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R.M.D ENGINEERING
COLLEGE
22EC403
ANALOG AND DIGITAL
COMMUNICATION
(Theory Course with
Laboratory Component)
Department : ECE

Batch/Year : 2022-2026 / II year

Created by : Dr. A. Sumaiya Begum

Dr. N. Vini Antony Grace

Ms. P. Santhoshini
Ms. S. Indumathi

Date : 20.02.2024
 Table of Contents

S.No Contents Page

Number

1 Course Objectives 7

2 Pre Requisites 8

3 Syllabus 9

4 Course outcomes 11

5 CO- PO/PSO Mapping 12

6 Unit III - PULSE MODULATION SYSTEMS 13

6.1 Lecture Plan 14

6.2 Activity based learning 15

6.3 Lecture Notes 18

6.3.1 Block Diagram of digital communication system, 19

Sampling
21
6.3.2 Sampling

6.3.3 Quantization 25

27
6.3.4 Uniform & nonuniform quantization

29
6.3.5 Quantization Noise

Companding (A law and µ law)

6.3.6 30

Pulse Code Modulation (PCM)

6.3.7 31

Differential pulse code modulation

6.3.8 40

5
S.No Contents Page
Numb
er
Delta modulation and demodulation
6.3.9 42

6.3.10 Adaptive Delta Modulation 46

6.4 Assignments 51

6.5 Part A Q & A 52

6.6 Part B Questions

6.7 Supportive online Certification courses

6.8 Real time Applications in day to day life and

62
to Industry

6.9 Content beyond the Syllabus 63

7 Assessment Schedule 64

8 Prescribed Text Books & Reference Books 65

9 Mini Project suggestions 66

6
1. COURSE OBJECTIVES

OBJECTIVES:

• To discuss the concepts of various AM modulation schemes

and their spectral characteristics
• To describe the Generation and Detection of Frequency
Modulation.
• To explain the performance of various Pulse coding
Techniques.
• To learn principles of different pass band transmission
schemes
• To calculate required parameters of Source and channel
coding Techniques
• To visualize the effects of sampling and Digital Modulations
Schemes
2. PRE REQUISITES

1. 22EC301 – SIGNALS AND SYSTEMS

By learning this course the student will have deep insight in to Fourier
transform, Trigonometric identities, Sampling and quantization.

2. 22EC302 – ANALOG ELECTRONICS

By learning this course the student will understand the operation of electronic
devices such as diodes, Transistors, FET, etc

22EC403 – ANALOG AND DIGITAL Semester IV

COMMUNICATION

22EC302 – ANALOG 22EC301 - SIGNALS

ELECTRONICS AND SYSTEMS

Semester III Semester III

3. SYLLABUS
22EC403 ANALOG AND DIGITAL COMMUNICATION L T P C
(Theory course with laboratory component)
3 0 2 4
COURSE OBJECTIVES:
• To discuss the concepts of various AM modulation schemes and their spectral
characteristics
• To describe the Generation and Detection of Frequency Modulation.
• To explain the performance of various Pulse coding Techniques.
• To learn principles of different pass band transmission schemes
• To calculate required parameters of Source and channel coding Techniques
• To visualize the effects of sampling and Digital Modulations Schemes
UNIT I AMPLITUDE MODULATION 9+6
Need for Modulation, Amplitude modulation – frequency spectrum of AM– Power and current
in AM wave –Generation of AM signal – AM Emitter Modulator and Collector Modulator, AM
demodulation - Envelope, DSB-SC, SSB-SC & VSB generation and demodulation modulation,
Synchronous detection, Comparison of AM modulation systems.
LIST OF EXPERIMENTS
1. AM Modulator and Demodulator

UNIT II ANGLE MODULATION 9+6

Principle of frequency and phase modulation – Relation between FM and PM waves –– Narrow
band and wide band FM, Carson’s rule, Frequency deviation, Bandwidth of FM – Direct and
Indirect Methods of FM Generation - FM detectors – slope detectors, Phase discriminators,
Ratio detectors, PLL Demodulators . Pre-emphasis and De-emphasis, Comparison of AM and
FM. Super-heterodyne receiver (AM and FM)
LIST OF EXPERIMENTS
2. FM Modulator and Demodulator.
UNIT III PULSE MODULATION SYSTEMS 9+6

Block Diagram of digital communication system, Sampling – Quantization – Uniform &

nonuniform quantization. – Quantization noise- Companding (A law and µ law) – Pulse Code
Modulation (PCM), Differential pulse code modulation-Delta modulation and Adaptive Delta
Modulation.
LIST OF EXPERIMENTS
3. Signal Sampling and reconstruction
4. Pulse Code Modulation and Demodulation
5. Delta Modulation and Demodulation
UNIT IV DIGITAL MODULATION TECHNIQUES 9+6
Geometric Representation of signals - Generation and detection of coherent systems -BASK,
BFSK, BPSK,QPSK, QAM, and Comparison of all digital Modulation Techniques.
LIST OF EXPERIMENTS
6. Simulation of ASK, FSK, and BPSK generation schemes
7. Simulation of ASK, FSK and BPSK detection schemes
8. Simulation of QPSK and QAM generation schemes
9. Simulation of signal constellations of BPSK, QPSK and QAM
UNIT V SOURCE AND CHANNEL CODING 9+6

Definition of - Discrete Memoryless source, Information, Entropy, Channel Capacity -Hartley law,
Shannon law, Source coding theorem -Shannon Fano & Huffman codes. Channel coding theorem
-Linear Block codes.
LIST OF EXPERIMENTS
10. Simulation of Linear Block

TOTAL: 45 PERIODS(THEORY) +30 PERIODS (LAB)=75

PERIODS
TEXT BOOKS:

1. Wayne Tomasi, Advanced Electronic Communications Systems, 6th Edition, Pearson

New
International Edition, Noida, India, 2014.
2. Simon Haykin, Communication Systems,5th Edition, Wiley, 2021.

REFERENCES:

1. Sanjay Sharma, Communication Systems (Analog and digital), 7th Edition, S.K. Kataria &
Sons, 2022.
2. Roddy and Coolen, Electronic Communication, 4th Edition, Pearson Education, Noida,
India, 2014.
3. Herbert Taub and Donald Schilling, Principles of Communication Systems, 4th Edition,
McGraw Hill, 2017.
4. HweiKsu and Debjani Mitra, Analog and Digital Communication: Schaum’s Outline
Series, 3rd Edition, McGraw Hill Education, New Delhi, India., 2017.
4. COURSE OUTCOMES

OUTCOMES: K LEVEL

On successful completion of this course, the student will be able to

CO1: Compare the Spectral efficiency of various Amplitude K3

Modulation Schemes.

CO2: : Summarize the concepts of Generation and Detection of

Frequency Modulation K3

CO3: Demonstrate the performance of various Pulse coding Techniques.

K2

CO4: Differentiate the different pass band transmission

schemes K2

CO5: Construct different Source and Error control codes

K4

CO6: Implement different Digital modulation schemes and coding

K4
techniques using simulation software
 5. CO- PO/PSO Mapping

MAPPING OF COURSE OUTCOMES WITH PROGRAM OUTCOMES:

Program
Program Outcomes Specific
Course Level
Outcomes
Outcom of
es CO

K3/
K3 K4 K6 K4/K6 K6 A3 A2 A3 A3 A3 A3 A2 K3/K4/ K3/K4/ K3
K6 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

C214.1 K3 3 2 1 - 3 - - 1 - - - - 1 1 -

C214.2 K3 3 2 1 - 3 - - 1 - - - - 1 1 -

C214.3 K2 2 1 - - 2 - - 1 - - - - 1 - -

C214.4 K2 2 1 - - 2 - - 1 - - - - 1 - -

C214.5 K4 3 3 1 3 3 1 - 1 - - - - 1 - -

C214.6 K4 3 3 1 - 3 1 - 1 - - - - 1 1 -

11
6 UNIT III

PULSE MODULATION SYSTEMS

12
 6.1 LECTURE PLAN
UNIT II – ANGLE MODULATION

of
Proposed Date

Taxonomy level
No. of Periods

Pertaining CO
Actual Date

Reason for
Deviation
S.No

Topic

Delivery
Mode
1 Block Diagram of
digital communication 1 CO3 PPT
K2
system
Sampling &
2 1 CO3 PPT
Quantization K2
3 Uniform &
nonuniform 1 CO3 K2 PPT
quantization
4 Quantization Noise 1 CO3 K2 PPT

5 Companding (A law and 1 CO3 K2 PPT

µ law)
6 Pulse Code Modulation 1 PPT
CO3 K2
(PCM)
7 Differential pulse code 1 PPT
CO3 K2
modulation

8 Delta modulation 1 PPT

CO3 K2

9 Adaptive Delta
Modulation 1 CO3 PPT
K2

Total No. of Periods : 9

13
6.2 ACTIVITY BASED LEARNING

1. Crossword Puzzle

14
6.3 Lecture Notes

15
INTRODUCTION
The conventional methods of communication used analog signals for long
distance communications, which suffer from many losses such as distortion,
interference, and other losses including security breach. In order to overcome these
problems, the signals are digitized using different techniques. The digitized signals
allow the communication to be more clear and accurate without losses.

BLOCK DIAGRAM OF DIGITAL COMMUNICATION SYSTEM

Fig.1 shows the basic block diagram of a Digital Communication System.

Source

The input source or the source of information is generally analog in nature. Example: A
Sound signal. These signals are non-electrical quantities and hence cannot be
processed directly in a digital communication system.

Input Transducer

A transducer is a device which converts one form of energy into another form. Since
the information from the input source is non-electrical quantity, the input transducer
converts this non-electrical quantity into electrical quantity. Example: microphone.
This block also consists of an analog to digital converter where a digital signal is needed
for further processes. A digital signal is generally represented by a binary sequence.

However, if the source signal is already in digital form such as computer then analog
to digital converter is not needed in such cases.

Source Encoder

The source encoder is used to compress the data into minimum number of bits. This helps
in effective utilization of the bandwidth. It removes the redundant bits or unnecessary
excess bits that are zeros from the input data.

Channel Encoder

The channel encoder, does the coding for error correction. During the transmission of the
signal, due to the noise in the channel, the signal may get altered and hence to avoid this,
the channel encoder adds some redundant bits to the transmitted data. These are the
error correcting bits.

Digital Modulator

In case of low speed wireless transmission, the base band signal can be transmitted
directly.

Now, for high speed transmission, the digital data is modulated with the help of a high
frequency carrier. This task is performed by the base-band modulator.

The signal is also converted to analog from the digital sequence, in order to make it travel
through the channel or medium.

Channel

The communication channel is the media through which the signal is transmitted from the
transmitting end to the receiving end. It is the part of model at which maximum noise is
added to the signal.

Digital Demodulator

The base-band decoder is the first step at the receiver end. The received signal is
demodulated as well as converted again from analog to digital. The signal gets
reconstructed here.

Channel Decoder

The channel decoder, after detecting the sequence, does some error corrections. During
the transmission, the signal might get distorted. This is corrected by adding some
redundant bits to the signal. This addition of bits helps in the complete recovery of the
original signal.
Source Decoder
The resultant signal is once again digitized by sampling and quantizing so that the pure digital
output is obtained without the loss of information. The source decoder recreates the source
output.
Output Transducer
This is the last block which converts the signal into the original physical form, which was at
the input of the transmitter. It converts the electrical signal into physical output (Example:
loud speaker). Then it converts the digital data into analog signal.
Output Signal
This is the output which is produced after the whole process. Example − The sound signal
received.

SAMPLING

Sampling is defined as the process of measuring the instantaneous values of

continuous-time signal in a discrete form.
Sample is a piece of data taken from the whole data which is continuous in the
time domain.When a source generates an analog signal and if that has to be digitized, having
1s and 0s i.e., High or Low, the signal has to be discretized in time. This discretization of
analog signal is called as Sampling.

The following figure indicates a continuous-time signal x( t) and a sampled signal xs (t)

When x (t) is multiplied by a periodic impulse train, the sampled signal xs (t) is obtained.
Natural Sampling: Natural sampling takes a slice of the waveform, and the top of the
slice preserves the shape of the waveform.
Flat-Top Sampling: A very common, and easily implemented method of sampling of an
analog signal uses the sample-and-hold operation. This produces flat top samples.

Major differences between Natural Sampling and Flat-Top Sampling

Natural Sampling Flat-Top Sampling

Natural sampling is a method where Flat-top sampling is a method where
samples are taken at the natural samples are taken at evenly spaced
frequency of the signal intervals with a specific duration.
Natural sampling is used when the Flat-top sampling is used when the
signal is not known beforehand. signal is known beforehand.
Natural sampling is used to analyze the Flat-top sampling is used to analyze a
frequency of a signal. specific frequency range.
Natural sampling is better suited for Flat-top sampling is better suited for
non-periodic signals. periodic signals.
Natural sampling is sensitive to Flat-top sampling is not sensitive to
frequency variations. frequency variations.
Sampling Rate
To discretize the signals, the gap between the samples should be fixed. That gap
can be termed as a sampling period Ts.
Sampling Frequency=1/Ts=fs

Where,
• Ts is the sampling time
• fs is the sampling frequency or the sampling rate

Sampling frequency is the reciprocal of the sampling period. This sampling

frequency, can be simply called as Sampling rate. The sampling rate denotes the
number of samples taken per second, or for a finite set of values.
For an analog signal to be reconstructed from the digitized signal, the sampling rate
should be highly considered. The rate of sampling should be such that the data in
the message signal should neither be lost nor it should get over-lapped. Hence, a
rate was fixed for this, called as Nyquist rate.

Nyquist Rate
Suppose that a signal is band-limited with no frequency components higher
than W Hertz. That means, W is the highest frequency. For such a signal, for
effective reproduction of the original signal, the sampling rate should be twice the
highest frequency.

Which means, fS=2W

Where,
• fS is the sampling rate
• W is the highest frequency
This rate of sampling is called as Nyquist rate.

A theorem called, Sampling Theorem, was stated on the theory of this Nyquist rate.
Sampling Theorem

The sampling theorem, which is also called as Nyquist theorem, delivers the theory
of sufficient sample rate in terms of bandwidth for the class of functions that are
bandlimited.

The sampling theorem states that, “a signal can be exactly reproduced if it is

sampled at the rate fs which is greater than twice the maximum frequency W.”

To understand this sampling theorem, let us consider a band-limited signal, i.e., a

signal whose value is non-zero between some –W and W Hertz.
Such a signal is represented as x(f)=0 for ∣f∣>W
For the continuous-time signal x(t), the band-limited signal in frequency domain,
can be represented as shown in the following figure.

We need a sampling frequency, a frequency at which there should be no loss of

information, even after sampling. For this, we have the Nyquist rate that the
sampling frequency should be two times the maximum frequency. It is the critical
rate of sampling.
We need a sampling frequency, a frequency at which there should be no loss of
information, even after sampling. For this, we have the Nyquist rate that the sampling
frequency should be two times the maximum frequency. It is the critical rate of sampling.
If the signal x(t) is sampled above the Nyquist rate, the original signal can be recovered,
and if it is sampled below the Nyquist rate, the signal cannot be recovered.
The following figure explains a signal, if sampled at a higher rate than 2w in the
frequency domain.
fs>2W - the original signal can be recovered

fs=2W - The information is replaced without any loss. Hence, this is also a good
sampling rate.
 fs<2W - Aliasing

We can observe from the above pattern that the over-lapping of information is done, which
leads to mixing up and loss of information. This unwanted phenomenon of over-lapping is
called as Aliasing.
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.

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 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.
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. 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.
Uniform Quantization
The type of quantization in which the quantization levels are uniformly
spaced is termed as a Uniform Quantization.
There are two types of uniform quantization. They are Mid-Rise type and Mid-Tread
type.
Mid-rise type uniform Quantization
Rise refers to the rising part. 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.
Mid-tread type uniform Quantization
Tread refers to the flat part. 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.
Advantages of Uniform Quantization
The advantages of Uniform Quantization are as follows:
High approximation compared to non-uniform Quantization.
Easy and simple to implement

Non-uniform Quantization

The quantized levels in the non-uniform quantization process are unequally spaced. The
relation between such quantization is generally logarithmic due to non-linear nature of
the signal.

Advantages of non-uniform Quantization

The advantages of non-uniform Quantization are as follows:

High Signal to Noise Ratio (SNR)

Low quantizer noise

Difference between Uniform Quantization and Non-uniform Quantization

The following table discusses the difference between uniform and non-uniform
quantization.
Parameter Uniform Quantization Non-uniform Quantization
Quantization levels Equally spaced quantization levels Unequally spaced quantization levels

Step size Same step size between all Variable step sizes between quantization
quantization levels levels

Distribution of Not accounted for input data Accounts for input data distribution
input data distribution

Quantization error Quantization error is uniform Lower quantization error for frequently
across all inputs occurring inputs

Complexity Simple to implement More complex to implement

Applications Used when input distribution is Used when input distribution is non-
uniform uniform, like image/audio signals

QUANTIZATION ERROR

The difference between an input value and its quantized value is called a Quantization
Error.

The following figure illustrates an example for a quantization error, indicating the
difference between the original signal and the quantized signal.
Quantization Noise

Quantization is the mapping of a range of analog voltage to a single value. The use of
quantization introduces an error defined as the difference between the input signal and
output signal. The error is called quantization noise.

Figure illustrates a typical variation of the quantization noise as a function of time assuming
the use of a uniform quantizer of midtread type

Let us consider the Staircase curve of a linear N Bit ADC Converter

In the figure Green curve is a scaled version of Vin without any quantization, Red curve is
the ADC Output and ∆ is the step size of the converter.
In the above figure Pink dots show that analog range that maps to an ADC Value. Black
arrows show the Quantization error for 2 points.

Quantization error is uniformly distributed. Figure shows the PDF of Quantization Error

PULSE CODE MODULATION

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.

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,
represent 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.

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.

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

❖ PCM systems are complex

❖ It requires large bandwidth than analog communication systems

Applications

• Long distance digital telephone

• Space communication

• Compact discs

COMPANDING
Companding is a technique of achieving non-uniform quantization. The
step size is variable in non-uniform quantization. In order to maintain proper signal to
quantization noise ratio, the step size must be variable according to the signal level.

Thus in order to achieve non-uniform quantization the process of

companding is used. It is a word formed by the combination of words compression and
expanding. Companding is done in order to improve SNR of weak signals.

The figure below represents the companding model in order to achieve

non-uniform companding:
As we can see that the companding model consists of a compressor, a uniform quantizer
and an expander.

Initially at the transmitting end, the signal is first provided to the compressor. The
compressor unit amplifies the low value or weak signal in order to increase the signal
level of the applied input signal.

While if the input signal is a high level signal or strong signal then compressor attenuates
that signal before providing it to the uniform quantizer present in the model.

This is done in order to have an appropriate signal level as the input to the uniform
quantizer. We know a high amplitude signal needs more bandwidth and also is more
likely to distort. Similarly, some drawbacks are associated with low amplitude signal and
thus there exist need for such a unit.

The operation performed by this block is known as compression thus the unit is called
compressor.

The output of the compressor is provided to uniform quantizer where the quantization of
the applied signal is performed.

At the receiver end, the output of the uniform quantizer is fed to the expander.

It performs the reverse of the process executed by the compressor. This unit when
receives a low value signal then it attenuates it. While if a strong signal is achieved then
the expander amplifies it.

This is done in order to achieve the originally transmitted signal at the output.

The input-output characteristics of the expander are the reverse as compared to the
compressor, as shown below:
The graph clearly represents that the compressor provides high gain to weak signal
and low gain to high input signal.

Expander performs reverse operation of the compander. So, it is clear from the
above figure that artificially boosted signals are attenuated to have the originally
transmitted signal.

The compressor and expander performs inverse operations thus in the above figure the
dotted line represents the linear characteristic of the compander indicating that the
originally transmitted signal is recovered at the receiver

There are two types of Companding techniques.

They are

(i) A-Law

(ii) µ-law

A-Law

European countries practice A-Law companding.

The mathematical expression for A-law compression in continuous domain (PDF) is given
as:
where:

x is the input signal

sgn(x) is the sign of the input signal

|x| is the absolute value of x

A = 87.6 is the compression parameter defined by Consultative Committee for

International Telephony And Telegraphy (CCITT) G.711

The corresponding A-law expansion equation (PDF) is:

where:

y is the companded signal

sgn(y) is the sign of the companded signal

|y| is the absolute value of y

A = 87.6 is the compression parameter defined by CCITT

• In the A-law companding , the compressor characteristic is piecewise , made up of

a linear segment for low level inputs and a logarithmic segment for high level
inputs.

• 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

The µ-law companding technique is deployed in North America and Japan.

The analog version of µ-law (PDF) is given as:

for compression and

for expansion

where:

x is the input signal

y is the companded signal

sgn(x) is the sign of the input signal

sgn(y) is the sign of the companded signal

|x| is the absolute value of x

|y| is the absolute value of y

µ = 255 is the compression parameter defined by CCITT G.711

In the μ-law companding , the compressor characteristic is continuous. It is

approximately linear for smaller values of input levels and logarithmic for high input
levels.

The characteristic corresponding to μ = 0 corresponds to the uniform quantization.

• 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.

The μ-law companding is used for speech and music signals. It is used for PCM
telephone systems in US, Canada and Japan.
DIFFERENTIAL PULSE CODE MODULATION(DPCM)
For the signals which does not change rapidly from one sample to next
sample, the PCM scheme is not preferred. When such highly correlated samples
are encoded the resulting encoded signal contains redundant information.
By removing this redundancy before encoding an efficient coded signal
can be obtained. One of such scheme is the DPCM technique. By knowing the
past behavior of a signal up to a certain point in time, it is possible to make some
inference about the future values. The transmitter and receiver of the DPCM
scheme is shown in the below figure respectively.
2.2.1 DPCM Transmitter
Let x(t) be the signal to be sampled and x(nTs) be its samples. In this
scheme the input to the quantizer is a signal.

e (nTs ) = x (nTs ) − x (nTs ) (2.4)

where x (nTs ) is the prediction for un-quantized sample x(nTs). This predicted

value is produced by using a predictor whose input, consists of a quantized

versions of the input signal x(nTs). The signal e(nTs) is called the prediction error.

Figure DPCM Transmitter

By encoding the quantizer output, in this method, we obtain a modified
version of the PCM called differential pulse code modulation (DPCM). The output
of the quantizer is given by,
v(nTs) = Q[e(nTs)]
v(nTs) = e(nTs) + q(nTs)
 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 receiver consists of a decoder to reconstruct the quantized error signal. The
quantized version of the original input is reconstructed from the decoder output
using the same predictor as used in the transmitter.
In the absence of noise the encoded signal at the receiver input is
identical to the encoded signal at the transmitter output. Correspondingly the
receive output is equal to u(nTs), which differs from the input x(nTs).only by the
quantizing error q(nTs).
DPCM Receiver
In the absence of channel noise the encoded signal is identical to the
encoded signal at transmitter output .Thus in a noise free environment the
predictors in the transmitter and receiver operate on the same sequence of
samples u(nTs).only to compensate this a feedback path is used in the
transmitter.

Figure Block diagram of DPCM Receiver

DPCM is subjected to slope overload distortion and quantization noise
Processing gain
The output signal to noise ratio of a signal coder is given by
(SNR) = σx /σ
2Q 2
σx2 variance of input, σQ 2variance of quantization error
 (SNR) = σx /σ€ 2 * σ2€ / σQ2 2

(SNR)0 = Gp (SNR)Q
(SNR)Q =prediction error to quantization noise ratio Gp=
prediction Gain

DELTA MODULATION (DM)

Delta Modulation is a special case of DPCM. In DPCM scheme if the base band
signal is sampled at a rate much higher than the Nyquist rate purposely to increase the
correlation between adjacent samples of the signal, so as to permit the use of a simple
quantizing strategy for constructing the encoded signal, Delta modulation (DM) is precisely
such as scheme.
Delta Modulation is the one-bit (or two-level) versions of DPCM.
n = 1 Rb = nfs
=fs
BW= Rb/2 = fs / 2 The
bandwidth requirement is less since n=1
No of representation levels L= 2n
L=2
The two levels are +δ & -δ and Step size Δ = 2 δ

Figure Input Output Characteristics Of Delta Modulator

DM provides a staircase approximation to the over sampled version of an
input base band signal. The difference between the input and the approximation is
quantized into only two levels, namely, ±δ corresponding to positive and negative
differences, respectively.
Thus, if the approximation falls below the signal at any sampling epoch, it is
increased by δ. Provided that the signal does not change too rapidly from sample
to sample, we find that the stair case approximation remains within ±δ of the
input signal. The symbol δ denotes the absolute value of the two representation
levels of the one-bit quantizer used in the DM.

x u (nTs Ts )

Figure Stair case Approximation

If the input signal lies above the stair case approximation then quantizer
round-off the sample to + δ and if the input signal lies above the stair case
approximation then quantizer round-off the sample -δ . The encoder in delta
modulator converts the +δ values to binary 1 & -δ values to binary 0.
DM Transmitter :
Let the input signal be x(t) and the staircase approximation to it is u(t).

Figure Block diagram of DM Transmitter

e (nTs ) = x (nTs ) − x (nTs )

To find x (nTs ) a delay circuit is used whose input is u(nTs).
x (nT ) = u (nTs − Ts )
substitute equation
e(nTs) = x(nTs)- u(nTs-Ts)

u(nTs) = u(nTs-Ts) + b(nTs)

where,
b(nTs)= δ sgn [e(nTs)]
Thus the binary output of delta modulator depends on the sign of e(nTs)
with a scaling factor of δ
DM Receiver
In the receiver the stair case approximation u(t) is reconstructed by
passing the incoming sequence of positive and negative pulses through an
accumulator in a manner similar to that used in the transmitter.
The out of band quantization noise in the staircase waveform is removed
by using a LPF.
Delta modulation offers two unique features:
i. No need for Word Framing because of one-bit code word.
ii. Simple design for both Transmitter and Receiver

Delta modulator with fixed step size is called as Linear Delta modulator (LDM)

Figure Block diagram for DM Receiver

Disadvantage of DM:

Delta modulation systems are subject to two types of quantization error:

(1) Slope –overload distortion
(2) Granular noise.
 Quantization Noise
Delta modulation systems are subject to two types of quantization error.
(1) slope –overload distortion, (2) granular noise
a) Slope –overload distortion
If a slope of the input signal x(t) is very high ,i.e the signal changes
rapidly then step size is 2 δ is not sufficient to catch up the input signal . As a
result u(t) falls behind x(t) as in figure . This condition is called slope –overload
and the resulting quantization noise is called slope –overload distortion.

Figure Slope overload Distortion & Granular Noise

b) Granular noise
When the step size is too large compared to the slope characteristics of
input signal the staircase approximation u(t) hunts around the flat segment of the
input waveform and results in noise called granular noise.
c) Condition to overcome slope –overload distortion
From the DPCM analysis
u(nTs) = x(nTs)+q(nTs) from
the equation we can write

u(nTs - Ts) = X(nTs - Ts) + q(nTs- Ts)

and we know that from equation (2.7) u(nTs -
Ts) = X(nTs ) - e(nTs)

Equating equation
X(nTs - Ts) + q(nTs- Ts) = X(nTs ) - e(nTs)

e(nTs) = X(nTs ) - X(nTs - Ts) - q(nTs- Ts) if q(nTs- Ts) is very small

e(nTs) = X(nTs ) - X(nTs - Ts)

Thus the input to quantizer is a first backward difference of the input signal. So when slope of X(t) is
maximum , x(nTs) is maximum and e(nTs) is also maximum which increases u(nTs ) to catch up the
input signal at maximum slope.
ADAPTIVE DELTA MODULATION:
The performance of a delta modulator can be improved significantly by
making the step size of the modulator assume a time-varying form. In particular,
during a steep segment of the input signal the step size is increased. Conversely,
when the input signal is varying slowly, the step size is reduced.
In this way, the size is adapted to the level of the input signal. The resulting
method is called adaptive delta modulation (ADM). There are several types of ADM,
depending on the type of scheme used for adjusting the step size. In this ADM, a
discrete set of values is provided for the step size.
During a steep segment of input signal the step size is increased to control
slope over load distortion and when input signal is varying slowly the step size is
reduced to control granular noise .Thus the step size is adapted to level of the
input signal. This method is called Adaptive Delta Modulation.
In a practical implementation the step size lies between maximum and
minimum value,
Min   (nTs )  Max

δ max controls slope overload distortion

δ min controls granular noise
Adaptive Delta Modulation Transmitter

Figure Block diagram of ADM Transmitter

Adaptive Delta Modulation Receiver
Transmitted output is given to the receiver input finding step limit size. The
staircase approximation u(t) is reconstructed by passing the incoming sequence
of positive and negative pulses through an accumulator in a manner similar to
that used in the transmitter.
 The out of band quantization noise in the high frequency staircase
waveform u(t) is rejected by passing through low pass filter with a bandwidth
equal to the original signal bandwidth.
(nTs ) =  x (nTs)

Figure Block diagram of ADM Receiver

The adaptation rule δ(nTs) can be generally expressed as

δ(nTs) = g(nTs)δ(nTs-Ts)
where time varying multiplier g(nTs) depends on binary output b(nTs)of delta
modulator and previous value of b(nTs).
g(nTs) = { K, if b(nTs) = b(nTs -Ts)
K-1 , if b(nTs) ≠ b(nTs -Ts) }
1.A DM transmitter with a fixed step of 0.5V, is given a sinusoidal message signal. If
the sampling frequency is twenty times the Nyquist rate, Determine
i) Maximum permissible amplitude of the message signal, if the slope
overload is to be avoided

Solution:
Let a0 the maximum permissible peak amplitude of the sinusoidal message
signal for avoiding slope overload. Then

a0 
2πf 0TS
Sampling frequency
fs = 20x2W

fs = 40W
The maximum permissible value is

a 0(max ) = (or ) fS
2πf 0TS 2πf 0
Let us assume, fo = W, the highest frequency in the pass band of the baseband
low pass filter of receiver
fS
a 0(max ) =
2πf 0

a0(max) = (0.5 x 40W)/ (2ΠW)

a0(max) = 3.18V
2. A 1 KHz signal (voice signal) is sampled at 10 KHz using 8-bit PCM and a DM
system. Find in each case (i) Signalling rate (ii) Bandwidth required.
Solution:
Signaling rate, r= v * fs
r = 8 X 10 KHz
r = 80 Kbps (for PCM System)
DM transmits only one bit per sample
Signaling rate, r= v * fs

r = 1 X 10 KHz
r = 10 Kbps (for DM System)
Bandwidth

For PCM System

BT = Signaling rate/2
BT = 80000/2

BT = 40 KHz
For DM System
BT = Signaling rate/2
BT = 10000/2
BT = 5 KHz
6. A signal having the bandwidth 4 KHz is to be encoded using
a) 8 bit PCM
b) DM System
If 10 cycles of the signal are digitized, state how many bits will be there in
digitized output? (In each case if sampling frequency is 12 KHz). Also find
bandwidth required in each case
Solution:
fs = 12000 samples/sec
DM transmits 1 bit/sample

PCM transmits 8 bits/sample

Here, W = 4 KHz, hence period of one cycle is
Therefore for 10 cycles
T10 = 10 X T T10 = 10 / 4000

T10 =1 / 400 sce

For PCM System

Signaling rate, r= v * fs r = 8 X 12 KHz
r = 96 Kbps
For DM System
Signaling rate, r= v * fs r = 1 X 12 KHz
r = 12 Kbps Bandwidth
BT = Signaling rate (r)/2 BT = 96000/2
BT = 48 KHz
Bandwidth
BT = Signaling rate (r)/2 BT = 12000/2

BT = 6 KHz
No of Bits (10 cycles) = r x T10
= 96000 x (1/400)
= 240 bits No of Bits (10 cycles) = r x T10
= 12000 x (1/400)
= 30 bits
 UNIT III
6.4.ASSIGNMENTS

Q.No Questions BT
CO
Level
Level
In a PCM, if we increase the quantization
1 levels from 2 to 8, how do the relative CO2 K4
bandwidth requirements vary?

2 In the case of PCM transmission if the number of CO2 K4

quantization levels is increased from 6 to 64, determine
the factor by which the required bandwidth increases.

3 An audio signal x(t) = 500 cos 2000πt is quantized using CO2 K4

10 bit PCM. What is the SQNR

4 In a PCM system each quantization level is encoded into CO2 K4

6 bits. Determine the SQNR

5 Determine the number of bits used in a 4096 level PCM. CO2 K4

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

S.No CO’S Bloom

PART A s Level

Define sampling theorem

Sampling :It is the process in which the original analog
signal is converted into a discrete time and continuous
1. amplitude signal . Mathematically, f s ≥ fmax ; fs = Sample
Rate/Nyquist Rate ; fmax = maximum message(analog) CO3 K2
signal frequency

What is the need for sampling?

To convert a signal from continuous time to discrete time, a
process called sampling is used.
2.
The value of the signal is measured at certain intervals in time.
When the continuous analog signal is sampled at a frequency F, CO3 K2
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 of the low frequencies in sampled signal spectrum.T his CO3 K2
is called aliasing
What are the steps to reduce aliasing?
Aliasing can be reduced by sampling at a rate higher than
Nyquist rate. In other words, Aliasing occurs when the signal is
4.
sampled at a rate less than Nyquist rate (2fmax samples/ sec). It CO3 K2
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
5. which the message signal is sampled, the amplitude of each CO3 K2
sample is rounded off to the nearest one of a finite set of
discrete levels and 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.
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
6. sampling frequency is fS ≥ 2fmax.
Nyquist rate: The minimum sampling rate of 2fmax samples per CO3 K2
second is called Nyquist rate. i.e., fS = 2fmax → Nyquist rate
Nyquist interval: Reciprocal of 2fmax is called the Nyquist
interval. Nyquist interval = 1/2fmax

56
 Bloom
S.No PART A CO’S
s Level

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 CO3 K2
7. 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?
8. Mid tread type quantizer.
CO3 K2
Mid riser type quantizer.
What is meant by quantization?
While converting the signal value from analog to digital,
9. quantization is performed. The analog value is assigned to CO3 K2
nearest digital value. This is 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.
Define quantization error?
10. Quantization error is the difference between the output and
input values of quantizer CO3 K2

11. What are the types of companding?

• A-law companding.
CO3 K2
• µ-law companding.
12. Write an expression for bandwidth of binary PCM with
N messages each with a maximum frequency of fm Hz.
If “v” number of bits are used to code each CO3 K2
input sample, then bandwidth of PCM is given as, BT ≥
N.v. fm. Here v. fmis the bandwidth required by one
message.

57
S.No PART A CO’S Bloom
s Level

Mention the merits of DPCM.

Bandwidth requirement of DPCM is less
13.
compared to PCM. Quantization error is reduced CO3 K2
because of prediction filter, Numbers of bits used to
represent one sample value are also reduced
compared to PCM.

What is the main difference in DPCM and DM?

DM encodes the input sample by one bit. It
sends the information about + δ or -δ, i.e. step rise or CO3 K2
14. fall.
DPCM can have more than one bit of encoding the
sample. It sends the information about difference
between actual sample value and the predicted sample
value.

What is meant by adaptive delta modulation?

In adaptive delta modulation, the step size is
15. adjusted as per the slope of the input signal. Step size CO3 K2
is made high if slope of the input signal is high. This
avoids slope overload distortion.

What is the advantage of delta modulation over

PCM?
Delta modulation uses one bit to encode on CO3 K2
14. sample.
Hence bit rate of delta modulation is low compared to
PCM.

What are the two limitations of delta modulation?

15. i)Slope of overload distortion – Occurs due to
Smaller Step Size. CO3 K2
ii)Granular noise - It occurs due to large step size
and very small amplitude variation in the input signal.

What are the advantages of the Delta modulation?

Delta modulation transmits only one bit for one
sample. Thus the signaling rate and transmission CO3 K2
16. channel bandwidth is quite small for delta modulation.
The transmitter and receiver implementation is
very much simple for delta modulation. There is no
analog to digital converter involved in delta modulation
58
6.6 Part B Q & A (with K level and CO)

S.No CO’S Blooms

PART B Level

Explain delta modulation in detail with the help

1. of transmitter and receiver. CO3 K2

Describe Delta Modulation system in detail with a neat block

2.
diagram. Also illustrate the two forms of quantization CO3 K2
error in Delta Modulation.
Describe and illustrate Delta Modulation and its Quantization
3. CO3 K2
error in detail.
(i) Explain the working of differential PCM modulation
4.
scheme. CO3 K2
(ii)Compare DPCM with Delta Modulation system.
In a DM system, the voice signal is sampled at a rate of 64
5.
KHz. The maximum signal amplitude is 1 Volt, voice signal
bandwidth is 3.5 KHz.
(i)Determine the minimum value of step size to avoid CO3 K2
slope overload.
(ii) Determine noise power.
(iii)Assuming signal to be sinusoidal, calculate signal
power and signal-to-noise ratio.

60
 6.7 NPTEL REFERENCE VIDEO LINKS
(For Extended Learning)
1. https://nptel.ac.in/courses/117/101/117101051/
Topics covered:
1. Introduction to Digital Communication
2. Sampling
3. Quantization
4. Encoding
5. PCM and Delta Modulation
6. Digital Modulation Techniques
7. Source Coding

2. https://nptel.ac.in/courses/108/102/108102096/
Topics covered:
All the topics of Unit I to V

3. https://ocw.mit.edu/courses/electrical-engineering-and-computer-science/6-450-
principles-of-digital-communications-i-fall-2006/video-lectures/
Topics covered:
All the topics of Unit I to V

4. http://www.infocobuild.com/education/audio-video-courses/electronics/modern-
digital-communication-iit-kharagpur.html
Topics covered:
All the topics of Unit I to V
6.8 Real time Applications in day to day life and
to Industry
1.PCM,DM and Adaptive DM

https://www.youtube.com/watch?v=yaFqc34kwNk&list=PLmzPxio-
afCCg5eVxvqtPRKXhdrF82pCw&index=36

2.DM & its Applications

https://www.youtube.com/watch?v=i8c4t9ck0cs

3. DM Applications

• Radio & Telephone communication systems

• For ECG/EEG Database waveform Reduction and real time signal
processing

https://www.youtube.com/watch?v=WyjGCEWU4zY

https://link.springer.com/article/10.1007/BF02368458

4. Delta Modulation Applications

https://ieeexplore.ieee.org/document/6767546
6.9 Content beyond Syllabus

Pulse Amplitude Modulation (PAM):

It is the most basic type of Pulse Modulation. Each sample in this sort of
modulation is proportional to the amplitude of the signal at the time of
sampling. As the signal traces out the path of the entire wave, the PAM signal
follows the amplitude of the original signal. In this case, a Nyquist-sampled
signal can be reconstructed by passing it through an efficient Low Pass Filter
(LPF) with a precise cutoff frequency. PAM is simple to create and
demodulate. This method sends data by encoding it in the amplitude of a
series of signal pulses.
Pulse Width Modulation (PWM):
Pulse Width Modulation (PWM), also known as Pulse Duration Modulation
(PDM), or Pulse Time Modulation (PTM), is an analog modulation system in
which the duration, width, or time of the pulse carrier varies proportionally to
the instantaneous amplitude of the message signal. The width of the pulse
varies in this manner, but the signal amplitude remains constant. Amplitude
limiters are used to keep the signal's amplitude consistent. These circuits limit
the noise by clipping the amplitude to the desired level.
Pulse Position Modulation (PPM):
The amplitude and width of the pulse are both kept constant in this sort of
modulation. We alter the position of each pulse about a specific pulse. In this
case, a single pulse with the required number of phase changes is sent. So,
pulse position modulation is an analog modulation system in which the
amplitude and breadth of the pulse remain constant while the position of the
pulse concerning the direction of a reference pulse varies depending on the
instantaneous value of the message signal.
Applications of PAM:
1. It is mostly used in Ethernet communication.
2. This approach is used by many microcontrollers to create control signals.
3. It is employed in photobiology.
4. It functions as a driver for LED circuits.

Applications of PPM:
1. It is utilized in air traffic control and telecommunications networks.
2. Pulse code modulation is used in remote-operated autos, planes, and trains.
3. It is used to compress data and, therefore, for storage.

Applications of PWM:
1. Drive a buzzer with varying levels of volume.
2. Control the motor's speed.
3. Control the movement of a servo.
4. Make an analog output available.
5. Create an audio signal.
6. Message encoding in telecommunications
 7. Assessment Schedule

Assessment Proposed Date Actual Date

Unit 1 Assignment
Assessment
Unit Test 1

Unit 2 Assignment
Assessment
Internal Assessment 1 12.02.2024

Retest for IA 1

Unit 3 Assignment
Assessment
Unit Test 2

Unit 4 Assignment
Assessment
Internal Assessment 2 01.04.2024

Retest for IA 2

Unit 5 Assignment
Assessment
Revision Test 1

Revision Test 2

Model Exam 20.04.2024

Remodel Exam

University Exam 11.05.2024

65
8. Prescribed Text Books & Reference Books

Text Books

1. Wayne Tomasi, Advanced Electronic Communications Systems, 6th

Edition, Pearson New International Edition, Noida, India, 2014.

2. Simon Haykin, Communication Systems,5th Edition, Wiley, 2021.

References

1. Sanjay Sharma, Communication Systems (Analog and digital), 7th

Edition, S.K. Kataria & Sons, 2022.

2. Roddy and Coolen, Electronic Communication, 4th Edition, Pearson

Education, Noida, India, 2014.

3. Herbert Taub and Donald Schilling, Principles of Communication

Systems, 4th Edition, McGraw Hill, 2017.

4. HweiKsu and Debjani Mitra, Analog and Digital Communication:

Schaum’s Outline Series, 3rd Edition, McGraw Hill Education, New Delhi,
India., 2017.

66
9. Mini Project suggestions

MINI PROJECTS LIST

S. NAME OF THE PROJECT COs BLOOMS LEVEL

No

1 Design of a PCM based digital Communication CO3 K4

system to transmit and receive audio signal

2 Design of a PCM based digital Communication CO3 K4

system to transmit and receive ECG signal

3 Design of a DPCM based digital Communication CO3 K4

system to transmit and receive audio signal

Design of a DM based digital Communication CO3 K4

4 system to transmit and receive audio signal

Design of a DPCM based digital Communication CO3 K4

5 system to transmit and receive any biological
signal of your choice

67
 Thank you

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