COMMUNICATION SYSTEMS

20A04402T

LECTURE NOTES

B.Tech – ECE – II-II Semester

Prepared by,
T. CHAKRAPANI
Associate Professor
ECE Department

Electronics and Communication Engineering

St.Johns College of Engineering & Technology
Yerrakota, Yemmiganur-518360, Kurnool(D), A.P.
 R20 Regulations
JAWAHARLAL NEHRU TECHNOLOGICAL UNIVERSITY ANANTAPUR
(Established by Govt. of A.P., ACT No.30 of 2008)
ANANTHAPURAMU – 515 002 (A.P) INDIA

Electronics & Communication Engineering

Course Code COMMUNICATION SYSTEMS L T P C

20A04402T 3 0 0 3
Pre-requisite Signals & Systems Semester IV

Course Objectives:
• To introduce various modulation and demodulation techniques of analog and digital
communication systems.
• To analyze different parameters of analog and digital communication techniques.
• To Know Noise Figure in AM & FM receiver systems.
• To understand Function of various stages of AM, FM transmitters and Know Characteristics of
AM &FMreceivers.
• To analyze the performance of various digital modulation techniques in the presence of AWGN.
• To evaluate the performance of each modulation scheme to know the merits and demerits
interms of bandwidth and power efficiency

Course Outcomes (CO):

CO1: Recognize/List the basic terminology used in analog and digital communication techniques for
transmission of information/data.
CO2: Explain/Discuss the basic operation of different analog and digital communication systems at
baseband and passband level.
CO3: Compute various parameters of baseband and passband transmission schemes by applying basic
engineering knowledge.
CO4: Analyze/Investigate the performance of different modulation & demodulation techniques to
solve complex problems in the presence of noise.
CO5: Evaluate/Assess the performance of all analog and digital modulation techniques to know the
merits and demerits of each one of them in terms of bandwidth and power efficiency.

UNIT - I Continuous Wave Modulation 15 Hrs

Introduction: The communication Process, Communication Channels, Baseband and Passband Signals,
Analog vs Digital Communications, Need for the modulation.
Amplitude Modulation(AM): AM and its modifications – DSB, SSB,VSB. Frequency Translation,
Frequency Division Multiplexing (FDM).
Angle Modulation:Frequency Modulation(FM), Phase Modulation, PLL, Nonlinear Effects in FM,
Superheterodyne Receivers.

UNIT - II Noise and Pulse Modulation 12 Hrs

Introduction to Noise: Types of Noise, Receiver Model,Noise in AM, DSB, SSB, and FM Receivers,
Pre-Emphasis and De-emphasis in FM.
Introduction to Pulse Modulation: The Sampling Process, PAM, TDM, Bandwidth-Noise Trade off,
Quantization process, PCM, Noise considerations in PCM systems, Delta Modulation, DPCM, Coding
speech at low bit rates.

UNIT - III Baseband Pulse Transmission 10 Hrs

Introduction, Matched Filter, Properties of Matched Filter, Error rate due to noise, Inter Symbol
Interference (ISI), Nyquist Criterion for distortion less baseband binary transmission, Correlative level
coding, Baseband M-ary PAM transmission, QAM, MAP and ML decoding, Equalization, Eye pattern.

UNKT - IV Digital Passband Transmission 8 Hrs

Introduction, Passband Transmission Model, Gram-Schmidt Orthogonalization Procedure, Geometric
Interpretation of Signals, Response of bank of correlators in noise, Correlation receiver, Probability of
Error, Detection of Signals with unknown phase.
 R20 Regulations
JAWAHARLAL NEHRU TECHNOLOGICAL UNIVERSITY ANANTAPUR
(Established by Govt. of A.P., ACT No.30 of 2008)
ANANTHAPURAMU – 515 002 (A.P) INDIA

Electronics & Communication Engineering

UNIT - V Digital Modulation Schemes & Information Theory 12 Hrs

Coherent Digital Modulation Schemes – ASK, BPSK, BFSK, QPSK, Non-coherent BFSK, DPSK. M-
ary Modulation Techniques, Power Spectra, Bandwidth Efficiency, Timing and Frequency
synchronization.
Information theory: Entropy, Mutual Information and Channel capacity theorem.
Textbooks:
1. Simon Haykin, “Communication Systems”, JohnWiley& Sons, 4th Edition, 2004.
2. B. P. Lathi, Zhi Ding “ Modern Digital and Analog Communication Systems”, Oxford press, 2011.

References:
1.Sam Shanmugam, “Digital and Analog Communication Systems”,JohnWiley& Sons, 1999.
2. Bernard Sklar, F. J. harris“Digial Communications: Fundamentals andApplications”, Pearson
Publications, 2020.
3. Taub and Schilling, “ Principles of Communication Systems”, Tata McGraw Hill, 2007.
 UNIT-I (a)

AMPLITUDE MODULATION
Introduction to Communication System

Communication is the process by which information is exchanged between individuals

through a medium.

Communication can also be defined as the transfer of information from one point in
spaceand time to another point.

The basic block diagram of a communication system is as follows.

 Transmitter: Couples the message into the channel using high frequency signals.
 Channel: The medium used for transmission of signals
 Modulation: It is the process of shifting the frequency spectrum of a signal to a
frequency range in which more efficient transmission can be achieved.
 Receiver: Restores the signal to its original form.
 Demodulation: It is the process of shifting the frequency spectrum back to the
original baseband frequency range and reconstructing the original form.

Modulation:

Modulation is a process that causes a shift in the range of frequencies in a signal.

• Signals that occupy the same range of frequencies can be separated.

• Modulation helps in noise immunity, attenuation - depends on the physical medium.

The below figure shows the different kinds of analog modulation schemes that are available
Modulation is operation performed at the transmitter to achieve efficient and reliable
information transmission.

For analog modulation, it is frequency translation method caused by changing the appropriate
quantity in a carrier signal.

It involves two waveforms:

 A modulating signal/baseband signal – represents the message.

 A carrier signal – depends on type of modulation.

• Once this information is received, the low frequency information must be removed from the
high frequency carrier. •This process is known as “Demodulation”.

Need for Modulation:

 Baseband signals are incompatible for direct transmission over the medium so,
modulation is used to convey (baseband) signals from one place to another.
 Allows frequency translation:
o Frequency Multiplexing
o Reduce the antenna height
o Avoids mixing of signals
o Narrowbanding
 Efficient transmission
 Reduced noise and interference

Types of Modulation:

Three main types of modulations:

Analog Modulation

 Amplitude modulation
Example: Double sideband with carrier (DSB-WC), Double- sideband
suppressed carrier (DSB-SC), Single sideband suppressed carrier (SSB-SC), vestigial
sideband (VSB)
  Angle modulation (frequency modulation & phase modulation)
Example: Narrow band frequency modulation (NBFM), Wideband frequency
modulation (WBFM), Narrowband phase modulation (NBPM), Wideband phase
modulation (NBPM)

Pulse Modulation

 Carrier is a train of pulses

 Example: Pulse Amplitude Modulation (PAM), Pulse width modulation (PWM) ,
Pulse Position Modulation (PPM)

Digital Modulation

 Modulating signal is analog

o Example: Pulse Code Modulation (PCM), Delta Modulation (DM), Adaptive
Delta Modulation (ADM), Differential Pulse Code Modulation (DPCM),
Adaptive Differential Pulse Code Modulation (ADPCM) etc.
 Modulating signal is digital (binary modulation)
o Example: Amplitude shift keying (ASK), frequency Shift Keying (FSK),
Phase Shift Keying (PSK) etc

Amplitude Modulation (AM)

Amplitude Modulation is the process of changing the amplitude of a relatively high
frequency carrier signal in accordance with the amplitude of the modulating signal
(Information).

The carrier amplitude varied linearly by the modulating signal which usually consists of a
range of audio frequencies. The frequency of the carrier is not affected.

 Application of AM - Radio broadcasting, TV pictures (video), facsimile transmission

 Frequency range for AM - 535 kHz – 1600 kHz
 Bandwidth - 10 kHz

Various forms of Amplitude Modulation

• Conventional Amplitude Modulation (Alternatively known as Full AM or Double

Sideband Large carrier modulation (DSBLC) /Double Sideband Full Carrier (DSBFC)

• Double Sideband Suppressed carrier (DSBSC) modulation

• Single Sideband (SSB) modulation

• Vestigial Sideband (VSB) modulation

Time Domain and Frequency Domain Description

It is the process where, the amplitude of the carrier is varied proportional to that of the
message signal.

Let m (t) be the base-band signal, m (t) ←→ M (ω) and c (t) be the carrier, c(t) = Ac
cos(ωct). fc is chosen such that fc >> W, where W is the maximum frequency component of
m(t). The amplitude modulated signal is given by

s(t) = Ac [1 + kam(t)] cos(ωct)

Fourier Transform on both sides of the above equation

S(ω) = π Ac/2 (δ(ω − ωc) + δ(ω + ωc)) + kaAc/ 2 (M(ω − ωc) + M(ω + ωc))

ka is a constant called amplitude sensitivity.

kam(t) < 1 and it indicates percentage modulation.

Amplitude modulation in time and frequency domain

Single Tone Modulation:

Consider a modulating wave m(t ) that consists of a single tone or single frequency
component given by
Expanding the equation (2), we get

Frequency Domain characteristics of single tone AM

Power relations in AM waves:
Consider the expression for single tone/sinusoidal AM wave

The ratio of total side band power to the total power in the modulated wave is given by
This ratio is called the efficiency of AM system

Generation of AM waves:
Two basic amplitude modulation principles are discussed. They are square law modulation
and switching modulator.

Switching Modulator

Switching Modulator
The total input for the diode at any instant is given by

When the peak amplitude of c(t) is maintained more than that of information
signal, the operation is assumed to be dependent on only c(t) irrespective of m(t).
When c(t) is positive, v2=v1since the diode is forward biased. Similarly, when
c(t) is negative, v2=0 since diode is reverse biased. Based upon above operation,
switching response of the diode is periodic rectangular wave with an amplitude unity
and is given by
 The required AM signal centred at fc can be separated using band pass filter.
The lower cut off-frequency for the band pass filter should be between w and fc-w
and the upper cut-off frequency between fc+w and 2fc. The filter output is given by
the equation
Detection of AM waves
Demodulation is the process of recovering the information signal (base band) from the
incoming modulated signal at the receiver. There are two methods, they are Square law
Detector and Envelope Detector

Envelope Detector

It is a simple and highly effective system. This method is used in most of the commercial AM
radio receivers. An envelope detector is as shown below.

Envelope Detector

During the positive half cycles of the input signals, the diode D is forward biased and
the capacitor C charges up rapidly to the peak of the input signal. When the input signal falls
below this value, the diode becomes reverse biased and the capacitor C discharges through
the load resistor RL.

The discharge process continues until the next positive half cycle. When the input
signal becomes greater than the voltage across the capacitor, the diode conducts again and the
process is repeated.

The charge time constant (rf+Rs)C must be short compared with the carrier period,
the capacitor charges rapidly and there by follows the applied voltage up to the positive peak
when the diode is conducting.That is the charging time constant shall satisfy the condition,

Where ‘W’ is band width of the message signal. The result is that the capacitor voltage or
detector output is nearly the same as the envelope of AM wave.

Advantages and Disadvantages of AM:

Advantages of AM:
 Generation and demodulation of AM wave are easy.
 AM systems are cost effective and easy to build.

Disadvantages:
 AM contains unwanted carrier component, hence it requires more
transmission power.
 The transmission bandwidth is equal to twice the message
bandwidth.

To overcome these limitations, the conventional AM system is modified at the cost of

increased system complexity. Therefore, three types of modified AM systems are discussed.

DSBSC (Double Side Band Suppressed Carrier) modulation:

In DSBC modulation, the modulated wave consists of only the upper and lower side
bands. Transmitted power is saved through the suppression of the carrier wave, but the
channel bandwidth requirement is the same as before.
SSBSC (Single Side Band Suppressed Carrier) modulation: The SSBSC modulated wave
consists of only the upper side band or lower side band. SSBSC is suited for transmission of
voice signals. It is an optimum form of modulation in that it requires the minimum
transmission power and minimum channel band width. Disadvantage is increased cost and
complexity.

VSB (Vestigial Side Band) modulation: In VSB, one side band is completely passed
and just a trace or vestige of the other side band is retained. The required channel bandwidth
is therefore in excess of the message bandwidth by an amount equal to the width of the
vestigial side band. This method is suitable for the transmission of wide band signals.

DSB-SC MODULATION

DSB-SC Time domain and Frequency domain Description:

DSBSC modulators make use of the multiplying action in which the modulating
signal multiplies the carrier wave. In this system, the carrier component is eliminated and
both upper and lower side bands are transmitted. As the carrier component is suppressed, the
power required for transmission is less than that of AM.

Consequently, the modulated signal s(t) under goes a phase reversal , whenever the message
signal m(t) crosses zero as shown below.
 Fig.1. (a) DSB-SC waveform (b) DSB-SC Frequency Spectrum

The envelope of a DSBSC modulated signal is therefore different from the message
signal and the Fourier transform of s(t) is given by
Generation of DSBSC Waves:

Balanced Modulator (Product Modulator)

A balanced modulator consists of two standard amplitude modulators arranged in

a balanced configuration so as to suppress the carrier wave as shown in the following
block diagram. It is assumed that the AM modulators are identical, except for the sign
reversal of the modulating wave applied to the input of one of them. Thus, the output of
the two modulators may be expressed as,

Hence, except for the scaling factor 2ka, the balanced modulator output is equal to
the product of the modulating wave and the carrier.

Ring Modulator

Ring modulator is the most widely used product modulator for generating DSBSC wave and
is shown below.
The four diodes form a ring in which they all point in the same direction. The
diodes are controlled by square wave carrier c(t) of frequency fc, which is applied
longitudinally by means of two center-tapped transformers. Assuming the diodes are
ideal, when the carrier is positive, the outer diodes D1 and D2 are forward biased where
as the inner diodes D3 and D4 are reverse biased, so that the modulator multiplies the
base band signal m(t) by c(t). When the carrier is negative, the diodes D1 and D2 are
reverse biased and D3 and D4 are forward, and the modulator multiplies the base band
signal –m(t) by c(t).

Thus the ring modulator in its ideal form is a product modulator for
square wave carrier and the base band signal m(t). The square wave carrier can be
expanded using Fourier series as

From the above equation it is clear that output from the modulator consists
entirely of modulation products. If the message signal m(t) is band limited to the
frequency band − w < f < w, the output spectrum consists of side bands centred at fc.

Detection of DSB-SC waves:

Coherent Detection:

The message signal m(t) can be uniquely recovered from a DSBSC wave s(t) by
first multiplying s(t) with a locally generated sinusoidal wave and then low pass filtering the
product as shown.
 It is assumed that the local oscillator signal is exactly coherent or synchronized, in
both frequency and phase, with the carrier wave c(t) used in the product modulator to
generate s(t). This method of demodulation is known as coherent detection or
synchronous detection.

Fig.6.Spectrum of output of the product modulator

From the spectrum, it is clear that the unwanted component (first term in the
expression) can be removed by the low-pass filter, provided that the cut-off frequency of
the filter is greater than W but less than 2fc-W. The filter output is given by

The demodulated signal vo(t) is therefore proportional to m(t) when the phase error ϕ
is constant.

Costas Receiver (Costas Loop):

Costas receiver is a synchronous receiver system, suitable for demodulating DSBSC

waves. It consists of two coherent detectors supplied with the same input signal,
 Fig.7. Costas Receiver

The frequency of the local oscillator is adjusted to be the same as the carrier
frequency fc. The detector in the upper path is referred to as the in-phase coherent detector or
I-channel, and that in the lower path is referred to as the quadrature-phase coherent detector
or Q-channel.

These two detector are coupled together to form a negative feedback system designed
in such a way as to maintain the local oscillator synchronous with the carrier wave. Suppose
the local oscillator signal is of the same phase as the carrier
c(t) = Accos(2πfct) wave used to generate the incoming DSBSC wave. Then we find that the
I-channel output contains the desired demodulated signal m(t), where as the Q-channel
output is zero due to quadrature null effect of the Q-channel. Suppose that the
local oscillator phase drifts from its proper value by a small angle ϕ radians. The I-channel
output will remain essentially unchanged, but there will be some signal
appearing at the Q-channel output, which is proportional to
sin(𝜙) ≈ 𝜙 for small ϕ.

This Q-channel output will have same polarity as the I-channel output for one
direction of local oscillator phase drift and opposite polarity for the opposite direction of local
oscillator phase drift. Thus by combining the I-channel and Q-channel outputs in a phase
discriminator (which consists of a multiplier followed by a LPF), a dc control signal is
obtained that automatically corrects for the local phase errors in the voltage-controlled
oscillator.
Introduction of SSB-SC
Standard AM and DSBSC require transmission bandwidth equal to twice the message
bandwidth. In both the cases spectrum contains two side bands of width W Hz,
each. But the upper and lower sides are uniquely related to each other by the virtue of
their symmetry about the carrier frequency. That is, given the amplitude and phase
spectra of either side band, the other can be uniquely determined. Thus if only one side
band is transmitted, and if both the carrier and the other side band are suppressed at the
transmitter, no information is lost. This kind of modulation is called SSBSC and spectral
comparison between DSBSC and SSBSC is shown in the figures 1 and 2.

Frequency Domain Description

side band is transmitted; the resulting SSB modulated wave has the spectrum shown in figure
6. Similarly, the lower side band is represented in duplicate by the frequencies
below fc and those above -fc and when only the lower side band is transmitted, the
spectrum of the corresponding SSB modulated wave shown in figure 5.Thus the
essential function of the SSB modulation is to translate the spectrum of the modulating
wave, either with or without inversion, to a new location in the frequency domain.
The advantage of SSB modulation is reduced bandwidth and the elimination of
high power carrier wave. The main disadvantage is the cost and complexity of its
implementation.

Generation of SSB wave:

Frequency discrimination method

Consider the generation of SSB modulated signal containing the upper side band
only. From a practical point of view, the most severe requirement of SSB generation
arises from the unwanted sideband, the nearest component of which is separated from the
desired side band by twice the lowest frequency component of the message signal. It
implies that, for the generation of an SSB wave to be possible, the message spectrum
must have an energy gap centered at the origin as shown in figure 7. This requirement
is naturally satisfied by voice signals, whose energy gap is about 600Hz wide.
 The frequency discrimination or filter method of SSB generation consists of a
product modulator, which produces DSBSC signal and a band-pass filter to extract the
desired side band and reject the other and is shown in the figure 8.

Application of this method requires that the message signal satisfies two conditions:
1. The message signal m(t) has no low-frequency content. Example: speech, audio, music.
2. The highest frequency component W of the message signal m(t) is much less than the
carrier frequency fc.

Then, under these conditions, the desired side band will appear in a non-overlapping
interval in the spectrum in such a way that it may be selected by an appropriate filter.

In designing the band pass filter, the following requirements should be satisfied:
1.The pass band of the filter occupies the same frequency range as the spectrum of the
desired SSB modulated wave.

2. The width of the guard band of the filter, separating the pass band from the stop
band, where the unwanted sideband of the filter input lies, is twice the lowest frequency
component of the message signal.

When it is necessary to generate an SSB modulated wave occupying a frequency band

that is much higher than that of the message signal, it becomes very difficult to design an
appropriate filter that will pass the desired side band and reject the other. In such a situation
it is necessary to resort to a multiple-modulation process so as to ease the filtering
requirement. This approach is illustrated in the following figure 9 involving two stages of
modulation.

The SSB modulated wave at the first filter output is used as the modulating wave
for the second product modulator, which produces a DSBSC modulated wave with a
spectrum that is symmetrically spaced about the second carrier frequency f2. The
frequency separation between the side bands of this DSBSC modulated wave is
effectively twice the first carrier frequency f1, thereby permitting the second filter to
remove the unwanted side band.

Time Domain Description:

The time domain description of an SSB wave s(t) in the canonical form is given
by the equation 1.
 Following the same procedure, we can find the canonical representation for an SSB
wave
s(t) obtained by transmitting only the lower side band is given by

Phase discrimination method for generating SSB wave:

Time domain description of SSB modulation leads to another method of SSB

generation using the equations 9 or 10. The block diagram of phase discriminator
is as shown in figure 15.
 The phase discriminator consists of two product modulators I and Q, supplied
with carrier waves in-phase quadrature to each other. The incoming base band signal m(t)
is applied to product modulator I, producing a DSBSC modulated wave that contains
reference phase sidebands symmetrically spaced about carrier frequency fc.

The Hilbert transform mˆ (t) of m (t) is applied to product modulator Q, producing a

DSBSC modulated that contains side bands having identical amplitude spectra to those of
modulator I, but with phase spectra such that vector addition or subtraction of the two
modulator outputs results in cancellation of one set of side bands and reinforcement of
the other set.

The use of a plus sign at the summing junction yields an SSB wave with
only the lower side band, whereas the use of a minus sign yields an SSB wave with only
the upper side band. This modulator circuit is called Hartley modulator.

Demodulation of SSB Waves:

Introduction to Vestigial Side Band Modulation
Vestigial sideband is a type of Amplitude modulation in which one side band is
completely passed along with trace or tail or vestige of the other side band. VSB is a
compromise between SSB and DSBSC modulation. In SSB, we send only one side
band, the Bandwidth required to send SSB wave is w. SSB is not appropriate way of
modulation when the message signal contains significant components at extremely low
frequencies. To overcome this VSB is used.

Vestigial Side Band (VSB) modulation is another form of an amplitude-modulated

signal in which a part of the unwanted sideband (called as vestige, hence the name vestigial
sideband) is allowed to appear at the output of VSB transmission system.

The AM signal is passed through a sideband filter before the transmission of SSB
signal. The design of sideband filter can be simplified to a greater extent if a part of the other
sideband is also passed through it. However, in this process the bandwidth of VSB system is
slightly increased.

Generation of VSB Modulated Signal

VSB signal is generated by first generating a DSB-SC signal and then passing it
through a sideband filter which will pass the wanted sideband and a part of unwanted
sideband. Thus, VSB is so called because a vestige is added to SSB spectrum.

The below figure depicts functional block diagram of generating VSB modulated
signal

Figure: Generation of VSB Modulated Signal

A VSB-modulated signal is generated using the frequency discrimination method, in

which firstly a DSB-SC modulated signal is generated and then passed through a sideband-
suppression filter. This type of filter is a specially-designed bandpass filter that distinguishes
VSB modulation from SSB modulation.the cutoff portion of the frequency response of this
filter around the carrier frequency exhibits odd symmetry, that is, (fc-fv)≤|f|≤(fc+fv).

Accordingly the bandwidth of the VSB signal is given as

BW=(fm+fv) Hz
Where fm is the bandwidth of the modulating signal or USB, and fv is the bandwidth of
vestigial sideband (VSB)

Time domain description of VSB Signal

Mathematically, the VSB modulated signal can be described in the time-domain as

s(t)= m(t) Ac cos(2πfct) ± mQ(t) Ac sin(2πfct)

where m(t) is the modulating signal, mQ(t) is the component of m(t) obtained by passing the
message signal through a vestigial filter, Ac cos(2πfct) is the carrier signal, and Ac sin(2πfct)
is the 90o phase shift version of the carrier signal.

The ± sign in the expression corresponds to the transmission of a vestige of the upper-
sideband and lower-sideband respectively. The Quadrature component is required to partially
reduce power in one of the sidebands of the modulated wave s(t) and retain a vestige of the
other sideband as required.

Frequency domain representation of VSB Signal

Since VSB modulated signal includes a vestige (or trace) of the second sideband, only a part
of the second sideband is retained instead of completely eliminating it. Therefore, VSB signal
can be generated from DSB signal followed by VSB filter which is a practical filter.

The below figure shows the DSB signal spectrum, the VSB filter characteristics, and the
resulting output VSB modulated signal spectrum.
Bandwidth Consideration in TV Signals

An important application of VSB modulation technique is in broadcast television. In

commercial TV broadcasting system, there is a basic need to conserve bandwidth.

 The upper-sideband of the video carrier signal is transmitted upto 4MHz without any
attenuation.
 The lower-sideband of the video carrier signal is transmitted without any attenuation
over the range 0.75 MHz (Double side band transmission) and is entirely attenuated at
1.25MHz (single sideband transmission) and the transition is made from one o
another between 0.75MHz and 1.25 MHz (thus the name vestige sideband)
 The audio signal which accompanies the video signal is transmitted by frequency
modulation method using a carrier signal located 4.5 MHz above the video-carrier
signal.
 The audio signal is frequency modulated on a separate carrier signal with a frequency
deviation of 25 KHz. With an audio bandwidth of 10 KHz, the deviation ratio is 2.5
and an FM bandwidth of approximately 70 KHz.
 The frequency range of 100 KHz is allowed on each side of the audio-carrier signal
for the audio sidebands.
 One sideband of the video-modulated signal is attenuated so that it does not interfere
with the lower- sideband of the audio carrier.

Advantages of VSB Modulation

VSB transmission system has several advantages which include

 Use of simple filter design

 Less bandwidth as compared to that of DSBSC signal
 As efficient as SSB
 Possibility of transmission of low frequency components of modulating signals

Facts to Know

VSB is mainly used as a standard modulation technique for transmission of video signals in
TV signals in commercial television broadcasting because the modulating video signal has
large bandwidth and high speed data transmission
Envelope detection of a VSB Wave plus Carrier
Comparison of AM Techniques:

Applications of different AM systems:

 Amplitude Modulation: AM radio, Short wave radio broadcast

 DSB-SC: Data Modems, Color TV’s color signals.
 SSB: Telephone
 VSB: TV picture signals
 UNIT-I (b)
ANGLE MODULATION
Introduction

There are two forms of angle modulation that may be distinguished – phase modulation
and frequency modulation

Basic Definitions: Phase Modulation (PM) and Frequency Modulation (FM)

Let θi(t) denote the angle of modulated sinusoidal carrier, which is a function of the
message. The resulting angle-modulated wave is expressed as

𝒔(𝒕) = 𝑨𝒄𝒄𝒐𝒔[𝜽𝒊(𝒕)] … … … … … (𝟏)

Where Ac is the carrier amplitude. A complete oscillation occurs whenever θi(t) changes by
2π radians. If θi(t) increases monotonically with time, the average frequency in Hz, over an
interval from t to t+∆t, is given by

𝜽𝒊(𝒕 + ∆𝒕) − 𝜽𝒊(𝒕)

𝒇∆𝒕 (𝒕) = … … … … … (𝟐)
𝟐𝝅∆𝒕
Thus the instantaneous frequency of the angle-modulated wave s(t) is defined as

𝒇𝒊(𝒕) = 𝐥𝐢𝐦 𝒇∆𝒕(𝒕)

∆𝒕→𝟎

𝜽𝒊(𝒕 + ∆𝒕) − 𝜽𝒊(𝒕)

𝒇𝒊 (𝒕) = 𝐥𝐢𝐦 [ ]
∆𝒕→𝟎 𝟐𝝅∆𝒕
𝟏 𝒅𝜽𝒊(𝒕)
𝒇 (𝒕) = … … … … … . (𝟑)
𝒊 𝟐𝝅 𝒅𝒕
Thus, according to equation (1), the angle modulated wave s(t) is interpreted as a rotating
Phasor of length Ac and angle θi(t). The angular velocity of such a Phasor is dθi(t)/dt, in
accordance with equ (3).In the simple case of an unmodulated carrier, the angle θi(t) is

𝜽𝒊(𝒕) = 𝟐𝝅𝒇𝒄𝒕 + ∅𝒄

And the corresponding Phasor rotates with a constant angular velocity equal to 2πfc.The
constant ϕc is the value of 𝜃𝑖(𝑡) at t=0.

There are an infinite number of ways in which the angle 𝜃𝑖(𝑡) may be varied in some manner
with the baseband signal.

But the 2 commonly used methods are Phase modulation and Frequency modulation.

Phase Modulation (PM) is that form of angle modulation in which the angle 𝜃𝑖(𝑡) is varied
linearly with the baseband signal m(t), as shown by
 𝜽𝒊(𝒕) = 𝟐𝝅𝒇𝒄𝒕 + 𝒌𝒑𝒎(𝒕) … … … … … … . . (𝟒)

The term 𝟐𝝅𝒇𝒄𝒕 represents the angle of the unmodulated carrier, and the constant 𝒌𝒑
represents the phase sensitivity of the modulator, expressed in radians per volt.

The phase-modulated wave s(t) is thus described in time domain by

𝒔(𝒕) = 𝑨𝒄𝐜𝐨 𝐬[𝟐𝝅𝒇𝒄𝒕 + 𝒌𝒑𝒎(𝒕)] … … … … … (𝟓)

Frequency Modulation (FM) is that form of angle modulation in which the instantaneous
frequency fi(t) is varied linearly with the baseband signal m(t), as shown by

𝒇𝒊(𝒕) = 𝒇𝒄 + 𝒌𝒇𝒎(𝒕) … … … … … … . (𝟔)

The term fc represents the frequency of the unmodulated carrier, and the constant kf
represents the frequency sensitivity of the modulator, expressed in hertz per volt.

Integrating equ.(6) with respect to time and multiplying the result by 2π, we get
𝒕
𝜽𝒊(𝒕) = 𝟐𝝅𝒇𝒄𝒕 + 𝟐𝝅𝒌𝒇 ∫ 𝒎(𝒕) 𝒅𝒕 … … … … … . (𝟕)
𝟎

Where, for convenience it is assumed that the angle of the unmodulated carrier wave is zero
at t=0. The frequency modulated wave is therefore described in the time domain by
𝒕
𝒔(𝒕) = 𝑨𝒄 𝐜𝐨𝐬 [ 𝟐𝝅𝒇𝒄𝒕 + 𝟐𝝅𝒌𝒇 ∫ 𝒎(𝒕) 𝒅𝒕] … … … … … … . . (𝟖)
𝟎

Relationship between PM and FM

Comparing equ (5) with (8) reveals that an FM wave may be regarded as a PM wave in which
𝑡
the modulating wave is ∫0 𝑚(𝑡)𝑑𝑡 in place of m(t).

A PM wave can be generated by first differentiating m(t) and then using the result as the
input to a frequency modulator.
Thus the properties of PM wave can be deduced from those of FM waves and vice versa

Single tone Frequency modulation

Consider a sinusoidal modulating wave defined by

𝒎(𝒕) = 𝑨𝒎𝐜𝐨 𝐬(𝟐𝝅𝒇𝒎𝒕) … … … … … (𝟏)

The instantaneous frequency of the resulting FM wave is

𝒇𝒊(𝒕) = 𝒇𝒄 + 𝒌𝒇𝑨𝒎𝐜𝐨 𝐬(𝟐𝝅𝒇𝒎𝒕)

𝒇𝒊(𝒕) = 𝒇𝒄 + ∆𝒇𝐜𝐨 𝐬(𝟐𝝅𝒇𝒎𝒕) … … … … . . (𝟐)

Where ∆𝒇 = 𝒌𝒇𝑨𝒎 ................................... (𝟑)

The quantity ∆f is called the frequency deviation, representing the maximum departure of the
instantaneous frequency of the FM wave from the carrier frequency fc.

Fundamental characteristic of an FM wave is that the frequency deviation ∆f is proportional

to the amplitude of the modulating wave, and is independent of the modulation frequency.

Using equation (2), the angle 𝜃𝑖(𝑡) of the FM wave is obtained as

𝒕
𝜽𝒊(𝒕) = 𝟐𝝅 ∫ 𝒇𝒊(𝒕)𝒅𝒕
𝟎

∆𝒇
𝜽𝒊(𝒕) = 𝟐𝝅𝒇𝒄𝒕 + 𝒔𝒊𝒏(𝟐𝝅𝒇𝒎𝒕) … … … … … … … (𝟒)
𝒇𝒎

The ratio of the frequency deviation ∆f to the modulation frequency fm is commonly called
the modulation index of the FM wave. Modulation index is denoted by β and is given as
∆𝒇
𝖰= … … … … … … … . (𝟓)
𝒇𝒎

And

𝜽𝒊(𝒕) = 𝟐𝝅𝒇𝒄𝒕 + 𝖰𝒔𝒊𝒏(𝟐𝝅𝒇𝒎𝒕) … … … … … … … (𝟔)

In equation (6) the parameter β represents the phase deviation of the FM wave, that is, the
maximum departure of the angle 𝜃𝑖(𝑡) from the angle 𝟐𝝅𝒇𝒄𝒕 of the unmodulated carrier.

The FM wave itself is given by

𝒔(𝒕) = 𝑨𝒄 𝐜𝐨𝐬[𝟐𝝅𝒇𝒄𝒕 + 𝖰𝒔𝒊𝒏(𝟐𝝅𝒇𝒎𝒕)] … … … … … … . . (𝟕)

Depending on the value of modulation index β, we may distinguish two cases of

frequency modulation. Narrow-band FM for which β is small and Wide-band FM for which β
is large, both compared to one radian.
Narrow-Band Frequency modulation

Consider the Single tone FM wave

𝒔(𝒕) = 𝑨𝒄 𝐜𝐨𝐬[𝟐𝝅𝒇𝒄𝒕 + 𝖰𝒔𝒊𝒏(𝟐𝝅𝒇𝒎𝒕)] … … … … . (𝟏)

Expanding this relation we get

Wide band frequency Modulation

The spectrum of the signle-tone FM wave of equation

𝒔(𝒕) = 𝑨𝒄 𝐜𝐨𝐬[𝟐𝝅𝒇𝒄𝒕 + 𝖰𝒔𝒊𝒏(𝟐𝝅𝒇𝒎𝒕)] … … … … . (𝟏)

For an arbitrary vale of the modulation index 𝖰 is to be determined.

An FM wave produced by a sinusoidal modulating wave as in equation (1) is by itself

nonperiodic, unless the carrier frequency fc is an integral multiple of the modualtion
frequency fm. Rewriting the equation in the form

𝑠̃(𝑡) is periodic function of time,with a fundamental frequency equal to the modulation

frequency fm. 𝑠̃(𝑡) in the form of complex Fourier series is as follows
The integral on the RHS of equation (7) is recognizedasthe nth order Bessel Function of the
first kind and argument 𝖰. This function is commonly denoted by the symbol Jn(𝖰), that is

Equ. (12) is the Fourier series representation of the single-tone FM wave s(t) for an arbitrary
value of 𝖰.

The discrete spectrum of s(t) is obtained by taking the Fourier transform of both sides of
equation (12); thus

In the figure below, we have plotted the Bessel function Jn(𝖰) versus the modulation index 𝖰
for different positive integer value of n.
Properties of Bessel Function

Thus using equations (13) through (16) and the curves in the above figure, following
observations are made
Spectrum Analysis of Sinusoidal FM Wave using Bessel functions
The above figure shows the Discrete amplitude spectra of an FM signal, normalized with
respect to the carrier amplitude, for the case of sinusoidal modulation of varying frequency
and fixed amplitude. Only the spectra for positive frequencies are shown.

Transmission Bandwidth of FM waves

This relation is known as Carson’s rule.

Generation of FM Signal
Direct methods for FM generation
Reactance modulator:
Indirect Method for WBFM Generation (ARMSTRONG’S Method):
Effect of frequency multiplication on a NBFM signal

Detection of FM Signal

Balanced Slope Detector

Phase Locked Loop
PRE-EMPHASIS AND DE-EMPHASIS NETWORKS

In FM, the noise increases linearly with frequency. By this, the higher frequency
components of message signal are badly affected by the noise. To solve this problem, we
can use a pre-emphasis filter of transfer function Hp(ƒ) at the transmitter to boost the higher
frequency components before modulation. Similarly, at the receiver, the de-emphasis filter
of transfer function Hd(ƒ)can be used after demodulator to attenuate the higher frequency
components thereby restoring the original message signal.
The pre-emphasis network and its frequency response are shown in Figure (a) and
(b) respectively. Similarly, the counter part for de-emphasis network is shown in Figure
below.
 Figure (a) Pre-emphasis network. (b) Frequency response of pre-emphasis network.

Figure (a) De-emphasis network. (b) Frequency response of De-emphasis network.

Comparison of AM and FM

S.NO AMPLITUDE MODULATION FREQUENCY MODULATION

1. Band width is very small which is one of It requires much wider channel (7 to 15
the biggest advantage times) as compared to AM.
2. The amplitude of AM signal varies The amplitude of FM signal is constant
depending on modulation index. and independent of depth of the
modulation.
3. Area of reception is large The area of reception is small since it is
limited to line of sight.
4. Transmitters are relatively simple & Transmitters are complex and hence
cheap. expensive.
5. The average power in modulated wave is The average power in frequency
greater than carrier power. This added modulated wave is same as contained in
power is provided by modulating source. un-modulated wave.
6. More susceptible to noise interference and Noise can be easily minimized amplitude
has low signal to noise ratio, it is more variations can be eliminated by using
difficult to eliminate effects of noise. limiter.
7. It is not possible to operate without It is possible to operate several
interference. independent transmitters on same
frequency.
8. The maximum value of modulation index No restriction is placed on modulation
= 1, otherwise over-modulation would index.
result in distortions.
 UNIT - II (a)
NOISE
 Noise in communication System,
 White Noise
 Narrowband Noise –In phase and Quadrature phase components
 Noise Bandwidth
 Noise Figure
 Noise Temperature
 Noise in DSB& SSB System
 Noise in AM System
 Noise in Angle Modulation System
 Threshold effect in Angle Modulation System
Noise in communication system

 Noise is unwanted signal that affects wanted signal

 Noise is random signal that exists in communication systems
Effect of noise

 Degrades system performance (Analog and digital)

 Receiver cannot distinguish signal from noise
 Efficiency of communication system reduces
Types of noise

 Thermal noise/white noise/Johnson noise or fluctuation noise

 Shot noise
 Noise temperature
 Quantization noise

Noise temperature
Equivalent noise temperature is not the physical temperature of amplifier, but a theoretical
construct, that is an equivalent temperature that produces that amount of noise power

𝑇𝑒 = (𝐹 − 1)

White noise
One of the very important random processes is the white noise process. Noises in
many practical situations are approximated by the white noise process. Most importantly, the
white noise plays an important role in modelling of WSS signals.

A white noise process is a random process that has constant power spectral density at
all frequencies. Thus
where is a real constant and called the intensity of the white noise. The corresponding
autocorrelation function is given by

where is the Dirac delta.

The average power of white noise

The autocorrelation function and the PSD of a white noise process is shown in Figure 1
below.

fig: auto correlation and psd of white noise

NARROWBAND NOISE (NBN)

In most communication systems, we are often dealing with band-pass filtering of signals.
Wideband noise will be shaped into band limited noise. If the bandwidth of the band limited
noise is relatively small compared to the carrier frequency, we refer to this as narrowband
noise.

the narrowband noise is expressed as as

where fc is the carrier frequency within the band occupied by the noise. x(t) and y(t)
are known as the quadrature components of the noise n(t). The Hibert transform of
 n(t) is
Proof.
The Fourier transform of n(t) is

Let N^ ( f ) be the Fourier transfor m of n^ ( t). In the frequency domain, N^

(f) = N(f)[-j sgn(f)]. We simply multiply all positive frequency components of N(f)
by -j and all negative frequency components of N(f) by j. Thus

The quadrature components x(t) and y(t) can now be derived from equations

x(t) = n(t)co2fct + n^(t)sin 2fct \

and
y(t) = n(t)cos 2fct - n^(t)sin 2fct

Fig: generation of narrow band noise

 Fig: Generation of quadrature components of n(t).

 Filters at the receiver have enough bandwidth to pass the

desired signal but not too big to pass excess noise.
 Narrowband (NB) fc center frequency is much bigger that the bandwidth.
 Noise at the output of such filters is called narrowband noise (NBN).
 NBN has spectral concentrated about some mid-band frequency fc
 The sample function of such NBN n(t) appears as a sine wave of frequency fc which
modulates slowly in amplitude and phase
Noise figure
The Noise figure is the amount of noise power added by the electronic circuitry in the
receiver to the thermal noise power from the input of the receiver. The thermal noise at the
input to the receiver passes through to the demodulator. This noise is present in the receive
channel and cannot be removed. The noise figure of circuits in the receiver such as amplifiers
and mixers, adds additional noise to the receive channel. This raises the noise floor at the
demodulator.

Noise Bandwidth
A filter’s equivalent noise bandwidth (ENBW) is defined as the bandwidth of a perfect
rectangular filter that passes the same amount of power as the cumulative bandwidth of the
channel selective filters in the receiver. At this point we would like to know the noise floor in
our receiver, i.e. the noise power in the receiver intermediate frequency (IF) filter bandwidth
that comes from kTB. Since the units of kTB are Watts/ Hz, calculate the noise floor in the
channel bandwidth by multiplying the noise power in a 1 Hz bandwidth by the overall
equivalent noise bandwidth in Hz.

NOISE IN DSB-SC SYSTEM:

Let the transmitted signal is

The received signal at the output of the receiver noise- limiting filter : Sum of this signal and
filtered noise .A filtered noise process can be expressed in terms of its in-phase and quadrature
components as

where nc(t) is in-phase component and ns(t) is quadrature component

Received signal (Adding the filtered noise to the modulated signal)

Demodulate the received signal by first multiplying r(t) by a locally generated sinusoid
cos(2 fct + ), where is the phase of the sinusoid.Then passing the product signal through

an ideal lowpass filter having a bandwidth W.

The low pass filter rejects the double frequency components and passes only the low pass
components.

the effect of a phase difference between the received carrier and a locally generated carrier at
2
the receiver is a drop equal to cos ( ) in the received signal

power. Phase-locked loop

The effect of a phase-locked loop is to generate phase of the received carrier at the receiver.

If a phase-locked loop is employed, then = 0 and the demodulator is

called a coherent or synchronous demodulator.

In our analysis in this section, we assume that we are employing a coherent demodulator.

With this assumption, we assume that =0

Therefore, at the receiver output, the message signal and the noise components are additive
and we are able to define a meaningful SNR. The message signal power is given by
Power PM is the content of the messagesignal

The noise power is given by

The power content of n(t) can be found by noting that it is the result of passing nw(t) through
a filter with bandwidth Bc.Therefore, the power spectral density of n(t) is given by

which is identical to baseband SNR.

In DSB-SC AM, the output SNR is the same as the SNR for a baseband system. DSB-SC AM
does not provide any SNR improvement over a simple baseband communication system.

NOISE IN SSB-SC SYSTEM:

Let SSB modulated signal is

Input to the demodulator

Assumption : Demodulation with an ideal phase reference.

Hence, the output of the lowpass filter is the in-phase component (with a
coefficient of ½) of the precedingsignal.

The signal-to-noise ratio in an SSB system is equivalent to that of a DSB system.

Noise in Conventional AM

Where a is the modulation index

mn(t) is normalized so that its minimum value is -1

If a synchronous demodulator is employed, the situation is basically similar to the

DSB case, except that we have 1 + amn(t) instead of m(t).
  In practical applications, the modulation index a is in the range of 0.8-0.9.

 Power content of the normalized message process depends on the message source.

 Speech signals : Large dynamic range, PM is about 0.1.

 The overall loss in SNR, when compared to a baseband system, is a

factor of 0.075 or equivalent to a loss of 11 dB.

The reason for this loss is that a large part of the transmitter power is used to send the
carrier component of the modulated signal and not the desired signal. To analyze the
envelope-detector performance in the presence of noise, we must use certain
approximations.

This is a result of the nonlinear structure of an envelope detector, which makes an exact
analysis difficult

In this case, the demodulator detects the envelope of the received signal and the noise
process.
The input to the envelope detector is

Therefore, the envelope of r ( t ) is given by

Now we assume that the signal component in r ( t ) is much stronger than the noise
component. Then

Therefore, we have a high probability that

After removing the DC component, we obtain

which is basically the same as y(t) for the synchronous demodulation without the ½
coefficient.
This coefficient, of course, has no effect on the final SNR. So we conclude that, under the
assumption of high SNR at the receiver input, the performance of synchronous and envelope
demodulators is the same.

However, if the preceding assumption is not true, that is, if we assume that, at the receiver
input, the noise power is much stronger than the signal power, Then
We observe that, at the demodulator output, the signal and the noise components are no
longer additive. In fact, the signal component is multiplied by noise and is no longer
distinguishable. In this case, no meaningful SNR can be defined. We say that this system is
operating below the threshold. The subject of threshold and its effect on the performance of
a communication system will be covered in more detail when we discuss the noise
performance in angle modulation.

Effect of threshold in angle modulation system:

FM THRESHOLD EFFECT FM threshold is usually defined as a Carrier-to-Noise ratio at

which demodulated Signal-to-Noise ratio falls 1dB below the linear relationship . This is the
effect produced in an FM receiver when noise limits the desired information signal. It occurs
at about 10 dB, as earlier stated in 5 the introduction, which is at a point where the FM signal-
to-Noise improvement is measured. Below the FM threshold point, the noise signal (whose
amplitude and phase are randomly varying) may instantaneously have amplitude greater than
that of the wanted signal. When this happens, the noise will produce a sudden change in the
phase of the FM demodulator output. In an audio system, this sudden phase change makes a
“click”. In video applications the term “click noise” is used to describe short horizontal black
and white lines that appear randomly over a picture

An important aspect of analogue FM satellite systems is FM threshold effect. In FM systems

where the signal level is well above noise received carrier-to-noise ratio and demodulated
signal-to-noise ratio are related by:

The expression however does not apply when the carrier-to-noise ratio decreases below a
certain point. Below this critical point the signal-to-noise ratio decreases significantly. This is
known as the FM threshold effect (FM threshold is usually defined as the carrier-to-noise
ratio at which the demodulated signal-to-noise ratio fall 1 dB below the linear relationship
given in Eqn 9. It generally is considered to occur at about 10 dB).

Below the FM threshold point the noise signal (whose amplitude and phase are randomly
varying), may instantaneously have an amplitude greater than that of the wanted signal.
When this happens the noise will produce a sudden change in the phase of the FM
demodulator output. In an audio system this sudden phase change makes a "click". In video
applications the term "click noise" is used to describe short horizontal black and white lines
that appear randomly over a picture, because satellite communications systems are power
limited they usually operate with only a small design margin above the FM threshold point
(perhaps a few dB). Because of this circuit designers have tried to devise techniques to delay
the onset of the FM threshold effect. These devices are generally known as FM threshold
extension demodulators. Techniques such as FM feedback, phase locked loops and frequency
locked loops are used to achieve this effect. By such techniques the onset of FM threshold
effects can be delayed till the C/N ratio is around 7 dB.

Noise in Angle Modulated Systems

Like AM, noise performance of angle modulated systems is characterized by parameter γ

Note: if bandwidth ratio is increased by a factor 2.Then increases by a factor 4

This exchange of bandwidth and noise performance is an important feature of FM

 UNIT-II (b)
PULSE MODULATION
Introduction:

Pulse Modulation

 Carrier is a train of pulses

 Example: Pulse Amplitude Modulation (PAM), Pulse width modulation (PWM) ,
Pulse Position Modulation (PPM)

Types of Pulse Modulation:

⚫ The immediate result of sampling is a pulse-amplitude modulation (PAM) signal

⚫ PAM is an analog scheme in which the amplitude of the pulse is proportional to the
amplitude of the signal at the instant of sampling

⚫ Another analog pulse-forming technique is known as pulse-duration modulation

(PDM). This is also known as pulse-width modulation (PWM)

⚫ Pulse-position modulation is closely related to PDM

Pulse Amplitude Modulation:

In PAM, amplitude of pulses is varied in accordance with instantaneous value of

modulating signal.

PAM Generation:

The carrier is in the form of narrow pulses having frequency fc. The uniform
sampling takes place in multiplier to generate PAM signal. Samples are placed Ts sec
away from each other.
 Figure PAM Modulator

⚫ The circuit is simple emitter follower.

⚫ In the absence of the clock signal, the output follows input.
⚫ The modulating signal is applied as the input signal.
⚫ Another input to the base of the transistor is the clock signal.
⚫ The frequency of the clock signal is made equal to the desired carrier pulse train
frequency.

⚫ The amplitude of the clock signal is chosen the high level is at ground level(0v) and
low level at some negative voltage sufficient to bring the transistor in cutoff region.

⚫ When clock is high, circuit operates as emitter follower and the output follows in the
input modulating signal.

⚫ When clock signal is low, transistor is cutoff and output is zero.

⚫ Thus the output is the desired PAM signal.

PAM Demodulator:
⚫ The PAM demodulator circuit which is just an envelope detector followed by a
second order op-amp low pass filter (to have good filtering characteristics) is as
shown below

Figure PAM Demodulator

Pulse Width Modulation:
⚫ In this type, the amplitude is maintained constant but the width of each pulse is varied
in accordance with instantaneous value of the analog signal.

⚫ In PWM information is contained in width variation. This is similar to FM.

⚫ In pulse width modulation (PWM), the width of each pulse is made directly
proportional to the amplitude of the information signal.

Pulse Position Modulation:

⚫ In this type, the sampled waveform has fixed amplitude and width whereas the
position of each pulse is varied as per instantaneous value of the analog signal.

⚫ PPM signal is further modification of a PWM signal.

PPM & PWM Modulator:

Figure PWM & PPM Modulator

• The PPM signal can be generated from PWM signal.

• The PWM pulses obtained at the comparator output are applied to a mono stable multi
vibrator which is negative edge triggered.
 • Hence for each trailing edge of PWM signal, the monostable output goes high. It
remains high for a fixed time decided by its RC components.

• Thus as the trailing edges of the PWM signal keeps shifting in proportion with the
modulating signal, the PPM pulses also keep shifting.

• Therefore all the PPM pulses have the same amplitude and width. The information is
conveyed via changing position of pulses.

Figure PWM & PPM Modulation waveforms

PWM Demodulator:

Figure PWM Demodulator

 ⚫ Transistor T1 works as an inverter.

⚫ During time interval A-B when the PWM signal is high the input to transistor T2 is
low.

⚫ Therefore, during this time interval T2 is cut-off and capacitor C is charged through
an R-C combination.

⚫ During time interval B-C when PWM signal is low, the input to transistor T2 is high,
and it gets saturated.

⚫ The capacitor C discharges rapidly through T2. The collector voltage of T2 during B-
C is low.

⚫ Thus, the waveform at the collector of T2is similar to saw-tooth waveform whose
envelope is the modulating signal.

⚫ Passing it through 2nd order op-amp Low Pass Filter, gives demodulated signal.

PPM Demodulator:

Figure PPM Demodulator

⚫ The gaps between the pulses of a PPM signal contain the information regarding the
modulating signal.

⚫ During gap A-B between the pulses the transistor is cut-off and the capacitor C gets
charged through R-C combination.

⚫ During the pulse duration B-C the capacitor discharges through transistor and the
collector voltage becomes low.

⚫ Thus, waveform across collector is saw-tooth waveform whose envelope is the

modulating signal.

⚫ Passing it through 2nd order op-amp Low Pass Filter, gives demodulated signal.
Multiplexing

Multiplexing is the set of techniques that allows the simultaneous transmission of multiple
signals across a single common communications channel.

Multiplexing is the transmission of analog or digital information from one or more sources to
one or more destination over the same transmission link.

Although transmissions occur on the same transmitting medium, they do not necessarily
occupy the same bandwidth or even occur at the same time.

Frequency Division Multiplexing

Frequency division multiplexing (FDM) is a technique of multiplexing which means

combining more than one signal over a shared medium. In FDM, signals of different
frequencies are combined for concurrent transmission.

In FDM, the total bandwidth is divided to a set of frequency bands that do not
overlap. Each of these bands is a carrier of a different signal that is generated and modulated
by one of the sending devices. The frequency bands are separated from one another by strips
of unused frequencies called the guard bands, to prevent overlapping of signals.

The modulated signals are combined together using a multiplexer (MUX) in the
sending end. The combined signal is transmitted over the communication channel, thus
allowing multiple independent data streams to be transmitted simultaneously. At the
receiving end, the individual signals are extracted from the combined signal by the process of
demultiplexing (DEMUX).

FDM system Transmitter

 Analog or digital inputs: mi (t); i = 1,2, ... n

 Each input modulates a subcarrier of frequency fi; i=1, 2, .... n
 Signals are summed to produce a composite baseband signal denoted as mb(t)
 fi is chosen such that there is no overlap.
Spectrum of composite baseband modulating signal

FDM system Receiver

 The Composite base band signal mb(t) is passed through n band pass filters with
response centred on fi
 Each si(t) component is demodulated to recover the original analog/digital data.
Time Division Multiplexing

TDM technique combines time-domain samples from different message signals (sampled at
same rate) and transmits them together across the same channel.

The multiplexing is performed using a commutator (switch). At the receiver a decommutator

(switch) is used in synchronism with the commutator to demultiplex the data.

The input signals, all band limited to fm (max) by the LPFs are sequentially sampled at the
transmitter by a commutator.

The Switch makes one complete revolution in Ts,(1/fs) extracting one sample from each
input. Hence the output is a PAM waveform containing the individual message sampled
periodically interlaced in time.

A set of pulses consisting of one sample from each input signal is called a frame.

At the receiver the de-commutator separates the samples and distributes them to a bank of
LPFs, which in turn reconstruct the original messages.

Synchronizing is provided to keep the de-commutator in step with the commutator.

Elements of Digital Communication Systems

Figure Elements of Digital Communication Systems

1. Information Source and Input Transducer:

The source of information can be analog or digital, e.g. analog: audio or video
signal, digital: like teletype signal. In digital communication the signal produced by
this source is converted into digital signal which consists of 1′s and 0′s. For this we
need a source encoder.
2. Source Encoder:
In digital communication we convert the signal from source into digital signal
as mentioned above. The point to remember is we should like to use as few binary
digits as possible to represent the signal. In such a way this efficient representation of
the source output results in little or no redundancy. This sequence of binary digits is
called information sequence.

Source Encoding or Data Compression: the process of efficiently converting

the output of whether analog or digital source into a sequence of binary digits is
known as source encoding.

3. Channel Encoder:
The information sequence is passed through the channel encoder. The purpose
of the channel encoder is to introduce, in controlled manner, some redundancy in the
binary information sequence that can be used at the receiver to overcome the effects
of noise and interference encountered in the transmission on the signal through the
channel.
For example take k bits of the information sequence and map that k bits to
unique n bit sequence called code word. The amount of redundancy introduced is
 measured by the ratio n/k and the reciprocal of this ratio (k/n) is known as rate of code
or code rate.
4. Digital Modulator:
The binary sequence is passed to digital modulator which in turns convert the
sequence into electric signals so that we can transmit them on channel (we will see
channel later). The digital modulator maps the binary sequences into signal wave
forms , for example if we represent 1 by sin x and 0 by cos x then we will transmit sin
x for 1 and cos x for 0. ( a case similar to BPSK)
5. Channel:
The communication channel is the physical medium that is used for
transmitting signals from transmitter to receiver. In wireless system, this channel
consists of atmosphere , for traditional telephony, this channel is wired , there are
optical channels, under water acoustic channels etc.We further discriminate this
channels on the basis of their property and characteristics, like AWGN channel etc.
6. Digital Demodulator:
The digital demodulator processes the channel corrupted transmitted
waveform and reduces the waveform to the sequence of numbers that represents
estimates of the transmitted data symbols.
7. Channel Decoder:
This sequence of numbers then passed through the channel decoder which
attempts to reconstruct the original information sequence from the knowledge of the
code used by the channel encoder and the redundancy contained in the received data

Note: The average probability of a bit error at the output of the decoder is a
measure of the performance of the demodulator – decoder combination.

8. Source Decoder:
At the end, if an analog signal is desired then source decoder tries to decode
the sequence from the knowledge of the encoding algorithm. And which results in the
approximate replica of the input at the transmitter end.

9. Output Transducer:
Finally we get the desired signal in desired format analog or digital.

Advantages of digital communication

 Can withstand channel noise and distortion much better as long as the noise and the
distortion are within limits.
 Regenerative repeaters prevent accumulation of noise along the path.
 Digital hardware implementation is flexible.
 Digital signals can be coded to yield extremely low error rates, high fidelity and
well as privacy.
 Digital communication is inherently more efficient than analog in realizing the
exchange of SNR for bandwidth.
 It is easier and more efficient to multiplex several digital signals.
  Digital signal storage is relatively easy and inexpensive.
 Reproduction with digital messages is extremely reliable without deterioation.
 The cost of digital hardware continues to halve every two or three years while
performance or capacity doubles over the same time period.

Disadvantages

 TDM digital transmission is not compatible with FDM

 A Digital system requires large bandwidth.

Elements of PCM System

Sampling:

 Process of converting analog signal into discrete signal.

 Sampling is common in all pulse modulation techniques
 The signal is sampled at regular intervals such that each sample is proportional to
amplitude of signal at that instant
 Analog signal is sampled every 𝑇𝑠̃ 𝑆𝑒𝑐𝑠̃, called sampling interval. 𝑓𝑠̃=1/𝑇𝑆 is called
sampling rate or sampling frequency.
 𝑓𝑠̃=2𝑓𝑚 is Min. sampling rate called Nyquist rate. Sampled spectrum (𝜔) is repeating
periodically without overlapping.
 Original spectrum is centered at 𝜔=0 and having bandwidth of 𝜔𝑚. Spectrum can be
recovered by passing through low pass filter with cut-off 𝜔𝑚.
 For 𝑓𝑠̃<2𝑓𝑚 sampled spectrum will overlap and cannot be recovered back. This is
called aliasing.
Sampling methods:

 Ideal – An impulse at each sampling instant.

 Natural – A pulse of Short width with varying amplitude.
 Flat Top – Uses sample and hold, like natural but with single amplitude value.

Fig. 4 Types of Sampling

Sampling of band-pass Signals:

 A band-pass signal of bandwidth 2fm can be completely recovered from its samples.
Min. sampling rate =2×𝐵𝑎𝑛𝑑𝑤𝑖𝑑𝑡ℎ

=2×2𝑓𝑚=4𝑓𝑚

 Range of minimum sampling frequencies is in the range of 2×𝐵𝑊 𝑡𝑜 4×𝐵𝑊

Instantaneous Sampling or Impulse Sampling:

 Sampling function is train of spectrum remains constant impulses throughout

frequency range. It is not practical.

Natural sampling:

 The spectrum is weighted by a sinc function.

 Amplitude of high frequency components reduces.

Flat top sampling:

 Here top of the samples remains constant.

 In the spectrum high frequency components are attenuated due sinc pulse roll off.
This is known as Aperture effect.
 If pulse width increases aperture effect is more i.e. more attenuation of high frequency
components.
PCM Generator
Transmission BW in PCM
PCM Receiver

Quantization

 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
 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.
 There are two types of Quantization
o Uniform Quantization
o 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.

Uniform Quantization:
• There are two types of uniform quantization.
– Mid-Rise type
– Mid-Tread type.
• The following figures represent the two types of uniform quantization.

• The Mid-Rise type is so called because the origin lies in the middle of a raising part
of the stair-case like graph. The quantization levels in this type are even in number.
• The Mid-tread type is so called because the origin lies in the middle of a tread of the
stair-case like graph. The quantization levels in this type are odd in number.
• Both the mid-rise and mid-tread type of uniform quantizer is symmetric about the
origin.
Quantization Noise and Signal to Noise ratio in PCM System
Derivation of Maximum Signal to Quantization Noise Ratio for Linear Quantization:
Non-Uniform Quantization:
In non-uniform quantization, the step size is not fixed. It varies according to certain
law or as per input signal amplitude. The following fig shows the characteristics of Non
uniform quantizer.
Companding PCM System
• Non-uniform quantizers are difficult to make and expensive.
• An alternative is to first pass the speech signal through nonlinearity before quantizing
with a uniform quantizer.
• The nonlinearity causes the signal amplitude to be compressed.
– The input to the quantizer will have a more uniform distribution.
• At the receiver, the signal is expanded by an inverse to the nonlinearity.
• The process of compressing and expanding is called Companding.
Differential Pulse Code Modulation (DPCM)
Redundant Information in PCM
Introduction to Delta Modulation
Condition for Slope overload distortion occurrence
Slope overload distortion will occur if
Expression for Signal to Quantization Noise power ratio for Delta Modulation
 UNIT- III

BASEBAND PULSE TRANSMISSION

Inter symbol Interference:

Generally, digital data is represented by electrical pulse, communication
channel is always band limited. Such a channel disperses or spreads a pulse
carrying digitized samples passing through it. When the channel bandwidth is
greater than bandwidth of pulse, spreading of pulse is very less. But when
channel bandwidth is close to signal bandwidth, i.e. if we transmit digital data
which demands more bandwidth which exceeds channel bandwidth, spreading
will occur and cause signal pulses to overlap. This overlapping is
called InterSymbol Interference. In short it is called ISI. Similar to
interference caused by other sources, ISI causes degradations of signal if left
uncontrolled. This problem of ISI exists strongly in Telephone channels like
coaxial cables and optical fibers.

The main objective is to study the effect of ISI, when digital data is transmitted
through band limited channel and solution to overcome the degradation of
waveform by properly shaping pulse

(Source:Brainkart)

The effect of sequence of pulses transmitted through channel is shown in

fig. The Spreading of pulse is greater than symbol duration, as a result adjacent
pulses interfere. i.e. pulses get completely smeared, tail of smeared pulse enter
into adjacent symbol intervals making it difficult to decide actual transmitted
pulse. First let us have look at different formats of transmitting digital data.In
base band transmission best way is to map digits or symbols into pulse
waveform. This waveform is generally termed as Line codes.
EYE PATTERN
The quality of digital transmission systems are evaluated using the bit
error rate. Degradation of quality occurs in each process modulation,
transmission, and detection. The eye pattern is experimental method that
contains all the information concerning the degradation of quality. Therefore,
careful analysis of the eye pattern is important in analyzing the degradation
mechanism.
• Eye patterns can be observed using an oscilloscope. The received wave is
applied to the vertical deflection plates of an oscilloscope and the saw tooth
wave at a rate equal to transmitted symbol rate is applied to the horizontal
deflection plates, resulting display is eye pattern as it resembles human eye.
• The interior region of eye pattern is called eye opening

(Source:Brainkart)

We get superposition of successive symbol intervals to produce eye

pattern as shown below.

Fig 3.1Eye pattern (Source:Brainkart)

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

Fig 3.2 Example of eye pattern: Binary-PAM with noise no ISI (Source:Brainkart)

EQUALISING FILTER
Adaptive equalization
• An equalizer is a filter that compensates for the dispersion effects of a
channel. Adaptive equalizer can adjust its coefficients continuously during the
transmission of data.
Pre channel equalization
 requires feed back channel
 causes burden on transmission.
Post channel equalization
Achieved prior to data transmission by training the filter with the guidance of a
training sequence transmitted through the channel so as to adjust the filter
parameters to optimum values.
Adaptive equalization
It consists of tapped delay line filter with set of delay elements, set of
adjustable multipliers connected to the delay line taps and a summer for adding
multiplier outputs.

Fig 3.3 Adaptive equalization (Source:Brainkart)

The output of the Adaptive equalizer is given by

Ci is weight of the ith tap Total number of taps are M .Tap spacing is equal to
symbol duration T of transmitted signal In a conventional FIR filter the tap
weights are constant and particular designed response is obtained. In the
adaptive equaliser the Ci's are variable and are adjusted by an algorithm.
Two modes of operation
1. Training mode
2. Decision directed mode

Mechanism of adaptation

Fig 3.4 Mechanism of adaptation (Source:Brainkart)

Training mode
A known sequence d(nT) is transmitted and synchronized version of it is
generated in the receiver applied to adaptive equalizer. This training sequence
has maximal length PN Sequence, because it has large average power and large
SNR, resulting response sequence (Impulse) is observed by measuring the filter
outputs at the sampling instants. The difference between resulting response
y(nT) and desired response d(nT)is error signal which is used to estimate the
direction in which the coefficients of filter are to be optimized using algorithms.
Matched Filter
It is obtained by correlating a known delayed signal, or template, with an
unknown signal to detect the presence of the template in the unknown
signal. This is equivalent to convolving the unknown signal with
a conjugated time-reversed version of the template. The matched filter is the
optimal linear filter for maximizing the signal-to-noise ratio (SNR) in the
presence of additive stochastic noise.
Matched filters are commonly used in radar, in which a known signal is sent
out, and the reflected signal is examined for common elements of the out-going
signal. Pulse compression is an example of matched filtering. It is so called
because the impulse response is matched to input pulse signals. Two-
dimensional matched filters are commonly used in image processing, e.g., to
improve the SNR of X-ray observations. Matched filtering is a demodulation
technique with LTI (linear time invariant) filters to maximize SNR. It was
originally also known as a North filter.

Pulse Shaping
It is the process of changing the waveform of transmitted pulses. Its
purpose is to make the transmitted signal better suited to its purpose or
the communication channel, typically by limiting the effective bandwidth of the
transmission. By filtering the transmitted pulses this way, the inter symbol
interference caused by the channel can be kept in control. In RF
communication, pulse shaping is essential for making the signal fit in its
frequency band.
Typically pulse shaping occurs after line coding and modulation.

Need for pulse shaping

Transmitting a signal at high modulation rate through a band-limited
channel can create inter symbol interference. As the modulation rate increases,
the signal's bandwidth increases. When the signal's bandwidth becomes larger
than the channel bandwidth, the channel starts to introduce distortion to the
signal. This distortion usually manifests itself as inter symbol interference.

The signal's spectrum is determined by the modulation scheme and data

rate used by the transmitter, but can be modified with a pulse shaping filter.
Usually the transmitted symbols are represented as a time sequence of dirac
delta pulses. This theoretical signal is then filtered with the pulse shaping filter,
producing the transmitted signal.

In many base band communication systems the pulse shaping filter is

implicitly a boxcar filter. Its Fourier transform is of the form sin(x)/x, and has
significant signal power at frequencies higher than symbol rate. This is not a big
problem when optical fibre or even twisted pair cable is used as the
communication channel. However, in RF communications this would waste
bandwidth, and only tightly specified frequency bands are used for single
transmissions. In other words, the channel for the signal is band-limited.
Therefore better filters have been developed, which attempt to minimize the
bandwidth needed for a certain symbol rate.

An example in other areas of electronics is the generation of pulses where

the rise time need to be short; one way to do this is to start with a slower-rising
pulse, and decrease the rise time, for example with a step recovery diode circuit.
Pulse shaping filters

(Source:Brainkart)

A typical NRZ coded signal is implicitly filtered with a sinc filter.

Not every filter can be used as a pulse shaping filter. The filter itself must
not introduce inter symbol interference — it needs to satisfy certain
criteria. The Nyquist ISI criterion is a commonly used criterion for evaluation,
because it relates the frequency spectrum of the transmitter signal to
intersymbol interference.

Examples of pulse shaping filters that are commonly found in communication

systems are:

 Sinc shaped filter

 Raised-cosine filter
 Gaussian filter
Sender side pulse shaping is often combined with a receiver side
matched filter to achieve optimum tolerance for noise in the system. In this case
the pulse shaping is equally distributed between the sender and receiver filters.
The filters' amplitude responses are thus point wise square roots of the system
filters.
Other approaches that eliminate complex pulse shaping filters have been
invented. In OFDM, the carriers are modulated so slowly that each carrier is
virtually unaffected by the bandwidth limitation of the channel.
Sinc filter

Fig 3.5 Amplitude response of raised-cosine filter with various roll-off factors (Source:Brainkart)

It is also called as Boxcar filter as its frequency domain equivalent is a

rectangular shape. Theoretically the best pulse shaping filter would be the sinc
filter, but it cannot be implemented precisely. It is a non-causal filter with
relatively slowly decaying tails. It is also problematic from a synchronization
point of view as any phase error results in steeply increasing inter symbol
interference.
Raised-cosine filter
Raised-cosine is similar to sinc, with the tradeoff of smaller side lobes for a
slightly larger spectral width. Raised-cosine filters are practical to implement
and they are in wide use. They have a configurable excess bandwidth, so
communication systems can choose a trade off between a simpler filter and
spectral efficiency.
Gaussian filter
This gives an output pulse shaped like a Gaussian function.

Nyquist criterion
When the baseband filters in the communication system satisfy
the Nyquist criterion, symbols can be transmitted over a channel with flat
response within a limited frequency band, without ISI. Examples of such
baseband filters are the raised-cosine filter, or the sinc filter as the ideal case.
 UNIT – IV
DIGITAL PASSBAND TRANSMISSION
 UNIT- V (a)
DIGITAL MODULATION TECHNIQUES

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

There are many types of digital modulation techniques and we can even use a combination of these
techniques as well. In this chapter, we will be discussing the most prominent digital modulation
techniques.

if the information signal is digital and the amplitude (lV of the carrier is varied proportional to
the information signal, a digitally modulated signal called amplitude shift keying (ASK) is
produced.
If the frequency (f) is varied proportional to the information signal, frequency shift keying (FSK) is
produced, and if the phase of the carrier (0) is varied proportional to the information signal,
phase shift keying (PSK) is produced. If both the amplitude and the phase are varied proportional to
the information signal, quadrature amplitude modulation (QAM) results. ASK, FSK, PSK, and
QAM are all forms of digital modulation:

a simplified block diagram for a digital modulation system.

Amplitude Shift Keying

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

Amplitude Shift Keying (ASK) is a type of Amplitude Modulation which represents the binary
data in the form of variations in the amplitude of a signal.

Following is the diagram for ASK modulated waveform along with its input.
 Any modulated signal has a high frequency carrier. The binary signal when ASK is modulated,
gives a zero value for LOW input and gives the carrier output for HIGH input.
Mathematically, amplitude-shift keying is

where vask(t) = amplitude-shift keying wave

vm(t) = digital information (modulating) signal (volts)
A/2 = unmodulated carrier amplitude (volts)
ωc= analog carrier radian frequency (radians per second, 2πfct)

In above Equation, the modulating signal [vm(t)] is a normalized binary waveform, where + 1 V =
logic 1 and -1 V = logic 0. Therefore, for a logic 1 input, vm(t) = + 1 V, Equation 2.12 reduces to

Mathematically, amplitude-shift keying is (2.12) where vask(t) = amplitude-shift keying wave

vm(t) = digital information (modulating) signal (volts) A/2 = unmodulated carrier amplitude (volts)
ωc= analog carrier radian frequency (radians per second, 2πfct) In Equation 2.12, the modulating
signal [vm(t)] is a normalized binary waveform, where + 1 V = logic 1 and -1 V = logic 0.
Therefore, for a logic 1 input, vm(t) = + 1 V, Equation 2.12 reduces to and for a logic 0 input, vm(t)
= -1 V,Equation reduces to

Thus, the modulated wave vask(t),is either A cos(ωct) or 0. Hence, the carrier is either "on “or
"off," which is why amplitude-shift keying is sometimes referred to as on-off keying (OOK).
it can be seen that for every change in the input binary data stream, there is one change in the ASK
waveform, and the time of one bit (tb) equals the time of one analog signaling element (t,).
B = fb/1 = fb baud = fb/1 = fb

Example :
Determine the baud and minimum bandwidth necessary to pass a 10 kbps binary signal using
amplitude shift keying. 10Solution For ASK, N = 1, and the baud and minimum bandwidth are
determined from Equations 2.11 and 2.10, respectively:

B = 10,000 / 1 = 10,000
baud = 10, 000 /1 = 10,000
The use of amplitude-modulated analog carriers to transport digital information is a relatively low-
quality, low-cost type of digital modulation and, therefore, is seldom used except for very low-
speed telemetry circuits.
ASK TRANSMITTER:
The input binary sequence is applied to the product modulator. The product modulator amplitude
modulates the sinusoidal carrier .it passes the carrier when input bit is ‘1’ .it blocks the carrier when
input bit is ‘0.’

Coherent ASK DETECTOR:

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

Frequency Shift Keying (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. The binary 1s and 0s are called Mark and Space frequencies.

FSK is a form of constant-amplitude angle modulation similar to standard frequency modulation

(FM) except the modulating signal is a binary signal that varies between two discrete voltage levels
rather than a continuously changing analog waveform.Consequently, FSK is sometimes called
binary FSK (BFSK). The general expression for FSK is
where

vfsk(t) = binary FSK waveform

Vc = peak analog carrier amplitude (volts)

fc = analog carrier center frequency(hertz)

f=peak change (shift)in the analog carrier frequency(hertz)

vm(t) = binary input (modulating) signal (volts)

From Equation 2.13, it can be seen that the peak shift in the carrier frequency ( f) is proportional to
the amplitude of the binary input signal (vm[t]), and the direction of the shift is determined by the
polarity.

The modulating signal is a normalized binary waveform where a logic 1 = + 1 V and a logic 0 = -1
V. Thus, for a logic l input, vm(t) = + 1, Equation 2.13 can be rewritten as

For a logic 0 input, vm(t) = -1, Equation becomes

With binary FSK, the carrier center frequency (fc) is shifted (deviated) up and down in the
frequency domain by the binary input signal as shown in Figure 2-3.

FIGURE: FSK in the frequency domain

As the binary input signal changes from a logic 0 to a logic 1 and vice versa, the output frequency
shifts between two frequencies: a mark, or logic 1 frequency (fm), and a space, or logic 0 frequency
(fs). The mark and space frequencies are separated from the carrier frequency by the peak frequency
deviation ( f) and from each other by 2 f.

Frequency deviation is illustrated in Figure 2-3 and expressed mathematically as

f = |fm – fs| / 2 (2.14)

where f = frequency deviation (hertz)

|fm – fs| = absolute difference between the mark and space frequencies (hertz)

Figure 2-4a shows in the time domain the binary input to an FSK modulator and the corresponding
FSK output.

When the binary input (fb) changes from a logic 1 to a logic 0 and vice versa, the FSK output
frequency shifts from a mark ( fm) to a space (fs) frequency and vice versa.

In Figure 2-4a, the mark frequency is the higher frequency (fc + f) and the space frequency is the
lower frequency (fc - f), although this relationship could be just the opposite.

Figure 2-4b shows the truth table for a binary FSK modulator. The truth table shows the input and
output possibilities for a given digital modulation scheme.
FSK Bit Rate, Baud, and Bandwidth

In Figure 2-4a, it can be seen that the time of one bit (tb) is the same as the time the FSK output is a
mark of space frequency (ts). Thus, the bit time equals the time of an FSK signaling element, and
the bit rate equals the baud.

The baud for binary FSK can also be determined by substituting N = 1 in Equation 2.11:

baud = fb / 1 = fb
The minimum bandwidth for FSK is given as
B= |(fs – fb) – (fm – fb)|

=|(fs– fm)| + 2fb

and since |(fs– fm)| equals 2 f, the minimum bandwidth can be approximated as
B= 2( f + fb) (2.15)

where
B= minimum Nyquist bandwidth (hertz)
f= frequency deviation |(fm– fs)| (hertz)
fb = input bit rate (bps)

Example 2-2

Determine (a) the peak frequency deviation, (b) minimum bandwidth, and (c) baud for a binary
FSK signal with a mark frequency of 49 kHz, a space frequency of 51 kHz, and an input bit rate of
2 kbps.

Solution

a. The peak frequency deviation is determined from Equation 2.14:

f= |149kHz - 51 kHz| / 2 =1 kHz

b. The minimum bandwidth is determined from Equation 2.15:
B = 2(100+ 2000)
=6 kHz
c. For FSK, N = 1, and the baud is determined from Equation 2.11 as
baud = 2000 / 1 = 2000

FSK TRANSMITTER:
Figure 2-6 shows a simplified binary FSK modulator, which is very similar to a conventional FM
modulator and is very often a voltage-controlled oscillator (VCO).The center frequency (fc) is
chosen such that it falls halfway between the mark and space frequencies.

A logic 1 input shifts the VCO output to the mark frequency, and a logic 0 input shifts the VCO
output to the space frequency. Consequently, as the binary input signal changes back and forth
between logic 1 and logic 0 conditions, the VCO output shifts or deviates back and forth between
the mark and space frequencies.

FIGURE 2-6 FSK modulator

A VCO-FSK modulator can be operated in the sweep mode where the peak frequency deviation is
simply the product of the binary input voltage and the deviation sensitivity of the VCO.
With the sweep mode of modulation, the frequency deviation is expressed mathematically as
f = vm(t)kl (2-19)

vm(t) = peak binary modulating-signal voltage (volts)

kl = deviation sensitivity (hertz per volt).
FSK Receiver
FSK demodulation is quite simple with a circuit such as the one shown in Figure 2-7.

FIGURE 2-7 Noncoherent FSK demodulator

The FSK input signal is simultaneously applied to the inputs of both bandpass filters (BPFs)
through a power splitter.The respective filter passes only the mark or only the space frequency on to
its respective envelope detector.The envelope detectors, in turn, indicate the total power in each
passband, and the comparator responds to the largest of the two powers.This type of FSK detection
is referred to as noncoherent detection.

Figure 2-8 shows the block diagram for a coherent FSK receiver.The incoming FSK signal is
multiplied by a recovered carrier signal that has the exact same frequency and phase as the
transmitter reference.

However, the two transmitted frequencies (the mark and space frequencies) are not generally
continuous; it is not practical to reproduce a local reference that is coherent with both of them.
Consequently, coherent FSK detection is seldom used.

FIGURE 2-8 Coherent FSK demodulator

PHASESHIFT KEYING:
The phase of the output signal gets shifted depending upon the input. These are mainly of two
types, namely BPSK and QPSK, according to the number of phase shifts. The other one is DPSK
which changes the phase according to the previous value.

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

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

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°.

BPSK is basically a DSB-SC (Double Sideband Suppressed Carrier) modulation scheme, for
message being the digital information.

Following is the image of BPSK Modulated output wave along with its input.
Binary Phase-Shift Keying
The simplest form of PSK is binary phase-shift keying (BPSK), where N = 1 and M =
2.Therefore, with BPSK, two phases (21 = 2) are possible for the carrier.One phase represents a
logic 1, and the other phase represents a logic 0. As the input digital signal changes state (i.e., from
a 1 to a 0 or from a 0 to a 1), the phase of the output carrier shifts between two angles that are
separated by 180°.

Hence, other names for BPSK are phase reversal keying (PRK) and biphase modulation. BPSK
is a form of square-wave modulation of a continuous wave (CW) signal.

FIGURE 2-12 BPSK transmitter

BPSK TRANSMITTER:

Figure 2-12 shows a simplified block diagram of a BPSK transmitter. The balanced modulator acts
as a phase reversing switch. Depending on the logic condition of the digital input, the carrier is
transferred to the output either in phase or 180° out of phase with the reference carrier oscillator.

Figure 2-13 shows the schematic diagram of a balanced ring modulator. The balanced modulator
has two inputs: a carrier that is in phase with the reference oscillator and the binary digital data. For
the balanced modulator to operate properly, the digital input voltage must be much greater than the
peak carrier voltage.

This ensures that the digital input controls the on/off state of diodes D1 to D4. If the binary input is
a logic 1(positive voltage), diodes D 1 and D2 are forward biased and on, while diodes D3 and D4
are reverse biased and off (Figure 2-13b). With the polarities shown, the carrier voltage is
developed across transformer T2 in phase with the carrier voltage across T

1. Consequently, the output signal is in phase with the reference oscillator.

If the binary input is a logic 0 (negative voltage), diodes Dl and D2 are reverse biased and off,
while diodes D3 and D4 are forward biased and on (Figure 9-13c). As a result, the carrier voltage is
developed across transformer T2 180° out of phase with the carrier voltage across T 1.

FIGURE 9-13 (a) Balanced ring modulator; (b) logic 1 input; (c) logic 0 input
 FIGURE 2-14 BPSK modulator: (a) truth table; (b) phasor diagram; (c) constellation
diagram
BANDWIDTH CONSIDERATIONS OF BPSK:

In a BPSK modulator. the carrier input signal is multiplied by the binary data.

If + 1 V is assigned to a logic 1 and -1 V is assigned to a logic 0, the input carrier (sin ωct) is
multiplied by either a + or - 1 .

The output signal is either + 1 sin ωct or -1 sin ωct the first represents a signal that is in phase with
the reference oscillator, the latter a signal that is 180° out of phase with the reference
oscillator.Each time the input logic condition changes, the output phase changes.
Mathematically, the output of a BPSK modulator is proportional to

BPSK output = [sin (2πfat)] x [sin (2πfct)] (2.20)

where
fa = maximum fundamental frequency of binary input (hertz)
fc = reference carrier frequency (hertz)

Solving for the trig identity for the product of two sine functions,

0.5cos[2π(fc – fa)t] – 0.5cos[2π(fc + fa)t]

Thus, the minimum double-sided Nyquist bandwidth (B) is

fc + fa fc + fa
-fc + fa
-(fc + fa) or
2fa
and because fa = fb / 2, where fb = input bit rate,

where B is the minimum double-sided Nyquist bandwidth.

Figure 2-15 shows the output phase-versus-time relationship for a BPSK waveform. Logic 1 input
produces an analog output signal with a 0° phase angle, and a logic 0 input produces an analog
output signal with a 180° phase angle.

As the binary input shifts between a logic 1 and a logic 0 condition and vice versa, the phase of the
BPSK waveform shifts between 0° and 180°, respectively.

BPSK signaling element (ts) is equal to the time of one information bit (tb), which indicates that the
bit rate equals the baud.

FIGURE 2-15 Output phase-versus-time relationship for a BPSK modulator

Example:

For a BPSK modulator with a carrier frequency of 70 MHz and an input bit rate of 10 Mbps,
determine the maximum and minimum upper and lower side frequencies, draw the output spectrum,
de-termine the minimum Nyquist bandwidth, and calculate the baud..

Solution

Substituting into Equation 2-20 yields

output = [sin (2πfat)] x [sin (2πfct)]; fa = fb / 2 = 5 MHz

=[sin 2π(5MHz)t)] x [sin 2π(70MHz)t)]

=0.5cos[2π(70MHz – 5MHz)t] – 0.5cos[2π(70MHz + 5MHz)t]
lower side frequency upper side frequency

Minimum lower side frequency (LSF):

LSF=70MHz - 5MHz = 65MHz

Maximum upper side frequency (USF):

USF = 70 MHz + 5 MHz = 75 MHz

Therefore, the output spectrum for the worst-case binary input conditions is as follows: The
minimum Nyquist bandwidth (B) is

B = 75 MHz - 65 MHz = 10 MHz

and the baud = fb or 10 megabaud.

BPSK receiver:.
Figure 2-16 shows the block diagram of a BPSK receiver.
The input signal maybe+ sin ωct or - sin ωct .The coherent carrier recovery circuit detects and
regenerates a carrier signal that is both frequency and phase coherent with the original transmit
carrier.
The balanced modulator is a product detector; the output is the product d the two inputs (the BPSK
signal and the recovered carrier).
The low-pass filter (LPF) operates the recovered binary data from the complex demodulated signal.

FIGURE 2-16 Block diagram of a BPSK receiver

Mathematically, the demodulation process is as follows.

For a BPSK input signal of + sin ωct (logic 1), the output of the balanced modulator is
output = (sin ωct )(sin ωct) = sin2ωct (2.21)
or sin2ωct = 0.5(1 – cos 2ωct) = 0.5 - 0.5cos 2ωct
filtered out
leaving output = + 0.5 V = logic 1
It can be seen that the output of the balanced modulator contains a positive voltage (+[1/2]V) and a
cosine wave at twice the carrier frequency (2 ωct ).

The LPF has a cutoff frequency much lower than 2 ωct, and, thus, blocks the second harmonic of
the carrier and passes only the positive constant component. A positive voltage represents a
demodulated logic 1.

For a BPSK input signal of -sin ωct (logic 0), the output of the balanced modulator is

output = (-sin ωct )(sin ωct) = sin2ωct

or
sin2ωct = -0.5(1 – cos 2ωct) = 0.5 + 0.5cos 2ωct

filtered out
leaving
output = - 0.5 V = logic 0
The output of the balanced modulator contains a negative voltage (-[l/2]V) and a cosine wave at
twice the carrier frequency (2ωct).
Again, the LPF blocks the second harmonic of the carrier and passes only the negative constant
component. A negative voltage represents a demodulated logic 0.

QUADRATURE PHASE SHIFT KEYING (QPSK):

This is the phase shift keying technique, in which the sine wave carrier takes four phase reversals
such as 0°, 90°, 180°, and 270°.

If this kind of techniques are further extended, PSK can be done by eight or sixteen values also,
depending upon the requirement. The following figure represents the QPSK waveform for two bits
input, which shows the modulated result for different instances of binary inputs.

QPSK is a variation of BPSK, and it is also a DSB-SC (Double Sideband Suppressed Carrier)
modulation scheme, which sends two bits of digital information at a time, called as bigits.
Instead of the conversion of digital bits into a series of digital stream, it converts them into bit-pairs.
This decreases the data bit rate to half, which allows space for the other users.

QPSK transmitter.
A block diagram of a QPSK modulator is shown in Figure 2-17Two bits (a dibit) are
clocked into the bit splitter. After both bits have been serially inputted, they are simultaneously
parallel outputted.
The I bit modulates a carrier that is in phase with the reference oscillator (hence the name "I" for "in
phase" channel), and theQ bit modulate, a carrier that is 90° out of phase.

For a logic 1 = + 1 V and a logic 0= - 1 V, two phases are possible at the output of the I balanced
modulator (+sin ωct and - sin ωct), and two phases are possible at the output of the Q balanced
modulator (+cos ωct), and (-cos ωct).

When the linear summer combines the two quadrature (90° out of phase) signals, there are four
possible resultant phasors given by these expressions: + sin ωct + cos ωct, + sin ωct - cos ωct, -sin
ωct + cos ωct, and -sin ωct - cos ωct.

Example:
For the QPSK modulator shown in Figure 2-17, construct the truthtable, phasor diagram, and
constellation diagram.

Solution

For a binary data input of Q = O and I= 0, the two inputs to the Ibalanced modulator are -1 and sin
ωct, and the two inputs to the Q balanced modulator are -1 and cos ωct.

Consequently, the outputs are

I balanced modulator =(-1)(sin ωct) = -1 sin ωct

Q balanced modulator =(-1)(cos ωct) = -1 cos ωct and the output of the linear summer is
-1 cos ωct - 1 sin ωct = 1.414 sin(ωct - 135°)

For the remaining dibit codes (01, 10, and 11), the procedure is the same. The results are shown in
Figure 2-18a.

FIGURE 2-18 QPSK modulator: (a) truth table; (b) phasor diagram; (c) constellation
diagram
In Figures 2-18b and c, it can be seen that with QPSK each of the four possible output phasors has
exactly the same amplitude. Therefore, the binary information must be encoded entirely in the
phase of the output signal
Figure 2-18b, it can be seen that the angular separation between any two adjacent phasors in QPSK
is 90°.Therefore, a QPSK signal can undergo almost a+45° or -45° shift in phase during
transmission and still retain the correct encoded information when demodulated at the receiver.

Figure 2-19 shows the output phase-versus-time relationship for a QPSK modulator.

FIGURE 2-19 Output phase-versus-time relationship for a PSK modulator

Bandwidth considerations of QPSK

With QPSK, because the input data are divided into two channels, the bit rate in either the I or the Q
channel is equal to one-half of the input data rate (fb/2) (one-half of fb/2 = fb/4).

QPSK RECEIVER:

The block diagram of a QPSK receiver is shown in Figure 2-21

The power splitter directs the input QPSK signal to the I and Q product detectors and the carrier
recovery circuit. The carrier recovery circuit reproduces the original transmit carrier oscillator
signal. The recovered carrier must be frequency and phase coherent with the transmit reference
carrier. The QPSK signal is demodulated in the I and Q product detectors, which generate the
original I and Q data bits. The outputs of the product detectors are fed to the bit combining circuit,
where they are converted from parallel I and Q data channels to a single binary output data stream.
The incoming QPSK signal may be any one of the four possible output phases shown in Figure 2-
18. To illustrate the demodulation process, let the incoming QPSK signal be -sin ωct + cos ωct.
Mathematically, the demodulation process is as follows.
 FIGURE 2-21 QPSK receiver

The receive QPSK signal (-sin ωct + cos ωct) is one of the inputs to the I product detector. The
other input is the recovered carrier (sin ωct). The output of the I product detector is

Again, the receive QPSK signal (-sin ωct + cos ωct) is one of the inputs to the Q product detector.
The other input is the recovered carrier shifted 90° in phase (cos ωct). The output of the Q product
detector is
The demodulated I and Q bits (0 and 1, respectively) correspond to the constellation diagram and
truth table for the QPSK modulator shown in Figure 2-18.

DIFFERENTIAL PHASE SHIFT KEYING (DPSK):

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

It is seen from the above figure that, if the data bit is LOW i.e., 0, then the phase of the signal is not
reversed, but is continued as it was. If the data is HIGH i.e., 1, then the phase of the signal is
reversed, as with NRZI, invert on 1 (a form of differential encoding).
If we observe the above waveform, we can say that the HIGH state represents an M in the
modulating signal and the LOW state represents a W in the modulating signal.

The word binary represents two-bits. M simply represents a digit that corresponds to the number of
conditions, levels, or combinations possible for a given number of binary variables.
This is the type of digital modulation technique used for data transmission in which instead of one-
bit, two or more bits are transmitted at a time. As a single signal is used for multiple bit
transmission, the channel bandwidth is reduced.
DBPSK TRANSMITTER.:
Figure 2-37a shows a simplified block diagram of a differential binary phase-shift keying
(DBPSK) transmitter. An incoming information bit is XNORed with the preceding bit prior to
entering the BPSK modulator (balanced modulator).

For the first data bit, there is no preceding bit with which to compare it. Therefore, an initial
reference bit is assumed. Figure 2-37b shows the relationship between the input data, the XNOR
output data, and the phase at the output of the balanced modulator. If the initial reference bit is
assumed a logic 1, the output from the XNOR circuit is simply the complement of that shown.

In Figure 2-37b, the first data bit is XNORed with the reference bit. If they are the same, the XNOR
output is a logic 1; if they are different, the XNOR output is a logic 0. The balanced modulator
operates the same as a conventional BPSK modulator; a logic I produces +sin ωct at the output, and
A logic 0 produces –sin ωct at the output.
 FIGURE 2-37 DBPSK modulator (a) block diagram (b) timing diagram

BPSK RECEIVER:
Figure 9-38 shows the block diagram and timing sequence for a DBPSK receiver. The received
signal is delayed by one bit time, then compared with the next signaling element in the balanced
modulator. If they are the same. J logic 1(+ voltage) is generated. If they are different, a logic 0 (-
voltage) is generated. [f the reference phase is incorrectly assumed, only the first demodulated bit is
in error. Differential encoding can be implemented with higher-than-binary digital modulation
schemes, although the differential algorithms are much more complicated than for DBPS K.

The primary advantage of DBPSK is the simplicity with which it can be implemented. With
DBPSK, no carrier recovery circuit is needed. A disadvantage of DBPSK is, that it requires
between 1 dB and 3 dB more signal-to-noise ratio to achieve the same bit error rate as that of
absolute PSK.
 FIGURE 2-38 DBPSK demodulator: (a) block diagram; (b) timing sequence

COHERENT RECEPTION OF FSK:

The coherent demodulator for the coherent FSK signal falls in the general form of coherent
demodulators described in Appendix B. The demodulator can be implemented with two correlators
as shown in Figure 3.5, where the two reference signals are cos(27r f t) and cos(27r fit). They must
be synchronized with the received signal. The receiver is optimum in the sense that it minimizes the
error probability for equally likely binary signals. Even though the receiver is rigorously derived in
Appendix B, some heuristic explanation here may help understand its operation. When s 1 (t) is
transmitted, the upper correlator yields a signal 1 with a positive signal component and a noise
component. However, the lower correlator output 12, due to the signals' orthogonality, has only a
noise component. Thus the output of the summer is most likely above zero, and the threshold
detector will most likely produce a 1. When s2(t) is transmitted, opposite things happen to the two
correlators and the threshold detector will most likely produce a 0. However, due to the noise nature
that its values range from -00 to m, occasionally the noise amplitude might overpower the signal
amplitude, and then detection errors will happen. An alternative to Figure 3.5 is to use just one
correlator with the reference signal cos (27r f t) - cos(2s f2t) (Figure 3.6). The correlator in Figure
can be replaced by a matched filter that matches cos(27r fit) - cos(27r f2t) (Figure 3.7). All
implementations are equivalent in terms of error performance (see Appendix B). Assuming an
AWGN channel, the received signal is

where n(t) is the additive white Gaussian noise with zero mean and a two-sided power spectral
density A',/2. From (B.33) the bit error probability for any equally likely binary signals is

where No/2 is the two-sided power spectral density of the additive white Gaussian noise. For
Sunde's FSK signals El = Ez = Eb, pI2 = 0 (orthogonal). thus the error probability is

where Eb = A2T/2 is the average bit energy of the FSK signal. The above Pb is plotted in Figure 3.8
where Pb of noncoherently demodulated FSK, whose expression will be given shortly, is also
plotted for comparison.
 Figure: Pb of coherently and non-coherently demodulated FSK signal.

NONCOHERENT DEMODULATION AND ERROR PERFORMANCE:

Coherently FSK signals can be noncoherently demodulated to avoid the carrier recovery.
Noncoherently generated FSK can only be noncoherently demodulated. We refer to both cases as
noncoherent FSK. In both cases the demodulation problem becomes a problem of detecting signals
with unknown phases. In Appendix B we have shown that the optimum receiver is a quadrature
receiver. It can be implemented using correlators or equivalently, matched filters. Here we assume
that the binary noncoherent FSK signals are equally likely and with equal energies. Under these
assumptions, the demodulator using correlators is shown in Figure 3.9. Again, like in the coherent
case, the optimality of the receiver has been rigorously proved (Appendix B). However, we can
easily understand its operation by some heuristic argument as follows. The received signal
(ignoring noise for the moment) with an unknown phase can be written as
The signal consists of an in phase component A cos 8 cos 27r f t and a quadrature component A sin
8 sin 2x f,t sin 0. Thus the signal is partially correlated with cos 2s fit and partiah'y correlated with
sin 27r fit. Therefore we use two correlators to collect the signal energy in these two parts. The
outputs of the in phase and quadrature correlators will be cos 19 and sin 8, respectively. Depending
on the value of the unknown phase 8, these two outputs could be anything in (- 5, y). Fortunately
the squared sum of these two signals is not dependent on the unknown phase. That is

This quantity is actually the mean value of the statistics I? when signal si (t) is transmitted and noise
is taken into consideration. When si (t) is not transmitted the mean value of 1: is 0. The comparator
decides which signal is sent by checking these I?. The matched filter equivalence to Figure 3.9 is
shown in Figure 3.10 which has the same error performance. For implementation simplicity we can
replace the matched filters by bandpass filters centered at f and fi, respectively (Figure 3.1 1).
However, if the bandpass filters are not matched to the FSK signals, degradation to

various extents will result. The bit error probability can be derived using the correlator demodulator
(Appendix B). Here we further assume that the FSK signals are orthogonal, then from Appendix B
the error probability is
 UNIT – V (b)
Information Theory
Information Theory

Information theory deals with representation and the transfer of information.

There are two fundamentally different ways to transmit messages: via discrete signals
and via continuous signals .....For example, the letters of the English alphabet are commonly
thought of as discrete signals.
Information sources

Definition:

The set of source symbols is called the source alphabet, and the elements of the set are
called the symbols or letters.
The number of possible answers ‘ r ’ should be linked to “information.”
“Information” should be additive in some sense.
We define the following measure of information:

Where ‘ r ’ is the number of all possible outcome so far an do m message U.

Using this definition we can confirm that it has the wanted property of additivity:

The basis ‘b’ of the logarithm b is only a change of units without actually changing the
amount of information it describes.

Classification of information sources

1. Discrete memory less.

2. Memory.

Discrete memory less source (DMS) can be characterized by “the list of the symbols, the
probability assignment to these symbols, and the specification of the rate of generating these
symbols by the source”.
1. Information should be proportion to the uncertainty of an outcome.
2. Information contained in independent outcome should add.
Scope of Information Theory

1. Determine the irreducible limit below which a signal cannot be compressed.

2. Deduce the ultimate transmission rate for reliable communication over a noisy channel.
3. Define Channel Capacity - the intrinsic ability of a channel to convey information.

The basic setup in Information Theory has:

– a source,
– a channel and
– destination.
The output from source is conveyed through the channel and received at the destination.
The source is a random variable S
which takes symbols from a finite alphabet i.e.,

S = {s0, s1, s2, ・ ・・, sk−1}

With probabilities

P(S = sk) = pk where k = 0, 1, 2, ・ ・・, k − 1

and
k−1,Xk=0 ,pk = 1

The following assumptions are made about the source

1. Source generates symbols that are statistically independent.

2. Source is memory less i.e., the choice of present symbol does not depend on the previous
choices.

Properties of Information

1. Information conveyed by a deterministic event is nothing

2. Information is always positive.
3. Information is never lost.
4. More information is conveyed by a less probable event than a more probable event

Entropy:
The Entropy (H(s)) of a source is defined as the average information generated by a
discrete memory less source.

Information content of a symbol:

Let us consider a discrete memory less source (DMS) denoted by X and having the
alphabet {U1, U2, U3, ……Um}. The information content of the symbol xi, denoted by I(xi) is
defined as

I (U) = log b = - log b P(U)

Where P (U) is the probability of occurrence of symbol U

Units of I(xi):

For two important and one unimportant special cases of b it has been agreed to use the
following names for these units:
b =2(log2): bit,

b = e (ln): nat (natural logarithm),

b =10(log10): Hartley.

The conversation of these units to other units is given as

log2a=

Uncertainty or Entropy (i.e Average information)

Definition:

In order to get the information content of the symbol, the flow information on the
symbol can fluctuate widely because of randomness involved into the section of symbols.

The uncertainty or entropy of a discrete random variable (RV) ‘U’ is defined as

H(U)= E[I(u)]=
 Where PU (·) denotes the probability mass function (PMF) 2 of the RV U, and where
the support of P U is defined as

We will usually neglect to mention “support” when we sum over PU (u) · log b PU (u), i.e., we
implicitly assume that we exclude all u
With zero probability PU (u) =0.

Entropy for binary source

It may be noted that for a binary source U which genets independent symbols 0 and 1
with equal probability, the source entropy H (u) is

H (u) = - log2 - log2 = 1 b/symbol

Bounds on H (U)

If U has r possible values, then 0 ≤ H(U) ≤ log r,

Where

H(U)=0 if, and only if, PU(u)=1 for some u,

H(U)=log r if, and only if, PU(u)= 1/r ∀ u.

Hence, H(U) ≥ 0.Equalitycanonlybeachievedif −PU(u)log2 PU(u)=0

For all u ∈ supp (PU), i.e., PU (u) =1forall u ∈ supp (PU).

To derive the upper bound we use at rick that is quite common in.

Formation theory: We take the deference and try to show that it must be non positive.
Equality can only be achieved if

1. In the IT Inequality ξ =1,i.e.,if 1r·PU(u)=1=⇒ PU(u)= 1r ,for all u;

2. |supp (PU)| = r.

Note that if Condition1 is satisfied, Condition 2 is also satisfied.

 Conditional Entropy

Similar to probability of random vectors, there is nothing really new about conditional
probabilities given that a particular event Y = y has occurred.
The conditional entropy or conditional uncertainty of the RV X given the event Y = y is
defined as

Note that the definition is identical to before apart from that everything is conditioned
on the event Y = y

Note that the conditional entropy given the event Y = y is a function of y. Since Y is
also a RV, we can now average over all possible events Y = y according to the probabilities
of each event. This will lead to the averaged.

Mutual Information

Although conditional entropy can tell us when two variables are completely
independent, it is not an adequate measure of dependence. A small value for H(Y| X) may
implies that X tells us a great deal about Y or that H(Y) is small to begin with. Thus, we
measure dependence using mutual information:

I(X,Y) =H(Y)–H(Y|X)

Mutual information is a measure of the reduction of randomness of a variable given

knowledge of another variable. Using properties of logarithms, we can derive several equiva-
lent definitions
 I(X,Y)=H(X)–H(X| Y)

I(X,Y) = H(X)+H(Y)–H(X,Y) = I(Y,X)

In addition to the definitions above, it is useful to realize that mutual information is a

particular case of the Kullback-Leibler divergence. The KL divergence is defined as:

KL divergence measures the difference between two distributions. It is sometimes called the
relative entropy. It is always non-negative and zero only when p=q; however, it is not a
distance because it is not symmetric.

In terms of KL divergence, mutual information is:

In other words, mutual information is a measure of the difference between the joint
probability and product of the individual probabilities. These two distributions are equivalent
only when X and Y are independent, and diverge as X and Y become more dependent.

Source coding
Coding theory is the study of the properties of codes and their respective fitness for
specific applications. Codes are used for data compression, cryptography, error-
correction, and networking. Codes are studied by various scientific disciplines—such as
information theory, electrical engineering, mathematics, linguistics, and computer
science—for the purpose of designing efficient and reliable data transmission methods.
This typically involves the removal of redundancy and the correction or detection of
errors in the transmitted data.
The aim of source coding is to take the source data and make it smaller.

All source models in information theory may be viewed as random process or random
sequence models. Let us consider the example of a discrete memory less source
(DMS), which is a simple random sequence model.

A DMS is a source whose output is a sequence of letters such that each letter is
independently selected from a fixed alphabet consisting of letters; say a1, a2 ,
 ……….ak. The letters in the source output sequence are assumed to be random
and statistically

Independent of each other. A fixed probability assignment for the occurrence of

each letter is also assumed. Let us, consider a small example to appreciate the
importance of probability assignment of the source letters.

Let us consider a source with four letters a1, a2, a3 and a4 with P(a1)=0.5,

P(a2)=0.25, P(a3)= 0.13, P(a4)=0.12. Let us decide to go for binary coding of these

four

Source letters While this can be done in multiple ways, two encoded representations
are shown below:

Code Representation#1:

a1: 00, a2:01, a3:10, a4:11

Code Representation#2:

a1: 0, a2:10, a3:001, a4:110

It is easy to see that in method #1 the probability assignment of a source letter has not
been considered and all letters have been represented by two bits each. However in

The second method only a1 has been encoded in one bit, a2 in two bits and the
remaining two in three bits. It is easy to see that the average number of bits to be used
per source letter for the two methods is not the same. ( a for method #1=2 bits per
letter and a for method #2 < 2 bits per letter). So, if we consider the issue of encoding
a long sequence of

Letters we have to transmit less number of bits following the second method. This
is an important aspect of source coding operation in general. At this point, let us
note
a) We observe that assignment of small number of bits to more probable letters and
assignment of larger number of bits to less probable letters (or symbols) may lead to
efficient source encoding scheme.
 b) However, one has to take additional care while transmitting the encoded letters. A
careful inspection of the binary representation of the symbols in method #2 reveals
that it may lead to confusion (at the decoder end) in deciding the end of binary
representation of a letter and beginning of the subsequent letter.

So a source-encoding scheme should ensure that

1) The average number of coded bits (or letters in general) required per source letter
is as small as possible and
2) The source letters can be fully retrieved from a received encoded sequence.

Shannon-Fano Code

Shannon–Fano coding, named after Claude Elwood Shannon and Robert Fano, is a
technique for constructing a prefix code based on a set of symbols and their probabilities. It is
suboptimal in the sense that it does not achieve the lowest possible expected codeword length
like Huffman coding; however unlike Huffman coding, it does guarantee that all codeword
lengths are within one bit of their theoretical ideal I(x) =−log P(x).

In Shannon–Fano coding, the symbols are arranged in order from most probable to least
probable, and then divided into two sets whose total probabilities are as close as possible to
being equal. All symbols then have the first digits of their codes assigned; symbols in the first
set receive "0" and symbols in the second set receive "1". As long as any sets with more than
one member remain, the same process is repeated on those sets, to determine successive
digits of their codes. When a set has been reduced to one symbol, of course, this means the
symbol's code is complete and will not form the prefix of any other symbol's code.

The algorithm works, and it produces fairly efficient variable-length encodings; when the two
smaller sets produced by a partitioning are in fact of equal probability, the one bit of
information used to distinguish them is used most efficiently. Unfortunately, Shannon–Fano
does not always produce optimal prefix codes.

For this reason, Shannon–Fano is almost never used; Huffman coding is almost as
computationally simple and produces prefix codes that always achieve the lowest expected
code word length. Shannon–Fano coding is used in the IMPLODE compression method,
which is part of the ZIP file format, where it is desired to apply a simple algorithm with high
performance and minimum requirements for programming.
Shannon-Fano Algorithm:
A Shannon–Fano tree is built according to a specification designed to define an
effective code table. The actual algorithm is simple:
For a given list of symbols, develop a corresponding list of probabilities or frequency
counts so that each symbol’s relative frequency of occurrence is known.

Sort the lists of symbols according to frequency, with the most frequently
occurring
Symbols at the left and the least common at the right.
Divide the list into two parts, with the total frequency counts of the left part being
as
Close to the total of the right as possible.
The left part of the list is assigned the binary digit 0, and the right part is assigned
the digit 1. This means that the codes for the symbols in the first part will all start
with 0, and the codes in the second part will all start with 1.

Recursively apply the steps 3 and 4 to each of the two halves, subdividing groups
and adding bits to the codes until each symbol has become a corresponding code leaf
on the tree.

Example:
The source of information A generates the symbols {A0, A1, A2, A3 and A4} with the
corresponding probabilities {0.4, 0.3, 0.15, 0.1 and 0.05}. Encoding the source symbols
using binary encoder and Shannon-Fano encoder gives

Source Symbol Pi Binary Code Shannon-Fano

A0 0.4 000 0
A1 0.3 001 10
A2 0.15 010 110
A3 0.1 011 1110
A4 0.05 100 1111
Lavg H = 2.0087 3 2.05
Shanon-Fano code is a top-down approach. Constructing the code tree, we get

14
 Binary Huffman Coding (an optimum variable-length source coding scheme)
In Binary Huffman Coding each source letter is converted into a binary code
word. It is a prefix condition code ensuring minimum average length per source letter in
bits.
Let the source letters a1, a 2, ……….aK have probabilities P(a1), P(a2),………….
P(aK) and let us assume that P(a1) ≥ P(a2) ≥ P(a 3)≥…. ≥ P(aK).

We now consider a simple example to illustrate the steps for Huffman coding.

Steps to calculate Huffman Coding

Example Let us consider a discrete memory less source with six letters having

P(a1)=0.3,P(a2)=0.2, P(a 3)=0.15, P(a 4)=0.15, P(a5)=0.12 and P(a6)=0.08.

Arrange the letters in descending order of their probability (here they are
arranged).
Consider the last two probabilities. Tie up the last two probabilities. Assign, say, 0
to the last digit of representation for the least probable letter (a 6) and 1 to the last
digit of representation for the second least probable letter (a5). That is, assign ‘1’
to the upper arm of the tree and ‘0’ to the lower arm.

(3) Now, add the two probabilities and imagine a new letter, say b1, substituting for a6
and a5. So P(b1) =0.2. Check whether a4 and b1are the least likely letters. If not,
reorder the letters as per Step#1 and add the probabilities of two least likely letters.
For our example, it leads to:
P(a1)=0.3, P(a2)=0.2, P(b1)=0.2, P(a3)=0.15 and P(a4)=0.15
 (4) Now go to Step#2 and start with the reduced ensemble consisting of a1 , a2 , a3 ,

a4 and b1. Our example results in:

Here we imagine another letter b1, with P(b2)=0.3.

Continue till the first digits of the most reduced ensemble of two letters are
assigned a ‘1’ and a ‘0’.

Again go back to the step (2): P(a1)=0.3, P(b2)=0.3, P(a2)=0.2 and P(b1)=0.2.
Now we consider the last two probabilities:

So, P(b3)=0.4. Following Step#2 again, we get, P(b3)=0.4, P(a1)=0.3 and

P(b2)=0.3.
Next two probabilities lead to:

With P(b4) = 0.6. Finally we get only two probabilities

 6. Now, read the code tree inward, starting from the root, and construct the
code words. The first digit of a codeword appears first while reading the code tree
inward.

Hence, the final representation is: a1=11, a2=01, a3=101, a4=100, a5=001, a6=000.
A few observations on the preceding example

1. The event with maximum probability has least number of bits

2. Prefix condition is satisfied. No representation of one letter is prefix for other.

Prefix condition says that representation of any letter should not be a part of any
other letter.

3. Average length/letter (in bits) after coding is

= ∑P (ai )ni = 2.5 bits/letter.

4. Note that the entropy of the source is: H(X)=2.465 bits/symbol. Average length
per source letter after Huffman coding is a little bit more but close to the source
entropy. In fact, the following celebrated theorem due to C. E. Shannon sets the
limiting value of average length of code words from a DMS.

Shannon–Hartley theorem

In information theory, the Shannon–Hartley theorem tells the maximum rate at which
information can be transmitted over a communications channel of a specified bandwidth in
the presence of noise. It is an application of the noisy-channel coding theorem to the
archetypal case of a continuous-time analog communications channel subject to Gaussian
noise. The theorem establishes Shannon's channel capacity for such a communication link, a
bound on the maximum amount of error-free information per time unit that can be transmitted
with a specified bandwidth in the presence of the noise interference, assuming that the signal
power is bounded, and that the Gaussian noise process is characterized by a known power or
power spectral density.
The law is named after Claude Shannon and Ralph Hartley.

Hartley Shannon Law

The theory behind designing and analyzing channel codes is called Shannon’s noisy
channel coding theorem. It puts an upper limit on the amount of information you can
send in a noisy channel using a perfect channel code. This is given by the following
equation:

where C is the upper bound on the capacity of the channel (bit/s), B is the
bandwidth of the channel (Hz) and SNR is the Signal-to-Noise ratio (unit less).

Bandwidth-S/N Tradeoff

The expression of the channel capacity of the Gaussian channel makes intuitive
sense:

1. As the bandwidth of the channel increases, it is possible to make faster

changes in the information signal, thereby increasing the information rate.

2 As S/N increases, one can increase the information rate while still preventing errors
due to noise.

3. For no noise, S/N tends to infinity and an infinite information rate is

possible irrespective of bandwidth.

Thus we may trade off bandwidth for SNR. For example, if S/N = 7 and B = 4kHz,
then the channel capacity is C = 12 ×103 bits/s. If the SNR increases to S/N = 15 and B
is decreased to 3kHz, the channel capacity remains the same. However, as B tends to
1, the channel capacity does not become infinite since, with an increase in bandwidth,
the noise power also increases. If the noise power spectral density is ɳ/2, then the total
noise power is N = ɳB, so the Shannon-Hartley law becomes