9/26/2024

CE 401
DIGITAL COMMUNICATION I

Dr. Saad Muhi Falih

TEXTBOOK
Digital Communication

T L SlNGAL
Professor
Department of Electronics and Communication Engineering School of Electronics and Electrical Engineering
Chitkara University
Rajpura, Punjab

Me Graw
Education

McGraw Hill Education (India) Private Limited

NEW DELHI
McGraw Hill Education Offices
New Delhi New York St Louis San Francisco Auckland Bogota Caracas Kuala Lumpur Lisbon London Madrid Mexico
City Milan Montreal San Juan Santiago Singapore Sydney Tokyo Toronto

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What is a communication system?.

 Communication systems are designed to transmit

information.
 There are two type of communication system
• Analog Communication Systems
• Digital Communication Systems
 Communication systems Design concerns:
• Selection of the information–bearing waveform;
• Bandwidth and power of the waveform;
• Effect of system noise on the received information;
• Cost of the system.

These factors will be discussed later in this course

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Digital and Analog Sources and Systems

Basic Definitions:
• Analog Information Source:
An analog information source produces messages which are defined on a
continuum. (E.g. :Microphone)

• Digital Information Source:

A digital information source produces a finite set of possible messages.
(E.g. :Typewriter)

x(t) x(t)

t t
Analog Digital

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Digital Communication

• analog information signal is converted to digital data (sequence of binary

symbols).

• To perform the processing digitally, there is a need for an interface between the
analog signal and the digital processor.

• This interface is called an analog-to-digital (A/D) converter. The output of the

A/D converter is a digital signal that is appropriate as an input to the digital
processor.

• waveform coding techniques such as pulse code modulation, differential pulse

code modulation, or delta modulation.

Digital Processing of Continuous Time Signals

Signals can be processed numerically by a digital computer or using a

DSP chip. We need:
1. Analog to Digital Converter (ADC): convert the signal to a
numerical sequence
2. Digital to Analog Converter (DAC): convert it back to analog, if we
need to.

ADC DSP

DAC

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Introduction: Digital Signal Processing?

Digital signal processing in an analog world

Analog Digital Analog

domain
domain domain

011010 100110
0101 0010
Analog-to- Digital-to-
Digital
Digital Analog
communication
Analog IN Conversion Conversion Analog OUT
Digital Digital
IN OUT
y (t )
x(t ) Digital

1.2 Digital Communication

 We know that analog signals are characterized by data whose values

vary over a continuous range of amplitude levels and are defined for
a continuous range of time.
 These types of analog signals are converted to digital data having
only a finite number of possible values using a device known as
codec (coder-decoder).
 The digital signal thus produced is known as digitized analog
data, and the process is also known as digitization.
 On the receiver side, a similar codec device converts the received bit
stream to the original analog information data.

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1.2 Digital Communication

The encoding of analog information to produce digital

data and then converting it to suitable signaling formats
makes digital transmission possible.
Most electronic communication systems, wire line or
wireless, are going “digital”.
The principle feature of a digital communication system
is that during a finite interval of time, it sends a
waveform from a finite set of possible waveforms at the
transmitter. At the receiver, the objective is just to
determine from a noise-affected signal which waveform
from the finite set of waveforms was sent by the
transmitter. So it is quite easy to regenerate digital
signals.

1.2 Digital Communication

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PCM, and Delta Modulation and Demodulation 1.3

 to be converted to digital pulses prior to transmission and converted

back to analog signals in the receiver.
 The conversion of analog signal to digitized pulses is known as
waveform coding.
 The digitized signals may be in the form of binary or any other form
of discrete-level digital pulses. Figure 1A depicts a simple model of
digital transmission of analog data by using waveform coding.
 An analog information signal is converted to digital data (sequence
of binary symbols) by waveform coding techniques such as pulse
code modulation, differential pulse code modulation, or delta
modulation. The occurrence of binary digits, 1s and 0s, is not
uniform in any digital data sequence. These binary digits are
converted into electrical pulses or waveforms.

Figure 1A Analog-to-Digital Encoding

Digital and Analog Sources and Systems

 A digital communication system transfers

information from a digital source to the intended
receiver (also called the sink).
 An analog communication system transfers
information from an analog source to the sink.
 A digital waveform is defined as a function of time
that can have a discrete set of amplitude values.
 An Analog waveform is a function that has a
continuous range of values.

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Digital Communication
 Digital communication systems offer several advantages over analog communication systems.
Here are some of the key advantages:

1. Noise Resistance: Digital signals are less susceptible to noise and interference compared to
analog signals. Digital signals can be easily reconstructed at the receiving end, reducing the
impact of noise.

2. Signal Quality: Digital signals can maintain their quality over long distances without significant
degradation, whereas analog signals can suffer from attenuation and distortion over long
transmission lines.

3. Error Detection and Correction: Digital communication systems can incorporate error detection
and correction techniques, such as checksums and parity bits, to ensure the accuracy of the
transmitted data.

4. Multiplexing: Digital signals can be easily multiplexed, allowing multiple data streams to be
transmitted simultaneously over the same channel, which is more efficient than analog
multiplexing techniques.

5. Compression: Digital data can be compressed efficiently, reducing the amount of data that
needs to be transmitted. This is essential for multimedia applications like video streaming and
music downloads.
7

Digital Communication
6. Flexibility: Digital systems are highly adaptable and can support a wide
range of services, including voice, video, and data transmission, using
the same infrastructure.
7. Signal Processing: Digital signals can undergo various types of signal
processing, such as filtering, equalization, and modulation, to improve
their quality and performance.
8. Security: Digital communication systems can implement advanced
encryption and security measures to protect the confidentiality and
integrity of transmitted data.
9. Storage and Reproduction: Digital data can be easily stored,
manipulated, and reproduced without loss of quality, making it ideal for
data storage and retrieval applications.
10. Compatibility: Digital systems can interface with a wide range of devices
and networks, making them compatible with modern technology and
facilitating integration with other digital systems.
11. Overall, digital communication systems offer superior performance,
reliability, and versatility compared to analog systems, which is why
they have largely replaced analog systems in most modern communication
technologies.
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Digital Communication
 Disadvantages
• Generally, more bandwidth is required than that for analog systems;
• Synchronization is required.

PCM, and Delta Modulation and Demodulation 1.5

A Typical Digital Communication Link

Analog
signal in

Estimate of analog
signal out

Figure 1B Block Diagram of a Typical Digital Communication Link

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PCM, and Delta Modulation and Demodulation 1.5

A Typical Digital Communication Link

1. The analog signal input is first sampled and quantized to convert an
analog signal into equivalent digitized analog signal.
2. In order to remove redundant information, the digitized analog signal is
source encoded without compromising the ability of the receiver to
provide a high-quality reproduction of the original signal.
3. The channel encoder introduces controlled redundancy bits into the
analog-encoded signal to provide protection against channel noise.
4. A wireless channel produces errors in the form of data bursts, mainly due
to deep signal fades.
5. To mitigate this particular channel impairment, an interleaver is used for
the purpose of pseudo-randomizing the order of the binary symbols in the
channel-encoded signal in a deterministic manner.
6. The function of a packetizer is to convert the encoded and interleaved
sequence of digitized analog data into successive packets. Each packet
occupies a significant part of a basic data

What are the differences between source coding and

channel coding?
 SOURCE CODING
• we decrease the number of redundant bits of information to reduce
bandwidth.
• How can one decide what is redundant information? The answer
is the probability of that message or information.
• It is as simple as if probability is higher ( that means information is
already known or event is fixed ) then the number of bits required to
represent that information will be less. Therefore average length
decreases. So bandwidth reduces! simple!
• CHANNEL CODING
• We add ( append ) extra bits with data in channel coding.
• Why do we add extra bits? - To protect information from errors.
• In simple language, extra bits are necessary for crosschecking or verifying
of received information.
• So now we are compromising with bandwidth for error detection and
correction.

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PCM, and Delta Modulation and Demodulation 1.7

1.1.1 Sampling Theorem

• The process of sampling is essentially based on the sampling theorem which
determines that the sampling rate must be large enough to allow the analog
signal to be reconstructed from the samples with adequate accuracy.

• The sampling theorem for a baseband signal (strictly band limited analog
signal of finite energy) may be stated in two equivalent parts:
1. A baseband signal having no frequency components higher than fm Hz may
be completely recovered from a knowledge of its samples taken at the rate of
at least 2 fm samples per second, that is, the sampling frequency fs ≥ 2 fm.
2. A baseband signal having no frequency components higher than fm Hz is
completely described by its sample values taken at uniform intervals less
than or equal to 1/(2fm) seconds apart, that is, the sampling interval Ts ≤
1/(2fm) seconds.

PCM, and Delta Modulation and Demodulation 1.7

Baseband Signal

• The minimum sampling rate fs = 2 fm samples per second is called the

Nyquist rate.
• The maximum sampling interval Ts = 1/(2fm) seconds is called the sampling
time, or Nyquist interval.

Example 1.1.1: Using the Nyquist Sampling theorem for a baseband signal, determine
the Nyquist rate for an analog signal represented by s(t) = 10 sin [2p (4 × 103)t].
Solution the Nyquist rate is given as
fs = 2fm samples per second.
For the given analog signal, s(t) = 10 sin [2p (4 × 103)t]; we deduce that fm = 4
kHz. Hence, Nyquist rate fs = 2 × 4 kHz = 8 kHz

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PCM, and Delta Modulation and Demodulation 1.7

Bandpass Signal

• For Bandpass signal has bandwidth of 𝐵 𝐻𝑧 = (𝑓 − 𝑓 ) 𝐻𝑧. The Nyquist

rate calculated as following:

1
𝑓 = 2𝑓 𝑎𝑛𝑑 the Nyquist interval 𝑇 =
𝑓
 Recommended sampling frequency = 2𝑓

• The Nyquist sampling theorem is also called uniform sampling theorem

because the samples are taken at uniform intervals.

PCM, and Delta Modulation and Demodulation 1.7

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1.10 Digital Communication

Ans.

1.1.2 Methods of Sampling

As discussed in the preceding section, sampling is required to faithfully represent an analog
signal by its discrete samples in the time domain. For practical needs, these samples are of
finite width in the form of pulses.
There are three distinct methods of sampling:
1. Ideal Sampling—An impulse at each instant of sampling. It is also known as impulse
sampling.
2. Natural Sampling—A pulse of short width with varying amplitude at each instant of sampling.
3. Flat-top Sampling—A pulse of short width with fixed amplitude at each instant of sampling.
In ideal sampling, an arbitrary analog signal is sampled by a train of impulses at uniform
intervals, Ts. An impulse (having virtually no pulse width) is generated at each instant of sampling.
Figure 1.1.1 shows the waveform for ideal sampling.

Ideal Sampling

Figure 1.1.1 Ideal Sampling

PCM, and Delta Modulation and Demodulation 1.11

Natural sampling refers to sampled signals when tops of the sampled pulses
retain their natural shape during the sample interval. In natural sampling, an
arbitrary analog signal is sampled by Natural sampling a train of pulses having
finite short pulse width occurring at uniform intervals. The amplitude of each
rectangular pulse follows the value of the analog information signal for the
duration of the pulse. Figure 1.1.2 shows the waveform for natural sampling.

The original analog signal can be recovered at the receiver by passing the
sampled signal through an ideal low-pass filter without any distortion provided
its bandwidth satisfies the condition fm < B < (fs – fm).
• It is quite evident that there are certain disadvantages of natural sampling.
It is difficult for an analog-to-digital converter to convert the natural sample
to a digital code. In fact, the output of analog-to-digital converter would
continuously try to follow the changes in amplitude levels and may never
stabilize on any code.

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PCM, and Delta Modulation and Demodulation 1.11

Flat-top sampling refers to sampled signals when the tops of the sampled pulses
remain constant during the sample interval. In flat-top sampling, an arbitrary analog
signal is sampled by a train of pulses having finite short pulse width occurring at
uniform intervals. A flat-top topped pulse has a constant amplitude established by the
sample value of the signal at some Sampling point within the pulse interval. We have
arbitrarily sampled the signal at the beginning of the pulse, retaining the amplitude of
each rectangular pulse at the value of the analog signal, as shown in Figure 1.1.3.

Flat-top sampling has the merit that it simplifies the design of the electronic
circuitry used to perform the sampling operation.

PCM, and Delta Modulation and Demodulation 1.11

A sample-and-hold circuit is used to keep the amplitude constant during each

pulse in flat-top sampling process.

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PCM, and Delta Modulation and Demodulation 1.11

In flat-top sampled signals, the high-frequency contents of the analog signal are lost
which results into distortion known as the aperture effect (‫)تأثير الثقب او الفجوة‬, or aperture
distortion.
• The aperture effect arises due to the finite pulse width used in flat-top sampling.
• Ideally, sampling should capture the exact instantaneous amplitude of the analog signal
at each sampling point. However, in flat-top sampling, the amplitude is held constant
during the pulse width, causing a loss of high-frequency content. This results in
distortion.
• Since the analog signal’s amplitude continues to change during the sampling period,
this leads to a difference between the original signal and the sampled version,
introducing what’s called aperture error.
In flat-top sampling, the analog signal cannot be recovered exactly by simply passing the
samples through an ideal low-pass filter. It can be easily seen that the use of flat-top
samples results into amplitude distortion. In addition, there is a delay by Tb/2, where Tb is
the width of the pulse, which results into lengthening of the samples during transmission.
At the receiver, amplitude distortion as well as delay causes errors in decoded signal.
However, the distortion may not be large.
In ideal, natural, or flat-top sampling, the sampling rate must be at least twice the highest

PCM, and Delta Modulation and Demodulation 1.13

aliasing distortion
What happens if sampling rate is less than that of the Nyquist rate?
When the sampling rate is reduced (sampling at too low a rate called
undersampling), such that fs < 2 fm, spectral components of adjacent samples will
overlap and some information will be lost. This phenomenon is called aliasing.
Aliasing can be defined as the distortion in the sampled analog signal due to the
presence of high-frequency components in the spectrum of the original analog
signal.
It is also known as aliasing distortion, or foldover distortion.

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PCM, and Delta Modulation and Demodulation 1.13

Note that an arbitrary rectangular waveform has been shown here for simplicity.
• If fs = 2 fm, the resultant sampled signal is just on the edge of aliasing.
• In order to separate the signals sufficiently apart, the sampling frequency fs should be
greater than 2 fm, as stated by the sampling theorem.
• If aliasing does take place, the interfering frequency component, called as aliasing
frequency, will be at a frequency
fa = fs – fm
where fa is the frequency component of the aliasing distortion (Hz), fs is the minimum
Nyquist sampling rate (Hz), fm is the maximum analog input (baseband) frequency (Hz).
• It is quite clear that the use of a low-pass reconstruction filter, with its pass-band
extending from –fs/2 to +fs /2, where fs is the sampled frequency, does not yield an
undistorted version of the original analog information signal.

PCM, and Delta Modulation and Demodulation 1.13

• Due to aliasing problem, not only we lose all the components of frequencies above
the folding frequency, fa/2, but aliased frequency components reappear as lower
frequency components. Application Such aliasing destroys the integrity of the
frequency components below the folding frequency.

1.1.4 Anti-aliasing Filter

• An anti-aliasing filter is a low-pass filter of sufficient higher order which is
recommended to be used prior to sampling.
• A practical procedure for the sampling of an analog signal whose frequency spectrum
is not strictly band-limited involves the use of the anti-aliasing filter, also known as
pre-alias filter.

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1.14 Digital Communication

• In practice, anti-aliasing filter is used at the front end of the impulse modulator (used
for sampling).
• This enables to exclude the frequency components greater than the required maximum
frequency component of the information signal. Thus, the application of sampling
process allows the reduction of continuously varying information waveform to a finite
limited number of discrete levels in a unit time interval. Figure 1.1.5 shows the use of
an anti-aliasing filter to minimize aliasing distortion.

1.14 Digital Communication

• It is emphasized that the anti-aliasing operation must be performed before the analog
signal is sampled. Accordingly, the reconstruction filter (low-pass filter) at the receiver
end is designed to satisfy the following characteristics:
1. The passband of the reconstruction filter should extend from zero to fm Hz.
2. The amplitude response of the reconstruction filter rolls off gradually from W Hz to
(fs – 2 fm) Hz.
3. The guard band has a width equal to (fs – 2 fm) Hz which is non-zero for (fs > 2 fm)
Hz.
• An anti-aliasing filter also helps reduce noise.
• Generally, noise has a wideband spectrum. Without anti-aliasing, the aliasing
phenomenon itself will cause the noise components outside the signal spectrum to
appear within the signal spectrum. Anti-aliasing suppresses the entire noise spectrum
beyond fs/2.

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1.15

SOLVED EXAMPLE 1.1.8

A baseband signal having maximum frequency of 30 kHz is required to be transmitted

using a digital audio system with a sampling frequency of 44.1 kHz. Estimate the aliasing
frequency component available at the output.
Solution For given value of fs = 44.1 kHz, and fm= 30 kHz, we observe that fs < 2 fm.
This would result into aliasing frequency which is given as
fa = fs – fm = 44.1 kHz – 30 kHz = 14.1 kHz
Thus, the output would have the original baseband frequency of 30 kHz as well as
aliasing frequency of 14.1 kHz.
SOLVED EXAMPLE 1.1.9
For a maximum audio input frequency of 4 kHz, determine the minimum sampling rate
and the aliasing frequency produced if a 5 kHz input signal were allowed to enter the
sampler circuit.
Solution For given maximum audio input analog frequency of fm = 4 kHz, the
recommended sampling rate is given by Nyquist’s sampling theorem as fs ≥ 2 fm.
Therefore, fs ≥ 8 kHz Ans.
If a 5 kHz audio signal enters the sampler circuit, an aliasing frequency fa of 3 kHz (8
kHz – 5 kHz) is produced. Hence, fa = 3 kHz Ans.

1.15
SOLVED EXAMPLE 1.1.10

Realizable filters require a nonzero bandwidth for the transition between the
passband and the required out-of-band attenuation, called the transition
bandwidth. Consider 20% transition bandwidth of the anti-aliasing filter used in
a system for producing a high-quality digitization of a 20-kHz bandwidth music
source. Determine the reasonable sampling rate.

Solution For given fm = 20 kHz, and 20% transition bandwidth of the anti-aliasing filter,
Transition bandwidth = 20 kHz × 0.2 = 4 kHz
Therefore, the practical Nyquist sampling rate, fs ≥ 2.2 fm
The practical Nyquist sampling rate fs ≥ 44.0 ksamples/second
The reasonable sampling rate for the digital CD audio player = 44.1 ksamples/second
Standard sampling rate for studio-quality audio = 48 ksamples/second.

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1.15

SOLVED EXAMPLE 1.1.11

Consider an analog signal given by s(t) = 2 cos (2p 100t) cos (2p 10t) is sampled at the
rate of 250 samples per seconds. Determine the maximum frequency component present
in the signal, Nyquist rate, and cut-off frequency of the ideal reconstruction filter so as to
recover the signal from its sampled version. Draw the spectrum of the resultant sampled
signal also.
Solution The analog signal, s(t) = 2 cos (2p 100t) cos (2p 10t) (Given)
Using the trigonometric identity 2 cos A cos B = cos (A + B) + cos (A – B), we have
s(t) = cos (2 p 110 t) + cos (2 p 90 t)
f1 = 110 Hz; f2 = 90 Hz
• the maximum frequency present in analog signal, fm = 110 Hz

• The sampling frequency, or Nyquist rate, fs ≥ 2 fm

fs ≥ 220 Hz Ans.
• The cut-off frequency of the ideal reconstruction filter should be more than the
Nyquist rate. Therefore, fs > 220 Hz

1.16 Digital Communication

Figure 1.1.6 Spectrum of Sampled Band-pass Signal

SOLVED EXAMPLE 1.1.12
Let the maximum frequency component (fm) in an analog signal be 3.3 kHz. Illustrate the
frequency spectra of sampled signals under the following relationships between the
sample frequency, fs and the maximum analog signal frequency, fm.
(a)fs = 2 fm
(b)fs > 2 fm
(c)fs < 2 fm

Solution
(a) For a given value of fm = 3.3 kHz,
Given, fs = 2 fm; fi fs = 2 × 3.3 kHz = 6.6 kHz
Figure 1.1.7 illustrates the frequency spectra of sampled signals for fs = 2 fm. (In ideal
sampling, spectrum will repeat for every fs Hz, as shown).

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1.16 Digital Communication

Figure 1.1.7 Frequency Spectra of Sampled Signals for fs = 2 fm

It may be noted here that any arbitrary waveform can be drawn with equal spacing
between their centers.
(b) For given value of fm= 3.3 kHz, fs is greater than 2 fm (i.e., 6.6 kHz).
Let fs = 8 kHz,
Therefore, guard band = fs – 2 fm = (8 – 2 × 3.3) kHz = 1.4 kHz
Figure 1.1.8 illustrates the frequency spectra of sampled signals for fs > 2 fm

PCM, and Delta Modulation and Demodulation 1.17

|XS(^| Filter characteristic to recover

waveform from sampled data

Figure 1.1.8 Frequency Spectra of Sampled Signals for fs > 2 fm

(c) For given value of fm= 3.3 kHz, fs is less than 2 fm (6.6 kHz).
Let fs = 6 kHz,
Therefore, overlap band = [fm – (fs – fm)] = [3.3 – (6 – 3.3)] kHz = 0.6 kHz
Figure 1.1.9 illustrates the frequency spectra of sampled signals for fs < 2 fm

The extent of overlapping between adjacent waveforms is clearly seen.

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PCM, and Delta Modulation and Demodulation 1.17

YOU ARE NOW READY TO ATTEMPT THE FOLLOWING PROBLEMS

Ex 1.1.13 An analog signal having a single frequency as 4 kHz is sampled with a 7 kHz
signal. Describe its frequency spectrum. What will be the output if the sampled
signals are passed through a low-pass filter having cut-off frequency at 3.5 kHz?
Ex 1.1.14 The specified voice spectrum is 300 Hz–3400 Hz. The sampling frequency
used is 8 kHz. In practice, the frequency spectrum of human voice extends much
beyond the highest frequency necessary for communication. Let the input analog
information signal contain a 5 kHz frequency component also. What would happen
at the output of the sampler? How can this problem be prevented?
Ex 1.1.15 An analog signal having a single frequency as 4 kHz is sampled with a 10 kHz
signal. Describe its frequency spectrum. What will be the output if the sampled
signals are passed through a low-pass filter having cut-off frequency at 5 kHz?

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