Digital Communication

Course Code CSE 2002

Course Name Digital Communications

Credits 5

Pre-Requisites ICS

L-T-P-C 3-1-2-5

COURSE OBJECTIVE

Upon successful completion of this course, students will be able to:

1. To know the principles of sampling & quantization
2. To study the various waveform coding schemes
3. To understand the various Bandpass signaling schemes
4. To know the fundamentals of satellite communication
LECTUREs WITH BREAKUP No. of
Lectures

Module-I: Introduction Random Theory 6

Random variable and random process, joint distribution and density
function, CDF, PDF, CCDF, Sampling and quantization.

Module-II: Quantization’s & Companding 8

PCM introduction, S/Nq for sinusoidal and arbitrary signal PCM
Generation and Reconstruction, Non-Uniform Quantization and
Companding, DPCM, Adaptive DPCM, DM and Adaptive DM, White Noise
in PCM

Module-III: Digital Modulation schemes 8

Digital Modulation Techniques: ASK- Modulator, Coherent ASK Detector,
FSK- Modulator, Non-Coherent FSK Detector, BPSK- Modulator,
Coherent BPSK Detection. BPSK error performance, Principles of QPSK,
Differential PSK and QAM, M-ary modulation schemes and performance
analysis.

Module-IV: Waveform Coding 8

Introduction to line codes, Line coding, properties, Power Spectral Density
of Unipolar / Polar RZ & NRZ, Bipolar NRZ, Manchester- ISI, Nyquist
criteria for distortion-less transmission, Pulse shaping – Correlative
coding – M-ary schemes – Eye pattern – Equalization

Module-V Advance communication systems 8

Spreading and its types, DSSS, FHSS,CDMA, SS7, OFDM- 64/128 QAM,
NOMA, Introduction to cellular communication & GSM. Introduction to
wireless communication, satellite communication and digital satellite
communication, types of satellite systems( LEO, MEO and GEO)

COURSE OUTCOMES

On completion of the course the student should be able to;

1. Design PCM systems

2. Design and implement base band transmission schemes
3. Analyze the spectral characteristics of band pass signaling schemes
and their noise performance
4. Know the wireless communication systems.

Text Book:
1. Principal of Communication system, Taub and Schilling, 2nd edition, TMH
publication
2. Digital Communication, John G. Proakis, 5th edition, Mc.Graw Hill
publications
Reference Book:

1. Digital Communication Systems Using MATLAB® and Simulink® by Dennis

Silage
2. Fundamentals of Wireless Communication by David Tse, and Pramod
Viswanath.
3. Digital Communications Fundamentals and applications by Bernard Sklar.
Prentice Hall

By
Dr. Jagadish S Jakati
Assistant Professor, SCSEA
D. Y. Patil International University,
Akurdi, Pune,411044,
Maharashtra, India
Email: [email protected]
Website: www.dypiu.ac.in
2nd Year DIGITAL COMMUNICATION CSE 2002

Module-01 RBT Level-L1, L2

Introduction Random Theory:

Random variable and random process, joint distribution and density function,
CDF, PDF, CCDF, Sampling and quantization.

Random Variable Probability is a mathematical concept that

assigns numerical values to outcomes of a sample space. They can
describe the outcomes of objective randomness (like tossing a coin).
We define a random variable as a function that maps from the
sample space of an experiment to the real numbers.
Mathematically, Random Variable is expressed as,
X: S →R

where,

X is Random Variable (It is usually denoted using capital letter)

S is Sample Space
R is Set of Real Numbers
Random Variable Example

Example 1
If two unbiased coins are tossed then find the random variable
associated with that event.

Solution:
Suppose Two (unbiased) coins are tossed
X = number of heads. [X is a random variable or function]
Here, the sample space S = {HH, HT, TH, TT}
Example 2

Suppose a random variable X takes m different values i.e. sample

space

X = {x1, x2, x3………xm} with probabilities

P(X = xi) = pi, where 1 ≤ i ≤ m
The probabilities must satisfy the following conditions:
0 ≤ pi ≤ 1; where 1 ≤ i ≤ m

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

p1 + p2 + p3 + ……. + pm = 1 Or we can say 0 ≤ pi ≤ 1 and ∑pi = 1

Hence possible values for random variable X are 0, 1, 2.
X = {0, 1, 2} where m = 3
P(X = 0) = (Probability that number of heads is 0) = P(TT) = 1/2×1/2 =
1/4
P(X = 1) = (Probability that number of heads is 1) = P(HT | TH) =
1/2×1/2 + 1/2×1/2 = 1/2
P(X = 2) = (Probability that number of heads is 2) = P(HH) = 1/2×1/2 =
1/4
Here, you can observe that, (0 ≤ p1, p2, p3 ≤ 1/2)
p1 + p2 + p3 = 1/4 + 2/4 + 1/4 = 1
For example,
Suppose a dice is thrown (X = outcome of the dice). Here, the
sample space S = {1, 2, 3, 4, 5, 6}. The output of the function will
be:
P(X=1) = 1/6, P(X=2) = 1/6 ,P(X=3) = 1/6 ,P(X=4) = 1/6 ,
P(X=5) = 1/6, P(X=6) = 1/6
Random process: A random process is a collection (or ensemble) of
random variables {X(s, t)} that are functions of a real variable,
namely time 't' where s ε S (Sample space) and t ε T (parameter set
or index set)

A random process is usually conceived of as a function of time,

but there is no reason to not consider random processes that are
functions of other independent variables, such as spatial
coordinates.
Difference between Random variable and Random Process

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Joint Distribution: A joint probability is a statistical measure used

to calculate the likelihood of two events taking place at the same
time. Both events must be independent of each other. Joint
probability is also called the intersection of two or more events.

Formula and Calculation of Joint Probability

P (X⋂Y) where: X,Y=Two different events that intersect

P(X and Y),P(XY)=The joint probability of X and Y

For example, the probability of drawing a red card from a deck of
cards is 1/2 = 0.5. This means there is an equal chance of drawing
a red and black card since there are 26 of each in a deck. As such,
there is a 50-50 probability of drawing a red card versus a black
card.
Joint probability measures two events that happen at the same
time. It can only be applied to situations where more than one
observation can occur at the same time. So the joint probability of
picking a card that is both red and 6 from a deck is P(6 ∩ red) =
2/52 = 1/26 since a deck of cards has two red sixes—the six of
hearts and the six of diamonds. Because the events red and 6 are
independent, you can also use the following formula to calculate the
joint probability: P(6∩red)=P(6)×P(red)=4/52×26/52=1/26
The symbol “∩” in a joint probability is referred to as an
intersection. The probability of event X and event Y happening is
the same thing as the point where X and Y intersect. Therefore, the
joint probability is also called the intersection of two or more events.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Examples
Example 1
What is the joint probability of rolling the number five twice in
a fair six-sided dice?
Event “A” = The probability of rolling a 5 in the first roll is 1/6 =
0.1666.
Event “B” = The probability of rolling a 5 in the second roll is 1/6 =
0.1666.
Therefore, the joint probability of event “A” and “B” is
P(1/6) x P(1/6) = 0.02777 = 2.8%.

Example 2
What is the joint probability of getting a head followed by a tail
in a coin toss?
Event “A” = The probability of getting a head in the first coin toss is
1/2 = 0.5.
Event “B” = The probability of getting a tail in the second coin toss
is 1/2 = 0.5.
Therefore, the joint probability of event “A” and “B” is
P(1/2) x P(1/2) = 0.25 = 25%.

Example 3
What is the joint probability of drawing a number ten card that
is black?
Event “A” = The probability of drawing a 10 = 4/52 = 0.0769
Event “B” = The probability of drawing a black card = 26/52 = 0.50
Therefore, the joint probability of event “A” and “B” is
P(4/52) x P(26/52) = 0.0385 = 3.9%.

Properties of Joint Distributions

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Cumulative Distribution Function(CDF)

The Cumulative Distribution Function (CDF), of a real-valued
random variable X, evaluated at x, is the probability function that X
will take a value less than or equal to x

Let X be a continuous random variable with a sample space Ω = R.

The cumulative distribution function (CDF) of X

The CDF defined for a discrete random variable and is given as

Fx(x) = P(X ≤ x)
Where X is the probability that takes a value less than or equal to x
and that lies in the semi-closed interval (a,b], where a < b.

Therefore the probability within the interval is written as

P(a < X ≤ b) = Fx(b) – Fx(a)

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

The CDF defined for a continuous random variable is given as;

Here, X is expressed in terms of integration of its probability density

function fx.
In case, if the distribution of the random variable X has the discrete
component at value b,
P(X = b) = Fx(b) – limx→b- Fx(x)

Cumulative Distribution Function Properties

The cumulative distribution function Fx(x) of a random variable has
the following important properties:

 Every CDF Fx is non decreasing and right continuous

limx→-∞Fx(x) = 0 and limx→+∞Fx(x) = 1

 For all real numbers a and b with continuous random variable

X, then the function fx is equal to the derivative of Fx, such
that Properties of CDF

 If X is a completely discrete random variable, then it takes the

values x1, x2, x3, … with probability pi = p(xi), and the CDF of
X will be discontinuous at the points x i:FX(x) = P(X ≤ x) =

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Probability Density Function (PDF)

Probability density function is the function that represents the

density of probability for a continuous random variable over the
specified ranges.

Graph for Probability Density Function

If X is continuous random variable and f(x) be the probability
density function. The probability for the random variable is given by
area under the pdf curve. The graph of PDF looks like bell curve,
with the probability of X given by area below the curve. The
following graph gives the probability for X lying between interval a
and b.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Probability Density Function Formula

Let Y be a continuous random variable and F(y) be the cumulative
distribution function (CDF) of Y. Then, the probability density
function (PDF) f(y) of Y is obtained by differentiating the CDF of Y.

If we want to calculate the probability for X lying between the

interval a and b, then we can use the following formula:

How to Find Probability from Probability Density Function

Step 1: First check the PDF is valid or not using the necessary
conditions.
Step 2: If the PDF is valid, use the formula and write the required
probability and limits.
Step 3: Divide the integration according to the given PDF.
Step 4: Solve all integrations.
Step 5: The resultant value gives the required probability.

Probability Density Function Properties

Let f(x) be the probability density function for continuous random
variable x. Following are some probability density function
properties:
 Probability density function is always positive for all the values
of x.
f(x) ≥ 0, ∀ x ∈ R
 Total area under probability density curve is equal to 1.

 For continuous random variable X, while calculating the

random variable probabilities end values of the interval can be
ignored i.e., for X lying between interval a and b
P (a ≤ X ≤ b) = P (a ≤ X < b) = P (a < X ≤ b) = P (a < X < b)
 Probability density function of a continuous random variable
over a single value is zero.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Probability Density Function Vs Cumulative Distribution

Function
Aspect Probability Density Cumulative
Function (PDF) Distribution Function
(CDF)
Definition The PDF gives the The CDF gives the probability
probability that a that a random variable is less
random variable takes than or equal to a specific
on a specific value value.
within a certain range.
Range of Defined for continuous Defined for both continuous
Values random variables. and discrete random
variables.
Mathematical f(x), where f(x)≥0 and F(x), where 0≤F(x)≤1 for all x,
Expression ∫−∞∞f(x)dx=1 and F(−∞)=0 and F(∞)=1
Interpretation Represents the Represents the probability
likelihood of the that the random variable is
random variable taking less than or equal to a
on a specific value. specific value.
Area Under The area under the PDF The value of the CDF at a
the Curve curve over a certain specific point gives the
interval gives the probability that the random
probability that the variable is less than or equal
random variable falls to that point.
within that interval.
Relationship The PDF can be The CDF can be obtained by
with CDF obtained by integrating the PDF with
differentiating the CDF respect to the random
with respect to the variable.
random variable.
Probability The probability of a The probability of a random
Calculation random variable falling variable being less than or
within a specific equal to a specific value x is
interval (a,b) is given by given by F(x).
∫abf(x)dx.
The PDF is always non- The CDF is a monotonically
negative: f(x)≥0 for all x. increasing function: F(x1) ≤
The total area under the F(x2) if x1 ≤ x2.
Properties PDF curve is equal to 1. 0≤F(x)≤1 for all x.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Complementary Cumulative Distribution Function (CCDF)

The CCDF is defined as the complement of the CDF (i.e., 1 minus

the CDF)

Therefore, the CCDF can be calculated from either the CDF or the
probability density function (PDF).

Sampling and quantization.

Sampling in digital communication is converting a continuous-time

signal into a discrete-time signal. It can also be defined as the
process of measuring the discrete instantaneous values of a
continuous-time signal.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Sampling Process in Digital Communication

The sampling process includes the following steps:

 The continuous signal is taken as an input.

 Sampling is performed to convert this signal into a digital
representation.
 In addition to sampling, quantization of a signal is performed.
 After the above step, encoding of the signal is done.

Important Terminologies of Sampling in Digital

Communication.
 Sampling
 Sample
 Sampling Rate or Sampling Frequency
 Nyquist Rate
 Nyquist Interval
 Quantization

Sampling: It is the process by which, we convert CTS (continuous

time signal) into DTS (discrete time signal)
Sample: It can be defined as the numeric value of an analog signal
at a specific time.
Sampling Rate or Sampling Frequency: It refers to the number of
samples or data points taken per unit of time from an analog signal
to convert it into a digital format. It is also known as sampling
frequency. It is measured in Hertz (Hz).
School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

The formula for sampling rate or sampling frequency is given

by:

Nyquist Rate: It is the minimum sampling rate required to

accurately capture an analog signal in digital form without
information loss. It is also known as Nyquist Frequency or Nyquist
Limit.
It is defined as twice the maximum frequency component present in
the analog signal. Mathematically it can be represented as:

Nyquist Interval: The Nyquist interval, also known as the Nyquist

period, is the time interval between consecutive samples in a digital
signal or digital sampling system. It is the reciprocal of the Nyquist
rate. Mathematically it can be represented as:

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Quantization.
It is the process to represent a continuous-valued signal with a
limited set of discrete values. In other words, it involves mapping a
continuous signal’s infinite range of potential values to a finite
collection of discrete values.

Types of Quantization
There are two types of Quantization
 Uniform Quantization
 Non-uniform Quantization.

The type of quantization in which the quantization levels are

uniformly spaced is termed as a Uniform Quantization.

There are two types of uniform quantization.

 Mid-Rise type
 Mid-Tread type

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

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.

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.

Difference between Uniform and Non-uniform Quantization

Aspects Uniform Quantization

Non-Uniform Quantization
Definition The quantization levels
The quantization levels are
are uniformly spaced is
unequal and mostly the
termed as a Uniform
relation between them is
Quantization. logarithmic, is termed as a
Non-uniform Quantization.
SNR SNR ratio varies with the SNR ratio can be maintained
input signal amplitude constant
Step Size Constant Variable
Relationship Linear Non-Linear
High SNR, Uniform Low SNR, Non-Uniform
Best Use Case Distribution Distribution
Distribution of Not accounted for input Accounts for input data
input data data distribution distribution

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Advantages of Quantization.
 It reduces the number of bits used to represent a signal.
 It enables uniform precision. Precision refers to the accuracy
or exactness required in a method.
 It represents the sampled value of the signal into the finite
number of levels, which helps in converting an input analog
signal to the digital signal.

Applications of Quantization.

 Digital Signal Processing: In digital signal processing,

quantization maps the large set of the input values to the
small set of the output values with a finite number of
elements.
 Image processing: In image processing, quantization reduces
the number of discrete value in the signal.
 Physics: Quantization in science is related to the
electromagnetic wave, quantum, or photons. It represents the
transitions from the classic mechanics to quantum mechanics.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Module-02 RBT Level-L1, L2

Quantization’s & Companding:

PCM introduction, S/Nq for sinusoidal and arbitrary signal PCM Generation
and Reconstruction, Non-Uniform Quantization and Companding, DPCM,
Adaptive DPCM, DM and Adaptive DM, White Noise in PCM.

Introduction to Pulse Code Modulation (PCM).

A signal is pulse code modulated to convert its analog information into a binary
sequence, i.e., 1s and 0s. The output of a PCM will resemble a binary
sequence. The following figure shows an example of PCM output with respect to
instantaneous values of a given sine wave.

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

coded pulses. This message signal is achieved by representing the signal in
discrete form in both time and amplitude.
Basic Elements of PCM
The transmitter section of a Pulse Code Modulator circuit consists
of Sampling, Quantizing and Encoding, which are performed in the analog-to-
digital converter section. The low pass filter prior to sampling prevents aliasing
of the message signal.
The basic operations in the receiver section are regeneration of impaired
signals, decoding, and reconstruction of the quantized pulse train. Following

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

is the block diagram of PCM which represents the basic elements of both the
transmitter and the receiver sections.

Low Pass Filter

This filter eliminates the high frequency components present in the input
analog signal which is greater than the highest frequency of the message
signal, to avoid aliasing of the message signal.
Sampler
This is the technique which helps to collect the sample data at instantaneous
values of message signal, so as to reconstruct the original signal. The sampling
rate must be greater than twice the highest frequency component W of the
message signal, in accordance with the sampling theorem.
Quantizer
Quantizing is a process of reducing the excessive bits and confining the data.
The sampled output when given to Quantizer, reduces the redundant bits and
compresses the value.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Encoder
The digitization of analog signal is done by the encoder. It designates each
quantized level by a binary code. The sampling done here is the sample-and-
hold process. These three sections LPF, Sampler, and Quantizer will act as an
analog to digital converter. Encoding minimizes the bandwidth used.
Regenerative Repeater
This section increases the signal strength. The output of the channel also has
one regenerative repeater circuit, to compensate the signal loss and reconstruct
the signal, and also to increase its strength.
Decoder
The decoder circuit decodes the pulse coded waveform to reproduce the original
signal. This circuit acts as the demodulator.
Reconstruction Filter
After the digital-to-analog conversion is done by the regenerative circuit and
the decoder, a low-pass filter is employed, called as the reconstruction filter to
get back the original signal.
Advantages and Disadvantages
Advantages
 Pulse Code Modulation is used in long-distance communication.
 The efficiency of the transmitter in PCM is high.
 Higher noise immunity is seen.
 Efficient method.
Disadvantages
 The bandwidth requirement is high.
 PCM is a complex process, since it involves encoding, decoding and
quantisation of the circuit.
Applications of Pulse Code Modulation
 It is used in telephony and compact discs.
 Pulse Code Modulation is used in satellite transmission systems and
space communications.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Signal-to-Noise Ratio (SNR) in Pulse Code Modulation (PCM)

A signal-to-noise ratio, often written S/N or SNR, is a measure of the strength
of the desired signal relative to background noise (undesired signal). The
ratio is typically expressed as a single numeric value in decibels (dB).
How do you calculate the signal-to-noise ratio?
The signal-to-noise ratio is typically measured in decibels and can be
calculated by using a base 10 logarithm. The exact formula depends on how
the signal and noise levels are measured, though.
S/N = 20 log10(Ps/Pn)
Where
Ps is the desired signal,
Pn is the background noise (undesired signal)

Non-Uniform Quantization and Companding.

Companding: It is a signal processing technique used in digital
communication to compress and expand signals, which improves their
dynamic range and reduces noise. It's used in telephony, audio systems, and
wireless microphones.
Model of Companding

We have already discussed that companding is formed by merging the

compression and expanding. Initially at the transmitting end the signal is
compressed and further at the receiving end the compressed signal is expanded
in order to have the original signal.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Initially at the transmitting end, the signal is first provided to the compressor.
The compressor unit amplifies the low value or weak signal in order to increase
the signal level of the applied input signal.
While if the input signal is a high-level signal or strong signal then compressor
attenuates that signal before providing it to the uniform quantizer present in
the model.
This is done in order to have an appropriate signal level as the input to the
uniform quantizer. We know a high amplitude signal needs more bandwidth
and also is more likely to distort. Similarly, some drawbacks are associated
with low amplitude signal and thus there exist need for such a unit.
The operation performed by this block is known as compression thus the unit
is called compressor.
The output of the compressor is provided to uniform quantizer where the
quantization of the applied signal is performed.
At the receiver end, the output of the uniform quantizer is fed to the expander.
It performs the reverse of the process executed by the compressor. This unit
when receives a low value signal then it attenuates it. While if a strong signal is
achieved then the expander amplifies it.
This is done in order to achieve the originally transmitted signal at the output.
Characteristic of Compander
As we know companding is composed of compression and expanding. So, here
in this session we will separately discuss the compressor and expander
characteristic.
Compressor characteristic: The figure below shows the graphical
representation of characteristic of the compressor:

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

The graph clearly represents that the compressor provides high gain to weak
signal and low gain to high input signal.
Expander characteristic: Here the figure shows the characteristic of
expander:

The expander performs reverse operation of the compander. So, it is clear from
the above figure that artificially boosted signals is attenuated to have the
originally transmitted signal.
The figure below represents the companding curve for PCM system:

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

The compressor and expander perform inverse operations thus in the above
figure the dotted line represents the linear characteristic of the compander
indicating that the originally transmitted signal is recovered at the receiver.
Advantages of Companding:
1. Improves SNR: One of the main advantages of companding is that it
improves the SNR of the transmitted signal. This is due to the fact that
the companding procedure lowers the signal's noise amplitude, which
raises the transmitted signal's SNR.
2. Reduces data transmission: Companding also lessens the quantity of
data that needs to be delivered, which is a benefit. This is because the
companding process reduces the dynamic range of the signal, which in
turn reduces the number of bits required to represent the signal. When
the communication channel's bandwidth is constrained, this might be
helpful.
3. Reduces quantization noise: Companding also helps to reduce the
effects of quantization noise. The noise that is added when a continuous
signal is transformed into a discrete signal is known as quantization
noise. The companding procedure lowers the quantization noise's
amplitude, enhancing the transmitted signal's SNR in turn.
4. Easy implementation: Companding is relatively easy to implement and
can be implemented in a variety of ways, such as logarithmic
companding, A-law companding, and μ-law companding.
Disadvantages of Companding:
1. Complexity: Companding is a complex technique that requires a
significant amount of processing power to implement. This can be a
disadvantage for systems with limited processing power.
2. Distortion: Companding can introduce distortion into the signal if not
implemented correctly. This can lead to a loss of quality in the
transmitted signal.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

3. Incompatibility: Companding is not compatible with all digital

communication systems. Systems that are not designed to handle
companded signals may not be able to process them correctly.
4. Limited dynamic range: Companding reduces the dynamic range of the
signal, which can be a disadvantage if the signal needs to be transmitted
over a long distance. This is because a signal with a limited dynamic
range is more susceptible to noise and interference.
5. Overall, companding is a powerful technique for improving the
performance of digital communication systems.
Non-uniform quantization: Non-Uniform quantization, uses variable size
steps.
Non-uniform quantization can be more difficult to implement but provides
better performance for specific applications.
Formula and Derivation of Non-Uniform Quantization

Non-uniform quantization often uses a companding approach, where the signal

is first compressed, uniformly quantized, and then expanded. The compression
and expansion functions can vary, but a common method is to use a
logarithmic companding function. For example, the μ-law companding used in
telephony is given by:

y=sgn(x)⋅ln(1+μ)/ln(1+μ∣x∣)
where:
 x is the input signal.
 y is the compressed signal.
 μ is the companding parameter.
 sgn(x) is the sign function, preserving the sign of x.

Derivation
The steps for non-uniform quantization using μ law companding are as follows:
1.Compression: Apply the compression function to the input signal x:
y=sgn(x)⋅ln(1+μ)/ln(1+μ∣x∣)
2. Uniform Quantization: Quantize the compressed signal y using uniform
quantization.
3. Expansion: Apply the inverse of the compression function to the quantized
value yq :

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

xq=sgn(yq)⋅(1/μ((1+μ)∣yq∣–1))
This process ensures that the quantization error is minimized in regions where
the signal amplitude is more likely to occur, providing better performance for
non-uniformly distributed signals.

Differential pulse-code modulation (DPCM)

Differential pulse-code modulation (DPCM) is a signal encoder that uses the
baseline of pulse-code modulation (PCM) to encode differences between the
current and predicted values of an analog signal.

DPCM Transmitter

The DPCM Transmitter consists of Quantizer and Predictor with two summer
circuits.

The sampled signal is denoted by x(nT s) and predicted signal is denoted by

xˆ(nTs).

The comparator finds out the difference between the actual sample value x(nT s)
and predicted sample value xˆ(nT s).

This is known as prediction error and it is denoted by e(nT s).

It can be defined as,

e(nTs) = x(nTs) – xˆ(nTs)……………………….(1)

The predicted value is produced by using a prediction filter.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

The quantizer output signal gap e q(nTs) and previous prediction is added and
given as input to the prediction filter. This signal is called x q(nTs).

This makes the prediction more and more close to the actual sampled signal.

We can observe that the quantized error signal e q(nTs) is very small and can be
encoded by using small number of bits.

Thus number of bits per sample are reduced in DPCM.

The quantizer output can be written as,

eq(nTs) = e(nTs) + q(nTs)………………………..(2)

Here, q(nTs) is the quantization error.

the prediction filter input xq(nTs) is obtained by sum xˆ(nTs) and quantizer
output. i.e.,

xq(nTs) = xˆ(nTs) + eq(nTs)……………………..(3)

Substituting the value of eq(nTs) from eq.(2) in the above eq. (3) , we get,

xq(nTs) = xˆ(nTs) + e(nTs) + q(nTs) ………………….(4)

eq.(1) is written as,

e(nTs) = x(nTs) – xˆ(nTs)

∴ e(nTs) + xˆ(nTs) = x(nTs)

Therefore, substituting the value of e(nT s) + xˆ(nTs) from the above equation
into eq. (4), we get,

xq(nTs) = x(nTs) + q(nTs) …………………..(5)

DPCM Receiver

The block diagram of DPCM Receiver consists of a decoder, a predictor, and a

summer circuit.

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2nd Year DIGITAL COMMUNICATION CSE 2002

The decoder first reconstructs the quantized error signal from incoming binary
signal.

The prediction filter output and quantized error signals are summed up to give
the quantized version of the original signal.

Thus the signal at the receiver differs from actual signal by quantization
error q(nTs), which is introduced permanently in the reconstructed signal.

Advantages of DPCM

1. As the difference between x(nTs) and xˆ(nTs) is being encoded and

transmitted by the DPCM technique, a small difference voltage is to be
quantized and encoded.

2. This will require a smaller number of quantization levels and hence a

smaller number of bits to represent them.

3. Thus, signaling rate and bandwidth of a DPCM system will be less than
that of PCM.

Delta Modulation

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

The type of modulation, where the sampling rate is much higher and in which
the stepsize after quantization is of a smaller value Δ, such a modulation is
termed as delta modulation.

Features of Delta Modulation

Following are some of the features of delta modulation.

 An over-sampled input is taken to make full use of the signal correlation.

 The quantization design is simple.

 The input sequence is much higher than the Nyquist rate.

 The quality is moderate.

 The design of the modulator and the demodulator is simple.

 The stair-case approximation of output waveform.

 The step-size is very small, i.e., Δ deltadelta.

 The bit rate can be decided by the user.

 This involves simpler implementation.

Delta Modulation is a simplified form of DPCM technique, also viewed as 1-bit

DPCM scheme. As the sampling interval is reduced, the signal correlation will
be higher.

Delta Modulator

The Delta Modulator comprises of a 1-bit quantizer and a delay circuit along
with two summer circuits. Following is the block diagram of a delta modulator.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

The predictor circuit in DPCM is replaced by a simple delay circuit in DM.

From the above diagram, we have the notations as −

 x(nTs) = over sampled input

 ep(nTs) = summer output and quantizer input

 eq(nTs) = quantizer output = v(nTs)

 xˆ(nTs) = output of delay circuit

 u(nTs) = input of delay circuit

Using these notations, now we shall try to figure out the process of delta
modulation.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Where,

 xˆ(nTs) = the previous value of the delay circuit

 eq(nTs) = quantizer output = v(nTs)

Hence

Which means,

The present input of the delay unit = The previous output of the delay
unit + the present quantizer output the present quantizer output

Delay unit output is an Accumulator output lagging by one sample.

A Stair-case approximated waveform will be the output of the delta modulator

with the step-size as delta (Δ). The output quality of the waveform is moderate.

Delta Demodulator

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2nd Year DIGITAL COMMUNICATION CSE 2002

The delta demodulator comprises of a low pass filter, a summer, and a delay
circuit. The predictor circuit is eliminated here and hence no assumed input is
given to the demodulator.

From the above diagram, we have the notations as −

 vˆ(nTs) is the input sample

 uˆ(nTs) is the summer output

 x¯(nTs) is the delayed output

A binary sequence will be given as an input to the demodulator. The stair-case

approximated output is given to the LPF.

Low pass filter is used for many reasons, but the prominent reason is noise
elimination for out-of-band signals. The step-size error that may occur at the
transmitter is called granular noise, which is eliminated here. If there is no
noise present, then the modulator output equals the demodulator input.

Advantages of Delta Modulation

 Design is easy and simple.

 It is a 1-bit quantizer.
 Modulator & demodulator can be designed easily.
 In delta modulation, the quantisation design is very simple
 The bit rate can be designed by the user

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2nd Year DIGITAL COMMUNICATION CSE 2002

Disadvantages of Delta Modulation

 When the value of the delta is small, slope overload distortion is seen,
which is a type of noise.
 When the value of delta is large, granular noise is seen, which is a type of
noise.
Adaptive Delta Modulation ADM
A larger step-size is needed in the steep slope of modulating signal and a
smaller step size is needed where the message has a small slope. The minute
details get missed in the process. So, it would be better if we can control the
adjustment of step-size, according to our requirement in order to obtain the
sampling in a desired fashion. This is the concept of Adaptive Delta
Modulation.

Transmitter
Input: The input of the transmitter is an analog signal that we want to encode
and transmit. This signal is the original continuous waveform that carries
information.

Logic for Step Size Control: This is a very important part of the transmitter. It
basically determines the step size or quantization level for encoding the
differences between consecutive samples of the input signal. The logic for step
size control adapts the step size based on the characteristics of the input
signal.

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2nd Year DIGITAL COMMUNICATION CSE 2002

One Bit Quantizer: The one-bit quantizer is responsible for converting the
continuously varying amplitude of the input signal into a binary
representation. It quantizes the difference between consecutive samples into a
binary value for example one bit (0 or 1). This quantized information is then
used for transmission.

Accumulator: The accumulator keeps track of the cumulative sum of the

quantized differences. It integrates the quantized values over time, providing a
running total. This accumulated value is then used to predict the next sample
of the input signal.

Output: The output of the transmitter is the modulated signal, which consists
of the quantized binary values and may also include information about the step
size or any necessary control signals. This modulated signal is then
transmitted to the receiver through the transmission medium or channel.

Receiver

Input: The input to the receiver is the modulated signal which is received from
the transmitter. This signal carries the quantized information about the
differences between consecutive samples of the original input signal.

Logic for Step Size Control: Similar to the transmitter, the receiver has logic
for step size control. This logic adapts the step size based on the received
information, adjusting to the changing characteristics of the transmitted signal.

Accumulator: The accumulator in the receiver is used to predict the next

sample of the input signal based on the received quantized information. It
accumulates the received quantized differences to estimate the original signal.

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2nd Year DIGITAL COMMUNICATION CSE 2002

Low Pass Filter: The low pass filter is applied to the output of the accumulator
to smooth out the high-frequency components and reduce noise, providing a
more accurate representation of the original signal.

Output: The output of the receiver reconstructs the analog signal, which
ideally closely resembles the original input signal. This reconstructed signal is
the output of the Adaptive Delta Modulation system on the receiving end.

Adaptive Delta Modulation waveform

Advantages of Adaptive Delta Modulation

 Adaptive delta modulation offers extremely high performance.

 This technique decreases the need for correction circuits in radio design
and error detection.
 Dynamic range is high since the variable step size covers a large range of
values.
 Slope overload error and granular error are not seen.
 Slope error is reduced.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Differences between Delta Modulation and Adaptive Delta Modulation

The differences between adaptive delta modulation and delta modulation are
listed below-
 In Delta Modulation step size is fixed for the whole signal. Whereas in
Adaptive delta modulation, the step size varies depending upon the input
signal.
 The slope overload and granular noise errors which are present in delta
modulation are not seen in this modulation.
 The dynamic range of Adaptive delta modulation is wider than delta
modulation.
 This modulation utilizes bandwidth more effectively than delta
modulation.
White noise in Pulse Code Modulation
The noise is additive, i.e., the received signal is equal to the transmitted signal
plus noise. This gives the most widely used equality in communication
systems.
r(t)=s(t)+w(t)

which is shown in the figure below. Moreover, this noise is statistically

independent of the signal. Remember that the above equation is highly
simplified due to neglecting every single imperfection a Tx signal encounters,
except the noise itself.
White Just like the white colour which is composed of all frequencies in the
visible spectrum, white noise refers to the idea that it has uniform power
across the whole frequency band. As a consequence, the Power Spectral
Density (PSD) of white noise is constant for all frequencies ranging
from −∞ to +∞, as shown in Figure below.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

White noise is characterized by the following properties:

 Uniform power: White noise has uniform power across the whole
frequency band.
 Gaussian distribution: The probability distribution of the noise samples
is Gaussian with a zero mean.
 Additive: The received signal is equal to the transmitted signal plus
noise.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Module-03 RBT Level-L1, L2

Digital Modulation schemes:

ASK- Modulator, Coherent ASK Detector, FSK- Modulator, Non-Coherent FSK
Detector, BPSK- Modulator, Coherent BPSK Detection. BPSK error
performance, Principles of QPSK, Differential PSK and QAM, M-ary modulation
schemes and performance analysis.

Amplitude Shift Keying (ASK) Modulator

Definition: A digital modulation technique in which the amplitude of the carrier

wave is altered according to the modulating signal (bitstream) is known as
Amplitude Shift Keying (ASK). It is the easiest and straightforward digital
modulation scheme.

ASK is sometimes known as On-Off keying because the carrier wave swings
between 0 and 1 according to the low and high level of input signal
respectively.

ASK Generation

ASK Waveforms

In ASK, frequency and phase of the carrier wave is kept constant and only the
amplitude is varied according to the digitized modulating signal. It is also
referred as Binary Amplitude Shift Keying (BASK) as its usual operation is
associated with only two levels.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Generation of ASK signal

The figure below shows the block diagram representing the process of
generation of an amplitude shift keying waveform.

It consists of a signal generator that produces a high-frequency sinusoidal

waveform, a message signal in digitized form and a bandpass filter.

The switch provided here gets open and closed according to the bits of the
message signal. When the digital bit is of level high i.e., 1 then the switch gets
closed. Thus, allows the carrier wave to get transmitted.

As against, in case of low-level bit i.e., 0 the switch gets open and restrict the
carrier wave.

The band-limiting filter, shapes the pulse depending upon the amplitude and
phase characteristics of the band-limiting filter or the pulse-shaping filter.

ASK Demodulation

Demodulation in Amplitude Shift Keying (ASK) involves extracting the original

digital message signal from the modulated carrier wave. The process reverses
the modulation applied to the carrier signal. The demodulation of ASK can be
performed through various methods, such as envelope detection or coherent
detection.

Types Of ASK Demodulation

There are mainly two types of Demodulation which is performed namely

envelope detection and coherent detection technique:

 Coherent detection (Synchronous demodulation)

 Noncoherent Detection (Asynchronous demodulation) or or Envelope
detection

ASK Coherent detection (Synchronous demodulation)

Coherent demodulation is a technique that recovers the original information

from a modulated signal by mixing it with a Carrier signal that has the same
frequency and phase.

This is method which is used for demodulation of ASK signal. It involves phase
synchronization with the carrier wave.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

The steps involved in this technique is as follows

 The received signal is mixed with a local oscillator, aligning its phase and
frequency with the original carrier signal. This process is called
Reception.
 Filtering isolates the part of the mixed signal that corresponds to the
data signal.
 The demodulator sets an amplitude threshold to distinguish between the
‘1’ and ‘0’ states.
 The amplitude of the received signal is compared against the threshold to
determine the transmitted data.
In this way the original message signal is recovered.

Noncoherent Detection (Asynchronous demodulation) or Envelope

detection

Envelop detection is a relatively simpler method as compared to coherent

detection which is used for demodulating Amplitude Shift Keying (ASK) signals.
It primarily focuses on extracting the envelope of the modulated signal to
recover the original digital data.
The steps involved in this technique is as follows
 The modulated ASK signal, which contains the carrier signal and the
digital data, is received by the demodulator. This is called receiving of
signal.
 The received signal is then passed through a diode. The diode rectifies the
signal, allowing only the positive portion of the waveform (the envelope)
to pass through it. This process is called Diode Rectification.
 Following the rectification process, a low-pass filter is applied to the
rectified signal. This filter helps remove or discard the high-frequency
carrier signal, leaving the varying amplitude that represents the original
digital data. This process is called Low-Pass Filtering of the modulated
signal.
 Once the signal has been filtered by passing it through the LPF, a
threshold detection mechanism is used to interpret the amplitude

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2nd Year DIGITAL COMMUNICATION CSE 2002

changes. Comparing the amplitudes to a predefined threshold helps to

determine the transmitted digital data. If the amplitude is above the
threshold, it might be interpreted as one bit ‘1’, and if it’s below the
threshold, it might be interpreted as the other bit ‘0’. This is called
Threshold Detection.
 The message signal is recovered from the modulated signal.

Advantages

There are various advantages of using ASK as a modulation technique.

 It is one of the simplest modulation techniques which makes it easy to

implement in both analog and digital systems.

 It is a cost-effective modulation technique, which is one reason for its

widespread use in various applications.

 It is compatible with various transmission media, including radio waves,

optical fibers, and wired communication systems.

 It is efficient in terms of bandwidth utilization as it only requires two

levels of amplitude i.e. 0 and 1.

Disadvantages

 Susceptibility to Noise: ASK is more sensitive to noise and interference

compared to other modulation techniques like Frequency Shift Keying

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2nd Year DIGITAL COMMUNICATION CSE 2002

(FSK) or Phase Shift Keying (PSK). Fluctuations in the amplitude caused

by noise can lead to errors in data recovery at the receiver end.

 Bandwidth Inefficiency: ASK can be less bandwidth efficient compared

to other modulation techniques. It requires more bandwidth because it
uses changes in amplitude to encode data, which can limit the number of
bits transmitted in a given bandwidth.

 Power Inefficiency: Transmitting signals at varying amplitudes requires

more power compared to constant amplitude transmission. This can be a
disadvantage in scenarios where power efficiency is critical.

 Limited Data Rate: ASK is generally limited in terms of achievable data

rates compared to other modulation techniques. This limitation arises
from the constraints on how fast the amplitude can be changed without
distortion.

 Signal-to-Noise Ratio Sensitivity: ASK’s performance is highly

dependent on the signal-to-noise ratio (SNR). As the ratio between the
signal power and noise power decreases, the accuracy and reliability of
data transmission can be significantly affected.

 Lack of Phase or Frequency Information: ASK only utilizes changes in

amplitude to represent data, unlike other modulation techniques that
use phase or frequency changes. This limitation might make it more
vulnerable to certain types of interference and distortions.

Applications

Here comes a wide range of application of ASK

 ASK is commonly used in wireless communication systems, such as

keyless entry systems, remote controls, and radio frequency
identification (RFID) tags.

 In optical fiber communication, ASK is used to transmit digital data over

long distances.

 ASK is often used in digital broadcasting, including television and radio

transmissions, for transmitting audio and video signals.

 ASK can be used for low to medium data rate communication, such as in
binary data transmission over short distances.

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2nd Year DIGITAL COMMUNICATION CSE 2002

 ASK is used in medical telemetry systems for monitoring and

transmitting patient data.

Frequency Shift Keying (FSK) Modulator

It is the digital modulation technique in which the frequency of the carrier

signal varies according to the digital signal changes. FSK is a scheme of
frequency modulation.

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.

FSK Modulator

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2nd Year DIGITAL COMMUNICATION CSE 2002

The FSK modulator block diagram comprises of two oscillators with a clock and
the input binary sequence.

The two oscillators, producing a higher and a lower frequency signals, are
connected to a switch along with an internal clock. To avoid the abrupt phase
discontinuities of the output waveform during the transmission of the message,
a clock is applied to both the oscillators, internally. The binary input sequence
is applied to the transmitter so as to choose the frequencies according to the
binary input.

FSK Demodulator

There are different methods for demodulating a FSK wave. The main methods
of FSK detection are asynchronous detector and synchronous detector. The
synchronous detector is a coherent one, while asynchronous detector is a non-
coherent one.

Coherent(synchronous) FSK Detection

 In this synchronous FSK detection, the modulated wave got affected by

noise while reaching the receiver.
 So, this noise can be eliminated from using the bandpass filter (BPF).
Here at multiplier stage, the noisy FSK modulated signal is multiplied
with the carrier signal from the local oscillator device.
 Then the resultant signal passes from the BPF. Here this bandpass filter
is assigned to cut off frequency which is equal to the binary input signal
frequency.
 So the same frequencies can be allowed to the decision device. Here this
decision device gives 0 and 1 for space and mark frequencies of the FSK
modulated waveforms.

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2nd Year DIGITAL COMMUNICATION CSE 2002

BPSK – Binary Phase Shift Keying

Binary Phase-shift keying (BPSK) is a digital modulation scheme that conveys

data by changing, or modulating, two different phases of a reference signal (the
carrier wave).

In BPSK we represent symbols as phase shifts in the carrier signal during the
modulation process. A binary 1 is sent without any change in phase while a
binary 0 is sent with a phase shift of 180 degrees. This information about
phase shifts is encoded into the carrier signal to facilitate data transmission.

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2nd Year DIGITAL COMMUNICATION CSE 2002

BPSK Demodulator

By recovering the band-limited message signal, with the help of the mixer
circuit and the band pass filter, the first stage of demodulation gets completed.
The base band signal which is band limited is obtained and this signal is used
to regenerate the binary message bit stream.

In the next stage of demodulation, the bit clock rate is needed at the detector
circuit to produce the original binary message signal. If the bit rate is a sub-
multiple of the carrier frequency, then the bit clock regeneration is simplified.
To make the circuit easily understandable, a decision-making circuit may also
be inserted at the 2nd stage of detection.

Advantages of BPSK

 Simplicity: BPSK, which stands for Binary Phase Shift Keying is a

modulation scheme. It simplifies implementation, in hardware and
software by utilizing two phase states; 0 degrees and 180 degrees.

 Effective Operation with Reliability: It has ability to operate effectively

in the presence of noise or interference from signals ensuring reliable
performance.

 Less Power Consumption: BPSK consumes power compared to

alternative methods making it advantageous for battery powered devices.

 Easy Detection: Receivers find it easy to comprehend BPSK accurately

determining the frequency and phase of the transmitted signal.

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2nd Year DIGITAL COMMUNICATION CSE 2002

 Compatible: BPSK serves as a building block for complex modulation

schemes like QPSK (Quadrature Phase Shift Keying) and higher
order Quadrature Amplitude Modulation (QAM).

Disadvantages of BPSK

 Low Data Sending Rate: However when it comes to data transmission

speed, BPSK has limitations as it can only send one piece of data at a
time.

 Less Efficient: It inefficiently uses the signal space, same like using a
whole road for just one small car.

 Can be Tricky: In situations such as communication, signal bouncing

can pose challenges, for BPSK as it weakens the signal strength and
leads to potential issues.

 Limited Error Correction: BPSK does not provide as much inherent

error correction capability as more complex modulation schemes,
therefore, error correction needs to be added separately, which can
increase system complexity.

 Not for Huge Data: If you need to send a lot of data fast then BPSK
might not be the best choice.

Quadrature Phase Shift Keying

QPSK Modulator

The QPSK Modulator uses a bit-splitter, two multipliers with local oscillator, a
2-bit serial to parallel converter, and a summer circuit. Following is the block
diagram for the same.

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2nd Year DIGITAL COMMUNICATION CSE 2002

At the modulator’s input, the message signal’s even bits (i.e., 2 nd bit, 4th bit,
6th bit, etc.) and odd bits (i.e., 1st bit, 3rd bit, 5th bit, etc.) are separated by the
bits splitter and are multiplied with the same carrier to generate odd BPSK
(called as PSKI) and even BPSK (called as PSKQ). The PSKQ signal is anyhow
phase shifted by 90° before being modulated.

QPSK Demodulator

The QPSK Demodulator uses two product demodulator circuits with local
oscillator, two band pass filters, two integrator circuits, and a 2-bit parallel to
serial converter. Following is the diagram for the same.

The two product detectors at the input of demodulator simultaneously

demodulate the two BPSK signals. The pair of bits are recovered here from the
original data. These signals after processing, are passed to the parallel to serial
converter.

Differential Phase Shift Keying

n Differential Phase Shift Keying (DPSK) 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.

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2nd Year DIGITAL COMMUNICATION CSE 2002

DPSK Modulator

DPSK is a technique of BPSK, in which there is no reference phase signal.

Here, the transmitted signal itself can be used as a reference signal. Following
is the diagram of DPSK Modulator.

DPSK encodes two distinct signals, i.e., the carrier and the modulating signal
with 180° phase shift each. The serial data input is given to the XNOR gate and
the output is again fed back to the other input through 1-bit delay. The output
of the XNOR gate along with the carrier signal is given to the balance
modulator, to produce the DPSK modulated signal.

DPSK Demodulator

In DPSK demodulator, the phase of the reversed bit is compared with the phase
of the previous bit. Following is the block diagram of DPSK demodulator.

From the above figure, it is evident that the balance modulator is given the
DPSK signal along with 1-bit delay input. That signal is made to confine to
lower frequencies with the help of LPF. Then it is passed to a shaper circuit,
which is a comparator or a Schmitt trigger circuit, to recover the original binary
data as the output.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

DPSK modulator example with data-0010010011

QAM MODULATOR AND DEMODULATOR

 QAM (Quadrature Amplitude Modulation) is a method of combining two

amplitude modulation (AM) signals into a single channel.

 This approach helps double its effective bandwidth.

 QAM is also used with pulse AM (PAM) in digital systems, like wireless
applications.

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2nd Year DIGITAL COMMUNICATION CSE 2002

QAM Tx QAM Rx

The input data signals I and Q are amplitude modulated by sine and cosine
carrier waves, respectively. The two modulated signals are merged and then
processed in accordance with RF specifications. These signals are then
frequency-converted and amplification is performed to satisfy the final
frequency and amplitude criteria.

QAM Rx

In contrast, in demodulation, signals from the transmitter enter the

demodulator and are divided into two signals that are sent into separate
mixers. The local oscillator generates the needed frequency demodulation
signals, which are in-phase and quadrature signals.

Advantages of QAM

Quadrature amplitude modulation aside from increased channel capacity,

QAM has various other advantages, which are stated below.

One of the most significant advantages of QAM is its ability to sustain a high
data rate. As a result, the carrier signal can carry a certain number of bits.
Because of these benefits, it is preferred in wireless communication networks.

 The noise immunity of QAM is quite strong. Noise interference is quite

low as a result of this.

 It has a low mistake probability value.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

 QAM makes excellent utilization of channel capacity.

 The carrier signal may transport more bits, resulting in a high data rate.

 Because of the high noise immunity, there is less noise interference

during data transmission.

 The likelihood of incorrect values is negligible.

Applications

QAM is most commonly used in radio communications and data transmission

systems.

 The QAM technique has many uses in the radio communications area
because as data rates increase, so does the noise, but with this QAM
approach, noise interference is not an issue. Hence, an easy mode of
signal transmission is achievable with this QAM.

 In cellular technology, quadrature amplitude modulation is preferred in

wireless device technology.

 QAM has several uses in digital signal transmissions, such as digital

cable television and internet services.

M-ary modulation schemes.

Multi-level (M-ary) modulation techniques are used in digital communications

as the digital inputs with more than two modulation levels are allowed on the
transmitter’s input. Hence, these techniques are bandwidth efficient.

There are many M-ary modulation techniques. Some of these techniques,

modulate one parameter of the carrier signal, such as amplitude, phase, and
frequency.

M-ary ASK

This is called M-ary Amplitude Shift Keying (M-ASK) or M-ary Pulse Amplitude
Modulation (PAM).

The amplitude of the carrier signal, takes on M different levels.

Representation of M-ary ASK

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

M-ary FSK

This is called as M-ary Frequency Shift Keying (M-ary FSK).

The frequency of the carrier signal, takes on M different levels.

Representation of M-ary FSK

Where

for some fixed integer n.

M-ary PSK

This is called as M-ary Phase Shift Keying (M-ary PSK).

The phase of the carrier signal, takes on M different levels.

Representation of M-ary PSK

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Module-04 RBT Level-L1, L2

Waveform Coding:
Introduction to line codes, Line coding, properties, Power Spectral Density of
Unipolar / Polar RZ & NRZ, Bipolar NRZ, Manchester- ISI, Nyquist criteria for
distortion-less transmission, Pulse shaping – Correlative coding – M-ary
schemes – Eye pattern – Equalization

Introduction to line codes

The long-haul communication systems typically involve serial transmission of
data either synchronous or asynchronous. Such serial transmission of data
over the coaxial cables, optical fiber or RF link requires the suitable encoding of
binary data so that highs and lows in the signal can be detected appropriately.
Also, the signal during transmission through any type of communication
channel wired or wireless suffers from attenuation and distortion, which is
more severe in long haul data transmission channel. Hence for efficient
communication, tuning the signal properties to the characteristics of physical
channel is necessary.

Information to be transmitted can be encoded in either digital or analog signal

using different encoding techniques.

Line coding refers to the process of converting a sequence of binary digits i.e.,
bits or digital data into a digital signal. Line coding is implemented for digital
transmission of binary information. As seen in fig 1, at the sending end digital
data are encoded into a digital signal and at the receiving end original digital
data is recovered by decoding the digital signal.

Data are encoded in a way that the timing information of the bit stream is
maintained and the logic levels 1’s and 0’s can be detected.

The major application of line coding in data communication is to perform base

band modulation of the digital data.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Characteristics of Line Coding

Data Rate

Signal Rate or Baud Rate

Bandwidth

DC components

Baseline wandering

Self Synchronization

Error Detection

Noise Immunity

Complexity and implementation

cost

1.Data Rate: It represents the number of data bits transmitted per second. The
unit is bits per second. The data rate should be increased to increase the data
transmission speed.

2.Signal Rate or Baud Rate: Baud rate is the measure of signaling element or
symbols sent per second. A symbol in some case could be a binary logic level 0

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

or 1. In that case, baud rate is equal to bit rate. In other cases, the symbol is
one of the several discrete signal amplitudes or phase shifts each representing
two or more data bits. The unit is baud. Baud rate can be more or less than the
data rate. The signal rate should be decreased to decrease the bandwidth
requirements.

3.Bandwidth: Bandwidth is the frequency range over which an information

signal is transmitted. It dictates the amount of data sent over time and
influences the choice of digital modulation scheme. Digital signal has
theoretically infinite bandwidth but practically the effective bandwidth is finite.
The baud rate determines the digital signal bandwidth required.

4.DC components: Very low frequencies around zero which are the
consequence of existence of constant voltage level in a digital signal are known
as Direct current or DC components. These components significantly induce
problems in the systems using transformer coupling and since most channels
are band pass, they may not support the low frequency signals. Hence line
coding schemes with no DC components are required for such systems.

5.Baseline wandering: A receiver will evaluate the average power of the

received signal (called the baseline) and use that to determine the value of the
incoming data elements. If the incoming signal does not vary over a long period
of time, the baseline will drift and thus cause errors in detection of incoming
data elements. A good line encoding scheme will prevent long runs of fixed
amplitude.

6.Self Synchronization: The clocks at the sender and the receiver must have
the same bit interval. If the receiver clock is faster or slower it will misinterpret
the incoming bit stream. A transmitted self-synchronizing digital signal
includes timing information in the data. Clock signal is recovered from the
receiving bit sequence by observing the transitions and if sufficient transitions
exist, a good recovery of the clock is guaranteed. Such a signal is said to be
self-clocking.

7.Error Detection: Ideally digital communication system should be free from

any error that gets incorporated into the data during transmission. Hence it is
desirable to implement some modifications in the data allowing built in error
detection capabilities. This can enhance the receiver’s capability to detect and
correct error. The transmission system used and adopted digital encoding
techniques influence the error detection and correction.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

8.Noise Immunity: The selected encoding scheme should be immune to noise

and external interferences. We are already aware that any signal during
transmission through any type of communication channel suffers from
attenuation and distortion. So the requirement is to encode the signal using an
encoding scheme imparting high noise immunity.

9.Complexity and implementation cost: An encoding scheme that is

complex, is costly to implement. Interpretation of any scheme that involves
more signal levels is difficult. So the encoding technique should be simple so as
to prevent high cost of implementation.

Various Line Coding Scheme.

Line codes are classified according to the assigned polarity of voltage levels to
represent the binary data. We can categorize line coding schemes into three
major types as shown in figure.

LINE
CODING

UNIPOLAR POLAR BIPOLAR

NRZ NRZ RZ BIPHASE AMI

PSEUDOTERNA
NRZ-L MANCHESTER
RY

DIFFERENTIAL
NRZ-I
MANCHESTER

Unipolar RZ and NRZ

Unipolar RZ:

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Example: Binary Data 10110100

Unipolar NRZ:

Example: Binary Data 10110100

Polar RZ & NRZ:

Polar RZ

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Example: Binary Data 10110100

Polar NRZ

Example: Binary Data 10110100

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2nd Year DIGITAL COMMUNICATION CSE 2002

Split Phase Manchester:

Example: Binary Data 10110100

Bipolar NRZ

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2nd Year DIGITAL COMMUNICATION CSE 2002

Example: Binary Data 10110100

Polar Quaternary NRZ (Natural Code): The massage bits are

grouped in the blocks of two. There are four possible combinations
00,01,10,11. To these four combinations four amplitudes are
assigned

Example: Binary Data 10110100

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Gray code:

Binary to Gray conversion :

The Most Significant Bit (MSB) of the gray code is always equal to
the MSB of the given binary code.

Other bits of the output gray code can be obtained by XORing

binary code bit at that index and previous index.

The message data:10110100

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Example: With given binary data sequence 011010110

construct the polar octal format of NRZ type using 1)Natural
coding 2)Gray code.

Natural coding

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Gray code

Example:For the given binary data sequence10011011 draw the

following case

1)Polar NRZ 2)Manchester RZ 3)Gray code NRZ

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Nyquist criteria for distortion-less transmission

Nyquist criterion: It is a set of conditions for sampling and

stability in systems that can be modeled by linear differential
equations.

Nyquist criterion mainly depending on two parameters

 Sampling
 Stability

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

 Sampling

A signal must be sampled at least twice its highest frequency

component to avoid losing information about the signal. If the
sampling frequency is lower, a phenomenon called aliasing occurs.

 Stability

The Nyquist criterion is a graphical technique for determining if an

unstable linear time invariant system can be stabilized using a
negative feedback loop.

The incoming binary data sequence {bk} consists of symbol 0 &1

each of bit duration Tb.

The pulse amplitude modulation modifies the binary data sequence

into new data sequence of short pulses whose amplitude a k is in
polor form.

The sequence of short pulse are applied to transmit filter to produce

the impulse response g(t), producing the transmitted signal

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

The signal s(t) is modified as a result of transmission through

channel of impulse response h(t).

In the channel it will add the random noise to the signal at the
receiver input.

The noisy signal x(t) is passed through receiver filter of impulse

response c(t).The resulting filter output y(t) is sampled
synchronously with transmitter with sampling instant clock or
timing signal is extracted from the receiver filter output.

Finally the sequence of samples applied to the decision device, the

amplitude of each sample compared with threshold λ. if threshold
λ greater than y(t) a decision made in the favor of symbol 1, if
threshold λ lesser than y(t) a decision made in the favor of symbol
0

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.
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2nd Year DIGITAL COMMUNICATION CSE 2002

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.

Inter Symbol Interference

This is a form of distortion of a signal, in which one or more

symbols interfere with subsequent signals, causing noise or
delivering a poor output. Causes of ISI The main causes of ISI are −
 Multi-path Propagation

 Non-linear frequency in channels

The ISI is unwanted and should be completely eliminated to get a

clean output. The causes of ISI should also be resolved in order to
lessen its effect. To view ISI in a mathematical form present in the
receiver output, we can consider the receiver output. The receiving
filter output y(t) is sampled at time ti=iTb (with i taking on integer
values), yielding –

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

In the above equation, the first term µa i is produced by the ith

transmitted bit. The second term represents the residual effect of all
other transmitted bits on the decoding of the ith bit. This residual
effect is called as Inter Symbol Interference. In the absence of ISI,
the output will be –

This equation shows that the ith bit transmitted is correctly

reproduced. However, the presence of ISI introduces bit errors and
distortions in the output. While designing the transmitter or a
receiver, it is important that you minimize the effects of ISI, so as to
receive the output with the least possible error rate.

Correlative Coding

So far, we’ve discussed that ISI is an unwanted phenomenon and

degrades the signal. But the same ISI if used in a controlled
manner, is possible to achieve a bit rate of 2W bits per second in a
channel of bandwidth W Hertz. Such a scheme is called as
Correlative Coding or Partial response signaling schemes.

Since the amount of ISI is known, it is easy to design the receiver

according to the requirement so as to avoid the effect of ISI on the
signal. The basic idea of correlative coding is achieved by
considering an example of Duo-binary Signaling.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Duo-binary Signaling

The name duo-binary means doubling the binary system’s

transmission capability. To understand this, let us consider a
binary input sequence {ak} consisting of uncorrelated binary digits
each having a duration Ta second. In this, the signal 1 is
represented by a +1 volt and the symbol 0 by a -1 volt. Therefore,
the duo-binary coder output ck is given as the sum of present
binary digit ak and the previous value ak-1 as shown in the following
equation.

ck= ak + ak-1
The above equation states that the input sequence of uncorrelated
binary sequence {ak} is changed into a sequence of correlated three
level pulses { ck }. This correlation between the pulses may be
understood as introducing ISI in the transmitted signal in an
artificial manner.

Eye Pattern

An effective way to study the effects of ISI is the Eye Pattern. The
name Eye Pattern was given from its resemblance to the human eye
for binary waves. The interior region of the eye pattern is called the
eye opening. The following figure shows the image of an eye-
pattern.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Jitter is the short-term variation of the instant of digital signal,

from its ideal position,which may lead to data errors.

When the effect of ISI increases, traces from the upper portion to
the lower portion of the eye-opening increases and the eye gets
completely closed, if ISI is very high.

An eye pattern provides the following information about a particular

system.

 Actual eye patterns are used to estimate the bit error rate and
the signal-to-noise ratio.
 The width of the eye opening defines the time interval over
which the received wave can be sampled without error from
ISI.
 The instant of time when the eye opening is wide, will be the
preferred time for sampling.
 The rate of the closure of the eye, according to the sampling
time, determines how sensitive the system is to the timing
error.
 The height of the eye opening, at a specified sampling time,
defines the margin over noise.

Hence, the interpretation of eye pattern is an important

consideration.

Equalization

For reliable communication to be established, we need to have a

quality output. The transmission losses of the channel and other
factors affecting the quality of the signal, have to be treated. The
most occurring loss, as we have discussed, is the ISI.

To make the signal free from ISI, and to ensure a maximum signal
to noise ratio, we need to implement a method called Equalization.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

The following figure shows an equalizer in the receiver portion of the

communication system.

The noise and interferences which are denoted in the figure, are
likely to occur, during transmission. The regenerative repeater has
an equalizer circuit, which compensates the transmission losses by
shaping the circuit. The Equalizer is feasible to get implemented.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Module-05 RBT Level-L1, L2

Advance communication systems:

Spreading and its types, DSSS, FHSS, CDMA, SS7, OFDM- 64/128 QAM,
NOMA, Introduction to cellular communication & GSM. Introduction to
wireless communication, satellite communication and digital satellite
communication, types of satellite systems (LEO, MEO and GEO)

What is Spread Spectrum?

The definition of Spread – Spectrum modulation may be stated in two parts.

1. Spread Spectrum is a mean of transmission in which the data sequence

occupies a BW (Bandwidth) in excess of the minimum BW necessary to
transmit it.
2. The Spectrum Spreading is accomplished before transmission through the
use of a code that is independent of the data sequence. The Same code is used
in the receiver to despread the received signal so that the original data
sequence may be recovered.

which provides advantages such as

 Increased resistance to interference.
 Improved security.
 Enhanced privacy.

Functions of each block:

Channel encoder: The channel encoder adds some redundant bits to the input
bit sequence in properly defined format.

Pseudorandom generator: It produced a pseudorandom or pseudonoise

binary valued sequence.

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2nd Year DIGITAL COMMUNICATION CSE 2002

Modulator: Pseudonoise (PN) binary valued sequence is impressed on the

transmitted signal at modulator.

Channel: Medium through which signal travels towards receiver.

Demodulator: The impressed PN sequence removed from the received signal at

the demodulator.

Channel decoder: The decoder at the receiver uses the coded bits to
reconstruct error free accurate bit sequence and reduce the effects of channel
noise and distortion.

Principles of Spread Spectrum process

To allow redundancy, it is necessary that the bandwidth allocated to each


station should be much larger than needed.
 The spreading process occurs after the signal is created by the source.
Conditions of Spread Spectrum are

The spread spectrum is a type of modulation where modulated signal BW


is much larger than the baseband signal BW i.e. spread spectrum is a
wide band scheme.
 A special code (pseudo noise) is used for spectrum spreading and the
same code is to be used to despread the signal at the receiver.
Characteristics of Spread Spectrum

Higher channel capacity.


 Ability to resist multipath propagation.
 They cannot easily intercept any unauthorized person.
 They are resistant to jamming.
 The spread spectrum provides immunity to distortion due to multipath
propagation.
 The spread spectrum offers multiple access capabilities.
Techniques used for Spread Spectrum

Frequency Hopping Spread Spectrum (FHSS)

 Direct Sequence Spread Spectrum (DSSS)
 Code Division Multiplexing Access(CDMA)
Frequency Hopping Spread Spectrum (FHSS)

Frequency hopping also known as frequency hopping spread spectrum (FHSS)

is a method of transmitting radio signals by rapidly switching the carrier

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2nd Year DIGITAL COMMUNICATION CSE 2002

between different frequency channels. Switching occurs based on a pseudo-

random sequence known to the transmitter and the receiver.

 The above figure shows the Block diagram of the FHSS system for both
transmitter and receiver.
 For transmission, binary data are fed into a modulator using some
digital-to-analog encoding scheme, such as Frequency Shift Keying
(FSK) or Binary Phase Shift Keying (BPSK).

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2nd Year DIGITAL COMMUNICATION CSE 2002

A PN source serves as an index into a table of frequencies each K bit on


the PN source specifies one of the 2k carrier frequencies.
 At each successive interval, a new carrier frequency is selected.
 This frequency is then modulated by the signal produced from the initial
modulator to produce a new signal with the same shape.
 On reception, the spread spectrum signal is demodulated using the same
sequence of PN-derived frequencies and then demodulated to produce
the output data.
Advantages of FHSS

Synchronization is not greatly dependent on distance.

 Processing Gain is higher than DSSS.
Disadvantages of FHSS

The bandwidth of the FHSS system is too large (in GHz).


 Complex and expensive Digital frequency synthesizers are required.
Applications of FHSS

FHSS is used in Bluetooth

 Military Communications
 Walkie-Talkies
 Wireless Local Area Networks (WLANs)
 Remote Controls
Direct Sequence Spread Spectrum (DSSS)

In Direct Sequence Spread Spectrum, the bandwidth of the original

signal is also expanded by a different technique. Here, each data bit
is replaced with n bits using a spreading code called chips, and the
bit rate of the chip is called as chip-rate. The chip rate is n times
the bit rate of the original signal. The below Figure shows the DSSS
block diagram.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

DSSS Sender

Step 1: Binary data and PN sequence applied to the multiplier to get

product signal

Step 2: Product or Spread signal is again modulated via a radio

modulation

Step 3: This shifts the signal to carrier frequency

Step 4: Signal is transmitted

DSSS Receiver

Step: 1 Demodulate the received signal

Step: 2 Generate the same pseudo random sequence as the

transmitter

Step: 3 Find the correlation with the pseudo random sequence by

finding the product and integrating the products

Step: 4 Decision unit decides if this sum represent a binary 1 or 0

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2nd Year DIGITAL COMMUNICATION CSE 2002

Advantages:

There are some advantages of direct sequence spread

spectrum which are given below,

o It has best discrimination against multipath signals.

o It avoids intentional interference such as jamming effectively.

o Performance of DSSS system in presence of noise is better than

FHSS system.

Disadvantages:

There are some disadvantages of direct sequence spread

spectrum which are given below,

o This system becomes slow due to large acquisition time.

o It requires wideband channel with small phase distortion.

o The pseudo noise generator should generate sequence at high

rates.

Applications:

There are some applications of direct sequence spread

spectrum which are given below,

o It is used in the low probability of intercept signal.

o It is used in military and many commercial applications.

o Code division multiple access with direct sequence spread

spectrum.

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2nd Year DIGITAL COMMUNICATION CSE 2002

Example1 A pseudo random sequence is generated using a feed

back shift register of length m=4. The chip rate is 107 chips per
second. Find the following a) PN sequence length b) Chip duration
of PN sequence c) PN sequence period

Solution a) Length of PN sequence N = 2m-1= 2 4 -1 =15

b) Chip duration Tc = 1/chip rate =1/107 = 0.1µsec

c) PN sequence period T = NTc = 15 x 0.1µsec = 1.5µsec

Example2 A direct sequence spread binary phase shift keying

system uses a feedback shift register of length 19 for the generation
of PN sequence. Calculate the processing gain of the system.
Solution

Given length of shift register = m =19

Therefore length of PN sequence N = 2 m - 1 = 2 19 - 1

Processing gain PG = Tb/Tc =N in db

=10log10N = 10 log10 (219 ) = 57d

Direct Sequence Code Division multiple Access (DS-CDMA)

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Signal transmission consists of the following steps.

 A pseudo-random code is generated, different for each channel

and each successive connection.
 The information data is spread by pseudorandom code. The
resulting signal modulates a carrier.
 The modulated carrier is amplified and broadcast.
 Signal reception consists of the following steps. The carrier is
received and amplified.
 The received signal is mixed with a local carrier to recover the
spread digital signal.
 A pseudorandom code is generated, matching the anticipated
signal.
 The receiver acquires the received code and phase locks its
own code to it.
 The received signal is correlated with the generated code,
extracting the information data

1. Advantages and Applications:

Advantages:

1. Efficient use of bandwidth: DS-CDMA allows multiple users to

share the same bandwidth.
2. Robustness against interference: Spread spectrum enhances
signal robustness.
3. Secure communication: Unauthorized users find the
transmitted signal as noise due to the unique spreading code.
Applications:

4. Mobile communication systems (e.g., 3G, 4G, and beyond)

5. Satellite communication

6. Military and secure communication

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2nd Year DIGITAL COMMUNICATION CSE 2002

Signaling System 7 (SS7)

Signaling System 7 (SS7) is an international telecommunication

protocol standard that defines how the network elements in a
public switched telephone network (PSTN) exchange information
and control signals. Nodes in an SS7 network are called signaling
points.

Terminologies Used

SSP: Signal switching point performs organization, termination, and

switching of calls
SCP: Signal Control Point is a database that takes care of advanced
call processing features.
STP: Signal transfer point performs routing functionalities and
allow the exchange of signaling information with each other.
PSTN: PSTN uses copper wires to transmit voice data as analog or
digital signals. A dedicated channel is created for each call,
connecting the caller's phone to the recipient's phone

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2nd Year DIGITAL COMMUNICATION CSE 2002

Tandem: It is called a Trunk Automatic Exchange that allows

connection between two exchanges that are not having a direct
trunk.
Trunk: the link between two exchanges.

Services offered by SS7 include the following:

 call setup, routing and teardown

 call forwarding
 automated voicemail
 call waiting
 conference calling
 caller ID
 subscriber authentication and extended billing
 toll-free (800 and 888) and toll (900) calls
 SMS
 mobile phone roaming and tracking

Orthogonal Frequency Division Multiplexing (OFDM)

In data communications and networking, orthogonal frequency-

division multiplexing (OFDM) is a method of digital data
modulation, whereby a single stream of data is divided into several
separate sub-streams for transmission via multiple channels.

OFDM uses the principle of frequency division multiplexing (FDM),

where the available bandwidth is divided into a set of sub-streams
having separate frequency bands. OFDM was introduced in 1966 by
Chang at Bell Labs and was improved by Weinstein and Ebert in
1971.

Working Principle of OFDM

OFDM is a specialised FDM having the constraint that the sub-

streams in which the main signal is divided, are orthogonal to each
other. Orthogonal signals are signals that are perpendicular to each

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

other. A main property of orthogonal signals is that they do not

interfere with each other.

When any signal is modulated by the sender, its sidebands spread

out either side. A receiver can successfully demodulate the data
only if it receives the whole signal.

In case of FDM, guard bands are inserted so that interference

between the signals, resulting in cross-talks, does not occur.
However, since orthogonal signals are used in OFDM, no
interference occurs between the signals even if their sidebands
overlap. So, guard bands can be removed, thus saving bandwidth.
The criteria that needs to be maintained is that the carrier spacing
should be equal to the reciprocal of the symbol period.

Usages
OFDM is used in the following area −
 Wi-Fi
 DSL internet access
 4G wireless communications
 digital television
 radio broadcast services

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

OFDM- 64/128 QAM

Serial to Parallel Converter: It accepts serial input data and

outputs parallel data

64-QAM mapper: A 64-QAM mapper works by taking binary input

and producing complex data symbols as outputs like Grouping
data, Determining I and Q values.

Inverse Fast Fourier Transform (IFFT): Inverse Fast Fourier

Transform (IFFT) works by converting a set of complex data points
into an equal number of points in the time domain.

Cyclic prefix (CP): The CP works by copying the end of the OFDM
symbol and placing it at the beginning of the symbol.

Parallel-to-serial converter: parallel-to-serial converter changes

parallel data into serial data so that it can be transmitted to a
peripheral device.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Digital-to-Analog Converter (DAC): A digital-to-analog converter

(DAC) works by taking digital samples and stitching them together
to create an analog waveform.

IQ modulator: An IQ modulator is a device that converts baseband

information into RF signals, by modulating with both in-phase (I)
and quadrature (Q) inputs.

Introduction to cellular communication & GSM.

A basic analog cellular system consists of three subsystems: a

mobile unit, a cell site, and a mobile telephone switching office
(MTSO), as Fig. 1.1 shows, with connections to link the three
subsystems.

Mobile units: A mobile telephone unit contains a control unit, a

transceiver, and an antenna system.

Base Station controller (BSC): It is a telecommunication network

component responsible for the control of one or more Base
Transceiver Stations (BTS)

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

MTSO: The switching office, the central coordinating element for all
cell sites, con-tains the cellular processor and cellular switch. It
interfaces with telephone company zone offices, controls call
processing, provides operation and maintenance, and han-dles
billing activities.

Home Location Register(HLR): which is a database that mobile

network providers use to track their customers. It contains
information about a mobile network's users, such as their phone
numbers, service entitlements, and location.

Visitor Location Register (VLR): A Visitor Location Register (VLR)

is a database in a cellular network that manages calls and tracks
the location of mobile devices when they are roaming outside of
their home network.

Gateway Mobile Switching Center (GMSC): A Gateway Mobile

Switching Center (GMSC) is a mobile network element that routes
calls to and from outside a mobile network.

Difference between 1g 2g 3g 4g 5g in mobile communication

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Introduction to wireless communication.

Wireless communication takes places over free space through RF

(radio frequency), one device, a Transmitter, sends signal to another
device, a Receiver. wo devices (transmitter and receiver) must use
same frequency (or channel) to be able to communicate with each
other. If a large number of wireless devices communicate at same
time, radio frequency can cause interference with each other.
Interference increases as no of devices increases.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Wireless - Advantages

Wireless communication has several advantages; the most

important ones are discussed below −

Cost effectiveness

Wired communication entails the use of connection wires. In

wireless networks, communication does not require elaborate
physical infrastructure or maintenance practices. Hence the cost is
reduced.

Flexibility

Wireless communication enables people to communicate regardless

of their location. It is not necessary to be in an office or some
telephone booth in order to pass and receive messages.

Convenience

Wireless communication devices like mobile phones are quite

simple and therefore allow anyone to use them, wherever they may
be. There is no need to physically connect anything in order to
receive or pass messages.

Speed

Improvements can also be seen in speed. The network connectivity

or the accessibility were much improved in accuracy and speed.

Accessibility

The wireless technology helps easy accessibility as the remote areas

where ground lines can’t be properly laid, are being easily
connected to the network.

Constant connectivity

Constant connectivity also ensures that people can respond to

emergencies relatively quickly.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Satellite communication and digital satellite communication.

Satellite communication is transporting information from one

place to another using a communication satellite in orbit around
the Earth.

Satellite Communication Block Diagram.

Three stages are involved, in the working of satellite

communications. These are:
 Uplink
 Transponders
 Downlink

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

In the first stage, the signal from the television broadcast on the
other side of the earth is first beamed up to the satellite from the
ground station on the earth. This process is known as uplink.

The second stage involves transponders such as radio receivers,

amplifiers, and transmitters. These transponders boost the
incoming signal and change its frequency so that the outgoing
signals are not altered. Depending on the incoming signal sources,
the transponders vary.

The final stage involves a downlink in which the data is sent to the
other end of the receiver on the earth. It is important to understand
that usually, there is one uplink and multiple downlinks.

Types of satellite systems (LEO, MEO and GEO).

The three main types of satellite systems are

 Low Earth Orbit (LEO)

 Medium Earth Orbit (MEO)
 Geostationary Orbit (GEO)

Low Earth Orbit (LEO): Satellites in LEO orbit are relatively close
to Earth's surface, typically at an altitude of less than 2,000
km. LEO satellites offer low latency, but require more satellites
because of their smaller coverage area.
Medium Earth Orbit (MEO): MEO satellites offer a balance between
latency and coverage.

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
2nd Year DIGITAL COMMUNICATION CSE 2002

Geostationary Orbit (GEO): GEO satellites are positioned to

remain fixed over a single point on Earth's surface. GEO satellites
have an orbital period of 24 hours, which is the same as Earth's
rotational period. GEO satellites offer continuous coverage over a
fixed area, but have higher latency.

Difference between LEO,MEO and GEO

ORBITS LEO MEO GEO

Orbital Period 1.5 to 2h 5 to10h 24h
Altitude 500 to 1500Km 8000 to 35863Km
Range 1800Km
Visibility 15 to 20 2 to 8 hr/pass Permanent
Duration min/pass
Elevation Rapid variations; Slow No variations;
high and low variations; low angles at
angles high angles high latitudes
Round Trip Several Ten ≈250 ms
Propagation milliseconds milliseconds
Delay
Instantaneous
ground
coverage ≈6000km ≈12000 to ≈16000km
(Diameter at 15000km
100
elevation)
Example of Iridium Odyssey Intelstat
System Globlaster Inmarsat Interspoutnik
Teledesic Inmarsat
Skybridge
Orcomm

School of Computer Science Engg. & Applications Dr. Jagadish S. Jakati, M.Tech. Ph.D
 School of Computer Science, Engineering and Applications
Digital Communication _CSE 2002_Question Bank

S.No. Questions CO RBT

Module-01
01 Distinguish between Random variable and Random Process 1 L1, L2
02 Define Joint Distribution and list out the Properties 1 L1, L2
03 Define Cumulative Distribution Function and list out the Properties 1 L1, L2
04 Define Probability Density Function and list the steps How to Find Probability 1 L1, L2
from Probability Density Function
05 Distinguish between Probability Density Function Vs Cumulative Distribution 1 L1, L2
Function
06 With neat block diagram explain Sampling Process in Digital Communication 1 L1, L2
07 Write short note on Quantization 1 L1, L2
08 Distinguish between Uniform and Non-uniform Quantization 1 L1, L2
09 List the Advantages & Applications of Quantization. 1 L1, L2
Module-02
01 With neat block diagram Explain DPCM Tx & Rx 2 L1, L2
02 With neat block diagram Explain PCM Tx & Rx 2 L1, L2
03 Write short note on White noise in Pulse Code Modulation 2 L1, L2
04 With neat block diagram Explain DM Tx & Rx 2 L1, L2
05 Define Companding and Explain Model of Companding 2 L1, L2
06 List the advantages and disadvantages of Compnamding 2 L1, L2
07 With neat block diagram Explain ADM Tx & Rx 2 L1, L2
08 Explain the characteristics of Compnamding. 2 L1, L2
09 Explain steps for non-uniform quantization using μ law companding 2 L1, L2
Module-03
01 With neat block diagram and waveforms, Explain Amplitude Shift Keying 3 L1, L2
(ASK)
02 With neat block diagram and waveforms, Explain Frequency Shift Keying 3 L1, L2
(FSK) Modulator
03 With neat block diagram and waveforms, Explain BPSK – Binary Phase Shift 3 L1, L2
Keying
04 With neat block diagram and waveforms, Explain Quadrature Phase Shift 3 L1, L2
Keying
05 With neat block diagram and waveforms, Explain Differential Phase Shift 3 L1, L2
Keying
06 For given binary data -0010010011 find DPSK modulator 3 L1, L2
07 With neat block diagram and waveforms, QAM modulator and demodulator 3 L1, L2
08 Write short note on M-ary modulation schemes 3 L1, L2
Module-04
01 Define line code? Mention the Characteristics of Line Coding 4 L1, L2
02 With given binary data sequence 011010110 4 L1, L2
construct 1)Polar NRZ 2)Manchester RZ 3)Gray code NRZ 4) Polar
Quaternary NRZ
03 For the given binary data sequence10011011 draw the following case 1)Polar 4 L1, L2
RZ & NRZ 2)Manchester RZ 3)Gray code NRZ 4) Unipolar RZ and NRZ 5)
Split Phase Manchester
04 With neat block diagram, Explain distortion-less transmission in digital 4 L1, L2
communication
05 Write short note on Inter Symbol Interference 4 L1, L2
06 Write short note on Correlative Coding 4 L1, L2
07 Write short note on Eye Pattern 4 L1, L2
08 With neat block diagram, Explain Equalization 4 L1, L2
Module-05
01 Define Spread Spectrum? With neat block diagram, Explain model of Spread 5 L1, L2
Spectrum
02 With neat block diagram, Explain Frequency Hopping Spread Spectrum 5 L1, L2
(FHSS)
03 With neat block diagram, Explain Direct Sequence Spread Spectrum (DSSS) 5 L1, L2
04 With neat block diagram, Explain Code Division Multiplexing Access(CDMA) 5 L1, L2
05 With neat block diagram, Explain Signaling System 7 (SS7) 5 L1, L2
06 With neat block diagram, Explain OFDM- 64/128 QAM 5 L1, L2
07 Explain cellular communication 5 L1, L2
08 Distinguish between 1g 2g 3g 4g 5g in mobile communication 5 L1, L2
09 Write short note on wireless communication 5 L1, L2
10 With neat block diagram, Explain Satellite Communication 5 L1, L2
11 Write short note on LEO, MEO and GEO 5 L1, L2
12 Distinguish between LEO,MEO and GEO 5 L1, L2

ALL THE BEST