IV Sem EC Analog & Digital Communication Summer-2023

Que. 1 (a)
What is modulation? What is the need of modulation in a communication
system? Explain in detail.

Solution :
Definition of Modulation:

Modulation is the process of varying one or more properties of a carrier

signal (such as amplitude, frequency, or phase) according to the informa-
tion signal to be transmitted. The carrier signal is usually a high-frequency
signal, and the information signal is a low-frequency message signal (such
as voice, data, or video).

Types of Modulation: There are three primary types of modulation:

1. Amplitude Modulation (AM) – The amplitude of the carrier wave is varied

in proportion to the message signal.

2. Frequency Modulation (FM) – The frequency of the carrier wave is varied

in proportion to the message signal.

3. Phase Modulation (PM) – The phase of the carrier wave is varied in pro-
portion to the message signal.

Need for Modulation in a Communication System:

Modulation is essential in communication systems due to several reasons:

1. To Transmit Over Long Distances:

(a) The message signal (voice, data, or video) typically has a low frequency,
which cannot travel long distances effectively.
(b) Modulation shifts the signal to a higher frequency, allowing it to be
transmitted efficiently.

2. To Reduce Signal Loss and Interference:

(a) Low-frequency signals are more susceptible to attenuation (weakening

of signals) and interference.
(b) Modulated signals can be transmitted with higher power and reduced
interference.

L: Dr. P.R. Bokde 1 PBCOE, Nagpur

Summer-2023 Analog & Digital Communication IV Sem EC

3. To Enable Multiplexing:

(a) Multiplexing is the technique of transmitting multiple signals simulta-

neously over a single communication channel.
(b) Modulation allows different signals to be transmitted at different fre-
quencies, enabling Frequency Division Multiplexing (FDM).

4. To Improve Signal-to-Noise Ratio (SNR):

(a) High-frequency signals are less affected by noise compared to low-

frequency signals.
(b) Modulation helps in reducing the effect of external noise on the trans-
mitted signal.

5. To Match Antenna Size Requirements:

(a) The size of an antenna is inversely proportional to the frequency of the

signal (λ = c/f , where λ is wavelength, c is the speed of light, and f is
frequency).
(b) Low-frequency signals require very large antennas, which are imprac-
tical.
(c) By modulating the signal to a higher frequency, the antenna size can
be made more compact.

6. To Allow Efficient Radiation of Energy:

High-frequency modulated signals can be efficiently radiated as electro-

magnetic waves, ensuring proper signal propagation in free space.

7. To Enable Bandwidth Utilization:

(a) Different types of information signals (such as voice, video, and data)
require different bandwidths.
(b) Modulation helps in allocating specific bandwidths for various com-
munication applications.

PBCOE, Nagpur 2 L: Dr. P.R. Bokde

IV Sem EC Analog & Digital Communication Summer-2023

Que. 1 (b)
Derive an expression for amplitude modulated wave. Draw necessary
waveforms.

Solution :
Amplitude Modulation (AM) is a type of modulation technique used in commu-
nication systems to transmit information by varying the amplitude of a carrier
wave. In AM, the amplitude of the carrier signal is varied in proportion to the
amplitude of the message signal (also called the modulating signal) that contains
the information to be transmitted. The resulting modulated signal is then trans-
mitted over the communication channel.
The high frequency signal whose characteristics is changed is called the carrier
signal and the term modulating signal is used for the voltage in accordance with
which the carrier is changed.
Let the carrier signal be represented by the equation as –

vc = Vc cos(ωc t + θ) (1)

where,
vc = instantaneous value of carrier voltage.
Vc = maximum value of carrier voltage (amplitude)
ωc = angular carrier frequency = 2πfc
θ = phase angle
t = time
Let th emodulating signal voltage is given by –

vm = Vm cos(ωm t) (2)

where, ωm is angular frequency of modulating signal and Vm is the amplitude of

modulating signal.
Let the carrier voltage is –

vc = Vc cos ωc t (3)

In amplitude modulation, amplitude of the carrier does not remain constant but
varies with the instantaneous value of modulating signal and varies with respect
to time as –
V (t) = Vc + Vm cos ωm t (4)

L: Dr. P.R. Bokde 3 PBCOE, Nagpur

Summer-2023 Analog & Digital Communication IV Sem EC

where, Vm cos ωm t represents the variation in amplitude of carrier signal.

Therefore, the instantaneous value of the modulated carrier voltage is given by –

v = V (t) cos ωc t (5)

Putting equation(4) in (5), we get,

v = [Vc + Vm cos ωm t] cos ωc t (6)

 
Vm
∴ v = Vc 1 + cos ωm t cos ωc t (7)
Vc
∴ v = Vc [1 + m cos ωm t] cos ωc t (8)

Vm
where, m = Vc
is modulation index or modulation factor or depth of modulation.

∴ v = Vc cos ωc t + Vc m cos ωm t cos ωc t (9)

Expanding the above equation (), we get,

mVc
v = Vc cos ωc t + [cos(ωc + ωm )t + cos(ωc − ωm )t] (10)
2
mVc mVc
∴ v = Vc cos ωc t + cos(ωc + ωm )t + cos(ωc − ωm )t (11)
2 2

Equation(11) represents the equation of AM wave.

From equation (11), it can be concluded that the AM wave contains the following
frequency components :

1. Original carrier signal Vc cos ωc t having angular frequency ωc .

mVc
2. Upper sideband term cos(ωc +ωm )t having angular frequency (ωc +ωm ).
2
mVc
3. Lower sideband term cos(ωc −ωm )t having angular frequency (ωc −ωm ).
2

Que. 2 (a)
Explain indirect generation of FM using Armstrong method.

Solution :
Figure 1 shows the block diagram of a indirect FM system.

• In indirect method, the message signal m(t) is first passed through an inte-
grator before applying it to the phase modulator as shown in figure 1.

PBCOE, Nagpur 4 L: Dr. P.R. Bokde

IV Sem EC Analog & Digital Communication Summer-2023

• The carrier signal is generated by using crystal oscillator because it provides

very high frequency stability.
The operation of indirect method is divided into two parts as follows :

1. Generate a NBFM wave using pjase modulator

2. Using the frequency multipliers and mixer to obtain the required val-
ues of frequency deviation and modulation index (i.e. WBFM).

Figure 1: Block diagram of indirect method for generating a wide band FM signal

• In order to minimize the distortion in the phase modulator, the maximum

phase deviation or modulation index β is kept small thereby resulting in a
NBFM signal.

• Let S1 (t) be the NBFM wave, then we have,

 Z t 
S1 (t) = Ac cos 2πf1 t + 2πKf m(t)dt (12)
0

where, fc is the frequency of the crystal oscillator and Kf is the frequency

Sensitivity constant in Hz/volt.

• For a single tone modulation signal defined by -

m(t) = Am cos(2πfm t) (13)

L: Dr. P.R. Bokde 5 PBCOE, Nagpur

Summer-2023 Analog & Digital Communication IV Sem EC

Then equation 12 becomes,

S1 (t) = Ac cos [2πf1 t + β1 sin 2πfm t] (14)

where, β1 is the modulation index for single tone modulation and is kept
below 0.3 radians to minimize the distortion.
The instantaneous frequency of equation 14 is

fi (t) = f1 + Kf m(t) (15)

Generation of WBFM

• The output of the Narrowband phase modulator is then multiplied by a

frequency multiplier, producing the desired WBFM wave as shown in figure
(2).

Figure 2: Frequency Multiplier

• A frequency multiplier consists of a memoryless non-linear device followed

by a band pass filter as shown in figure 2.
The input-output relationship of such a non-linear device may be expressed
in the general form -

V (t) = a1 S1 (t) + a2 S12 (t) + ........ + an S1n (t) (16)

where, a1 , a2 , .......an are coefficients and n is the highest order of non-linearity.

Substituting equation 12 in equation 16 and simplifying, we find the fre-
quency modulated wave having carrier frequencies f1 , 2f1 , ......., nf1 with
frequency deviation △f1 , 2△f1 , ........n△f1 .
The bandpass filter has two functions to perform :

1. To pass the FM wave centered at carrier frequency nf1 and having the
frequency deviation n△f1 .
2. To supress all other FM spectra.

PBCOE, Nagpur 6 L: Dr. P.R. Bokde

IV Sem EC Analog & Digital Communication Summer-2023

Que. 2 (b)

An audio frequency signal 10 sin(2 × 500t) is used to amplitude modulation

of the carrier of 50 sin(2 × 105 t). Calculate -
1. Modulation Index
2. BW required
3. Amplitude of each sideband frequency.
4.Side band frequency
5. Total power delivered to the load of 600 Ω.

Solution :
Given Audio frequency signal is - 10 sin(2 × 500t)
Comparing with standard equation of sinusoidal signal Vm sin(ωm t), we get,
∴ Vm = 10 V ωm = 2 × 500 ∴ 2πfm = 1000
1000
∴ fm = = 159.23Hz
2π
Similarly given carrier signal - 50 sin(2×105 t) Comparing with standard equa-
tion of sinusoidal signal VC sin(ωc t), we get,
∴ VC = 50 V ωc = 2 × 105 ∴ 2πfc = 2 × 105
5
2 × 10
∴ fc = = 31.847 × 103 Hz
2π
(1)Modulation index of an AM is given by -

Vm 10
m= = = 0.2
VC 50

(2) Bandwidth required for an AM is -

Bandwidth, BW = 2fm = 2 × 159.23Hz = 318.46Hz

(3) Amplitude of each sideband frequency is given by -

mVC 0.2 × 50
VLSB = VU SB = = = 5V
2 2

(4) Sideband frequencies is given by -

fU SB = fc + fm = 31.847 × 103 + 159.23Hz = 32.006 × 103 Hz

fLSB = fc − fm = 31.847 × 103 − 159.23Hz = 31.687 × 103 Hz

(5) To calculate the total power delivered to the load of 600Ω

L: Dr. P.R. Bokde 7 PBCOE, Nagpur

Summer-2023 Analog & Digital Communication IV Sem EC

Carrier Power is given -

 2
50
√
VC2 (rms) 2
PC = = = 2.08 watts
R 600

Therefore, the total power in AM is given by -

m2
 
Pt = P C 1 +
2
(0.2)2
 
∴ Pt = 2.08 1 +
2
∴ Pt = 2.08 (1 + 0.02)
∴ Pt = 2.1216 watts

Que. 3 (a)
Explain the working of envelop detector.

Solution :
An envelop detector is a simple and highly effective device that is well suited
for the demodulation of narrow-band AM wave (i.e. the carrier frequency is
large compared with the modulating signal bandwidth) for which the percentage
modulation is less than 100 %. In an envelop detector, the output of the detector
follows the envelope of the modulated signal, hence the name envelope detector.
Figure 3 shows the envelope detector circuit.

Figure 3: Circuit of Envelope Detector

It consists of a diode and a resistor-capacitor filter. this circuit is also known

as diode detector. In the positive half cycle of AM wave, diode conducts and cur-
rent flows through R whereas, in the negative half cycle, diode is reverse biased
and no current flows through R. As a result only positive half cycle of AM wave
appears across RC as shown in figure 4. Let us see how RC filter responses to

PBCOE, Nagpur 8 L: Dr. P.R. Bokde

IV Sem EC Analog & Digital Communication Summer-2023

this positive half of AM wave. On the positive half cycle, the diode is forward bi-
ased and the capacitor charges up rapidly to peak value of the input signal when
the input signal falls below this value, the diode becomes reverse biased ad the
capacitor C discharges slowly through the load resistor R. The discharging pro-
cess continues untill the next positive half cycle. When the input signal becomes
greater than the voltage across capacitor, the diode conducts again and process is
repeated.

Figure 4: Input-Output waveforms of envelop detector

As shown in figure 3, we have assumed that the AM wave applied to the enve-
lope detector is supplied by a voltage source of internal impedance Rs . To rapidly
charge the capacitor to peak value of input signal, the charging time constant Rs C
must be short compared to the carrier period 1/fc .

1
i.e. Rs C ≪
fc

On the other hand, the discharging time constant RC must be long enough to
ensure that the capacitor discharges slowly through the load resistance R between
positive peaks of the carrier wave, but not so long that the capacitor voltage will
not discharge at the maximum rate of change of modulating wave.

1 1
i.e. ≪ RC ≪
fc W

where, W is the message bandwidth.

L: Dr. P.R. Bokde 9 PBCOE, Nagpur

Summer-2023 Analog & Digital Communication IV Sem EC

Que. 3 (b)
Define radio receiver. State and explain its performance characteristics.

Solution :

A radio receiver is an electronic device that receives radio waves transmit-

ted by a radio transmitter, extracts the desired information (such as audio,
video, or data), and converts it into a usable form. It demodulates the re-
ceived signal to recover the original message signal.

Performance Parameters of Receiver

The performance of a Radio receiver is measured on the basis of its selectivity,
sensitivity, fidelity and image frequency rejection.

1. Selectivity :
The selectivity is the ability of the receiver to select a signal of a desired fre-
quency while rejecting all others. The selectivity of the receiver is obtained
partially by RF amplifier and mainly by IF amplifiers. The selectivity shows
the attenuation that the receiver offers to signals at frequencies near to the
one to which it is tuned.

2. Sensitivity :
The ability of the receiver to pick up weak signals and amplify them is called
sensitivity. If is often defined in terms of the voltage that must be applied
to the receiver input terminals to give the standard output power, measured
at the output terminals. As the gain of the receiver is increased, sensitivity
is also increased. The sensitivity is expressed in micro volts or decibels.

3. Fidelity :
The ability of the receiver to reproduce all the range of modulating frequen-
cies equally is called fidelity of the receiver. A good fidelity requires wide
band of frequencies to be amplified. Hence for good fidelity, more band-
width of RF and IF stages is required. but this results in poor selectivity.
AM receivers are not good fidelity receivers, since bandwidth in AM is low.

4. Image Frequency Rejection :

We know that local oscillator frequency is made higher than the signal fre-
quency such that fo − fs = fi . Here fi is IF. that is fo = fs + fi . The IF

PBCOE, Nagpur 10 L: Dr. P.R. Bokde

IV Sem EC Analog & Digital Communication Summer-2023

stage passes only fi . If the frequency fsi = fs + 2fi appears at the input of
the mixer, then mixer will produce difference frequency equal to fi . This is
equal to IF. The frequency fsi is called image frequency and is defined as
the signal frequency plus twice the IF. This image frequency is converted in
the IF range and it is also amplified by IF amplifiers. This is the effect of
two stations being received simultaneously. The image frequency rejection
is done by tuned circuits in the RF stage. It depends upon the selectivity of
the RF stage. The image rejection should be done before the IF stages.

Que. 4 (a)
What do you mean by noise? Explain the types of noise.

Solution :

Definition of Noise

Noise in a communication system refers to any unwanted or random signal

that interferes with the transmission and reception of the desired message
signal. It can distort the signal, reduce clarity, and degrade the overall per-
formance of the communication system.

Types of Noise
1. External Noise : are noises whose sources are external to the receiver or
communication system. The examples of external noises are atmoshperic
noises, extra terrestrial noises and man made or industrial noise.

2. Internal Noise are noises which are generated within the system. They
include the thermal noise, short noise, transit time noise, flicker noise, par-
tition noise etc.

External Noise
1. Atmospheric Noise :
The noise generated by thunderstorms and lightening constitute this noise.
These noises, which are electrical in nature, act as spurious signals get su-
perimposed on the signals being transmitted. The receiver unable to dif-
ferentiate between these signals, picks up both signals and the signal thus

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Summer-2023 Analog & Digital Communication IV Sem EC

received is distorted in nature. But at higher frequencies the effect of this

noise is lesser.

2. Extra-terrestrial Noises :
It includes solar and cosmic noise. sun radiates electrical energy which is
spread over wide spectrum including the spectrum used for radio commu-
nication. Hence when we find that reception of signals at night is better
than in the day that is due to the disturbances caused by solar noise.
Distant stars are also suns and have high temperatures. These stars radiate
noise in the same way as our sun. The noise received from these distant
stars is thermal noise and is distributed almost uniformly over the entire
sky. The noise received from the centre of our own galaxy (milky way) and
from other galaxies also constitutes a noise called galactic noise.

3. Industrial Noises :
These noises generated due to sources such as automobiles, aircrafts, elec-
trical machines and pollution from industries etc whose sounds may create
disturbances in transmission of radio signals.

Internal Noise
(a) Thermal Noise :
This noise is the noise generated in the resistor or resistive component
of a complex impedance. This is due to rapid and random motion
of the molecules, atoms and electrons. According to kinetic theory of
thermodynamics, temperature express in internal kinetic energy. As
per this theory the K.E. due to the motion of the particles become ap-
proximately zero at absolute zero i.e. 0o K.
Therefore we say that the noise power generated in a resistor is propor-
tional to absolute temperature. This noise which depends on the mo-
tion of electrons which in turn is dependent on temeperature is called
thermal noise.
(b) Shot Noise : Normally it is assumed that the current in an electronic
device, such as diode or transistor under d.c. condition is constant at
every instant of time. Actually, the current consists of a stream of indi-
vidual electrons and holes, and it is only the time average flow which
is constant. The fluctuations in the number of electrons (or holes) con-
stitute the shot noise.

PBCOE, Nagpur 12 L: Dr. P.R. Bokde

IV Sem EC Analog & Digital Communication Summer-2023

i. Shot noise has a uniform spectrum density similar to thermal noise,

and the mean square noise current depends directly on the direct
component of current.
ii. Shot noise is also dependent upon the operating conditions of the
device.
(c) Partition Noise :
When current has to divide between two or more paths there are ran-
dom fluctuations in the division of current, a noise results which is
called the partition noise.
In bipolar transistors, there is a noise because of the random motion
of the carriers crossing emitter-base and base-collector junctions and
random recombination of holes and electrons in the base.
An emitter is divided into base and collector current and when there
are random fluctuations in the division of current between collector
and base, this noise called partition noise arises. Pentode has more
partition noise because there are five electrodes and current gets irreg-
ularly divided resulting in noise.
(d) Flicker Noise :
This noise is limited to transistors operating at low audio frequencies.
This noise arises because of the fluctuations in carrier density. Conduc-
tivity of a semiconducting material depends on carrier density. This
fluctuations may arrive due to random change in emission of electrons
flicker noise varies inversely with frequency.
(e) Transit Time Noise :
Generally, transit time is defined as the time taken by the carriers to
cross a junction. The periodic time of the signal is equal to reciprocal of
signal frequency.When signal frequency is high periodic time becomes
very small and therefore can be compared to transit time of carriers. At
such time, some of the carriers may diffuse back to source. Due to this
conductance component of input admittance increases with frequency.
A noise generator is always associated with this conductance. Since
conductance increases with frequency this noise called transit time noise
also increases with frequency. This phenomenon generally occurs at
very high frequencies in the upper VHF range and beyond.
(f) Burst Noise :
The burst noise appears as a series of bursts at two or more levels. It

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Summer-2023 Analog & Digital Communication IV Sem EC

appears in bipolar transistors and is of low frequency nature.

i. The burst noise produces popping sounds in an audio system.
Hence it is also called poncorn noise.
ii. The spectral density of burst noise increase as the frequency de-
creases.
(g) Avalanche Noise :
In the avalanche region, the atoms are ionized by collision. These colli-
sions occur at random and they produce noise spikes in the avalanche
current. These noise spikes from the avalanche noise.
i. Avalanche noise is normally observed in reverse biased condition
of zener diodes especially in avalanche region.
ii. In the avalanche region, the electrons and holes gains sufficient
energy from a reverse biased field to ionize atoms by collision.
iii. The spectral density of avalanche noise is flat.
iv. Avalanche noise is used in noise measurements.

4. White Noise :
White noise contains all frequency components in equal proportion. White
noise is not the noise source. It is the classification of noise. The noise which
has gaussian distribution and have flat spectral density over a wide range
of frequencies.
White light contains all visible spectral components. The white noise also all
frequency components in equal proportion. Hence the name ’white noise’
is given.
The power spectral density of white noise is independent of frequency and
is N2o for all frequencies. The parameter No is defined as – No = kTe . Here
k is the Boltzmans constant and Te is the equivalent noise temperature of
the receiver. The equivalent noise temperature of a system is defined as
the temperature at which the noisy resistor has to be maintained such that
by connecting the resistor to the input of a noiseless version of a system, it
produces the same available noise power at the output of the system as that
produced by all the sources of noise in the actual system.

PBCOE, Nagpur 14 L: Dr. P.R. Bokde

IV Sem EC Analog & Digital Communication Summer-2023

Que. 4 (b)
Explain superheterodyne receiver with the help of a block diagram.

Solution :
The problems of TRF receiver are overcome in this receiver. The superheterodyne
receiver converts all incoming RF frequencies to a fixed lower frequency, called
intermediate frequency (IF). This IF is then amplified and detected to get the orig-
inal signal. Figure 5 shows the block diagram of superheterodyne receiver.

Figure 5:

The antenna receives all the frequency signals and gives it to RF amplifier.
The RF stage amplifies the signals in the required range of frequencies. Thus it
provides initial gain and selectivity. The output of the RF amplifier is given to
the mixer stage. The local oscillator output is also applied to the mixer. Let us
assume that local oscillator frequency is fo and signal frequency is fs . The signal
frequency fs and local oscillator frequency fo are mixed in the mixer in such a
way that frequency difference (fo − fs ) is produced at the output of mixer. This
difference fo − fs is called Intermediate Frequency (IF). The signal at this IF con-
tains the same modulation as the incoming signal. The IF is amplified by one or
more IF amplifier stages and given to the detector. Most of the gain and selectiv-
ity is provided by these IF amplifiers. Normally IF is fixed for the AM recivers.
To select a particular station, the local oscillator frequency fo is changed in such
a way that the frequency fs of that station and fo has the difference equal to IF.
Thus whatever is the station being tuned, the IF is fixed. Thus the IF amplifiers
and detector operate at the single frequency i.e. IF. Hence the bandwidth of the
IF amplifiers is relatively narrow.

L: Dr. P.R. Bokde 15 PBCOE, Nagpur

Summer-2023 Analog & Digital Communication IV Sem EC

A part of output is taken from the detector and it is applied to RF amplifier,

mixer and If amplifiers for gain control. This is called Automatic Gain Control or
AGC. This AGC maintains the constant output voltage level over a wide range
of RF input signal levels. The detector obtains the modulating signal from the
modulated IF. The output of detector is amplified and given to speaker.

Advantages of Superheterodyne Receiver

1. The selectivity of this receiver is better since its IF amplifiers are narrow-
band, and operate only at IF.

2. The design of IF amplifiers is relatively simple since they operate only at IF.

Que. 5 (a)
State sampling theorem. Explain types of sampling.

Solution :

Sampling Theorem
The Sampling Theorem, also known as Nyquist Theorem, states:

"A band-limited signal can be completely reconstructed from its

samples if it is sampled at a rate at least twice the highest fre-
quency present in the signal."

If a continuous-time signal x(t) has a maximum frequency component fm ,

then the minimum sampling frequency fs must be:

fs ≥ 2fm

1. If a signal is not sampled at a rate of atleast 2fm , aliasing occurs, causing

loss of information.

2. Sampling at exactly 2fm is called Nyquist rate, while sampling above 2fm is
called oversampling.

3. Practical systems often use a sampling rate slightly higher than 2fm to avoid
aliasing and reconstruction errors.

PBCOE, Nagpur 16 L: Dr. P.R. Bokde

IV Sem EC Analog & Digital Communication Summer-2023

Sampling and its types

Sampling is the process where an analog signal is converted into a corresponding
sequence of samples that are usually spaced uniformly in time. i.e. process of
converting continuous time signal into discrete time signal.
There are two types of sampling :

1. Ideal sampling or Impulse sampling or Instantaneous sampling

2. Practical Sampling

(a) Natural sampling or chopper sampling

(b) Flat top sampling or Sample & Hold Sampling

Natural Sampling or Chopper Sampling

In natural sampling the pulse has finite width τ . Natural sampling is some times
called chopper sampling because the waveform of the sampled signal appears to
be chopped off from the original signal waveform.

Figure 6: Natural Sampler

Let us consider an analog continuous time signal x(t) to be sampled at the rate
of fs Hz and fs is higher than Nyquist rate such that sampling theorem is satis-
fied. A sampled signal s(t) is obtained by multiplication of a sampling function
and signal x(t). Sampling function c(t) is a train of periodic pulses of width τ
and frequency equal to fs Hz. Figure ?? shows a functional diagram of natural
sampler. When c(t) goes high, a switch ′ s′ is closed. Therefore,

s(t) = x(t) when c(t) = A

s(t) = 0 when c(t) = 0

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Summer-2023 Analog & Digital Communication IV Sem EC

Here, A is amplitude of c(t).

The waveforms of x(t), c(t) and s(t) are shown in figure ??. Signal s(t) can also
be defined mathematically as –

s(t) = c(t) x(t) (17)

Here, c(t) is the periodic train of pulses of width τ and frequency fs .

Figure 7: Natural Sampling

Flat Top Sampling or Rectangular Pulse Sampling

This is also a practically possible sampling method. Natural sampling is little
complex, but it is easy to get flat top samples. The top of the samples remains

PBCOE, Nagpur 18 L: Dr. P.R. Bokde

IV Sem EC Analog & Digital Communication Summer-2023

constant and equal to instantaneous value of baseband signal x(t) at the start of
1
sampling. The duration of each sample is τ and sampling rate is equal to fs = .
Ts
Figure 9 shows the functional diagram of sample and hold circuit generating flat
top samples and figure 9 (b) shows waveforms.

Figure 8: Sample and hold circuit generating flat top sampling

Normally the width of the pulse in flat top sampling and natural sampling is
increased as far as possible to reduce the transmission bandwidth.

Here we see from figure 9 (b) that only starting edge of the pulse represents
instantaneous value of the baseband signal x(t). the flat tope pulse of s(t) is math-
ematically equivalent to the convolution of instantaneous sample and pulse h(t)
is shown in figure ??. i.e. the width of the pulse in s(t) is determined by width
of h(t), and sampling instant is determined by delta function. In the waveforms
showsn in figure 9(b), the starting edge of pulse represents the point where base-
band signal is sampled and width is determined by function h(t). Therefore s(t)
will be given as –

L: Dr. P.R. Bokde 19 PBCOE, Nagpur

Summer-2023 Analog & Digital Communication IV Sem EC

Figure 9: Convolution of any function with delta function is equal to that function

s(t) = xδ (t) ∗ h(t) (18)

By the replication property of delta function, we know that

x(t) ∗ δ(t) = x(t) (19)

The delta function in equation 19 is instantaneously sampled signal xδ (t) and

function h(t) is convolved with xδ (t). Clearly ovserve that we are not directly
applying equation 19 here, but we are using it similarly. In equation 19, δ(t) is
constant amplitude delta function. But in figure ?? (b), xδ (t) is varying amplitude
train of impulses. Therefore on convolution of xδ (t) and h(t), we get a pulse
whose duration is equal to h(t) only but amplitude is defined by xδ (t).
xδ (t) is given as –
X∞
xδ (t) = x(nTs )δ(t − nTs ) (20)
n=−∞

Que. 5 (b)
Explain in detail pulse code modulation with the help of a diagram.

Solution :
Pulse Code Modulation (PCM) is an analog to digital converter where the in-
formation contained in the instantaneous samples of an analog signal are repre-

PBCOE, Nagpur 20 L: Dr. P.R. Bokde

IV Sem EC Analog & Digital Communication Summer-2023

sented by digital codes in a serial bit stream manner.

Figure 10:

The block diagram of a PCM system is shown in figure 10. It consists of –

1. Transmitter
2. Regenerative repeater
3. Receiver
PCM Transmitter :

1. Low Pass filter :

In practice the low pass filter (pre-alias filter) is used before sampler in order
to limit the frequency greater tha W Hz. Hence message signal is bandlim-
ited to W Hz.

2. Sampler :
The incoming message signal is sampled with a train of narrow rectangu-
lar pulses. The sampling rate fs is selected above Nyquiest rate to avoid
aliasing i.e, fs ≥ 2W .

3. Quantization :
The sampled signal is fed to the quantizer. The quantizer approximates each
input signal level to the nearest prefixed level.

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Summer-2023 Analog & Digital Communication IV Sem EC

The output of the quantizer is discrete time discrete valued signal known as
"quantized signal".

4. Encoding :
The quantized samples are then encoded in the encoder. The process of
encoding involves allocating some digital code to each level. These coded
levels are transmitted as a bitstream of data i.e. 0’s and 1’s.

5. Regenerative Repeater :
The PCM signal is reconstructed by means of a regenerative repeater located
at sufficiently closed spacing along the transmission path.
The regenerative networks are used at intermediate points between trans-
mitter and receiver in order to boost up the pulse amplitude.

PCM Receiver :

1. Decoder :
The first operation in the receiver is to generate the received pulses.
The decoder converts binary coded signal to a approximated pulses of dis-
crete magnitude.

2. Reconstruction Filter :
The final operation in the receiver is to recover the original analog signal.
This is done by passing the decoder output through a low pass filter. The
output of low pass filter is an analog signal.

Advantages of PCM
1. Since PCM is a digital technique, it is less affected by noise and interference
compared to analog modulation.
2. Digital signals can be compressed, multiplexed, and efficiently transmitted
over long distances.
3. PCM supports error detection and correction techniques, ensuring reliable
data transmission.
4. Due to quantization and encoding, PCM signals maintain quality over long
distances without degradation.
5. Digital signals can be encrypted for secure communication.
6. PCM is used in digital telephony, VoIP, and modern multimedia communi-
cation systems.

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IV Sem EC Analog & Digital Communication Summer-2023

Disadvantages of PCM :
1. PCM requires more bandwidth compared to analog modulation techniques
due to binary encoding.

2. Quantization error can occur due to the rounding off of analog signal values
to discrete levels.

3. The encoding and decoding process requires complex hardware and pro-
cessing power.

4. Receiver synchronization is essential for proper decoding of PCM signals.

Applications of PCM :
1. Used in telephone networks (PSTN) and Voice over IP (VoIP) for clear com-
munication.

2. PCM is used in CDs, DVDs, and high-quality audio recording systems.

3. PCM ensures secure and error-free data transmission in space communica-

tion.

4. Used in medical imaging and signal processing (e.g., ECG, MRI).

5. PCM is used in secure military communication and radar systems.

6. PCM is widely used in digital TV, video streaming, and broadcasting.

Que. 6 (a)
Explain delta modulation technique in detail. Also explain slope overload
distortion and granular noise.

Solution :
Delta modulation transmits only one bit per sample i.e. the present sample value
is compared with the previous sample value and the indication, whether the am-
plitude is increased or decreased is sent.

1. The input signal x(t) is approximated to step signal by the delta modulator.
The difference between input signal x(t) and staircase approximated signal
is quantized into only two levels i.e. +∂ or −∂.

2. If the difference is positive, then approximated signal is increased by one

step i.e. +∂ and bit 1 is transmitted.

3. If the difference is negative, then approximated signal is reduced by one

step i.e. −∂ and bit 0 is transmitted.

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Summer-2023 Analog & Digital Communication IV Sem EC

4. Thus for each sample only one bit is transmitted.

Figure 11: Illustration of Delta Modulation

Delta Modulator Transmitter :

Figure 12: Delta Modulation Transmitter

• The error between the sampled value x(nTs ) and last approximated sample
is given by –
e(nTs ) = x(nTs ) − x̂(nTs ) (21)

• Let u(nTs ) be the present sample approximation of staircase output.

From figure,

x̂(nTs ) = u(n − 1)Ts (22)

∴ x̂(nTs ) = u(nTs − Ts ) (23)

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IV Sem EC Analog & Digital Communication Summer-2023

Substituting equation (23) in equation (21), we get,

e(nTs ) = x(nTs ) − u(nTs − Ts ) (24)

• The binary quantity b(nTs ) is the algebraic sign of the error e(nTs ), except
for the scaling factor ∂.

b(nTs ) = ∂Sgn [e(nTs )] (25)

Here, b(nTs ) depends on the sign of error e(nTs ), the sign of step-size ∂ will
be decided.

i.e. b(nTs ) = +∂ , if x(nTs ) ≥ x̂(nTs ) (26)

b(nTs ) = −∂ , if x(nTs ) ≤ x̂(nTs ) (27)
(28)

• If b(nTs ) = +∂, then binary 1 is transmitted.

If If b(nTs ) = −∂, then binary 0 is transmitted.

u(nTs ) = u [nTs − Ts ] + b(nTs ) (29)

• The previous sample approximation u[nTs − Ts ] is restored by delaying one

sample period Ts .

Delta Modulation Receiver :

Figure 13 shows the block diagram of DM receiver.

Figure 13: Delta Modulation Receiver

• The accumulator generates the staircase approximated signal output and is

delayed by one sampling period Ts . It is then added to the input signal.

• If input is binary 1, then it adds +∂ step to the previous output.

• If input is binary 0, then one step ∂ is subtracted from the delayed signal.

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Summer-2023 Analog & Digital Communication IV Sem EC

• The low pass filter is used to remove step variation and to get smooth re-
constructed message signal x(t).

Advantages of Delta Modulation :

The DM has the following advantages over PCM.

1. DM transmits only one bit for one sample. Thus the signalling rate and
transmission channel bandwidth is quite small for DM.

2. Simiplicity of design for both the transmitter and the receiver.

3. A one bit code word for the output, which eliminates the need for word
processing.

Disadvantages of Delta Modulation

Delta modulation systems are subjected to two types of quantization errors :

1. Slope overload distortion

2. Granular Noise.

Figure 14: Illustration of quantization error in delta modulation

1. Slope Overload Distortion :

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IV Sem EC Analog & Digital Communication Summer-2023

(a) Slope overload distortion arises because of the large dynamic range of
the input signal.
(b) In figure 14, it cn be seen that, the rate of rise of input signal x(t) is
so high that the staircase signal cannot approximate it, the step size ∂
becomes too small for staircase signal x(t) to follow the steep segment
of x(t). Thus large error between the staircase approximated signal
and the original input signal x(t). This error is called slope overload
distortion.
(c) To reduce this error, the step size should be increased when slope of
the signal x(t) is high. i.e. slope of the staircase u(t) ≥ slope of the
message signal.
∂ d
≥ max [x(t)]
Ts dt
2. Granular Noise :
(a) This noise occurs when the step size is too large compared to small
variations in the input signal i.e. for very small variations in the input
signal, the staircase signal is changed by large amount because of large
step size ∂.
(b) In figure 14, the input signal is almost flat, the staircase signal u(t)
keeps on oscillating by ±∂ around the signal.
(c) The error between the input and approximated signal is called Granu-
lar noise. The solution of this problem is to make step size small.

Que. 6 (b)
Write a short note on : (i) Aliasing and aperture effect (ii) Companding in
PCM

Solution :

(i) Aliasing and Aperture effect

While proving sampling theorem we considered that fs = 2W . Consider the case
of fs < 2W . Then the spectrum of Xδ (f ) shown in figure 15 will be modified as
follows:

1. The spectrums located at X(f ), X(f − fs ), X(f − 2fs ), ..... overlap on each
other.

2. Consider the spectrums of X(f ) and X(f −f s) shown as magnified in figure

15. The frequencies from (fs − W ) to W are overlapping in these spectrums.

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Summer-2023 Analog & Digital Communication IV Sem EC

3. The high frequencies near ω in X(f − fs ) overlap with low frequencies (fs −
W ) in X(f ).

Figure 15:

When the high frequency interferes with low frequency and appears as low
frequency, then the phenomenon is called aliasing.

Effects of aliasing :

• Since high and low frequencies interfere with each other, distortion is gen-
erated.

• The data is lost and it cannot be recovered.

Different ways to avoid aliasing

Aliasing can be avoided by two methods :

• Sampling rate fs ≥ 2W .

• Strictly bandlimit the signal to ’W’

When the sampling rate is made higher than 2W, then the spectrums will not
overlap and there will be sufficient gap between the individual spectrums. This
is shown in figure 16.

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IV Sem EC Analog & Digital Communication Summer-2023

Figure 16:

When the signal is sampled at a rate much higher than Nyquist rate, it is called
oversampling. It is necessary to avoid aliasing error in the signal. But it increases
transmission bandwidth.

(ii)Companding in PCM
1. Speech communication is very important in digital communication systems.
If uniform quantization is used, the step size will be contants.

2. The system that uses equally spaced quantization levels, the quantization
noise is same for all signal amplitudes. Hence small amplitude samples are
more affected than the bigger sample values. Therefore to keep signal to
quantization noise ratio high, we must use a signal which is large in com-
parison with step size. This requirement is not satisfied when signal is small
i.e. we need smaller step size for low magnitude signal samples and higher
step size for higher magnitude signals.

3. Changing step size according to signal magnitude is not preferable one. In-
stead, change the characteristics of the signal such that lower amplitudes
are amplified without changing maximum value of the signal.

4. Maintaining of constant SNR throughout the signal range is called "Robust

Quantization".

5. To achieve this, the signal is passed through a combination of Compressor-

Expander circuit respectively at the transmitter and receiver. This technique
is known as Companding.

6. To achieve robust quantization, the signal is passed through a network which

has an input-output characteristics as shown in figure (17).

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Summer-2023 Analog & Digital Communication IV Sem EC

7. The signal is changed such that small amplitude signals are boosted up
without altering the maximum amplitude of the signal, small amplitude
signals range through more quantization levels.

8. Any signal when passed through such a network gets compressed leading
to signal distortion.

9. To remove this distortion, the signal is passed through an inverse network

at the receiver called as Expander.

10. The complete process of compressing and expanding the signal is referred
to as Companding.

Figure 17:

There are two types of companding :

(a) µ – law companding

(b) A – law companding

Que. 7 (a)
Explain generation and detection of DPSK in detail.

Solution :
Differential Phase Shift Keying (DPSK) isdifferentially coherent modulation method.
DPSK does not need a synchronous (coherent) carrier at the demodulator. The in-
put sequence of binarybits is modified such that the next bit depends upon the
previous bit. Therefore in the receiver the prevous received bits are used to detect
the present bit.

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IV Sem EC Analog & Digital Communication Summer-2023

Non-coherent detection of BPSK is not possible since message information lies

in phases. Hence DPSK is also called non-coherent version of BPSK.

Generation of DPSK signal

Figure18 shows the scheme to generate DPSK signal.

Figure 18: Block diagram of DPSK transmitter

Figure 19 shows the waveforms of the above circuit. Here b(t) = d(t)⊕b(t−Tb ).
The initial value of b(t − Tb ) is assumed zero.
The differentially encoded signal b(t) then performs BPSK modulation of the
√
carrier 2P cos(2πfo t).
Figure 19 shows the phase shift of the carrier after modulation. Observe that
phase of the carrier changes by 180o only when d(t) = 1.

Figure 19: Waveforms of DPSK

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Summer-2023 Analog & Digital Communication IV Sem EC

Always two successive bits of d(t) are checked for any change of level. Hence
one symbol has two bits.

Symbol duration (T) = Duration of two bits(2Tb )

∴ T = 2Tb

As shown in figure ??, the sequence b(t) is applied to a balanced modulator.

√
The balanced modulator is also supplied with a carrier 2P cos(2πfo t)..
The modulator output is,
√
s(t) = b(t) 2P cos(2πfo t)
√
= ± 2P cos(2πfo t)

DPSK Receiver

Figure 20 shows the method to recover the binary sequence from DPSK signal.
Figure ?? (a) and (b) are equivalent to each other. Figure 20 (b) represents DPSK
receiver using correlator. Figure ??(a) shows multiplier and integrator separately.

Figure 20: DPSK Receiver

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IV Sem EC Analog & Digital Communication Summer-2023

Operation of Receiver

1. Phase shift in received signal : During the transmission, the DPSK signal
undergoes some phase shift 0. Therefore the signal received at the input of
the receiver is -
√
Received signal = b(t) 2P cos(2πfo t + 0)

2. Synchronous detector or Multiplier output : The signal is multiplied with

its delayed version by one bit. Therefore the output of the multiplier is,

Multiplier Output = b(t)b(t − Tb )(2P ) cos(2πfo t + 0) cos[2πfo (t − Tb ) + θ]

We know that,

1
cos(A) cos(B) = [cos(A − B) + cos(A + B)]
2

Here, A = 2πfo t + 0 and B = 2πfo (t − Tb ) + 0

    
Tb
Multiplier Output = b(t)b(t − Tb )p cos 2πfo Tb + cos 4πfo t − + 2θ
2

fo is the carrier frequency and Tb is one bit period. Tb contains integral

number of cycles of fo . We know that,

1
fb =
Tb

If Tb contains n cycles of fo , then we can write,

fo = nfb
n
∴ fo =
Tb
∴ fo Tb = n

Putting fo T = n in the first cosine term we get,

    
Tb
Multiplier output = b(t)b(t − Tb )P cos 2πn + cos 4πfo t − + 2θ
2

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Summer-2023 Analog & Digital Communication IV Sem EC

Since cos 2πn = 1, the above equation will be,

   
Tb
Multiplier output = b(t)b(t − Tb )P + b(t)b(t − Tb )P cos 4πfo t − + 2θ
2

3. Integrator : The above signal is given to the integrator. In the k th bit interval,
the integrator output can be written as,

Z kTb
so (kTb ) = b(kTb )b [(k − 1)Tb ] P dt
(k−1)Tb
Z kTb
   
Tb
+ b(kTb )b [(k − 1)Tb ] P cos 4πfo t − + 2θ dt
(k−1)Tb 2

The integration of the second term will be zero since it is integration of

carrier over one bit duration. The carrier has integral number of cycles over
one bit period hence integration is zero. Therefore we can write,

so (kTb ) = b(ktb )b [(k − 1)Tb ] P [kTb − (k − 1)Tb ]

= b(kTb )b [(k − 1)Tb ] P Tb

Here we know that P Tb = Eb ; i.e. energy of one bit. The product b(kTb )b [(k − 1)Tb ]
decides the sign of P Tb .

The transmitted data bit d(t) can be verified easily from product b(kTb )b [(k − 1)Tb ].
We know from figure 19 when b(t) = b(t − Tb ), d(t) = 0. That is if both are
+1 V or -1 V, then b(t)b(t − Tb ) = 1. Alternatively we can write,

If b(t)b(t − Tb ) = 1 V d(t) = 0

We know that b(t) = b(t − Tb ), then d(t) = 1. That is b(t) = −1 V, b(t − Tb ) =

+ 1V and vice versa.Therefore b(t)b(t − Tb ) = −1. Alternately we can write,

If b(t)b(t − Tb ) = −1 V d(t) = 1

4. Decision Device : The decision device is shown in figure 20. We know that,

so (kTb ) = b(kTb )b [(k − 1)Tb ] P Tb

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IV Sem EC Analog & Digital Communication Summer-2023

so (kTb ) = −P Tb , then d(t) = 1 and

= +P Tb , then d(t) = 0

Que. 7 (b)
A memoryless source emits 6 messages with probabilities : 0.3, 0.25,
0.15, 0.12, 0.1 and 0.08. Determine : (i) Compact Huffman binary code
(ii)Average Code word length (iii) Entropy of the source (iv) Efficiency (v)
Redundancy

Solution :

Entropy of the source is -

n
X
H(m) = Pi log2 Pi
i=1

= 0.3log2 (0.3) + 0.25log2 (0.25) + 0.15log2 (0.15)

+ 0.12log2 (0.12) + 0.1log2 (0.1) + 0.08log2 (0.08)
∴ H(m) = 2.42 bits/message

The Huffman code is constructed as follows:

Message P (Mi ) S1 S2 S3 S4
M1 0.3 (00) 0.3 (00) 0.3 (00) 0.43 (1) 0.57 (0)

M2 0.25(10) 0.25 (10) 0.27(01) 0.3 (00) 0.43 (1)

M3 0.15(010) 0.18 (11) 0.25 (10) 0.27 (01)

M4 0.12 (011) 0.15 (010) 0.18 (11)

M5 0.1 (110) 0.12 (011)

M6 0.08(111)

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Summer-2023 Analog & Digital Communication IV Sem EC

Message Probability No. of Encoded Bits (n)

M1 0.3 2
M2 0.25 2
M3 0.15 3
M4 0.12 3
M5 0.1 3
M6 0.08 3

Average Code Word Length is -

n
X
L= P (m1 )ni
i=1

∴ L = 0.3(2) + 0.25(2) + 0.15(3) + 0.12(3) + 0.1(3) + 0.08(3)

∴ = 2.45 bits/message

Code efficiency is given by -

H(m)
η=
L
2.42
=
2.45
∴ η = 0.9881
∴ %η = 98.81%

Redundancy is given by -

r = 1 − η = 1 − 0.9881
∴ r = 0.0119

Que. 8 (a)
State and prove Hartley-Shannon’s Channel Capacity theorem.

Solution :
Channel capacity of discrete memoryless channel is maximum information that
can be transmitted per second. It is denoted as ’C’ bits/sec.
The channel capacity of a channel is limited by bandwidth and signal to noise
ratio (S/N ratio) of the system.
In a channel distributed by White Gaussian noise, one can transmit informa-

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IV Sem EC Analog & Digital Communication Summer-2023

tion at a rate of maximum ’C’ bits/sec efficienty.

 
S
C = B log2 1+ bits/sec (30)
N

This is known as Hartley-Shanon’s Law.

Here,
C = Channel Capacity in bits/sec
B = Bandwidth of signal (Hz)
S = Signal Power
N = Noise Power

Proof : This theorem is derived with assumption that, if a signal is mixed with
noise, then signal amplitude can be recognized only within root mean square
noise voltage. Let us consider the average signal power and noise power as ’S’
watts and ’N’ watts respectively.

Let the load of R = 1 Ω.

Power ’P’ is given by -

V2
P =VI = = I 2R (31)
R

But, R = 1 Ω
∴P =V2 (32)

But,

P =S+N (33)
∴V2 =S+N (34)
√
∴V = S+N volts (35)

Above equation represents the root mean square value of received signal.
√
Therefore RMS value of noise voltage is N volts.
Therefore, Number of distinct voltage levels that can be distinguished without
√
S+N
noise = √ N
.

r
S
∴M = 1+ (36)
N

Where, M = No. of voltage levels without noise.

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Summer-2023 Analog & Digital Communication IV Sem EC

Maximum amount of information carried by each pulse is -

I = log2 (M ) (37)
r
S
I = log2 1+ (38)
 N 
1 S
∴ I = log2 1 + bits (39)
2 N

If the channel can transmit ’K’ pulses/sec then maximum amount of informa-
tion transmitted per second by a channel is -
 
K S
C= log2 1 + (40)
2 N

A system with bandwidth ’B’ Hz can transmit a maximum of ’2B’ pulses per
second. Therefore K = 2B.
 
2B S
∴C= log2 1 + (41)
2 N
 
S
∴ C = B log2 1 + (42)
N

Que. 8 (b)
Find the LZ source coding for the given sequence -
000101110010100101. Assume that the binary symbols 0 and 1
are already in the code book.

Solution :
Given Sequence is - 000101110010100101
Given that 0 and 1 are already in the code book.

Numerical Se- 1 2 3 4 5 6 7 8 9
quence
Subsequence 0 1 00 01 011 10 010 100 101
Number representa- – – 11 12 42 21 41 61 62
tion
Binary Encoded Se- – – 0010 0011 1001 0100 1000 1100 1101
quence (L-Z coding)
So, the encoded sequence is - 00100011110010100100011001101

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IV Sem EC Analog & Digital Communication Summer-2023

Que. 9(a)
Explain linear block codes and cyclic codes in detail.

Solution :
In digital communication systems, error detection and correction is crucial to en-
sure data integrity. Linear Block Codes and Cyclic Codes are two important types
of error-correcting codes used to detect and correct errors in transmitted data.

Linear Block Codes

A Linear Block Code is an error-correcting code in which a set of k -bit message
blocks is mapped into a set of n -bit codewords n > k in such a way that the
combination of any two codewords also results in a valid codeword. This ensures
that errors can be detected and corrected using algebraic properties.

Properties of Linear Block Codes:

1. Each codeword consists of n bits: k message bits and (n - k) redundant

(parity) bits.

2. Linearity Property: The sum (XOR operation) of two valid codewords is

also a valid codeword.

3. Can detect and correct single-bit and multiple-bit errors, depending on the
code design.

4. Hamming Distance: The minimum number of bit changes required to con-

vert one codeword into another. It determines error detection and correc-
tion capability.

Example of a Linear Block Code (Hamming Code (7,4)):

• k=4 message bits

• n=7 total bits (4 message + 3 parity)

• Can detect and correct single-bit errors.

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Summer-2023 Analog & Digital Communication IV Sem EC

Cyclic Codes

A Cyclic Code is a special type of Linear Block Code in which cyclic shifts (ro-
tations) of a valid codeword result in another valid codeword. These codes are
widely used in digital communication because of their efficient encoding and de-
coding mechanisms.

Properties of Cyclic Codes:

1. A cyclic shift of any valid codeword produces another valid codeword.

2. Can be implemented using polynomial algebra.

3. Efficient error detection and correction, making them useful for real-time
communication.

Types of Cyclic Codes:

1. Cyclic Redundancy Check (CRC) Codes:

(a) Used in error detection (e.g., network protocols, storage devices).

(b) Computed using polynomial division (e.g., CRC-32 in Ethernet).

2. Bose-Chaudhuri-Hocquenghem (BCH) Codes:

Capable of correcting multiple errors.

3. Reed-Solomon (RS) Codes:

Used in CDs, DVDs, satellite communication, and QR codes.

Example of a Cyclic Code:

Consider a generator polynomial G(x) used to encode a message M(x). The

encoded codeword is obtained as:

C(x) = G(x) · M (x)

The receiver checks for errors using division by G(x).

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IV Sem EC Analog & Digital Communication Summer-2023

Que. 9(b)

A convolutional encoder is described by v1 = [111] and v2 = [101].

(i) Draw the convolutional encoder (ii) Draw the state diagram (iii) Draw
the code tree (iv) Determine the input sequence for Y = 110111 using Veterbi
algorithm.

Solution :
(i) Encoder :
V1 = S1 ⊕ S2 ⊕ S3
V2 = S1 ⊕ S3

(ii) To obtain state table

The two bits m1 and m2 in the shift register will indicate the state of the encoder.
Let these states be defined as follows:

S2 S3 State
0 0 a
0 1 b
1 0 c
1 1 d

State Transition Table :

Input Present State Next State Output

S2 S3 State S1S2 State V1 V2
S1
0 00 a 00 a 0 0
1 00 a 10 c 1 1
0 01 b 00 a 1 1
1 01 b 10 c 0 0
0 10 c 01 b 1 0
1 10 c 11 d 0 1
0 11 d 01 b 0 1
1 11 d 11 d 1 0

(1) State Diagram :

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Summer-2023 Analog & Digital Communication IV Sem EC

11 01

00 a 10 00 d 01

11 01

(2) Code Tree :

Figure 21:

(3) Trellis Diagram :

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IV Sem EC Analog & Digital Communication Summer-2023

Figure 22:

(4) Calculation of input sequence for Y = 110111 Using Veterbi algorithm, the
input sequence can be found from trellis diagram. Make a group of output as a
group of two bits. Observe the outgoing path 0 (solid line) and 1 (dotted line)
corresponding to the group of output bits with minimum difference (metric). As
observed from the Trellis diagram the input sequence is -

Output Bits 11 01 11
Corresponding detected input bit us- 1 1 1
ing viterbi algorithm

Que. 10
Write short notes on any three :

1. Direct Sequence spread spectrum

2. PN sequence

3. OFDM

4. CDMA

Solution :

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Summer-2023 Analog & Digital Communication IV Sem EC

1. Direct Sequence Spread Spectrum

Figure 23: Idealized model of baseband spread spectrum system (a) Transmitter
(b) Channel (c) Receiver

1. The baseband DS-SS does not use any digital modulation techniques.

2. The digital signals bandwidth is widen by using only spreading codes (PN
sequence).

3. The baseband signal b(t), which is of low bandwidth (Narrowband) is mul-

tiplied with the wideband signal c(t) to obtain spread spectrum signal m(t)
as shown in figure 23.
m(t) = b(t) · c(t) (43)

4. The m(t) is transmitted through the channel where additive noise i(t) is
added (figure 23).

5. The received signal consists of transmitted signal m(t) plus an additive in-
terference denoted by i(t).

r(t) = m(t) + i(t) (44)

Substituting equation 43 in equation 44, we get,

r(t) = c(t)b(t) + i(t) (45)

6. To recover the original data sequence b(t), the received signal r(t) is applied
to a demodulator that consists of a multiplier followed by a low pass filter.

7. The multiplier is supplied with a locally generated PN-sequence i.e. an

PBCOE, Nagpur 44 L: Dr. P.R. Bokde

IV Sem EC Analog & Digital Communication Summer-2023

exact replica of that used in the transmitter.

z(t) = c(t) · r(t) (46)

Substituting equation 45 in equation 46, we get,

z(t) = c(t) [c(t) · b(t) + i(t)] (47)

z(t) = c2 (t) · b(t) + c(t) · i(t) (48)
∴ z(t) = b(t) + c(t) · i(t) (49)

where, c2 (t) = 1 for all ’t’.

8. From equation 49, b(t) can be recovered by passing z(t) through a low pass
filter, which removes the effect of the interference represented by c(t) · i(t)
and reproduces the original data.

Figure 24:

Advantages of Direct Sequence - Spread Spectrum (DS-SS)

1. Best antijam performance.

2. Best noise performance.

L: Dr. P.R. Bokde 45 PBCOE, Nagpur

Summer-2023 Analog & Digital Communication IV Sem EC

3. Coherent demodulation of the SS signal is possible.

4. The generation of the coded signal is easy. It can be done by a simple mul-
tiplication.

5. DS-SS has best discrimination against multipath signals.

Disadvantages of DS-SS

1. It has long acquisition time.

2. Susceptible to the near-far problem.

3. The PN generator should generate sequence at high rates.

4. It requires wideband channel with small phase distortion.

2. PN Sequence

1. A Pseudo-Noise (PN) sequence is defined as a Coded Sequence of 0’s and

1’s with certain auto-correlation properties.

2. The PN sequence used in Spread Spectrum communication are periodic.

3. The length of the PN-sequence is given by -

N = 2m − 1 (50)

where, m is the number of flip-flops.

4. Using m-stage shift registers (m-FF’s), it is possible to generate a periodic

sequence i.e 2m − 1 bits. Such sequences are also called Maximum Length
(ML) sequence.

PBCOE, Nagpur 46 L: Dr. P.R. Bokde

IV Sem EC Analog & Digital Communication Summer-2023

Figure 25: 3-stage maximum length sequence generator

5. The shift register operation is controlled by a sequence of clock pulses. For

every clock pulse, the contents of each stage of the register is shifted by one
position to the right.

6. For every clock pulse the contents of second and third stages are modulo-2
added and the result is fed back to the first stage.

7. The output of the shift register is taken at the last stage of the flip-flop i.e.
Q3 .

8. The length of PN-Sequence is given by -

N = 2m − 1
∴ N = 23 − 1
∴ N = 7 bits

9. For an initial state 1,1,0 the output sequence is -

0, 1, 1, 1, 0, 0, 1, 0, 1, 1, 1, 0, 0, 1
| {z }| {z }

L: Dr. P.R. Bokde 47 PBCOE, Nagpur

Summer-2023 Analog & Digital Communication IV Sem EC

10. The PN sequence is periodic with period equal to 7-bits.

Note:

• Suppose initially all the contents of the shift registers are zero i.e. Q1 = Q2 =
Q3 = 0.

• Modulo-2 addition of Q2 and Q3 is ’0’, which is fed back to stage 1.

• Hence, when the clock pulses are applied, only 0’s are shifted at input D1 ,
shift register state remain at state 000 and the output sequence i.e. Q3 is a
sequence of all 0’s.

Properties of PN-Sequence
There are 3 properties of PN sequence :

1. Balanced property

2. Run property

3. Auto-correlation property

Balanced Property
In each period of a ML-sequence, the number of 1’s is always one more than the
number of 0’s (i.e. numer of 1’s exceeds the number of 0’s by one).
Example : For 3-stage shift register, N = 23 − 1 = 7 i.e. 0010111.
Number of 1’s = 4
Number of 0’s = 3

Run Property
A run is defined as a subsequence of identical symbols within the ML-sequence.
The length of the subsequence is known as the run-length.

(N + 1)
The total number of runs = (51)
2
In ML-sequence :

1. One-half the runs are of length one.

PBCOE, Nagpur 48 L: Dr. P.R. Bokde

IV Sem EC Analog & Digital Communication Summer-2023

2. One-fourth the runs are of length two.

3. One-eighth the runs are of length three.

Example : 0010111

N +1 7+1
Total Number of runs = = = 4 runs
2 2

00, 1, 0, 111 = 4 runs.

• 1,0 → Two runs are of length one.

• 00 → One run are of length two

• 111 → One run are of length three.

Auto-correlation property
The auto-correlation function of a ML-sequence is periodic and binary valued.

N
1 X
Rc (k) = Cn Cn−k (52)
N n=1

where, N is the length or period of the PN sequence and k is the lag of the auto-
correlation sequence.

Rc (k) = 1 ; fork = ln
1
=− k ̸= ln
N

where, ’1’ is any integer

3. OFDM
The principle of OFDM is transmitting data by dividing the data stream into
multiple parallel bit streams that have a much lower bit rate and using these
sub-streams to modulate several carriers. OFDM is more resistant to frequency
selective fading than single carrier systems are.

OFDM system
1. Orthogonal frequency division multiplexing (OFDM) is a multicarrier trans-
mission technique which is based on frequency division multiplexing (FDM).

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Summer-2023 Analog & Digital Communication IV Sem EC

2. In conventional FDM multiple-frequency signals are transmitted simulta-

neously in parallel where the data contained in each signal is modulated
onto subcarriers and therefore the subcarrier multiplexed signal typically
contains a wide range of frequencies.

3. Each subcarrier is separated by a guard band to avoid signal overlapping.

4. The subcarriers are then demodulated at the receiver by using filters to sep-
arate the frequency bands.

5. OFDM employs several subcarrier frequencies orthogonal to each other (i.e.

perpendicular) and therefore they do not overlap.

6. Hence this technique can squeeze multiple modulated carriers tightly to-
gether at a reduced bandwidth without the requirement for guard bands
while at the same time keeping the modulated signals orthogonal so that
they do not interfere with each other, as illustrated in Figure 26.

Figure 26: Orthogonal Frquency Division Multiplexing (OFDM) compared with

conventiona frequency division multiplexing (FDM)

7. In the upper spectral diagram 10 non-overlapping subcarrier frequency sig-

nals arranged in parallel depicting conventional FDM are shown, each be-
ing separated by a finite guard band.

8. OFDM is displayed in the bottom spectral diagram where the peak of one
signal coincides with the trough of another signal.

PBCOE, Nagpur 50 L: Dr. P.R. Bokde

IV Sem EC Analog & Digital Communication Summer-2023

9. Each subcarrier must maintain the Nyquist criterion separation with the
minimum time period of T for each subcarrier OFDM uses the inverse fast
Fourier transform (IFFT) for the purpose of modulation and the fast Fourier
transform (FFT) for demodulation.

10. This is a consequence of the FFT operation by which subcarriers are posi-
tioned perpendicularly and hence the reason why the technique is referred
to as orthogonal FDM.

11. It may be observed that a large bandwidth saving in comparison with con-
ventional FDM is identified in Figure ?? resulting from the orthogonal place-
ment of the subcarriers.

12. Since the orthogonal feature allows high spectral efficiency near the Nyquist
rate where efficient bandwidth use can be obtained, OFDM generally ex-
hibits a nearly white frequency spectrum.

13. OFDM, also being tolerant to signal dispersion, thus enables high-speed
data transmission across a dispersive channel and it has been widely used
in high-bit-rate cable and wireless communication systems.

14. For applications within optical fiber communications it is necessary to in-

corporate an optical source to convert the OFDM signals into an optical
signal format before coupling onto an optical fiber, while at the receiving
end the intensity modulated signal can be recovered to as optical OFDM
(OOFDM).

15. Although the multiplexing approach is similar to optical SCM, the orthog-
onal nature of the subcarriers is unique to OOFDM.

(4) CDMA

In this method every user is assigned the unique code sequence or signature se-
quence. The signal is then spread across the complete frequency band with the
help of this code. As the receiver, the signal is recovered with the help of same
code.

L: Dr. P.R. Bokde 51 PBCOE, Nagpur

Summer-2023 Analog & Digital Communication IV Sem EC

Figure 27: CDMA Transmission

Since the signals in CDMA spread over the complete frequency band, it is aso
called spread spectrum multiple access (SSMA).
Access to the user is given randomly. Hence signal transmissions from various
overlap in time as well as frequency. Figure 27 ilustrates CDMA concept.

Advantages
1. Maximum utilization of the channel takes place.

2. Synchronization is not necessary.

Disadvantages
1. Chance of data collision because of overlap.

2. Protocols are necessary to avoid the collision.

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