Digital Communication

Unit - I BASICS OF DIGITAL COMMUNICATION
OBJECTIVES
• To study about the basic elements or building blocks that constitutes a digital
communication system.
• To study about various types of digital communication channels.
• To study about the various types of signals.
• To study about data transmission.
1.1 INTRODUCTION
The purpose of a communication system is to transmit information-bearing
signals from a source, located at one location, to a user destination, located at
another distant location. Based on the nature of signal processing applied to the
information-bearing signal, communication systems may be broadly divided into two
major systems. They are:
1) Analog Communication System
2) Digital Communication System
In an analog communication system, the information bearing analog signal is
continuously varying in both amplitude and time. It is used directly to modify some
characteristics of a high frequency sinusoidal carrier wave, such as amplitude, phase
or frequency. Speech signal, video signal, temperature signal, pressure signal etc.,
are some examples of analog signal.
In digital communication system, the information bearing digital signal is
processed such that it can be represented by a sequence of binary digits (discrete
messages). Then it is used for ON/OFF keying of some characteristic of a high
frequency sinusoidal carrier wave, such as amplitude, phase or frequency. If the
input message signal is in analog form, then it is converted to digital form by the
processes of sampling, quantizing and encoding. Computer data and telegraph
signals are some examples of digital signal. The key feature of a digital
communication system is that it deals with a finite set of discrete messages.
Digital communication systems are becoming increasingly attractive due to
the ever-growing demand for data communication. Because digital transmission
offers data processing options and flexibilities not available with analog transmission.
Further, developments in digital techniques have led to more and more powerful
microprocessors, larger and larger memory devices and a number of programmable
logic devices. Availability of these devices has made the design of digital
communication systems highly convenient.
Basics of Digital Communication
1.2 DIGITAL COMMUNICATION SIGNAL PROCESSING

The transmission of information (voice, video, or data) over a path (channel)
may consist of wires, waveguides, or space. The principle feature of a digital
communication system is that during a finite interval of time, it sends a signal
waveform from a finite set of possible waveforms. During propagation, the amplitude
and shape of the signal waveform gets degraded. The objective of the receiver is to
determine from a noise-perturbed signal which waveform from the finite set of
waveforms was sent by the transmitter.
Figure 1.1 illustrates an ideal binary digital pulse propagating along a
transmission line. The shape of the waveform is affected by two basic mechanisms.
1) Due to some non-ideal frequency transfer function of all transmission lines
and circuits.
2) Unwanted electrical noise or other interference further distorts the pulse
waveform.
Both of these mechanisms cause the pulse shape to degrade as a function of
line length, as shown in the figure 1.1.
Figure 1.1 Pulse degradation and regeneration
During propagation, before the pulse is degraded to an ambiguous state,

some corrective signal processing methods have to be done. This process is called
as Regeneration.
In Regeneration, the pulse is amplified by a digital amplifier that recovers its
original ideal shape. The pulse is thus “reborn” or regenerated. Circuits that perform
this function at regular intervals along a transmission system are called Regenerative
repeaters.
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1.3 ADVANTAGES OF DIGITAL COMMUNICATION OVER ANALOG

COMMUNICATION
1. The use of “Regenerative repeaters” generate strong error free signal at a

good power level.
2. Digital circuits are less subject to distortion and interference than analog
circuits.
3. With digital techniques, extremely low error rates producing high signal
fidelity are possible through error detection and correction.
4. Digital circuits are more reliable and can be produced at a lower cost than
analog circuits.
5. Digital hardware lends itself to more flexible implementation than analog

circuits.
6. The combining of digital signals using Time Division Multiplexing (TDM) is
simpler than the combining of analog signals using Frequency Division
Multiplexing(FDM).
7. Different types of digital signals (data, telegraph, telephone, television) can
be treated as identical signals in transmission and switching - a bit is a bit.
8. Digital techniques lend themselves naturally to signal processing functions

that protect against interference and jamming or that provide encryption and
privacy.
9. Also, much data communication is from computer to computer, or from digital
instruments or terminal to computer. Such digital terminations are naturally
best served by digital communication links.
10. Storage and retrieval of voice, data or video at intermediate points (in the
transmission path) is easy and is inexpensive in terms of storage space.
11. Signal processing and image-processing operations like compression of
voice and image signals, etc. can easily be carried out.
12. Adaptive equalization can be implemented.
13. Very powerful encryption and decryption algorithms are available for digital
data so as to maintain a high level of secrecy of communication.
14. Availability of powerful microprocessors, larger memory devices, and number
of programmable logic devices has made the design of digital communication
systems highly convenient.
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15. The mathematical theory of logic circuits called as switching theory is a very
useful concept in digital communication.
16. The effect of noise, temperature and parameter variations is very small in
digital circuits.
1.4 DISADVANTAGES OF DIGITAL COMMUNICATION OVER ANALOG

COMMUNICATION
1. Digital systems tend to be very signal-processing intensive compared with

analog systems.
2. Digital systems need to allocate a significant share of their resources to the

task of synchronization at various levels.
3. Sometimes non-graceful degradation occurs in digital communication
systems, ie., when the signal-to-noise ratio drops below a certain threshold,
the quality of service can change suddenly from very good to very poor.
4. Digital communication systems generally need more bandwidth than analog
communication systems.
5. Digital components generally consume more power as compared to analog
components.
1.5 TYPICAL BLOCK DIAGRAM AND TRANSFORMATIONS

The principal feature of a Digital communication system is that during a finite
interval of time, the transmitter sends a waveform from a finite set of possible
waveforms. The receiver has to determine from a noise perturbed signal which
waveform from the finite set of waveforms was sent by the transmitter.
The block diagram of a typical digital communication system with only the
essential blocks is shown in the figure 1.2(a). The functions of encryption,
multiplexing, spreading, multiple access and equalization are optional.
The upper blocks- Formatter, Source encoder, channel encoder, Baseband
processor/ Band pass modulator- denote signal transformations from the source to
the transmitter. The lower blocks-Baseband decoder/Bandpass demodulator,
channel decoder, source decoder, Deformatter – denote signal transformations from
the receiver to the sink. The lower blocks essentially reverse the signal processing
steps performed by the upper blocks. We shall discuss the basic functions of each
of these blocks.
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Transmitter Section
Discrete
channel
Information Baseband
Source and input processor/
Formatter Source encoder Channel encoder Band pass
transducer
modulator
Si(t)
Noise
Synchronization Channel hc(t)
n(t)
Receiver Section x(t)

Output
Signal
Output transducer Baseband decoder
and Information / Band pass
Deformatter Channel
Sink Source Decoder demodulator
Decoder
Figure 1.2(a): Block diagram of a Typical Digital Communication System
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TRANSMITTER SECTION
1) Information source
The Source is where the information to be transmitted, originates. The
information / message may be available in digital form (eg: computer data, tele-type
data). If the information / message available is a non-electrical signal, (eg: video
signal, voice signal) then it is first converted into a suitable electrical signal using an
input transducer. Then the analog electrical signal is sampled and digitized using an
analog to digital converter to make the final source output to be in digital form.
2) Formatter
Formatting transforms the source information into binary digits (bits). The bits
are then grouped to form digital messages or message symbols. Each such symbol
(mi, where i = 1,2,3……M) can be regarded as a member of a finite alphabet set
containing M members. Thus for M=2, the message symbol m i is binary (it
constitutes just a single bit). For M>2, such symbols are each made up of a
sequence of two or more bits (M-ary)
3) Source encoder
The process of efficiently converting the output of either an analog or digital
source into a sequence of binary digits is called source encoding or data
compression. Source coding produces analog-to-digital (A/D) conversion for analog
sources. It also removes redundant (unneeded) information. By reducing data
redundancy, source codes can reduce a system’s data rate (ie., reduced bandwidth).
Formatting and source coding are similar processes, in that they both involve
data digitization. However, source coding involves data compression in addition to
digitization. Hence, a typical digital communication system would either use
formatter, (for digitizing alone) or source encoder (for both digitizing and
compressing).
4) Channel encoder
The channel encoder introduces some redundancy in the binary information
sequence, in a controlled manner. Such introduction of controlled redundancy can
be used at the receiver to provide error correction capability to the data being
transmitted. This minimises the effects of noise and interference encountered in the
transmission of the signal through the channel. Hence channel coding increases the
reliability of the received data and improves the fidelity of the received signal.
Channel coding is used for reliable transmission of digital data.
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5) Base band processor

For low speed wired transmission, each symbol to be transmitted is
transformed from a binary representation (voltage levels representing binary ones
and zeros) to a baseband waveform. The baseband refers to a signal whose
frequency range extends from DC up to a few MHz. The baseband processor is a
pulse modulation circuit. When pulse modulation is applied to binary symbols, the
resulting binary waveform is called Pulse Code-Modulation (PCM) waveform. In
telephone applications, the PCM waveforms are often called as Line codes. After
pulse modulation, each message symbol takes the form of a baseband waveform,
gi(t), where i=1,2…M.
6) Band pass Modulator

For transmission of high speed digital data (eg. Computer communication
systems), the digital signal needs to be modulated. The primary purpose of the
digital modulator is to map the binary information sequence into high frequency
analog signal waveforms (carrier signals).The term band pass is used to indicate that
the baseband waveform gi(t) is frequency translated by a carrier wave to a frequency
that is much larger than the spectral content of gi(t). The digitally modulated signal is
a band pass waveform Si(t), where i=1,2,…..M. The digital modulator may simply
map the binary digit 0 into a waveform S1(t) and the binary digit 1 into a waveform
S2(t). We call this as binary modulation (M=2).
Alternatively, the modulator may transmit K coded information bits at a time by
using M=2K distinct waveforms Si(t), i=1,2,……..M, one waveform for each of the 2K
possible bit sequences. We call this as M-ary modulation (M>2). The band pass
modulator is used for efficient transmission of digital data. The baseband processor
block is not required, if the bandpass modulator block is present. Therefore, these
two blocks are shown as mutually exclusive blocks.
CHANNEL
The communication channel is the physical medium that is used to send the
signal from the transmitter to the receiver. In wireless transmission, the channel may
be the atmosphere (free space). On the other hand, telephone channels usually
employ a variety of physical media, including wirelines, optical fibre cables, and
wireless (microwave radio).
The transmitted signal is corrupted in a random manner by a variety of
possible mechanisms, such as additive thermal noise generated by electronic
15
devices, man-made noise, eg., automobile ignition noise and atmospheric noise, eg.,
electrical lightning discharges during thunderstorms.
As the transmitted signal Si(t) propagates over the channel, it is impacted by
the channel characteristics, which can be described in terms of the channel’s
impulse response hc(t). Also, at various points along the signal route, additive
random noise n(t) distorts the signal. Hence the received signal x(t) must be termed
as the corrupted version of the transmitted signal Si(t). The received signal x(t) can
be expressed as
x(t)= Si(t) hc(t) + n(t) i=1,2…..M
where * represents a convolution operation and n(t) represents a noise process.
RECEIVER SECTION
1. Baseband decoder
The baseband decoder block converts back the line coded pulse waveform to
transmitted data sequence.
2. Band pass demodulator

The receiver front end and/or the demodulator provides frequency down
conversion for each of the received band pass waveform x(t). Digital demodulation
is defined as recovery of a waveform (base band pulse). The demodulator restores
x(t) to an optimally shaped baseband pulse z(t) in preparation for detection.
Detection is defined as decision-making regarding the digital meaning of that
waveform.
Typically there are several filters associated with the receiver and
demodulator
(i) Filtering to remove unwanted high frequency terms (in the Frequency
down conversion of band pass waveforms.)
(ii) Filtering for pulse shaping.
(iii) Filtering option by equalisation to reverse any degrading effects on the
signal caused by the poor impulse response of the channel.
Finally the detector transforms the shaped pulse to an estimate of the
transmitted data symbols (binary or M-ary).
Demodulator is typically accomplished with the aid of reference waveforms.
When the reference used is a measure of the entire signal attributes (particularly
phase), the process is termed coherent. When phase information is not used, the
process is termed non-coherent.
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3. Channel decoder
The estimates of the transmitted data symbols are passed to the channel
decoder. The channel decoder attempts to reconstruct the original information
sequence from knowledge of the code used by the channel encoder and the
redundancy contained in the received data. A measure of how well the demodulator
and decoder perform is the frequency with which errors occur in the decoded
sequence. This is the important measure of system performance called as
Probability of bit error (Pe).
4. Source decoder
The source decoder accepts the output sequence from the channel decoder.
From the knowledge of the source encoding method used, it attempts to reconstruct
the original signal from the source. Because of channel decoding errors and possible
distortion introduced by the source decoder, the signal at the output of the source
decoder is an approximation to the original source output. The difference of this
estimate and the original digital signal is the distortion introduced by the digital
5. Deformatter
If the original information source was not in digital data form and the output of
the receiver needs to be in the original form of information, a deformatter block is
needed. It converts back the digital data to either discrete form (like keyboard
characters) or analog form (speech signal).
6. Information sink
If an analog output is needed in non-electrical form, the output transducer
converts the estimate of digital signal to the required analog output. The information
sink may be computer, data terminal equipment or an user.
7. Synchronization
Synchronization and its key element, a clock signal, is involved in the control
of all signal processing within the digital communication system. It actually plays a
role in regulating the operation of almost every block. Synchronization involves the
estimation of both time and frequency. Coherent systems need to synchronize their
frequency reference with the carrier in both frequency and Phase. For non-coherent
systems, phase synchronization is not needed.
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BASIC DIGITAL COMMUNICATION TRANSFORMATIONS

The basic signal processing functions which may be viewed as
transformations can be classified into the following nine groups.
1. Formatting and Source Coding:

Formatting and source coding are similar processes, in that they both involve
data digitization. Source coding also involves data compression in addition to
digitization.
2. Baseband Signaling
Baseband signaling process involves generation of PCM waveforms or line
codes.
3. Bandpass signaling
During demodulation, when the references used are a measure of all the
signal attributes (particularly phase), the process is termed coherent. When phase
information is not used, the process is termed non coherent.
4. Equalization
An equalization filter is needed for those systems where channel induced ISI
(Intersymbol interference) can distort the signals.
5. Channel Coding
Waveform coding and structured sequences are the two methods of channel
coding. Waveform coding involves the use of new waveforms. Structured
sequences involve the use of redundant bits.
6. Multiplexing and multiple access

Multiplexing and multiple access both involve the idea of resource sharing.
Multiplexing takes place locally and multiple access takes place remotely.
7. Spreading
Spreading is used in military applications for achieving interference protection
and privacy. Signals can be spread in frequency, in time, or in both frequency and
time.
8. Encryption
Encryption and decryption are the basic goals, which are communication
privacy and authentication. Maintaining privacy means preventing unauthorized
persons from extracting information (eavesdropping) from the channel. Establishing
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authentication means preventing unauthorized persons from injecting spurious

signals (spoofing) into the channel.
9. Synchronization
Synchronization involves the estimation of both time and frequency. Coherent
systems need to synchronize their frequency reference with the carrier in both
frequency and phase. For non coherent systems, phase synchronization is not
needed.
The figure 1.2(b) shows the basic digital communication transformations.
Fig. 1.2(b) Basic digital communication transformations
Performance criteria
A digital communication system transmits signals that represent digits. These
digits form a finite set or alphabet, and the set is known a priori to the receiver. A
figure of merit for digital communication systems is the probability of incorrectly
detecting a digit or the probability of error (Pe).
1.6 CHANNELS FOR DIGITAL COMMUNICATION

The transmission of information across a communication network is
accomplished in the physical layer by means of a communication channel. One
common problem in signal transmission through any channel is additive noise.
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1.8 INFORMATION CAPACITY

The information capacity is defined as the maximum rate at which information
can be transmitted across the channel without error. It is measured in bits per
second.
A key issue in evaluating the performance of a digital communication system
is that the maximum rate at which reliable communication can take place over the
channel.
1.9 SHANNON’S LIMIT FOR INFORMATION CAPACITY

Shannon-Hartley capacity theorem:
The Shannon’s channel capacity theorem defines the fundamental limit on the
rate of error free transmission for a power-limited, band-limited Gaussian Channel.
The information capacity C of a channel perturbed by additive white Gaussian Noise
(AWGN) is a function of the average received signal power S, the average noise
power N, and the bandwidth B. The information capacity relationship (Shannon-
Hartley theorem) can be stated as
𝑆
C = B log 2 ( 1 + 𝑁) , bits/s (1.13)
We can rewrite the noise power as N=NoB, where No is the noise power
spectral density. Hence, the theorem can be written as
𝑆
C = B log 2 ( 1 + ) , bits/s (1.14)
𝑁𝑜 𝐵
The significance of the channel capacity is as follows:
(i) If the information rate R from the source is less than or equal to
channel capacity C (R ≤ C), then it is possible to achieve reliable
(error-free) transmission through the channel by appropriate coding.
(ii) If the information rate R from the source is greater than the channel
capacity C (R > C), it is not possible to find a code that can achieve
reliable (error-free) transmission through the channel.
Thus, Shannon established basic limits on communication of information and
gave birth to a new field that is now called Information Theory.
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Example problem 1.1: Calculate the capacity of a standard 4kHz telephone

channel with a 32dB signal-to-noise ratio.
Solution:
The standard telephone channels occupy the frequency range of 300Hz to
3400Hz. Hence, the bandwidth is B=3400-300=3100HZ
Signal-to-noise ratio (S/N) in decibels = 32dB

𝑆
Hence, 10 log10 ( ) = 32
𝑁
𝑆 32
log10 (𝑁) = 10
= 3.2
𝑆
 = anti log(3.2) = 1584.89
𝑁
𝑆
Therefore, = 1585
𝑁
𝑆
Capacity of a channel, C = B.log 2 (1 + 𝑁)
𝑆
On substituting the values of B and𝑁, we have
C = 3100 x log 2 (1 + 1585)
= 3100 x log 2 (1586)

log10 1586 3.2003
= 3100 x = 3100 x 0.3010
log10 2
= 3100 x 10.63 = 32953
capacity, C = 32953 bits per second
Exercise Problem 1.8.1: A system has bandwidth of 4kHz and a signal-to-

noise ratio of 28dB at the input to the receiver. Calculate its information
carrying capacity.
1.10 DATA TRANSMISSION

The data transmission involves the transmission of information such as
digitized voice, digitized image and video, computer generated data, and so on. For
data transmission and reception, a data network is established. A data network is a
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Formatting and Baseband Modulation
Formatting transforms the source information into bits, thus assuring

compatibility between the information and the signal processing steps within the
digital communication system. The information remains in the form of a bit stream
upto the pulse modulation block.
Information sources can be analog or discrete. Hence, the output of an
information source may be digital information, textual information or an analog
information. Data already in a digital format would bypass the formatting function.
Textual information is transformed into binary digits by the use of coder. If the data
is in the form of alphanumeric text, then it will be character encoded with one of
several standard formats such as ASCII, EBCDIC, Baudot, and Hollerith.
Analog information is formatted using three separate processes: Sampling,
quantization and coding. For all types of information sources, the formatting step
results in a sequence of binary digits.
The pulse modulator converts the bit stream into a sequence of pulse
waveforms. The characteristics of this sequence of pulses correspond to the digits
being sent. These pulse waveforms are then transmitted through a baseband
channel, such as pair of wires or a coaxial cable.
After transmission through the channel, the pulse waveforms, are recovered
(demodulated) and detected to produce an estimate of the transmitted digits. The
final step is the reverse formatting, which recovers an estimate of the source
information.
2.3 FORMATTING ANALOG INFORMATION

Analog information sources can be transformed into digital sources through
the use of sampling and quantization. We utilize sampling to convert a continuous
time signal to a discrete time signal, process the discrete-time signal using a discrete
time system and then convert back to continuous-time signals.
2.3.1 The Sampling theorem

Sampling of the signals is the fundamental operation in signal-processing. A
continuous-time signal is first converted to discrete-time signal by sampling process.
Sampling theorem gives the complete idea about the sampling of signals. The
output of the sampling process is called pulse amplitude modulation (PAM).
Because the successive output intervals can be described as a sequence of pulses
with amplitudes derived from the input waveform samples. The analog waveform
can be approximately retrieved from a PAM waveform by simple low-pass filtering.
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The statement of sampling theorem can be given in two parts as below:

(i) A band limited signal of finite energy, which has no frequency
components higher than fm Hertz, is completely described by its sample
1
values at uniform intervals less than or equal to 2𝑓 seconds apart.
𝑚
(ii) A band limited signal of finite energy, which has no frequency

components higher than fm Hertz, may be completely recovered from
the knowledge of its samples taken at the rate of 2f m samples per
second.
Combining the two parts, the uniform sampling theorem may be stated as
follows:
“A continuous-time signal may be completely represented in its samples and
recovered back if the sampling frequency is fs ≥ 2fm”.
Here fs is the sampling frequency and fm is the maximum frequency present in
the signal.
2.3.2 Nyquist theorem

The Nyquist theorem provides a prescription for the nominal sampling interval
required to avoid aliasing. It may be stated as follows:
“The sampling frequency (fs) must be at the rate equal to or greater than twice
the highest frequency component (fm) present in the signal ie.,fs ≥ 2fm , in order to
recover the signal exactly.”
(i) When the sampling rate becomes exactly equal to 2fm samples per
second, then it is called as Nyquist rate. Nyquist rate is also defined as
the minimum sampling rate. It is given by
fs = 2fm (2.1)
(ii) Similarly, maximum sampling interval is called as Nyquist interval. It is
given by
1
Nyquist interval, Ts = 2𝑓𝑚
seconds, (2.2)
1
where, fs =
𝑇𝑠
(iii) The restriction of fs≥ 2fm, stated in terms of the sampling rate, is known as
the Nyquist criterion. The Nyquist criterion is a theoretically sufficient
condition to allow an analog signal to be reconstructed completely from a
set of uniformly spaced discrete time samples.
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2.4 SAMPLING TECHNIQUES

The sampling of a continuous-time signal is done in several ways. Basically,
there are three types of sampling techniques. They are:
1. Impulse sampling
2. Natural sampling
3. Flat top sampling (Sample and hold operation)
2.4.1 Impulse sampling or Ideal sampling

If the sampling function is a train of impulses, then the method is called
Impulse sampling or Ideal sampling. Figure 2.2(c) shows this sampling function.
Figure 2.2(g) shows a circuit to produce this sampling. This circuit is known as the
switching sampler.
X(t) fs Xs(t)
(g)
Figure 2.2 Impulse Sampling
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1
Let us choose Ts=2𝑓 , so that the Nyquist criterion is just satisfied. The circuit
𝑚
simply consists of a switch. If we assume that the closing time ‘t’ of the switch
approaches zero, then the output xs(t) will contain only instantaneous value of the
input signal x(t). This instantaneous sampling gives a train of impulses of height
equal to the instantaneous value of the input signal x(t) at the sampling instant.
The train of impulses (sampling function) may be represented as
x(t) = 𝑛=−∞ (𝑡 − 𝑛𝑇𝑠 )

∑∞ (2.3)
Where Ts is the sampling period and (t) is the unit impulse or Dirac delta
function. The sampled signal xs(t) is expressed as the multiplication of x(t) and x(t).
Thus, xs(t) = x(t).x(t) (2.4)
xs(t) = 𝑛=−∞ (𝑡 − 𝑛𝑇𝑠 )

x(t).∑∞ (2.5)
xs(t) = 𝑛=−∞ 𝑥(𝑛𝑇𝑠 ). (𝑡 − 𝑛𝑇𝑠 )

∑∞ (2.6)
Demerits of Impulse sampling:

Impulse sampling results in the samples whose width T approaches zero.
Due to this, the power content in the instantaneously sampled pulse is negligible.
Thus, this method is not suitable for transmission purpose.
Spectrum:
The spectrum X(f) of the sampled signal xs(t) is shown in the figure 2.2(f) for
fs=2fm. Using the frequency convolution property of the Fourier transform, we can
transform the time domain product x(t).x(t) of equation(2.6) to the frequency domain
convolution X(f)*X(f).
Therefore, Xs(f) = X(f)  X(f)
1
= 𝑛=−∞ (𝑓 − 𝑛𝑓𝑠 )]
X(f) [ 𝑇 ∑∞
𝑠
1
Xs(f) = ∑∞
𝑛=−∞ 𝑋(𝑓 − 𝑛𝑓𝑠 ) (2.7)
𝑇𝑠
(i) From the figure, we infer that, if the sampling rate is chosen such that
fs = 2fm, then each spectral replicate is separated from each of its neighbours by a
frequency band exactly equal to fs Hertz. Therefore, the analog waveform can
theoretically be completely recovered from the samples, by the use of filtering.
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(ii) If the sampling rate is chosen such that fs>2fm, the spectral replications will
move farther apart in frequency, as shown in Figure 2.3(a),
Figure 2.3 Spectra for various sampling rates
Now, it is easier to perform the filtering operation. A typical low-pass filter

characteristic that might be used to separate the baseband spectrum from those at
higher frequencies is also shown in the figure.
(iii) When the sampling rate is reduced, such that f s<2fm, the spectral
replications will overlap, as shown in the figure 2.3(b). Therefore, some information
will be lost. This phenomenon is called “aliasing” which results from under sampling
(sampling at too low a rate).
Conclusion
• The Nyquist rate, fs=2fm is the sampling rate below which aliasing occurs.
• To avoid aliasing, the Nyquist criterion, fs≥2fm must be satisfied.
2.4.2 Natural Sampling

Natural sampling is a practical method. Here the sampling function is a pulse
train or switching waveform xp(t). Figure 2.4(c) shows this sampling function. Figure
2.4(g) shows a functional diagram of a natural sampler.
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Xp(t)
X(t) Xs(t)
(g)
Figure 2.4 Natural Sampling
1
Here also, we choose Ts=2𝑓 , so that the Nyquist criterion is just satisfied.
𝑚
The circuit simply consists of a switch. The pulse train xp(t) is applied to the switch.
1
Each pulse in xp(t) has width T and amplitude 𝑇. The multiplication of input analog
signal x(t) by the pulse train xp(t) can be viewed as the opening and closing of the
switch. The resulting sampled data sequence, xs(t) is shown in figure 2.4(e). It can
be represented as
xs (t) = x(t).xp(t) (2.8)
This process is called natural sampling, since the top of each pulse in the
sampled data sequence retains the shape of its corresponding analog segment
during the pulse interval. We can express the periodic pulse train as a Fourier series
in the form
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xp(t) = ∑∞
n=−∞ Cn e
j2πnfs t
(2.9)
1 𝑛𝑇 1
where Cn = 𝑇𝑠
sinc ( 𝑇 ), T is the pulse width and 𝑇 is the pulse amplitude.
𝑠
Hence, the sampled data sequence is given by
xs(t) = x(t)∑∞
n=−∞ Cn e
j2πnfs t
(2.10)
Disadvantages
Each pulse in the sampled data sequence has varying top according to signal
variation. During transmission, noise interferes the top of pulses. Then it becomes
difficult to determine the shape of top of the pulse at the receiver.
Spectrum
The spectrum of the naturally sampled signal is shown in Figure 2.4(f). The
transform Xs(f) of the sampled waveform is found as follows:
Xs(f) = ℱ{x(t).∑∞
n=−∞ Cn e
j2πnfs t
} (2.11)
We can solve for Xs(f) as below,
Xs(f) = ∑∞
n=−∞ Cn X(f − nfs ) (2.12)
Equation (2.12) and Figure 2.4(f) illustrate that Xs(f) is a replication of X(f),
periodically repeated in frequency every fs Hertz. However, we see that Xs(f) is
weighted by the Fourier series coefficients (Cn) of the pulse train, compared with a
constant value in the impulse sampling.
2.4.3 Flat Top Sampling or Sample-and-Hold operation

In case of natural sampling, the pulse has varying top according to the signal
variation. Therefore, amplitude detection of the pulse is not exact and errors are
introduced in the signal. This problem will be solved by having flat top pulses. A
sample and hold circuit is used to generate flat top pulses.
Figure 2.5 shows the functional diagram of a sample and hold circuit. The
circuit consists of two field effect transistor (FET) switches and a capacitor.
47
Sampling
Switch
G1
x(t) C Xs(t)
G2
Discharge
switch
Figure 2.5 Sample and hold circuit for flat top sampling
By applying a short
pulse to the gate G1, the
sampling switch is closed for
a very small period. During
this period, the capacitor ‘C’
is quickly charged upto a
voltage equal to the x(t) xg(t)
instantaneous sample value
of the incoming signal x(t).
The sampling switch is now
opened and the capacitor
holds the charge. The
discharge switch is then
closed by a pulse applied to
the gate G2 to discharge
capacitor to zero volts. The xs(t) = p(t) * [x(t) xg(t)]
discharge switch is then
opened and thus capacitor
has no voltage. After the
period of Ts, sampling
switch is closed to take new
Figure 2.6 Flat Top Sampling
sample. This periodic
gating of sample and hold
circuit generates a sequence of flat top samples as shown in the figure 2.6.
48
It may be noted that only starting edge of the pulse represents instantaneous
value of the baseband signal x(t). Sample and hold can be described by the
convolution of the sampled pulse train, [x(t)x(t)], with a unity amplitude rectangular
pulse P(t) of pulse width Ts. Hence, convolution results in the flat top sampled
sequence.
xs(t) = P(t)  [x(t)x(t)]
= 𝑛=−∞ (𝑡 − 𝑛𝑇𝑠 )]

P(t)  [x(t)∑∞ (2.13)
Spectrum
The Fourier transform, xs(f), of the time convolution in equation (2.13) is the
frequency-domain product of the transform P(f) of the rectangular pulse and the
periodic spectrum of the impulse-sampled data. Therefore,
Xs(f) = 𝑛=−∞ (𝑡 − 𝑛𝑇𝑠 )}

P(f) ℱ {x(t) ∑∞
1
= 𝑛=−∞ (𝑓 − 𝑛𝑓𝑠 )]}
P(f){X(f)  [𝑇 ∑∞
𝑠
1
Xs(f) = P(f).𝑇 ∑∞
𝑛=−∞ 𝑋(𝑓 − 𝑛𝑓𝑠 ) (2.14)
𝑠
The effect of this product operation results in a spectrum similar in

appearance to the naturally sampled signal shown in the figure 2.4(f). The main
effect of the hold operation is the significant attenuation of the higher frequency
spectral replicates, which is a desired effect.
Example 2.1: A continuous-time signal is given as x(t) = 8cos 200t. Determine the
minimum sampling rate ie., Nyquist rate required to avoid aliasing.
Solution:
The continuous time signal,
x(t) = 8cos 200t
We have,
x(t) = A cos (2f)t = A cost
 A cos (2f)t = 8cos 200t
49
On comparing the values,

2f = 200
200
 f = = 100Hz
2
Hence, the highest frequency component of the given continuous time signal
is fm=100Hz. Therefore, minimum sampling rate required to avoid aliasing is the
Nyguist rate given by
fs = 2fm = 2 x100 = 200Hz.
2.5 ALIASING
When a continuous-time band-limited signal is sampled at a rate lower than
Nyquist rate, fs<2fm, it is termed as undersampling. The spectrum of the sampled
signal is shown in the figure 2.7 and 2.8.
Figure 2.7 Sampled Spectrum for fs< 2 fm
Figure 2.8 Aliasing in the frequency domain
50
Aliasing is a phenomenon which results from the effect of undersampling.

The aliased spectral components represent ambiguous data that appear in the
frequency band between (fs-fm) and fm. Aliasing may be defined as the phenomenon
in which a high frequency component in the frequency-spectrum of the signal takes
identity of a lower-frequency component in the spectrum of the sampled signal.
𝑓
Let the frequencies above half the sampling frequency (> 𝑠 ) be called as the
2
fold over frequencies. From the figure, we see some overlapping in the periodic
replications. This overlapping of successive periods of the spectrum causes the fold
over frequencies in the original signal to appear as frequencies below half the
𝑓
sampling frequency (< 𝑠 ), in the sampled signal. This will cause distortion in the
2
reconstructed signal. This phenomenon is called Aliasing.
Methods to prevent aliasing:

There are two ways of eliminating aliasing using antialiasing filters.
(i) The analog signal is prefiltered using a low pass filter. The bandwidth of the
𝑓
filter is less than or equal to half the sampling frequency (ie.,𝑓𝑚′ ≤ 2𝑠 ). Thus,
there are no aliased components seen in the sampled signal spectrum as
shown in the Figure 2.9.
Figure 2.9 Prefiltering
51
Eliminating the aliasing terms prior to sampling is good engineering practice.

(ii) When the signal structure is well known, the aliased terms can be eliminated
after sampling. Here, the low pass filter operates on the sampled data. The
figure 2.10 shows how the aliased components are removed by post filtering
after sampling.
Figure 2.10 Post Filtering
Table 2.1: Performance comparison of three sampling techniques
Flat Top
Parameter Ideal sampling Natural Sampling
Sampling
1) Sampling It uses multiplication. It uses chopping It uses sample
Principle principle. and hold circuit.
2) Generation Figure 2.2g Figure 2.4g Figure 2.5
circuit
3) Sampling rate Sampling rate tends Sampling rate Sampling rate
to be infinite. satisfies Nyquist satisfies Nyquist
criteria. criteria.
4) Noise Noise interference is Noise interference Noise
interference maximum. is minimum. interference is
maximum
5) Feasibility This is not a This method can This method is
practically possible be used practically. used practically.
method.
52
2.6 SIGNAL INTERFACE FOR A DIGITAL SYSTEM

We know that a digital system deals with a finite number of values. An analog
signal, such as voice, has a continuous range of amplitudes. When it is sampled,
the samples also cover a continuous amplitude range. Because, within the finite
amplitude range of the signal we find an infinite number of amplitude levels. Hence,
the sampled data are not compatible with a digital system.
Figure 2.11 illustrates four ways in which analog source information can be
described.
Figure 2.11 Source Data
(i) Figure 2.11(a) shows the original analog waveform.
(ii) Figure 2.11(b) represents natural-sampled version of the original analog

waveform. Here, the amplitude of each natural sample still has an infinite
number of possible values. Hence, it is not compatible with a digital system.
(iii) Figure 2.11(c) illustrates the original waveform represented by discrete
pulses. Here the pulses have flat tops and the pulse amplitude values are
limited to a finite set. Each pulse is expressed as a level from a finite number
of predetermined levels. These pulses are referred to as quantized samples.
Such a format is the best choice for interfacing with a digital system.
(iv) The format in Figure 2.11(d) may be viewed as the output of a sample and
hold circuit. When the sample values are quantized to a finite set, this format
can also interface with a digital system.
Therefore, the existence of a finite number of discrete amplitude levels is a

basic condition for interfacing with a digital system. The conversion of analog
(continuous) sample of the signal into a digital (discrete) form is called the quantizing
process.
53
2.7 QUANTIZATION
The process of representing a large (possibly infinite) set of values with a
much smaller set of values is called quantization.
Quantizer input Quantizer output
x y=Q(x)
Quantization noise
Figure 2.12 Quantizer
ut
In a linear analog system, the transfer characteristic representing the relation
between the input and the output is a straight line. For a quantizer, the transfer
characteristic is staircase like in appearance
The quantizing process has a two-fold effect:
1) The peak-to-peak range of input sample values is subdivided into a finite set
of decision levels or decision threshold that are aligned with the “risers” of the
staircase, and
2) The output is assigned a discrete value selected from a finite set of
representation levels or reconstruction values that are aligned with the
“treads” of the staircase.
The combination of sampler and quantizer is called Analog-to-Digital(A/D)
converter or digitizer.
2.8 SOURCES OF CORRUPTION

The analog signal recovered from the sampled, quantized, and transmitted
pulses will contain corruption from several sources. The sources of corruption are
related to
1) Sampling and quantizing effects (Quantization noise, Quantizer saturation and
Timing jitter)
2) Channel effects (channel noise and Intersymbol interference)
2.8.1 Sampling and quantizing effects

2.8.1.1 Quantization noise
The distortion inherent in quantization is a round-off or truncation error. The
sample values of an analog baseband signal are rounded-off to the nearest
54
permissible representation levels of the quantizer. This rounding-off or

approximation involves discarding some of the original analog information. The
distortion introduced by the need to approximate the analog waveform with
quantized samples, is referred to as quantization noise. The amount of such noise is
inversely proportional to the number of levels employed in the quantization process.
2.8.1.2 Quantizer saturation

The quantizer (or analog-to-digital converter) allocates L levels to the task of
approximating the continuous range of inputs with a finite set of outputs. The range
of inputs for which the difference between the input and output is small is called the
operating range of the converter.
If the input exceeds this range, the difference between the input and the
output becomes large. At this condition, the converter is operating in saturation.
Generally, saturation is avoided by the use of Automatic Gain Control (AGC), which
effectively extends the operating range of the converter.
2.8.1.3 Timing jitter

We know that the samples of the analog signal are uniformly spaced. If there
is a slight jitter in the position of the sample, the sampling is no longer uniform. The
jitter is usually a random process and thus the sample positions are not accurately
known.
The effect of the jitter is equivalent to frequency modulation (FM) of the
baseband signal.
(i) If the jitter is random, a low-level wideband spectral contribution is
induced. The properties are very close to those of quantizing noise.
(ii) If the jitter exhibits periodic components, the periodic FM will induce low-
level spectral lines in the data.
Timing jitter can be controlled with very good power supply isolation and
stable clock references.
2.8.2 Channel effects

2.8.2.1 Channel noise
Channel noise may be introduced anywhere along the transmission path. The
channel noise is the combined effect of thermal noise, interference from other users,
and interference from circuit switching transients. Channel noise may be modeled as
55
Additive White Gaussian Noise (AWGN) with zero mean and Power spectral
𝑁
density 2𝑜.
The effect of channel noise is to introduce errors in detecting the pulses

carrying the digitized samples. Channel-induced errors can degrade the
reconstructed signal quality quite quickly. This rapid degradation of output signal
quality with channel induced errors is called a threshold effect.
If the channel noise is small, it does not corrupt the reconstruct signals. The
only noise present in the reconstruction is the quantization noise. On the other hand,
if the channel noise is large enough to affect our ability to detect the waveforms, then
there will be reconstruction errors. A large difference in behaviour can occur for very
small changes in channel noise level.
2.8.2.2 Intersymbol interference

The channel is always band limited. A band limited channel disperses or
spreads a pulse waveform passing through it.
(i) When the channel bandwidth is much greater than the pulse band width, the
spreading of the pulse will be slight.
(ii) When the channel bandwidth is close to the signal bandwidth, the spreading
will exceed symbol duration and cause signal pulses to overlap. This
overlapping is called Intersymbol interference (ISI).
ISI causes system degradation (higher error rates). We may use an adaptive
equaliser to correct the channel induced degradations. Also, if we transmit a sinc
pulse instead of a rectangular pulse, then the ISI can be reduced to zero. This is
known as Nyquist Pulse Shaping.
2.9 Pulse Code Modulation (PCM)

Pulse Code Modulation (PCM) refers to the class of baseband signals
obtained from the quantized PAM signals by encoding each quantized sample into a
digital word. The essential operations in the transmitter of a PCM system are
sampling, quantizing and encoding as shown in the Figure 2.13.
Continuous PCM wave

time Low Pass
Sampler Quantizer Encoder
Filter
Message
signal
Figure 2.13 PCM transmitter
56
The source information is sampled and quantized to one of ‘L’ levels. Then
each quantized sample is digitally encoded into an l-bit codeword, where l=log 2 𝐿.
For baseband transmission, the codeword bits will then be transformed to pulse
waveforms. The essential features of binary PCM are shown in the Figure 2.14.
Figure 2.14 (a) Quantization Levels
The Figure 2.14(a) illustrates an L-level linear quantizer for an analog signal
with a peak-to peak voltage range of Vpp=Vp-(-Vp) = 2 Vp volts. The quantized pulses
assume positive and negative values. The stepsize between quantization levels,
called the quantile interval, is denoted by q volts. When the quantization levels are
uniformly distributed over the full range, the quantizer is called a uniform or linear
quantizer. Each sample value of the analog waveform is approximated with a
quantized pulse. The degradation of the signal due to quantization is therefore
𝑞
limited to half a quanitle interval, ± volts.
2
Figure 2.14(b) shows an analog signal x(t) limited in its excursions to the
range -4 to +4V. The stepsize between quantization levels has been set at 1V.
Thus, eight quantization levels are employed. These are located at -3.5,
-2.5,……+3.5V. Assign the code number 0 to the level at -3.5V, code number 1 to
the level at -2.5V, and so on, until the level at 3.5V, which is assigned the code
number 7.
57
Figure 2.14 (b) Pulse Code Modulation
Each code number has its representation in binary arithmetic, ranging from
000 for code number 0 to 111 for code number 7. The quantile intervals between the
levels should be equal. The ordinate in Figure 2.14(b) is labeled with quantization
levels and their code numbers. Each sample of the analog signal is assigned to the
quantization level closest to the value of the sample. There are four representations
of x(t) as follows: the natural sample values, the quantized sample values, the code
numbers, and the PCM sequence.
Here, each sample is assigned to one of eight levels or a three-bit PCM
sequence. Increasing the number of levels will reduce the quantization noise. If we
double the number of levels to 16, each analog sample will be represented as a four-
bit PCM sequence. But when there are more bits per sample, the data rate is
increased, and the cost is a greater transmission bandwidth. Thus, we can obtain
better fidelity at the cost of more transmission bandwidth.
2.10 UNIFORM AND NON-UNIFORM QUANTIZATION

In pulse code modulation both the parameters time and amplitude are
expressed in discrete form. The sampling process converts the continuous time
58
values of the analog signal into discrete time values. The quantization process
converts the continuous amplitude values into a finite (discrete) set of allowable
values. This process is called “discretization” in time and amplitude. Here, we shall
study about the quantization process. Basically, quantization process may be
classified as follows:
Quantization
Uniform Non Uniform

Quantization Quantization
Midtread type Midrise type
2.10.1 Uniform quantization

When the quantization levels are uniformly distributed over the full amplitude
range of the input signal, the quantizer is called an uniform or linear quantizer. In
uniform quantization, the stepsize between quantization levels remains the same
throughout the input range. The quantizer characteristic can also be midtread or
midrise type, as shown in the Figure 2.15.
a) Midtread Type b) Midrise Type
Figure 2.15 Two types of Uniform Quantization
59
(a) For the uniform quantizer of midtread type, the origin lies in the middle of a
tread of the staircase like graph.
(b) For the Uniform quantizer of midrise type, the origin lies in the middle of a
rising part of the staircase like graph.
Both the midtread and midrise types of uniform quantizers are symmertric
about the origin. Hence they are also called as symmetric quantizer.
2.10.2 Non-uniform quantization

If the quantizer characteristic is nonlinear, then the quantization is known as
non-uniform quantization. In non-uniform quantization, the step size is not constant.
The step size is variable, depending on the amplitude of input signal.
2.10.2.1 Companding
The non-uniform quantization is practically achieved through a process called
companding. Figure 2.16 shows a companding model. The compressor amplifies
weak signals and attenuates strong signals.
Input Output
Uniform
Compressor Expander
quantizer
Figure 2.16 A companding model
This process is called compression. At the receiver, the expander does the
opposite function of compression. Thus the expander provides expansion.
Therefore, the compression of the signal at the transmitter and the expansion at the
receiver is combined to be called as companding.
Companding = Compressing + Expanding
The non-uniform quantizer characteristic is shown in the figure 2.17
60
(d)
Figure 2.17 Non uniform Quantizer Characteristic
2.10.2.2 Companding Characteristics

We need linear compressor characteristics for small amplitudes of the input
signal and a logarithmic characteristic elsewhere. In practice, this is achieved by
using following two methods
(i) -law companding and (ii) A-law companding
The figure 2.18 shows the compression characteristics.
a) -law b) A-law
Figure 2.18 Compression Characteristics
(i) In North America, a -law compression characteristic is used. It is expressed

mathematically as
|𝑥|⁄
loge [1+μ( xmax )]
Y = Ymax loge (1+μ)
𝑆𝑔𝑛 𝑥 (2.15)
61
In telephony applications, these PCM waveforms are often called as Line

Codes. When pulse modulation is applied to a non-binary symbol, the resulting
waveform is called M-ary pulse modulation waveform.
Several types of PCM
waveforms are illustrated in
Figure 2.21.
The PCM waveforms can
be classified into the following
four groups.
1. Non return to Zero (NRZ)
2. Return to Zero (RZ)
3. Phase encoded
4. Multilevel binary
2.12.1 Non-Return to Zero

(NRZ)
The NRZ group is
probably the most commonly
used PCM waveform. If the
waveform stays at any non-zero
level for the whole bit interval T,
then it is called Non Return to
Zero (NRZ) waveform. It can be
subdivided into the following
subgroups, NRZ – L (L for
level), NRZ – M (M for Mark)
and NRZ – S (S for Space).
Figure 2.21 Various PCM Waveforms
1. NRZ – L is used
extensively in digital logic circuits. A binary one is represented by one voltage
level and a binary zero is represented by another voltage level. There is a
change in level whenever the data change from a one to a zero or from a zero
to a one.
65
2. With NRZ – M, the one, or mark, is represented by a change in level, and the
zero, or space, is represented by no change in level. This is often referred to
as differential encoding. NRZ – M is used primarily in magnetic tape
recording.
3. NRZ – S is the complement of NRZ – M. A one is represented by no change
in level, and a zero is represented by a change in level.
2.12.2 Return to Zero (RZ)

If the waveform comes back to zero level after a portion of bit interval T, then
it is called RZ waveform. RZ group can be subdivided into the following subgroups;
Unipolor RZ, Bipolor RZ, and RZ-AMI.
These codes find application in baseband data transmission and in magnetic
recording.
1) With unipolar RZ, a one is presented by a half-bit-wide pulse, and a zero is
represented by the absence of a pulse.
2) With bipolar-RZ, the ones and zeros are represented by opposite level pulses
that are one-half bit wide. There is a pulse present in each bit interval.
3) RZ-AMI (AMI for “Alternate Mark Inversion”) is a signaling scheme used in
telephone systems. The ones are represented by equal amplitude alternating
pulses. The zeros are represented by the absence of pulses.
2.12.3 Phase encoded

In phase encoded scheme, the time position of the occurrence or transition of
a pulse waveform is utilized to distinguish between different logic levels. The phase
encoded group consists of bi--L (bi-phase-level), better known as Manchester
coding; bi--M (bi-phase-Mark); bi--S (bi-phase-space); and delay modulation (DM)
or Miller coding. The phase-encoding schemes are used in magnetic recording
systems and optical communications and in some satellite telemetry links.
1) With bi--L, a one is represented by a half-bit –wide pulse positioned during
the first half of the bit interval. A zero is represented by a half-bit-wide pulse
positioned during the second half of the bit interval.
2) With bi--M, a transition occurs at the beginning of every bit interval. A one is
represented by a second transition one-half bit interval later. A zero is
represented by no second transition.
66
3) With bi--S, a transition also occurs at the beginning of every bit interval. A
one is represented by no second transition. A zero is represented by a
second transition one-half bit interval later.
4) With delay modulation, a one is represented by a transition at the midpoint of
the bit interval. A zero is represented by no transition, unless it is followed by
another zero. In this case, a transition is placed at the end of the bit interval
of the first zero.
2.12.4 Multilevel Binary

The binary waveforms which use three levels to encode the binary data,
instead of two levels, are referred as multilevel binary waveforms. Bipolar RZ and
RZ-AMI schemes belong to this group. This group also contains formats called
dicode and duobinary.
1) With dicode NRZ, the one-to-zero, or zero-to-one data transition changes the
pulse polarity. Without a data transition, the zero level is sent.
2) With dicode-RZ, the one-to-zero, or zero-to-one transition produces a half
duration polarity change. Otherwise, a zero level is sent.
3) Duobinary signalling: Duobinary signalling is also referred to as correlative
coding and partial response signaling. This technique introduced some
controlled amount of ISI into the data stream. This improves bandwidth
efficiency at the expense of an increase in power.
2.13 SELECTION OF A PCM WAVEFORM

There are a variety of PCM waveform formats to represent binary digits. We
now consider the selection of a particular waveform format for the transmission of
baseband signals through the channel. In choosing a PCM waveform for a particular
application, some of the parameters to be considered are given below.
1) DC component: Eliminating the dc energy from the signal’s power spectrum
enables the system to be ac coupled.
2) Self Clocking (Self Synchronization): Symbol or bit synchronization is
required for any digital communication system. The Manchester code has a
transition in the middle of every bit interval whether a one or a zero is being
sent. This guaranteed transition provides a clocking signal.
3) Error detection: Duobinary signaling scheme provides the means of
detecting data errors without introducing additional error detection bits into the
data sequence.
67
Baseband Coding Techniques
3.8.4 Cyclic Redundancy Check Code (CRC)

Cyclic codes are extremely well-suited for error detection. Because they can
be designed to detect many combinations of likely errors. Also, the implementation
of both encoding and error detecting circuits is practical. For these reasons, all the
error detecting codes used in practice are of cyclic code type. Cyclic Redundancy
Check (CRC) code is the most important cyclic code used for error detection in data
networks & storage systems. CRC code is basically a systematic form of cyclic code.
CRC Generation (encoder)

The CRC generation procedure is shown in the figure 3.8.
• First we append a string of ‘q’ number of 0s to the data sequence. For
example, to generate CRC-6 code, we append 6 number of 0s to the data.
• We select a generator polynomial of (q+1) bits long to act as a divisor. The
generator polynomials of three CRC codes have become international
standards. They are
(i) CRC – 12 code : p12 + p11 + p3 + p2 + p + 1
(ii) CRC – 16 code : p16 + p15 + p2 + 1
(iii) CRC – CCITT Code : p16 + p12 + p5 + 1
 ‘k’ bits  ‘q’ bits 
Data Sequence 00......0
(q+1) bits Divisor
Remainder
‘q’ bits CRC

Codeword X
Data Sequence CRC
 ‘k’ bits  ‘q’ bits 
Figure 3.8 CRC Generation

• We divide the data sequence appended with 0s by the divisor. This is a
binary division.
• The remainder obtained after the division is the ‘q’ bit CRC. Then, this ‘q’ bit
CRC is appended to the data sequence. Actually CRC is a sequence of
redundant bits.
• The code word generated is now transmitted.
112
CRC checker
The CRC checking procedure is shown in the figure 3.9
• The same generator polynomial (divisor) used at the transmitter is also used
at the receiver.
Received codeword
Data Sequence CRC
Data Sequence CRC
(q+1) bits Divisor
If remainder is 0, then
Remainder there is no error
Figure 3.9 CRC checker
• We divide the received code word by the divisor. This is also a binary
division.
• If the remainder is all 0s, then there are no errors in the received codeword,
and hence must be accepted.
• If we have a non-zero remainder, then we infer that error has occurred in the
received code word. Then this received code word is rejected by the receiver
and an ARQ signalling is done to the transmitter.
Example 3.5
Generate the CRC code for the data word of 1 1 1 0. The divisor polynomial
is p3 + p + 1
Solution
Data Word (Message bits) = 1110
Generator Polynomial (divisor) = p3 + p + 1
Divisor in binary form = 10 11
The divisor will be of (q + 1) bits long.
Here the divisor is of 4 bits long.
Hence q = 3. We append three 0s to the data word.
113
Now the data sequence is 1 1 1 0 0 0 0. We divide this data by the divisor of

1 0 1 1. Binary division is followed.
1 1 0 0
1 01 1 1 1 1 0 0 0 0
1 0 1 1
0 1 0 1 0
1 0 1 1
0 0 0 1 0 0
Remainder
The remainder obtained from division is 100. Then the transmitted codeword
is 1 1 1 0 1 0 0.
Example 3.6
A codeword is received as 1 1 1 0 1 0 0. The generator (divisor) polynomial is
p3 + p + 1. Check whether there is error in the received codeword.
Solution
Received Codeword = 1110100
Divisor in binary form = 1011
We divide the received codeword by the divisor.
1 1 0 0
1 01 1 1 1 1 0 1 0 0
1 0 1 1
0 1 0 1 1
1 0 1 1
0 0 0 0 0 0
Remainder
The remainder obtained from division is zero. Hence there is no error in the
received codeword.
114
10. What are the advantages and disadvantages of using error detection with
retransmission method or ARQ system?
Advantages
• This method has lower probability of error.
• Selective repeat ARQ provides the best throughput efficiency.
• It is an adaptive method, since information is retransmitted only when errors
occur.
Disadvantages
• The ARQ system is slow, because of large overall delay.
• Expansive input and output buffers are required.
• The implementation cost is high.
11. List the error detection codes and error correction codes.
I. Error detection Codes
1. Constant ratio Codes
2. Redundant Codes
3. Parity check Codes
4. Cyclic Redundancy Check (CRC) Codes
II. Error Correction Codes
A. Linear Block Codes
1. Hamming Codes
2. Cyclic Codes
3. Bose-Chauduri-Hocguenghem (BCH) Codes
4. Reed-Solomon (RS) Codes
B. Convolutional Codes
1. Self Orthogonal Codes
2. Trial and Error Codes
3. Recursive Systematic Codes
12. State the types of errors

There are mainly two types of errors introduced during data transmission.
1. Random Error: Random errors are caused by Additive White Gaussian
Noise (AWGN) in the channel. Here noise affects the transmitted symbols
independently.
128
2. Burst Error: Burst errors are caused by impulse noise in the channel.
Impulse noise affects several consecutive bits and errors tend to occur in
clusters.
There is a possibility that both the Gaussian noise and impulse noise will
affect the channel. Therefore, if there is a mixture of random and burst errors, then
such errors are called as compound errors.
13. Mention some applications of error control coding techniques

• For AWGN Channels, forward error correction (FEC) codes are employed.
Typical applications include line-of-sight radio links such as satellite and deep
space communication links.
• For compound-error channels, Automatic Repeat Request (ARQ) methods
are employed. Typical applications include telephone channels and radio
channels.
• Block codes are widely used to provide error control for magnetic tapes, mass
storage systems, magnetic disks, and other data storage systems.
• Trellis-coded Modulation (TCM) technique combines convolutional coding and
modulation into a single function. TCM is applied in the new generation of
modems being developed for telephone channel.
14. Define Code Rate.

The code rate ‘r’ is defined as the ratio of the message bits (k) and the
encoder output (codeword) bits (n).
𝑚𝑒𝑠𝑠𝑎𝑔𝑒 𝑏𝑖𝑡𝑠 𝑘
Code rate, r = =
𝑇𝑟𝑎𝑛𝑠𝑚𝑖𝑡𝑡𝑒𝑑 𝑐𝑜𝑑𝑒𝑤𝑜𝑟𝑑 𝑏𝑖𝑡𝑠 𝑛
where 0 < r < 1
15. What is hamming distance?

The hamming distance (d) between the two code vectors is equal to the
number of elements in which they differ. Eg, Let X = 101 and Y = 110. Then
hamming distance between X and Y code vectors is 2.
The smallest hamming distance between the valid codevectors is termed as
the minimum hamming distance (dmin).
16. Define codeword and codevector.

The encoded block of ‘n’ bits is called a codeword. It contains ‘k’ message
bits and ‘q’ check bits.
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An ‘n’ bit code word can be visualized in an N-dimensional space as a vector

whose elements or co-ordinates are the bits in the codeword.
17. What are linear and non-linear codes?

In a linear code, modulo-2 sum of any two codevectors produces another
valid code vector. The codes used in practical applications are almost always linear
codes.
In a non-linear code, modulo-2-sum of any two codevectors does not
necessarily produces another valid code vector.
18. What are systematic codes and non-systematic codes?

In a systematic code, the check bits are added in such a way that the
message bits appear first and then check bits.
In a non-systematic code, it is not possible to identify message bits and check
bits. They are mixed in the block.
19. What are linear block codes?

The input binary data sequence is divided into block of ‘k’ message bits. For
each block of ‘k’ message bits, (n-k) check bits are added to produce ‘n’ bits
codeword. Such codes are called (n, k) block codes.
Message bits Check bits

k q
 ‘n’ bits code word 
(n = k + q)
If the block codes satisfy linearity property, then they are called as linear block
codes.
20. What are Hamming Codes?

Hamming codes are (n, k) linear block codes. They can be generated either
systematically or non-systematically. The systematic form of Hamming codes satisfy
the following conditions.
• Number of check bits, q  3
• Codeword length, n = 2q – 1
• Number of message bits, k = n – q
• Minimum hamming distance, dmin = 3
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21. State the error detection and correction capability of linear block code or
Hamming code.
• Detect upto ‘s’ errors per codeword, dmin  s+1.
• Correct upto ‘t’ error per codeword, dmin  2t + 1.
• For Hamming code, dmin = 3, s = 2, t = 1
22. What is retransmission?

At the receiver, the channel decoder decodes the received codewords and
look for errors. If no error is detected, the decoder sends a positive
acknowledgement (ACK) through the return transmission channel. If any error is
detected, it discards that part of the data sequence and sends a negative
acknowledgement (NAK). The transmitter then once again transmits that part of the
codeword sequence in which error was detected. This process is called
retransmission.
23. What are binary cyclic codes?

Binary cyclic codes are a subclass of the linear block codes. A linear block
code is called as cyclic code if every cyclic shift of the code vector produces another
code vector. A cyclic code exhibits both the linearity property and cyclic property.
24. What are the advantages and disadvantages of cyclic codes

Advantages
• Cyclic codes can correct burst errors that span many successive bits.
• They have an excellent mathematical structure. This makes the design of
error correcting codes with multiple-error correction capability relatively easier.
• The encoding and decoding circuits can be easily implemented using shift
registers.
• The error correcting and decoding methods eliminate the storage (large
memories) needed for lookup table decoding. Therefore the cyclic codes
become powerful and efficient.
Disadvantages
• Even though the error detection is simpler, the error correction is slightly more
complicated. This is due to the complexity of the combinational logic circuit
used for error correction.
25. Define Cyclic Redundancy Check (CRC) code.

Cyclic Redundancy Check (CRC) code is an most important cyclic code used
for error detection in data networks and storage systems. CRC code is basically a
systematic form of cyclic code.
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Digital Modulation Techniques
Unit – IV DIGITAL MODULATION TECHNIQUES
OBJECTIVES
• To know the Digital Modulation techniques
• To study about Coherent and Non-Coherent modulation schemes
• To learn about TDM frame structure
• To study about Coherent and Non-Coherent detection schemes
4.0 INTRODUCTION
We have discussed Baseband pulse transmission in Unit II. In baseband
pulse transmission, the input data is represented in the form of a discrete PAM signal
(Line codes). The baseband signals have an adequately large power at low
frequencies. So they can be transmitted over a pair of wires or coaxial cables.
But, it is not possible to transmit the baseband signals over radio links or
satellites, since impractically large antennas would be required. Hence, the spectrum
of the message signal has to be shifted to higher frequencies. This is achieved by
using the baseband digital signal to modulate a high frequency sinusoidal carrier.
The modulated signals are transmitted over a band pass channel, such as
microwave radio link, satellite channel, optical fibre link etc. This process is called as
digital carrier modulation or digital passband communication.
4.1 DIGITAL MODULATION:

Digital modulation may be defined as mapping a sequence of input binary
digits into a set of corresponding high frequency signal waveforms. These modulated
waveforms may differ in either amplitude or frequency or phase or some combination
of two signal parameters (Amplitude and phase or frequency and phase).
4.1.1 Digital Modulation Techniques

The digital modulation techniques may be classified into two categories.
1. Coherent digital modulation techniques
2. Non-Coherent digital modulation techniques
1. Coherent Digital Modulation Techniques

Coherent digital modulation techniques employ coherent detection. In
coherent detection, the local carrier generated at the receiver is phase locked with
the carrier at the transmitter. Thus detection is done by correlating received noisy
133
signal and locally generated carrier. The coherent detection is a synchronous

detection. Coherent detection techniques are complex but provide better
performance.
2. Non-Coherent Digital Modulation Techniques

These techniques employ Non-Coherent detection. The detection process
does not need receiver carrier to be phase locked with the transmitter carrier. Non-
Coherent detection techniques are less complex. However the probability of error is
high compared to Coherent detection.
4.1.2 Listing of various types:

Based on the mapping techniques, we can broadly classify the digital
modulation methods.
I. Binary Scheme / M-ary Scheme:

In binary scheme, we send any one of the two possible signals during each
signaling interval of duration Tb. Examples are
1. Amplitude Shift Keying (ASK), 2. Frequency Shift Keying (FSK) and
3. Phase Shift Keying (PSK)
M-ary Scheme:
In M-ary scheme, we can send any one of the M possible signals during each
signaling interval of duration Tb. Examples are
1. M-ary ASK
2. M-ary FSK
3. M-ary PSK
4. Minimum shift keying (MSK) is a special form of continuous phase frequency

shift keying (CPFSK).
5. Quadriphase shift keying (QPSK) is an example of M-ary PSK with M=4. Both
MSK and QPSK are examples of quadrature carrier multiplexing system.
6. M-ary Quadrature Amplitude Modulation (M-ary QAM)
We may combine discrete changes in both the amplitude and phase of a

carrier to produce M-ary Amplitude-Phase Keying (APK). M-ary QAM is a special
form of this hybrid modulation.
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II. Based on the performance of the modulation scheme and properties of

modulated signal.
1. Power efficient scheme / Bandwidth efficient scheme
2. Continuous phase (CP) modulation / In phase Quadrature phase (IQ)
modulation
3. Constant envelope modulation / Non-Constant envelope modulation
4. Linear modulation / Non-linear modulation
5. Modulation scheme with memory / modulation scheme without memory.
4.1.3 Design Goals of Digital Communication System

There are so many modulation/detection schemes available to the designer of
a digital communication system. Each scheme offers system trade-offs of its own.
The selection of a particular modulation/detection scheme is determined by the
usage of available primary communication resources, transmitted power and channel
bandwidth. In particular, the choice is based on achieving as many of the following
design goals as possible.
1. Maximum data rate
2. Minimum possibility of symbol error
3. Minimum transmitted power
4. Minimum channel bandwidth
5. Maximum resistance to interfering signals
6. Minimum circuit complexity.
4.1.4 Gram-Schmidt Orthogonalization Procedure

The task of transforming an incoming message m i, where i = 1, 2, …. M, into a
modulated wave Si(t) may be divided into separate discrete time and continuous time
operations. The Gram-Schmidt orthogonalization procedure permits the
representation of any set of M energy signals, {𝑆𝑖 (𝑡)}, as linear combinations of N
orthonormal basis functions. Hence we may represent the given set of real-valued
energy signals Si(t), S2(t)….. Sm(t), each of duration T seconds, in the form
𝑖 = 1,2, … . 𝑀
Si (t) = 𝑗=1 𝑆𝑖𝑗 𝑗 (𝑡),
∑𝑁 (4.1)
𝑜 ≤𝑡 ≤𝑇
The real valued basis functions 1(t), 2(t), ……N(t) are orthonormal. Hence
we have
𝑇 1 𝑖𝑓 𝑖 = 𝑗
∫0 𝑖 (𝑡)𝑗 (𝑡)𝑑𝑡 = { (4.2)
0 𝑖𝑓 𝑖 ≠ 𝑗
The first condition states that each basis function is normalized to have unit
energy. The second condition states that the basis functions 1(t), 2(t), ……N(t) are
orthogonal with respect to each other over the interval 0≤ 𝑡 ≤ 𝑇.
135
In equation (4.1), the coefficients of the expansion are defined by

𝑇 𝑖 = 1, 2, … . 𝑀
Sij = ∫0 𝑆𝑖 (𝑡)𝑗 (𝑡)𝑑𝑡 𝑗 = 1, 2, … . 𝑁 (4.3)
Modulator Design:
Let the set of coefficients {Sij}, j = 1, 2, … N operating as input. Then we may
use the scheme shown in Figure 4.1 to generate the signal S i(t), i = 1, 2, … M as per
equation (4.1).
Multiplier
Si1
1(t)
Si2
2(t)  Si(t)
:
summer
:
SiN
N(t)
Figure 4.1 Scheme for generating the signal Si (t)
It consists of a bank of N multipliers, with each multiplier supplied with its own
basis function, followed by a summer. This scheme is performing a similar role to
that of modulator in the transmitter.
Detector Design:
Correlator
Integrator
Multiplier
𝑇
∫ 𝑑𝑡 Si1
0
1(t)
𝑇
S1(t) ∫ 𝑑𝑡 Si2
0
2(t)
𝑇
∫ 𝑑𝑡 SiN
0
N(t)
Figure 4.2 Scheme for generating the set of coefficients {Sij}
136
Let the set of signals {𝑆𝑖 (𝑡)}, 𝑖 = 1, 2, … 𝑀, operating as input. We may use
the scheme shown in figure 4.2 to calculate the set of coefficients {𝑆𝑖𝑗 }, j = 1, 2, ….N
as per equation (4.3). This scheme consists of a bank of N product integrators or
correlators with a common input. Each multiplier is supplied with its own basis
function. This scheme is performing a similar role to that of detector in the receiver.
4.2 COHERENT BINARY MODULATION TECHNIQUES

We know that binary modulation scheme has three basic forms.
1. Binary Amplitude Shift Keying (BASK)
2. Binary Frequency Shift Keying (BFSK)
3. Binary Phase Shift Keying (BPSK)
When these modulation schemes employ coherent detection at the receiver,

then they are called as coherent binary modulation techniques. In coherent
detection, the local carrier generated at the receiver is phase locked with the carrier
at the transmitter.
4.2.1 Coherent Binary Phase Shift Keying (BPSK)

In phase shift keying, the modulation process involves switching or keying the
phase of the carrier signal in accordance with the incoming data. The Figure 4.3(a)
shows the block diagram for binary PSK transmitter.
Input Binary Product Binary PSK

Sequence Modulator Wave
1(t)
Figure 4.3(a) Binary PSK transmitter
137
Input Binary
Sequence
Mark
Switch
1(t)
Oscillator
Inverter Summing Modulated
Amplifier PSK signal
180o Phase
Shift
Space
Switch
Figure:4.3 (b) Binary PSK modulator
In a coherent binary PSK system, the pair of signals, S1(t) and S2(t) are used
to represent binary symbols 1 and 0 respectively. They are defined by
2𝐸
S1(t) = √ 𝑇 𝑏 𝑐𝑜𝑠(2𝜋𝑓𝑐 𝑡) (4.4)
𝑏
2𝐸
S2(t) = √ 𝑇 𝑏 𝑐𝑜𝑠(2𝜋𝑓𝑐 𝑡 + 𝜋)
𝑏
2𝐸
= −√ 𝑇 𝑏 𝑐𝑜𝑠(2𝜋𝑓𝑐 𝑡) (4.5)
𝑏
where Eb transmitted signal energy per bit

Tb bit duration, 0 ≤ t ≤ Tb
We require only one basis function of unit energy.
2
1(t) = √𝑇 𝑐𝑜𝑠(2𝜋𝑓𝑐 𝑡), 0 ≤ t ≤ Tb (4.6)
𝑏
We have to represent the input binary sequence in polar form with symbols 1
and 0 by constant amplitude levels of √𝐸𝑏 and √−𝐸𝑏 , respectively. This binary wave
and a sinusoidal carrier 1 (t) are applied to a product modulator. The desired PSK
wave is obtained at the modulator output. An alternate method of generating binary
PSK is shown in Figure 4.3(b). In this method we use two balanced modulators as
mark and space switch. The input binary data is applied directly to mark switch and
after inverting to the space switch. The carrier signal 1 (t) is fed directly to mark
switch and 180° phase shifted to space switch. For binary input 1, the mark switch is
138
closed and PSK wave is generated. For binary input 0, the space switch is closed
and PSK wave is generated. The summing amplifier combines the output from mark
and space switches.
Wave forms:
The Figure 4.4 shows the waveforms for coherent binary PSK modulation.
Figure 4.4 - Waveforms for BPSK
Merits of BPSK:
• BPSK requires lower bandwidth than BFSK
• BPSK has the minimum value of probability of error. Hence it provides best
performance compared to BFSK and BASK schemes.
• It has very good noise immunity.
Demerits of BPSK:
In PSK, the information lies in the phase, and hence, it cannot be detected
non-coherently.
4.2.2 Coherent Binary Frequency Shift Keying (BFSK):

In Frequency shift keying, the modulation process involves switching or
keying the frequency of the carrier signal in accordance with the incoming data. The
Figure 4.5 shows the block diagram of binary FSK transmitter.
139
1(t)
Summing Binary FSK
Input binary 
amplifier wave
Sequence
Inverter
2(t)
Figure 4.5 (a) BFSK Modulator
Input Binary Mark

Sequence Switch
Mark
1(t) Oscillator
Inverter Summing Modulated
Amplifier FSK signal
Space
2(t) Oscillator
Space
Switch
Figure 4.5 (b) BFSK Modulator
In a coherent binary FSK system, the pair of signals, S 1(t) and S2(t) are used
2𝐸𝑏
S1(t) = √𝑇 𝑐𝑜𝑠(2𝜋𝑓1 𝑡) (4.7)
𝑏
2𝐸𝑏
S2(t) = √𝑇 𝑐𝑜𝑠(2𝜋𝑓2 𝑡) (4.8)
𝑏
Here we require two basic functions of unit energy.

2
1(t) = √𝑇 𝑐𝑜𝑠(2𝜋𝑓1 𝑡) (4.9)
𝑏
2
2(t) = √𝑇 𝑐𝑜𝑠(2𝜋𝑓2 𝑡) (4.10)
𝑏
140
Here 1(t) is applied to the upper product modulator (Also referred as Mark
Switch). 2(t) is applied to lower product modulator (Also referred as space switch).
The input binary data is applied directly to the mark switch and through an inverter to
the space switch. For binary input 1, the mark switch is closed and FSK wave S 1(t)
is generated. For binary input 0, the space switch is closed and FSK wave S2(t) is
generated. The summing amplifier combines the output from Mark and Space
switches. In BFSK, the frequency of the modulated wave is shifted with a continuous
phase, in accordance with the input binary wave. Hence phase continuity is always
maintained including the inter-bit switching times. Therefore BFSK is also referred
as continuous phase frequency shift keying (CPFSK).
Waveforms:
The figure 4.6 shows the waveforms for coherent binary FSK modulation.
Figure 4.6 Waveforms for BFSK
Merits of BFSK
• It is relatively easy to implement.
• It has better noise immunity than ASK.
Demerits of BFSK
• BFSK requires high bandwidth compared to BPSK and BASK.
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4.2.3 Coherent Binary Amplitude Shift Keying (BASK)

In Amplitude shift keying, the modulation process involves switching or keying
the amplitude of the carrier signal in accordance with the incoming data. The Figure
4.7 shows the block diagram of binary ASK transmitter.
Input Binary Digitally Modulated

Sequence Controlled ASK output
Switch
1(t)
Figure 4.7 BASK Modulator
In a coherent binary ASK system, the pair of signals S 1(t) and S2(t) are used
2𝐸
S1(t) = √ 𝑇 𝑏 𝑐𝑜𝑠(2𝜋𝑓𝑐 𝑡) (4.11)
𝑏
S2(t) = 0 (4.12)
We require only one basis function of unit energy.
2
1(t) = √𝑇 𝑐𝑜𝑠(2𝜋𝑓𝑐 𝑡) (4.13)
𝑏
The binary wave and the sinusoidal carrier 1 (t) are applied to a product
modulator. The product modulator acts like a digitally controlled switch. For binary
input 1, the switch is closed and the carrier signal 1 (t) is obtained as output signal.
For binary input 0, the switch is open and hence there is no output signal. The
resulting output will be the ASK waveform. The modulator simply does the on-off
function. Hence BASK is also called as On-Off Keying (OOK).
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Wave forms:
The Figure 4.8 shows the wave forms for coherent binary ASK modulation.
Figure 4.8: Wave forms for BASK
Merits of BASK:
• BASK is easy to generate and detect
Demerits of BASK:
• Bask is very sensitive to noise
4.2.4 Performance Comparison:

Table 4.1 shows the performance comparison of three basic digital
modulation techniques.
Sl.
Parameters BASK BFSK BPSK
No.
1. Switching or keying of Amplitude Frequency Phase
2. Bandwidth 2fb 4fb 2fb
3. Noise immunity Low High High
4. Probability of error High Low Low
5. Performance in presence Poor Better than Best of three
of noise ASK schemes
6. System complexity Simple Moderately Very
complex Complex
7. Bit rate or data rate Suitable upto Suitable upto Suitable upto
100 bits / sec 1200 bits / sec high bit rates
8. Demodulation method Envelope Envelope Coherent
detection detection detection
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4.3 NON-COHERENT BINARY MODULATION TECHNIQUES:

The modulation scheme in which the detection process does not need
receiver carrier to be phase locked with the transmitter carrier is said to be Non-
Coherent modulation technique. The Non-Coherent binary modulation techniques
are
1. Differential Phase Shift Keying (DPSK)
3. Binary Frequency Shift Keying (BFSK).
For BASK and BFSK, the modulator sections are the same for both coherent
and non-coherent modulation techniques. We have already explained the
modulator sections of BASK and BFSK. Now we shall see about Differential PSK.
The non-coherent binary FSK and DPSK schemes are treated as special
cases of non-coherent orthogonal modulation.
Differential Phase Shift Keying (DPSK):

In the binary phase shift keying, we cannot have “Non-Coherent PSK”,
because detection without phase information is not possible. Hence, there is a
“Pseudo PSK” technique called Differential Phase Shift Keying (DPSK). DPSK may
be viewed as the non- coherent form of PSK.
DPSK eliminates the need for a coherent reference signal at the receiver by
combining two basic operations at the transmitter.
1. Differential encoding of the input binary wave
2. Phase shift keying
To send symbol 1, we leave the phase of the current signal waveform
unchanged. To send symbol 0, we phase advance the current signal waveform by
180°. The receiver is equipped with a storage capability, so that it can measure the
relative phase difference between the waveforms received during two successive bit
intervals. The Figure 4.8 shows the block diagram of a DPSK transmitter.
{dk}
Input Binary Logic Amplitude Product DPSK
Sequence Network level shifter Modulator Signal
{bk}
{dk-1}
Delay 1(t)
Tb
Figure 4.8 DPSK transmitter
144
It consists of a logic network and a one-bit delay element interconnected so

as to convert the raw binary sequence {bk} into a differentially encoded sequence
{dk}. This sequence is amplitude level encoded and then used to modulate a carrier
2
wave [1 (𝑡) = √ 𝑐𝑜𝑠(2𝜋𝑓𝑐 𝑡)], thereby producing the desired DPSK signal.
𝑇 𝑏
The differential encoding process starts with an arbitrary first bit, serving as
reference. Let {dk} denote the differentially encoded sequence with this added
reference bit.
(i) If the incoming binary symbol bk is 1, leave the symbol dk unchanged with
respect to the previous bit.
(ii) If the incoming binary symbol bk is 0, change the symbol dk with respect to
the previous bit.
The differentially encoded sequence {dk} thus generated is used to phase-shift
a carrier with phase angles 0 and 𝜋 radians representing symbols 1 and 0,
respectively. Table 4.2 illustrates the differential phase encoding process. Here, d k
is the complement of the modulo-2 sum of bk and dk-1.
Table 4.2 Illustrating the generation of DPSK signal
{bk} 1 0 0 1 0 0 1 1
{dk-1} 1 1 0 1 1 0 1 1
Differentially encoded 1 1 0 1 1 0 1 1 1
sequence {dk}
Transmitted phase (radians) 0 0  0 0  0 0 0
Merits of DPSK:
• DPSK scheme does not need carrier at the receiver end. Hence it has
reduced system complexity.
• The bandwidth required is less than that required for BPSK.
Demerits of DPSK:
• It has higher value of probability of error than that of BPSK.
• Noise interference is more.
• In DPSK, previous bit is used to detect next bit. Hence, there is possibility of
errors appearing in pairs.
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4.4 COHERENT QUADRATURE MODULATION TECHNIQUES

One important goal in the design of a digital communication system is the
efficient utilization of channel bandwidth. There are two bandwidth conserving
Quadrature-Modulation schemes for the transmission of binary data.
They are:
1. Quadriphase-Shift Keying (QPSK)
2. Minimum Shift Keying (MSK)
These two schemes are both examples of the quadrature-carrier multiplexing

system. They produce a modulated wave described as
S(t) = SI(t) cos (2fct) – SQ(t) Sin(2fct) (4.14)
where SI(t) is the in-phase component of the modulated wave, and S Q(t) is the
quadrature component. QPSK is a quadrature-carrier signaling technique, which is
an extension of binary PSK. MSK is a special form of continuous Phase Frequency
Shift Keying (CPFSK).
4.4.1 Quadri Phase-Shift Keying (QPSK):

In QPSK, as with binary PSK, information carried by the transmitted signal is
contained in the phase. The mapping or assignment of k information bits to the M=2k
possible phases may be done in a number of ways. The preferred assignment is
Gray encoding. For QPSK, we have k=2, and hence M=2 2=4. Therefore, the
number of bits per symbol is two bits. Then the information bits 1 0, 0 0, 0 1, and 1 1
𝜋 𝜋 𝜋 𝜋
(Gray encoding) represent the phase values 4 , 3 4 , 5 4 and 7 4 (45° , 135° , 225° ,
and 315°). For this set of values we may define the transmitted signal as
2𝐸 𝜋
√ 𝑐𝑜𝑠 [2𝜋𝑓𝑐 𝑡 + (2𝑖 − 1) ] , 0 ≤ 𝑡 ≤ 𝑇
Si(t) = { 𝑇 4 (4.15)
0, 𝑒𝑙𝑠𝑒𝑤ℎ𝑒𝑟𝑒
where i = 1, 2, 3, 4; E is the transmitted signal energy per symbol (E = 2 Eb) and T is

the symbol duration (T = 2Tb).
We can rewrite the equation (4.15) as
2𝐸 𝜋 2𝐸 𝜋
Si(t) =√ 𝑇 cos(2𝜋𝑓𝑐 𝑡) . 𝑐𝑜𝑠 [(2𝑖 − 1) 4 ] − √ 𝑇 sin(2𝜋𝑓𝑐 𝑡) . 𝑠𝑖𝑛 [(2𝑖 − 1) 4 ]
(4.16)
146
Where 0 ≤ t ≤ T
There are two orthonormal basis functions 1(t) and 2(t).
2
1(t) = √ cos(2𝜋𝑓𝑐 𝑡)
𝑇
and (4.17)
2
2(t) = √ sin(2𝜋𝑓𝑐 𝑡)
𝑇
(4.18)
Transmitter
The figure 4.9 shows the block diagram of QPSK transmitter.
a1(t)
1(t) +
Input binary
Polar NRZ  QPSK
data Demultiplexer
level encoder Signal
Sequence +
a2(t)
2(t)
Figure 4.9 QPSK Transmitter
The incoming binary data sequence is first transformed into polar form by a
non-return-to-zero (NRZ) level encoder. This binary wave is next divided by means
of a demultiplexer into two separate binary waves consisting of the odd-and even-
numbered input bits. These two binary waves are denoted by a 1(t) and a2(t).
These two binary waves a1(t) and a2(t) are used to modulate a pair of
quadrature carriers 1(t) and 2(t) respectively. The result is a pair of binary PSK
signals. Finally, the two binary PSK signals are added to produce the desired QPSK
signals.
Wave forms:
The figure 4.10 illustrates the sequences and waveforms involved in the
generation of a QPSK signal.
147
Figure 4.10 QPSK waveforms
Receiver:
The Figure 4.11 shows the block diagram of coherent QPSK receiver.
In-phase Channel Threshold = 0
𝑇 x1
Decision
∫ 𝑑𝑡
0 device
Estimate of
1(t)
Received Multiplexer transmitted
binary
Signal sequence
Quadrature
Channel 𝑇 x2
∫ 𝑑𝑡 Decision
0 device
2(t) Threshold = 0
Figure 4.11 Coherent QPSK receiver
The received signal x(t) is applied to a pair of product integrators or

correlators. The multiplier is supplied with locally generated coherent carrier signals
1(t) in the In-phase channel and 2(t) in the quadrature channel. The correlator
148
outputs of x1 and x2 are produced in response to the received signal x(t). They are
each compared with a threshold of Zero.
For the In phase channel, if x1>0, a decision is made in favour of symbol 1,
and if x1< 0, a decision is made in favour of symbol 0. Similarly, for the quadrature
channel, if x2> 0, a decision is made in favour of symbol 1 and if x2< 0, a decision is
made in favour of symbol 0. Finally, these two binary sequences at the in-phase and
quadrature channel outputs are combined in a multiplexer. This will reproduce the
original binary sequence at the transmitter input. The minimum average probability
𝐸
of symbol error for QPSK is given by Pe = erfc[√𝑁𝑏 ].
𝑜
Signal space diagram

For any modulation scheme, the analysis is based on the signal space
diagram assuming an Additive White Gaussian Noise (AWGN) model. Signal-space
approach is a plotting of possible message points. Such a set of possible message
points is also referred to as a “Signal Constellation”.
In QPSK, there are four message points. The associated signal vectors are
defined by
𝜋
√𝐸 𝑐𝑜𝑠 [(2𝑖 − 1) 4 ]
Si = [ 𝜋
], i = 1, 2, 3, 4 (4.19)
−√𝐸𝑠𝑖𝑛 [(2𝑖 − 1) 4 ]
The elements of the signal vectors, namely, si1 and si2 have their values
shown in Table 4.3.
Table 4.3 Signal Space Characterization of QPSK
Input dibit Phase of QPSK Coordinates of message points

0≤t≤T signal (radians) Si1 Si2
10 𝜋 𝐸 𝐸
4 +√2 -√ 2
𝜋
00 3 𝐸 𝐸
4 -√ -√
2 2
𝜋
01 5 𝐸
4 𝐸 +√
−√ 2
2
𝜋
11 7 𝐸 𝐸
4 +√ +√
2 2
Accordingly, a QPSK signal is characterized by having a two dimensional

signal constellation (ie. N = 2) and four message points (ie., M = 4), as illustrated in
Figure 4.12.
149
Figure 4.12 Signal space diagram for coherent QPSK system
Merits of QPSK:
• QPSK has very good noise immunity.
• More effective utilization of the available bandwidth of the transmission
channel.
• It has low error probability
Demerits of QPSK:
• The generation and detection of QPSK Is complex.
How QPSK is better than BPSK:

• Due to multilevel modulation used in QPSK, it is possible to increase the bit
rate to double the bit rate of BPSK without increasing the bandwidth.
• Available channel bandwidth is utilized in a better way by the QPSK system
than the BPSK system.
• The noise immunity of QPSK is same as that of BPSK system.
4.4.2 Minimum Shift Keying (MSK):

Minimum shift keying (MSK) is a special form of binary CPFSK signal. A
Continuous Phase Frequency Shift Keying (CPFSK) signal with a deviation ratio of
1
h = is referred to as MSK. Using MSK, it is possible to improve the noise
2
performance of the receiver significantly, by the proper use of the phase information.
This improvement is achieved at the expense of increased receiver complexity.
Consider a CPFSK signal, defined for the interval 0 ≤ t ≤ Tb as below:
150
2𝐸
√ 𝑇 𝑏 𝑐𝑜𝑠[2𝜋𝑓1 𝑡 + 𝜃(0)]for symbol 1
𝑏
S(t) = (4.20)
2𝐸𝑏
√ 𝑐𝑜𝑠[2𝜋𝑓2 𝑡 + 𝜃(0)]for symbol 0
{ 𝑇𝑏
where Eb →transmitted signal energy per bit and Tb→bit duration.
The phase (0) denotes the value of phase at time t=0. The frequencies f 1
and f2 are sent in response to binary symbols 1 and 0 appearing at the modulator
input respectively.
Another useful way of expressing the CPFSK signal S(t) is to represent it in
the form of an angle modulated signal as follows:
2𝐸
S(t) = √ 𝑇 𝑏 𝑐𝑜𝑠[2𝜋𝑓𝐶 𝑡 + 𝜃(𝑡)] (4.21)
𝑏
where (t) is the phase of S(t). The phase (t) of a CPFSK signal increases or
decreases linearly with time during each bit duration of T b seconds, as shown by
𝜋ℎ
(t) = (0) ± 𝑡, 0 ≤ t ≤ Tb (4.22)
𝑇𝑏
𝜋ℎ
ie., (t) = (0) + 𝑡  for sending symbol 1
𝑇𝑏
𝜋ℎ
and (t) = (0) - 𝑡  for sending symbol 0
𝑇𝑏
Substituting equation (4.22) into equation (4.21), and then comparing the
angle of the cosine function with that of equation (4.20), we deduce the following pair
of relations:
ℎ
fc + = f1 (4.23)
2𝑇𝑏
ℎ
fc - = f2 (4.24)
2𝑇𝑏
Solving equations (4.23) and (4.24) for fc and h, we get

1
fc = 2
(f1 + f2) (4.25)
and h = 𝑇𝑏 (f1 - f2) (4.26)
The nominal carrier frequency fc is therefore the arithmetic mean of the

frequencies f1 and f2.
Deviation Ratio (h):

The difference between the frequencies f 1 and f2, normalized with respect to
the bit rate 1⁄𝑇 defines the dimensionless parameter h, which is referred to as the
𝑏
deviation ratio.
151
Phase Trellis:
From equation (4.22), for sending symbol 1, we have
𝜋ℎ
𝜃(t) = 𝜃(0) + 𝑇𝑏
t
At time t = Tb
𝜋ℎ
𝜃(Tb) = 𝜃(0) + 𝑇𝑏
. 𝑇𝑏  𝜃(Tb) - 𝜃(0) = h
For sending symbol 0, we have

𝜋ℎ
𝜃(t) = 𝜃(0) - .𝑡
𝑇𝑏
At time t = Tb
𝜋ℎ
𝜃(Tb) = 𝜃(0) - . 𝑇𝑏  𝜃(Tb) - 𝜃(0) = - h
𝑇𝑏
Hence we may write

𝜋ℎ for symbol 1
(Tb) - (0) = { (4.27)
−𝜋ℎ for symbol 0
Therefore, sending of symbol 1 increases the phase of a CPFSK signal S(t)
by 𝜋h radians. Sending of symbol 0 decreases the phase by h radians. We can
plot the variation of phase 𝜃(t) with respect to time t. Such a plot is called as a
phase tree as shown in the Figure 4.13.
Figure 4.13 Phase Tree
152
The plot of phase tree follows a path consisting of a sequence of straight

lines, the slope of which represents frequency changes.
The phase tree described in Figure 4.13 is a manifestation of phase
continuity, which is an inherent characteristic of a CPFSK signal. In BFSK, which is
a CPFSK scheme, the deviation ratio h is exactly unity. Hence the phase change
over one bit interval is ± radians.
1
For MSK scheme, the deviation ratio h is assigned the special value of 2.
𝜋
Then the phase can take on only the two values ± 2 at odd multiples of Tb, and only
the two values 0 and 𝜋 at even multiples of Tb as shown in the Figure 4.14.
Figure 4.14 phase trellis (for the sequence 1101000)
This plot is called as a phase trellis, since a “trellis” is a tree like structure with
remerging branches. Each path from left to right through the trellis corresponds to a
specific binary sequence input. The path shown in boldface in the figure 4.14
corresponds to the binary sequence 1101000 with 𝜃(0) = 0.
Why the name MSK?

From equation (4.26), we have
h = Tb (f1 – f2)
1
On substituting h =
2
1 1
2
= Tb (f1 – f2)  f1 – f2 = 2𝑇𝑏
1
Since bit rate Rb= 𝑇 , we can write
𝑏
𝑅𝑏
f1 – f2 = 2
(4.28)
Hence the frequency deviation (f 1 – f2) equals half the bit rate. This is the
minimum frequency spacing that allows the two FSK signals representing symbols 1
and 0 as in equation 4.20 to be coherently orthogonal ie., they do not interfere with
one another in the process of detection. It is for this reason, a CPFSK signal with a
153
1
deviation ratio of h = 2
is referred to as Minimum shift keying (MSK). MSK is also
referred to as fast FSK.
4.4.2.1 MSK Transmitter

The Figure 4.15 shows the block diagram of MSK transmitter.
Figure 4.15 Block diagram of MSK transmitter communication system
The advantage of this method of generating MSK signals is that the signal
coherence and deviation ratio are largely unaffected by variation in the input data
𝑛
rate. Two input sinusoidal waves, one of frequency f c= 4𝑇𝑐 for some fixed integer nc,
𝑏
1
and the other of frequency 4𝑇 are first applied to a product modulator.
𝑏
This produces two phase-coherent sinusoidal waves at frequencies f1 and f2.

1
They are related to the carrier frequency fc and the bit rate 𝑇 such that
𝑏
ℎ
fc + 2𝑇 = f1
𝑏
ℎ 1
fc - 2𝑇 = f2 , where h = 2
𝑏
These two sinusoidal waves are separated from each other by two narrow
band filters, one centered at f1 and the other at f2. The resulting filter outputs are
linearly combined to produce the pair of quadrature carriers. The orthonormal basis
functions used as quadrature carriers are
2 𝜋
1(t) = √𝑇 cos (2𝑇 𝑡) 𝑐𝑜𝑠(2𝜋𝑓𝑐 𝑡), 0 ≤ t ≤ Tb (4.29)
𝑏 𝑏
2 𝜋
2(t) = √𝑇 sin (2𝑇 𝑡) 𝑠𝑖𝑛(2𝜋𝑓𝑐 𝑡), 0 ≤ t ≤ Tb (4.30)
𝑏 𝑏
Finally, 1(t) and 2(t) are multiplied with two binary waves a1(t) and a2(t)
1
having a bit rate equal to 2𝑇𝑏. The two binary waves a1(t) and a2(t) are extracted
154
from the incoming binary sequence. The two multiplier outputs are summed to get
the MSK signal output. We may express the MSK signal in the form of
s(t) = s11(t) + s22(t), 0 ≤ t ≤ Tb (4.31)
WAVE FORMS:
The Figure 4.16 shows the sequences and waveforms involved in the
generation of MSK signal for the binary sequence 1101000
(a) Input Binary Sequence (c) Function S22(t)
(b) Function S11(t) (d) MSK Signal
Figure 4.16 sequence and waveforms for MSK signal
4.4.2.2 MSK Receiver:

The Figure 4.17 shows the block diagram of coherent MSK receiver.
Figure 4.17 Block diagram of MSK receiver
155
The received signal x(t) is correlated with locally generated replicas of the
coherent reference signals 1(t) and 2(t). In both cases the integration interval is
2Tb seconds. Also, the integration in the quadrature channel is delayed by T b
seconds with respect to that in the in-phase channel.
The resulting in-phase and quadrature channel correlator outputs, x1 and x2,
are each compared with a threshold of zero.
• For the in-phase channel, if x1>0, then choose the phase estimate 𝜃̂(0) = 0. If
x1<0, then choose the estimate 𝜃̂(0)= .
𝜋
• For the quadrature channel, if x2>0, then choose the phase estimate 𝜃̂(Tb)=- .
2
𝜋
If x2<0, then choose the estimate 𝜃̂ (Tb) =
2
Finally, these phase decisions are interleaved so as to reconstruct the original

input binary sequence. The minimum average probability of symbol error in an
𝐸
AWGN channel, for MSK is given by pe=erfc[√ 𝑏⁄𝑁 ]
𝑜
Signal Space Diagram:

The signal constellation for an MSK signal is two-dimensional (ie., N = 2), with
four message points (ie., M = 4), as shown in the Figure 4.18.
Figure 4.18 signal space diagram for MSK system
156
The coordinates of message points are given as

S1 = √𝐸𝑏 cos[𝜃(0)] and (4.32)
S2 = −√𝐸𝑏 sin[𝜃(𝑇𝑏 )] (4.33)
Table 4.4 presents a summary of the values of 𝜃(0) and 𝜃(Tb), as well as the
corresponding values of S1 and S2.
Table 4.4: Signal-Space characterization of MSK
Coordinates of message
Transmitted Phase states (radians)
points
binary symbol
0 ≤ t ≤ Tb
(0) (Tb) S1 S2
𝜋
1 0 +2 + √𝐸𝑏 -√𝐸𝑏
𝜋
0  + - √𝐸𝑏 -√𝐸𝑏
2
𝜋
1  - -√𝐸𝑏 +√𝐸𝑏
2
𝜋
0 0 - +√𝐸𝑏 +√𝐸𝑏
2
Merits of MSK:
• MSK scheme has constant envelope (ie., there are no amplitude variations).
• It has coherent detection performance equivalent to that of QPSK.
• The MSK signal has a continuous phase (ie., there are no phase changes in
the MSK signal)
• Filters to suppress the sidelobes which causes interchannel interference are

not required.
Demerits of MSK:
• The generation and detection of MSK signal is more complicated.
• For a wireless communication system using MSK, the adjacent channel

interference is not low enough to satisfy the practical requirements of such a
multiuser communications environment.
157
4.4.3 Performance Comparison:

Table 4.5 shows the performance comparison of the quadrature modulation
schemes of QPSK and MSK.
Table 4.5 Performance comparison
Sl.
Parameter QPSK MSK
No.
1. Switching or keying of Phase Frequency
2. Bandwidth fb 1.5 fb
3. Bits per symbol Two Two
4. System complexity Moderately complex Very complex
5. Demodulation method Coherent detection Coherent detection
6. Noise immunity High High
7. Probability of symbol error Low Low
8. Carrier signal A pair of quadrature A pair of sinusoidally

carriers modulated quadrature
carriers
4.5 DETECTION OF SIGNALS:

In previous sections, we described various types of modulation methods that
may be used to transmit digital information through a communication channel. The
modulator at the transmitter performs the function of mapping the digital binary input
sequence into corresponding signal waveforms. Here, we shall study the
performance characteristics of receivers for the various modulation methods. The
Figure 4.19 shows the two basic steps in a digital receiver.
Receiver
Channel Sample at t
Received Message
=T
Transmitted signal Symbol
signal, Si(t)  Demodulator Detector
x(t)
Noise, n(t)
Figure 4.19 Two basic steps in digital receiver
158
The transmitted signal Si(t) is degraded by noise n(t) and impulse response of
the channel hc(t). Hence the received signal is given by
x(t) = Si(t) * hc(t) + n(t) (4.34)
On receiving the signal x(t), the digital receiver performs two basic functions
of demodulation and detection.
1. Demodulator
The demodulator is a frequency down conversion block. The function of the
signal demodulator is to convert the received waveform x(t) in to an N-dimensional
vector x=[x1, x2, … xN] where N is the dimension of the transmitted signal waveforms.
Signal demodulator can be realized in two ways. They are
A) Based on the use of signal correlators (product integrators)
B) Based on the use of matched filters.
2. Detector:
The function of the detector is to decide which of the M possible signal
waveforms was transmitted based on the vector x. The optimum detector is
designed to minimize the probability of error.
4.5.1 Correlation Receiver:

The basic function of a correlator is to product integrate the received noisy
signal with each of the reference carrier signals. It decomposes the received signal
into N-dimensional vectors (x1, x2, …xN). The Figure 4.20 shows the block diagram
of correlation type receiver, using a bank of N correlators.
𝑇 x1
∫ 𝑑𝑡
0
1(t)
𝑇 X2
Received Message
∫ 𝑑𝑡
Signal 0 Decision Symbol
x(t) device
2(t)
:
: 𝑇 XN
∫ 𝑑𝑡
0
N(t) Sample at t = T
Demodulator Detector
Figure 4.20 Correlator receiver
159
Here the demodulators imply the use of analog hardware (multipliers and
integrators) and continuous signals. The mathematical operation of a correlator is
correlation; a signal is correlated with a replica of itself. The demodulator outputs
are sampled at the rate t=T to obtain the vector x=x1, x2, ….xN.
A decision device is used as a detector. The function of the detector is to
decide which of the symbols was actually transmitted. The decision rule for the
detector is to choose a symbol based on location of received vector x in the
particular decision regions of the signal space.
4.5.2 Matched Filter Receiver:
Figure 4.21 Matched filter characteristics
The mathematical operation of a matched filter is convolution; a signal is

convolved with the impulse response of a filter. A matched filter is a linear filter
designed to provide the maximum signal-to- noise power ratio at its output for a
given transmitted symbol waveform. Also, the impulse response of the filter is a
delayed version of the mirror image (rotated on the t = 0 axis) of the signal
waveform. Therefore, if the signal waveform is S(t), its mirror image is S(-t), and the
mirror image delayed by T seconds is S(T-t), as shown in the Figure 4.21.
Thus a matched filter can be implemented using digital hardware and

sampled waveforms. The figure 4.22 shows the block diagram of matched filter
receiver. Here we use a bank of N linear filters followed by envelope detectors.
Envelope detectors are used to avoid poor sampling that arises in the absence of
prior information about the phase 𝜃.The demodulator outputs are sampled at the rate
t=T to obtain the vectors x=x1, x2,….xN. The decision device follows the decision rule
to decide which of the symbols was actually transmitted.
160
Matched Envelope x1
filter, 1 detector
Matched Envelope X2 Message

Received
Signal filter, 2 detector Decision Symbol
x(t) device
Matched Envelope XN
filter, N detector
Sample at
t=T
Demodulator Detector
Figure 4.22 Matched filter receiver
4.6 COHERENT DETECTION OF PSK

The figure 4.23 shows the block diagram of coherent binary PSK receiver.
Correlator
𝑇
Received x1 𝑐ℎ𝑜𝑜𝑠𝑒 1 𝑖𝑓 𝑥1 > 0
Signal x(t)  {
𝑐ℎ𝑜𝑜𝑠𝑒 0 𝑖𝑓 𝑥1 < 0
0 device
Threshold = 0
1(t)
Figure 4.23 coherent binary PSK receiver
The noisy BPSK signal x(t) received from the channel is applied to a
correlator. The correlator is also supplied with a locally generated coherent
reference signal 1(t). The correlator output, x1, is compared with a threshold of zero
volts. If x1>0, the receiver decides in favour of symbol 1. If x1<0, it decides in favour
of symbol 0. If x1 is exactly zero, the receiver makes a random guess in favour of 0
or 1. The average probability of symbol error or, equivalently, the bit error rate for
coherent BPSK is
1 𝐸
Pe = 2
erfc [√𝑁𝑏 ] (4.35)
0
161
Signal - Space Diagram:

A coherent BPSK system
is characterized by having a
signal space that is one-
dimensional (ie., N=1), and with
two message points (ie.,M=2),
as shown in figure 4.24
The coordinates of
message points are given by
Figure 4.24 Signal-Space diagram for coherent BPSK
S11 =
+√𝐸𝑏 and
S21 = -√𝐸𝑏
Hence the message point corresponding to S1(t) is located at S11 =+√𝐸𝑏 . The
message point corresponding to S2(t) is located at S21 = -√𝐸𝑏 . Now the decision rule
is
• If the received signal point falls in region Z1, guess that symbol 1 was
transmitted.
• If the received signal point falls in region Z2, guess that symbol 0 was
transmitted.
4.7 SAMPLED MATCHED FILTER

A matched filter is a linear filter designed to provide the maximum signal to
noise power ratio. The impulse response of the matched filter is a delayed version of
the mirror image (rotated on the t = 0 axis) of the input signal waveform. Therefore,
if the signal waveform is S(t), its mirror image is S(-t), and the mirror image delayed
by T seconds is S(T-t). Thus the impulse response h(t) of a filter matched to S(t) is
described by
𝑆(𝑇 − 𝑡) 0≤𝑡≤𝑇
h(t) = { (4.36)
0 𝑒𝑙𝑠𝑒𝑤ℎ𝑒𝑟𝑒
Matched filter can be implemented using digital techniques and sampled

waveforms. The figure 4.25 shows how a matched filter can be implemented using
digital hardware.
The input signal X(t) comprises a prototype signal S i(t) plus noise n(t). The
1
bandwidth of the signal is W = 2𝑇, where T is the symbol time. Thus, the minimum
1
Nyquist sampling rate is fs = 2W = . The sampling time Tb needs to be equal to or
𝑇
162
less than the symbol time. Therefore, there must be atleast one sample per symbol.
In real systems, sampling is usually performed at a rate that exceeds the Nyquist
minimum by a factor of 4.
Shift Register
x(t) x(k)
……
-w 0 1
w= Sample at
2𝑇𝑏
t = kTb
Ci(0) Ci(1) Ci(N-1)
y(k) = ∑𝑁−1
𝑛=0 𝑥(𝑘 − 𝑛)𝑐𝑖 (𝑛)
Figure 4.25 Sampled matched filter
At the clock times of t = kTb, the samples are shifted into the register, so that
earlier samples are located to the right of later samples. Once the received signal
has been sampled, the continuous time notation t is changed to kT b, or simply to k.
Then the simple discrete notation is
x(k) = Si(k) + n(k) i = 1, 2 k = 0, 1, …… (4.37)

where k is the sampling time index.
In the shift register, its coefficients or weights ci (n) approximate a matched
filter. Here n = 0, 1, …. N – 1 represents the time index of the weights and register
stages. N represents the number of stages in the register and the number of
samples per symbol. By using the discrete form of the convolution integral, the
output at a time corresponding to the kth sample is given by
Yi (k) = ∑𝑁−1
𝑛=0 𝑥(𝑘 − 𝑛)𝑐𝑖 (𝑛) k = 0, 1, …..modulo – N (4.38)
Following the summer, a symbol decision will be made after N time samples
have entered the registers.
163
Relation between correlator and matched filter:

Similarity
Even though the mathematical operation of an matched filter to be convolution
of a signal with the impulse response of the filter, the end result appears to be the
correlation of a signal with a replica of that same signal. That is why it is valid to
describe a correlator as an implementation of a matched filter.
Difference
There is an important distinction between the matched filter and correlator.
Since the correlator yields a single output value per symbol, it must have side
information, such as the start and stop times over which the product integration
should take place. If there are timing errors in the correlator, then the sampled
output fed to the detector may be badly degraded.
On the other hand, the matched filter yields a time series of output values
(reflecting time shifted input samples multiplied by fixed weights). Then with the use
of additional circuitry, the best time for sampling the matched filter output can be
learned.
4.8 COHERENT DETECTION OF FSK

The figure 4.26 shows the block diagram of coherent binary FSK receiver.
𝑇𝑏
x1
∫ 𝑑𝑡
0
+ l
Received Decision 𝑐ℎ𝑜𝑜𝑠𝑒 1 𝑖𝑓 𝑙 > 0
1(t)  {
Signal x(t) 𝑇𝑏 - device 𝑐ℎ𝑜𝑜𝑠𝑒 0 𝑖𝑓 𝑙 < 0
∫ 𝑑𝑡
0 x2
2(t)
Figure 4.26 coherent binary FSK receiver
The noisy BFSK signal x(t) received from the channel is applied to the pair of
correlators. The two correlators are supplied with locally generated coherent
reference signals 1(t) and 2(t). The correlator outputs x1 and x2 are then subtracted
one from the other. The resulting difference, l is compared with a threshold of zero
volts. If l > 0, the receiver decides in favour of symbol 1. If l < 0, it decides in favour
of symbol 0. If l is exactly zero, the receiver makes a random guess in favour of
0 or 1. The average probability of symbol error for coherent BFSK is
1 𝐸
Pe = 2
𝑒𝑟𝑓𝑐 [√2𝑁𝑏 ] (4.39)
0
164
Signal space diagram

A coherent BFSK system is characterized by having a signal space that is
two-dimensional (ie. N = 2) with two message points (ie., M = 2) as shown in Figure
4.27.
The two message points are defined by the signal vectors
0
S1 = [√𝐸𝑏 ] S2 = [ ]
0 √𝐸 𝑏
Figure 4.27 signal space diagram for BFSK
The distance between the two message points is equal to √2𝐸𝑏 . Now the
decision rule is
• If the received signal vector x falls in region Z1 (such that x1> x2), guess
that symbol 1 was transmitted.
• If the received signal vector x falls in region Z2 (such that x1< x2), guess
that symbol 0 was transmitted.
4.9 NON-COHERENT DETECTION

When it is impractical to have knowledge of the carrier phase at the receiver,
we use non coherent detection process. ‘Non coherent’ means doing without phase
information. Non coherent detection techniques are less complex. However, the
probability of error is high compared to coherent detection.
We treat non coherent binary FSK and DPSK signals as special cases of non-
coherent orthogonal modulation. Hence, we shall see about the non-coherent
receiver for detection of binary FSK and DPSK signals.
165
4.9.1 Non coherent detection of BFSK

In BFSK scheme, the transmitted signals are,
2𝐸
S1 (t) = √ 𝑇 𝑏 𝑐𝑜𝑠(2𝜋𝑓1 𝑡) (4.40)
𝑏
2𝐸
𝑏
S2 (t) = √ 𝑇 𝑐𝑜𝑠(2𝜋𝑓2 𝑡) (4.41)
𝑏
The transmission of frequency f1 represents symbol 1, and the transmission of

frequency f2 represents symbol 0.
The Figure 4.28 shows the non-coherent receiver for detection of binary FSK
signals.
Figure 4.28 Non-coherent receiver for BFSK Digital communications
The receiver consists of a pair of matched filters followed by envelope

2
detectors. The filter in the upper path of the receiver is matched to √𝑇 𝑐𝑜𝑠(2𝜋𝑓1 𝑡)
𝑏
2𝐸
ie.,1(t). The filter in the lower path is matched to √ 𝑇 𝑏 𝑐𝑜𝑠(2𝜋𝑓2 𝑡) ie.,2(t).
𝑏
The resulting envelope detector outputs are sampled at t = T b. The envelope

samples of the upper and lower paths are l1 and l2 respectively. The comparison
device compares the values of l1 and l2. On comparison,
• If l1> l2, the receiver decides in favour of symbol 1.
• If l1< l2, the receiver decides in favour of symbol 0.

166
If l1 = l2, the receiver simply makes a guess in favour of symbol 1 or 0. The

average probability of error for non-coherent binary FSK is given by
1 𝐸𝑏
pe = 𝑒𝑥𝑝 (− ). (4.42)
2 2𝑁0
4.9.2 Non-coherent detection of binary differential PSK

The term differentially coherent detection of DPSK, refers to a detection
scheme often classified as non-coherent because it does not require a reference in
phase with the received carrier.
The Figure 4.29 shows the block diagram of non-coherent receiver for
detection of DPSK signals.
Correlator
𝑇𝑏
x(t) Bandpass l 𝑐ℎ𝑜𝑜𝑠𝑒 1 𝑖𝑓 𝑙 > 0
 {
𝑐ℎ𝑜𝑜𝑠𝑒 0 𝑖𝑓 𝑙 < 0
Filter 0 device
Delay
Tb
Figure 4.29 Non-coherent receiver for DPSK
The received DPSK signal plus noise is passed through a Bandpass filter
centred at the carrier frequency f c, so as to limit the noise power. The filter output is
delayed by one bit interval Tb. Both the filter output and its delayed version are
applied to a correlator.
The resulting correlator output is proportional to the cosine of the difference
between the carrier phase angles in the two correlator inputs. The correlator output l
is finally compared with a threshold of zero volts. The decision is taken such that
• If l > 0, the phase difference between the waveforms received during the
𝜋 𝜋
pair of bit intervals lies inside the range − to . The receiver decides in
2 2
favour of symbol 1.
𝜋 𝜋
• If l < 0, the phase difference lies outside the range − to , modulo 2. The
2 2
receiver decides in favour of symbol .
DPSK is a special case of non-coherent orthogonal modulation with T = 2T b
and E = 2Eb. The average probability of error for DPSK is given by
1 𝐸𝑏
Pe = 𝑒𝑥𝑝 (− ) (4.43)
2 𝑁𝑜
167
4.10 ALLOCATION OF THE COMMUNICATIONS RESOURCE

One of the important design goals in a digital communication system is to
achieve maximum throughput (ie., maximum data rate). Three basic ways are used
to achieve maximum data rate.
1. Increase the transmitter’s EIRP (Effective Isotropic Radiated Power) or reduce
the system losses.
2. Provide more channel bandwidth.
3. More efficient distribution / utilization of the communication resources.
Multiplexing and Multiple Access are the two methods for more efficient
distribution / utilization of the communication resources. The communication
resources utilized are frequency, time, wavelength, code, space and polarization.
Multiplexing
Multiplexing may be defined as the process of simultaneously transmitting two
or more individual signals over a single communication channel. Using multiplexing,
more information can be transmitted at a time. The typical applications of
multiplexing are in telemetry and telephony or in satellite communication. There are
three basic types of multiplexing.
1. Time Division Multiplexing (TDM).

2. Frequency Division Multiplexing (FDM).
3. Wavelength Division Multiplexing (WDM).
Generally, the FDM and WDM systems are used to handle analog
information. TDM systems are often used to handle the digital information. Here we
shall see about TDM in detail.
4.10.1 Time Division Multiplexing (TDM)

In TDM, group of signals are sampled sequentially in time at a common
sampling rate and then multiplexed for transmission over a common channel. This
enables us to combine several digital signals, such as computer outputs, digitized
voice signals, digitized facsimile and television signals, into a single data stream with
a higher bit rate.
168
4.10.1.1 A PAM / TDM system

The concept of a PAM / TDM system is shown in the Figure 4.30.
There are N analog message signals in the input. Each message signal is
restricted in bandwidth by a low-pass pre-alias filter. The pre-alias filter outputs are
applied to a commutator. The commutator is an electronic switching circuitry which
takes a narrow sample of each of the N input messages at rate f s. Such multiplexed
samples are then applied to a pulse-amplitude modulator. It transforms the
multiplexed signal into a form suitable for transmission over the communication
channel.
Figure 4.30 Block diagram of PAM / TDM system
The received signal is applied to a pulse amplitude demodulator. The short

pulses produced at the demodulator output are distributed to the appropriate low-
pass reconstruction filters by means of a decommutator. The decommutator
operates in synchronism with the commutator. The transmitted message signals are
reproduced at the corresponding filter outputs.
4.10.1.2 Digital TDM

The figure 4.31 shows the concept of digital TDM.
The digital data can be multiplexed by using a bit-by-bit or byte-by-byte
interleaving procedure. This can be achieved by using a selector switch (MUX). The
switch sequentially selects a bit or byte from each input and places it over the high
speed transmission channel. At the receiving end, another switch (DEMUX)
separates the bit or byte and delivers them to their respective destinations.
169
Data Sources Destinations
1 Multiplexer Demultiplexer 1
2 2
Highspeed
DEMUX
3 3
MUX Transmission
Channel .
.
.
.
.
.
.
.
N
N
Figure 4.31 concept of digital TDM
In TDM, the communication resources are shared by assigning each of N

signals or users the full spectral occupancy of the system for a short duration of time
called a Time Slot as shown in Figure 4.32.
Guard Time
Time Time Time Time

Guard Time
Guard Time
Slot Slot Slot Slot ....

1 2 3 4
Time
Figure 4.32 TDM Time Slots
4.10.1.3 Types of TDM

There are two types of TDM in use. They are 1) Synchronous TDM and
Asynchronous TDM.
1) Synchronous or Deterministic TDM

It has a constant delay and bandwidth for a given individual communication
channel. Time slots have constant length (Capacity) and used in a synchronous
periodical manner. It is used in techniques like ISDN, PDH and SDH.
170
2) Asynchronous or statistical TDM

It has a variable delay and bandwidth for a given individual communication
channel. Time slots have variable length and are used on demand. It is used in
technologies like X25, Frame relay, ATM or IP.
4.10.1.4 TDM Frame structure

There are some basic requirements involved in the design of a digital TDM
multiplexer.
1. Digital signals cannot be directly interleaved into a format that allows for their
eventual separation unless their bit rates are locked to a common clock.
Accordingly, provision has to be made for synchronization of the incoming digital
signals, so that they can be properly interleaved.
2. The multiplexed signal must include some form of framing, so that its individual
components can be identified at the receiver.
Hence the TDM system uses a frame structure for placing data in each time
slot following a synchronized pattern. The TDM frame structure is shown in Figure
4.33. The TDM system divides the data stream into frames which repeat indefinitely.
Each TDM frame is then divided into equal timeslots which are allocated to individual
message signal or user. The individual users then transmit or receive only in their
own time slot.
Data
Stream
Single
1 2 3 …… N
Frame
Single 1 0 0 1 1 1 0 1 Data
Timeslot
Guard time
Figure 4.33 TDM Frame structure
The frame formats of synchronous and statistical TDM are shown in the
Figure 4.34.
171
F 1 2 3 …… N F  Framing bit
1, 2, .... N Time Slots
(a) Synchronous TDM Frame
Flag Address Control Data FCS Flag
(b) Statistical TDM Frame
Figure 4.34 Frame formats of TDM
4.11 ASCII FRAMING

The frames which are structured using character oriented protocols and
character stuffing are referred as character oriented frames. They use a normal
frame format and carry characters coded with protocol code. ASCII (American
Standard Code for Information Interchange) is one example of such protocol code.
A host PC uses ASCII characters to send commands to a device and then receives
responses back from that device. The ASCII command set is used to configure
devices, send data to devices and to read data and status information back from
devices. If the character oriented frame uses ASCII serial communication protocol,
then it is called as an ASCII frame.
ASCII Message Frame

The Figure 4.35 shows an example ASCII frame. The ASCII serial
communication protocol is used in this frame. It is used to transfer data between a
master computer station and a slave device such as power meter, or panel meter.
Field No. 1 2 3 4 5 6 7
Message Slave Message Message Check Trailer

Contents SYNC
length address type body sum (CRLF)
Length,
1 3 2 1 0 to 246 1 2
Character
Figure 4.35 Example ASCII Message Frame
The following specifies the ASCII message frame.

SYNC: Synchronization character: One character “!” (ASCII 33) is used for starting
synchronization.
172
Message length
The length of the message including only number of bytes in fields #2, #3, #4
and #5. It contains three characters between ‘006’ and ‘252’.
Slave Address
Two characters from ‘00’ to ‘99’. The instrument with address ‘00’ responds
to requests with any incoming address.
Message Type
One character representing the type of a host request. A list of message
types is shown in Tables 4.6 and 4.7.
Message Body
It contains the message parameters in ASCII representation. The data fields
vary in length depending on the data type, from 0 to 246.
Check sum
Arithmetic sum, calculated in a 2 byte word over fields #2, #3, #4 and #5 to
produce a one byte check sum in the range of 22to 7E (hexadecimal).
Trailer
Two ASCII characters Carriage Return (CR) (ASCII 13) and Line Feed (LF)
(ASCII 10) are used.
Table 4.6 Specific ASCII Requests

Message Type
Description
Char ASCII Hex
0 30h Read basic data registers
1 31h Read basic setup
2 32h Write basic setup
3 33h Read Instrument status
4 34h Reset / Clear functions
8 38h Reset the instrument
173
Table 4.7 Direct Read / Write ASCII Request

Message Type
Description
Char ASCII Hex
A 41h Long-size direct read
a 61h Long-size direct write
X 58h Variable-size direct read
x 78h Variable-size direct write
4.12 ARCHITECTURES OF SYNCHRONOUS TDM

Only a hierarchical digital multiplexing infrastructure can connect millions of
customers across the city / country / world. The local network used is simple star
and the wide area network is point to point trunks or ring topologies.
There are two main architectures for standard based synchronous TDM on
trunk lines for carrying PCM-coded digital telephony.
1. Plesiochronous Digital Hierarchy (PDH)
2. Synchronous Digital Hierarchy (SDH)
Plesiochronous means nearly synchronous. The network is not synchronized
but fast enough to synchronize sender and receiver. In PDH, there are two framing
methods based on the carrier system. They are
1. E1 Framing based on E1 carrier system.
2. T1 Framing based on T1 carrier system.
4.12.1 E1 Framing for Telephone

The E1 framing based on E1 carrier is an European hierarchy. The
conference of European post and Telecom (CEPT) administrations originally
standardized the E carrier system. E1 carrier standards are framed by European
Territory Standards Institute (ETSI). An E1 link operates over two separate sets of
wires, usually coaxial cable. It employs A-law algorithm for companding. This is
most suitable for voice transmission.
The most commonly used voice coding is PCM where the analog signal is
sampled at the rate of 8000 samples per second, and quantized by an 8 bit coder.
Hence the data rate of one PCM voice channel is 8000 x 8 = 64 kbps.
The telephone companies implement TDM through hierarchy of digital signals.
This is called as Digital Signal (DS) service. The table 4.8 lists the European digital
signals and their data rates.
174
Table 4.8 E1 hierarchy

Signal / Service Carrier / Line No of Channels Data Rate
DSO E0 1 64 kbps
CEPT 1 E1 32 2.048 mbps
CEPT 2 E2 128 8.448 mbps
CEPT 3 E3 512 34.368 mbps
CEPT 4 E4 2048 139.264 mbps
CEPT 5 E5 8192 565.148 mbps
The format of E1 framing is shown in the figure 4.36
Framing (8 bits) Signalling (8 bits)

 
TS0 TS1 TS2 TS3 …… TS16 …… TS28 TS29 TS30 TS31
TS Time slot each of 8 bits, one voice channel sample
TS1 to TS15 and TS17 to TS31  30 voice channels
1 Frame = 32 x 8 = 256 bits
Figure 4.36 E1 Framing format
The E1 framing data rate is 2.048 Mbps (full duplex). The frame is split into
32 time slots, each being allocated 8 bits. The timeslots are numbered from TSO to
TS31. The E1 frame repetition rate is 8KHZ.
Time Slot TS0

This slot is used for synchronization. It is reserved for framing purpose to
indicate the start of each frame.
Timeslots TS1 to TS15 and TS17 to TS31

These 30 time slots are used for carrying user data.
Time Slot TS16

This slot is used for signaling information. This includes control, call setup
and teardown. In E1 carrier, a 4-bit signaling information is used per time slot in
every 16th frame. This is called channel Associated Signaling (CAS) used for channel
synchronization.
175
4.12.2 T1 Framing for Telephone

The T1 framing based on T1 carrier is a North American hierarchy. The T1
carrier standards are framed by American National Standards Institute (ANSI). The
T1 link operates over special low capacitance two separate sets of shielded twisted
pair cabling. In some cases unshielded twisted pair cable is also used with
precautions to avoid cross talk. It employs -law algorithm for companding. This is
also most suitable for voice transmission.
The most commonly used voice coding is PCM. Hence the data rate of one
PCM voice channel is 64 kbps. T1 carrier is also implemented using the Digital
signal (DS) service. The table 4.9 lists the American digital signals and their data
rates.
Table 4.9 T1 hierarchy
Signal / Service Carrier / Line No of Channels Data Rate

DSO - 1 64 Kbps
DS1 T1 24 1.544 Mbps
DS1C T1C 48 3.152 Mbps
DS2 T2 96 6.312 Mbps
DS3 T3 672 44.736 Mbps
DS4 T4 4032 274.176 Mbps
The single PCM voice channel with data rate 64 Kbps is called as digital
signal at level zero (DSO). When 24 such PCM voice channels are multiplexed
using TDM, the multiplexed signal is the Digital signal at level one (DS1). Hence the
T1 carrier uses the DS1 signal at the line data rate of 1.544 Mbps. The format of T1
framing is shown in the figure 4.37.
Framing (1 bit)

F TS1 TS2 TS3 TS4 …………………. TS20 TS21 TS22 TS23 TS24
24 voice channels each of 8 bits
1 Frame = (24 x 8) + 1 = 193 bits
Figure 4.37 T1 Framing format
176
The T1 frame is split into 24 time slots each being allocated 8 bits and a
framing single bit at the start. The time slots are numbered from TS1 to TS24, each
representing a voice channel sample of 8 bits. The T1 frame repetition rate is
8 KHZ.
F bit
The frame synchronizing bit ‘F’ is used to provide synchronisation as well as
to indicate the start of a frame.
TS1 to TS24
These 24 time slots are used for carrying user data.
In T1 carrier, there is no dedicated time slot for channel associated signaling
(CAS). Instead ‘Robbed bit’ signaling is used. Using CAS, the signaling information
is transmitted by robbering certain bits, which are normally used for data. The
signaling is placed in the LSB of every timeslot in the 6th and 12th frame of every D4
super frame.
4.12.3 Comparison of E1 and T1 carriers

The Table 4.10 lists the performance comparison of E1 and T1 carriers.
Table 4.10 comparison of E1 and T1 frames

Sl.
Parameter E1 Frame / Carrier T1 Frame / Carrier
No.
1. Implemented in the Country Europe North America
2. Standard CEPT, ETSI ANSI
3. Digital Signal DSO = 64 KHZ DSO = 64 KHZ
PCM voice channel PCM voice channel
4. Hierarchy E1, E2, E3, E4 T1, T2, T3, T4
5. Cable used Coaxial Cable Twisted pair (STP,
UTP)
6. Line data rate 2.048 Mbps 1.544 Mbps
7. Companding A-law -law
8. No. of Time slots 32 24
9. No. of channels 30 24
177
SHORT QUESTIONS AND ANSWERS
1. Define Digital Modulation.

Digital modulation may be defined as mapping a sequence of input binary
digits into a set of corresponding high frequency signal waveforms.
2. List the various types of digital modulation techniques.

I. Based on the method of detection:
1. Coherent digital modulation
2. Non-coherent digital modulation
II. Based on the mapping techniques:

Quaternary
Binary Scheme M-ary Scheme Hybrid Scheme
Scheme
1. BASK 1. QPSK 1. M-ary ASK 1. QAM
2. BFSK 2. MSK 2. M-ary FSK 2. APK
3. BPSK 3. M-ary PSK
III. Based on the performance of the modulation scheme and properties of the
modulated signal
1. Power efficient scheme / Bandwidth efficient scheme.
2. Continuous Phase (CP) Modulation / In phase – Quadrature Phase (IQ)
Modulation.
3. Constant Envelope Modulation / Non-constant Envelope Modulation.
4. Linear Modulation / Non-linear Modulation.
5. Modulation scheme with memory / Modulation scheme without memory.
3. Mention the design goals of digital communication system.

1. Maximum data rate
2. Minimum possibility of symbol error.
3. Minimum transmitted power.
4. Minimum channel bandwidth.
5. Maximum resistance to interfering signals.
6. Minimum circuit complexity.
178
4. What is meant by coherent binary modulation technique?

The binary modulation scheme has three basic forms. They are
3. Binary Phase Shift Keying (BFSK)
When these binary modulation schemes employ coherent detection at the
receiver, then they are called as coherent binary modulation techniques.
In coherent detection, the local carrier generated at the receiver is phase
locked with the carrier at the transmitter.
5. Define ASK, FSK and PSK

ASK: In Amplitude Shift Keying (ASK), the modulation process involves switching or
keying the amplitude of the carrier signal in accordance with the incoming data.
FSK: In Frequency Shift Keying (FSK), the modulation process involves switching or
keying the frequency of the carrier signal in accordance with the incoming data.
PSK: In Phase Shift Keying (PSK) the modulation process involves switching or
keying the phase of the carrier signal in accordance with the incoming data.
6. What are the merits and demerits of BPSK?

Merits
• BPSK requires lower bandwidth than BFSK.
• BPSK has the minimum value of probability of error. Hence, it provides best
performance compared to BFSK and BASK schemes.
• It has very good noise immunity.
Demerits
• In PSK, the information lies in the phase, and hence, it cannot be detected
non-coherently.
7. What are the merits and demerits of BFSK?

Merits
• It is relatively easy to implement.
• It has better noise immunity than ASK.
Demerits
• BFSK requires high-bandwidth compared to BPSK and BASK.
179
8. What are the merits and demerits of BASK?

Merit
• BASK is easy to generate and detect.
Demerit
• It is very sensitive to noise.
9. What is meant by Non-coherent binary modulation technique?

The modulation scheme in which the detection process does not need
receiver carrier to be phase locked with the transmitter carrier is said to be non-
coherent modulation technique. The non-coherent binary modulation techniques are
1. Differential Phase Shift Keying (DPSK)
10. Define DPSK

Differential Phase Shift Keying (DPSK) is a “Pseudo PSK” technique and can
be viewed as the non-coherent form of BPSK. It eliminates the need for a coherent
reference signal at the receiver by combining two basic operations at the transmitter.
They are
1. Differential encoding of the input binary wave
2. Phase shift keying
11. What are the merits and demerits of DPSK?

Merits
• DPSK scheme does not need carrier at the receiver end. Hence it has
reduced system complexity.
• The bandwidth required is less than that required for BPSK.
Demerits
• It has higher value of probability of error than that of BPSK.
• Noise interference is more.
• In DPSK, previous bit is used to detect next bit. Hence, there is possibility of
errors appearing in pairs.
180
12. What is meant by coherent quadrature modulation technique?

The M-ary modulation scheme with m=4 is said to be quadrature modulation
scheme. The quadrature modulation scheme in which coherent detection is
employed at the receiver is called as the coherent quadrature modulation technique.
There are two bandwidth conserving quadrature modulation schemes. They are
1. QuadriPhase Shift Keying (QPSK)
2. Minimum Shift Keying (MSK)
13. Define QPSK

In QPSK, as with BPSK, information carried by the transmitted signal is
contained in the phase. For QPSK, the bits per symbol is k = 2 and hence m = 2 k =
22 = 4. Hence, two successive bits (dibit) in the data sequence are used to modulate
two quadrature carriers.
14. What is meant by signal constellation?

For any modulation scheme, the analysis is based on the signal space
diagram. Signal space approach is a plotting of possible message points. Such a
set of possible message points is also referred to as a signal constellation.
15. What are the merits and demerits of QPSK?

Merits
• QPSK has very good noise immunity.
• More effective utilization of the available bandwidth of the transmission
channel.
• It has low error probability.
Demerits
• The generation and detection of QPSK is complex.
16. Define MSK

Minimum Shift Keying (MSK) is a special form of binary CPFSK signal. A
Continuous Phase Frequency Shift Keying (CPFSK) signal with a deviation ratio of
h = ½ is referred to as MSK.
181
17. Define Deviation Ratio.

The difference between the frequencies f 1 and f2, normalised with respect to
1
the bit rate defines the dimensionless parameter h, which is referred to as the
𝑇𝑏
deviation ratio. For MSK, deviation ratio h = ½.
18. Justify the name Minimum Shift Keying (MSK)

The deviation ratio is h = Tb (f1 – f2)
1
For MSK, we have h = . Therefore,
2
1
2
= Tb (f1 – f2)
1 𝑅𝑏
 (f1 – f2) = 2𝑇𝑏
= 2
Where Rb  bit rate

Hence the frequency deviation (f 1 – f2) equals half the bit rate. This is the
minimum frequency spacing that allows the two FSK signals representing symbols 1
and 0 to be coherently orthogonal in the sense that they do not interfere with one
another in the process of detection. It is for this reason, a CPFSK signal with a
deviation ratio of one-half is referred to as Minimum Shift Keying (MSK).
19. What are the merits and demerits of MSK?

Merits
• MSK scheme has constant envelope (ie., there are no amplitude variations)
• It has coherent detection performance equivalent to that of QPSK.
• The MSK signal has a continuous phase.
• Filters to suppress the sidelobes which causes interchannel interference are
not required.
Demerits
• The generation and detection of MSK signal is more complicated.
• For a wireless communication system using MSK, the adjacent channel
interference is not low enough to satisfy the practical requirements of such a
multiuser communications environment.
20. What are the two basic steps in a digital receiver? Explain
The two basic steps in a digital receiver are 1) Demodulation and 2) Detection
1. Demodulator:
The demodulator is a frequency down conversion block. The function of the
signal demodulator is to convert the received waveform X(t) into an N-dimensional
182
vector X = [X1, X2, ..... XN] where N is the dimension of the transmitted signal
waveforms.
2. Detector
The function of the detector is to decide which of the M possible signal
waveforms was transmitted based on the Vector X. The optimum detector is
designed to minimize the probability of error.
21. What is meant by coherent and non-coherent detection in a digital

receiver?
Coherent Detection
In coherent detection, the local carrier generated at the receiver is phase
locked with the carrier at the transmitter. The receiver knows the instants of time
when the modulation changes state. The coherent detection is a synchronous
detection scheme.
Non-Coherent detection
In non-coherent detection, the detection process does not need receiver
carrier to be phase locked with the transmitter carrier. There is no time and phase
synchronization between the transmitter and receiver. Hence the non-coherent
detection is an asynchronous detection scheme.
22. Define sampled matched filter

A matched filter may be defined as a filter whose impulse response is a
delayed version of the mirror image (rotated on the t = 0 axis) of the input signal
waveform. Thus the impulse response h(t) of a filter matched to s(t) is given by
𝑠(𝑇 − 𝑡) 0 ≤ 𝑡 ≤ 𝑇
h(t) = {
0 𝑒𝑙𝑠𝑒𝑤ℎ𝑒𝑟𝑒
If the matched filter is implemented using digital techniques and sampled
waveforms, then it is called as sampled matched filter.
23. Define Correlator.

The basic function of a correlator is to product integrate the received noisy
signal with each of the reference carrier signals. It decomposes the received signal
into N-dimensional vectors (X1, X2, .... XN).
183
24. Define Time Division Multiplexing (TDM)

In TDM, group of signals are sampled sequentially in time at a common
sampling rate and then multiplexed for transmission over a common channel. TDM
is used to handle digital information. This enables us to combine several digital
signals, such as computer outputs, digitized voice signals, digitized facsimile and
television signals into a single data stream with a higher bit rate.
25. What are the types of TDM?

There are two types of TDM in use. They are
1. Synchronous TDM and 2) Asynchronous TDM
1. Synchronous or Deterministic TDM:
It has a constant delay and bandwidth for a given individual communication
channel. Time slots have constant length (capacity) and used in a synchronous
periodical manner. It is used in techniques like ISDN, PDH and SDH.
2. Asynchronous or statistical TDM:
It has a variable delay and bandwidth for a given individual communication
channel. Time slots have variable length and are used on demand. It is used in
techniques like X25, Frame relay, ATM or IP.
26. What is ASCII Framing?

American Standard Code for Information Interchange (ASCII) Codes in
hexadecimal notation is used in ASCII frame for TDM. ASCII frames are structured
using character oriented protocols.
27. Write short notes on EI framing.

• The EI framing based on EI carrier is an European hierarchy.
• The number of PCM encoded voice data channels is 30.
• The transmission line data rate is 2.048 Mbps.
• It employs A-law algorithm for companding and operates over coaxial cable.
28. Write short notes on TI framing

• The TI framing based on TI carrier is a North American hierarchy.
• The number of PCM encoded voice data channels is 24.
• The transmission line data rate is 1.544 Mbps.
• It employs -law algorithm for companding and operates over shielded twisted
pair cables.
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Spread Spectrum Techniques
Unit – V SPREAD SPECTRUM TECHNIQUES
OBJECTIVES
• To understand the spread spectrum communication.

• To understand about Pseudo noise sequences.
• To study about the major types of spreading techniques.
• To study about the commercial applications of spread spectrum

communication.
5.0 INTRODUCTION
In any digital communication system, the basic design factors are 1) efficient
utilisation of channel bandwidth and 2) minimizing the transmitted power.
Some of the major problems encountered in specific communication systems
are
1) Combating or suppressing the detrimental effects of interference due to
jamming, interference arising from other users of the channel, and self-
interference due to multipath propagation.
2) Hiding a signal by transmitting it at low power and making it difficult for an
unintended listener to detect the signal.
3) Achieving message privacy in the presence of other listeners.
These problems can be successfully solved by using a technique called
spread spectrum modulation. We shall discuss this modulation technique in detail in
this chapter.
5.1 SPREAD SPECTRUM COMMUNICATION SYSTEM

A system is defined to be a spread spectrum communication system if it fulfills
the following requirements.
1) The signal occupies a bandwidth much in excess of the minimum bandwidth
necessary to send the information.
2) Spreading is accomplished by means of a spreading signal, often called a
code signal, which is independent of the data.
185
3) At the receiver, despreading (recovering the original data) is done by the

correlation of the received spread signal with a synchronized replica of the
spreading signal used to spread the information.
In the transmitter of a digital communication system, such frequency
spreading of signal is achieved along with the Bandpass modulator circuit.
5.2 MODEL OF SPREAD SPECTRUM DIGITAL COMMUNICATION SYSTEM

The block diagram shown in Figure 5.1 illustrates the basic elements of a
spread spectrum digital communication system.
Figure 5.1 Model of spread spectrum digital communication system
The channel encoder / decoder and the modulator / demodulator are the basic
elements of the digital communication system, we have already discussed. In
addition to these elements we have two identical pseudorandom pattern generators.
One interfaces with the modulator at the transmitting end. The second interfaces
with the demodulator at the receiving end. These pseudorandom pattern generators
generate a Pseudonoise (PN) binary-valued sequence which is impressed on the
transmitted signal at the modulator and removed from the received signal at the
demodulator.
5.3 BENEFICIAL ATTRIBUTES OF SPREAD SPECTRUM SYSTEMS

Spread spectrum modulation was originally developed for military applications
where resistance to jamming (interference) is of major concern. However there are
civilian applications that also benefit from the unique characteristics of spread
spectrum modulation. We hereby list the following beneficial attributes of spread
spectrum systems.
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1) Interference suppression benefits:

(i) In combating intentional interference (jamming), the transmitter introduces an
element of unpredictability or randomness (pseudorandomness) in each of
the transmitted coded signal waveforms. This is known to the intended
receiver only, but not to the jammer. Thus interference due to jamming is
suppressed.
(ii) Resolvable multipath components resulting from time dispersive propagation
through a channel may be viewed as a form of self-interference. This type of
interference may also be suppressed by the introduction of pseudorandom
pattern in the transmitted signal.
2) Multiple Access
Spread spectrum methods can be used as a multiple access technique in
order to share a communication resource among numerous users in a coordinated
manner. Interference from the other users arises in multiple access communication
systems in which a number of users share a common channel bandwidth. The
transmitted signals in this common channel spectrum may be distinguished from one
another by superimposing a different pseudorandom pattern, also called a code, in
each transmitted signal. Thus, a particular receiver can recover the transmitted
information intended for it by knowing the code or key, used by the corresponding
transmitter. This type of communication technique, which allows multiple users to
simultaneously use a common channel for transmission of information, is called
Code Division Multiple Access (CDMA).
3) Energy Density Reduction

A message may be hidden in the background noise by spreading its
bandwidth with coding and transmitting the resultant signal at a low average power.
Because of its low power level, the transmitted signal is said to be “covert”. It has a
low probability of being intercepted (detected) by a casual listener. Hence, it is also
called as a Low-Probability of Intercept (LPI) signal.
A Radiometer is a simple power measuring instrument that can be used to
detect the presence of spread spectrum signals within some bandwidth, B.
4) Fine Time Resolution

Spread spectrum signals are used to obtain accurate range (time delay) and
range rate (velocity) measurements in radar and navigation. Distance can be
determined by measuring the time delay of a pulse as it traverses the channel.
187
5) Message Privacy
Message privacy may be obtained by superimposing a pseudorandom pattern
on a transmitted message. The message can be demodulated by the intended
receivers, who know the pseudorandom pattern or key used at the transmitter, but
not by any other receivers, who do not know the key.
5.4 SPREAD SPECTRUM APPROACHES (HISTORICAL BACKGROUND)

There are two spread-spectrum approaches called Transmitted Reference
(TR) and Stored Reference (SR).
(i) In a TR system, the transmitter send two versions of truly random spreading
signal (wideband carrier) – one modulated by data and the other unmodulated.
The receiver used the unmodulated carrier as the reference signal for
despreading (correlating) the data modulated carrier.
(ii) In a SR system, the spreading code signal is independently generated at both
the transmitter and the receiver. Since the same code must be generated
independently at two locations, the code sequence must be deterministic, even
though it should appear random to unauthorized listeners. Such random
appearing deterministic signals are called pseudonoise (PN), or
pseudorandom signals.
Modern spread spectrum systems use Stored Reference (SR) approach which
uses a Pseudonoise (PN) or pseudorandom code signal.
5.5 PSEUDONOISE SEQUENCES

A pseudonoise (PN) sequence may be defined as a coded sequence of 1’s
and 0’s with certain autocorrelation properties.
The PN sequence is a deterministic, periodic signal that is known to both the
transmitter and receiver. Even though the signal is deterministic it appears to have
the statistical properties of sampled white noise. Hence, it appears to be a truly
random signal, to an unauthorised listener.
5.5.1 Randomness properties

PN sequences have many of the properties possessed by a truly random
binary sequence. A random binary sequence is a sequence in which the presence
of a binary symbol 1 or 0 is equally probable. There are three basic properties that
can be applied to any periodic binary sequence as a test for the appearance of
randomness. They are described as follows.
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1) Balance Property:
In each period of the sequence, the number of 1’s is always one more than
the number of 0’s. This property is called the balance property.
2) Run Property:
Among the runs of 1’s and of 0’s in each period of the sequence, one-half the
runs of each kind are of length one, one-fourth are of length two, one-eighth are of
length three, and so on. This property is called the Run property. A run is defined
as a sequence of a single type of binary digit(s). The appearance of the alternate
digit in a sequence starts a new run. The length of the run is the number of digits in
the run.
3) Correlation Property
The autocorrelation function of a sequence is periodic and binary valued.
This property is called the correlation property.
5.5.2 Pseudo Noise (PN) sequence generator

The class of sequences used in spread spectrum communications is usually
periodic in that a sequence of 1s and 0s repeats itself exactly with a known period.
The maximum length sequence, a type of cyclic code represents a commonly used
periodic PN sequence.
The maximum length sequences or PN sequences can be generated easily
using shift register circuits with feedback from one or more stages. A PN sequence
generator using a 3-stage shift register is shown in Figure 5.2.
Clock
Maximum Length Sequence

X1 X2 X3 Output
Flip Flops
Feedback(𝑥1′ )
Modulo 2 adder
Figure 5.2 PN sequence or Maximum Length Sequence Generator
189
The 3-stage shift register consists of 3 flipflops regulated by a single timing

clock. At each pulse of the clock, the state of each flipflop is shifted to the next one.
The feedback function is obtained by using modulo-2 addition of the outputs of
flipflops x2 and x3. The feedback term is applied to the input of the first flipflop x 1.
The maximum length sequence output is obtained by noting the contents of flipflop
x3 at each clock pulse. The maximum-length sequence so generated is always
periodic with a period of
N = 2m-1 (5.1)
where m is the length of the shift register. Here, m = 3 and so N = 23-1 = 7.
For the PN sequence generator of Figure 5.2, if we assume that the shift
register contents are initially 111, then with each clocking pulse, the contents will
change as shown in the following table 5.1
Table 5.1 operation of the PN sequence generator
Shift register contents

Shifts 𝒙′𝟏 = X2 X3
X1 X2 X3
0 1 1 1
1 11=0 0 1 1
2 11=0 0 0 1
3 01=1 1 0 0
4 00=0 0 1 0
5 10=1 1 0 1
6 01=1 1 1 0
7 10=1 1 1 1
Hence for one period, the output PN sequence is 1 1 1 0 0 1 0, with a

sequence length of 7. Thereafter, the sequence will be repeated.
190
5.5.3 Important Observations

• The length of the PN sequence is N = 2m-1, where m is the number of shift
register stages.
• The PN sequence repeats itself after every ‘N’ clock cycles.
• The PN sequence is an NRZ type pulse signal with logic l represented by + 1
and logic 0 represented by -1, as shown in Figure 5.3.
1 1 1 0 0 1 0 1 1
+1
-1 Sequence

Tc repeats
hereafter
Tb = NTc
Figure 5.3 PN Sequence waveform
• The duration of every bit is known as the chip duration T c. The chip rate Rc is
defined as the number of bits (chips) per second.
1 1
Tc = (or) Rc = (5.2)
𝑅𝑐 𝑇𝑐
• The period of the PN sequence is Tb = NTc

• The autocorrelation function R() is a periodic function of time and it is a two
valued function.
R()
+1.0
-Tc Tc

−1⁄𝑁
−1
-NTc +NTc
𝑁
Figure 5.4 Autocorrelation function of a PN sequence

Example 5.1
A four stage shift register with feedback connections taken from the outputs of
stages 4 and 1 through a modulo – 2 adder, is used for PN sequence generation.
Assuming the initial contents of the shift register to be 0100, determine the output
sequence. What is the length of the sequence?
191
Solution: The PN sequence generator is shown in Figure 5.5

Clock
PN Sequence
X1 X2 X3 X4
Output
Flip Flops
Feedback(𝑥1′ )
Modulo 2 adder
Figure 5.5 PN Sequence Generator
If the initial contents of the shift register are 0100, then with each clocking
pulse, the contents will change as shown in the following table 5.2.
Table 5.2 Operation of PN sequence generator
Feedback Shift register contents

Shifts
𝒙′𝟏 = X4 X1 X1 X2 X3 X4
0 0 1 0 0
1 00=0 0 0 1 0
2 00=0 0 0 0 1
3 10=1 1 0 0 0
4 01=1 1 1 0 0
5 01=1 1 1 1 0
6 01=1 1 1 1 1
7 11=0 0 1 1 1
8 10=1 1 0 1 1
9 11=0 0 1 0 1
10 10=1 1 0 1 0
11 01=1 1 1 0 1
12 11=0 0 1 1 0
13 00=0 0 0 1 1
14 10=1 1 0 0 1
15 11=0 0 1 0 0
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The output PN sequence is
001000111101011
After 15 shiftings, the initial contents of the shift registers are once again
obtained. For further shiftings, the same cycle of events will repeat. Thus, the
length of one period of the PN sequence is, N = 2m – 1 = 24 – 1 = 15. Hence the
sequence is a maximal length sequence.
5.5.4 Testing of PN sequence for Randomness Properties
Let us consider example 5.1 for testing of PN sequence, for Randomness

properties.
1) Balance Property:
The output PN sequence is given by 0 0 1 0 0 0 1 1 1 1 0 1 0 1 1. There are
seven 0s and eight 1s in the sequence. Hence balance property is satisfied.
2) Run Property:
Consider the zero runs - there are four of them. One-half are of length 1, one-
fourth are of length 2. The same is true for the one runs. Hence run property is
satisfied.
3) As shown in Figure 5.4, the autocorrelation function R() will be a periodic
function of time and will be a two valued function. Hence the correlation property
is also satisfied.
4) For an m-stage linear feedback shift register the sequence repetition period in
clock pulses is
N = 2m - 1
Thus it can be seen that the sequence generated by the shift register
generator of Figure 5.5 is an example of maximum length sequence.
5.5.5 Demerits of spread spectrum system

The use of a spreading code in the transmitter produces a wideband
transmitted signal that appears noise like to a receiver that has no knowledge of the
spreading code. Naturally, this technique provides improved protection against
interference. But there are also some demerits involved in this method. They are
• Increased transmission bandwidth
• System complexity
• Processing delay
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Hence, spread spectrum systems are employed only for those applications
where security of transmission is our primary concern.
5.6 CLASSIFICATION OF SPREAD SPECTRUM MODULATION TECHNIQUES

The SS modulation techniques are broadly classified into two categories
namely, the averaging type systems and the avoidance type systems. The
averaging systems reduce the interference by averaging it over a long period. The
Direct Sequence Spread Spectrum (DS-SS) system is an averaging system.
Spread spectrum modulation techniques
Averaging type systems Avoidance type systems
Direct sequence
spread spectrum
(DS-SS) system Frequency Time Chirp Hybrid
hopping hopping methods
Figure 5.6 Classification of SS Modulation techniques
The avoidance type systems reduce the interference by making the signal
avoid the interference over a large fraction of time. Some of the avoidance type
systems are Frequency Hopping (FH) system, Time hopping (TH) system, Chirp and
hybrid modulation system.
5.7 DIRECT SEQUENCE SPREAD SPECTRUM SYSTEMS

The most important advantage of spread spectrum modulation is that it
provides protection against externally generated interfering signals such as jamming
signals. The Direct Sequence Spread Spectrum (DS-SS) technique can be used in
practice for such interference suppression. For this transmission of information
signal is carried over a band pass channel (eg. Satellite channel). For such an
application, the coherent Binary Phase Shift Keying (BPSK) is used in the
In the Direct sequence spread spectrum (DS-SS) systems, the use of a PN
sequence to modulate a phase shift keyed signal achieves instantaneous spreading
of the transmission bandwidth.
194
DS-BPSK Transmitter
The Figure 5.7 shows the transmitter section of the Direct Sequence Spread
Spectrum with coherent BPSK.
The transmitter section uses two stages of modulation. In the first stage the
input data sequence is first converted into an NRZ sequence b(t) by the NRZ
encoder. This sequence b(t) is used to modulate a wide band pseudo-noise
sequence c(t) by applying these two sequences to the product modulator or
multiplier. Both sequences are in polar form. The product sequence m(t) = b(t) . c(t)
will have a spectrum which will be same as that of c(t). The modulated signal m(t) is
used to modulate the local carrier for BPSK modulation at the second stage. We can
also use QPSK modulation.
Product modulator
or Multiplier
Binary data
b(t) m(t)
sequence Polar NRZ BPSK S(t)
encoder modulator To Channel
c(t)
PN Sequence
generator ~ Carrier
Figure 5.7 DS-BPSK Transmitter
The second stage modulated output s(t) is thus a Direct Sequence Spread
binary phase shift keyed (DS | BPSK) signal. The phase modulation (t) of S(t) has
one of the two values, 0 and , depending on the polarities of the data sequence and
PN sequence, as shown in the Table 5.3.
Table 5.3 Truth table for phase modulation (t), Radians
Polarity of Data Sequence b(t) at time ‘t’
+ -
+ 0 
Polarity of PN
Sequence C(t) at time ‘t’
-  0
195
Waveforms
Figure 5.8 illustrates the wave forms for the first stage of modulation.
+1
(a) Data b(t) 0 t
-1
Tb
+1
(b) Spreading code c(t) 0 t
-1
Tc
NTc
+1
(c) Product signal m(t) 0 t
-1
Figure 5.8 Waveforms for First stage of modulation
Figure 5.9 illustrates the waveforms for the second stage of modulation for
one period of the PN sequence.
(a) Product signal m(t) = c(t) b(t) (b) Sinusoidal carrier
(c) DS/BPSK signal

Figure 5.9 Waveforms for Second Stage of Modulation
196
DS-BPSK Receiver
The figure 5.10 shows the Receiver section of DS-BPSK system
Synchronous Detector
Choose
Multiplier
Received 1 if v>0
Product
𝑇𝑏 v Decision
signal r(t)
LPF ∫ 𝑑𝑡 device 0 if v<0
Modulator 0
Data
Local PN Threshold = 0
~ Local Sequence
Carrier generator
Figure 5.10 DS-BPSK Receiver
The receiver section consists of two stages of demodulation. In the first stage
the received signal r(t) is subjected to coherent detection using the locally generated
carrier signal. This carrier signal is arranged to be in phase and frequency
synchronism with the carrier used at the transmitter.
In the second stage, the output of the coherent detector is subjected to de-
spreading. It is multiplied with a locally generated PN sequence, which is in
synchronism with the one at the transmitter. After despreading, it is integrated over
a bit duration to get the observed random signal v. This is used for decision making,
which provides an estimate of the original data sequence.
Important Observation
• In practice, the transmitter and receiver of Figures 5.7 and 5.10 are followed.
In the transmitter spectrum spreading is performed prior to phase modulation.
Also phase demodulation is done first and then despreading is done second,
in the receiver.
• In the model of DS spread spectrum BPSK system used for analysis, the
order of these two operations are interchanged. In the transmitter, BPSK is
done first and spectrum spreading is done subsequently. Similarly, at the
receiver also, spectrum despreading is done first and then phase
demodulation is done second.
• This is possible, because the spectrum spreading and BPSK are both linear
operations.
197
Advantages of DS-SS System

1. This system combats the intentional interference (jamming) most effectively.
2. This system has a very high degree of discrimination against the multipath
signals. Therefore, the interference caused by the multipath reception is
minimized successfully.
3. The performance of DS-SS system in the presence of noise is superior to
other systems.
Disadvantages of DS-SS system

1. The PN code generator output must have a high rate. The length of such a
sequence needs to be long enough to make the sequence truly random.
2. With the serial search system, the acquisition time is too large. This makes
the DS-SS system be slow.
3. Synchronization is affected by the variable distance between the transmitter
and receiver.
4. The DS-SS signal is not very effective against broadband interference.
Major applications of DS-SS system

1. Providing immunity against a jamming signal – Anti-jamming application.
2. Low detectability signal transmission – the signal is purposely transmitted at a
very low power level. Hence the signal has a Low Probability of being
intercepted (LPI) and it is called an LPI signal.
3. Accommodating a number of simultaneous signal transmissions on the same
channel, ie. Code Division Multiple Access (CDMA) or spread spectrum
multiple access (SSMA).
5.8 PERFORMANCE PARAMETERS OF DS-SS SYSTEM

The important performance parameters of a direct sequence spread spectrum
system are 1) Processing gain, 2) Probability of Error and 3) Jamming Margin.
1) Processing Gain
The processing gain of a DS-SS system represents the gain achieved by
processing a spread spectrum signal over an unspread signal. It may also be
198
defined as the ratio of the bandwidth of the spread spectrum signal to the bandwidth
of the unspread signal.
𝐵𝑎𝑛𝑑𝑤𝑖𝑑𝑡ℎ 𝑜𝑓 𝑠𝑝𝑟𝑒𝑎𝑑 𝑠𝑖𝑔𝑛𝑎𝑙
Therefore, Processing Gain (PG) = 𝐵𝑎𝑛𝑑𝑤𝑖𝑑𝑡ℎ 𝑜𝑓 𝑢𝑛𝑠𝑝𝑟𝑒𝑎𝑑 𝑠𝑖𝑔𝑛𝑎𝑙
• With reference to Figure 5.8, the bit rate of the binary data entering the
transmitter input refers to the bandwidth of unspread signal. It is given by
1
Rb = 𝑇𝑏
(5.3)
• Also, the chip rate of the PN sequence refers to the bandwidth of spread
spectrum signal. It is given by
1
Rc = (5.4)
𝑇𝑐
• Therefore, Processing gain is given by

1⁄
𝑅𝑐 𝑇𝑐 𝑇𝑏
PG = = 1⁄ =
𝑅𝑏 𝑇𝑏 𝑇𝑐
𝑇𝑏
 PG = (5.5)
𝑇𝑐
• Also with reference to Figure 5.8, we note that T b = NTc. This can be rewritten
as
𝑇𝑏
N = (5.6)
𝑇𝑐
where N is the number of chips per information bit, and also called as the spread
factor.
• On comparing equations (5.5) and (5.6), we infer that both PG and N are
equal. Hence
𝑇𝑏
PG = N = 𝑇𝑐
(5.7)
• The Processing Gain (PG) is also called as the bandwidth expansion factor
(Be) since it represents the advantage gained over the jammer that is obtained
by expanding the bandwidth of the transmitted signal.
2) Probability of Error
• The probability of error Pe for a coherent BPSK system is given by
1 𝐸
Pe = 2
𝑒𝑟𝑓𝑐 √𝑁𝑏 (5.8)
𝑜
𝑁𝑜
where Eb is the energy per bit and 2
is the power spectral density of white noise.
199
• In a DS-SS BPSK system, the interference may be treated as a wideband

𝑁
noise signal with a power spectral density of 2𝑜. For the spread signal, we
may write No as
No = JTc (5.9)
where J refers to the average interference power and Tc refers to chip duration or
interval.
• On substituting the value of No in equation (5.8), we can write the probability
of error for the DS-SS-BPSK system as
1 𝐸𝑏
Pe = 𝑒𝑟𝑓𝑐 √ (5.10)
2 𝐽𝑇𝑐
3) Jamming Margin (Antijam characteristics)

𝐸𝑏
• We express the bit energy to noise density ratio as ⁄𝑁 . For the DS-SS-
𝑜
BPSK system, we may write No as equal to JTC (No = JTc). The bit energy Eb
is given by
Eb = PsTb (5.11)
where Ps is the average signal power and Tb is the bit duration or interval.
𝐸
• Hence 𝑏⁄𝑁 can be written as
𝑜
𝐸𝑏 𝑃𝑠 𝑇𝑏 𝑃 𝑇
= = ( 𝐽𝑠 ) ( 𝑇𝑏) (5.12)
𝑁𝑜 𝐽𝑇𝑐 𝑐
𝑇𝑏
𝐽 ⁄𝑇 𝑃𝐺
 = 𝐸𝑏
𝑐
= (𝐸𝑏 ) (5.13)
𝑃𝑠 ⁄𝑁 ⁄𝑁
𝑜 𝑜
𝑇𝑏
Since we know that PG=
𝑇𝑐
𝐽
• This ratio 𝑃𝑠
is called as the jamming margin. Therefore, the jamming margin
may be defined as the ratio of average interference power J and the average
signal power Ps.
• If the jamming margin and the processing gain are both expressed in
decibels, equation(5.13) can be written as
𝐸
(Jamming margin)dB = (Processing gain)dB - 10log10 ( 𝑏 ) (5.14)
𝑁𝑜 𝑚𝑖𝑛
𝐸
where( 𝑏 ) is the minimum bit energy-to-noise density ratio needed to support a
𝑁𝑜 𝑚𝑖𝑛
prescribed average probability of error.
200
Example 5.2 A spread spectrum communication system is characterised by

the following parameters
Information bit duration, Tb = 4.095 ms
PN chip duration, Tc = 1s
𝑬𝒃
Determine the processing gain and jamming margin if 𝑵𝒐
=10 and the
average probability of error, Pe=0.5 x 10-5.
Solution:
𝑇𝑏 4.095𝑚𝑠
(i) the processing gain, PG = 𝑇𝑐
= 1s
4.095 𝑥 10−3
PG = = 4.095 x 103 = 4095
1 x 10−6
Hence, PG = 4095. Since PG = Spread factor, N, we have PG = N = 4095.

(ii) The jamming margin is
𝐸
(Jamming margin)dB = (Processing gain)dB - 10log10 (𝑁𝑏 )
𝑜 𝑚𝑖𝑛
= 10log10 (4095) - 10log10 (10)

= 36.1225 – 10 =26.1225
(Jamming margin)dB = 26.1225dB
Important observation:
From the above example, we infer that the information bits at the receiver
output can be detected reliably, even when the noise or interference at the receiver
input is up to 409.5 times the received signal power. Clearly, this is a powerful
advantage against interference (jamming), which is obtained by the use of spread
spectrum.
5.9 FREQUENCY HOPPING SPREAD SPECTRUM SYSTEMS (FH-SS)

In the Direct sequence spread spectrum systems (DS-SS), the use of a PN
of the transmission bandwidth. The frequency hopping spread spectrum (FH-SS)
system is an alternative method. In FH-SS, the spectrum of the transmitted signal is
spread sequentially by randomly hopping the data modulated carrier from one
frequency to the next.
Hence, the type of spread spectrum in which the carrier hops randomly from
one frequency to another is called Frequency-hopped Spread Spectrum (FH-SS)
system.
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Basic Principle
In a FH-SS communication system the available channel bandwidth is
subdivided into a large number of contiguous frequency slots. In any signalling
interval, the transmitted signal occupies one or more of the available frequency slots.
The selection of the frequency slot(s) in each signalling interval is made
pseudorandomly according to the output from a PN generator. The figure 5.11
illustrates a particular FH pattern in the time-frequency plane.
Figure 5.11 An example of a frequency – hopped (FH) pattern
Reason for employing M-ary FSK modulation

A common modulation format for FH systems is that of M-ary frequency shift
keying (MFSK). The combination is referred to simply as FH/MFSK. Although PSK
modulation gives better performance than FSK in AWGN channel, it is difficult to
maintain phase coherence in
(i) the synthesis of frequencies used in the hopping pattern.
(ii) the propagation of the signal over the channel as the signal is hopped from
one frequency to another over a wide bandwidth.
Therefore, FSK modulation with non-coherent detection is usually employed with
FH spread spectrum signals.
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Types of Frequency hopping

Since frequency hopping does not cover the entire spread spectrum
instantaneously, we consider the rate at which the hops occur. We identify two basic
(technology-independent) characterizations of frequency-hopping. They are
1) Slow-frequency hopping
2) Fast-frequency hopping
5.9.1 Slow-frequency hopping:

In FH system, if the hopping is performed at the symbol rate, we have a slow-
hopped signal. Hence in slow-frequency hopping, the symbol rate Rs of the MFSK
signal is an integer multiple of the hop rate Rh ie., several symbols are transmitted on
each frequency hop.
Transmitter:
Figure 5.12 shows the block diagram of a slow-frequency hopping FH-MFSK
transmitter.
First, the incoming binary data are applied to an M-ary FSK modulator. The
resulting M-ary FSK modulated signal is applied to a Mixer. The Mixer consists of a
multiplier followed by a band pass filter (BPF).
Mixer
FH-MFSK
Binary data Signal
M-ary FSK Band Pass
Modulator Filter
Frequency
Synthesizer
…….....
PN code
generator
Figure 5.12 FH-MFSK Transmitter
The other input to the mixer block is obtained from a digital frequency
synthesizer. The frequency synthesiser is controlled by a PN code generator. Hence
the M-ary FSK modulated signal is again modulated by a carrier produced by the
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frequency synthesizer. The Mixer produces two outputs of the sum frequency and
the difference frequency. The band pass filter that follows the mixer selects only the
sum frequency signal, which is the FH-MFSK signal. This signal is then transmitted.
• Using the M-ary FSK system, M symbols can be transmitted, where M=2 K.
Here k is the number of bits of the input binary data that form one symbol.
• The M-ary FSK modulator will assign a distinct frequency for each of these M
symbols.
• The synthesizer output at a given instant of time is the frequency hop.
• The output bits of the PN generator change randomly. Hence the synthesizer
output frequency will also change randomly.
• Each frequency hop is mixed with the MFSK signal to produce the transmitted
signal.
• If the number of successive bits at the output of PN generator is n, then the

total number of frequency hops will be 2n.
• The total bandwidth of the transmitted FH-MFSK signal is equal to the sum of
all the frequency hops. Therefore, the bandwidth of the transmitted FH-MFSK
signal is very large of the order of few GHz.
Receiver:
Figure 5.13 shows the block diagram of a slow-frequency hopping FH-MFSK
receiver.
Mixer
Received Estimate of
signal binary data
Band Pass M-ary FSK
Filter detector
(Non-coherent)
Frequency
Synthesizer
…….....
Local PN code
generator
Figure 5.13 FH-MFSK Receiver
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• The received signal is applied as input to the Mixer. The other input to the
mixer is obtained from the digital frequency synthesizer.
• The frequency synthesizer is driven by a PN code generator. This generator
is synchronized with the PN code generator at the transmitter.
• Therefore, the frequency hops produced at the synthesizer output will be
identical to those at the transmitter.
• The mixer produces two outputs of the sum frequency and the difference
frequency. The band pass filter selects only the difference frequency, which
is the MFSK signal. Thus the mixer removes the frequency hopping.
• The MFSK signal is then applied to a non-coherent MFSK detector. A bank of
M, non-coherent matched filters are used for non-coherent MFSK detection.
Each matched filter is matched to one of the tones of the MFSK signal.
• An estimate of the original symbol transmitted is obtained by selecting the
largest filter output.
• For an FH/MFSK system,
(i) The chip rate, Rc = max (Rh, Rs) (5.14)
where Rh is the hop rate and Rs is the symbol rate

(ii) A slow FH/MFSK system is characterized by having multiple symbols
transmitted per hop. Hence, each symbol of a slow FH/MFSK system
is a chip.
(iii) We can relate all rates as
𝑅𝑏
Rc = Rs = ≥ Rh (5.15)
𝑘
where k = log 2 𝑀
Bandwidth of Spread signal
(iv) Processing gain, PG = Bandwidth of unspread signal
Let fs be the symbol frequency and 2n be the number of frequency hops

2𝑛 𝑓𝑠
Then, Processing gain, PG = 𝑓𝑠
= 2n (5.16)
1 𝑅𝑐
(v) Probability of error, Pe = 𝑒 −𝑟𝑏 (5.17)
2 2
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5.9.2 Frequency hopping example:

Figure 5.14 illustrates the frequency hopping by an example.
Figure 5.14 Frequency Hopping Example
• The input binary sequence data rate, Rb=150bits/s

• The modulation is 8-ary FSK.
𝑅 150
• Then the symbol rate is Rs= 𝑘𝑏 = log = 50 bits/s
28
1 1
• The symbol interval is Ts = 𝑅 = 50 = 20ms
𝑠
• The frequency is hopped once per symbol. Hence the hopping rate is
Rh=50hops/s.
• In the time-bandwidth plane of the figure, the abscissa (x-axis) represents
time and the ordinate(y-axis) represents the hopping bandwidth.
• A set of 8-ary FSK symbol-to-tone assignments is given. f0 refers to centre
frequency of the data band, which is not fixed.
1 1
• The tone separation is f = 𝑇 =20𝑚𝑠= 50Hz.
𝑠
• A typical binary data sequence is given at the top. Since the modulation is 8-
ary FSK, the bits are grouped three at a time to form symbols.
• A single-sideband tone (offset from fo) would be transmitted according to
symbol-to-tone assignment.
• For each new symbol, f0 hops to a new position in the hop bandwidth. For the
first symbol in the data sequence 011, f o+25Hz assignment is done. In the
206
figure, fo is shown with a dashed line and the symbol tone f o+25Hz is shown
with a solid line.
• Likewise, for the second symbol 110, f o - 125Hz assignment is done. For the
third symbol 001, fo + 125Hz assignment is done. For each symbols, the
centre frequency fo hops to a new position.
5.9.3 Frequency hopping with diversity:

In communication the transmitted signal’s ability to withstand impairments
from the channel, such as noise, jamming, fading, and so on is termed as
Robustness. A signal configured with multiple replicate copies, each transmitted on
a different frequency, has a greater likelihood of survival than does a single such
signal.
Diversity may be defined as multiple transmissions of the same signal at
different frequencies which are spread in time. The greater the diversity, the more
robust the signal against random interference.
To illustrate the beneficial effect of diversity, we can extend the frequency
hopping example shown in Figure 5.14. We introduce frequency hopping diversity
by a chip repeat factor of N=4. Figure 5.15 illustrates the effect of diversity.
Figure 5.15 Frequency hopping with diversity (N = 4)
207
• During each 20ms symbol interval, there are now four columns,
corresponding to the four separate chips to be transmitted for each symbol.
• Now, each symbol is transmitted four times. For each transmission, the
centre frequency f0 is hopped to a new region of the hopping band.
𝑇𝑠 20𝑚𝑠
• The chip interval is Tc = 𝑁
= 4
= 5ms.
𝑅 150 𝑥 4
• The hopping rate is Rh= log𝑏 8 . N = 3
= 200 hops/s.
2
• Also the spacing between frequency tones must change to satisfy

1 4
orthogonality. Hence the tone separation isf = .N = = 200Hz.
𝑇𝑠 20𝑚𝑠
• Hence, the resulting transmissions yield a more robust signal than that without
such diversity.
5.9.4 Fast-frequency hopping:

In FH system, if there are multiple hops per symbol, we have fast-hopped
signal. Hence in fast-frequency hopping, the hop rate Rh is an integer multiple of the
MFSK symbol rate Rs ie., the carrier frequency will change or hop several times
during the transmission of one symbol. Hence, in a fast FH-MFSK system, each hop
is a chip.
Figure 5.16 Fast FH-MFSK demodulator
In general, fast frequency hopping is used to defeat a smart jammer’s tactic

that involves two functions: measurement of the spectral content of the transmitted
signal, and returning of the interfering signal to that portion of the frequency band.
208
To overcome the jammer, the transmitted signal must be hopped to a new carrier
frequency before the jammer is able to complete the processing of these two
functions.
For data recovery at the receiver, non-coherent detection is used. However,
the detection procedure is different from that used in a slow FH-MFSK receiver. The
Figure 5.16 shows a typical fast FH-MFSK demodulator.
First, the signal is dehopped using a PN generator identical to that used in
transmitter. Then, filtering is done with a low pass filter having a bandwidth equal to
the data bandwidth. The filtered signal is demodulated using a bank of ’M’ envelope
detectors.
Each envelope detector is followed by a clipping circuit and an accumulator.
The clipping circuit serves an important function in the presence of an intentional
jammer or other strong unpredictable interference. The demodulator does not make
symbol decisions on a chip-by-chip basis. The energy from the N chips are
accumulated. After the energy from the Nth chip is added to the N-1 earlier ones, the
demodulator makes a symbol decision by choosing the symbol that corresponds to
the accumulator with maximum energy.
Advantages of FH-SS system:

1. The processing gain PG is higher than that of DS-SS system.
2. Synchronization is not greatly dependent on the distance.
3. The serial search system with FH-SS needs shorter time for acquisition.
Disadvantages of FH-SS system:

1. The bandwidth of FH-SS system is too large (in GHz).
2. Complex and expensive digital frequency synthesizers are required.
Applications of FHSS system:

1) CDMA systems based on FH spread spectrum signals are particularly
attractive for mobile communication.
2) Wireless local area networks (WLAN) standard for Wi-Fi.
3) Wireless Personal area network (WPAN) standard for Bluetooth.
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5.9.5 Fast hopping Versus Slow hopping:

Table 5.4 compares the performance of fast hopping and slow hopping
systems.
Table 5.4
SI No. Slow frequency hopping Fast frequency hopping
1. More than one symbols are More than one frequency hops are
transmitted per frequency hop. required to transmit one symbol.
2. Chip rate is equal to symbol rate. Chip rate is higher than Symbol
rate.
3. Symbol rate is higher than hop rate. Hop rate is higher than Symbol
rate.
4. Same carrier frequency is used to One symbol is transmitted over

transmit one or more symbols. multiple carriers in different hops.
5. A jammer can detect this signal if A jammer cannot detect this signal
the carrier frequency in one hop is because one symbol is transmitted
known. using more than one carrier
frequencies.
Slow hopping and fast hopping performance may also compared by the
following two examples:
1) Figure 5.17 shows chip in the context of an FH-MFSK system.
(a) Fast frequency hopping
(b) Slow frequency hopping

Figure 5.17 Chip in the context of an FM-MFSK System
210
• Figure 5.17(a) illustrates an example of fast frequency hopping. The data

symbol rate is 30 symbol/s and the frequency hopping rate is 60 hops/s. The
1
figure illustrates the waveform s(t) over one symbol duration ( s). The
30
waveform change in (the middle of) s(t) is due to a new frequency hop.
• Figure 5.17(b) illustrates an example of slow frequency hopping. The data

symbol rate is still 30symbols/s, but the frequency hopping rate has been
reduced to 10hops/s. The waveform s(t) is shown over a duration of three
1
symbols (10s).
2) Figure 5.18 shows the comparison for a binary FSK system.

• Figure 5.18(a) illustrates an example of fast frequency hopping for a binary
FSK system. The diversity is N=4. There are 4 chips transmitted per bit.
Here, the chip duration is the hop duration.
(a) Fast Frequency Hopping
(b) Slow Frequency Hopping

Figure 5.18 Comparison for a binary system
211
• Figure 5.18(b) illustrates an example of slow frequency hopping for a binary

FSK system. In this case, there are 3 bits transmitted during the time duration
of a single hop. Here, the chip duration is the bit duration.
5.10 SYNCHRONIZATION
5.10.1 Need for Synchronization
The process in which the locally generated carrier at the receiver must be in
frequency and phase synchronism with the carrier at the transmitter is called
synchronization. In spread spectrum communication systems, there should be
perfect alignment between the transmitted and received PN codes, for satisfactory
operation.
Because
(i) Carrier frequency as well as the PN clock may drift with time.
(ii) If there is relative motion between the transmitter and receiver, as in the
case of mobile and satellite spread spectrum systems, the carrier and PN
clock will suffer Doppler frequency shift.
Hence, synchronization of the PN sequence of the receiver with that of the
transmitter is essential.
5.10.2 Synchronization steps:

The process of synchronizing the locally generated spreading signal with the
received spread spectrum signal is usually accomplished in two steps. They are
1) Acquisition: The first step, called acquisition, consists of bringing the two
spreading signals into coarse alignment with one another.
2) Tracking: Once the received spread spectrum signal has been acquired, the
second step, called tracking, takes over for fine alignment.
Both acquisition and tracking make use of the feedback loop.
5.10.3 Acquisition:
Acquisition schemes can be classified into three types. They are
1) Serial search acquisition
2) Parallel search acquisition
3) Sequential search acquisition
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1. Serial search acquisition:

A) DS Spread spectrum systems:
Figure 5.19 shows the serial search scheme for Direct Sequence spread
spectrum systems.
There is always an initial timing uncertainty between the receiver and the
transmitter. Let us suppose that the transmitter has N chips and the chip duration is
Tc. If initial synchronization is to take place in the presence of additive noise and
other interference, it is necessary to dwell for T d=NTc in order to test synchronism at
each time instant. We search over the time uncertainty interval in(coarse) time steps
1
of Tc.
2
Figure 5.19 Direct Sequence spread spectrum systems – Serial Search Acquisition
The locally generated PN signal is correlated with the incoming PN signal. At

fixed search intervals of NTc (search dwell time), the output signal is compared to a
preset threshold. If the output is below the threshold, the locally generated code
1
signal is advanced in time by 2Tc seconds. The correlation process is repeated
again. These operations are performed until a signal is detected or the threshold is
exceeded. Then the PN code is assumed to have been acquired.
Thus, if initially the misalignment between the two codes was n chips, the total
time taken for acquisition is given by
Tacq = 2nNTc seconds (5.18)
213
B) FH spread spectrum systems
Figure 5.20 shows the serial search scheme for frequency hopping spread
spectrum systems.
Here the non-coherent matched filter consists of a mixer followed by a

bandpass filter (BPF) and a square law envelope detector. The PN code generator
controls the frequency hopper. Acquisition is accomplished when the local hopping
is aligned with that of the received signal.
Let fi be the frequency of the frequency synthesizer at the transmitter.

Suppose fj be the frequency of the signal produced by the frequency synthesizer in
the acquisition circuit of the receiver. If f i ≠ fj, then only a small voltage less than the
threshold will be produced at the output of BPF. At a later instant of time during
searching, if fi = fj, then a large voltage exceeding the threshold will be produced at
the output of BPF. This indicates the alignment of local hopping with that of the
received signal.
Figure 5.20 Frequency hopping serial search acquisition
2. Parallel search acquisition

The parallel search acquisition scheme introduces some degree of parallelism
by having two or more correlators operating in parallel. They will search over non-
overlapping time slots. In this scheme, the search time is reduced at the expense of
a more complex and costly implementation.
214
3. Sequential search acquisition

In this scheme, the dwell time at each delay in the search process is made
variable by employing a correlator with a variable integration period whose (biased)
output is compared with two thresholds. Hence the sequential search method
results in a more efficient search in the sense that the average search time is
minimised.
5.10.4 Tracking
Once the signal is acquired, the initial search process is stopped and fine
synchronization and tracking begins. The tracking maintains the PN Code generator
at the receiver in synchronism with the incoming signal. Tracking includes both fine
chip synchronization and, for coherent demodulation, carrier phase tracking.
A) DS Spread spectrum system:

The commonly used tracking loop for a Direct sequence spectrum signal is
the Delay-locked loop (DLL) as shown in the Figure 5.21.
Figure 5.21 Delay-Locked Loop (DLL) for PN code tracking
The received DS spread spectrum signal is applied simultaneously to two

multipliers. One of the multipliers is fed with PN code delayed by , a fraction of the
chip interval. The other multiplier is fed with the same PN code advanced by . The
output from each multiplier is fed to a BPF centred on f 0.
The output of each BPF is envelope detected and then subtracted. This
difference signal is applied to the loop filter that drives the voltage controlled
oscillator. The VCO serves as the clock for the PN Code generator. If the
215
synchronization is not exact, the filtered output from one correlator will exceed the
other. Hence the VCO will be appropriately advanced or delayed. At the equilibrium
point, the two filtered correlator outputs will be equally displaced from the peak
value. Then the PN code generator output will be exactly synchronized to the
received signal that is fed to the demodulator.
B) FH Spread spectrum system:

A typical tracking technique for FH spread spectrum signals is illustrated in
Figure 5.22.
Although initial acquisition has been achieved, there is a small timing error
between the received signal and the receiver clock. The BPF is tuned to a single
1
intermediate frequency and its bandwidth is of the order of 𝑇 , where Tc is the chip
𝑐
interval. Its output is envelope detected and then multiplied by the clock signal to
produce a three-level signal. This drives the loop filter.
VCO
Figure 5.22 Tracking loop for FH signals
Suppose that the chip transitions from the locally generated sinusoidal
waveform do not occur at the same time as the transitions in the incoming signal.
Then the output of the loop filter will be either positive or negative, depending on
whether the VCO is lagging or advanced relative to the timing of the input signal.
This error signal from the loop filter will provide the control signal for adjusting the
VCO timing signal so as to drive the frequency synthesizer output to proper
synchronism with the received signal.
216
5.11 PERFORMANCE COMPARISON OF DS-SS AND FH-SS
Direct sequence spread Frequency hopping

S.No. Parameter
spectrum spread spectrum
1 Definition The PN sequence makes The PN sequence makes
the transmitted signal the carrier hop over a
assume a noise like number of frequencies in a
appearance by spreading pseudo-random manner,
its spectrum over a broad with the result that the
range of frequencies spectrum of the transmitted
simultaneously. signal is spread in a
sequential manner.
2 Chip rate 1 Rc = max(Rh,Rs)
Rc =
𝑇𝑐
3 Modulation BPSK M-ary FSK

technique
4 Processing 𝑇𝑏 PG = 2n
PG = 𝑇𝑐
= N
gain(PG)
5 Error probability 1
Pe =
1
𝑒𝑟𝑓𝑐√𝐽𝑇
𝐸 Pe = 𝑒 −𝑟𝑏 𝑅𝑐 ⁄2
2
2 𝐶
6 Acquisition time Long time Short time

7 Effect of This system is distance Effect of distance is less.
distance relative.
5.12 JAMMING CONSIDERATIONS

5.12.1 Jamming:
Jamming refers to the intentional interference in a communication system.
The goals of the communicator are to develop a jam-resistant communication
system under the following assumptions:
(i) Complete invulnerability is not possible.
(ii) The jammer has apriori knowledge of most system parameters, such as
frequency bands, timing, traffic and so on.
(iii) The jammer has no apriori knowledge of the PN spreading or hopping codes.
The signalling waveform should be designed, so that the jammer cannot gain
any appreciable jamming advantage by choosing a jammer waveform and strategy
other than wideband Gaussian noise.
217
5.12.1.1 Jammer waveforms

• There are many different waveforms that can be used for jamming
communication systems. The most appropriate choice depends on the targeted
system.
• We shall assume that the jammer waveform is wideband noise and the jammer
strategy is to jam the entire bandwidth.
5.12.1.2 Tools of the communicator:

The usual design options for an antijam (AJ) communication system are
1) Frequency diversity by the use of direct sequence and frequency hopping
spread spectrum techniques.
2) Time diversity, by the use of time hopping.
3) Spatial discrimination, by the use of a narrow beam antenna.
5.12.1.3 J/S Ratio:

• The ratio (J/S)reqd is a figure of merit that provides a measure of how
invulnerable a system is to interference. It is given by
𝐽 𝑃𝐺
( ) = 𝐸𝑏 (5.19)
𝑆 𝑟𝑒𝑞𝑑 ( ⁄𝐽 )
𝑜 𝑟𝑒𝑞𝑑
where J  average received jammer power

S  received signal power
PG  processing gain
𝐸
( 𝑏⁄𝐽 )  bit energy per jammer noise power spectral density
𝑜 𝑟𝑒𝑞𝑑 required for maintaining the link at a specified error
probability.
5.12.1.4 Anti-jam margin:

• Anti-jam (AJ) margin usually means the safety margin against a particular
threat. It is defined as
𝐸 𝐸
MAJ (dB) = ( 𝐽𝑏 ) (𝑑𝐵) - ( 𝐽𝑏 ) (𝑑𝐵) (5.20)
𝑜 𝑟 𝑜 𝑟𝑒𝑞𝑑
𝐸 𝐸
where, ( 𝑏 )  ( 𝑏 ) actually received
𝐽𝑜 𝑟 𝐽𝑜
𝐸 𝐸
( 𝐽𝑏 )  ( 𝐽𝑏 )actually required
𝑜 𝑟𝑒𝑞𝑑 𝑜
218
5.12.2 Broadband noise jamming

• The jamming signal may be modelled as a zero-mean wide-sense stationary
Gaussian noise process.
• If the jammer strategy is to jam the entire spread spectrum bandwidth, with its
fixed power, then the jammer is referred to as a wide band or broad band
jammer.
5.12.3 Partial-band noise jamming:

• In the case of partial-band jamming, a specific transmitted symbol will be
received unjammed, with probability (1-P). Also, it will be perturbed by
jammer power with probability P.
• Forward error Correction (FEC) coding with appropriate interleaving can
mitigate this degradation.
5.12.4 Multiple-tone jamming:

• In the case of multiple-tone jamming, the jammer divides its total received
power, J into distinct, equal power, random phase CW tones.
• These are distributed over the spread spectrum bandwidth according to some
strategy.
5.12.5 Pulse jamming

• A pulse-noise jammer transmits pulses of band limited white Gaussian noise
having a time-averaged received power J.
• Forward error correction (FEC) coding with appropriate interleaving can
almost fully restore this degraded performance.
5.12.6 Repeat-back jamming

• The repeat-back jammers or frequency-follower (FF) jammers monitor a
communicator’s signal.
• They can increase the jamming power in the communicator’s instantaneous
bandwidth.
• To defeat the repeat back jammer, one method is to simply hop so fast that by
the time the jammer receives, detects and transmits the jamming signal, the
communicator is already transmitting at a new hop.
219
• Another technique capable of defeating the repeat-back jammer is a system

named as Buffalo Laboratories Application of Digitally Exact Spectra or
BLADES.
5.13 COMMERCIAL APPLICATIONS OF SPREAD SPECTRUM TECHNIQUES

Spread spectrum signals are used for
jamming (Intentional interference). It can be used in military applications also.
2) Accommodating multiple users to transmit messages simultaneously over the
same channel bandwidth. This type of digital communication in which each
user (transmitter-receiver pair) has a distinct PN code for transmitting over a
common channel bandwidth is called as Code Division Multiple Access
(CDMA) or Spread Spectrum Multiple Access (SSMA). This technique is
popularly used in digital cellular communications.
3) Reducing the unintentional interference arising from other users of the
channel.
4) Suppressing self-interference due to multipath propagation.
5) Hiding a signal by transmitting it at low power and, thus, making it difficult for
an unintended listener to detect in the presence of background noise. It is
also called a Low Probability of Intercept (LPI) signal.
7) Obtaining accurate range (time delay) and range rate (velocity)
measurements in radar and navigation.
5.14 CDMA - DIGITAL CELLULAR SYSTEM

The most important application of spread spectrum technique is the Digital
cellular CDMA system. Here, we shall explain in detail about this CDMA digital
cellular system based on Direct Sequence (DS) spread spectrum.
This digital cellular communication system was proposed and developed by
Qualcomm corporation. It has been standardized and designated as Interim
Standard 95 (IS-95) by the Telecommunications Industry Association (TIA) for use in
the 800 MHz and in the 1900 MHz frequency bands.
The nominal bandwidth used for transmission from a base station to the
mobile receivers (Forward link or channel) is 1.25 MHz. A separate channel, also
with a bandwidth of 1.25 MHz is used for signal transmission from mobile receivers
220
to a base station (reverse link or channel). The signals transmitted in both the
forward and reverse links are DS Spread spectrum signals having a chip rate of
1.288 x 106 chips per second (Mchips/s).
5.14.1 Forward link or channel

The signal transmission from a base station to the mobile receivers is referred
as the Forward link or channel. The figure 5.23 shows the block diagram of IS-95
forward link.
Source coding
The speech (source) coder is a code-excited linear predictive (CELP) coder.
It generates data at the variable rates of 9600, 4800, 2400 and 1200bits/s. The data
rate is a function of the speech activity of the user, in frame intervals of 20ms.
Channel coding
The data from the speech coder is encoded by a rate 1⁄2, constraint length
K = 9 convolutional code. For lower speech activity, the output symbols from the
convolutional encoder are repeated. If the data rate is 4800 bits/s, then the output
symbols are repeated twice, so as to maintain a constant bit rate of 9600 bits/s.
Figure 5.23 IS-95 Forward Link
221
Block interleaver:
The encoded bits for each frame are passed through a block interleaver. It is
needed to overcome the effects of burst errors that may occur in transmission
through the channel. The data bits at the output of the block interleaver occur at a
rate of 19.2kbits/s.
Symbol scrambler
The data bits from the block interleaver are scrambled by multiplication with
the output of a long code (period N=242-1) generator. This generator is running at
the chip rate of 1.288M chips/s, but the output is decimated by a factor of 64 to 19.2
kchips/s. The long code is used to uniquely identify a call of a mobile station on the
forward and reverse links.
Hadamard Sequence
Each user of the channel is assigned a Hadamard (or Walsh) sequence of
length 64. There are 64 orthogonal Hadamard sequences assigned to each base
station. Thus there are 64 channels available.
One Hadamard sequence is used to transmit a pilot signal. The pilot signal is
used for measuring the channel characteristics, including the signal strength and the
carrier phase offset. Another Hadamard sequence is used for providing time
synchronization. Another one sequence may be used for messaging (paging)
service. Hence there are 61 channels left for allocation to different-users. The data
sequence is now multiplied by the assigned Hadamard sequence of each user.
Modulator
The resulting binary sequence is now spread by multiplication with two PN
sequences of length 215 and rate 1.2288 Mchips/s. This operation creates in-phase
and quadrature signal components. Thus, the binary data signal is converted to a
four-phase signal. Then, both I and Q signals are filtered by baseband spectral
shaping filters.
Different base stations are identified by different offsets of these PN
sequences. The signals for all the 64 channels are transmitted synchronously.
Finally, heterodyning of a carrier wave with BPSK modulation and QPSK spreading,
is done. The summed output is the CDMA signal.
222
Mobile receiver
At the receiver, a RAKE demodulator is used to resolve the major multipath
signal components. Then, they are phase-aligned and weighted according to their
signal strength using the estimates of phase and signal strength derived from the
pilot signal. These components are combined and passed to the Viterbi Soft
decision decoder.
5.14.2 Reverse link or channel

The signal transmission from mobile transmitters to a base station is referred
as the Reverse link or channel. The Figure 5.24 shows the block diagram of IS-95
reverse link.
Limitations
In the reverse link, the signals transmitted from various mobile transmitters to
the base station are asynchronous. Hence, there is significantly more interference
among users. Also the mobile transmitters are usually battery operated and
therefore, these transmissions are power limited. We have to design the reverse link
in order to compensate for these two limitations.
Figure 5.24 IS-95 Reverse link
Source coding
The reverse link data may also be at variable rates of 9600, 4800, 2400 and
1200 bits/s. The data rate is a function of the speech activity of the user, in frame
intervals of 20ms.
223
Channel coding
The data from the speech coder is encoded by a rate 1⁄3, constraint length
K=9 convolutional code. This coder has higher coding gain in a fading channel. This
compensates for the above mentioned limitations.
For lower speech activity, the output bits form the convolutional encoder are
repeated either two, or four, or eight times.
Block interleaver
The encoded bits for each frame are passed through a block interleaver. It is
needed to overcome the effects of burst errors. For each 20ms frame, the 576
encoded bits are block-interleaved. However, the coded bit rate is 28.2 kbits/s.
Hadamard sequence
The data is modulated using an M=64 orthogonal signal set using Hadamard
sequences of length 64. Thus, a 6-bit block of data is mapped into one of the 64
Hadamard sequences. The result is a bit (or chip) rate of 307.2 kbits/s at the output
of the modulator.
Symbol scrambler
To reduce interference to other users, the time position of the transmitted
code symbol repetitions is randomized. Hence, at the lower speech activity,
consecutive bursts do not occur evenly spaced in time.
The signal is also spread by the output of the long code generator running at
a rate of 1.2288 Mchips/s. This is done for channelization (addressing), for privacy,
scrambling, and spreading.
Modulator
The resulting 1.2288 Mchips/s binary sequence at the output of the multiplier
is then further multiplied by two PN sequences of length N=2 15 with rate 1.2288
Mchips/s. This operation creates in phase and quadrature signals. Both the I and Q
signals are filtered by baseband spectral shaping filters.
The Q-channel signal is delayed in time by one half PN chip time relative to
the I-channel signal prior to the base band filter. The signal at the output of the two
baseband filters is an offset QPSK signal. Finally, the filtered signals are passed to
quadrature mixers. The summed output is the CDMA signal.
224
Base station Receiver

The base station dedicates a separate channel in order to receive the
transmissions of each active user in the cell. Although the chips are transmitted as
an offset QPSK signal, the demodulator at the base station receiver employs non-
coherent demodulation. A fast Hadamard transform is used to reduce the
computational complexity in the demodulation process. The output of the
demodulator is then fed to the Viterbi detector, whose output is used to synthesize
the speech signal.
SHORT QUESTIONS AND ANSWERS
1. What is spread spectrum communication?

A system is defined to be a spread spectrum communication system if it fulfills
the following requirements.
1. The signal occupies a bandwidth much in excess of the minimum bandwidth
necessary to send the information.
2. Spreading is accomplished by means of a spreading signal, often called a
code signal, which is independent of the data.
3. At the receiver, despreading (recovering the original data) is done by the
correlation of the received spread signal with a synchronized replica of the
spreading signal used to spread the information.
2. Mention the beneficial attributes of spread spectrum systems

Spread Spectrum Systems are useful for both military and civilian
applications. The beneficial attributes of spread spectrum system are listed below.
1) Interference suppression 2) Multiple access 3) Energy density reduction
4) Fine time resolution and 5) Message privacy
3. What is meant by Pseudonoise sequence?

A pseudonoise (PN) sequence may be defined as a coded sequence of 1’s
and 0’s with certain autocorrelation properties.
The PN sequence is a deterministic, periodic signal that is known to both the
transmitter and receiver. It appears to have the statistical properties of sampled
white noise. Hence, it appears to be a truly random signal, to an unauthorized
listener.
225
4. What are Randomness properties?

A random binary sequence is a sequence in which the presence of a binary
symbol 1 or 0 is equally probable. PN sequences have many of the properties
possessed by a truly random binary sequence. There are three basic properties that
can be applied to any periodic binary sequence as test for the appearance of
randomness. They are 1) Balance property 2) Run property and 3) Correlation
property.
5. State the Balance Property.

In each period of the sequence, the number of 1’s is always one more than
the number of 0’s. This property is called the Balance property.
6. State the Run Property

Among the runs of 1’s and of 0’s in each period of the sequence, one-half the
runs of each kind are of length one, one-fourth are of length two, one-eighth are of
length three, and so on. This property is called the Run property.
7. State the correlation property.

The autocorrelation function of a sequence is periodic and binary valued.
This property is called the correlation property.
8. How a pseudonoise (PN) sequence can be generated?

The class of sequences used in spread spectrum communications is usually
periodic in that a sequence of 1’s and 0’s repeats itself exactly with a known period.
The maximum length sequence, a type of cyclic code represents a commonly used
periodic PN sequence.
The maximum length sequences or PN sequences can be generated easily
using shift register circuits with feedback from one or more stages. The length of the
PN sequence is N = 2m–1, where m is the number of shift register stages.
9. What are the demerits of spread spectrum system?

1. Increased transmission bandwidth
2. System complexity
3. Processing delay
226
10. State the classification of spread spectrum modulation techniques.

I. Averaging type systems
1. Direct Sequence Spread Spectrum (DS-SS) System
II. Avoidance type systems
1. Frequency hopping Spread Spectrum (DS-SS) System
2. Time hopping system
3. Chirp
4. Hybrid Methods
11. Define Direct Sequence Spread Spectrum (DS-SS) system

In the Direct sequence spread spectrum (DS-SS) systems, the use of a PN
of the transmission bandwidth.
12. What are the advantages and disadvantages of Direct Sequence Sprtead
Spectrum (DS-SS) system
Advantages
1. This system combats the intentional interference (jamming) most effectively.
2. It has a very high degree of discrimination against the multipath signals.
Therefore the interference caused by the multipath reception is minimized
successfully.
3. The performance of DS-SS system in the presence of noise is superior to
other systems.
Disadvantages
1. The PN code generator output must have a high rate. The length of such a
sequence needs to be long enough to make the sequence truly random.
2. With the serial search system, the acquisition time is too large. This makes
the DS-SS system be slow.
13. What are the major applications of DS-SS system?

1. Providing immunity against a jamming signal-Anti jamming application.
2. Low detectability signal transmission – the signal is purposely transmitted at a
very low power level. Hence the signal has a low probability of being
intercepted (LPI) and it is called an LPI signal.
227
3. Accommodating a number of simultaneous signal transmissions on the same

channel, ie., code division multiple access (CDMA) or Spread Spectrum
Multiple Access (SSMA).
14. What are the performance parameters of Direct Sequence Spread Spectrum
DS-SS) system?
The important performance parameters of Direct Sequence Spread Spectrum
(DS-SS) system are 1) Processing gain 2) Probability of error and 3) Jamming
Margin.
15. Define Processing Gain.
The processing gain of DS-SS system represents the gain achieved by
processing a spread spectrum signal over an unspread signal. It may also be
defined as the ratio of the bandwidth of the spread spectrum signal to the bandwidth
of the unspread signal.
Bandwidth of Spread Signal
Processing gain (PG) =
Bandwidth of Unspread Signal
𝑇𝑏
Also , PG = , where Tb  bit duration, Tc  Chip duration
𝑇𝑐
16. State the probability of error for DS-SS BPSK system

The probability of error for the DS-SS BPSK system is
1 𝐸
Pe = 𝑒𝑟𝑓𝑐√𝐽𝑇𝑏
2 𝑐
where Eb  energy per bit

J  Average interference power
Tc  Chip duration
17. Define Jamming Margin

The jamming margin may be defined as the ratio of average interference
power J and the average signal power Ps.
𝐽 𝑃𝐺
Jamming Margin = 𝑃𝑠
= 𝐸𝑏 ⁄𝑁0
where PG  Processing gain

𝐸𝑏 ⁄𝑁0  bit energy to noise density ratio
18. Define Frequency hopping spread spectrum (FH-SS) System

In the Frequency hopping spread spectrum (FH-SS) systems, the spectrum of
the transmitted signal is spread sequentially by randomly hopping the data
modulated carrier from one frequency to the next.
228
19. State and define the types of frequency hopping.

There are two basic (technology-independent) characterizations of frequency
hopping.
1. Slow frequency hopping
In frequency hopping (FH) system, if the hopping is performed at the symbol
rate, we have a slow hopped signal. Here the chip rate is equal to the symbol rate.
2. Fast Frequency hopping
In frequency hopping (FH) system, if there are multiple hops per symbol, we
have fast hopped signal. Here the chip rate is higher than symbol rate.
20. Define frequency hopping with diversity.

Diversity may be defined as multiple transmissions of the same signal at
different frequencies which are spread in time. In frequency hopping with diversity, a
signal is configured with multiple replicate copies, each transmitted on a different
frequency. This signal has a greater likelihood of survival than does a single such
signal.
21. What are the advantages and disadvantages of Frequency Hopping Spread
Spectrum (FH-SS) System?
Advantages
1. The processing gain PG is higher than that of DS-SS system.
2. Synchronization is not greatly dependent on the distance.
3. The serial search system with FH-SS needs shorter time for acquisition.
Disadvantages
1. The bandwidth of FH-SS system is too large (in GHz)
2. Complex and expensive digital frequency synthesizers are required.
22. Mention the applications of FH-SS system

1. CDMA systems based on FH spread spectrum signals are particularly
attractive for mobile communication.
2. Wireless local area networks (WLAN) standard for Wi-Fi.
3. Wireless Personal Area Network (WPAN) standard for Bluetooth.
229
23. Compare slow hopping and fast hopping systems.
SI No. Slow frequency hopping Fast frequency hopping

1. More than one symbols are More than one frequency hops are
transmitted per frequency hop. required to transmit one symbol.
2. Chip rate is equal to symbol rate. Chip rate is higher than Symbol
rate.
3. Symbol rate is higher than hop rate. Hop rate is higher than Symbol
rate.
4. Same carrier frequency is used to One symbol is transmitted over
transmit one or more symbols. multiple carriers in different hops.
5. A jammer can detect this signal if A jammer cannot detect this signal
the carrier frequency in one hop is because one symbol is transmitted
known. using more than one carrier
frequencies.
24. Define synchronization

The process in which the locally generated carrier at the receiver must be in
frequency and phase synchronism with the carrier at the transmitter is called
synchronization. In spread spectrum communication systems, there should be
perfect alignment between the transmitted and received PN codes, for satisfactory
operation.
25. State and define the synchronization steps

The process of synchronizing the locally generated spreading signal with the
received spread spectrum signal is usually done in two steps. They are
1) Acquisition
The first step called acquisition consists in bringing the two spreading signals
into coarse alignment with one another.
2) Tracking
Once the received spread spectrum signal has been acquired, the second
step, called tracking, takes over for fine alignment.
26. List the acquisition and tracking schemes

Acquisition schemes can be classified into three types. They are
1. Serial Search Acquisition
2. Parallel Search Acquisition
3. Sequential Search Acquisition
230
Tracking includes both fine chip synchronization and, for coherent

demodulation, carrier phase tracking. The commonly used tracking loops are
1. Delay-locked loop (DLL)
2. Tau-dither loop (TDL)
27. What is jamming?

Jamming refers to the intentional interference in a communication system.
The signalling waveform should be designed, so that the jammer cannot gain any
appreciable jamming advantage by choosing a jammer waveform and strategy.
28. List the design options for an Antijam (AJ) communication System
1. Frequency diversity by the use of direct sequence and frequency hopping
spread spectrum techniques.
2. Time diversity by the use of time hopping.
3. Spacial discrimination by the use of a narrow beam antenna.
𝑱
29. Define 𝑺 Ratio
𝐽
The ratio ( ) is a figure of merit that provides a measure of how
𝑆 𝑟𝑒𝑞𝑑
invulnerable a system is to interference. It is given by

𝐽 𝑃𝐺
(𝑆 ) = (𝐸𝑏 ⁄𝐽𝑜 )𝑟𝑒𝑞𝑑
𝑟𝑒𝑞𝑑
where J  Average Received Jammer Power

S  Received Signal Power
PG  Processing Gain
(𝐸𝑏 ⁄𝐽𝑜 )𝑟𝑒𝑞𝑑  bit energy per jammer noise power spectral
density required for maintaining the link at a
specified error probability.
30. Define Anti-jam margin.

Anti-jam (AJ) margin usually means the safety margin against a particular
threat. It is defined as
𝐸 𝐸
𝑀𝐴𝐽 (𝑑𝐵) = ( 𝐽𝑏 ) − ( 𝐽𝑏 ) (𝑑𝐵)
𝑜 𝑟 𝑜 𝑟𝑒𝑞𝑑
231
𝐸 𝐸
where ( 𝐽𝑏 )  ( 𝐽𝑏 ) actually received
𝑜 𝑟 𝑜
𝐸 𝐸
( 𝐽𝑏 )  ( 𝐽𝑏 ) actually required
𝑜 𝑟𝑒𝑞𝑑 𝑜
31. List the commercial applications of spread spectrum techniques

Spread spectrum signals are used for
jamming (Intentional interference). It can be used in military applications also.
2) Accommodating multiple users to transmit messages simultaneously over the
same channel bandwidth. This type of digital communication in which each
user (transmitter-receiver pair) has a distinct PN code for transmitting over a
common channel bandwidth is called as Code Division Multiple Access
(CDMA) or Spread Spectrum Multiple Access (SSMA). This technique is
popularly used in digital cellular communications.
3) Reducing the unintentional interference arising from other users of the
channel.
4) Suppressing self-interference due to multipath propagation.
5) Hiding a signal by transmitting it at low power and, thus, making it difficult for
an unintended listener to detect in the presence of background noise. It is
also called a Low Probability of Intercept (LPI) signal.
7) Obtaining accurate range (time delay) and range rate (velocity)
measurements in radar and navigation.
232
MODEL QUESTION – I
Time : 3 Hours
Maximum Marks : 75
[N.B: (1) Answer any FIVE questions in each PART-A and PART-B, Q.No..8 in
PART-A and Q.No.16 in PART-B are compulsory.
(2) Answer division (a) or division (b) of each question in PART-C.
(3) Each question carries 2 marks in PART-A, 3 marks in PART-B and
10 marks in PART-C]
PART – A
1. Define Information Capacity.
2. What is aliasing?
3. What is Retransmission?
4. What are linear block codes?
5. Define digital modulation.
6. Define DPSK.
7. Mention the beneficial attributes of spread spectrum systems.
8. What is NRZ waveform?
PART - B
9. Define periodic and non-periodic signals.
10. Define Sampling theorems.
11. Explain the types of errors.
12. What is CRC code? Mention two of its applications.
13. What are the merits and demerits of MSK?
14. What are the major applications of DS-SS system?
15. Define Jamming Margin.
16. Define sampled matched filter.
233
PART - C
17. (a) Draw the typical block diagram of Digital Communication System and
Explain in detail.
(or)
(b) What is Data Transmission? Explain about synchronous and
asynchronous transmission.
18. (a) With neat sketches explain the various sampling techniques.
(or)
(b) With neat sketches explain the PCM waveform types.
19. (a) Explain in detail about the error control coding methods.
(or)
(b) Explain about Hamming codes with a suitable example.
20. (a) With neat sketches explain about BPSK. What are its merits and
demerits?
(or)
(b) Explain about (i) ASCII framing (ii) T1 framing for telephone.
21. (a) With neat sketches explain in detail about the Direct Sequence Spread
Spectrum Systems.
(or)
(b) With a neat block diagram, explain the Working of Forward link in CDMA
Digital Cellular System.
234
MODEL QUESTION – II
Time : 3 Hours
Maximum Marks : 75
[N.B: (1) Answer any FIVE questions in each PART-A and PART-B, Q.No..8 in
PART-A and Q.No.16 in PART-B are compulsory.
(2) Answer division (a) or division (b) of each question in PART-C.
(3) Each question carries 2 marks in PART-A, 3 marks in PART-B and
10 marks in PART-C]
PART – A
1. Define Unit Impulse Function.
2. Define PCM Wordsize.
3. List the error detection codes and error correction codes.
4. What is E1 framing for telephone?
5. List the various types of digital modulation techniques.
6. Define synchronization.
7. What are randomness properties?
8. What is forward error correction method?
PART - B
9. Mention the advantages of digital communication over analog communication.
10. What is quantisation noise?
11. Discuss the rationale for coding.
12. Define code rate and hamming distance.
13. Mention the design goals of digital communication system.
14. Draw the TDM frame structure.
15. Compare slow hopping and fast hopping systems.
16. What is Jamming? List the design options for an Antijam (AJ) communication
system.
235
PART - C
17. (a) Explain in detail about the various channels for digital communication.
(or)
(b) What is Data Transmission? Explain about serial and parallel
transmission.
18. (a) What is PCM? Explain about uniform and non-uniform quantization.
(or)
(b) (i) Briefly explain about the spectral attributes of PCM waveforms.
(ii) Write short notes on M-ary Pulse Modulation Waveforms.
19. (a) Explain the principles of linear block codes with a suitable example.
(or)
(b) (i) Explain about CRC Code
(ii) Explain about convolution code.
20. (a) With neat block diagrams, explain the working of MSK transmitter and
receiver.
(or)
(b) (i) Explain about sampled matched filter.
(ii) Explain about the Non-coherent detection of binary differential PSK.
21. (a) With neat sketches explain in detail about Slow Frequency hopping
Spread Spectrum Systems.
(or)
(b) With neat sketches explain any one method of acquisition and tracking.
236

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Unit - I BASICS OF DIGITAL COMMUNICATION

1.2 DIGITAL COMMUNICATION SIGNAL PROCESSING

Figure 1.1 Pulse degradation and regeneration

During propagation, before the pulse is degraded to an ambiguous state,

1.3 ADVANTAGES OF DIGITAL COMMUNICATION OVER ANALOG

1. The use of “Regenerative repeaters” generate strong error free signal at a

5. Digital hardware lends itself to more flexible implementation than analog

8. Digital techniques lend themselves naturally to signal processing functions

1.4 DISADVANTAGES OF DIGITAL COMMUNICATION OVER ANALOG

1. Digital systems tend to be very signal-processing intensive compared with

2. Digital systems need to allocate a significant share of their resources to the

1.5 TYPICAL BLOCK DIAGRAM AND TRANSFORMATIONS

Receiver Section x(t)

Figure 1.2(a): Block diagram of a Typical Digital Communication System

5) Base band processor

6) Band pass Modulator

2. Band pass demodulator

BASIC DIGITAL COMMUNICATION TRANSFORMATIONS

1. Formatting and Source Coding:

6. Multiplexing and multiple access

authentication means preventing unauthorized persons from injecting spurious

Fig. 1.2(b) Basic digital communication transformations

1.6 CHANNELS FOR DIGITAL COMMUNICATION

1.8 INFORMATION CAPACITY

1.9 SHANNON’S LIMIT FOR INFORMATION CAPACITY

The significance of the channel capacity is as follows:

Example problem 1.1: Calculate the capacity of a standard 4kHz telephone

Signal-to-noise ratio (S/N) in decibels = 32dB

C = 3100 x log 2 (1 + 1585)

= 3100 x log 2 (1586)

= 3100 x 10.63 = 32953

capacity, C = 32953 bits per second

Exercise Problem 1.8.1: A system has bandwidth of 4kHz and a signal-to-

1.10 DATA TRANSMISSION

Formatting transforms the source information into bits, thus assuring

2.3 FORMATTING ANALOG INFORMATION

2.3.1 The Sampling theorem

The statement of sampling theorem can be given in two parts as below:

(ii) A band limited signal of finite energy, which has no frequency

2.3.2 Nyquist theorem

2.4 SAMPLING TECHNIQUES

2.4.1 Impulse sampling or Ideal sampling

Figure 2.2 Impulse Sampling

x(t) = 𝑛=−∞ (𝑡 − 𝑛𝑇𝑠 )

Thus, xs(t) = x(t).x(t) (2.4)

xs(t) = 𝑛=−∞ (𝑡 − 𝑛𝑇𝑠 )

xs(t) = 𝑛=−∞ 𝑥(𝑛𝑇𝑠 ). (𝑡 − 𝑛𝑇𝑠 )

Demerits of Impulse sampling:

Figure 2.3 Spectra for various sampling rates

Now, it is easier to perform the filtering operation. A typical low-pass filter

2.4.2 Natural Sampling

Figure 2.4 Natural Sampling

xs (t) = x(t).xp(t) (2.8)

Hence, the sampled data sequence is given by

We can solve for Xs(f) as below,

2.4.3 Flat Top Sampling or Sample-and-Hold operation

xs(t) = P(t)  [x(t)x(t)]

= 𝑛=−∞ (𝑡 − 𝑛𝑇𝑠 )]

Xs(f) = 𝑛=−∞ (𝑡 − 𝑛𝑇𝑠 )}

The effect of this product operation results in a spectrum similar in