which is not
continuously varying with time. Rather the message signal is intermittent with respect to time. The output of
discrete information source such as teletype or the numerical output of the computer consists of a sequence
of discrete symbols of letters. An analog information source may be transformed into a discrete information
sources through the process of sampling and quantizing. Discrete information sources are characterized by
the following parameters:
 Source Alphabet: These are the letters, digits or special characters available from the information
source.
 Symbol Rate: It is the rate at which the information source generates source alphabets. It is
generally represented in symbols/sec unit.
 Source Alphabet Probabilities: Each source alphabet from the source has independent occurrence
rate in the sequence. As an example, letters A, E, I etc. occur frequently in the sequence. Hence,
probability of the occurrence of each source alphabet can become one of the important properties
which is useful in digital communication.
 Probabilistic Dependence of Symbols in a Sequence: The information carrying capacity of each
source alphabet is different in a particular sequence. This parameter defined average information
content of the symbols. The entropy of a source describes the average information content per
symbol in long message. Entropy may be defined in terms of bits per symbol. This means that the
source information rate is the product of symbol rate and source entropy, i.e., Information rate =
Symbol rate * Source entropy (Bits/sec) (Symbols/sec) (Bits/Symbol)
Thus, the information rate represents minimum average data rate required to transmit information from
source to the destination.
Source Encoder and Decoder: In source coding, the encoder maps the digital signal generated at the source
output into another signal in digital form. The mapping is one to one and the objective is to eliminate or
reduce the redundancy so as to provide an efficient representation of the source output. The source decoder
simply performs the inverse mapping and thereby delivers to the user destination, a reproduction of the
digital source output. The advantage of source coding is to reduce the bandwidth of transmission.
Source Encoder
 Sampling
 makes signal discrete in time
 signals can be sampled without introducing distortion
 Quantization
 makes signal discrete in amplitude
 Good quantizers are able to use few bits and introduce small distortion
• Source Coding
 compression of digital data to eliminate redundant information
 does not introduce distortion
Channel Encoder and Decoder: The purpose of channel encoder is to map the incoming digital signal into
a channel input and for the decoder to map the channel output into an output digital signal in such a way that
the effect of channel noise is minimized. That is the combined roll of channel encoder and decoder is to
provide reliable communication. This provision is satisfied by introducing redundancy in a prescribed
fashion. In the channel encoder and exploiting it in the decoder, to reconstruct the original encoder input as
accurately as possible.
Channel Encoder
Encryption- ensures data privacy
Channel coding
 Provides protection against transmission errors by selectively inserting redundant data
 plays an extremely important role in system design
Modulation
 Converts digital data to a continuous waveform suitable for transmission (usually a sinusoidal wave)
  Information is transmitted by varying one or more parameters of the transmitted signal such as
PSK, FSK and ASK

Electromagnetic spectrum

And also need to study:

Chapter-1 Electronic Communication Systems by Kennedy D 6th Edition
Chapter-2 Communication Systems by Haykin, Simon
 Representation of Signals and Systems
(Chapter II) Simon Haykin, Communication Systems

1. Definition of the Fourier transform

2. Properties of the Fourier transform
3. Dirac Delta Function
4. Filters (low-pass, band-pass)
5. Band-pass signal
6. Numerical computation of the Fourier Transform
 Fourier Transform
• Provides the link between the time-domain and
frequency-domain descriptions of a signal
(waveform and its spectrum).
• The Fourier transform of signal g(t) is:

G ( f )   g (t )e  j 2ft dt

Inverse Fourier transform is:

g (t )  G( f )e j 2ft df


For signal g(t) to have F-transform it is sufficient to

have Dirichlet's conditions:
1. it is single-valued,
2. it has a finite number of discontinuities,
3. it is integrable.
 Fourier Transform
• All energy signals are F-transformable

 g (t ) dt  
2

Fourier pair are described by two variables f (Hz) and t (s) or
by one variable  = 2f (angular frequency(rad/s).
Fourier transforms are described as linear operations:
G(f) = F[g(t)]
g(t) = F-1[G(f)]
or as a Fourier pair
g(t)  G(f)
Fourier transform is represented as a complex function.
j ( f )
G( f )  G( f ) e
where |G(f)| is called continuous amplitude spectrum and (f)
continuous phase spectrum.
Amplitude spectrum is an even function. G(-f) = G(f)
Phase spectrum is odd function. (-f) = -(f).
 Fourier Transform- Ex: Rectangular pulse
• Rectangular pulse g(t) of duration T and amplitude A
and its Fourier transform.
 t
g (t ) A  Rect 
T  sin(fT )
G ( f ) A  T sinc  fT   A  T
fT

u Time-domain and Frequency-domain signal

description has inverse relationship. A pulse narrow in
time has wide frequency description and vice versa.
 Properties of Fourier Transform
1. Linearity (Superposition)
2. Time scaling
3. Duality
4. Time shifting
5. Frequency shifting
6. Area under g(t)
7. Area under G(f)
8. Differentiation in the time
domain
9. Integration in the time domain
10. Conjugate functions
11. Multiplication in the time
domain
12. Convolution in the time domain
 Properties of Fourier Transform
1. When a function is shifted in the positive direction by an amount t0 the effect is
equivalent to multiplying its Fourier transform G(f) by factor exp(-j2ft0).
2. This means that the amplitude of G(f) is unaffected but its phase is changed by the
linear factor of -2ft0.
3. Multiplication of function g(t) by the factor of exp(j2fct) is equivalent of shifting
F(f) by the amount of fc (modulation theorem).
t e j 2fct  e  j 2fct
g (t )  Arect( ) cos(2f c t ) cos(2f c t ) 
T 2
AT
G( f )  sinc(T ( f  f c )  sinc(T ( f  f c )
2
4. The area under a function g(t) is equal value of its Fourier transform G(f) at f= 0.
5. The value of a function g(t) at t= 0 is equal to the area under its Fourier transform
G(f).
6. The differentiation of time function g(t) has an effect of multiplying G(f) by j2f.
7. The integration of time function g(t) has an effect of dividing G(f) by j2f assuming
G(0) is zero.
8. The multiplication of two signals in the time domain is transformed into the
convolution of their individual Fourier transforms in the frequency domain.
 Rayleigh’s energy theorem
• The total energy of signal is
 2
E   g (t ) dt


The integrand is an energy intensity that varies with

time.
It can be expressed as product of two time functions
g(t) and g*(t).
Redefining the total energy of signal g(t)
 2  2
E   g (t ) dt   G ( f ) df
 

This is Rayleigh’s energy theorem.

To calculate energy we need only know the
amplitude spectrum |G(f)| of the signal.
 Time and frequency relationship
• They have inverse relationship.
• If description of signal in time domain is changed the frequency
domain description is changed in an inverse manner.
• If signal is limited in frequency (strictly frequency limited or
strictly band limited), the time-domain description of the signal
will trail on infinity.
• If signal is strictly limited in time the spectrum of the signal is
infinite.
• Bandwidth provides a measure of the extent of significant
spectral content of the signal for positive frequencies.
• Signal is low-pass if its significant spectral content is centered
around the origin.
• Signal is band-pass if its significant spectral content is centered
around fc.
 Bandwidth
• With low-pass signals the bandwidth is defined as W/2.

• For band-pass signals the bandwidth is the width of the

main lobe for positive frequencies (null-to-null
bandwith).

• Bandwidth can be expressed as 3-dB bandwidth. It is the

separation between peak value and the positive frequency
at which the amplitude drops below 1/sqrt(2).

• Bandwidth can be expressed also in root mean square

form (rms).

• The nice property of bandwidth is:

(duration)* (bandwidth)= constant
 Dirac delta function
Dirac delta function (t) is an unit impulse.
It has zero amplitude except at t= 0, where it is infinitely
large (contains unit area under its curve).
Useful properties:
1. Combines the Fourier series as Fourier transform into a
unified theory,
2. Includes power signals in the list of signals to which
Fourier transform can be applied.
Shifting property: product of g(t) and time shifted delta
function. 

g (t ) (t  t0 )dt   (t0 )
• Convolution of any function with the delta function leaves
that function unchanged.


g ( ) (t   )dt  g (t )  g (t )   (t   )  g (t )
 Other useful properties
• For DC signal: 1  (f)
• Complex exponential function : exp(j2fct)  (f- fc)

• Sinusoidal function:
cos(2fct)  ½[(f- fc) + (f + fc)]sin(2fct) 
(1/2j)[(f- fc) - (f + fc)]
• Cosine-function consists of a pair of delta functions at
f=±fc each of which is weighted by the factor of 1/2 (look
figure 2.14).
• Signum function: sgn(t)  1/(jf)

• Unit step function: u(t)  0.5*[1/(jf)+ (f )]

 Periodic signal
Periodic signal can be represented as a sum of complex
exponential functions.

g (t )  n
c e
n
j 2nf 0t

where cn is the complex Fourier coefficient.

• For periodic signal there exists Fourier-transform pair:

 

 g (t  mT )  f  G(nf
m
0 0
n
0 ) ( f  nf 0 )

The periodicity in the time domain has the effect of changing the
frequency-domain description of spectrum of the signal into a
discrete form defined at integer multiples of the fundamental
frequency (look figure 2.18).
 System
• System refers to any physical device that produces an output signal
in response to an input signal.
• Filters and communication channels are linear systems.
• Filter is a device which limits the spectrum of signal to some band
of frequencies.
• Channel is a transmission medium that connects the transmitter and
receiver to communication system.
• In time domain linear system is described in terms of its impulse
response, which is the response of the system to a unit impulse or
delta function (t) applied to the input of the system.
• The system is time invariant if the shape of impulse response is the
same no matter when the unit impulse is applied to system.
• To find total response y(t) at some time t we use convolution

integral:
y (t )   x( )h(t   )d

 Linear systems
• System has three times associated with it:
t = response time , = excitation time and t-= system-
memory time
• Using commutative

property:
y (t )   h( ) x(t   )d  Y ( f )  H ( f ) X ( f )

Linear time-invariant system can be expressed as tapped-delay-
line filter:
N 1
y (nt )   wk x(n  k )
k 0
 Causality and stability
• System is causal if it does not respond before the excitation
is applied (h(t) = 0, t < 0). The system does not anticipate.

• System is stable if the output signal is bounded for all

bounded input signals (BIBO).
• The sufficient condition for BIBO


 h( ) dt  

Transfer function H(f) is a complex function, so it can be
expressed as
H ( f )  H ( f ) exp( j ( f )
where first term is amplitude response (gain G or attenuation
G-1) and and latter one is phase response.
 Filter
• Filter is a frequency-selective device.
• It is used to limit the spectrum of a signal to some band of
frequencies.
• Its frequency response is Frequency response is
characterized by pass-band and stop-band.
• The frequencies inside pass-band are transmitted with or
no distortion.
• The frequenices inside stop-band are rejected.
• Between these two bands is transition-band.
• Filter types are:
– low-pass,
– high-pass,
– band-pass, exp( j 2ft0 ), B  f  B
H( f )  
– band-stop.  0
• Ideal low-pass filter can be expressed (B is the bandwidth)
 Filter
• Inverse Fourier transform of transfer function is (Note!
bounded by bandwidth)
B
h(t )   exp( j 2f (t  t0 ))df
B

Let’s solve the integral

B 1
h(t )   exp( j 2f (t  t0 ))
 B j 2 (t  t )
0


1
j 2 (t  t0 )
e j 2B ( t t0 )  e  j 2B ( t t0 ) 

1
 2 j sin( 2B (t  t0 ))
j 2 (t  t0 )
sin( 2B (t  t0 ))
  2 B sinc( 2 B (t  t0 ))
 (t  t0 )

From figure 2.23 we can see that the maximum value 2B is

reached at location t0.
 Design of filters
• With low-pass filter there exists oscillation which can be
measured as overshoot percentage.
• It is independent of the filter bandwidth.
• A filter is characterized by specifying its impulse response
h(t) or its transfer function H(f).
• Design is typically carried out in frequency domain.
• It takes two steps:
1 Approximation of the prescribed frequency response,
2 Realization of transfer function by physical device.
• Transfer function is typically described as rational function
( s  z0 )( s  z1 )( s  z2 )( s  z3 )
H ' ( s)  K
( s  p0 )( s  p1 )( s  p2 )( s  p3 )
where zi are zeroes and pi are poles.
For low-pass and band-pass filters, the number of zeroes is
less than number of poles.
 Design of filters
• The system is said to be causal if all the poles are inside the left half
of the s-plane.
• Minimum-phase system is where all zeroes and poles lie inside the
left half of the s-plane.
• Non-minimum-phase system can have zeroes and poles on the
imaginary axis as well as the right half of the s-plane.

• Two families of filters:

– Butterworth (the poles lie on a circle with origin and 2B as its
radius),
– Chebyshev (the poles lie on an ellipse with origin).
• Filter can be realized as:
– Analog filter,
– Discrete-time filter (sampled in time but their amplitude is
continuous),
– Digital filter (sampled in time and quantized by amplitude).
 Band-pass signals
• A signal g(t) is a band-pass signal if its Fourier transform G(f) is non-
negligible only in a band of frequencies of the total extent 2W
centered about frequency fc which is referred as carrier frequency. If
fc is large compared with 2W the signal is referred as narrow-band
signal.
• Band-pass signal can be defined as:
g (t )  Reg~(t ) exp( j 2f ct )
g~(t )  g I (t )  jgQ (t )
g (t )  g I (t ) cos(2f ct )  gQ (t ) sin(2f ct )
u Signal is a sum of in-phase and the quadrature component and it
can be described as
 Phase and group delay
• Whenever a signal is transmitted through a
dispersive device such as a filter or communication
channel some delay is introduced.

• Phase delay of channel is the time taken by the

received signal phasor to sweep out phase lag.

• Envelope or group delay occurs when modulated

signal is transmitted through a communication
channel. There is a delay between the envelope of
the input signal and that of the received signal.
 Some problems
Drill Problem 2.2 Determine the inverse Fourier transform of the frequency function
defined by the amplitude and phase spectra shown in Fig.

Drill Problem 2.7 Prove the following properties of the convolution process:
(a) The commutative property
(b) The associative property
(c) The distributive property
Drill Problem 2.9: Determine the Fourier transform of the squared sinusoidal signals:
Drill Problem 2.12 Discuss the following two issues, citing examples for your
answers:
(a) Is it possible for a linear time-invariant system to be causal but unstable?
(b) Is it possible for such a system to be non-causal but stable?
Drill Problem 2.14 A tapped-delay-line filter consists of N weights, where N is odd. It
is symmetric with respect to the center tap; that is, the weights satisfy the condition
(a) Find the amplitude response of the filter.
(b) Show that this filter has a linear phase response. What is the implication of this
property? 22
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40 Chapter 2 ∙ Signal and Linear System Analysis

2.4.5 Transform Theorems: Proofs and Applications

Several useful theorems10 involving Fourier transforms can be proved. These are useful for
deriving Fourier-transform pairs as well as deducing general frequency-domain relationships.
The notation 𝑥(𝑡) ⟷ 𝑋(𝑓 ) will be used to denote a Fourier-transform pair.
Each theorem will be stated along with a proof in most cases. Several examples giving
applications will be given after the statements of all the theorems. In the statements of the
theorems 𝑥(𝑡), 𝑥1 (𝑡), and 𝑥2 (𝑡) denote signals with 𝑋(𝑓 ), 𝑋1 (𝑓 ), and 𝑋2 (𝑓 ) denoting their
respective Fourier transforms. Constants are denoted by a, 𝑎1 , 𝑎2 , 𝑡0 , and 𝑓0 .

Superposition Theorem

𝑎1 𝑥1 (𝑡) + 𝑎2 𝑥2 (𝑡) ⟷ 𝑎1 𝑋1 (𝑓 ) + 𝑎2 𝑋2 (𝑓 ) (2.83)

Proof: By the defining integral for the Fourier transform,

∞[ ]
ℑ{𝑎1 𝑥1 (𝑡) + 𝑎2 𝑥2 (𝑡)} = 𝑎1 𝑥1 (𝑡) + 𝑎2 𝑥2 (𝑡) 𝑒−𝑗2𝜋𝑓𝑡 𝑑𝑡
∫−∞
∞ ∞
= 𝑎1 𝑥1 (𝑡)𝑒−𝑗2𝜋𝑓𝑡 𝑑𝑡 + 𝑎2 𝑥2 (𝑡) 𝑒−𝑗2𝜋𝑓𝑡 𝑑𝑡
∫−∞ ∫−∞
= 𝑎1 𝑋1 (𝑓 ) + 𝑎2 𝑋2 (𝑓 ) (2.84)

Time-Delay Theorem
( )
𝑥 𝑡 − 𝑡0 ⟷ 𝑋 (𝑓 ) 𝑒−𝑗2𝜋𝑓𝑡0 (2.85)

Proof: Using the defining integral for the Fourier transform, we have
∞
ℑ{𝑥(𝑡 − 𝑡0 )} = 𝑥(𝑡 − 𝑡0 )𝑒−𝑗2𝜋𝑓 𝑡 𝑑𝑡
∫−∞
∞
= 𝑥(𝜆)𝑒−𝑗2𝜋𝑓 (𝜆+𝑡0 ) 𝑑𝜆
∫−∞
∞
= 𝑒−𝑗2𝜋𝑓𝑡0 𝑥 (𝜆) 𝑒−𝑗2𝜋𝑓 𝜆 𝑑𝜆
∫−∞
= 𝑋 (𝑓 ) 𝑒−𝑗2𝜋𝑓𝑡0 (2.86)

where the substitution 𝜆 = 𝑡 − 𝑡0 was used in the first integral.

Scale-Change Theorem
( )
1 𝑓
𝑥 (𝑎𝑡) ⟷ 𝑋 (2.87)
|𝑎| 𝑎

10 See Tables F.5 and F.6 in Appendix F for a listing of Fourier-transform pairs and theorems.

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 2.4 The Fourier Transform 41

Proof: First, assume that 𝑎 > 0. Then

∞
ℑ{𝑥(𝑎𝑡)} = 𝑥(𝑎𝑡)𝑒−𝑗2𝜋𝑓𝑡 𝑑𝑡
∫−∞
∞ ( )
𝑑𝜆 1 𝑓
= 𝑥(𝜆)𝑒−𝑗2𝜋𝑓 𝜆∕𝑎 = 𝑋 (2.88)
∫−∞ 𝑎 𝑎 𝑎

where the substitution 𝜆 = 𝑎𝑡 has been used. Next considering 𝑎 < 0, we write
∞ ∞
𝑑𝜆
ℑ{𝑥(𝑎𝑡)} = 𝑥 (− |𝑎| 𝑡) 𝑒−𝑗2𝜋𝑓𝑡 𝑑𝑡 = 𝑥(𝜆)𝑒+𝑗2𝜋𝑓 𝜆∕|𝑎|
∫−∞ ∫−∞ |𝑎|
( ) ( )
1 𝑓 1 𝑓
= 𝑋 − = 𝑋 (2.89)
|𝑎| |𝑎| |𝑎| 𝑎

where use has been made of the relation − |𝑎| = 𝑎 if 𝑎 < 0.

Duality Theorem

𝑋(𝑡) ⟷ 𝑥(−𝑓 ) (2.90)

That is, if the Fourier transform of 𝑥(𝑡) is 𝑋(𝑓 ), then the Fourier transform of 𝑋(𝑓 ) with
𝑓 replaced by 𝑡 is the original time-domain signal with 𝑡 replaced by −𝑓 .

Proof: The proof of this theorem follows by virtue of the fact that the only difference
between the Fourier-transform integral and the inverse Fourier-transform integral is a minus
sign in the exponent of the integrand.

Frequency-Translation Theorem
( )
𝑥(𝑡)𝑒𝑗2𝜋𝑓0 𝑡 ⟷ 𝑋 𝑓 − 𝑓0 (2.91)

Proof: To prove the frequency-translation theorem, note that

∞ ∞
𝑥(𝑡)𝑒𝑗2𝜋𝑓0 𝑡 𝑒−𝑗2𝜋𝑓𝑡 𝑑𝑡 = 𝑥(𝑡)𝑒−𝑗2𝜋(𝑓 −𝑓0 )𝑡 𝑑𝑡 = 𝑋(𝑓 − 𝑓0 ) (2.92)
∫−∞ ∫−∞

Modulation Theorem
1 1
𝑥(𝑡) cos(2𝜋𝑓0 𝑡) ⟷ 𝑋(𝑓 − 𝑓0 ) + 𝑋(𝑓 + 𝑓0 ) (2.93)
2 2

Proof: The proof of this theorem follows by writing cos(2𝜋𝑓0 𝑡) in exponential form as
1 ( )
2
𝑒𝑗2𝜋𝑓0 𝑡 + 𝑒−𝑗2𝜋𝑓0 𝑡 and applying the superposition and frequency-translation theorems.

Differentiation Theorem
𝑑 𝑛 𝑥 (𝑡)
⟷ (𝑗2𝜋𝑓 )𝑛 𝑋 (𝑓 ) (2.94)
𝑑𝑡𝑛

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42 Chapter 2 ∙ Signal and Linear System Analysis

Proof: We prove the theorem for 𝑛 = 1 by using integration by parts on the defining
Fourier-transform integral as follows:
{ } ∞
𝑑𝑥 (𝑡) −𝑗2𝜋𝑓𝑡
𝑑𝑥
ℑ = 𝑒 𝑑𝑡
𝑑𝑡 ∫−∞ 𝑑𝑡
∞
|∞
= 𝑥(𝑡)𝑒−𝑗2𝜋𝑓𝑡 | + 𝑗2𝜋𝑓 𝑥(𝑡)𝑒−𝑗2𝜋𝑓𝑡 𝑑𝑡
|−∞ ∫−∞
= 𝑗2𝜋𝑓 𝑋 (𝑓 ) (2.95)
where 𝑢 = 𝑒−𝑗2𝜋𝑓𝑡
and 𝑑𝑣 = (𝑑𝑥∕𝑑𝑡)𝑑𝑡 have been used in the integration-by-parts formula,
and the first term of the middle equation vanishes at each end point by virtue of 𝑥(𝑡) being an
energy signal. The proof for values of 𝑛 > 1 follows by induction.

Integration Theorem
𝑡
1
𝑥(𝜆) 𝑑𝜆 ⟷ (𝑗2𝜋𝑓 )−1 𝑋(𝑓 ) + 𝑋(0)𝛿(𝑓 ) (2.96)
∫−∞ 2

Proof: If 𝑋(0) = 0, the proof of the integration theorem can be carried out by using
integration by parts as in the case of the differentiation theorem. We obtain
{ 𝑡 }
ℑ 𝑥 (𝜆) 𝑑 (𝜆)
∫−∞
{ 𝑡 }( )∞
1 −𝑗2𝜋𝑓𝑡 ||
∞
1
= 𝑥 (𝜆) 𝑑 (𝜆) − 𝑒 | + 𝑥 (𝑡) 𝑒−𝑗2𝜋𝑓𝑡 𝑑𝑡 (2.97)
∫−∞ 𝑗2𝜋𝑓 | 𝑗2𝜋𝑓 ∫−∞
|−∞
∞
The first term vanishes if 𝑋(0) = ∫−∞ 𝑥(𝑡)𝑑𝑡 = 0, and the second term is just 𝑋(𝑓 )∕ (𝑗2𝜋𝑓 ).
For 𝑋(0) ≠ 0, a limiting argument must be used to account for the Fourier transform of the
nonzero average value of 𝑥(𝑡).

Convolution Theorem
∞
𝑥1 (𝜆)𝑥2 (𝑡 − 𝜆) 𝑑𝜆
∫−∞
∞
≜ 𝑥1 (𝑡 − 𝜆)𝑥2 (𝜆)𝑑𝜆 ↔ 𝑋1 (𝑓 )𝑋2 (𝑓 ) (2.98)
∫−∞

Proof: To prove the convolution theorem of Fourier transforms, we represent 𝑥2 (𝑡 − 𝜆)

in terms of the inverse Fourier-transform integral as
∞
𝑥2 (𝑡 − 𝜆) = 𝑋2 (𝑓 )𝑒𝑗2𝜋𝑓 (𝑡−𝜆) 𝑑𝑓 (2.99)
∫−∞
Denoting the convolution operation as 𝑥1 (𝑡) ∗ 𝑥2 (𝑡), we have
∞ [ ∞ ]
𝑗2𝜋𝑓 (𝑡−𝜆)
𝑥1 (𝑡) ∗ 𝑥2 (𝑡) = 𝑥 (𝜆) 𝑋 (𝑓 )𝑒 𝑑𝑓 𝑑𝜆
∫−∞ 1 ∫−∞ 2
∞ [ ∞ ]
−𝑗2𝜋𝑓 𝜆
= 𝑋 (𝑓 ) 𝑥 (𝜆)𝑒 𝑑𝜆 𝑒𝑗2𝜋𝑓𝑡 𝑑𝑓 (2.100)
∫−∞ 2 ∫−∞ 1

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 2.4 The Fourier Transform 43

where the last step results from reversing the orders of integration. The bracketed term inside
the integral is 𝑋1 (𝑓 ), the Fourier transform of 𝑥1 (𝑡). Thus,
∞
𝑥1 ∗ 𝑥2 = 𝑋1 (𝑓 )𝑋2 (𝑓 )𝑒𝑗2𝜋𝑓𝑡 𝑑𝑓 (2.101)
∫−∞

which is the inverse Fourier transform of 𝑋1 (𝑓 )𝑋2 (𝑓 ). Taking the Fourier transform of this
result yields the desired transform pair.

Multiplication Theorem
∞
𝑥1 (𝑡)𝑥2 (𝑡) ⟷ 𝑋1 (𝑓 ) ∗ 𝑋2 (𝑓 ) = 𝑋1 (𝜆)𝑋2 (𝑓 − 𝜆) 𝑑𝜆 (2.102)
∫−∞

Proof: The proof of the multiplication theorem proceeds in a manner analogous to the
proof of the convolution theorem.

EXAMPLE 2.11
Use the duality theorem to show that
( )
𝑓
2AW sinc (2𝑊 𝑡) ⟷ 𝐴Π (2.103)
2𝑊

Solution
From Example 2.8, we know that
( )
𝑡
𝑥(𝑡) = 𝐴Π ⟷ 𝐴𝜏 sinc 𝑓 𝜏 = 𝑋(𝑓 ) (2.104)
𝜏

Considering 𝑋(𝑡), and using the duality theorem, we obtain

( )
𝑓
𝑋(𝑡) = 𝐴𝜏 sinc (𝜏𝑡) ⟷ 𝐴Π − = 𝑥 (−𝑓 ) (2.105)
𝜏

where 𝜏 is a parameter with dimension (s)−1 , which may be somewhat confusing at first sight! By letting
𝜏 = 2𝑊 and noting that Π (𝑢) is even, the given relationship follows.
■

EXAMPLE 2.12
Obtain the following Fourier-transform pairs:

1. 𝐴𝛿(𝑡) ⟷ 𝐴
2. 𝐴𝛿(𝑡 − 𝑡0 ) ⟷ 𝐴𝑒−𝑗2𝜋𝑓𝑡0
3. 𝐴 ⟷ 𝐴𝛿(𝑓 )
4. 𝐴𝑒𝑗2𝜋𝑓0 𝑡 𝑡 ⟷ 𝐴𝛿(𝑓 − 𝑓0 )

www.it-ebooks.info
Modulation: It is the process of superimposing the information content of a base band modulating signal by
altering the characteristics of a high frequency carrier wave {Amplitude, frequency and phase}.

c  t   Ac cos ct 

Ac  Amplitude
c  frequency
c t  phase

- The modulation process is a frequency translation process that translates a low frequency base band signal into
a high frequency band pass signal.

- Since a linear device cannot provide frequency translation. So, the modulation process can only be generated
by a non-linear device.

# Need for Modulation:

1. Avoids mixing of signals: The modulation process translates different base band signal at different
carrier frequencies. So, that the spectrum overlapping does not takes place and the mixing of signals can
be avoided.

2. Allows multiplexing of signals: Multiplexing means transmission of two or more signals

simultaneously over same channel. Modulation process translates different base band signals at different
carrier frequencies and now they can be transmitted simultaneously over same channel without any loss
of information.

3. Reduces height of antenna: The height of antenna required for transmission and reception of radio waves
in a radio transmission is a function of frequency used. The minimum height of antenna is given as
 
 .
4

4. Increases range of communication: At low frequencies the radiation is poor and the signal gets highly
attenuated therefore base band signals cannot be transmitted directly over longer distance. Modulation
 effectively increases the frequency of the signal to be radiated and thus increases the distance over
which signals can be transmitted faithfully.

5. Improves Quality of reception: The signal communication using modulation techniques such as FM and
PCM reduces the effect of noise to great extent. Reduction in noise improves quality of reception.

# Amplitude Modulation [AM]:

- In AM, the amplitude of high frequency carrier signal is varied in accordance with instantaneous value
of base band modulating signal keeping frequency and phase constant.
- In AM, a low frequency baseband signal is translated into a high frequency narrow band signal.

1. Simple Amplitude Modulation [AM] or Double side band with carrier [DSB-C, DSB-FC]:

Let

m  t   Arbitrary modulating signal with maximum frequency m

c  t   Ac cos c t   carrier signal
x AM  t    Ac  m  t   cos c t 
x AM  t   Ac cos c t   m  t  cos c t 
-Frequency domain representation:

1
X AM    Ac    c      c     M   c   M   c  
2
Bandwidth  BW   c  m   c  m   2m
BW  2m rad/sec
BW  2 f m Hz
Notes:-

1. In AM, three components are transmitted i.e. the carrier signals, upper side band [USB] and lower side
band [LSB].
2. The information is contains in simultaneously by upper as well as lower side band. The carrier does not
contain any information.
3. The information bandwidth of AM signal is 2m in rad/sec and 2 f m in Hz.
4. Since, the bandwidth 2m is very small as compared to higher cutoff frequency c  m  . So, Am
signal is a high frequency narrow band signal.

- Sinusoidal AM:

m  t   Am cos mt 
c  t   Ac cos c t 
x AM  t    Ac  Am cos mt   cos c t 
 A 
= Ac 1  m cos mt   cos c t 
 Ac 
A
Where, modulation index  m   m
Ac
Then,
x AM  t  = Ac 1  m cos mt   cos c t 
= Ac cos c t   mAc cos m t  cos c t 

cos c  m  t  cos c  m  t

mAc mAc
x AM  t   Ac cos c t   ...... 1
2 2
x AM  t   carrier + upper side band + lower side band
Taking fourier transform of eq 1
X AM    Ac    c      c  
mAc
    c  m       c  m   
2  
mAc
    c  m       c  m   
2  
-Power relation in AM:

Ac 2 m 2 Ac 2 m 2 Ac 2
Pt   
2 8 8
Ac 2  m 2   m2 
= 1    Pc 1  
2  2   2 
 m2 
Pt  Pc 1  
 2 
Where, Pt  Total power, Pc  Carrier power

-Current relation in AM:

 m2 
Pt  Pc 1  
 2 
 m2 
I R  I R 1 
t
2 2
c 
 2 
 m2 
I  I 1 
t
2 2
c 
 2 

It  m2 
 1  
Ic  2 
Where, I t = Current across antenna or transmitter when modulated output is transmitted,
I t = Current across antenna or transmitter when unmodulated carrier is transmitted,
R=Resistance of antenna or transmitter

-Transmission efficiency in AM   :

It is the ratio of use full power to the total transmitted power.

 Pc m 2
 2  100%
 m2 
Pc 1 
 2 
 m2 
  2 
 100%
 2m 

-Time domain representation of AM wave:

A  Ac  Am cos mt 
Amax  Ac  Am
Amin  Ac  Am
Amax  Amin
Am 
2
A  Amin
Ac  max
2
A A  Amin
m  m  max
Ac Amax  Amin

-concepts of modulation index:

- Modulation index gives the depth of modulation to which modulation has occurred.

- For m=0, it represents no modulation

- for m=1, it represents maximum modulation has occurred.

-for m>1, overlapping of envelope take place that results in distortion of envelope. This condition is known as
over modulation and should always be avoided.
2. Double side band - suppressed carrier [DSB-SC]:

The transmission of full AM signal is not advisable because:-

- Since, the carrier is also transmitted which does not contain any information.
- For m=1, only 2/3 rd of total transmitted power appears in the carrier which is a complete wastage.

So, instant of transmitting full AM signal, the carrier is suppressed before transmission and such type of
modulation is known as Double Side Band Suppressed Carrier modulation [DSB-SC].

c  t   Ac cos ct 
S DSB  SC  t   y  t   Ac m  t  cos ct 
Ac
Y     M   c   M   c  
2 

BW  2m rad/sec
=2f m Hz
-Ring Modulator:

- During +ve half cycle of carrier, diode D1 & D2 are forward biased and D3 & D4 are reverse biased.
- During -ve half cycle of carrier, diode D1 & D2 are reverse biased and D3 & D4 are forward biased.
- In any case the output is always equal to zero because the current flowing in opposite direction. So,
output is always zero until message signal is applied.

Message input Carrier input Out put

+ve +ve +ve, D1 & D2 is ON
+ve -ve -ve, D3 & D4 is ON
-ve +ve -ve, D1 & D2 is ON
-ve -ve +ve, D3 & D4 is ON

-percentage of power saving in DSB-SC Modulation:

total power saved

percentage of power saving =  100%
total transmitted power
Pc 2
%age of power saving =  100%
 m 
Pc 1 
2
 
2  m2
 2 
3. Single side band - suppressed carrier [SSB-SC]:

- In case of DSB-SC modulation, the two side bands are transmitted which contains the same information. So,
DSB-SC transmission is further redundant. Instead of transmitting both the side band only one side band can be
transmitted to achieve maximum efficiency. Such type of modulation in which only one side band is transmitted
is known as single side band – suppressed carrier [SSB-SC].

There are two methods for generation of SSB-SC signals.

A. Frequency Discrimination method: - It is also known as filter method. In this method band pass filter
is used to generate SSB-SC signal from DSB-SC signal.
 Disadvantage- In case of frequency discrimination method, the band-pass filter should be as ideal as
possible.
Ideal filters are not practically possible. So, this method is suitable for the transmission of such signal
where upper and lower side band do not meet at carrier frequency such as voice signals.
B. Phase discrimination Method:
Hilbert Transform: It is a method of separating the signals with respect to their phase content and not
with respect to their frequency contents.
The easiest phase generation is 180 degree. Hilbert transformer is a special transformer that seperates the
signals with a phase shift of 90 degree.
The Hilbert transform of any arbitrary signal is given by

x  
ˆx  t    d

 t  
1
xˆ  t   x  t  
t
Xˆ    X      j sgn   

1
y1  t   m  t  cos ct  
FT
 M   c   M   c  
2
1 ˆ
y2  t   mˆ  t  sin ct  
FT
 M   c   Mˆ   c  
2j  
 Disadvantage- The shift method is based upon exact 0 degree phase shift between m  t  and m̂  t  which
is not practically possible at high modulating frequencies. So, this method can only be used for
modulating frequencies up to few KHz.

- Power saving in SSB-SC modulation:

total power saving

% power saving =
total transmitted power
 m2 
Pc 1  
 4  4  m2
% power saving =  100%
 m 2  4  2m 2
Pc 1  
 2 

# Demodulation of AM Waves:- The recovery of baseband signal from the modulated signal is known as
demodulation or detection of waves. There are two methods of detection of AM waves.

1. Synchronous Detection:- In this method, a local carrier is generated at the receiving end whose phase is
exactly synchronized with the transmitted carrier phase. The received signal is multiplied with this locally
generated carrier and the product is passed through a low pass filter to detect the original base band signal.

A. Detection of DSB-C signal:-

x AM  t    Ac  m  t   cos ct 
y1  t    Ac  m  t   cos 2 ct 
 1  cos  2c t  
y1  t    Ac  m  t    
 2 
A m t 
y t   c 
2 2
It passes through capacitor which blocks DC signals.
m t 
y t  
2
B. Detection of DSB-SC signal:-
x AM  t   Ac m  t  cos ct 
y1  t   Ac m  t  cos 2 ct 
 1  cos  2c t  
y1  t   Ac m  t   
 2 
After LPF passing through we get
Ac m  t 
y t  
2
C. Detection of SSB-SC signal:-
x AM  t   m  t  cos ct   mˆ  t  sin ct 
y1  t   m  t  cos 2 ct   mˆ  t  cos ct  sin ct 
 1  cos  2c t   1
y1  t   m  t     mˆ  t  sin  2ct 
 2  2
After LPF passing through we get
m t 
y t  
2
2. Asynchronous Detection:- It is also known as envelope detector or diode detector.
1 1
RC
fc fm
To remove diagonal clipping we are
1 1  m2
RC
m m
Superheterodyne Receiver Block Diagram
The block diagram for the superheterodyne receiver shows its operation and the signal flow.

Having looked at the concepts it is helpful to look at a superheterodyne receiver block diagram of a basic
superhet. In this way it is possible to see the overall operation of the receiver.

There are several different circuit blocks that make up the overall receiver, each one has its own function.

Whilst the superheterodyne receiver block diagram below is the most basic format, it serves to illustrate the
operation. More complicated receivers with more complicated block diagrams are often seen as these radios are
able to offer better performance and more facilities.

Superheterodyne receiver circuit blocks

There are some key circuit blocks that form the basic superheterodyne receiver. Although more complicated
receivers can be made, the basic circuit is widely used – further blocks can add improved performance or
additional functionality and their operation within the whole receiver is normally easy to determine once the
basic block diagram is understood.

 RF tuning & amplification: This RF stage within the overall block diagram for the receiver provides initial
tuning to remove the image signal. It also provides some amplification. If noise performance for the receiver is
important, then this stage will be designed for optimum noise performance. This RF amplifier circuit block will
also increase the signal level so that the noise introduced by later stages is at a lower level in comparison to the
wanted signal.
  Local oscillator: The local oscillator circuit block can take a variety of forms. Early receivers used
free running local oscillators. Today most receivers use frequency synthesizers, normally based around
phase locked loops. These provide much greater levels of stability and enable frequencies to be
programmed in a variety of ways.
  Mixer: Both the local oscillator and incoming signal enter this block within the superheterodyne
receiver. The wanted signal is converted to the intermediate frequency.
 IF amplifier & filter: This superheterodyne receiver block provides the majority of gain and selectivity. High
performance filters like crystal filters may be used, although LC or ceramic filters may be used within domestic
radios.

 Demodulator: The superheterodyne receiver block diagram only shows one demodulator, but in reality
radios may have one or more demodulators dependent upon the type of signals being receiver.
   Automatic Gain Control, AGC: An automatic gain control is incorporated into most superhet radios.
Its fubnction is to reduce the gain for strong signals so that the audio level is maintained for amplitude
sensitive forms of modulation, and also to prevent overloading

 Audio amplifier: Once demodulated, the recovered audio is applied to an audio amplifier block to be amplified
to the required level for loudspeakers or headphones. Alternatively the recovered modulation may be used for
other applications whereupon it is processed in the required way by a specific circuit block.

Superheterodyne receiver block diagram explanation

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

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

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

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