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R.M.D ENGINEERING COLLEGE

21EC401 COMMUNICATION SYSTEMS

Department : ECE

Batch/Year : 2021-2025 / II year

Created By : Dr. N. Vini Antony Grace

Mr. M. Jyothi Prasad
Mrs. P. Santhoshini

Date : 24.04.2023
 Table of Contents
S. Contents Page
No. Number

1 Course Objectives 7

2 Pre-Requisites 8

3 Syllabus 9

4 Course outcomes 10

5 CO- PO/PSO Mapping 11

6 Unit 5 – Pulse Modulation Techniques 12

13
6.1 Lecture Plan
14
6.2 Activity based learning
18
6.3 Lecture Notes

➢ Sampling of Bandpass signals/ Sampling 19

Theorem
➢ Nyquist rate or Nyquist Interval 19

➢ Representation of CT signals intern of its 19

samples
23
➢ Reconstruction of CT signals from its samples

➢ Aliasing and effects to avoid aliasing 23

➢ Quantization- Uniform & non-uniform 25

quantization
➢ Quantization noise 27

➢ Logarithmic Companding 32

➢ Pulse Amplitude Modulation (PAM), Pulse 37

Position Modulation (PPM), Pulse Width
Modulation(PWM)
S.No Contents Page
Number
28
➢ Pulse Code Modulation (PCM)- Time
Division Multiplexing (TDM), Frequency
Division Multiplexing (FDM)
52
6.4 Assignments
53
6.5 Part A Q & A
58
6.6 Part B Questions
59
6.7 Supportive online Certification courses
6.8 Real time Applications in day to day life
60
and to Industry
61
6.9 Content beyond the Syllabus
7 Assessment Schedule 64

8 Prescribed Text Books & Reference Books 65

9 Mini Project suggestions 66

1. COURSE OBJECTIVES

To discuss the concepts of various analog modulation

schemes and their spectral characteristics
To summarize various types of noises in communication
system
To analyze the effect of noise in communication systems
To describe the concepts of sampling and quantization
To discuss the concepts of pulse modulation techniques

7
2. PRE REQUISITES

1. 21EC301 – SIGNALS AND SYSTEMS

By learning this course the student will have deep insight in to

Fourier transform, Trigonometric identities, Sampling and
quantization.

2. EC8252 – ELECTRONIC DEVICES

By learning this course the student will understand the operation of

electronic devices such as diodes, Transistors, FET, etc

21EC401- COMMUNICATION Semester IV

SYSTEMS

EC8252- ELECTRONIC 21EC301 - SIGNALS

DEVICES AND SYSTEMS

Semester II Semester III

8
 3. SYLLABUS

21EC401 COMMUNICATION SYSTEMS LTPC

3 003
UNIT I AMPLITUDE MODULATION 9

Amplitude Modulation- DSBSC, DSBFC, SSB, VSB - Modulation index, Spectra, Power
relations and Bandwidth – AM Generation – Square law and Switching modulator, AM
detection - Envelope detector, DSBSC Generation – Balanced and Ring Modulator, DSBSC
detection – Coherent detector, SSB Generation – Filter, Phase Shift and Third Methods, VSB
Generation – Filter Method –comparison of different AM techniques.

UNIT II ANGLE MODULATION 9

Phase and frequency modulation, Narrow Band and Wide band FM – Modulation index,
Spectra, Power relations and Transmission Bandwidth - FM modulation –Direct and Indirect
methods, FM Demodulation – FM to AM conversion, FM Discriminator - PLL as FM
Demodulator.

UNIT III NOISE CHARACTERIZATION 9

Noise sources – Noise figure, noise temperature and noise bandwidth – Noise in cascaded
systems. Hilbert Transform, Pre-envelope & complex envelope - Representation of Narrow
band noise –In- phase and quadrature components, Envelope and Phase components.

UNIT IV NOISE PERFORMANCE IN AM AND FM SYSTEMS 9

AM Super heterodyne Receiver - Noise performance analysis in AM systems, AM Threshold

effect, FM Super heterodyne Receiver, Noise performance analysis in FM systems – FM
Threshold effect, Pre-emphasis and de-emphasis for FM, Comparison of noise performance
of AM and FM Systems.

UNIT V PULSE MODULATION TECHNIQUES 9

Baseband sampling – Aliasing-Quantization - Uniform & non-uniform quantization -

quantization noise - Logarithmic Companding – PAM, PPM, PWM, PCM – TDM, FDM.

TOTAL: 45 PERIODS

9
4. COURSE OUTCOMES

After successful completion of the course, the students should be able to

Highest
Course Outcomes Cognitive
Level
Compare different Amplitude Modulation Schemes for
C212.1 K2
their efficiency and bandwidth

C212.2 Summarize the concepts of Angle Modulation Systems K2

Explain different types of Noise in Communication

C212.3 K3
Systems
Analyze the behavior of a communication system in
C212.4 K3
presence of noise
Summarize the principles of Sampling and
C212.5 K2
Quantization

C212.6 Describe the concepts of Pulse modulation Techniques K3

10
5. CO- PO/PSO Mapping

MAPPING OF COURSE OUTCOMES WITH PROGRAM OUTCOMES:

Program
Program Outcomes Specific
Course Level
Outcom of Outcomes
K3,
es CO K3 K4 K4 K5 K5, A3 A2 A3 A3 A3 A3 A2 K5 K5 K3
K6
PO-1 PO-2 PO-3 PO-4 PO-5 PO-6 PO-7 PO-8 PO-9 PO-10 PO-11 PO-12 PSO-1 PSO-2 PSO-3
C212.1 K2 2 1 - - - - - - - - - - - - 2
C212.2 K2 2 1 - - - - - - - - - - - - 2
C212.3 K3 3 2 2 2 - - - - - - - - - 2 3
C212.4 K3 3 2 2 2 - - - - - - - - - 2 3
C212.5 K2 2 1 - - - - - - - - - - - - 2
C212.6 K3 3 2 2 2 - - - - - - - - - - 2
C211 - 3 2 2 2 - - - - - - - - - 3 3

11
 6 UNIT V – PULSE
MODULATION TECHNIQUES

12
6.1 LECTURE PLAN
UNIT V – PULSE MODULATION TECHNIQUES

Mode of Delivery
Taxonomy level
Proposed Date
No. of Periods

Pertaining CO
Actual Date

Reason for
Deviation
S.No

Topic

Low pass
1 sampling
1 26.04.2023 CO5 K2 PPT

2 Aliasing 1 26.04.2023 CO5 K2 PPT

Signal
3 Reconstruction
1 26.04.2023 CO5 K2 PPT

Quantization -
Black Board
4 Uniform 1 27.04.2023 CO5 K2
Teaching
quantization
Non-uniform Black Board
5 quantization
1 27.04.2023 CO5 K2
Teaching
Logarithmic
Black Board
6 Companding of 1 28.04.2023 CO5 K2
Teaching
speech signal
Black Board
7 PAM,PPM,PWM 1 28.04.2023 CO6 K3
Teaching

Pulse Code Black Board

8 Modulation
1 29.04.2023 CO6 K3
Teaching

9 TDM,FDM 1 29.04.2023 CO6 K2 PPT

Total No. of Periods : 9

13
6.2 ACTIVITY BASED LEARNING

14
6.2 ACTIVITY BASED LEARNING

UNIT V
PULSE MODULATION TECHNIQUES

S.No Activity Topic

1 Quiz Sampling and Quantization

2 Poster Presentation PCM and Its applications

15
 TECHNICAL QUIZ
1. The process which convert continuous time signal into discrete time signal
a. Quantization
b. Encoding
c. Sampling
d. None of the above
Ans: Sampling
2. The rate at which the signal is sampled.
a. Sampling Interval
b. Sampling Frequency
c. Nyquist interval
d. All the above
Ans: Sampling Frequency
3. The time interval between two successive samples is called sampling interval
/Sampling period.
a. Sampling Interval
b. Sampling Frequency
c. Nyquist interval
d. None of the above
Ans: Sampling Interval
4. If the sampling frequency is twice of the highest frequency content of the signal.
𝑓𝑠 = 2W
a. Sampling Interval
b. Sampling Frequency
c. Nyquist rate
d. Nyquist interval
Ans: Nyquist rate
5. Due to which effect the original signal cannot be recovered
a. Sampling
b. Presence of guard band
c. Aliasing
d. All the above
Ans: Aliasing

16
6. The gap between the two spectrum which is used for avoiding interference
a. Nyquist interval.
b. Guard band
c. Both a and b
d. None of the above.
Ans: Guard band
7. The condition of sampling that avoids interference.
a. 𝑓𝑠 ≥2 𝑓𝑚
b. 𝑓𝑠 ≤ 2 𝑓𝑚
c. Both a and b
d. None
Ans: 𝒇𝒔 ≤ 2 𝒇𝒎
8. Antialiasing filter is a
a. High Pass Filter
b. Low Pass Filter
c. Band Pass Filter
d. Band stop Filter.

Ans: Low Pass Filter

9. Quantizing noise occurs in

a. TDM
b. FDM
c. PCM
d. PWM
Ans: PCM
10. Quantizing noise can be reduced by increasing the number of samples per second. It is
true
a. Yes, it is
b. No, its not
c. Not necessary.
d. None of these

Ans: None of these

17
6.3 Lecture Notes
UNIT V
PULSE MODULATION TECHNIQUES

Baseband sampling – Aliasing-Quantization - Uniform & non-

uniform quantization - quantization noise - Logarithmic
Companding – PAM, PPM, PWM, PCM – TDM, FDM.

18
UNIT V PULSE MODULATION TECHNIQUES

INTRODUCTION:
Discrete time signals can be analyzed by using Z transform and Fourier
transform, which are the sampled counterpart of Laplace transform. A CT signal will
be converted into DT signal by the process of sampling.

SAMPLING:
Sampling is the process of converting a continuous time signal into a discrete time
signal.

SAMPLING OPERATION:

Sampling Process
Sampling operation can be represented by a switch. The switch is closed for a very
short interval of time ꞇ, during which the signal is available at the output.
If the input is x (t), then the output is x (nT) ; n=0,1,2,3…
x (nT) is the sampled sequence of x (t).

SAMPLING INTERVAL: (T)

The time interval between two successive samples is called sampling
interval (T).

SAMPLING FREQUENCY: (fs)

The rate at which the signal is sampled is called the sampling rate or sampling
frequency (fs).

𝒇𝒔 =1/T

19
SAMPLING THEOREM:
STATEMENT:
• A band limited signal of finite energy, which has no frequency components
higher than ‘ W‘ Hz is completely described by specifying the values signal
at instants of time separated by 1/2W seconds.
• A band limited signal of finite energy, which has no frequency components
higher than ‘W’ Hz maybe completely recovered by its samples taken at the
rate of two W samples per second.
• A Continuous limit signal can be completely represented by its samples and
recovered back. If the sampling frequency is twice of the highest frequency
content of the signal.
𝒇𝒔 > 2W
which is called as nyquist rate.

PROOF:
1. Representation of CT signal interms of its samples.
2. Reconstruction of CT signals from its samples.

1. REPRESENTATION OF CT SIGNAL INTERMS OF ITS

SAMPLES:

x (t) 𝑥𝑠 (t)
X

𝛿𝑇 (t)

𝑥𝑠 (t) = x (t) . 𝛿𝑇 (t)

Representation of Signals
20
 Consider any arbitrary signal x (t) and any impulse train of pulses 𝛿𝑇 (t) and
let the sampled signal be 𝑥𝑠 (t).
T- is the sampling period and 𝑓𝑠 – sampling frequency.
𝑥𝑠 (t) – is the product of the signal x (t) and impulse train 𝛿𝑇 (t)
𝑥𝑠 (t) = x (t) . 𝛿𝑇 (t) →1
where 𝛿𝑇 (t) = σ∞
𝑛=−∞ 𝛿(𝑡 − 𝑛𝑇)

By exponential Fourier series,

𝑛=−∞ 𝛿(𝑡 − 𝑛𝑇) =σ𝑛=−∞ 𝑐𝑛 𝑒
σ∞ ∞ 𝑗𝑛𝜔𝑠 𝑡

1 𝑇/2
where 𝑐𝑛 =𝑇 ‫׬‬−𝑇/2 𝛿(𝑡) . 𝑒 −𝑗𝑛𝜔𝑠 𝑡
1 1
= [1]=
𝑇 𝑇
1
𝛿𝑇 (t)= σ∞
𝑛=−∞ 𝛿(𝑡 − 𝑛𝑇) = 𝑇 σ𝑛=−∞ 𝑒
∞ 𝑗𝑛𝜔𝑠 𝑡
→2

Sub 2 in 1,
1
𝑥𝑠 (t) = 𝑇 σ∞
𝑛=−∞ 𝑥(𝑡). 𝑒
𝑗𝑛𝜔𝑠 𝑡

Take FT on both sides,

1
F[𝑥𝑠 (t)] = F[σ∞
𝑛=−∞ 𝑥(𝑡). 𝑒
𝑗𝑛𝜔𝑠 𝑡
]
𝑇
1
𝑋𝑠 (ω) = [ σ∞
𝑛=−∞ 𝐹[ 𝑥 𝑡 . 𝑒
𝑗𝑛𝜔𝑠 𝑡
]]
𝑇
1
𝑋𝑠 (𝜔) = 𝑇 σ∞
𝑛=−∞ 𝑋( 𝜔 − 𝑛𝜔𝑠 ) [by freq. shifting property]

(i.e) F[ x(t)𝑒 𝑗𝜔𝑜 𝑡 = X[ j(𝜔 − 𝜔𝑜 )]

(or)
1
𝑋𝑠 (f) = 𝑇 σ∞
𝑛=−∞ 𝑋(𝑓 − 𝑛𝑓𝑠 )

𝑋𝑠 (f) = 𝑓𝑠 σ∞
𝑛=−∞ 𝑋 (𝑓 − 𝑛𝑓𝑠 )

Thus the Fourier transform of the sampled signal is given by an infinite sum
of shifted replicas of the Fourier transform of the original signal.

X(ω)
1

−𝜔𝑚 𝜔𝑚 ω

Fig. a
21
 Xs(ω)
1/T

−𝜔𝑚 −𝜔𝑠 𝜔𝑚 −𝜔𝑚 𝜔𝑚 −𝜔𝑚 𝜔𝑠 𝜔𝑚 ω

Fig .b
Xs(ω)
1/T

−𝜔𝑠 0 𝜔𝑠 ω

Fig.c
Xs(ω)
1/T

−𝜔𝑠 0 𝜔𝑠 ω
Fig .d Overlapping of
• Fig.a denotes the spectrum of the original signal x(t). spectral components
• Fig .b denotes the spectrum of the sampled signal, 𝑥𝑠 (t) when 𝑓𝑠 > 2𝑓𝑚
• In this case can be recovered even with non ideal filters.
• Fig .c denotes the spectrum of the sampled signal 𝑥𝑠 (𝑡), when 𝑓𝑠 = 2𝑓𝑚 .
• In this case the signal can be recovered with an ideal low pass filter with sharp cut off
frequencies.
• Fig .d denotes the spectrum of the sampled signal 𝑥𝑠 (𝑡), when 𝑓𝑠 < 2𝑓𝑚 .
• In this case the low frequency components in 𝑋𝑠 (𝜔), overlaps with the high frequency
components of X(𝜔).
• This type of distortion is called Aliasing. Due to Aliasing effect the signal cannot be
recovered. To overcome aliasing effect Anti-aliasing can be used.
NYQUIST RATE:
The rate at which the sampling frequency ‘𝑓𝑠 ′is equal to twice that of the max. frequency
component.
(i.e) 𝑓𝑠 = 2𝑓𝑚 . At nyquist rate the signal can be recovered by using an ideal Lowpass filter
(LPF).

22
RECONSTRUCTION OF SIGNAL FROM ITS SAMPLES:
Consider the sampled signal 𝑥𝑠 𝑡

𝑥𝑠 𝑡 = ෍ 𝑥 𝑛𝑇 . 𝛿(𝑡 − 𝑛𝑇)
𝑛=−∞
The sampled signal 𝑥𝑠 𝑡 is passed through a LPF with an impulse response
h(t), to get the original signal x(t).

𝑥𝑠 (t) h(t) 𝑥(𝑡)

LPF
Reconstruction of signal
By convolution integral,
x(t) = 𝑥𝑠 𝑡 ∗ h(t)
∞ ∞
= ‫׬‬−∞ 𝑥𝑠 (𝜏).ℎ(𝑡 − 𝜏) 𝑑𝜏 =‫׬‬−∞ σ∞
𝑛=−∞ 𝑥 𝑛𝑇 . 𝛿 𝜏 − 𝑛𝑇 . ℎ 𝑡 − 𝜏 𝑑𝜏

∞
= σ∞
𝑛=−∞ 𝑥 𝑛𝑇 ‫׬‬−∞ 𝛿 𝜏 − 𝑛𝑇 . ℎ 𝑡 − 𝜏 𝑑𝜏

x(t)= σ∞
𝑛=−∞ 𝑥 𝑛𝑇 . ℎ 𝑡 − 𝑛𝑇 →2
Consider the frequency response H(𝜔) of LPF,

H(ω)
T 𝜔𝑠
𝑇 ; 𝑓𝑜𝑟 𝜔 ≤ ൗ2
𝐻 𝜔 = ൝
0 ; 𝑜𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒

−𝜔𝑠 /2 0 𝜔𝑠 /2 ω
From Inverse Fourier transform,
1 𝜔𝑠 /2
h(t) = ‫׬‬ 𝐻 𝜔 . 𝑒 𝑗𝜔𝑡 𝑑𝜔
2𝜋 −𝜔𝑠 /2

1 𝜔𝑠 /2
= ‫׬‬−𝜔 𝑇. 𝑒 𝑗𝜔𝑡 𝑑𝜔
2𝜋 𝑠 /2

𝑇 𝑒 𝑗𝜔𝑡 𝜔𝑠 /2
= [ 𝑗𝑡 ] −𝜔 /2
2𝜋 𝑠

𝑇 𝑒 𝑗𝜔𝑠 𝑡/2 −𝑒 −𝑗𝜔𝑠 𝑡/2

= [ ]
2𝜋 𝑗𝑡

𝑇 𝑒 𝑗𝜔𝑠 𝑡/2 −𝑒 −𝑗𝜔𝑠 𝑡/2

= [[ ]
𝜋𝑡 2𝑗

𝑇 𝜔 𝑡
= [sin 2𝑠 ]
𝜋𝑡

𝑇 𝜋𝑓𝑠 𝑡 𝑇
= [ sin 2 ] = [ sin 𝜋𝑓𝑠 𝑡/2]
𝜋𝑡 2 𝜋𝑡
23
 𝑇
h(t) = sin 𝜋𝑡/𝑇 [ 𝑓𝑠 = 1/𝑇]
𝜋𝑡

𝑠𝑖𝑛 𝜋𝑡/𝑇
= = sin 𝑐 𝑡/𝑇
𝜋𝑡/𝑇

∴ ℎ 𝑡 = sin 𝑐 𝑡/𝑇

(𝑡 − 𝑛𝑇)
∴ ℎ 𝑡 − 𝑛𝑇 = sin 𝑐 →3
𝑇
sub 3 in 2, The reconstructed signal is

∞
(𝑡 − 𝑛𝑇)
∴ 𝑥 𝑡 = ෍ 𝑥 𝑛𝑇 . sin 𝑐
𝑇
𝑛=−∞

𝑥𝑠 (t)

……. …….

-4T -3T -2T -T 0 T 2T 3T nT

Original Signal

-4T -3T -2T -T 0 T 2T 3T

Superposition of Sinc Function

24
Process of quantization
The following figure represents an analog signal

Analog 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.

Quantization process

Types of Quantization
Quantization process can be classified into two types
➢Uniform quantization
➢Non uniform quantization

Uniform quantization
A quantizer is called as uniform quantizer if the step size remains
constant throughout the input range and the representation levels are
uniformly spaced

There are two types of uniform quantizers

➢Midtread type
➢Midriser type
25
 Midtread Type Midriser type
• 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

Non-Uniform Quantizer

If the step size varies depending upon the input signal values, then the
quantizer is known as the non uniform quantizer.
Nonuniform quantizers increase quantization intervals as magnitude
of value. Interval proportional to value implies logarithmic curve.

Non-Uniform Quantizer
27
Quantization Error or Noise

The difference between the instantaneous value of the quantized signal and
input signal is called as quantization error or quantization noise

Quantization Error

Quantization Noise
Signal to quantization noise ratio (SNRQ)
SNRq=(4.8+6N)dB

27
 Pulse code modulation PCM
Definition

Pulse code modulation is essentially an analog to digital conversion process ,

where the information contained in the instantaneous sample of analog signal are
represented by digital codes and are transmitted as a serial bit stream.

The PCM system includes the following basic elements

➢ Transmitter

➢ Transmission path

➢ Receiver

PCM transmitter

The basic operations performed in the transmitter of a PCM system are

➢ Sampling

➢ quantizing

➢ Encoding

➢ The quantizing and encoding operations are usually performed in the same circuit
which is called analogue to digital converter.

PCM transmitter

LPF: The LPF are used to aliasing of the message signal by adjusting the
frequencies
greater than fm so that a proper sampling rate can be obtained at PCM
transmitter.

Sampler:
A train of narrow rectangular pulse are used to sample the message
signal
fs ≥ 2fm
28
 Quantizer
The process of making the signal discrete in amplitude by
approximating the sampled signal to the nearest predefined or representation
level is called quantization.
It has two types
Uniform quantization, Non uniform quantization
Encoder
The function of encoder is to encode the discrete set of samples.
The process of allocating some digital code to each level is called encoding

PCM transmission path

The PCM transmission path is referred as the path between PCM
transmitter and PCM receiver over which the PCM signal travel.
The ability of controlling the effects caused by distortion and noise in
the transmission of PCM wave is the important feature of PCM system.
PCM achieves the capacity with the help of a chain of regenerative
repeaters.

PCM transmission
PCM receiver

PCM receiver

29
➢The regeneration circuit of the PCM receiver reshape the pulse by removing the noise

➢The noise removed pulses are applied to decoder.

➢ A sample and hold circuit in the decoder can be used to convert the digitized word
into to its analog value.

➢Message signal is recovered by passing the decoder output through a LPF, whose
cutoff frequency is equal to the message bandwidth fm bandwidth of PCM= 1/2 Nfs,
N=No of samples.

Quantization in PCM
Quantization
Definition
Each sampled value at the input of the quantizer is approximated or rounded off
to the nearest standard predicted voltage level. These standard levels are known as
quantization levels.
Sampling results in a series of pulses of varying amplitude values ranging
between two limits: a min and a max.
• The amplitude values are infinite between the two limits.
We need to map the infinite amplitude values onto a finite set of known values.
This is achieved by dividing the distance between min and max into L zones, each of
height 
= (max - min)/L

Quantization Levels
The midpoint of each zone is assigned a value from 0 to L-1 (resulting in L values)
Each sample falling in a zone is then approximated to the value of the midpoint.

Quantization Zones
Assume we have a voltage signal with amplitudes , Vmin=-20V and Vmax=+20V.
We want to use L=8 quantization levels.
Zone width  = (20 - -20)/8 = 5
The 8 zones are: -20 to -15, -15 to -10, -10 to -5, -5 to 0, 0 to +5, +5 to +10, +10 to
+15, +15 to +20
The midpoints are: -17.5, -12.5, -7.5, -2.5, 2.5, 7.5, 12.5, 17.5

30
Assigning Codes to Zones

Each zone is then assigned a binary code.

The number of bits required to encode the zones, or the number of bits per
sample as it is commonly referred to, is obtained as follows:

nb = log2 L

Given our example, nb = 3

The 8 zone (or level) codes are therefore: 000, 001, 010, 011, 100, 101, 110, and
111
Assigning codes to zones:
000 will refer to zone -20 to -15
001 to zone -15 to -10, etc.

Quantization and encoding of a sampled signal

31
Companding PCM
Definition
The compression of signal at the transmitter and expansion at the receiver
is combined to be called as companding
companding= compressing+ expanding

Companding
Compressor

Compressor provides higher gain to the weak signals and smaller gain to the
strong input signals

Expander
This is inverse of the compressor
Characteristics
It brings original amplitudes from the compressor to the receiver.

Companding curves for PCM

Types of compressor
µ- law companding A- law companding

32
 Telephone systems use ITU standardized compression formula.
μ-law: North America and Japan. For μ = 255 (for 8-bit codes),
A-law: Europe, rest of world.

The standard value is A = 87.7.

For both laws, the input to the compressor is

Comparison of μ-Law and A-Law

33
 μ-law provides slightly larger dynamic range than A-law.
A-law has smaller proportional distortion for small signals.
A-law is used for international connections if at least one country uses it.
μ-Law Implementation
Both μ-law and A-law expanders are piecewise linear.
This table shows how 7 bits are expanded.

This table is used by the μ-law encoder.

34
μ-Law Signal-to-Noise Ratio
The average power of a compressed signal is closer to the peak power.

Quantization Error and SNQR

Signals with lower amplitude values will suffer more from quantization error
as the error range: /2, is fixed for all signal levels.
Non-linear quantization is used to alleviate this problem. Goal is to keep
SNQR fixed for all sample values.
Two approaches:
• The quantization levels follow a logarithmic curve. Smaller ’s at lower amplitudes
and larger ’s at higher amplitudes.
• Companding: The sample values are compressed at the sender into logarithmic
zones, and then expanded at the receiver. The zones are fixed in height.
Bandwidth of PCM
In binary PCM, we have a group of n bits corresponding to L levels with n bits. Thus,
L = 2n or n = log2 (L)
Signal m(t) is band-limited to B Hz, which requires 2B samples per second.
For 2nB elements of information, we must transfer 2nB bits/second. Thus,
the minimum bandwidth BT needed to transmit 2nB bits/second is
BT = nB Hz
Practically speaking, we usually choose the transmission bandwidth to be a
little higher than the minimum bandwidth required.

35
 Advantages of PCM

➢ PCM systems have very high noise immunity.

➢ PCM is good in removing the distortion and noise.

➢ PCM signal is easily stored.

➢ Multiplexing is easily possible.

Disadvantages of PCM

Due to encoding decoding the PCM systems are complex

Applications

➢ Long distance digital telephone

➢ Space communication

Pulse Modulation

Pulse modulation is a technique in which the signal is transmitted with the

information by pulses. This is divided into Analog Pulse Modulation and Digital Pulse
Modulation.

36
Analog pulse modulation is classified as
• Pulse amplitude Modulation
• Pulse width Modulation
• Pulse Position Modulation

PAM (Pulse amplitude Modulation)

The amplitude of the pulse carrier is changed in proportion with the

instantaneous amplitude of the modulating signal.

Generation of PAM Signal

Types of PAM

Depending upon the shape of the PAM pulse, it is classified into two types.

➢ Natural PAM.
➢ Flat top PAM

Need for flat top PAM

During the transmission, the noise interferes with the flat top of the
transmitted pulses and this noise can be easily removed.

37
Natural PAM Generation of natural PAM

➢ The modulating signal is passed through low pass filter which will band limit this
signal to fm.
➢ The pulse train generator generates a pulse train of frequency fs such that fs >
2fm
➢ The uniform sampling takes place at the multiplier block to generate the PAM
signal. The information in the modulating signal is contained in the "amplitude
variations" of the pulsed carrier

Detection of Natural PAM

PAM signal can be detected by passing it through a LPF which is tuned to fm.
So all high frequency ripples are removed and the original modulating signal is recovered back

Detection of signal waveform

Natural PAM Modulated Signal

Natural PAM modulating Signal

Flat Top PAM Diagram for Flat Top PAM

38
Operation:

➢ A sample of hold circuit is used to produce flat top sampled PAM. This
consists of two FET switches and a capacitor.

➢ Flat top PAM signals are generated by applying the input modulating signal
changing switch. Sampling switch is closed for a short duration by a short
pulse applied to gate G1 of the transistor.
➢ During this period, the capacitor C quickly charged up to a voltage equal to
the instantaneous sample value of the incoming signal.
➢ Now, sampling switch is opened and capacitor C holds the charge. The
discharge switch is then closed by a pulse applied to gate G2, Due to this the
capacitor C is discharged to zero volts.

Flat Top PAM signal waveform

Detection of Flat Top PAM

Diagram of Detector (Flat top PAM)

39
 o Detector contains a low pass reconstruction filter with cut off frequency slightly higher
than the maximum frequency e present in the message signal.

o The equalizer compensates for the full effect.

Detection of Flat top PAM signal waveform

Flat Top Modulated Signal

Flat Top Modulating Signal

Mathematical Representation of PAM

We may express the PAM signal as

where
Ts=sampling period
m(nTs)=sample value of m(t) obtained at t=nTs
h(t)=standard rectangular pulse of unit amplitude and duration T and it is defined as

40
 The spectrum of flat-top PAM signal is

Naturally Sampled PAM signal

The natural sampling is basically pulse amplitude modulation. Therefore, it is
called naturally sampled PAM signal.
•The time-domain representation of a naturaly sampled PAM signal will be given
as

The frequency spectrum of naturally sampled PAM signal will be given as

Demodulation of PAM

System for recovering message signal m(t) from PAM signal s(t)

The distortion caused by the use of PAM to transmit an analog information

bearing signal is referred to as the aperture effect. This distortion may be
corrected by connecting an equalizer in cascade with the low-pass reconstruction
filter as shown in fig
The equalizer has the effect of decreasing the in-band loss of the reconstruction
filter as the frequency increases in such a manner as to compensate for the
aperture effect
Ideally, the magnitude response of the equalizer is given by

41
 The amount of equalization needed in practice is usually small.

Transmission Bandwidth of PAM

In PAM signal the pulse duration τ is assumed to be very small compared to time
period Ts between the two samples i.e τ<Ts

•If the maximum frequency in the modulating signal x(t) is fm then sampling
frequency fs is given by fs>=2fm Or 1/Ts>=2fm or Ts<=1/2fm

τ<<Ts<=1/2fm

If ON and OFF time of PAM pulse is equal, then maximum frequency of PAM pulse
will be fmax=1/τ+τ=1/2τ

Therefore, Transmission bandwidth >=fmax

But, fmax=1/2τ

B.W>=1/2τ>>fm

Advantages of PAM

• It is the simple and simple process for modulation and demodulation

• Transmitter and receiver circuits are simple and easy to construct

Disadvantages of PAM

• In PAM noise cannot be e removed easily.

• Transmission bandwidth required is too large.

• Transmission power is not constant.

42
PULSE WIDTH MODULATION (PWM) / PULSE DURATION MODULATION(PDM)

Definition:

A modulation technique where the width of the pulses of the pulsed carrier
wave is changed according to the modulating signal is known as Pulse Width
Modulation (PWM). It is also known as Pulse duration modulation (PDM).

Basics of Pulse Width Modulation

It is a type of Pulse Time Modulation (PTM) technique where the timing of the
carrier pulse is varied according to the modulating signal.

In pulse duration modulation (PDM), the amplitude of the pulse is kept constant and
only the variation in width is noticed. As the information component is present in width
of the pulses. Thus, during signal transmission, the signal undergoes pulse width
modulation.

Due to constant amplitude property, it gets less affected by noise. However, during
transmission channel noise introduces some variation in amplitude as it is additive in
nature. But that is totally easy removable at the receiver by making use of limiter
circuit.

As the width of the pulses contains information. Thus the noise factor does not cause
much signal distortion. Hence the immunity to the noise of a PWM system is better
than the PAM system.

Generation of PWM signal Waveform representation

The figure below shows the process of pulse width modulation. It is commonly known
as an indirect method of PWM generation.

Generation of PWM signal

43
The message signal and the carrier waveform is fed to a modulator which
generates PAM signal. This pulse amplitude modulated signal is fed to the non-
inverting terminal of the comparator.

A ramp signal generated by the saw tooth generator is fed to the inverting
terminal of the comparator.

These two signals are added and compared with the reference voltage of the
comparator circuit. The level of the comparator is so adjusted to have the
intersection of the reference with the slope of the waveform.

The PWM pulse begins with the leading edge of the ramp signal and the width of
the pulse is determined by the comparator circuit.

The width of the PWM signal is proportional to the omitted portion of the ramp
signal by the comparator level.

The figure below will help you to understand in a better way how PWM signal is
generated by the comparator:

Waveform representation of PWM signal generation

44
 Here, the first image i.e., (a) shows the waveform of the sinusoidal modulating signal
and the second one (b) shows the pulsed carrier. After modulation, a PAM signal
is generated that is shown in (c). This PAM signal, when added with ramp signal
shown in (d), is compared with the reference voltage of the comparator shown in
figure (e).

Lastly, figure (f) shows the PWM signal.

We have already mentioned that the width of the pulse is directly dependent on
the portion of the waveform that lies above the comparator level.

This is how a pulse width modulated signal is generated.

Detection of PWM signal

As we know during signal transmission, some noise gets added to the PWM
signal. So firstly to remove the noise introduced in the transmitted signal, the
incoming signal is fed to a pulse generator. This regenerates the PWM signal.

This regenerated PWM pulse is then given to a reference pulse generator that
generates pulses of constant amplitude along with constant width.

The regenerated pulses are also given to the ramp signal generator, that
generates a ramp signal of constant slope, whose duration is similar to the pulse
duration. Thus we have ramp signal height proportional to the PWM pulse width.

The figure below shows the PWM detection circuit, that provides the original
message signal from the modulated one

Detection of PWM signal

45
The constant amplitude pulses are then provided to a summation unit in order to get
added with the ramp signal. The added output is then fed to a clipper, this clips off the
signal up to its threshold value thereby generating a PAM signal at its output.

This PAM signal is then given to an LPF in order to generate the original message signal
from the modulated one.

The figure below will provide you with the waveform representation of the process of
PWM detection.

Waveform representation of PWM signal detection

The first image (i) shows the distorted PWM wave and the next one (ii) shows the
regenerated PWM pulse.

The operation of the ramp generator is shown in (iii) and (iv) shows the output of the
reference pulse generator. The summation operation and clipping off the signal is shown
in (v).

The final image of the above figure (vi) represents the PAM pulses from which the
original message signal is recovered.
47
 Pulse Position Modulation (PPM)
In PPM, the position of the pulse relative to its un-modulated time occurrence is
varied in accordance with the message signal

PWM / PPM waveform representation

Generation of PPM
• PPM generator consists of differentiator and monostable multivibrator.

• The differentiator generates positive and negative spikes corresponding to leading

and trailing edges of the PWM waveform.

• Diode D1 is used to bypass the positive spikes

• The negative spikes are used to trigger the multivibrator.

• The monostable multivibrator then generates the pulses of same width and
amplitude with reference to the trigger to give PPM waveform as shown in
figure.

Generation of PPM

47
 PPM waveform generation

Demodulation of PPM
• This utilizes the fact that the gaps between the pulses of a PPM signal
contain the information regarding the modulating signal.
• During the gap A- B between the pulses, the transmitter is cut- off, and
the capacitor C gets charged through the R-C combination. During the
pulse duration B-C, the capacitor discharges through the transistor,
and the collector voltage becomes low.
• Thus, the waveform at the collector is approximately a saw tooth
waveform whose envelope is the modulating signal.
• When this is passed through a second order OP- AMP low pass filter, we
get the desired demodulated output.

Detection of PPM

48
 Advantages of PPM:
• Like PWM, in PPM, amplitude is held constant thus less noise interference.

• Signal and noise separation is very easy

• Because of constant pulse widths and amplitudes, transmission power for

each pulse is same

Disadvantages of PWM:

• Synchronization between transmitter and receiver is required.

• Large bandwidth is required for the PPM as compared to PAM

Comparison of PAM, PWM and PPM

S.No PAM PWM/PDM PPM

The relative position of

Amplitude of the pulse is
Width of the pulse is the pulse is
proportional to the
1 proportional to amplitude of proportional to the
amplitude of modulating
modulating signal. amplitude of
signal
modulating signal.
The bandwidth of the Bandwidth of
Bandwidth of transmission
transmission channel transmission channel
2 channel depends on rise time
depends on width of the depends on rise time
of the pulse.
pulse of the pulse.

The instantaneous The instantaneous

power of the The instantaneous power of power of the
3 transmitter varies the transmitter varies with transmitter remains
with amplitude of width of pulses constant with width
pulses. of pulses.

Noise interference is Noise interference is Noise interference

4
high minimum is minimum
Simple is
5 System is complex Simple is implement
implement
Similar to Amplitude Similar to frequency Simple to Phase
6
modulation modulation modulation

49
Frequency Division Multiplexing(FDM)
FDM system used for telephone systems where frequency is divided for
communication channels
The multiplexer generates FDM signal from many telephone channels. Each
telephone channel is assigned a fixed frequency band in FDM signal

Frequency Division Multiplexing

The signal is then up converted to uplink frequency and transmitted to

satellite
The signal from satellite is down converted
It is then the demodulated by the FM Demodulator which is demultiplexed to
generate different telephone channels.

50
Time Division Multiplexing (TDM)
The TDM system which enables the joint utilization of a common channel by
independent message signals without mutual interference
The concept of TDM is illustrated by the block diagram. Each input signal is
first restricted in bandwidth by LPF to remove that frequencies.

Time Division Multiplexing

Suppose that the N message signals to be multiplexed have similar spectral

properties
Tx=Ts/N

Tx- denote the time spacing between adjacent samples in the time multiplexed
signal
Ts- sampling period N- message signals
At the receiving end of the system the received signal is applied to a pulse
amplitude modulator to demodulator.
Then the synchronization is essential for a satisfactory operation of the TDM
system between PAM modulator and demodulator.

51
 6.4 ASSIGNMENTS

S.No Questions K Level CO

Determine the Nyquist rate and interval for the following signal
1 K2 C212.5
x(t) = 5 cos(2000t)+7 sin(7000t)
A signal is sampled at Nyquist rate of 8 KHz and is quantized
2 using 8 bit uniform quantizer. Assuming SNR for a sinusoidal K2 C212.5
signal, calculate the bit rate, SNR and BW
A television signal with a bandwidth of 4.2 MHz is transmitted
3 using binary PCM. The number of quantization level is K2 C212.6
512.Calculate transmission bandwidth
A PCM system uses a uniform quantizer followed by 7-bit
4 encoder. The bit rate is 50mbps. Calculate the sampling K2 C212.6
frequency and BW.
A speech signal bandlimited to 4 KHz having maximum
amplitude of 1V is to be delta modulated with sampling
5 K2 C212.5
frequency of 20 KHz. What is the appropriate step size for the
same?
The signal x(t) = 100 Cos(20t) Cos(200t)is sampled at the
rate of 500 sps

6 (a) Determine the Nyquist rate. K2 C212.5

(b) Calulate the cut-off frequency of ideal reconstruction filter.

A signal m(t) = cos200πt + 0.25cos700πt is ideally sampled at fs

= 400 Hz. If the sampled signal is passed through an ideal LPF
7 K2 CO212.6
with a cut off frequency of 250 Hz, what frequency components
will appear at the output?
A message has zero mean value and a peak value of 10V. It is
to be quantized using a step size of 0.1 V with one level
8 K2 C212.5
coinciding to 0V. Find the number of bits required for encoding
the quantized signal.
A binary channel with bit-rate of 56 Kbps is available for uniform
PCM voice transmission having signal bandwidth of 4 KHz.
Calculate C212.6C
(a) Sampling frequency 212.6
9 K2
(b) Number of bits per sample
(c) Quantization levels with step size
(d) Transmission bandwidth for 10 channels
(e) Signal to noise ratio
An analog waveform with bandwidth 20 KHz is to be quantized
10 with 200 levels and transmitted via binary PCM signal. Find the K2 C212.6
rate of transmission and bandwidth required.

52
 6.5 Part A Q & A (with K level and CO)

SI.
K CO
N Questions
level level
O

Define sampling theorem and quantization.

▪ Sampling :It is the process in which the original analog signal is
converted into a discrete time and continuous amplitude signal
1. ▪ Mathematically, f s ≥ fmax ; fs → Sample Rate/Nyquist Rate ; K1 C212.5
fmax → maximum message(analog) signal frequency;
▪ Quantization :It is the process in which the analog sample of the
original signal is converted into a digital form.

What is the need for sampling?

▪ To convert a signal from continuous time to discrete time, a process
called sampling is used.
2. K2 C212.5
▪The value of the signal is measured at certain intervals in time.
When the continuous analog signal is sampled at a frequency F, the
resulting discrete signal has more frequency components than did the
analog signal.

What is aliasing?
▪When the continuous time signal g(t) is sampled at the rate less
3. than Nyquist rate, frequencies higher than fmax takes on the identity K2 C212.5
of the low frequencies in sampled signal spectrum.T his is called
aliasing.

What are the steps to reduce alasing?

Aliasing can be reduced by sampling at a rate higher than Nyquist
4. rate. In other words, Aliasing occurs when the signal is sampled at a K1 C212.5
rate less than Nyguist rate (2fmax samples/ sec). It is prevented by
using Guard Bands Pre-alias Filter (LPF)

What is meant by PCM?

Pulse code modulation (PCM) is a method of signal coding in which

the message signal is sampled, the amplitude of each sample is
5. rounded off to the nearest one of a finite set of discrete levels and K1 C212.6
encoded so that both time and amplitude are represented in discrete
form.. This allows the message to be transmitted by means of a
digital waveform.

53
 K CO
SI.NO Questions
level level

Define Nyquist rate and Nyquist interval.

According to sampling theorem, a continuous time signal can be
completely represented in its samples and recovered back if the sampling
frequency is fS ≥ 2fmax
6. Nyquist rate: The minimum sampling rate of 2fmax samples per second is K2 C212.5
called Nyquist rate.
i.e., fS = 2fmax → Nyquist rate
Nyquist interval: Reciprocal of 2fmax is called the Nyquist interval.
Nyquist interval = 1/2fmax

What are the two-fold effects of quantizing process?

The peak-to-peak range of input sample values subdivided into a finite set
of decision levels or decision thresholds
7. K2 C212.5
The output is assigned a discrete value selected from a finite set of
representation levels are reconstruction values that are aligned with the
treads of the staircase.

Name the types of uniform quantizer?

Mid tread type quantizer.
8. K1 C212.5
Mid riser type quantizer.

Define Mid tread quantizer

Origin of the signal lies in the middle of a tread of the staircase.

9. K2 C212.5

What is meant by idle channel noise and prediction error?

Idle channel noise is the coding noise measured at the receiver output
with zero transmitter input.
10. K2 C212.5
The difference between the actual sample of the process at the time of
interest and the predictor output is called a prediction error.

54
SI. K CO
Questions
NO level level
Define Companding.
Companding is the process of compressing, then expanding. With
13 K2 C212.5
companded systems, the higher amplitude analog signals are compressed
prior to transmission, then expanded at the receiver.

Define Mid-riser quantizer?

Origin of the signal lies in the middle of a rise of the staircase

14. K2 C212.5

What is meant by quantization?

While converting the signal value from analog to digital, quantization is

15. performed. The analog value is assigned to nearest digital value. This is K2 C212.5
called quantization. The quantized value is then converted into equivalent
binary value. The quantization levels are fixed depending upon the number
of bits. Quantization is performed in every Analog to Digital Conversion.

The signal to quantization noise ratio in a PCM system

depends on what criteria?

The signal to quantization noise ratio in PCM is given as,

16. K2 C212.6
(S/N) db ≤(4.8+6v)dB

Here v is the number of bits used to represent samples in PCM. Hence

signal to quantization noise ratio in PCM depends upon the number of bits
or quantization levels.

List the type of PAM and its Uses.

Types Of PAM:
Natural Pam
17. Flat Top PAM K1 C212.6
Applications/Uses of PAM
It is used in Ethernet communication.
It is used in many micro-controllers for generating the control signals.
It is used in Photo-biology.
It is used as an electronic driver for LED lighting.

55
SI. K CO
Questions
NO level level

Define quantization error?

C212.
18 Quantization error is the difference between the output and input K2
5
values of quantizer.

What you mean by non-uniform quantization?

Step size is not uniform. Non-uniform quantizer is characterized
C212.
19 by a step size that increases as the separation from the origin of K2
5
the transfer characteristics is increased. Non-uniform quantization
is otherwise called as robust quantization. .

Draw the quantization error for the mid tread and mid-rise
type of quantizer?

C212.
20 K2
5

What is the disadvantage of uniform quantization over the

non-uniform quantization?

SNR decreases with decrease in input power level at the uniform C212.
21 K2
quantizer but non-uniform quantization maintains a constant SNR 5
for wide range of input power levels. This type of quantization is
called as robust quantization.

What are the types of companding:

• A-law companding. C212.

22 K1
5
• µ-law companding.

56
SI. K CO
Questions
NO level level

Draw the block diagram of compander? Mention the types of

companding?

23 K2 C212.5

What is TDM?

Time-division multiplexing (TDM) is a method of putting multiple data

24 streams in a single signal by separating the signal into many segments, K2 C212.6
each having a very short duration.
Each individual data stream is reassembled at the receiving end based on
the timing.

What is FDM?

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

means combining more than one signal over a shared medium.
25 In FDM, the total bandwidth is divided to a set of frequency bands that do K2 C212.6
not overlap. Each of these bands is a carrier of a different signal that is
generated and modulated by one of the sending devices.
The frequency bands are separated from one another by strips of unused
frequencies called the guard bands, to prevent overlapping of signals.

What is FAM?

PAM is the pulse amplitude modulation. In pulse amplitude modulation, the

26 K2 C212.6
amplitude of a carrier consisting
of a periodic train of rectangular pulses is varied in proportion to sample
values of a message signal.

57
6.6 Part B Q & A (with K level and CO)

SI.N K CO
Questions
O level level

Discuss an ideal sampling process and derive the Nyquist

1. Condition for reconstructing the original signal from the K2 C212.5

sequence of samples.

State the sampling theorem and Describe the process of

sampling.
2. (a)Representation of Analog Signal K2 C212.5

(b)Reconstruction of analog signal from samples

Also illustrate the effect of aliasing with neat sketch.

With the block diagram of a Simplex PCM system, explain

3. the process of sampling and Quantization involved in PCM. K2 C212.6
Derive the expression for signal to noise ratio in PCM.

Define uniform and non-uniform quantization methods.

4. Derive an expression for quantization error and SNR for K2 C212.5
uniform quantization
(i)Explain the characteristics of Non-uniform quantization
with diagrams.
5. K2 C212.5
(ii) Explain companding? Explain A-law and µ-law
companding

Discuss about the generation of Pulse Amplitude Modulation

6. K2 C212.6
(PAM) and its demodulation

7. Explain in detail about PWM with suitable diagrams. K2 C212.6

8. Explain in detail PAM, PWM and PPM with suitable diagrams. K2 C212.6

Explain the types of multiplexing techniques and show the

9. block diagram of multiplexing K2 C212.6

58
6.7 Supportive online Certification courses (NPTEL,
Swayam, Coursera, Udemy, etc.,)

ONLINE COURSE NPTEL:

https://onlinecourses.nptel.ac.in/noc21_ee30/preview

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This course covers introduction to digital communications: Interfaces and

channels for digital communications , Geometric representation of signals, Gram-
Schmidt orthogonalization, Geometric interpretation of signals, Review of random
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The course is covers the practical basics of digital communication systems, Source
Coding, Characterization of Communication Signals & Systems, Signal space
Representation, Representation of Memory less Modulation Methods, Nonlinear
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Fundamentals of estimation and detection theory used in digital communication,
Carrier phase and symbol timing synchronization techniques, Channel estimation
and equalization techniques, Power Adaptation methods for colored noise channel

59
6.8 Real time Applications in day to day life and to
Industry

❖ Television broadcasting application using Frequency Division

Multiplexing technique (FDM)

YouTube Link to view mechanism involved in TV Broadcasting.

https://youtu.be/OmV9aJCXHIg

❖ How PWM signal is used in controlling the speed of DC motors?

Webpage Link to view mechanism involved in DC motor speed control

https://www.electronics-tutorials.ws/blog/pulse-width-modulation.html

60
6.9 CONTENTS BEYOND THE SYLLABUS

Orthogonal Frequency Division Multiplexing (OFDM)

Orthogonal Frequency Division Multiplexing (OFDM) is a digital multi-carrier
modulation scheme that extends the concept of single subcarrier modulation by
using multiple subcarriers within the same single channel. Rather than transmit a
high-rate stream of data with a single subcarrier, OFDM makes use of a large
number of closely spaced orthogonal subcarriers that are transmitted in parallel.
Each subcarrier is modulated with a conventional digital modulation scheme (such
as QPSK, 16QAM, etc.) at low symbol rate. However, the combination of many
subcarriers enables data rates similar to conventional single-carrier modulation
schemes within equivalent bandwidths.

OFDM is based on the well-known technique of Frequency Division Multiplexing

(FDM). In FDM different streams of information are mapped onto separate parallel
frequency channels. Each FDM channel is separated from the others by a
frequency guard band to reduce interference between adjacent channels.
The OFDM scheme differs from traditional FDM in the following interrelated ways:
1. Multiple carriers (called subcarriers) carry the information stream,
2. The subcarriers are orthogonal to each other, and
3. A guard interval is added to each symbol to minimize the channel delay
spread and intersymbol interference.
The following figure illustrates the main concepts of an OFDM signal and the
inter-relationship between the frequency and time domains. In the frequency
domain, multiple adjacent tones or subcarriers are each independently modulated
with complex data. An Inverse FFT transform is performed on the frequency-
domain subcarriers to produce the OFDM symbol in the time-domain. Then in the
time domain, guard intervals are inserted between each of the symbols to prevent
inter-symbol interference at the receiver caused by multi-path delay spread in the
radio channel. Multiple symbols can be concatenated to create the final OFDM
burst signal. At the receiver an FFT is performed on the OFDM symbols to recover
the original data bits.

61
 The OFDM signal can be described as a set of closely spaced FDM subcarriers. In
the frequency domain, each transmitted subcarrier results in a sinc function
spectrum with side lobes that produce overlapping spectra between subcarriers,
see "OFDM Signal Frequency Spectra" figure below. This results in subcarrier
interference except at orthogonally spaced frequencies. At orthogonal
frequencies, the individual peaks of subcarriers all line up with the nulls of the
other subcarriers. This overlap of spectral energy does not interfere with the
system’s ability to recover the original signal. The receiver multiplies (i.e.,
correlates) the incoming signal by the known set of sinusoids to recover the
original set of bits sent.

The use of orthogonal subcarriers allows more subcarriers per bandwidth

resulting in an increase in spectral efficiency. In a perfect OFDM signal,
Orthogonality prevents interference between overlapping carriers. In FDM
systems, any overlap in the spectrums of adjacent signals will result in
interference. In OFDM systems, the subcarriers will interfere with each other only
if there is a loss of orthogonality. For example, frequency error will cause the
subcarrier frequencies to shift so that the spectral nulls will no longer be aligned
resulting in inter-subcarrier-interference

Simple Digital OFDM system Implementation

The concepts used in the simple analog OFDM implementation can be extended
to the digital domain by using a combination of Fast Fourier Transform (FFT) and
Inverse Fast Fourier Transform (IFFT) digital signal processing. These transforms
are important from the OFDM perspective because they can be viewed as
mapping digitally modulated input data (data symbols) onto orthogonal
subcarriers.
62
In principle, the IFFT takes frequency-domain input data (complex numbers
representing the modulated subcarriers) and converts it to the time-domain
output data (analog OFDM symbol waveform).

In a digitally implemented OFDM system, the input bits are grouped and mapped
to source data symbols that are a complex number representing the modulation
constellation point (e.g., the BPSK or QAM symbols that would be present in a
single subcarrier system). These complex source symbols are treated by the
transmitter as though they are in the frequency-domain and are the inputs to an
IFFT block that transforms the data into the time-domain. The IFFT takes in N
source symbols at a time where N is the number of subcarriers in the system.
Each of these N input symbols has a symbol period of T seconds. Recall that the
output of the IFFT is N orthogonal sinusoids. These orthogonal sinusoids each
have a different frequency and the lowest frequency is DC.

he input symbols are complex values representing the mapped constellation point
and therefore specify both the amplitude and phase of the sinusoid for that
subcarrier. The IFFT output is the summation of all N sinusoids. Thus, the IFFT
block provides a simple way to modulate data onto N orthogonal subcarriers. The
block of N output samples from the IFFT make up a single OFDM symbol.

After some additional processing, the time-domain signal that results from the
IFFT is transmitted across the radio channel. At the receiver, an FFT block is used
to process the received signal and bring it into the frequency domain which is
used to recover the original data bits

63
7. Assessment Schedule

Assessment Proposed Date Actual Date

Unit 1 Assignment
Assessment

Unit Test 1

Unit 2 Assignment
Assessment

Internal Assessment 1 27.02.2023

Retest for IA 1

Unit 3 Assignment
Assessment

Unit Test 2

Unit 4 Assignment
Assessment

Internal Assessment 2 18.04.2023

Retest for IA 2

Unit 5 Assignment
Assessment

Revision Test 1

Revision Test 2

Model Exam 02.05.2023

Remodel Exam

University Exam

64
8. Prescribed Text Books & Reference Books

TEXT BOOK

J.G.Proakis, M.Salehi, ―Fundamentals of Communication Systems‖, Pearson

Education 2014. (UNIT I-IV)

Simon Haykin, ―Communication Systems‖, 4th Edition, Wiley, 2014.(UNIT I-

V) T3. Shibu. K.V, “Introduction to Embedded Systems”, 2e, Mc graw Hill,
2017.

REFERENCES

B.P.Lathi, - Modern Digital and Analog Communication Systems‖, 3rd Edition,

Oxford University Press, 2007.

D.Roody, J.Coolen, ―Electronic Communications, 4th edition PHI 2006.

A.Papoulis, ―Probability, Random variables and Processes, McGraw Hill, 3rd

edition, 1991.

B.Sklar, Digital Communications Fundamentals and Applications, 2nd Edition

Pearson Education 2007

H P Hsu, Schaum Outline Series - ―Analog and Digital Communications

TMH 2006

65
9. Mini Project suggestions

COMMUNICATION THEORY MINI PROJECTS LIST

S.No Name of The Project

Intelligent approach to video transmission over 2.4 GHz wireless

1 technology
Grasping money game used in wireless channel resource allocation
2
Bluetooth Controlled Electronic Home Appliances
3
GSM Based Home Security Alarm System Using Arduino
4
Remote Industrial Security System
5
Centralized Monitoring System for Taxies
6
Wireless Weather Monitoring based on GSM
7
Cell Phone Detector
8
High fidelity and way intercom with musical ring alert system for point
to point communication
9
Wireless stepper motor Control using RF Communication
10
IR based escaping cock robot
11
Wireless Stepper motor control using IR Communication
12
Analog Transmitter and Receiver: Optimization Of Power Dissipation
13 And Maximization Of Bandwidth
ASK Modulation using OPAMP
14
Wireless Stepper Motor Control using IR Communication
15

66
 Thank you

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