UNIT-2 INFORMATION THEORY AND NOISE

AIM & OBJECTIVES

 To understand the fundamental concepts of information theory.

 To understand channel capacity and coding Techniques.
 To understand noise in Analog Modulation Techniques.

PRE-TEST MCQ

[Link] radio signal on assigned frequency.

a) Splatter
b) RFI
c) Noise
d) EMI

[Link] system performance is limited by

a) Available signal power
b) Background noise
c) Bandwidth limits.
d) All the above

3.A binary source is a device that generates

a) 0
b) 1
c) Both
d) Only 0

[Link] of an event depends only on

a) Its probability of occurrence
b) Dependent on its content.
c) Both
d) None of the above

[Link] source can be classified as having

a) Memory
b) Being memoryless
c) Both
d) Only memory
[Link] factor for a system is defined as the ratio of

a. Input noise power (Pni) to output noise power (Pno)

b. Output noise power (Pno) to input noise power (Pni)
c. Output noise power (Pno) to input signal power (Psi)
d. Output signal power (Pso) to input noise power (Pni)

[Link] Factor(F) and Noise Figure(NF) are related as

a. NF = 10 log10(F)
b. F = 10 log10(NF)
c. NF = 10 (F)
d. F = 10 (NF)

[Link] noise temperature at a resistor depends upon

a. Resistance value
b. Noise power
c. Both a and b
d. None of the above

[Link] power at the resistor is affected by the value of the resistor as

a. Directly proportional to the value of the resistor

b. Inversely proportional to the value of the resistor
c. Unaffected by the value of the resistor
d. Becomes half as the resistance value is doubled

[Link] noise is produced by lighting discharges in thunderstorms?

a) White noise
b) Extraterrestrial noise
c) Industrial noise
d) Atmospheric noise

PREREQUISITES
Basic Knowledge of Electronic Devices, Digital System Design,
Information theory.
 2.1 Entropy

Information:
Information of an event depends only on its probability of
occurrence and is not dependent on its content. The randomness of
happening of an event and the probability of its prediction as a news
is known as information. The message associated with the least
likelihood event contains the maximum information.

Information is a non-negative quantity: I (p) ≥ 0.

If an event has probability 1, we get no information from the
occurrence of the event: I (1) = 0.
If two independent events occur (whose joint probability is the
product of their individual probabilities), then the information we get
from observing the events is the sum of the two information:
I (p1 * p2 ) = I (p1 ) + I (p2 ).

I (p) is monotonic and continuous in p.

An information source may be viewed as an object which produces

an event, the outcome of which is selected at random according to a
probability distribution. The set of source symbols is called the
source alphabet and the elements of the set are called symbols or
letters Information source can be classified as having memory or
being memory-less. A source with memory is one for which a current
symbol depends on the previous symbols.
A memory-less source is one for which each symbol produced is
independent of the previous symbols. A discrete memory-less source
(DMS) can be characterized by the list of the symbol, the probability
assignment of these symbols and the specification of the rate of
generating these symbols by the source.

The amount of information contained in an event is closely related to

its uncertainty. A mathematical measure of information should be a
function of the probability of the outcome and should satisfy the
following axioms
a) Information should be proportional to the uncertainty of an
outcome
b) Information contained in independent outcomes should add up.
Information Content of a Symbol
(i.e. Logarithmic Measure of Information):

Let us consider a DMS denoted by ‗x‘ and having alphabet

{x1, x2, ……, xm}.
The information content of the symbol xi, denoted by I (𝑥𝑖) is defined
by
𝐼 (𝑥𝑖) = 𝑙𝑜𝑔𝑏 (1/ 𝑝(𝑥𝑖) ) = −𝑙𝑜𝑔𝑏 𝑝(𝑥𝑖)

where p(𝑥𝑖) is the probability of occurrence of symbol 𝑥𝑖.

For any two independent source messages xi and xj with probabilities

𝑃𝑖 and 𝑃𝑗 respectively and with joint probability P (𝑥𝑖 , 𝑥𝑗) = Pi Pj, the
information of the messages is the addition of the information in
each message. 𝐼𝑖𝑗 = 𝐼𝑖 + 𝐼𝑗 .

Note that I(xi ) satisfies the following properties.

1. I(xi ) = 0 for P(xi ) = 1
2. I(xi ) ≥ 0
3. I(xi ) > I(xj ) if P(xi ) < P(xj )
4. I(xi , xj ) = I(xi ) + I(xj ) if xi and xj are independent

Unit of I (xi ): The unit of I (xi ) is the bit (binary unit) if b = 2, Hartley
or decit if b = 10 and nat (natural unit) if b = e. it is standard to use
b = 2.
log2(a) = ln(a) / ln(2) = log(a) / log(2)

Entropy (i.e. Average Information):

Entropy is a measure of the uncertainty in a random variable.
The entropy, H, of a discrete random variable X is a measure of the
amount of uncertainty associated with the value of X. For
quantitative representation of average information per symbol we
make the following assumptions:

 The source is stationary so that the probabilities may remain

constant with time.
 The successive symbols are statistically independent and come
from the source at an average rate of ‗r‘ symbols per second.

The quantity H(X) is called the entropy of source X. it is a measure of

the average information content per source symbol. The source
entropy H(X) can be considered as the average amount of uncertainty
within the source X that is resolved by the use of the alphabet.

H(X) = E [I(xi)] = - ΣP(xi) I(xi) = - ΣP(xi)log2 P(xi) b/symbol.

Entropy for Binary Source:

H(X) = − 1/2 log2 (1/2) − 1/2 log2 (1/2) = 1 bit/symbol.

The source entropy H(X) satisfies the relation: 0 ≤ H(X) ≤ log 2 m,

where m is the size of the alphabet source X.

Properties of Entropy:
1) 0 ≤𝐻 (𝑋) ≤ log2𝑚 ; m = no. of symbols of the alphabet of source X.
2) When all the events are equally likely, the average uncertainty
must have the largest value i.e. log2𝑚 ≥𝐻(𝑋)
3) H (X) = 0, if all the P(xi ) are zero except for one symbol with P = 1.

Information Rate:

If the time rate at which X emits symbols is ‗r‘ (symbols s), the
information rate R of the source is given by
R = r H(X) b/s
R is the information rate. H(X) = Entropy or average information.

Using the input probabilities P(xi), output probabilities P(yj),

transition probabilities P(yj/xi) and joint probabilities P(xi,yj), various
entropy functions for a channel with m inputs and n outputs are
defined
H (X) is the average uncertainty of the channel input and H (Y) is the
average uncertainty of the channel output. The conditional entropy H
(X/Y) is a measure of the average uncertainty remaining about the
channel input after the channel output has been observed. H (X/Y) is
also called equivocation of X with respect to Y. The conditional
entropy H (Y/X) is the average uncertainty of the channel output
given that X was transmitted.

The joint entropy H (X, Y) is the average uncertainty of the

communication channel as a whole. Few useful relationships among
the above various entropies are as under:
a. H (X, Y) = H (X/Y) + H (Y)
b. H (X, Y) = H (Y/X) + H (X)
c. H (X, Y) = H (X) + H (Y)
d. H (X/Y) = H (X, Y) – H (Y)
X and Y are statistically independent.

The conditional entropy or conditional uncertainty of X given random

variable Y is the average conditional entropy over Y. The joint entropy
of two discrete random variables X and Y is merely the entropy of
their pairing: (X, Y), this implies that if X and Y are independent,
then their joint entropy is the sum of their individual entropies.
 2.2 Discrete Memoryless Channels

Channel Representation:
A communication channel may be defined as the path or
medium through which the symbols flow to the receiver end. A DMC
is a statistical model with an input X and output Y. Each possible
input to output path is indicated along with a conditional probability
P (yj |xi ), where P (yj |xi ) is the conditional probability of obtaining
output yj given that the input is x1 and is called a channel transition
probability. A channel is completely specified by the complete set of
transition probabilities. The channel is specified by the matrix of
transition probabilities [P(Y|X)]. This matrix is known as Channel
Matrix.

Figure 2.1 Channel Representation

𝑃 (𝑌|𝑋) = 𝑃(𝑦1 /𝑥1) ⋯ 𝑃(𝑦𝑛 /𝑥𝑚)

⋮ ⋮
𝑃(𝑦1 /𝑥𝑚) ⋯ 𝑃(𝑦𝑛 /𝑥1)

Since each input to the channel results in some output, each row of
the column matrix must sum to unity.
Now, if the input probabilities P(X) are represented by the row matrix,
we have
𝑃(𝑋) = [𝑃(𝑥1) 𝑃(𝑥2) … 𝑃(𝑥𝑚)]

Also the output probabilities P(Y) are represented by the row matrix,
we have
𝑃(Y) = [𝑃(y1) 𝑃(y2) … 𝑃(yn)]

Then [𝑃(𝑌)] = [𝑃(𝑋)] [𝑃(𝑌/X)]

Now if P(X) is represented as a diagonal matrix, we have

[𝑃(𝑋)]𝑑 = 𝑃(𝑥1) ⋯ 0
⋮ ⋱ ⋮
0 ⋯ 𝑃(𝑥𝑚)

Then [𝑃(X,𝑌)] = [𝑃(𝑋)]d [𝑃(𝑌/X)]

Where the (i, j) element of matrix [P(X,Y)] has the form P(xi,yj).
The matrix [P(X, Y)] is known as the joint probability matrix.
The element P (xi, yj) is the joint probability of transmitting xi and
receiving yj.

Types of Channels
Other than discrete and continuous channels, there are some
special types of channels with their own channel matrices.
They are as follows:
Lossless Channel: A channel described by a channel matrix with only
one non zero element in each column is called a lossless channel.

𝑃 (𝑌/𝑋) = 3/4 1/4 0 0

0 0 2/3 0
0 0 0 1

Deterministic Channel: A channel described by a channel matrix

with only one non – zero element in each row is called a deterministic
channel.

1 0 0
𝑃 (𝑌/𝑋) = 1 0 0
0 1 0
0 0 1
Noiseless Channel: A channel is called noiseless if it is both lossless
and deterministic. For a lossless channel, m = n

𝑃 (𝑌/𝑋) = 1 0 0
0 1 0
0 0 1

Binary Symmetric Channel:

BSC has two inputs (x1 = 0 and x2 = 1) and two outputs (y1 = 0
and y2 = 1). This channel is symmetric because the probability of
receiving a 1 if a 0 is sent is the same as the probability of receiving a
0 if a 1 is sent.

𝑃 (𝑌/𝑋) = 1-p p
p 1-p
 2.2 Channel capacity

The channel capacity represents the maximum amount of

information that can be transmitted by a channel per second. To
achieve this rate of transmission, the information has to be
processed properly or coded in the most efficient manner. Channel
Capacity per Symbol CS

The channel capacity per symbol of a discrete memory-less channel

(DMC) is defined as
𝐶𝑠 = max I(X; Y) bits/symbol i.e. {P(xi)}

Where the maximization is over all possible input probability

distributions {P (xi )} on X.

Channel Capacity per Second C: If ‗r‘ symbols are being transmitted

per second, then the maximum rate or transmission of information
per second is ‗rCS ‘.

This is the channel capacity per second and is denoted by C (b/s) i.e.
𝐶 = 𝑟𝐶𝑆 𝑏/s

Lossless Channel:
For a lossless channel, H (X/Y) = 0 and I (X; Y) = H(X).
Thus the mutual information is equal to the input entropy and no
source information is lost in transmission.

𝐶𝑆 = max 𝐻(𝑋)= log2𝑚 { P(xi)}

Where m is the number of symbols in X.

Deterministic Channel:
For a deterministic channel, H (Y/X) = 0 for all input
distributions P (xi ) and I (X; Y) = H(Y). Thus the information transfer
is equal to the output entropy. The channel capacity per symbol will
be
𝐶𝑆 = max𝐻(Y) = log2 n {P(xi )}
where n is the number of symbols in Y

Noiseless Channel:
Since a noiseless channel is both lossless and deterministic,
we have I (X; Y) = H (X) = H (Y) and the channel capacity per symbol
is
𝐶𝑆 = log2 𝑚 = log2 𝑛

Binary Symmetric Channel:

For the BSC, the mutual information is

𝐼( 𝑋;𝑌) = 𝐻(𝑌) + 𝑝 log2 𝑝 + (1 − 𝑝) log2 (1 −𝑝)

And the channel capacity per symbol will be

𝐶𝑆= 1 + 𝑝 log2 𝑝+(1−𝑝) log2 (1 −𝑝)
 2.3 Hartley Shannon Law

The Shannon – Hartley law underscores the fundamental role of

bandwidth and signal – to – noise ration in communication channel.
It also shows that we can exchange increased bandwidth for
decreased signal power for a system with given capacity C.

In an additive white Gaussian noise (AWGN) channel, the channel

output Y is given by Y = X + n
Where X is the channel input and n is an additive bandlimited white
Gaussian noise for zero mean and variance σ2.

The capacity C of an AWGN channel is given by

𝐶𝑠 = max 𝐼(𝑋; 𝑌) = 1/2 log2 (1+S/N) bit/sample {P(xi)}

Where S/N is the signal – to – noise ratio at the channel output.

If the channel bandwidth B Hz is fixed, then the output y(t) is also a

bandlimited signal completely characterized by its periodic sample
values taken at the Nyquist rate 2B samples/s.

Then the capacity C (b/s) of the AWGN channel is limited by

𝐶 = 2𝐵 ∗𝐶𝑆 =𝐵log2 (1+S/N)𝑏/𝑠

This above equation is known as the Shannon – Hartley Law.

The bandwidth and the noise power place a restriction upon the rate
of information that can be transmitted by a channel. Channel
capacity C for an AWGN channel is expressed as

𝐶 = 𝐵𝑙𝑜𝑔2(1 + 𝑆/𝑁)

Where B = channel bandwidth in Hz; S = signal power; N = noise

power;
Assuming signal mixed with noise, the signal amplitude can be
recognized only within the root mean square noise voltage.
Assuming average signal power and noise power to be S watts and N
watts, respectively, the RMS value of the received signal is √ (𝑆 + 𝑁)
and that of noise is √𝑁.
Therefore the number of distinct levels that can be distinguished
without error is expressed as
𝑀 = √( 𝑆 + 𝑁) / √𝑁 = √ (1 + S/N)

The maximum amount of information carried by each pulse having

√ (1 + S/N) distinct levels is given by

𝐼 = log2 (√ (1 + S/N)) = 1/2 log2 (√ (1 + S/N) bits

The channel capacity is the maximum amount of information that

can be transmitted per second by a channel. If a channel can
transmit a maximum of K pulses per second, then the channel
capacity C is given by

C = 𝐾/2 log2 (1 + 𝑆/𝑁) bits/Second

A system of bandwidth nfm Hz can transmit 2nfm independent pulses

per second. It is concluded that a system with bandwidth B Hz can
transmit a maximum of 2B pulses per second.

Replacing K with 2B, we eventually get

C = 𝐵 log2 (1 + 𝑆/𝑁) bits/Second

The bandwidth and the signal power can be exchanged for one
another.
 2.4 Source Coding theorem

A conversion of the output of a discrete memory less source (DMS)

into a sequence of binary symbols i.e. binary code word is called
Source Coding. The device that performs this conversion is called the
Source Encoder.
An objective of source coding is to minimize the average bit rate
required for representation of the source by reducing the redundancy
of the information source

Code word Length:

Let X be a DMS with finite entropy H (X) and an alphabet
{𝑥1… … . . 𝑥𝑚 } with corresponding probabilities of occurrence P(xi )
(i = 0, …. , M-1). Let the binary code word assigned to symbol xi by
the encoder have length ni, measured in bits. The length of the code
word is the number of binary digits in the code word.

Average Code word Length:

The average code word length L, per source symbol is given by
M−1

𝐿= Ʃ p(xi)𝑛𝑖
i=0
The parameter 𝐿 represents the average number of bits per source
symbol used in the source coding process.

Code Efficiency:
The code efficiency η is defined as η = 𝐿𝑚𝑖𝑛 / 𝐿

Code Redundancy:
The code redundancy γ is defined as 𝜸 =𝟏 –ƞ

Classification of Code
1. Fixed – Length Codes
2. Variable – Length Codes
3. Distinct Codes
4. Prefix – Free Codes
5. Uniquely Decodable Codes
6. Instantaneous Codes
7. Optimal Codes
Fixed – Length Codes:
A fixed – length code is one whose code word length is fixed.
Code 1 and Code 2 of above table are fixed – length code words with
length.

Variable – Length Codes:

A variable – length code is one whose code word length is not
fixed. All codes of above table except Code 1 and Code 2 are variable
– length codes.

Distinct Codes:
A code is distinct if each code word is distinguishable from each
other. All codes of above table except Code 1 are distinct codes.

Prefix – Free Codes:

A code in which no code word can be formed by adding code
symbols to another code word is called a prefix- free code. In a prefix
– free code, no code word is prefix of another. Codes 2, 4 and 6 of
above table are prefix – free codes.

Uniquely Decodable Codes:

A distinct code is uniquely decodable if the original source
sequence can be reconstructed perfectly from the encoded binary
sequence. A sufficient condition to ensure that a code is uniquely
decodable is that no code word is a prefix of another. Thus the prefix
– free codes 2, 4 and 6 are uniquely decodable codes. Prefix – free
condition is not a necessary condition for uniquely decidability. Code
5 albeit does not satisfy the prefix – free condition and yet it is a
uniquely decodable code since the bit 0 indicates the beginning of
each code word of the code.

Instantaneous Codes:
A uniquely decodable code is called an instantaneous code if
the end of any code word is recognizable without examining
subsequent code symbols. The instantaneous codes have the
property previously mentioned that no code word is a prefix of
another code word. Prefix – free codes are sometimes known as
instantaneous codes.

Optimal Codes:
A code is said to be optimal if it is instantaneous and has the
minimum average L for a given source with a given probability
assignment for the source symbols.
 2.5 Huffman & Shannon-Fano codes

The design of a variable length code such that its average code word
length approaches the entropy of DMS is often referred to as Entropy
Coding.
There are basically two types of entropy coding. They are

1) Shannon – Fano Coding

2) Huffman Coding

Shannon – Fano Coding:

An efficient code can be obtained by the following simple
procedure, known as Shannon–Fano algorithm.

1) List the source symbols in order of decreasing probability.

2) Partition the set into two sets that are as close to equi-probables
as possible and assign 0 to the upper set and 1 to the lower set.
3) Continue this process, each time partitioning the sets with as
nearly equal probabilities as possible until further partitioning is
not possible
4) Assign code word by appending the 0s and 1s from left to right

Shannon –Fano Coding

Example Let there be six (6) source symbols having probabilities as

x1 = 0.30, x2 = 0.25, x3 = 0.20, x4 = 0.12, x5 = 0.08 x6 = 0.05. Obtain
the Shannon – Fano Coding for the given source symbols.
Shannon Fano Code words

H (X) = 2.36 b/symbol H(X) = - ΣP(xi)log2 P(xi)

m-1

𝑳 = 2.38 b/symbol 𝐿= Ʃ p(xi)𝑛𝑖

i=0

η = H (X)/ 𝑳 = 0.99

Huffman Coding:

Huffman coding results in an optimal code. It is the code that has

the highest efficiency.
The Huffman coding procedure is as follows:

1) List the source symbols in order of decreasing probability.

2) Combine the probabilities of the two symbols having the lowest
probabilities and reorder the resultant probabilities; this step is
called reduction 1. The same procedure is repeated until there
are two ordered probabilities remaining.
3) Start encoding with the last reduction, which consists of exactly
two ordered probabilities. Assign 0 as the first digit in the code
word for all the source symbols associated with the first
probability; assign 1 to the second probability.
4) Now go back and assign 0 and 1 to the second digit for the two
probabilities that were combined in the previous reduction step,
retaining all the source symbols associated with the first
probability; assign 1 to the second probability.
5) Keep regressing this way until the first column is reached.
6) The code word is obtained tracing back from right to left.

Huffman coding Example

Let there be six (6) source symbols having probabilities as x 1 = 0.30,

x2 = 0.25, x3 = 0.20, x4 = 0.12, x5 = 0.08 x6 = 0.05. Obtain the
Huffman coding for the given source symbols.
 Source Sample xi P(xi ) Codeword
X1 0.30 00
X2 0.25 01
X3 0.20 11
X4 0.12 101
X5 0.08 1000
X6 0.05 1001

H (X) = 2.36 b/symbol H(X) = - ΣP(xi)log2 P(xi)

m-1

𝑳 = 2.38 b/symbol 𝐿= Ʃ p(xi)𝑛𝑖

i=0

η = H (X)/ 𝑳 = 0.99

Redundancy:

Redundancy in information theory refers to the reduction in

information content of a message from its maximum value For
example, consider English having 26 alphabets.

Assuming all alphabets are equally likely to occur, P (xi ) = 1/26. For
all the 26 letters, the information contained is therefore
log2 26 = 4.7 𝑏𝑖𝑡𝑠/𝑙𝑒𝑡𝑡𝑒𝑟.

Assuming that each letter to occur with equal probability is not

correct, if we assume that some letters are more likely to occur than
others, it actually reduces the information content in English from its
maximum value of 4.7 bits/symbol.

We define relative entropy on the ratio of H (Y/X) to H (X) which gives

the maximum compression value and Redundancy is then expressed
as
Redundancy = H (Y/X) / H(X)
 2.6 Noise in amplitude and frequency modulation systems

Modulated signals, regardless of their kind, are perturbed by

noise and by the imperfect characteristics of the channel during
transmission. Noise can broadly be defined as any unknown signal
that affects the recovery of the desired signal. There may be many
sources of noise in a communication system, but often the major
sources are the communication devices themselves or interference
encountered during the course of transmission. There are several
ways that noise can affect the desired signal, but one of the most
common ways is as an additive distortion. That is, the received signal
is modeled as
r(t) = s(t) + w(t)

where s(t) is the transmitted signal and w(t) is the additive noise.

If we knew the noise exactly, then we could subtract it from and

recover the transmitted signal exactly. Unfortunately, this is rarely
the case. Much of communication system design is related to
processing the received signal r(t) in a manner that minimizes the
effect of additive noise.

 Minimizing the effects of noise is a prime concern in analog

communications, and consequently the ratio of signal power to
noise power is an important metric for assessing analog
communication quality.
 Amplitude modulation may be detected either coherently
requiring the use of a synchronized oscillator or non-coherently
by means of a simple envelope detector. However, there is a
performance penalty to be paid for non-coherent detection.
 Frequency modulation is nonlinear and the output noise
spectrum is parabolic when the input noise spectrum is flat.
Frequency modulation has the advantage that it allows us to
trade bandwidth for improved performance.

The mean of the random process

Both noise and signal are generally assumed to have zero mean.
The autocorrelation of the random process.
With white noise, samples at one instant in time are uncorrelated
with those at another instant in time regardless of the separation.

The autocorrelation of white noise is described by

where the Dirac delta function and N0 is the two-sided power

spectral density.

The spectrum of the random process. For additive white Gaussian

noise the spectrum is flat and defined as

To compute noise power, we must measure the noise over a specified

bandwidth. Equivalent-noise bandwidth is BT

N=N0 BT

The transmitted signal is distorted by additive white noise and the

combination is passed through a filter of bandwidth BT.

Fig 2.2 Block diagram of signal plus noise before and after filtering

If the filter bandwidth is greater than the signal bandwidth, then we

retain all of the desired signal energy. However, if the filter is no
larger than required to pass the signal undistorted, then it will
minimize the amount of noise passed. Consequently, the bandwidth
BT is referred to as the transmission bandwidth of the signal. The
matching of the receiver filter to the bandwidth of the transmitted
signal is the basis of many optimum detection schemes.
In the following, we shall represent the signal after initial filtering as
x(t) = s(t) + n(t) , where n(t) is narrowband noise, as contrasted to w(t)
which is assumed to be white.

Signal-To-Noise Ratios

The received signal in many communication systems can be modeled

as the sum of the desired signal, s(t) and a narrowband noise signal,
n(t) as shown by
x(t)= s(t) + n(t)

The signal is random due to the unpredictability of its information

content, and the noise is random. The two simplest parameters for
partially describing a random variable are the mean and variance.

For zero-mean processes, a simple measure of the signal quality is

the ratio of the variances of the desired and undesired signals.

where E is the expectation operator. For a communication signal, a

squared signal level is usually proportional to power. Consequently,
the signal-to-noise ratio is often considered to be a ratio of the
average signal power to the average noise power.

Equivalently, it can be considered to be a ratio of the average signal

energy per unit time to the average noise energy per unit time.

The signal-to-noise ratio is clearly measured at the receiver, but

there are several points in the receiver where the measurement may
be carried out. In fact, measurements at particular points in the
receiver have their own particular importance and value.

Fig 2.3 High-level block diagram of a communications receiver.

For instance:
If the signal-to-noise ratio is measured at the front-end of the
receiver, then it is usually a measure of the quality of the
transmission link and the receiver front-end.
If the signal-to-noise ratio is measured at the output of the receiver,
it is a measure of the quality of the recovered information-bearing
signal whether it be audio, video, or otherwise.

To illustrate these two points, consider the block diagram of a typical

analog communication receiver presented in Fig. 2.3. The signal plus
white Gaussian noise is passed through a band-pass filter to produce
the band-pass signal,x(t).

The signal x(t) is processed by the demodulator to recover the original

message signal m(t) The SNR measured at the input to the
demodulator is referred to as the pre-detection signal-to-noise ratio.
Of equal or greater importance is the signal-to-noise ratio of the
recovered message at the output of the demodulator.

This metric defines the quality of the signal that is delivered to the
end user. We refer to this output SNR as the post-detection signal-to-
noise ratio.
It should be noted that the signal and noise characteristics may
differ significantly between the pre-detection and post-detection
calculations. The calculation of the post-detection signal-to-noise
ratio involves the use of an idealized receiver model, the details of
which naturally depend on the channel noise and the type of
demodulation used in the receiver.

The message power is the same as the modulated signal power of the
modulation scheme under study.

The baseband low-pass filter passes the message signal and rejects
out-of-band noise.
Accordingly, we may define the reference signal-to-noise ratio, as
SNRref. It is defined as the ratio of average power of the modulated
message signal to the average power of noise measured in the
message bandwidth.

The reference signal-to-noise ratio may be used to compare different

modulation and demodulation schemes by using it to normalize the
post-detection signal-to-noise ratios. That is, we may define a figure
of merit for a particular modulation–demodulation scheme as follows
Fig 2.4 Reference transmission model for analog communications.

post detection SNR

Figure of merit = ---------------------------
reference SNR

The higher the value that the figure of merit gas, the better the noise
performance of the receiver will be

 To summarize our consideration of signal-to-noise ratios:

 The pre-detection SNR is measured before the signal is
demodulated. The post-detection SNR is measured after the
signal is demodulated. The reference SNR is defined on the
basis of a baseband transmission model.
 The figure of merit is a dimensionless metric for comparing
different analog modulation-demodulation schemes and is
defined as the ratio of the post-detection and reference SNRs.

Band-Pass Receiver Structures

Fig. 2.5 shows an example of a superheterodyne receiver

AM radio transmissions
Common examples are AM radio transmissions, where the RF
channels‘ frequencies lie in the range between 510 and 1600 kHz,
and a common IF is 455 kHz v FM radio.
Another example is FM radio, where the RF channels are in the range
from 88 t o 108 MHz and the IF is typically 10.7 MHz.
The filter preceding the local oscillator is centered at a higher RF
frequency and is usually much wider, wide enough to encompass all
RF channels that the receiver is intended to handle.
With the same FM receiver, the band-pass filter after the local
oscillator would be approximately 200 kHz wide; it is the effects of
this narrower filter that are of most interest to us.

Noise in Linear Receivers Using Coherent Detection:

Double-sideband suppressed-carrier (DSB-SC) modulation, the
modulated signal is represented as
s(t) = Ac m(t) cos(2πfc t +θ)

Where fc is the carrier frequency, m(t) is the message signal.

The carrier phase θ. In Fig. 2.6, the received RF signal is the sum of
the modulated signal and white Gaussian noise w(t).

After band-pass filtering, the resulting signal is

x(t) = s(t) + n(t)

Fig 2.6 A linear DSB-SC receiver using coherent demodulation.

In figure 2.7 the assumed power spectral density of the band-pass

noise is illustrated. For the signal s(t) the average power of the signal
component is given by expected value of the squared magnitude. The
carrier and modulating signal are independent
E[ s2(t)] = E[( Ac cos(2πfct+ θ ))2 ] E[ m2(t )]

P = E[ m2(t )] E[ s2(t)] = A2c P/2

 Fig 2.7 Power spectral density of band-pass noise.

Pre-detection signal-to-noise ratio of the DSB-SC system. A noise

bandwidth BT. The signal-to-noise ratio of the signal is

A2c P
SNR = ---------------
2N0BT

The signal at the input to the coherent detector of Fig. 2.6

x(t) = s(t) + n1(t)cos(2πfct) - nQ(t)sin(2πfct)

v(t) = x(t)cos(2πfct)

= 0.5(Ac m(t)+n1(t)) + 0.5 (Ac m(t)+n1(t)) cos(4πfct) –

0.5 nQ(t)sin(4πfct)

Cos A cos A = (1+ cos 2A) / 2 sin A cos A = sin 2A / 2

These high-frequency components are removed with a low-pass filter

Y(t) = 0.5 (Ac m(t)+n1(t))

The message signal m(t) and the in-phase component of the filtered
noise n1(t) appear additively in the output.

The quadrature component of the noise is completely rejected by the

demodulator. Post-detection signal to noise ratio.

The message component is 0.5(Ac m(t),so analogous to the

computation of the pre detection signal power, the post-detection
signal power is ¼ (AcP) where P is the average message power.
Post-detection SNR
A2c P
SNR = ---------------
2N0W

Figure of merit for this receiver is 1.

Comparing the results for the different amplitude modulation

schemes. Single-sideband modulation achieves the same SNR
performance as the baseband reference model but only requires half
the transmission bandwidth of the DSC-SC system.
SSB requires more transmitter processing.

Detection of Frequency Modulation (FM)

The frequency-modulated signal is given by

Pre-detection SNR
The pre-detection SNR in this case is simply the carrier power
A c /2 divided by the noise passed by the bandpass filter N0BT
2

A slope network or differentiator with a purely imaginary frequency

response that varies linearly with frequency. It produces a hybrid-
modulated wave in which both amplitude and frequency vary in
accordance with the message signal.

An envelope detector that recovers the amplitude variation and

reproduces the message signal.

Fig 2.8 FM Receiver

Post-detection SNR

The noisy FM signal after band-pass filtering may be

represented as
x(t) = s(t) + n(t)
n(t) = n1(t)cos(2πfct) - nQ(t)sin(2πfct)

We may equivalently express n(t) in terms of its envelope and phase

as n(t) = r(t)cos[2πfct)+φn(t)]

Where the envelope is r(t) = [ n12(t) + n Q 2 (t)] 1/2

And the phase is

To proceed, we note that the phase of s(t) is

The noisy signal at the output of the band-pass filter may be

expressed as

The phase φ(t)]of the resultant is given by

Fig 2.8 Phasor diagram for FM signal plus narrowband noise

assuming high carrier-to-noise ratio.
Under this condition, the expression for the phase simplifies to

the quadrature component of the noise is

The ideal discriminator output

The noise term nd(t) is defined by

Fig 2.9 Noise analysis of FM receiver.

a)Power spectral density of quadrature component nQ(t) of

narrowband noise n(t)
b) Power spectral density nd(t) at discriminator output.
c) Power spectral density of noise no(t) at receiver output.
The corresponding power spectral density of the noise nd(t)

Therefore, the power spectral density SNo(f) of the noise no(t)

appearing at the receiver output is defined by
 2.7 Pre-emphasis and De-emphasis

From the square-law nature of the output noise spectrum of an

FM receiver, the noise is most severe at large values of |f |.
This becomes a significant issue in FM stereo transmission where the
upper channel, (ml(t) –mr(t),suffers significantly more noise than the
lower channel, (ml(t) + mr(t).

Suppose the demodulator includes a low-pass filter which gradually

increases attenuation as |f | approaches W rather than being
approximately flat for |f | < W and cutting off sharply at W.

Such a filter with transfer function Hde(f) is presented in Fig. 2.10.

This filter will de-emphasize the effects of noise at high frequency as
illustrated in the figure.

Fig 2.10 Use of pre-emphasis and de-emphasis in an FM system.

(a) Output noise spectrum before de-emphasis.

(b) Frequency response of de-emphasis filter.
(c) Noise spectrum after de-emphasis.
As well as reducing the noise, the de-emphasis filter will distort the
received signal. To compensate this distortion, we appropriately pre-
distort or pre-emphasize the baseband signal at the transmitter,
prior to FM modulation, using a filter with the frequency response

With a matching combination of pre-emphasis and de-emphasis as

described) and the signal is recovered undistorted and, most
important, with reduced noise levels. The de-emphasis filter is often
a simple resistance-capacitance (RC) circuit with

This filter is approximately flat |f | < f3dbfor the 3-dB bandwidth of

the filter. With this choice, the noise spectrum for |f | > f3db becomes
flat over most of the message bandwidth.

At the transmitting end, the pre-emphasis filter is

Thus the pre-emphasized signal is the sum of the original signal plus
its derivative. Consequently, the modulated signal is approximately

where  = 1/ f3db .Thus, pre-emphasized FM is really a combination

of frequency modulation and phase modulation.

Pre-emphasis is used in many applications other than FM stereo

broadcasting. Pre-emphasis can be used to advantage whenever
portions of the message band are degraded relative to others.
That is, portions of the message band that are most sensitive to noise
are amplified (emphasized) before transmission. At the receiver, the
signal is de-emphasized to reverse the distortion introduced by the
transmitter; at the same time the de-emphasis reduces the noise that
falls in the most sensitive part of the message band. For example, the
Dolby system for tape recording pre-emphasizes high frequencies for
sound recording so that high-frequency surface noise can be de-
emphasized during playback.
 2.8 White noise – Narrowband noise

White Noise
The noise analysis of communication systems is often based on
an idealized noise process called white noise. The power spectral
density of white noise is independent of frequency. White noise is
analogous to the term ―white light‖ in the sense that all frequency
components are present in equal amounts. We denote the power
spectral density of a white noise process W(t) as

where the factor1/2 has been included to indicate that half the
power is associated with positive frequencies and half with negative
frequencies, as illustrated in Fig. 2.11.

Fig 2.11 Characteristics of white noise.

(a) Power spectral density. (b) Autocorrelation function.

The dimensions of No are watts per hertz. The parameter No is usually

measured at the input stage of a communications receiver. Since
there is no delta function at the origin in the power spectral density,
white noise has no dc power.
That is, its mean or average value is zero. Since the autocorrelation
function is the inverse Fourier transform of the power spectral
density. The autocorrelation of white noise is given by

The autocorrelation function of white noise consists of a delta

function weighted by the factor No/2 located at  = 0.

We note that RW()is zero for  ≠ 0.

Consequently, any two different samples of white noise, no matter

how close together in time they are taken, are uncorrelated. White
noise has infinite average power and, as such, it is not physically
realizable. Nevertheless, white noise has convenient mathematical
properties and is therefore useful in system analysis. Utility of the
white noise process is parallel to that of an impulse function or delta
function in the analysis of linear systems.

The effect of the impulse is observed only after it has passed through
a system with finite bandwidth. Likewise, the effect of white noise is
observed only after passing through a system of finite bandwidth. We
may therefore state that as long as the bandwidth of a noise process
at the input of a system is appreciably larger than that of the system
itself, we may model the noise process as white noise. This is usually
the case in practical communication systems.

Narrowband Noise
A communication receiver includes multiple signal-processing
stages. A common signal processing stage for passband systems is a
narrowband filter whose bandwidth is just large enough to pass the
modulated component of the received signal essentially undistorted,
but not so large as to admit excessive noise into the receiver. The
noise process appearing at the output of such a filter is called
narrowband noise.
If the narrowband noise has a spectrum centered at the mid-band
frequencies fcas illustrated in Figure, we find that a sample function
of the narrowband noise process is somewhat similar to a sine wave
of frequency fc that varies slowly in amplitude and phase.
 Fig 2.12 (a) Power spectral density of narrowband noise.
(b) Sample function of narrowband noise.

Narrowband noise can be represented mathematically using in-phase

and quadrature components, just as we used them to represent
narrowband signals. For the narrowband noise process N(t) of
bandwidth 2B and centered on frequency fc, we may represent in the
form

where NI(t)is called the in-phase component of N(t)and NQ(t) is the

quadrature component.

Both NI(t) and NQ(t) are low-pass random processes; that is, their
spectra are confined to 0 < |f| < B . Knowledge of the in-phase and
quadrature components, as well as the center frequency fc fully
characterizes the narrowband noise.

Given the narrowband noise sample function n(t),the in-phase and

quadrature components may be extracted using the scheme shown.
The two low-pass filters are assumed to be ideal with bandwidth
equal to B. This scheme follows directly from the representation of
narrowband noise. Alternatively, if we are given the in-phase and
quadrature components, we may generate the narrowband noise n(t).
Fig 2.13 (a) Extraction of in-phase and quadrature components of
Narrowband noise process.
(b) Generation of narrowband noise process from its in
phase and quadrature components.

The in-phase and quadrature components of narrowband noise have

the following important properties:

1. The in-phase component NI(t) and quadrature component NQ(t ) of

narrowband noise N(t) have zero mean.
2. If the narrowband noise N(t) is Gaussian, then its in-phase and
quadrature components are Gaussian.
3. If the narrowband noise N(t) is stationary, then its in-phase and
quadrature components are stationary.
4. Both the in-phase component NI(t) and NQ(t ) have the same
power spectral density SN(f)

This power spectral density is related to the power spectral density of

the narrowband density by

The in-phase component NI(t) and quadrature component NQ(t ) have

the same variance as the narrowband noise N(t) .
 2.9 Threshold Effect in Angle Modulation

Although we can increase the output SNR by increasing β,

having a large β means having a large B T (Carson‘s rule), which also
means a large noise power at the input of the demodulator. The large
input SNR approximation will not hold. „

Threshold Effect: The signal will be lost in the noise if the input SNR
is small. „

The formula of defining the post-detection SNR ratio of an FM

receiver is valid only if the pre-detection SNR, measured at the
discriminator input, is high compared to unity. If the pre-detection
SNR is lowered, the FM receiver breaks down. At first, individual
Clicks are heard in the receiver output, and as the pre-detection SNR
decreases further, the clicks merge to a crackling or sputtering
sound. At and below this breakdown point, equation fails to
accurately predict the post-detection SNR. This phenomenon is
known as the threshold effect;

POST-TEST MCQ

[Link] unit of average mutual information is

a) Bits
b) Bytes
c) Bits per symbol
d) Bytes per symbol

[Link] X and Y are statistically independent, then I (x,y) is

a) 1
b) 0
c) Ln 2
d) Cannot be determined
[Link] self information of random variable is

a) 0
b) 1
c) Infinite
d) Cannot be determined

[Link] M equally likely messages, the average amount of information

H is
a. H = log10M
b. H = log2M
c. H = log10M2
d. H = 2log10M

[Link] channel capacity is

a. The maximum information transmitted by one symbol
over the channel
b. Information contained in a signal
c. The amplitude of the modulated signal
d. All of the above

[Link] image frequency of a superheterodyne receiver ___________

a) is created within the receiver itself

b) is due to insufficient adjacent channel rejection
c) is not rejected by the IF tuned circuits
d) is independent of the frequency to which the receiver is tuned

7.A pre-emphasis is usually a ________.

a. high-pass filter
b. band-stop filter
c. low-pass filter
d. bandpass filter

[Link] noise analysis of communication systems is often based on an

idealized noise process called

a) Shot noise
b) White noise
c) Flicker noise
d) Thermal noise
9. The received signal is modeled as
a) r(t) = s(t) + w(t)
b) w(t) = a(t) + s(t)
c) w(t) = p(t) + s(t)
d) r(t) = p(t) + w(t)

10. Pre-emphasis and de-emphasis always reduce noise probability

in
a) AM
b) PM
c) FM
d) ASK

APPLICATIONS

 Error Correcting and detecting.

 Data Compression
 It is used for HF radio links
 Cryptology
 Linguistics

CONCLUSION:

Upon completion of this, Students should be able to

 Understand the Information theory technique.

 Understand the channel coding techniques.
 Understand the noises in AM and FM.

REFERENCES

1. [Link] Sharma, ―Analog and Digital communication‖,

seventh edition, K KATARIA & amp; SON‘S publication, 2017.
2. Haykin.S and Michel Moher,‖ Introduction to Analog and Digital
communication‖, Second edition, John Wiley and sons Inc,
2012.
3. Prokias J.G.,‖ Digital communications‖, 4th edition, Tata
McGraw Hill, 2000.
4. [Link]
ASSIGNMENT

[Link] there be source symbols having probabilities as x1 = 0.40, x2 =

0.25, x3 = 0.25, x4 = 0.1. Obtain the Shannon – Fano Coding for the
given source symbols.

[Link] there be source symbols having probabilities as x1 = 0.30, x2 =

0.20, x3 = 0.15, x4 = 0.15 , x3 = 0.10, x4 = 0.10. Obtain the Shannon –
Fano Coding for the given source symbols.

3. Let there be source symbols having probabilities as x1 = 0.30, x2 =

0.20, x3 = 0.15, x4 = 0.15 , x3 = 0.10, x4 = 0.10. Obtain the Huffmann
Coding for the given source symbols.

[Link] about Noise in AM system.

[Link] about Noise in FM system.

 UNIT-3 PULSE MODULATION

AIM & OBJECTIVES

 To understand the fundamental concepts of Sampling
 To understand Pulse Modulation Techniques.
 To understand Line codes and noise in PCM Techniques.

PRE-TEST MCQ

1. To avoid aliasing
a) Reduce the bandwidth
b) Cut out high frequency
c) Reduce the bandwidth & Cut out high frequency
d) None of the mentioned

2. Advantages of digital communication are

a) Easy multiplexing
b) Easy processing
c) Reliable
d) All of the mentioned

3. What is necessary for digital communication?

a) Precision timing
b) Frame synchronization
c) Character synchronization
d) All of the mentioned

4. What are the disadvantages of digital communication?

a) Needs more bandwidth
b) Is more complex
c) Needs more bandwidth & Is more complex
d) None of the mentioned

5. Examples of digital communication are

a) ISDN
b) Modems
 c) Classical telephony
d) All of the mentioned

6. Which system uses digital transmission?

a) ISDN
b) LANs
c) ISDN & LANs
d) None of the mentioned

7. Analog to digital conversion includes

a) Sampling
b) Quantization
c) Sampling & Quantization
d) None of the mentioned

8. Digital communication is _______ to environmental changes?

a) Less sensitive
b) More sensitive
c) Does not depend
d) None of the mentioned

9. Which are the common transmission media used in digital

communication system?
a) Coaxial cable
b) Twisted copper cable
c) Radio frequency bands
d) All of the mentioned

10. Modulation channel consists of

a) Amplifier
b) Signal processing units
c) Amplifier & Signal processing units
d) None of the mentioned

PRE-REQUISITE
Basic Knowledge of Electronic Devices, Digital System Design,
Signals & Systems
 3.1 Sampling Process

A continuous signal or an analog signal can be represented in

the digital version in the form of samples. Here, these samples are
also called as discrete points. In sampling theorem, the input signal
is in an analog form of signal and the second input signal is a
sampling signal, which is a pulse train signal and each pulse is
equidistance with a period of ―Ts‖.
This sampling signal frequency should be more than twice of
the input analog signal frequency. If this condition satisfies, analog
signal perfectly represented in discrete form else analog signal may
be losing its amplitude values for certain time intervals. How many
times the sampling frequency is more than the input analog signal
frequency, in the same way, the sampled signal is going to be a
perfect discrete form of signal. And these types of discrete signals are
well performed in the reconstruction process for recovering the
original signal.

The sampling theorem can be defined as the conversion of an analog

signal into a discrete form by taking the sampling frequency as twice
the input analog signal frequency. Input signal frequency denoted by
Fm and sampling signal frequency denoted by Fs.

The output sample signal is represented by the samples. These

samples are maintained with a gap, these gaps are termed as sample
period or sampling interval (Ts). And the reciprocal of the sampling
period is known as ―sampling frequency‖ or ―sampling rate‖. The
number of samples is represented in the sampled signal is indicated
by the sampling rate.

Sampling Theorem Statement

Sampling theorem states that ―continues form of a time-variant
signal can be represented in the discrete form of a signal with help of
samples and the sampled (discrete) signal can be recovered to
original form when the sampling signal frequency Fs having the
greater frequency value than or equal to the input signal frequency
Fm.
If the sampling frequency (Fs) equals twice the input signal frequency
(Fm), then such a condition is called the Nyquist Criteria for
sampling. When sampling frequency equals twice the input signal
frequency is known as ―Nyquist rate‖.

If the sampling frequency (Fs) is less than twice the input signal
frequency, such criteria called an Aliasing effect.

Fs<2Fm
So, there are three conditions that are possible from the sampling
frequency criteria. They are sampling, Nyquist and aliasing states.
Now we will see the Nyquist sampling theorem.

Nyquist Sampling Theorem

In the sampling process, while converting the analog signal to a
discrete version, the chosen sampling signal is the most important
factor. And what are the reasons to get distortions in the sampling
output while conversion of analog to discrete? These types of
questions can be answered by the ―Nyquist sampling theorem‖.

Nyquist sampling theorem states that the sampling signal frequency

should be double the input signal‘s highest frequency component to
get distortion less output signal. As per the scientist‘s name, Harry
Nyquist this is named as Nyquist sampling theorem.

Fs=2Fm
Sampling Output Waveforms
The sampling process requires two input signals. The first input
signal is an analog signal and another input is sampling pulse or
equidistance pulse train signal. And the output which is then
sampled signal comes from the multiplier block. The sampling
process output waveforms are shown below.

Fig 3.1 Sampling-output-waveforms

The sampling theorem is one of the efficient techniques in
the communication concepts for converting the analog signal into
discrete and digital form. Later the advances in digital computers
Claude Shannon, an American mathematician implemented this
sampling concept in digital communications for converting the
analog to digital form. The sampling theorem is a very important
concept in communications and this technique should follow the
Nyquist criteria for avoiding the aliasing effect.

Fig 3.2 Sampling of an analog signal

Fig 3.3 Sampling and Round off

Sampling Theorem for Low Pass Signals
The low pass signals having the low range frequency and
whenever this type of low-frequency signals need to convert to
discrete then the sampling frequency should be double than these
low-frequency signals to avoid the distortion in the output discrete
signal. By following this condition, the sampling signal does not
overlap and this sampled signal can be reconstructed to its original
form.

 Bandlimited signal xa(t)

 Fourier signal representation of xa(t) for reconstruction Xa(F)

Proof of Sampling Theorem

The sampling theorem states that the representation of an
analog signal in a discrete version can be possible with the help of
samples. The input signals which are participating in this process
are analog signal and sample pulse train sequence.

Input analog signal is s(t)

The sample pulse train is

The spectrum of an input analog signal is,

Fig 3.4 Input signal spectrum

Fourier series representation of the sample pulse train is

The spectrum of the sample output signal is,

 Fig 3.5 Spectrum-of-the-sample-output-signal

When these pulse train sequences are multiples with the analog
signal we will get the sampled output signal which is indicated by
here as g(t).

When the signal related to equation passes from the LPF, only
Fm to –Fm signal only passed to the output side and the remaining
signal is going to be eliminated. Because LPF is assigned to the cut
off frequency which is equal to the input analog signal frequency
value. In this way at one side analog signal going to converted to
discrete and recovered to its original position simply passing from a
low pass filter.

Aliasing

We can observe the over-lapping of information, which leads to

mixing up and loss of information. This unwanted phenomenon of
over-lapping is called as Aliasing.
Aliasing can be referred to as ―the phenomenon of a high-frequency
component in the spectrum of a signal, taking on the identity of a
low-frequency component in the spectrum of its sampled version.‖
The corrective measures taken to reduce the effect of Aliasing are −
 In the transmitter section of PCM, a low pass anti-aliasing
filter is employed, before the sampler, to eliminate the high
frequency components, which are unwanted.
 The signal which is sampled after filtering is sampled at a rate
slightly higher than the Nyquist rate.
This choice of having the sampling rate higher than Nyquist rate,
also helps in the easier design of the reconstruction filter at the
receiver.

Sampling Techniques
They are basically three types of Sampling techniques, namely:
1. Natural Sampling
2. Flat top Sampling
3. Ideal Sampling

Natural Sampling:
Natural Sampling is a practical method of sampling in which pulse
have finite width equal to τ. Sampling is done in accordance with the
carrier signal which is digital in nature.

Fig 3.6 Natural Sampled Waveform and Natural Sampler

With the help of functional diagram of a Natural sampler, a sampled

signal g(t) is obtained by multiplication of sampling function c(t) and
the input signal x(t).
Spectrum of Natural Sampled Signal is given by:
G(f) = Aτ/ Ts .[ Σ sin c(n fs.τ) X(f-n fs)]

2. Flat Top Sampling:

Flat top sampling is like natural sampling i.e; practical in nature. In
comparison to natural sampling flat top sampling can be easily
obtained. In this sampling techniques, the top of the samples
remains constant and is equal to the instantaneous value of the
message signal x(t) at the start of sampling process. Sample and hold
circuit are used in this type of sampling.

Fig 3.7 Block Diagram and Waveform

Figure (a), shows functional diagram of a sample hold circuit

which is used to generate fat top samples.

Figure (b), shows the general waveform of the flat top samples. It
can be observed that only starting edge of the pulse represent the
instantaneous value of the message signal x(t).

Spectrum of Flat top Sampled Signal is given by:

G(f) = fs .[ Σ X(f-n fs). H(f)]

3. Ideal Sampling:
Ideal Sampling is also known as Instantaneous sampling or
Impulse Sampling. Train of impulse is used as a carrier signal for
ideal sampling. In this sampling technique the sampling function is a
train of impulses and the principle used is known as multiplication
principle.

Fig 3.8 Ideal Sampling Waveform

Here,
Figure (a), represent message signal or input signal or signal to be
sampled.
Figure (b), represent the sampling function.
Figure (c), represent the resultant signal.

Spectrum of Ideal Sampled Signal is given by:

G(f) = fs .[ Σ X(f-n fs)]

NYQUIST RATE:
Nyquist rate is the rate at which sampling of a signal is done so
that overlapping of frequency does not take place. When the
sampling rate become exactly equal to 2fm samples per second, then
the specific rate is known as Nyquist rate. It is also known as the
minimum sampling rate and given by: fs =2fm
Effect of Under sampling: ALIASING
It is the effect in which overlapping of a frequency components
takes place at the frequency higher than Nyquist rate. Signal loss
may occur due to aliasing effect. We can say that aliasing is the
phenomena in which a high frequency component in the frequency
spectrum of a signal takes identity of a lower frequency component in
the same spectrum of the sampled signal.
Because of overlapping due to process of aliasing, sometimes it is not
possible to overcome the sampled signal x(t) from the sampled signal
g(t) by applying the process of low pass filtering since the spectral
components in the overlap regions . Hence this causes the signal to
destroy.
The Effect of Aliasing can be reduced:
1) Pre alias filter must be used to limit band of frequency of the
required signal fm Hz.
2) Sampling frequency fs must be selected such that fs > 2fm.
3.2 Pulse Amplitude Modulation (PAM)-Pulse Position Modulation (PPM)

The process of transmitting signals in the form of pulses

(discontinuous signals) by using special techniques.

They are
Pulse Amplitude Modulation
Pulse Width Modulation
Pulse Position Modulation
Pulse Code Modulation

Pulse Amplitude Modulation

Pulse Amplitude Modulation (PAM) is an analog modulating
scheme in which the amplitude of the pulse carrier varies
proportional to the instantaneous amplitude of the message signal.
The pulse amplitude modulated signal, will follow the amplitude of
the original signal, as the signal traces out the path of the whole
wave. In natural PAM, a signal sampled at the Nyquist rate is
reconstructed, by passing it through an efficient Low Pass
Frequency (LPF) with exact cutoff frequency
The following figures explain the Pulse Amplitude Modulation.

Fig 3.9 PAM signal

Though the PAM signal is passed through an LPF, it cannot recover
the signal without distortion. Hence to avoid this noise, flat-top
sampling is done as shown in the following figure

Fig 3.10 Flat – Top PAM signal

Flat-top sampling is the process in which sampled signal can be

represented in pulses for which the amplitude of the signal cannot be
changed with respect to the analog signal, to be sampled. The tops of
amplitude remain flat. This process simplifies the circuit design.

Pulse Width Modulation

Pulse Width Modulation (PWM) or Pulse Duration Modulation
(PDM) or Pulse Time Modulation (PTM) is an analog modulating
scheme in which the duration or width or time of the pulse carrier
varies proportional to the instantaneous amplitude of the message
signal.
The width of the pulse varies in this method, but the amplitude of
the signal remains constant. Amplitude limiters are used to make
the amplitude of the signal constant. These circuits clip off the
amplitude, to a desired level and hence the noise is limited.
The following figures explain the types of Pulse Width Modulations.

Fig 3.11 PCM signal

There are three variations of PWM. They are
 The leading edge of the pulse being constant, the trailing edge
varies according to the message signal.
 The trailing edge of the pulse being constant, the leading edge
varies according to the message signal.
 The center of the pulse being constant, the leading edge and
the trailing edge varies according to the message signal.
These three types are shown in the above given figure, with timing
slots.

Pulse Position Modulation

Pulse Position Modulation (PPM) is an analog modulating
scheme in which the amplitude and width of the pulses are kept
constant, while the position of each pulse, with reference to the
position of a reference pulse varies according to the instantaneous
sampled value of the message signal.
The transmitter has to send synchronizing pulses (or simply sync
pulses) to keep the transmitter and receiver in synchronism. These
sync pulses help maintain the position of the pulses. The following
figures explain the Pulse Position Modulation.

Fig 3.12 PPM signal

Pulse position modulation is done in accordance with the pulse
width modulated signal. Each trailing of the pulse width modulated
signal becomes the starting point for pulses in PPM signal. Hence,
the position of these pulses is proportional to the width of the PWM
pulses.

Comparison between PAM, PWM, and PPM

The comparison between the above modulation processes is
presented in a single table.

PAM PWM PPM

Amplitude is varied Width is varied Position is varied

Bandwidth depends Bandwidth depends Bandwidth depends

on the width of the on the rise time of on the rise time of the
pulse the pulse pulse

Instantaneous Instantaneous Instantaneous

transmitter power transmitter power transmitter power
varies with the varies with the remains constant
amplitude of the amplitude and width with the width of the
pulses of the pulses pulses

System complexity is System complexity is System complexity is

high low low

Noise interference is Noise interference is Noise interference is

high low low

It is similar to It is similar to It is similar to phase

amplitude modulation frequency modulation
modulation
 3.3 Quantization

The digitization of analog signals involves the rounding off of

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

Quantizing an Analog Signal

The analog-to-digital converters perform this type of function to
create a series of digital values out of the given analog signal. The
following figure represents an analog signal. This signal to get
converted into digital has to undergo sampling and quantizing.

Fig 3.13 Analog signal

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

with a number of quantization levels. Quantization is representing
the sampled values of the amplitude by a finite set of levels, which
means converting a continuous-amplitude sample into a discrete-
time signal.
The following figure shows how an analog signal gets quantized. The
blue line represents analog signal while the brown one represents
the quantized signal.
 Fig 3.14 Analog signal and Quantized Signal

Both sampling and quantization result in the loss of information.

The quality of a Quantizer output depends upon the number of
quantization levels used. The discrete amplitudes of the quantized
output are called as representation levels or reconstruction levels.
The spacing between the two adjacent representation levels is called
a quantum or step-size.
The following figure shows the resultant quantized signal which is
the digital form for the given analog signal.

Fig 3.15 Quantized signal

This is also called as Stair-case waveform, in accordance with its

shape.
Types of Quantization
There are two types of Quantization - Uniform Quantization and
Non-uniform Quantization.
The type of quantization in which the quantization levels are
uniformly spaced is termed as a Uniform Quantization. The type of
quantization in which the quantization levels are unequal and
mostly the relation between them is logarithmic, is termed as a Non-
uniform Quantization.
There are two types of uniform quantization. They are Mid-Rise type
and Mid-Tread type. The following figures represent the two types of
uniform quantization.

Figure 1 shows the mid-rise type and figure 2 shows the mid-tread
type of uniform quantization.
 The Mid-Rise type is so called because the origin lies in the
middle of a raising part of the stair-case like graph. The
quantization levels in this type are even in number.
 The Mid-tread type is so called because the origin lies in the
middle of a tread of the stair-case like graph. The quantization
levels in this type are odd in number.
 Both the mid-rise and mid-tread type of uniform quantizers are
symmetric about the origin.

Quantization Error
For any system, during its functioning, there is always a difference
in the values of its input and output. The processing of the system
results in an error, which is the difference of those values.
The difference between an input value and its quantized value is
called a Quantization Error. A Quantizer is a logarithmic function
that performs Quantization rounding off the value rounding off the
value. An analog-to-digital converter (ADC) works as a quantizer.
The following figure illustrates an example for a quantization error,
indicating the difference between the original signal and the
quantized signal.

Fig 3.16 Quantization error

Quantization Noise
It is a type of quantization error, which usually occurs in
analog audio signal, while quantizing it to digital. For example, in
music, the signals keep changing continuously, where regularity is
not found in errors. Such errors create a wideband noise called
as Quantization Noise.
 3.4 Pulse Code Modulation PCM

We know that Modulation is the process of varying one or more

parameters of a carrier signal in accordance with the instantaneous
values of the message signal.
The message signal is the signal which is being transmitted for
communication and the carrier signal is a high frequency signal
which has no data, but is used for long distance transmission.
There are many modulation techniques, which are classified
according to the type of modulation employed. Of them all, the
digital modulation technique used is 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.

Fig 3.17 PCM Signal

Instead of a pulse train, PCM produces a series of numbers or digits,
and hence this process is called as digital. Each one of these digits,
though in binary code, represents the approximate amplitude of the
signal sample at that instant.
In Pulse Code Modulation, the message signal is represented by a
sequence of coded pulses. This message signal is achieved by
representing the signal in discrete form in both time and amplitude.

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

Fig 3.18 PCM block diagram

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.
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.
Hence, the Pulse Code Modulator circuit digitizes the given analog
signal, codes it and samples it, and then transmits it in an analog
form. This whole process is repeated in a reverse pattern to obtain
the original signal.

Companding in PCM
The word Companding is a combination of Compressing and
Expanding, which means that it does both. This is a non-linear
technique used in PCM which compresses the data at the
transmitter and expands the same data at the receiver. The effects of
noise and crosstalk are reduced by using this technique.
There are two types of Companding techniques. They are

A-law Companding Technique

 Uniform quantization is achieved at A = 1, where the
characteristic curve is linear and no compression is done.
 A-law has mid-rise at the origin. Hence, it contains a non-zero
value.
 A-law companding is used for PCM telephone systems.

µ-law Companding Technique

 Uniform quantization is achieved at µ = 0, where the
characteristic curve is linear and no compression is done.
 µ-law has mid-tread at the origin. Hence, it contains a zero
value.
 µ-law companding is used for speech and music signals.
µ-law is used in North America and Japan
 3.5 Differential Pulse Code Modulation DPCM

For the samples that are highly correlated, when encoded by

PCM technique, leave redundant information behind. To process this
redundant information and to have a better output, it is a wise
decision to take a predicted sampled value, assumed from its
previous output and summarize them with the quantized values.
Such a process is called as Differential PCM DPCM technique.

DPCM Transmitter
The DPCM Transmitter consists of Quantizer and Predictor with
two summer circuits. Following is the block diagram of DPCM
transmitter.

Fig 3.19 DPCM Transmitter

The signals at each point are named as −
 x(nTs) is the sampled input
 x^(nTs) is the predicted sample
 e(nTs) is the difference of sampled input and predicted output,
often called as prediction error
 v(nTs) is the quantized output
 u(nTs) is the predictor input which is actually the summer
output of the predictor output and the quantizer output

The predictor produces the assumed samples from the previous

outputs of the transmitter circuit. The input to this predictor is the
quantized versions of the input signal x(nTs).
Quantizer Output is represented as −
v(nTs)=Q[e(nTs)]

=e(nTs)+q(nTs)
Where q (nTs) is the quantization error

Predictor input is the sum of quantizer output and predictor output,

u(nTs)=x^(nTs)+v(nTs)

u(nTs)=x^(nTs)+e(nTs)+q(nTs)

u(nTs)=x(nTs)+q(nTs)
The same predictor circuit is used in the decoder to reconstruct the
original input.

DPCM Receiver
The block diagram of DPCM Receiver consists of a decoder, a
predictor, and a summer circuit. Following is the diagram of DPCM
Receiver.

Fig 3.20 DPCM Receiver

The notation of the signals is the same as the previous ones. In the
absence of noise, the encoded receiver input will be the same as the
encoded transmitter output.
As mentioned before, the predictor assumes a value, based on the
previous outputs. The input given to the decoder is processed and
that output is summed up with the output of the predictor, to obtain
a better output.
 3.6 Delta Modulation DM

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

rate, to achieve better sampling. If this sampling interval in
Differential PCM is reduced considerably, the sample to-sample
amplitude difference is very small, as if the difference is 1-bit
quantization, then the step-size will be very small i.e., Δ delta.

Delta Modulation
The type of modulation, where the sampling rate is much
higher and in which the step size 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 delta.
 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.
 Fig 3.20 DM Modulator

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)
 x^(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.
ep(nTs)=x(nTs)−x^(nTs) ----eqn 1

=x(nTs)−u([n−1]Ts)

=x(nTs)−[xˆ[[n−1]Ts]+v[[n−1]Ts]] ----eqn 2
Further,
v(nTs)=eq(nTs)=[Link].[ep(nTs)] ------eqn 3

u(nTs)=x^(nTs)+eq(nTs)
Where,
 x^(nTs) = the previous value of the delay circuit
 eq(nTs) = quantizer output = v(nTs)v(nTs)
Hence,
u(nTs)=u([n−1]Ts)+v(nTs) ------eqn 4
Which means,
The present input of the delay unit is equal to
The previous output of the delay unit (+) The present quantizer
output.
Assuming zero condition of Accumulation,
n
u(nTs)=S∑ sig[ep(jTs)]
j=1

Accumulated version of DM output

n
= ∑ v(jTs) --- eqn 5
j=1
Now, note that
x^(nTs)=u([n−1]Ts)

n-1
= ∑ v(jTs) --- eqn 6
j=1

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

From eqns 5 & 6, we get a possible structure for the demodulator.
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
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.
Following is the diagram for delta demodulator.
 Fig 3.21 DM Demodulator

From the above diagram, we have the notations as

 v^(nTs) is the input sample
 u^(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 DM Over DPCM

 1-bit quantizer
 Very easy design of the modulator and the demodulator
However, there exists some noise in DM.
 Slope Over load distortion (when Δ is small)
 Granular noise (when Δ is large)
 3.7 Line Codes

A line code is the code used for data transmission of a digital signal
over a transmission line. This process of coding is chosen so as to
avoid overlap and distortion of signal such as inter-symbol
interference.

Properties of Line Coding

Following are the properties of line coding
 As the coding is done to make more bits transmit on a single
signal, the bandwidth used is much reduced.
 For a given bandwidth, the power is efficiently used.
 The probability of error is much reduced.
 Error detection is done and the bipolar too has a correction
capability.
 Power density is much favorable.
 The timing content is adequate.
 Long strings of 1s and 0s is avoided to maintain transparency.

Types of Line Coding

There are 3 types of Line Coding

 Unipolar
 Polar
 Bi-polar

Unipolar Signaling
Unipolar signaling is also called as On-Off Keying or
simply OOK.
The presence of pulse represents a 1 and the absence of pulse
represents a 0.
There are two variations in Unipolar signaling

 Non Return to Zero NRZ

 Return to Zero RZ
Unipolar Non-Return to Zero NRZ
In this type of unipolar signaling, a High in data is represented
by a positive pulse called as Mark, which has a duration T0 equal to
the symbol bit duration. A Low in data input has no pulse.
The following figure clearly depicts this.

Fig 3.22 Unipolar Non-Return to Zero Waveform

Advantages
The advantages of Unipolar NRZ are

 It is simple.
 A lesser bandwidth is required.
Disadvantages
The disadvantages of Unipolar NRZ are
 No error correction done.
 Presence of low frequency components may cause the signal
droop.
 No clock is present.
 Loss of synchronization is likely to occur (especially for long
strings of 1s and 0s).
Unipolar Return to Zero RZ
In this type of unipolar signaling, a High in data, though
represented by a Mark pulse, its duration T0 is less than the symbol
bit duration. Half of the bit duration remains high but it
immediately returns to zero and shows the absence of pulse during
the remaining half of the bit duration.
It is clearly understood with the help of the following figure.

Fig 3.23 Unipolar Return to Zero Waveform

Advantages
The advantages of Unipolar RZ are

 It is simple.
 The spectral line present at the symbol rate can be used as a
clock.
Disadvantages
The disadvantages of Unipolar RZ are

 No error correction.
 Occupies twice the bandwidth as unipolar NRZ.
 The signal droop is caused at the places where signal is non-
zero at 0 Hz.
Polar Signaling
There are two methods of Polar Signaling. They are

 Polar NRZ
 Polar RZ

Polar NRZ
In this type of Polar signaling, a High in data is represented by a
positive pulse, while a Low in data is represented by a negative
pulse. The following figure depicts this well.

Fig 3.24 Polar Non-Return to Zero Waveform

Advantages
The advantages of Polar NRZ are

 It is simple.
 No low-frequency components are present.
Disadvantages
The disadvantages of Polar NRZ are
 No error correction.
 No clock is present.
 The signal droop is caused at the places where the signal is
non-zero at 0 Hz.
Polar RZ
In this type of Polar signaling, a High in data, though
represented by a Mark pulse, its duration T0 is less than the symbol
bit duration. Half of the bit duration remains high but it
immediately returns to zero and shows the absence of pulse during
the remaining half of the bit duration.
However, for a Low input, a negative pulse represents the data, and
the zero level remains same for the other half of the bit duration.
The following figure depicts this clearly.

Fig 3.25 Polar Return to Zero Waveform

Advantages
The advantages of Polar RZ are

 It is simple.
 No low-frequency components are present.
Disadvantages
The disadvantages of Polar RZ are
 No error correction.
 No clock is present.
 Occupies twice the bandwidth of Polar NRZ.
 The signal droop is caused at places where the signal is non-
zero at 0 Hz.
Bipolar Signaling
This is an encoding technique which has three voltage levels
namely +, - and 0. Such a signal is called as duo-binary signal.
An example of this type is Alternate Mark Inversion AMIAMI. For
a 1, the voltage level gets a transition from + to – or from – to +,
having alternate 1s to be of equal polarity. A 0 will have a zero
voltage level.
Even in this method, we have two types.

 Bipolar NRZ
 Bipolar RZ
From the models so far discussed, we have learnt the difference
between NRZ and RZ. It just goes in the same way here too. The
following figure clearly depicts this.

Fig 3.26 Bi-Polar NRZ & RZ Waveform

The above figure has both the Bipolar NRZ and RZ waveforms. The
pulse duration and symbol bit duration are equal in NRZ type, while
the pulse duration is half of the symbol bit duration in RZ type.
Advantages

 It is simple.
 No low-frequency components are present.
 Occupies low bandwidth than unipolar and polar NRZ schemes.
 This technique is suitable for transmission over AC coupled
lines, as signal drooping doesn‘t occur here.
 A single error detection capability is present in this.

Disadvantages

 No clock is present.
 Long strings of data causes loss of synchronization.
 3.8 Noise Consideration in PCM

Noise Consideration in PCM Systems

Two major noise sources in PCM systems. They are

 (Message-independent) Channel noise

 (Message-dependent) Quantization noise

The quantization noise is often under designer‘s control, and can be

made negligible by taking adequate number of quantization levels.

The main effect of channel noise is to introduce bit errors. Notably,

the symbol error rate is quite different from the bit error rate. A
symbol error may be caused by one-bit error, or two bit error, or
three-bit error, or …; so, in general, one cannot derive the symbol
error rate from the bit error rate (or vice versa) unless some special
assumption is made.

Considering the reconstruction of original analog signal, a bit error in

the most significant bit is more harmful than a bit error in the least
significant bit

Error Threshold (Eb/N0)

Eb: Transmitted signal energy per information bit

E.g., information bit is encoded using three-times repetition code, in

which each code bit is transmitted using one BPSK symbol with
symbol energy Ec.

Then Eb = 3 Ec.

N0: One-sided noise spectral density

The bit error rate (BER) is a function of Eb/N0 and transmission

speed (and implicitly bandwidth, etc).
Influence of Eb/N0 on BER at 105 bit per second (bps)

The usual requirement of BER in practice is 10-5.

Error threshold: The minimum Eb/N0 to achieve the required BER.

By knowing the error threshold, one can always add a regenerative

repeater when Eb/N0 is about to drop below the threshold; hence,
long-distance transmission becomes feasible.

Unlike the analog transmission, of which the distortion will

accumulate for long-distance transmission.
 3.9 Time Division Multiplexing

In FDM, multiple signals are transmitted over a single channel, each

signal being allocated a portion of the spectrum within that
bandwidth.

In time-division multiplexing (TDM), each signal occupies the entire

bandwidth of the channel. Each signal is transmitted for only a brief
period of time.

An important feature of sampling process is a conservation of-time.

In principle, the communication link is used only at the sampling
time instances. Hence, it may be feasible to put other message‘s
samples between adjacent samples of this message on a time-shared
basis. This forms the time-division multiplex (TDM) system. A joint
utilization of a common communication link by a plurality of
independent message sources.

Fig 3.27 Basic Multiplexing Switch

The rotary switch is a manual switch that can be used to select

individual data or signal lines simply by turning its inputs ―ON‖ or
―OFF‖.In digital electronics, multiplexers are constructed from
individual analog switches encased in a single IC package as
compared to the ―mechanical‖ type selectors such as normal
conventional switches and relays.

Generally, the selection of each input line in a multiplexer is

controlled by an additional set of inputs called control lines and
according to the binary condition of these control inputs, either
―HIGH‖ or ―LOW‖ the appropriate data input is connected directly to
the output.
Normally, a multiplexer has an even number of 2 N data input lines
and a number of ―control‖ inputs that correspond with the number of
data inputs.

Fig 3.28 Single- Channel PCM-TDM System

Fig 3.29 Two Channel PCM-TDM System

Block Diagram shows the simplified block diagram for a PCM carrier
system comprised of two DS-0 channels that have been time division
multiplexed. Each channel input is sampled at an 8KHz rate and
then converted to an eight bit PCM code. While PCM code for channel
1 is being transmitted. Channel 2 is sampled and converted to a
PCM code. While the PCM code from channel 2 is being transmitted,
the next sample is taken from channel 1 and converted to PCM code.

The process continues and samples are taken alternately from each
channel, converted to PCM codes and transmitted.

Multiplexer is simply an electronically controlled digital switch with

two inputs and one output. Channel 1 and channel 2 are alternately
selected and connected to transmission line through the multiplexer.

Fig 3.30 TDMA Frame

One 8-bit PCM code from each channel (16-bits total) is called a
TDM frame, and time it takes to transmit one TDM frame is called
frame time.

Frame time is reciprocal of sample rate (1/f s) or 1/8000 =125μs.

The PCM code for each channel occupies a fixed time slot within the
total TDM frame.

With Two channel system, one sample is taken from each channel
during each frame, and time allocated to transmit PCM bits from
each channel is equal to one half the total frame time. Therefore eight
bits from each channel must be transmitted during each frame (a
total of 16 PCM bits per frame). Thus line speed at output of
multiplexer is

Each channel is producing and transmitting only 64Kbps, the bits

must be clocked out onto line at a 128 KHz rate to allow eight bits
from each channel to be transmitted in a 1211μs time slot.
Synchronization is essential for a satisfactory operation of the TDM
system.
One possible procedure to synchronize the transmitter clock and the
receiver clock is to set aside a code element or pulse at the end of a
frame, and to transmit this pulse every other frame only.

Fig 3.31 TDMA Frame

 3.10 Digital Multiplexers

The introduction of digital multiplexer enables us to combine digital

signals of various natures, such as computer data, digitized voice
signals, digitized facsimile and television signals.

Fig 3.32 Digital Multiplexer

The multiplexing of digital signals is accomplished by using a bit-by-

bit interleaving procedure with a selector switch that sequentially
takes a (or more) bit from each incoming line and then applies it to
the high-speed common line.

Fig 3.33 Digital Multiplexer using bit-by-bit

Digital multiplexers are categorized into two major groups.

1st Group:
Multiplex digital computer data for TDM transmission over
public switched telephone network. n Require the use of modem
technology.
2nd Group:
Multiplex low-bit-rate digital voice data into high-bit-rate voice
stream. Accommodate in the hierarchy that is varying from one
country to another.

Usually, the hierarchy starts at 64 Kbps, named a digital signal zero

(DS0).

POST-MCQ:

[Link] signals which are obtained by encoding each quantized signal

into a digital word is called as
a) PAM signal
b) PCM signal
c) FM signal
d) Sampling and quantization

[Link] noise can be reduced by ________ the number of

levels.
a) Decreasing
b) Increasing
c) Doubling
d) Squaring

[Link] PCM the samples are dependent on ________

a) Time
b) Frequency
c) Quanization leavel
d) Interval between quantization level

[Link] pulse modulation technique is least expensive?

a) Pulse amplitude modulation
b) Pulse width modulation
c) Pulse position modulation
d) Pulse code modulation

[Link] of the following is the process of ‗aliasing‘?

a) Peaks overlapping
b) Phase overlapping
c) Amplitude overlapping
d) Spectral overlapping
[Link] of the following is false with respect to pulse position
modulation?
a) Can be transmitted in broadband
b) Modulates a high frequency carrier
c) Pulse is narrow
d) Pulse width changes in accordance with the amplitude of
modulating signal

[Link] encodes the PCM values based on

a) Quantization level
b) Difference between the current and predicted value
c) Interval between levels
d) None of the mentioned

[Link] modulation uses _____ bits per sample.

a) One
b) Two
c) Four
d) Eight

[Link] provides constant delay?

a) Synchronous TDM
b) Non synchronous TDM
c) Synchronous & Non synchronous TDM
d) None of the mentioned

[Link] is based on orthogonality?

a) TDM
b) FDM
c) TDM & FDM
d) None of the mentioned

Applications:

 It is used in the satellite transmission system.

 It is also used in space communication.
 Used in Telephony.
 One of the recent applications is the compact disc.
 It used in ISDN (Integrated Services Digital Network) telephone
lines.
 It is used in PSTN (public switched telephone network).
  It is used for some telephone system.
 It is used in wire line telephone lines.

CONCLUSION:

Upon completion of this, Students should be able to

 Understand the sampling techniques.

 Understand the Pulse code Modulation technique.
 Understand the Line code technique.
 Understand the TDM , Multiplexers.

REFERENCES

1. [Link] Sharma, ―Analog and Digital communication‖,

seventh edition, K KATARIA & amp; SON‘S publication, 2017.
2. Haykin.S and Michel Moher,‖ Introduction to Analog and Digital
communication‖, Second edition, John Wiley and sons Inc,
2012.
3. Prokias J.G.,‖ Digital communications‖, 4th edition, Tata
McGraw Hill, 2000.
4. [Link]

ASSIGNMENT:

1. Explain sampling techniques.

2. Explain about PAM,PPM and PWM.
3. Explain about PCM and DPCM.
4. Explain about DM.