Modulation Techniques

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Modulation Techniques

Fairozmani Binti Sulaiman

Politeknik Kota Bharu

Published and printed by:
Department of Electrical Engineering
Politeknik Kota Bharu
KM. 24, Kok Lanas, 16450 Ketereh, Kelantan
Modulation Techniques
Published Edition 2022
© Fairozmani Binti Sulaiman

All rights reserved. No part of this publication may be reproduced, stored in a retrieval
system, or transmitted in any form or by any means, electronic, photocopying, recording
or otherwise, without the prior written permission of the Department of Electrical
Engineering, Politeknik Kota Bharu.

Modulation Techniques / Fairozmani Binti Sulaiman

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APPRECIATION

Thanks to Allah s.w.t for the strength that has been given to me in preparing this book. I
would also like to take this opportunity to thank the Head of the Electrical Engineering
Department, Puan Fatimah binti Ab Rahman for the encouragement given. The same
speech was also addressed to the E-Learning Coordinator of the Department of Electrical
Engineering, namely Mrs. Sheilani binti Shaari who to some extent has contributed ideas
to further strengthen the writing of this book. Many thanks also to those who have played
a direct or indirect role in the success of this book. Lastly, not to be missed also to the
beloved family who has given me a lot of moral support.

Fairozmani Binti Sulaiman
Department of Electrical Engineering
Politeknik Kota Bharu
KM. 24, Kok Lanas, 16450 Ketereh, Kelantan

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ABSTRACT

This book is written based on Chapter 2 contained in the Communication System
Fundamentals course taken by semester three students in the Department of Electrical
Engineering at Politeknik Malaysia. This book will introduce us to the concepts of
modulation techniques. Why use modulation at all? Modulation allows us to send a signal
over a bandpass frequency range. If every signal gets its own frequency range, then we
can transmit multiple signals simultaneously over a single channel, all using different
frequency ranges. So, in the scope of communication systems, modulation plays hold
crucial responsibility in the communication system to encode information digitally in the
analog world. To be clear, the writer dive into the detailed concept of knowing what
modulation is, its different types in it, and what are the types of modulation techniques
used in communication systems. These modulation techniques are classified into two
major types: analog and digital or pulse modulation. Prior to discussing further, the
different types of modulation techniques, let us understand the importance of modulation
and demodulation. Also, it is more important to have a clear idea of what are the
disadvantages of modulation and demodulation. Hopefully, this book can preferably be
utilized by all lecturers and students.

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ABOUT AUTHOR

Fairozmani binti Sulaiman was born in 1978 in Tanah Merah,
Kelantan. Obtained primary education at Sekolah Kebangsaan
Chetok, Pasir Mas and secondary education at Sekolah Menengah
Tanah Merah (1), Kelantan. After that, the author continued her
studies in the field of Electronic Engineering (Communication) at
Politeknik Kota Bharu and then continued her studies for the
Bachelor of Electrical Engineering (Information Technology) at the
Tun Hussein Onn University College of Technology (KUiTTHO, 2003). Thereafter,
continue studies in the same place for a Master of Technical and Vocational Education
(KUiTTHO, 2004). The author started her career as a lecturer at the Department of
Electrical Engineering, Politeknik Kota, Kuala Terengganu in 2005 and is currently
working at the Department of Electrical Engineering, Politeknik Kota Bharu.

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TABLE OF CONTENTS ii
iii
APPRECIATION iv
ABSTRACT v
ABOUT AUTHOR 1
TABLE OF CONTENTS 1
1.0 INTRODUCTION TO MODULATION AND DEMODULATION 2
2.0 MODULATION 3
3.0 DEMODULATION 4
4.0 THE NEED FOR MODULATION AND DEMODULATION 4
5.0 TYPES OF MODULATION 9
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5.1 Analog Modulation
5.2 Digital Modulation
REFERENCES

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1.0 INTRODUCTION TO MODULATION AND DEMODULATION

In the real-world modulation and demodulation are used everywhere. Both use the same principle
to create signals that can be properly sent. As signals are used everywhere in today’s world from
transmitting music to even sending confidential files securely, a modulation and demodulation
technique can be seen in all of them. A device that performs both modulation and demodulation
is called a modem, a name created by combining the first letters of Modulator and Demodulator.

A message signal cannot travel a long distance because of its low signal strength. In addition to
this, physical surroundings, the addition of external noise and travel distance will further reduce
the signal strength of a message signal. So, to send the message signal a long distance, we need
to increase the signal strength of a message signal. This can be achieved by using a high
frequency or high energy signal called carrier signal. A high energy signal can travel a larger
distance without getting affected by external disturbances. We take the help of such high energy
signal to transmit the message signal. This high energy or high frequency signal is known as
carrier signal. The low energy message signal is mixed with the high energy or high frequency
carrier signal to produce a new high energy signal which carries information to a larger distance.
The frequency band occupied by the modulation signal is called the baseband. While the higher
frequency band occupied by the modulated carrier is called the passband.

2.0 MODULATION

Modulation is a process of changing one or more properties of the high frequency analog carrier
signal in proportion with the values of information signal. In this process, the characteristics of the
carrier signal are changed but the message signal characteristics will not be changed. The carrier
signal does not contain any information so even if we change the characteristics of the carrier
signal, the information contained in it will not be changed. However, the message signal contains
information so if we change the characteristics of the message signal, the information contained
in it will also change. Therefore, we always change the characteristics of the carrier signal but not
the message signal. The information (modulating) signal modulates the carrier signal by changing
either its amplitude, frequency, or phase to produce modulated signal. Modulated signal is the
carrier signal that has been modified by information signal. Below is the general form for how a
signal is modulated.

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Figure 2.1: Modulation Block Diagram

Referring to the modulation block diagram in figure 2.1, the information signal (low
frequency) combines with the carrier signal (high frequency) are performed in a transmitter by a
device called a modulator to produce a modulated signal (high frequency). The information can
be in analog or digital form, and the modulator can perform either analog or digital modulator.
Apart from modulator, there are other devices also used to convert the original signal information
into a form more suitable that are performed in a transmitter through transmission media such as
multiplexer, encoder, transducer, light source, and others. In the real-world modulation is used
to send radio signals and other TV signals as well.

3.0 DEMODULATION
Demodulation is the process of extracting information from the transmitted signal (modulated-
carrier signal) or knows as a reverse process of modulation. Demodulated signal is an original
information signal.

Figure 3.1: Demodulation Block Diagram

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Refer to the demodulation block diagram in figure 3.1, the modulated signal (high frequency) is
extracted that performed in a receiver by a device called a demodulator to produce a demodulated
signal (low frequency). A demodulator is an electronic circuit that is used to recover the
information content from the modulated carrier wave. The output signal via a demodulator may
describe the sound, images, or binary data. Demodulators typically include multiple stages of
amplification and filtering to eliminate interference. Apart from demodulator, there are other
devices also used to extracting the information from the transmitted signal that is performed in a
receiver such as de-multiplexer, decoder, and others.

4.0 THE NEED FOR MODULATION AND DEMODULATION
Modulation and Demodulation is extremely necessary in the communication system because of
the following reasons:
1) To avoid mixing signals
2) To increase the bandwidth of the signal
3) Reducing the effect of noise
4) To reduce the height and size of antenna
5) To multiplex greater number of signals
6) To reduce equipment complexity

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5.0 TYPES OF MODULATION

Figure 5.1: Types of Modulation

Figure 5.1 shows the types of modulation which are basically divided into two types of namely
Analog Modulation and Digital Communication.

5.1 Analog Modulation

Analog modulation is the analog carrier signal is change according to the analog information
signal which is refers to the process of transferring an analog baseband (low frequency) signal,
like an audio or TV signal over a higher frequency signal such as a radio frequency band. In
Analog Modulation, both Information Signal and Carrier signal are in analog waveform. Analog
modulation is divided into three types of namely Amplitude modulation (AM), Frequency
modulation (FM), and Phase modulation (PM).

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i) Amplitude Modulation (AM)

Amplitude Modulation (AM) is the process of changing the amplitude of analog carrier
signal in proportion with the amplitude of the analog information signal. In AM, the
amplitude (V) of the carrier signal is change proportional to the information signal, while
the frequency (f) and phase (Ɵ) of carrier signal are remains unchanged.

Figure 5.2: Amplitude Modulation (AM)

Based on figure 5.2, the carrier amplitude is simply changed according to the amplitude
of the information signal. When the information signals amplitude is increased, the carrier
signal amplitude also increased and vice versa. The information (modulating) signal will
modulate the amplitude of carrier signal to produce high frequency AM modulated signal
by using AM Modulator circuit. The shape of AM modulated signal is called AM envelope.
This “envelope” contains the information signal.

Modulation Index

The modulation index or m for AM can be described as the degree to which the carrier is
being modulated by the message signal. To avoid distortion, we must set m to be as close
to or equal to 1 but not exceed 1 (over modulation). Figure 5.3 shows the modulation index
for AM.

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Figure 5.3: Modulation Index for Amplitude Modulation (AM)

Advantage of Amplitude Modulation (AM)

AM is the simplest type of modulation. Hardware design of both transmitter and receiver
is quite simple and less cost effective.

Disadvantage of Amplitude Modulation (AM)

AM is very susceptible to noise.

Application of Amplitude Modulation (AM)

AM radio broadcast.

ii) Frequency Modulation (FM)

FM is the process of changing the frequency of analog carrier signal in proportion with the
amplitude of the analog information signal. In FM, the frequency (f) of the carrier signal is
change proportional to the information signal, while the amplitude (V) and phase (Ɵ) of
carrier signal are remains unchanged.

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Figure 5.4: Frequency Modulation (FM)
Based on figure 5.4, the carrier frequency is simply changed according to the amplitude
of the information signal which is when the information signals amplitude is increased, the
carrier frequency increases and vice versa.

Advantage of Frequency Modulation (FM)
Modulation and demodulation do not catch any channel noise.
Disadvantage of Frequency Modulation (FM)

Circuit needed for FM modulation and demodulation is bit complicated than AM.

Application of Frequency Modulation (FM)

FM radio broadcast.

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iii) Phase Modulation (PM)

PM is the process of changing the phase of analog carrier signal in proportion with the
amplitude of the information signal. In PM, the phase (Ɵ) of the carrier signal is change
proportional to the information signal, while the amplitude (V) and frequency (f) of carrier
signal are remains unchanged.

Figure 5.5: Phase Modulation (PM)

Based on figure 5.5, the carrier phase is simply changed according to the amplitude of the
information signal which is when the information signals amplitude is increased, the carrier
phase increases and vice versa.

Advantage of Phase Modulation (PM)

Modulation and demodulation do not catch any channel noise.

Disadvantage of Phase Modulation (PM)

Circuit needed for PM modulation and demodulation is bit complicated than AM and FM.

Application of Phase Modulation (PM)

Satellite communication.

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5.2 Digital Communication

Digital Communication is defined as the process by which digital devices communicate
information digitally where it is a mode of communication of the information is encoded digitally
as discreet signals and electronically transferred to the recipients. The communication that occurs
in our daily life is in the form of signals such as sound signals. These sound or audio signals are
normally analog in nature. For such communication via sound signals, we must be near the source
of a sound signal, at least within our hearing range. But if the source and receiver are long-
distance, communication needs to be established over a distance, the analog signals are sent
through the wire, using different techniques for effective and efficient transmission.

Basic Elements of Digital Communication

Figure 5.6 shows the basic elements of a digital communication system. For each function in the
transmitting station, there is an inverse operation in the receiver. The analog input signal (such
as an audio or video signal) must first be converted to a digital signal by an analog-to-digital (A/D)
converter. If no analog message is involved, a digital signal (such as the output of a teletype
machine, which is discrete in time and has a finite number of output characters) can be directly
input.

Figure 5.6: Block Diagram for Basic Elements of Digital Communication

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Digital Communication Process
a) Source

The source consists of an analog signal.
For example: A Sound signal

b) Input Transducer

This block consists of an input transducer which takes a physical input and converts it to
an electrical signal.
For example: Microphone

c) Analog to Digital Converter

This electrical signal from Input Transducer is further processed and converted into
Digital Signal by Analog to Digital Converter.

Figure 5.7: Analog to Digitals Conversion

d) Source Encoder

The source encoder compresses the data into the lowest number of bits. This procedure
helps in the efficient operation of the bandwidth. It removes unnecessary bits.

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e) Channel Encoder

The channel encoder, here the coding is done for error correction. During the transmission
of the signal, due to the sound in the channel, the signal may get distorted. To avoid this,
the channel encoder adds some unnecessary bits to the transmitted data. These bits are
error-correcting bits.

f) Digital Modulator

Here the signal which is to be transmitted is modulated by a carrier. The carrier is used
for effective long-distance transmission of data.

g) Digital to Analog Converter

The digital signal extracted from the carrier is then converted again into analog so that the
signal can be passed effectively through the channel or medium.

h) Channel

The channel provides a path for the signal and permits the analog signal to transmit from
the transmitter end to the receiver end.

i) Digital Demodulator

This is the place from where the data retrieving process is started at the receiver end. The
received signal is demodulated and again converted from analog to digital. The signal gets
rebuilt here.

j) Channel Decoder

The channel decoder does the error corrections post sequence detection. The distortions
which might take place during the transmission are corrected by adding some additional
bits. Addition of these bits help in the complete recovery of the original signal.

k) Source Decoder

The resulting signal is again digitized by sampling and quantizing. This is done to obtain
the unadulterated digital output without any loss of information. The source decoder
creates again the source output.

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l) Output Transducer

This is the final block which converts the signal into its original form (which was at the
input of the transmitter). It converts the electrical signal into physical output.
For example: Speaker

m) Output Signal

This is the output for which the entire process is done.
For example: The sound signal received

Figure 5.8: Output Signals

The Need for Digital Communication

The traditional methods of communication were using analog signals for long-distance
communications. Due to the long distance, the analog signal must go through many losses such
as distortion, intervention, interference, and even security breaches too. To minimize and
overcome these types of losses, the signals are now digitized using different techniques. With the
use of digitized signals, communication becomes clearer and more accurate with minimum or no
losses.

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Figure 5.9: Representation of Analog and Digitals Signals

Figure 5.9 represents analog and digital signals. The digital signals consist of 1s and 0s which
indicate High and Low values, respectively.

The Advantages of Digital Communication

As the signals are digitized, there are many advantages of digital communication over analog
communication, such as −

• The effect of distortion, noise, and interference is much less in digital signals as they are
less affected.

• Digital circuits are more reliable.
• Digital circuits are easy to design and cheaper than analog circuits.
• The hardware implementation in digital circuits, is more flexible than analog.
• The occurrence of crosstalk is very rare in digital communication.
• The signal is un-altered as the pulse needs a high disturbance to alter its properties, which

is very difficult.
• Signal processing functions such as encryption and compression are employed in digital

circuits to maintain the secrecy of the information.
• The probability of error occurrence is reduced by employing error detecting and error

correcting codes.
• Spread spectrum technique is used to avoid signal jamming.
• Combining digital signals using Time Division Multiplexing (TDM) is easier than combining

analog signals using Frequency Division Multiplexing (FDM).

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• The configuring process of digital signals is easier than analog signals.
• Digital signals can be saved and retrieved more conveniently than analog signals.
• Many of the digital circuits have almost common encoding techniques and hence similar

devices can be used for several purposes.
• The capacity of the channel is effectively utilized by digital signals.

Application of Digital Communication

• ADC (Analog to Digital Converter)
• DAC (Digital to Analog Converter)
• MODEM – Modulator-Demodulator
• Digital Camera
• Digital Video
• Broadband digital subscriber lines (DSL)
• Telemetry
• Teleconferencing
• Compact Disk (CD)
• Hard Disk Drive
• Personal Communication System (PCS)
• Satellite Communication System

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Types of Digital Communication

Figure 5.10: Types of Digital Communication

Figure 5.10 shows the term Digital Communication covers a broad range of communication
techniques including Digital Radio and Digital Transmission.

1) Digital Radio

Digital Radio is a transmittal of digitally modulated analog carrier signals between two or more
points in a communication system. With digital radio, the information signal and demodulated
signal are in digital form. While the carrier signal and modulated signal are in analog form. The
digital pulses could be originated from computer-generated data or digital transmission system or
digitally encoded analog information signal.
In digital radio system, digital pulses modulate the analog carrier signal to produce digitally-
modulated carrier signal in analog form. Since the modulated signal is in analog form, therefore

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the transmission medium could be a wireless transmission medium (free space) or physically
facility (metallic or fiber optic cable). Digital Radio covered the Digital Modulation methods.

Digital Modulation

Digital Signal cannot be transmitted through free space (wireless) medium but analog signal does.
Therefore, digital data needs to be converted into analog signal by doing the Digital Modulation
techniques. Digital Modulation is the process an analog carrier signal is varied according to the
digital information signal. It is further divided into types such as Amplitude Shift Keying (ASK),
Frequency Shift Keying (FSK), Phase Shift Keying (PSK), and Quadrature Amplitude Modulation
(QAM). In Digital Modulation, Information Signal is in digital waveform, while Carrier signal is in
analog waveform.

i) Amplitude Shift Keying (ASK)

Amplitude Shift Keying (ASK) is an amplitude of an analog carrier signal is varied
according to the digital information signal. In ASK, the amplitude of the carrier signal is
change proportional to the information signal, while both frequency and phase remain
constant. When the binary input data is logic ‘1’, the carrier signal has the changes
amplitude (Vp = A cosωct). When the data is logic ‘0’, the carrier signal has no amplitude
(Vp = 0V).

Figure 5.11: Amplitude Shift Keying (ASK)

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ii) Frequency Shift Keying (FSK)

Frequency Shift Keying (FSK) is an frequency of an analog carrier signal is varied
according to the digital information signal. In FSK, the frequency of the carrier signal is
change proportional to the information signal, while both amplitude and phase remain
constant. When the binary input data is changes from a logic 0 to a logic 1 and vice versa,
the output frequency shifts between two frequencies: logic 1 - frequency (f1) – the carrier
signal is closely waveform and logic 0 - frequency (f0) – the carrier signal is widely
waveform.

Figure 5.12: Frequency Shift Keying (FSK)

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iii) Phase Shift Keying (PSK)

Phase Shift Keying (PSK) is phase of an analog carrier signal is varied according to the
digital information signal. In PSK, the phase of the carrier signal is change proportional to
the information signal, while both amplitude and frequency remain constant. Therefore,
two phases are possible (21 = 2) for the carrier which are logic ‘1’ and logic ‘0’. One phase
represents a logic ‘1’ and other phase represents logic ‘0’. When the binary input data
changes from a 1 to a 0 or from a 0 to a 1, the phase of the output carrier shifts between
two angles that are separated by 180º. Data input is 1 to a 0 - the carrier signal is look
like alphabet (w) or data input 0 to a 1- the carrier signal is look like alphabet (m).

Figure 5.13: Phase Shift Keying (PSK)

iv) Quadrature Amplitude Modulation (QAM)

Quadrature Amplitude Modulation (QAM) is both Amplitude and Phase are varied
according to the digital information signal. In QAM, a signal in which two carrier shifted in
phase by 90 degrees are modulated and the resultant output consists of both amplitude
and phase variations. It is a combination of amplitude shifting and phase shifting. The
process to Quadrature Amplitude Modulation (QAM)can been in Figure 5.14.

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Figure 5.14: Quadrature Amplitude Modulation (QAM)

2) Digital Transmission

Digital transmission is show how even analogue signals may be converted to a digital format prior
to transmission and it is a true digital system where digital signals are transferred between two or
more points in a communication system. The digital signals could be a binary digit (bit 0 and bit
1) or other form of discrete-level digital pulses.

In digital transmission, there is NO analog carrier, and the original source information may be in
digital or analog form. If the information signal is in analog forms, it must be converted to digital
pulses prior to digital transmission and converted back to analog form at the receive end. The

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analog signal is converted into digital signal by using Pulse Code Modulation (PCM) technique.
Since digital pulses CANNOT be propagated through a wireless transmission medium (free
space); therefore, the digital transmission required physically medium such as a metallic cable
(twisted, coaxial cable) or a fiber optic cable. Digital Transmission covered the Pulse Modulation
methods.

Pulse Modulation

Pulse Modulation is a type of modulation in which the carrier signal is transmitted in the form of
pulses. It can be used to transmit analogue information signal. In other words, the carrier consists
of (discrete) pulse trains that are varied in accordance with the information (message)
signal. Pulse modulation is further divided into two types are Analog Pulse Modulation and Digital
Pulse Modulation.

a) Analog Pulse Modulation

Analog Pulse Modulation is an indication of sample amplitude that infinitely variable techniques.
The Analog Pulse Modulation is mainly classified into Pulse Amplitude Modulation (PAM), Pulse
Width Modulation (PWM) or Pulse Duration Modulation (PDM), and Pulse Position Modulation
(PPM). PAM, PWM, and PPM signals are digital signal (discrete-time signal), but those signals
do not represent in a single binary digit (bits). Therefore, PCM technique is needed to convert the
discrete sampled signal (usually PAM) to serial bits.

i) Pulse Amplitude Modulation (PAM)

PAM is an analog pulse modulation technique in which amplitude of the pulses (discrete)
signal is varied according to the analog amplitude information signal. The figure given
below represents a PAM signal.

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Figure 5.15: Pulse Amplitude Modulation (PAM)

As we can see in the figure 5.15, the amplitude of the pulses is varying with respect to the
amplitude of analog modulating signal, like in case of amplitude modulation. But the major
difference is that unlike AM, here the carrier wave is a pulse train rather than continuous
wave signal. In PAM, the amplitude of the pulse train signal is change proportional to the
analog amplitude information signal, while both width and position remain constant. The
higher amplitude of Information signal, the higher amplitude of pulse

ii) Pulse Width Modulation (PWM)

PWM is an analog pulse modulation technique in which width of the pulses (discrete)
signal is varied according to the analog amplitude information signal. As we can see in the
figure shown below, the width of the pulses is varying with respect to the amplitude of
analog modulating signal, like in case of frequency modulation. In PWM, the width of the
pulse train signal is change proportional to the analog amplitude information signal, while
both amplitude and position remain constant. The higher amplitude of Information signal,
the wider of pulse.

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Figure 5.16: Pulse Width Modulation (PWM)

iii) Pulse Position Modulation (PPM)

PPM is an analog pulse modulation technique in which position of the pulses (discrete)
signal is varied according to the analog amplitude information signal. The figure
5.17 represents a PPM signal.

Figure 5.17: Pulse Position Modulation (PPM)

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Based on the figure 5.17, the position of the pulses is varying with respect to the amplitude
of analog modulating signal. In PPM, the position of the pulse train signal is change
proportional to the analog amplitude information signal, while both amplitude and width
remain constant. It is to be noted here that the position of the pulse changes according to
the reference pulses. And these reference pulses are nothing but PWM pulses. Basically,
the falling edge of PWM pulses acts as the starting of the PPM pulses. The higher
amplitude of Information signal, the farther to the right the pulse is positioned.

b) Digital Pulse Modulation

In Digital Pulse Modulation, a code is used to represent the amplitude of the samples that have
been divided into various levels. It indicates sample amplitude at the nearest predetermined level.
The Digital Pulse Modulation techniques are mainly classified into Pulse Code Modulation (PCM)
and Delta Modulation (DM). The other techniques in Digital Pulse Modulation are Differential PCM
(DPCM), Adaptive Differential PCM (ADPCM), Adaptive Delta Modulation (ADM), and Delta-
Sigma Modulation. Both Differential PCM (DPCM) and Adaptive Differential PCM (ADPCM) are
the types of the PCM technique.

i) Pulse Code Modulation (PCM)

In digital transmission, any analog information data should be changed into digital signal
for the digital transmission. PCM is the only digitally encoded modulation technique that
is commonly used for digital transmission. Pulse Code Modulation is a digital pulse
modulation technique to convert the analog signal to digital signal. Basically, it is a
technique needed to convert the discrete sampled signal to serial bits. In PCM, a signal is
in form of 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.

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The following figure 5.18 shows an example of PCM output with respect to instantaneous
values of a given sine wave.

Figure.5.18: PCM output

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 (LPF) prior to sampling prevents aliasing of the message signal. The
basic operations in the receiver section are regenerative repeater,
decoding, and reconstruction of the quantized pulse train. Figure 5.19 shows the block
diagram of PCM which represents the basic elements of both the transmitter and the
receiver sections.

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Figure.5.19: Block Diagram of Pulse Code Modulation (PCM)

1. Low Pass Filter (Transmitter)
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.

2. Sampler & Hold
This sampler is a section to convert the analog input signal to sampling pulses
signal then convert it to the sampled PAM signal.

3. Quantizer
This quantizer is a section to convert the sampled PAM signal to quantized PAM
signal by rounding off the amplitude of sampled signal to quantization levels, L.

4. Encoder
This encoder is a section to convert the quantized PAM signal to parallel code
number, convert the code number to serial binary pulses (encoded word). The
digitization of analog signal is done by the encoder. Encoding minimizes the
bandwidth used.

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5. Repeater
This section increases the signal strength which is to amplify and regenerate the
weakened digital pulses during transmission on the transmission line like
compensate the signal loss, reconstruct the signal, and increase its strength.

6. Decoder & Hold
The decoder circuit decodes the pulse coded waveform to reproduce the original
signal. This circuit acts as the demodulator (convert back the digital pulses signal
to multilevel PAM signals).

7. Low Pass Filter (Receiver)
After the digital-to-analog conversion is done by the repeater and the decoder, a
low-pass filter is employed, called as the low pass filter to smooth the staircase
amplitude of PAM signals to get back the original signal into an analog 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.

Process in PCM

Generally, the sampling, quantizing, and encoding are the three steps or process to
digitize an analog signal that occurs at source encoder section in Pulse Code Modulation
(PCM). Before sampling process, the information signal should be filtered to limit the
maximum information frequency (fmmax) that can enter the sampler as it affects the
sampling rate (fs). Filtering should ensure that do not distort the signal, i.e., remove high
frequency components that affect the signal shape. The three basic steps for PCM are
shown in Figure 5.20.

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Figure 5.20: Three Steps For PCM

To better understand the process of the PCM technique, we can see it according to the
illustrated below. The analog signal is first sampled at an 8-kHz sampling rate. Then each
sample is quantized into 1 of 256 levels and then encoded into digital eight-bit words.

Figure 5.21: The Process Steps In PCM

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Sampling

Sampling is a process where the information signal (in analog signal) is sampled by
sampling pulse signal to the sampled PAM signal which is generated at certain sampling
rate (fs). Sampling process will convert an analog signal (in continuous-time signal) to a
sampled signal (in discrete-time signal) either in PAM, PWM or PPM.

After sampling, the signal value is known only at discrete points in time, called sampling
instants. If these points have a sufficiently close spacing, a smooth curve drawn through
them allows us to interpolate intermediate values to any degree of accuracy, according to
the Shannon’s Theorem; to reconstruct the original signal (analog signal), the sampling
rate (fs) must be equal to or greater than twice the highest frequency of the message
signal (2fmmax). If the sampling rate is less than 2fmax, some of the highest frequency
components in the analog input signal will not be correctly represented in the digitized
output. This phenomenon is called “aliasing”. Under aliasing condition, the analog signal
cannot be reconstructed. If the sampling frequency, fs, is higher than two times the highest
frequency component of the analog signal, fmmax, the original analog signal is completely
described by these instantaneous samples alone, that is, fs ≥2 fmmax. This rate of
sampling is called as Nyquist rate and the sampling theorem is called as Nyquist theorem.

Info
signal

Sampling
Pulse
Signal

PAM
Signal

Figure 5.22: Sampling Process

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According to the figure above it shows the sampling process where an analog information
signal is sampled every TS seconds. Ts is referred to as the Sampling Interval, Sampling
Time or Sampling Period. These sampling, Ts = 1/ fs ≤ 1/2fmmax and 1 /2fmmax is the
maximum Sampling Interval. While fs = 1/Ts is called the Sampling Rate or Sampling
Frequency. From this formula it shows that the higher the sampling rate (fs) the smaller
sampling interval, (Ts), then the closer the recovered signal approaches the original signal.
Ideally, an infinite sampling rate would be desirable in terms of reproducing the original
signal, but it is not practical due to the bandwidth limitation.
There are 3 methods of sampling which are:
a) Ideal Sampling

Idea Sampling is the analog information signal is sampled instantaneously by
pulses. This sampling is not practical and cannot be easily implemented.

Figure 5.23: Ideal Sampling

b) Natural Sampling
Nature Sampling is the more practical sampling which is performed by high-speed
switching.

Figure 5.24: Natural Sampling

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c) Flat-Top Sampling
Flat-Top Sampling is the simplest and the most popular sampling method which
is performed by Sample-and-Hold circuit. However, this circuit creates a flat-top
(staircase) sampled signal.

Figure 5.24: Flat-Top Sampling

Quantization

The second process after the sampling is the quantization process. Quantization is a
process convert sampled PAM signal to the quantized PAM signal where to transmit the
sampled values via a digital system, we must represent each sample value in numerical
form. This requires quantizing where each accurate sample value is rounded off to the
closest numerical value in each numerical set. In the quantizing process the information
in accurate signal values is lost because of rounding off and the original signal cannot be
reproduced exactly anymore.

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Figure 5.25: Quantization

For binary coding, the number of quantization levels is:

L = 2n

where L is the number of quantization levels also known as the midpoint and n is the
number of bits per level in the binary code words. The number of bits, n for each level of
code number depends on the number of quantization level, L used to quantize the samples
which can be determined using M-ary formula of:

n = Log2 L

The more quantization levels, L is used means that an analog signal can be described
more accurate during signal recovery at receiver. This is because the greater number of
bits (n) and quantization level (L) are used, the more quantization error (Qe) could be
reduced and the more accurate the recovery original signal. This leads to the requirement
of a higher bit rate for transmission of the data representing the original message (in real-
time) and, finally, larger bandwidth. However, a PCM code could have only 8 bits
maximum which equates to only L= 28 or 256 levels.

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There are two types of Quantizing method which are:

a) Uniform Quantization – uniform step size, ∆
b) Non-uniform Quantization – non-uniform step size, ∆

The previous example is Uniform quantization where the step size is uniform for each
zone. However, a non-uniform quantization is commonly used because the uniform
quantization is not efficient for a signal that has smaller amplitude. For example, in speech
communication (voice signal), the signal is found have smaller amplitudes rather than
larger amplitudes. Thus, uniform quantization scheme is wasteful for speech signals
because many of quantization levels, L are rarely used. The non-uniform quantizing
method is more suitable because the step size could be adjusted depends on the
amplitude of signal (smaller step size for lower amplitude and larger for higher amplitude).

When a signal is quantized, we introduce an error because Quantization is an
approximation process. The difference between sampled and quantized value is referred
to the quantization error (Qe).

Qe = Quantizied signal – Sampled Signal

Quantization error (Qe) is also called Quantization noise (Qn) where the maximum error
formula for Uniform Quantization is:

Qe = ± ∆/2

where ∆ is a symbol of the step size and the formula for Step size is.

∆ = 2 Vmax/ L or ∆ = (Vmax - Vmin) /L

Meanwhile the Signal-to-Quantization Noise Power Ratio (SQR) is defined by the
equation.

SQR (dB) = 6.02n + 1.76dB

From the SQR equation, it could be seen that the values of SQR is depends on the number
of bits, n. The higher number of bits, the higher value of SQR could be achieved, the more

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quantization error could be reducing, and the more accurate recovery signal could be
achieved. This is because when the number of bits, n is increased, the number of
quantization level, L also increase and the step size, ∆ will become smaller. When the step
size become smaller, the amplitude difference (gap) between sampled signal and
quantized signal could be minimized (or maybe no gap) which results in smaller errors.

Thus, the recovery original signal is more accurate when the quantization errors are
reduced. In conclusion, the quality of sampled PAM signal can be improved by using a
PCM code with more bits, n. But the more bits will introduce higher bit rate. Bit Rate is the
number of bits transmitted during one second and is expressed in bits per second (bps).
The Bit Rate for PCM could be found from the formula:

Bit Rate = fs x n

Encoding

Encoding refers to the process of conversion by which data are symbolized through some
specific symbols, or characters, where it is converting the quantized PAM signal into the
serial bit number. In other words, encoding is a process of translating the quantized signal
into a decimal code number. Then this decimal code number is converted to its
representative binary sequence. In PCM encoding process each sample is represented
as one in the set of eight-bit binary words. In other words, encoding is a process of
translating the quantized signal into a decimal code number. Then this decimal code
number is converted to its representative binary sequence. This process brings more
security to the communication system. That is why the process is important. For long
transmission there is always possibility of unwanted interferences. Encoding saves us
from those attacks.

Advantages of PCM

• It transmits signals uniformly.
• PCM has an efficient SNR.
• PCM always offers efficient regeneration.

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Disadvantages of PCM

• Attenuation occurs due to noise and cross-talks.
• PCM needs a larger bandwidth for transmission.
• Other errors are also observed during transmission.

ii) Differential Pulse Code Modulation (DPCM)

If the redundancy is reduced, then the overall bitrate will decrease, and the number of bits
required to transmit one sample will also reduce. This type of digital pulse modulation
technique is called Differential Pulse Code Modulation (DPCM). Differential Pulse Code
Modulation (DPCM) is a technique of analog to digital signal conversion. The DPCM works
on the principle of prediction. The value of the present sample is predicted from the
previous samples. The prediction may not be exact, but it is very close to the actual sample
value. This technique samples the analog signal and then quantizes the difference
between the sampled value and its predicted value, then encodes the signal to form a
digital value.

DPCM Transmitter

The below figure shows the DPCM transmitter. The transmitter consists of a comparator,
quantizer, prediction filter, and an encoder.

Figure 5.26: DPCM Transmitter

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DPCM Receiver

To reconstruct the received digital signal, the DPCM receiver (shown in the below figure)
consists of a decoder and prediction filter. In the absenteeism of noise, the encoded
receiver input will be the same as the encoded transmitter output.

Figure 5.27: DPCM Receiver

In DPCM, the value is taken based on the previous outputs. The input given to the decoder
is processed and that output is summed up with the predicted output, to obtain better
output. That means here first the decoder will reconstruct the quantized form of the
original signal. Therefore, the signal at the receiver differs from the actual signal by
quantization error, which is introduced permanently in the reconstructed signal.

iii) Adaptive Differential Pulse Code Modulation (ADPCM)

ADPCM which is known as Adaptive differential pulse code modulation. The modulation
like ADPCM was developed in the year 1970 by Bell Labs. It is an extremely efficient
digital coding of signals for voice coding purposes. In the year 1990, this kind of
modulation was used by IMA which is known as Interactive Multimedia Association for the
legacy audio codec development. So, this modulation is also called ADPCM IMA ADPCM,
DVI, or DVI4. Some of the methods of ADPCM are used in VoIP communications.
ADPCM (Adaptive Differential Pulse Code Modulation) is also known as Delta Pulse Code
Modulation. For a signal, it is a compression coding using differential values like DPCM or
Differential Pulse Code Modulation, where the quantization stages scaling is adapted
additionally based on the signal curve. This modulation is used in the structure of a variety
of ITU-T standards like G.726 in audio signals.

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In these applications, the data rate of the output can be adjusted dynamically among 16
kbit/s & 64 kbit/s. Another example of this is DECT (Digital Enhanced Cordless
Telecommunications) which is used in cordless telephones. The ADPCM uses the signal
performance in the past and predicts it in the future and the output signal will signify the
prediction error, which has no importance. So, the signal should be decoded to restore a
strong and original signal. The main function of ADPCM is to transmit sound signals with
the help of fiber-optic long-distance lines. This modulation is very helpful particularly for
associations that arrange digital lines among remote sites for transmitting both voices as
well as data. Before they transmitted, the voice signals are digitized

The ADPCM method is mainly used in the telecommunication field for compression of
speech because this technique will reduce the flow of bit without reducing its quality. This
kind of modulation can be used in all waveforms, images, and audio with high-quality &
data in other moderns.

ADPCM operating principle is, it is one kind of pulse code modulation (PCM) using a
prediction function. Once the signal is processed, an effort can be made to forecast the
additional path of the signal in the next segment. For the signal quantization within the
next time step only the disparity among predicted as well as the real signal can be used.
Because of this different configuration, a smaller number of bits can be utilized to explain
the signal.

By using this technique, both the quantization level & prediction function are adapted one
more with every work step. As compared to DPCM, this control loop gives an enhanced
signal prediction. This kind of modulation is used for imitating computers as well as arcade
games.

The Adaptive Differential Pulse Code Modulation block diagram is simplified like encoder
and decoder are shown in Figure 5.28.

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Figure 5.28: ADPCM

iv) Delta Modulation (DM)

Delta Modulation is a technique that converts or encodes message signal into a binary bit
stream. This technique allows transmission of only 1 bit per sample, where only 1 bit is
used to encode 1 voltage level. As PCM has the property of converting message signal
directly into a sequence of a binary coded pulse, this resultantly increases the bandwidth
requirement of the system. So, to remove the drawbacks of PCM, Delta Modulation is
used.

The operating principle of DM is such that, a comparison between present and previously
sampled value is performed, the difference of which decides the increment or decrement
in the transmitted values. Simply put, when the two sample values are compared, either
we get difference having a positive polarity or negative polarity.

If the difference polarity is positive, then the step of the signal denoted by Δ is increased
by 1. As against in case when difference polarity is negative then step of the signal is
decreased i.e., reduction in Δ. When +Δ is noticed i.e., increase in step size, then 1 is
transmitted. However, in the case of –Δ i.e., decrease in step size, 0 is transmitted.

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Hence, allowing only a single binary bit to get transmitted for each sample. The block
diagram for the delta modulated signal is shown below:

Figure 5.29: Delta Modulation Transmitter

Figure 5.30: Delta Modulation Receiver

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Advantages of Delta Modulation

• Design is easy and simple.
• It is a 1-bit quantizer.
• Modulator & demodulator can be designed easily.
• In delta modulation, the quantization design is very simple
• The bit rate can be designed by the user

Disadvantages of Delta Modulation

• When the value of the delta is small, slope overload distortion is seen, which is a
type of noise.

• When the value of delta is large, granular noise is seen, which is a type of noise.

Delta Modulation (DM) is widely used in radio communication devices and digital voice
storage and voice information transmission where signal quality is less important.

v) Adaptive Delta Modulation (ADM)

An advanced form of Delta Modulation is Adaptive Delta Modulation (ADM). It is the refined
form of delta modulation. This method was introduced to solve the granular noise and
slope overload error caused during Delta modulation. This modulation method is similar
to Delta modulation except that the step size is variable according to the input signal in
Adaptive Delta Modulation whereas it is a fixed value in Delta Modulation.

In Adaptive Delta Modulation, the step size of the staircase signal is not fixed and changes
depending upon the input signal. Here first the difference between the present sample
value and previous approximation is calculated. This error is quantized i.e., if the present
sample is smaller than the previous approximation, quantized value is high or else it is
low. The output of the one-bit quantizer is given to the Logic step size control circuit where
the step size is decided.

At the logic step size control circuit, the output is decided based on the quantizer output.
If the quantizer output is high, then the step size is doubled for the next sample. If the
quantizer output is low, the step size is reduced by one step for the next sample. The block
diagram for the delta modulated signal is shown below:

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Figure 5.31: Adaptive Delta Modulation Transmitter

Figure 5.32: Adaptive Delta Modulation Receiver

Advantages of Adaptive Delta Modulation

• Adaptive delta modulation offers extremely high performance.
• This technique decreases the need for correction circuits in radio design and error

detection.
• Dynamic range is high since the variable step size covers a large range of values.
• Slope overload error and granular error are not seen.
• Slope error is reduced.

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vi) Delta-Sigma Modulation

Delta-sigma (ΔΣ; or sigma-delta, ΣΔ) modulation is a method for encoding analog
signals into digital signals as found in an analog-to-digital converter (ADC) that converts
an analog voltage signal into a pulse frequency, or pulse density, which can be understood
as pulse-density modulation (PDM). This process introduces quantization error noise.
Delta-Sigma has a relationship with to the Delta Modulation where the first step in a Delta-
Sigma Modulation is a using a Delta Modulation, where the delta modulation is the coding
step of a signal coding/compression system. While the Sigma-delta modulation re-
arranges blocks in the modulator/demodulator system to provide a very similar coding,
with much better signal characteristics. As we can see, the Delta-Sigma modulation is
inspired by delta modulation as shown in figure below.

Figure 5.33: Derivation of Sigma-Delta Modulation from Delta Modulation

∆Σ ADCs are now ideal for converting analog signals over a wide range of frequencies,
from DC to several megahertz. Basically, these converters consist of an oversampling
modulator followed by a digital/ decimation filter that together produce a high-resolution
data-stream output.
The goal of delta-sigma modulation is to achieve higher transmission efficiency by
transmitting only the changes (delta) in value between consecutive samples, rather than
the actual samples themselves. ADCs and digital-to-analog converters (DACs) both can
use delta-sigma modulation.

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