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CANARA

ENGINEERING COLLEGE
Benjanapadavu, Mangalore-574219

NOTES
Digital Communication
6th Semester (CBCS)
Course Code: 18EC61

PREPARED BY:

SUJIT S PAI
ASSISTANT PROFESSOR
DEPARTMENT OF
ELECTRONICS & COMMUNICATION
ENGINEERING
MODULE-1 VI SEM DIGITAL COMMUNICATION [18EC61]

MODULE - 1

SYLLABUS:
Bandpass Signal to Equivalent Low pass: Hilbert Transform, Pre-envelopes, Complex envelopes,
Canonical representation of bandpass signals, Complex low pass representation of bandpass systems,
Complex representation of band pass signals and systems.
Line codes: Unipolar, Polar, Bipolar (AMI) and Manchester code and their power spectral densities.
Overview of HDB3, B3ZS and B6ZS

TEXT, REFERENCE & ADDITIONAL REFERENCE BOOKS:

BOOK TITLE/AUTHORS/PUBLICATION

Simon Haykin, “Digital Communication Systems”, John Wiley & sons, First Edition,
T-1
2014, ISBN 978-0-471-64735-5.

B.P.Lathi and Zhi Ding, “Modern Digital and Analog communication Systems”,
AR-1
Oxford University Press, 4th Edition, 2010, ISBN: 978-0-198-07380-2.

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

MODULE - 2

SYLLABUS:
Signaling over AWGN Channels: Introduction, Geometric representation of signals, Gram- Schmidt
Orthogonalization procedure, Conversion of the continuous AWGN channel into a vector channel, Optimum
receivers using coherent detection: ML Decoding, Correlation receiver, matched filter receiver.

TEXT, REFERENCE & ADDITIONAL REFERENCE BOOKS:

BOOK TITLE/AUTHORS/PUBLICATION

Simon Haykin, “Digital Communication Systems”, John Wiley & sons, First Edition,
T-1
2014, ISBN 978-0-471-64735-5.

B.P.Lathi and Zhi Ding, “Modern Digital and Analog communication Systems”,
AR-1
Oxford University Press, 4th Edition, 2010, ISBN: 978-0-198-07380-2.

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

INTRODUCTION:
The conversion of analog waveforms into coded pulses represents the transition from analog
communications to digital communications. This transition has been empowered by several factors:
• Ever-increasing advancement of digital silicon chips, digital signal processing, and computers,
which, in turn, has prompted further enhancement in digital silicon chips, thereby repeating the cycle
of improvement.
• Improved reliability, which is afforded by digital communications to a much greater extent than is
possible with analog communications.
• Broadened range of multiplexing of users, which is enabled by the use of digital modulation
techniques.
• Communication networks, for which, in one form or another, the use of digital communications is
the preferred choice.

TYPICAL DIGITAL COMMUNICATION SYSTEM:


➢ A typical model of digital communication system is shown in Fig. 1.

Fig. 1: Model of Digital communication system

➢ The source output consists of a sequence of 1’s and 0’s with each binary symbol being emitted every T b
seconds. The transmitting part of the digital communication system takes the 1’s and 0’s emitted by the
source computer and encodes them into distinct signals denoted by s1(t) and s2(t), respectively, which are
suitable for transmission over the analog channel.
➢ Both s1(t) and s2(t) are real-valued energy signals, as shown by

----------- (1)
➢ The received signal is defined by

----------- (2)
where w(t) is the channel noise.

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Fig. 2: AWGN model of a channel

➢ The receiver has the task of observing the received signal x(t) for a duration of Tb seconds and then
making an estimate of the transmitted signal si(t) or equivalently the ith symbol, i = 1, 2. Because of the
error introduced in the channel, the receiver will make occasional errors. Hence it is required to design,
the receiver so as to minimize the average probability of symbol error, defined as

---------- (3)
where π1 and π2 are the prior probabilities of transmitting symbols 1 and 0 respectively and is the
estimate of the symbol 1 or 0 sent by the source which is computed by the receiver.

➢ In order to minimize the average probability of symbol error between the receiver output and the symbol
emitted by the source and make the digital communication system as more reliable the following two
basic issues are considered:
• How to optimize the design of the receiver so as to minimize the average probability of symbol error.
➢ How to choose the set of signals s1(t), s2(t), ……, sM(t) for representing the symbols m1, m2, ……, mM
respectively. (M-ary symbols denoted by m1, m2, ……, mM)

GEOMETRIC REPRESENTATION OF SIGNALS:


➢ The essence of geometric representation of signals is to represent any set of M energy signals {si(t)} as
linear combinations of N orthonormal basis functions, where N < M. ø1(t), ø2(t), ……, øN(t) form
orthonormal functions or basis functions. i.e., given a set of real-valued energy signals s1(t), s2(t), ……,
sM(t) each of duration T seconds, we write

------------- (4)
where the coefficients of the expansion are defined by

-------- (5)
➢ The real-valued basis functions ø1(t), ø2(t), ……, øN(t) form an orthonormal set, by which we mean

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------------ (6)
where δij is the Kronecker delta. (6) states that
• Each basis function is normalized to have unit energy.
• Also, the basis functions ø1(t), ø2(t), ……, øN(t) are orthogonal with respect to each other over the
interval 0 ≤ t ≤ T.
𝑁
➢ For prescribed i, the set of coefficients {𝑠𝑖𝑗 }𝑗=1may be viewed as an N-dimensional signal vector,
denoted by si. The important point to note here is that the vector si bears a one-to-one relationship with
the transmitted signal si(t):
• Given the N elements of the vector si operating as input, we may use the scheme shown in Fig. 3(a)
to generate the signal si(t), which follows directly from (4). This figure consists of a bank of N
multipliers with each multiplier having its own basis function followed by a summer. The scheme of
Fig. 3(a) may be viewed as a synthesizer.
• Conversely, given the signals si(t), i = 1, 2, ……, M, operating as input, we may use the scheme
shown in Fig. 3(b) to calculate the coefficients s1(t), s2(t), ……, sM(t) which follows directly from
(5). This second scheme consists of a bank of N product-integrators or correlators with a common
input and with each one of them supplied with its own basis function. The scheme of Fig. 3(b) may
be viewed as an analyzer.

Fig. 3: (a) Synthesizer for generating the signal si(t) (b) Analyzer for reconstructing the signal vector {si}

➢ Accordingly, we may state that each signal in the set {si(t)} is completely determined by the signal
vector

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-------- (7)
➢ If we conceptually extend our conventional notion of two and three-dimensional Euclidean spaces to an
N-dimensional Euclidean space, we may visualize the set of signal vectors {si|i = 1, 2, …., M} as
defining a corresponding set of M points in an N-dimensional Euclidean space, with N mutually
perpendicular axes labeled ø1, ø2, ……, øN. This N-dimensional Euclidean space is called the signal
space.
➢ It provides the mathematical basis for the geometric representation of energy signals in a conceptually
satisfying manner. This form of representation is illustrated in Fig. 4 for the case of a two-dimensional
signal space with three signals i.e., N = 2 and M = 3.
➢ In an N-dimensional Euclidean space, we may define lengths of vectors and angles between vectors. It is
customary to denote the length (also called the absolute value or norm) of a signal vector s i by the
symbol ||si||. The squared length of any signal vector si is defined to be the inner product or dot product
of si with itself, as shown by

--------- (8)
th
where sij is the j element of si and the superscript T denotes matrix transposition.

Fig. 4: Illustrating the geometric representation of signals for the case when N = 2 and M = 3

➢ By definition, the energy of a signal si(t) of duration T seconds is given by

------- (9)
Therefore, substituting (4) into (9), we get
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➢ Interchanging the order of summation and integration, as both linear operations and then rearranging
terms we get

--------- (10)
➢ Since, by definition, the øj(t) form an orthonormal set-in accordance with the two conditions of (6), we
find that (10) reduces simply to

---------- (11)
(8) and (11) show that the energy of an energy signal si(t) is equal to the squared length of the
corresponding signal vector si(t).
➢ In the case of a pair of signals si(t) and sk(t) represented by the signal vectors si and sk, respectively, we
may also show that

----------- (12)
(12) states that “The inner product of the energy signals si(t) and sk(t) over the interval [0, T] is equal to
the inner product of their respective vector representations si and sk.”
𝑁
➢ Note that the inner product siTsk is invariant to the choice of basis functions, {ø𝑗 (𝑡)}𝑗=1 in that it only
depends on the components of the signals si(t) and sk(t) projected onto each of the basis functions.
➢ Another useful relation involving the vector representations of the energy signals si(t) and sk(t) is
described by

------------ (13)
where ||si - sj|| is the Euclidean distance dik between the points represented by the signal vectors si and sk.
➢ To complete the geometric representation of energy signals, we need to represent for the angle θ ik
subtended between two signal vectors si and sk. By definition, the cosine of the angle θik is equal to the
inner product of these two vectors divided by the product of their individual norms, as shown by

---------- (14)
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The two vectors si and sk are thus orthogonal or perpendicular to each other if their inner
product is zero, in which case θik = 90°.

THE SCHWARZ INEQUALITY:


Consider any pair of energy signals s1(t) and s2(t). The Schwarz inequality states that:

------------- (15)
The equality holds if and only if, s2(t) = c.s1(t), where c is any constant.
SOLUTION:
➢ To prove this important inequality, let s1(t) and s2(t) be expressed in terms of the pair of orthonormal
basis functions ø1(t) and ø2(t) as follows:

where ø1(t) and ø2(t) satisfy the orthonormality conditions over the time interval (-ꝏ, ꝏ):

➢ On this basis, we may represent the signals s1(t) and s2(t) by the following respective pair of vectors, as
illustrated in Fig. 5.

Fig. 5: Vector representations of signals s1(t) and s2(t) providing the background picture for proving the
Schwarz inequality

➢ From Fig. 5 we see that the cosine of angle θ subtended between the vectors s1 and s2 is

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---------- (16)
➢ Considering |cosθ| ≤ 1, the Schwarz inequality of (15) immediately follows from (16). Also, from the
first line of (16) we note that | cosθ | = 1 if and only if, s2 = c.s1 where c is an arbitrary constant. Thus,

Hence proved.

NOTE:
Schwarz inequality applies to real-valued signals. It may be readily extended to complex-valued signals, in
which case (15) is reformulated as

----- (17)
where the asterisk denotes complex conjugation and the equality holds if and only if, s2(t) = c.s1(t), where c
is a constant.

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MODULE - 3

SYLLABUS:
Digital Modulation Techniques: Phase shift Keying techniques using coherent detection: generation,
detection and error probabilities of BPSK and QPSK, M–ary PSK, M–ary QAM
Frequency shift keying techniques using Coherent detection: BFSK generation, detection and error
probability
Non coherent orthogonal modulation techniques: BFSK, DPSK Symbol representation, Block diagrams
treatment of Transmitter and Receiver, Probability of error (without derivation of probability of error
equation)
TEXT, REFERENCE & ADDITIONAL REFERENCE BOOKS:

BOOK TITLE/AUTHORS/PUBLICATION

Simon Haykin, “Digital Communication Systems”, John Wiley & sons, First Edition,
T-1
2014, ISBN 978-0-471-64735-5.

B.P.Lathi and Zhi Ding, “Modern Digital and Analog communication Systems”,
AR-1
Oxford University Press, 4th Edition, 2010, ISBN: 978-0-198-07380-2.

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

INTRODUCTION
➢ When it is required to transmit digital data over a bandpass channel, it is necessary to modulate the
incoming data onto a carrier wave with fixed frequency limits imposed by the channel.
➢ The data may represent digital computer or PCM waves generated by digitizing voice or video signals.
The channel may be a Telephone channel, Microwave radio link, Satellite channel or an Optical fiber.
➢ The different types of basic modulation schemes used in transmission of digital data are Amplitude Shift
Keying (ASK), Frequency Shift Keying (FSK) and Phase Shift Keying (PSK).
➢ For data transmission at higher rates through bandlimited channels, combined modulation schemes are
also used. Usually Amplitude Phase Keying (APK) and Quadrature Amplitude Modulation (QAM) are
used.

DIGITAL MODULATION AND DEMODULATION SCHEMES


➢ In Digital Communications, the digital data is usually in the form of binary. It can also be block n binary
bits encoded into M = 2n possible discrete amplitudes or M discrete phases or M discrete frequencies.
Such schemes are generally called as M-ary Signaling. M = 2 corresponds to binary with symbols being
0 or 1. The three basic binary modulation schemes are:

❖ ASK
• In ASK, the amplitude of the carrier will have A volts for binary symbol 1 and the value zero for
symbol 0. It is also called as ON-OFF Keying.
• The information resides only in the Amplitude of the carrier signal.

❖ FSK
• Here the carrier amplitude is fixed but the frequency has a value f0 Hz for symbol 0 and a different
value f1 Hz for symbol 1. The instantaneous frequency of the carrier switches between f0 Hz and f1
Hz corresponding to symbols 0 and 1.
• Here the information resides in the frequency of carrier.

❖ PSK
• If the carrier amplitude and frequency are fixed but the phase has a value zero radians for symbol 1
and π radians for symbol 0. The modulation process is called PSK or BPSK. (Binary PSK)
• Here the information resides in the phase of the carrier.

➢ Demodulation can be done using either a Coherent Reciever or Non-Coherent Reciever. The
demodulation process then respectively called Coherent Detection and Non-Coherent Detection.
➢ In the case of Coherent detection, synchronized local reference of the transmitter signal must be
available i.e., the receiver should have the exact knowledge of the carrier waves phase reference. The
receiver is said to be phase locked to the transmitter.
➢ In Non-Coherent detection, the knowledge of the phase of carrier is not needed and hence the
complexity of the receiver is reduced.
➢ The choice of modulation schemes is made on the following desirable requirements:
• Maximum Data Rate
• Minimum Transmitter Power
• Minimum Channel Bandwidth
• Minimum Immunity to interfering signals

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

• Minimum Circuit Complexity and hence minimum cost


• Minimum probability of Symbol error

Fig.1: Waveforms for ASK, FSK and PSK


NOTE:
In general, representation of signal in terms of energy is done by
s(t) = m(t) c(t)
where m(t) is 1/0
c(t) = Ac cos 2πfct
i.e. s(t) = Ac cos 2πfct

𝐴 2 𝐴2𝑐
Average Power of Ac cos 2πfct = ( 𝑐 ) = =P
√2 2

𝐸
But P = E/T = joules/seconds = 𝑇𝑏 for symbol duration 0 ≤ t ≤ Tb
𝑏
𝐴2𝑐 𝐸𝑏 2𝐸
Equating = or Ac = √ 𝑇 𝑏
2 𝑇𝑏 𝑏

𝟐𝑬
⸫ s(t) = √ 𝑻 𝒃 cos 2πfct
𝒃

Since bit duration is fixed, hence fc, the carrier frequency is fixed.
s(t) can be represented in terms of basis function as,
2
s(t) = √𝐸𝑏 √𝑇 cos 2πfct
𝑏

s(t) = √𝐸𝑏 ø1(t)


2
where ø1(t) = √𝑇 cos 2πfct
𝑏

Similarly, if there are two carrier frequencies, the basis functions are required are two.

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

BINARY PHASE SHIFT KEYING USING COHERENT DETECTION

SIGNAL REPRESENTATION:
➢ In a binary PSK system, the pair of signals s1(t) and s2(t) used to represent binary symbols 1 and 0,
respectively, is defined by
2𝐸
s1(t) = √ 𝑇 𝑏 cos 2πfct ; 0 ≤ t ≤ Tb, for Symbol 1 ---------- (1)
𝑏

2𝐸
s2(t) = √ 𝑇 𝑏 cos (2πfct + π)
𝑏

2𝐸
s2(t) = -√ 𝑇 𝑏 cos 2πfct ; 0 ≤ t ≤ Tb, for Symbol 0 --------- (2)
𝑏

where Tb is the bit duration and Eb is the transmitted signal energy per bit.

➢ The carrier frequency fc is chosen equal to nc/Tb for some fixed integer nc. A pair of sinusoidal waves
that differ only in a relative phase-shift of 180° defined in (1) and (2), is referred to as an antipodal
signal.

SIGNAL SPACE DIAGRAM:


➢ From the pair of equations (1) and (2), it is clear that, in the case of binary PSK, there is only one basis
function of unit energy:
2
ø1(t) = √𝑇 cos 2πfct ; 0 ≤ t ≤ Tb ----------- (3)
𝑏
➢ The transmitted signals s1(t) and s2(t) is expressed in terms of ø1(t) as
s1(t) = √𝐸𝑏 ø1(t) --------- (4)
s2(t) = -√𝐸𝑏 ø1(t) -------- (5)
➢ A binary PSK system is, therefore, characterized by having a signal space that is one-dimensional (i.e., N
= 1), with a signal constellation consisting of two message points (i.e., M = 2). The respective
coordinates of the two message points are:
𝑇
s11 = ∫0 𝑏 s1(t) ø1(t) dt = √𝐸𝑏

𝑇
s21 = ∫0 𝑏 s2(t) ø1(t) dt = -√𝐸𝑏
➢ The signal space diagram for binary PSK is shown in Fig. 2

Fig.2: Signal space diagram for Binary PSK


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➢ The decision rule may be applied by setting up the appropriate decision regions. The decision boundary
is decided by taking the average between the two message points. If the received signal point lies in the
region Z1, then symbol 1 is decided. If received signal point lies in the region Z2, decision is made as
symbol 0.

BLOCK DIAGRAM:
➢ The block diagram for generating binary PSK follows directly from equation (3) to (5) which is as
shown in Fig. 3

Fig. 3: Block diagram for binary PSK Transmitter

➢ The Transmitter consists of two components:


• Polar NRZ-level encoder which represents symbols 1 and 0 of the incoming binary sequence by
amplitude levels +√𝐸𝑏 and -√𝐸𝑏 respectively.
• Product modulator which multiplies the output of the Polar NRZ encoder by the basis ø1(t).
➢ To make an optimum decision on the received signal x(t) in favor of symbol 1 or symbol 0, the receiver
is synchronized with the transmitter basis function ø1(t) as shown in Fig. 4

Fig. 4: Block diagram for Coherent binary PSK Receiver

➢ The two basic components in the binary PSK receiver:


• Correlator which correlates the received signal x(t) with the basis function ø1(t) on a bit-by-bit basis.
• Decision device which compares the correlator output against a zero-threshold, assuming that binary
symbols 1 or 0 are equiprobable. If the threshold is exceeded, a decision is made in favor of symbol
1, if not, the decision is made in favor of symbol 0.

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

ERROR PROBABILITY/PROBABILITY OF ERROR OF BINARY PSK USING COHERENT


DETECTION:
To calculate the probability of making an error the decision region associated with symbol 1 or signal s1(t) is
described by
Z1: 0 < x1 < ꝏ
STEP 1:
➢ The received signal x(t) is
x(t) = s(t) + w(t)
where w(t) is Additive White Gaussian Noise
➢ For BPSK,
s(t) = +√𝐸𝑏 ø1(t) ; for symbol 1; 0 ≤ t ≤ Tb
= −√𝐸𝑏 ø1(t) ; for symbol 0; 0 ≤ t ≤ Tb
⸫ x(t) = s1(t) + w(t) = +√𝐸𝑏 + w(t) ; for symbol 1
= s2(t) + w(t) = −√𝐸𝑏 + w(t) ; for symbol 0

STEP 2:
➢ For White Gaussian Noise, the mean and variance are 0 and N0/2 respectively. Assuming symbol 1 is
transmitted, the mean of the received signal is
µ = E [x1] = E [s1] + E [wch] = √𝐸𝑏 ; E [wch] = 0
➢ Similarly,
σ2 = var [x1] = var [s1] + var [wch] = N0/2 ; var [s1] = 0

STEP 3:
➢ The conditional probability density function of random variable X1 given that symbol 1 is transmitted is
defined by

1 −(𝑥1 −µ)2
𝑓𝑋1 (x1/1) = √2𝜋𝜎2 exp [ ]
2𝜎2

1 −(𝑥1 −√𝐸𝑏 )2
= exp [ 𝑁 ]
𝑁
√2𝜋( 0 ) 2( 0 )
2
2

1 −(𝑥1 −√𝐸𝑏 )2
= exp [ ]
√𝜋𝑁0 𝑁0

2
1 (𝑥1 −√𝐸𝑏 )
= exp [− ( ) ]
√𝜋𝑁0 √𝑁0

STEP 4:
➢ The conditional probability of the Reciever deciding in favour of symbol 0 given that the symbol 1 was
transmitted is therefore
0
Pe (1) = ∫−∞ 𝑓𝑋1 (𝑥1 /1) 𝑑𝑥1

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

2
√2(𝑥1 −√𝐸𝑏 ) 1
0 1 [−( ) ]
√𝑁0 2
= ∫−∞ √𝜋𝑁 𝑒 𝑑𝑥1
0

2
➢ Let √𝑁 (𝑥1 − √𝐸𝑏 )= y
0

2
dy = 𝑑𝑥1 √𝑁
0

𝑁
𝑑𝑥1 = dy√ 20
➢ If x1 = -ꝏ ; y = -ꝏ
2𝐸
x1 = 0 ; y = −√ 𝑁 𝑏
0

2𝐸𝑏 2 1
−√ 1 (−[√ (𝑥 −√𝐸𝑏 )]2 ) 𝑁
𝑁0 1
⸫ Pe (1) = ∫−∞ 𝑁0
𝑒 2
𝑑𝑦 √ 20
√𝜋𝑁0

2𝐸𝑏 2 1
1 −√ (−[√ (𝑥 −√𝐸𝑏 )]2 )
𝑁0 1 2
= ∫ 𝑁0
𝑒 𝑑𝑦
√2𝜋 −∞
−𝑦 2
➢ Since exponential function 𝑒 is symmetry about y = 0, the integral in above equation can be written
as
2𝐸𝑏
1 −√ 2 /2
Pe (1) = ∫ 𝑁0 𝑒 (−𝑦)
√2𝜋 −∞
𝑑𝑦

2𝐸
Pe (1) = Q (√ 𝑁 𝑏 )
0

or

1 𝐸
Pe (1) = 2 erfc (√𝑁𝑏 )
0

➢ Similarly, for Pe (0) the conditional probability of the receiver deciding in favour of symbol 1 given that
Symbol 0 is transmitted is given by
1 𝐸
Pe (0) = 2 erfc (√𝑁𝑏 )
0

STEP 5:
➢ Total Average Probability of error is given by
Pe = Pe (0) P (0) + Pe (1) P (1)

1 1 𝐸 1 1 𝐸
=2 erfc (√𝑁𝑏 ) + 2 erfc (√𝑁𝑏 )
2 0 2 0

𝟏 𝑬
Pe = 𝟐 erfc (√𝑵𝒃 ) is error for BPSK
𝟎

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

➢ Thus, we find that the average probability of Symbol error or equivalent BER for binary PSK using
coherent detection, assuming equiprobable symbols given by
𝟐𝑬
Pe = Q (√ 𝑵 𝒃)
𝟎

QUADRIPHASE SHIFT KEYING


SIGNAL REPRESENTATION:
➢ It is one of the Bandwidth conserving scheme.
➢ Here the phase of the carrier takes on one of four equally spaced values such as π/4, 3π/4, 5π/4 and 7π/4.
➢ The Transmitted signal s(t) is defined as
2E 𝜋
s(t) = { √ T cos(2𝜋𝑓𝑐 𝑡 + (2𝑖 − 1) 4 ) ; 0 ≤ t ≤ T ---------------- (1)
0 ; Elsewhere

where i = 1, 2, 3, 4
E is Transmitted signal energy per symbol
T is Symbol duration
➢ Each possible value of phase corresponds to unique dibit (pair of bits). For example, we may choose the
set of phase values to represent Gray Code and Natural Code as shown in Table 1.

SIGNAL SPACE DIAGRAM:


➢ Expanding equation (1) using cos (a+b) = cos a cos b – sin a sin b, we get
2E 𝜋 2E 𝜋
s(t) = √ T cos(2𝑖 − 1) 4 cos2πfct - √ T sin(2𝑖 − 1) 4 sin2πfct ---------- (2)

INPHASE COMPONENT QUADRATURE PHASE COMPONENT


where i = 1, 2, 3, 4
➢ Based on this representation we make two observations:
1. There are two orthogonal basis functions defined by a pair of Quadrature Carriers.
2
ø1(t) = √𝑇 cos 2πfct ; 0 ≤ t ≤ T

2
ø2(t) = √𝑇 sin 2πfct ; 0 ≤ t ≤ T

2. The co-ordinates of message points are given by


𝜋 𝜋
si(t) = [√𝐸 cos [(2𝑖 − 1) 4] - [√𝐸 sin [(2𝑖 − 1) 4] --------- (3)
where i = 1, 2, 3, 4
➢ Elements of the signal vectors namely 𝑠𝑖1 and 𝑠𝑖2 have their values summarized as in Table 1.
i si(t) 𝐬𝐢𝟏 𝐬𝐢𝟐 Dibit Phase for Gray Code Phase for Natural Code
1 s1(t) √𝐸/2 −√𝐸/2 10 7π/4 π/4
2 s2(t) −√𝐸/2 −√𝐸/2 00 5π/4 3π/4
3 s3(t) −√𝐸/2 √𝐸/2 01 3π/4 5π/4
4 s4(t) √𝐸/2 √𝐸/2 11 π/4 7π/4
Table 1: Signal Space characterization of QPSK

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

➢ Thus, QPSK is a two-dimensional constellation (N = 2) and four message points (M = 4). The signal
space diagram is shown in Fig. 5

Fig. 5: Signal space diagram of QPSK

➢ The input binary sequence 01101000 is shown in Fig. 6(a). This sequence is divided into two other
sequences, consisting of odd- and even-numbered bits of the input sequence. These two sequences are
shown in the top lines of Fig. 6(b) and 6(c).
➢ The waveforms representing the two components of the QPSK signal, namely 𝑠𝑖1 ø1(t) and 𝑠𝑖2 ø2(t) are
also shown in 6(b) and 6(c), respectively. These two waveforms may individually be viewed as
examples of a binary PSK signal. Adding them, we get the QPSK waveform shown in Fig. 6(d)

Fig. 6: (a) Input binary sequence (b) Odd-numbered dibits of input sequence and associated binary PSK
signal (c) Even-numbered dibits of input sequence and associated binary PSK signal (d) QPSK waveform
defined as s(t) = 𝑠𝑖1 ø1(t) + 𝑠𝑖2 ø2(t)

GENERATION AND COHERENT DETECTION OF QPSK:


➢ Block diagram for QPSK Transmitter and QPSK Receiver is shown in Fig. 7(a) and Fig. 7(b)
respectively.

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

Fig. 7: Block diagram of (a) QPSK transmitter and (b) Coherent QPSK receiver

➢ The demultiplexer in the QPSK transmitter divide the binary wave produced by the polar NRZ-level
encoder into two separate binary waves, one of which represents the odd-numbered dibits in the
incoming binary sequence and the other represents the even-numbered dibits.
➢ QPSK transmitter may be viewed as two binary PSK generators that work in parallel, each at a bit rate
equal to one-half the bit rate of the original binary sequence at the QPSK transmitter input.
➢ The functional composition of the QPSK receiver is as follows:
1. Pair of correlators, which have a common input x(t). The two correlators are supplied with a pair of
locally generated orthonormal basis functions ø1(t) and ø2(t), which means that the receiver is
synchronized with the transmitter. The correlator outputs, produced in response to the received signal
x(t), are denoted by x1 and x2, respectively.
2. Pair of decision devices, which act on the correlator outputs x1 and x2 by comparing each one with a
zero-threshold; here, it is assumed that the symbols 1 and 0 in the original binary stream at the
transmitter input are equally likely. If x1 > 0, a decision is made in favor of symbol 1 for the in-phase
channel output. If x1 < 0, then a decision is made in favor of symbol 0. Similar binary decisions are
made for the Quadrature channel.
3. Multiplexer, the function of which is to combine the two binary sequences produced by the pair of
decision devices. The resulting binary sequence so produced provides an estimate of the original
binary stream at the transmitter input.

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

ERROR PROBABILITY OF QPSK:


STEP 1:
➢ The received signal x(t) = s(t) + w(t)
➢ Let xi = si + w, assuming symbol 4 is transmitted where i = 1, 2, 3, 4
➢ If symbol s4 is transmitted than
x1 = √E/2 + w and x2 = √E/2 + w

STEP 2:
➢ Mean is given by
µ1 = E [x1] = √E/2 and µ2 = E [x2] = √E/2
➢ Variance for both x1 and x2 is given by
σ2 = N0/2

STEP 3:
➢ Conditional Probability Density function
−(𝑥1 −µ)2
1
𝑓𝑋1 (x1/ s4) = √2𝜋𝜎2 𝑒 2𝜎2

−(𝑥1 −√𝐸/2 )2
1
= 𝑒 𝑁0
√𝜋𝑁0

2
(𝑥 −√𝐸/2 )
1 −( 1 )
𝑓𝑋1 (x1/ s4) = 𝑒 √𝑁0 ------------(4)
√𝜋𝑁0
➢ Similarly
2
(𝑥 −√𝐸/2 )
1 −( 2 )
𝑓𝑋2 (x2/ s4) = 𝑒 √𝑁0 ------------(5)
√𝜋𝑁0

STEP 4:
➢ When signal s4(t) is transmitted, the received signal point lies in the decision region Z4. If x1 > 0 and x2 >
0 leading to correct decision. Thus, the probability of correct decision Pc [s4(t)] when signal s4(t) is
transmitted is given by
Pc [s4(t)] = P (x1 > 0 and x2 > 0)
➢ Since the random variables X1 and X2 are statistically independent, their joint probability is equal to the
product of individual probabilities. Thus, when s4(t) is transmitted the probability of correct decision is
Pc [s4(t)] = P (x1 > 0) P (x2 > 0)
∞ ∞
= [∫0 𝑓𝑋1 (𝑥1 / 𝑠4 ) 𝑑𝑥1 ] [∫0 𝑓𝑋2 (𝑥2 / 𝑠4 ) 𝑑𝑥2 ]
2 2
(𝑥1 −√𝐸/2 ) (𝑥2 −√𝐸/2 )
1 ∞ −( ) 1 ∞ −( )
= [ ∫ 𝑒 √𝑁0 𝑑𝑥1 ] [ ∫ 𝑒 √𝑁0 𝑑𝑥2 ]
√𝜋𝑁0 0 √𝜋𝑁0 0

Modifying the terms by multiplying and dividing by 2, we get

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

2 2
2 𝐸 1 2 𝐸 1
(− (√ (𝑥1 −√ )) ) (− (√ (𝑥2 −√ )) )
1 ∞ 𝑁0 2 2 1 ∞ 𝑁0 2 2
Pc [s4(t)] =
√𝜋𝑁0
∫0 𝑒 𝑑𝑥1
√𝜋𝑁0
∫0 𝑒 𝑑𝑥2 ------------ (6)
[ ][ ]
2 𝐸 2 𝐸
➢ Let √𝑁 (𝑥1 − √ 2 ) = √𝑁 (𝑥2 − √2 ) = z
0 0

2 𝑁
⸫ dz = 𝑑𝑥1 √𝑁 or 𝑑𝑥1 = √ 20 dz = 𝑑𝑥2
0

𝐸
Also, when 𝑥1 = 0 ; z = −√𝑁
0

𝑥1 = ꝏ ; z = ꝏ
Equation (6) reduces to,
2
1 ∞ (− 𝑧 2 /2) 𝑁0
Pc = [ ∫ 𝐸 𝑒 √ dz ]
√𝜋𝑁0 −√ 2
𝑁0
2
1 ∞ (− 𝑧 2 /2)
=[ ∫ 𝐸 𝑒 dz ]
√2𝜋 −√
𝑁0

➢ From Complementary Error function,


2
𝐸
Pc = [1 − 𝑄 [√𝑁 ]]
0

Expanding,
𝐸 𝐸
Pc = 1 + 𝑄 2 [√𝑁 ] – 2 𝑄 [√𝑁 ]
0 0
𝐸 2
In this region, when 𝑁 >> 1, we may ignore 𝑄 term
0
𝐸
Pc = 1 - 2 𝑄 [√𝑁 ]
0

STEP 5:
➢ The average probability of symbol error for QPSK is
Pe = 1 - P c
𝐸
= 1 – (1 – 2 𝑄 [√ ])
𝑁0

𝐸
Pe = 2 𝑄 [√𝑁 ]
0

➢ In QPSK system, there are 2 bits per symbol and hence the transmitted signal energy per symbol is twice
the transmitted energy per bit i.e., E = 2 Eb. Hence, the average probability of symbol error is
2𝐸
P e = 2 𝑄 [√ 𝑁 𝑏 ]
0

➢ Hence, BER for QPSK is given by


𝟐𝑬
BER = 𝑸 [√ 𝑵 𝒃 ]
𝟎

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

OFFSET QPSK (O – QPSK):


➢ The 180˚ and 90˚ phase shifts in the carrier phase of QPSK can result in changes in the carrier amplitude
i.e., envelope of the QPSK signal during the cause of transmission over the channel thereby causing
additional symbol errors on detection at the receivers.
➢ To overcome the shortcoming of QPSK, we need to reduce the amplitude fluctuations. Thus, we may use
offset QPSK.
➢ In offset QPSK, the bit stream responsible for generating the Quadrature Components is delayed (i.e.,
offset) by half a symbol interval with respect to the bit stream responsible for generating the In phase
component.
➢ The two basis functions of offset QPSK are defined by
2
ø1(t) = √ cos 2πfct ; 0 ≤ t ≤ T
𝑇

2
ø2(t) = √𝑇 sin 2πfct ; T/2 ≤ t ≤ 3T/2
➢ Unlike QPSK, the phase transitions likely to occur in offset QPSK are confined to ±90˚ as indicated in
signal space diagram. We find that amplitude fluctuations in offset QPSK due to filtering having a
smaller amplitude than in case of QPSK.

Fig. 8: Possible paths for switching between the message points in (a) QPSK and (b) offset QPSK.

NOTE:
Despite the delay T/2 applied to basis function ø2(t), compared with the probability of error the O - QPSK
has some probability error as that of QPSK.
M – ary PSK:
➢ QPSK is the form of PSK commonly referred to as M-ary PSK (MPSK). Here, the phase of the carrier
takes on one of M possible values θ1 = 2 (i - 1) π/M where i = 1, 2, …. M
➢ One of M possible signals can be represented by
2𝐸 2π
si(t) = √ 𝑇 cos [2π𝑓𝑐 t + (𝑖 − 1)] ------- (1)
𝑀
where i = 1, 2, …. M
E = signal energy per symbol
Also, The carrier frequency fc = nc/T for some fixed integer nc
➢ Each si(t) may be expanded in terms of the same two basis functions ø1(t) and ø2(t), therefore the signal
constellation is of two-dimensional. (In QPSK, M = 4 and N = 2)

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

➢ The M message points are equally spaced on a circle of radius √𝐸 and center at the origin, as illustrated
in Fig. 9. This is called Octa phase shift-keying (i.e., M = 8). We see that signal space diagram is
Circularly Symmetric.

Fig. 9: (a) Signal-space diagram for Octa phase shift-keying (i.e., M = 8). The decision boundaries are
shown as dashed lines. (b) Signal-space diagram illustrating the application of the union bound for Octa
phase shift-keying.

CHANNEL BANDWIDTH:
➢ Channel Bandwidth required to pass M – ary PSK signals through an Analog Channel as
B = 2/T ------(1)
where T = Tb is a Symbol duration
But Bit rate Rb = 1/Tb = loge M/ Tb
➢ Equation (1) can be written as
B = 2 Rb / loge M
➢ The Bandwidth efficiency of M – ary PSK signal is given by
ρ = Rb / B = log2 M/2
➢ Table 2 gives Bandwidth efficiency of M – ary PSK signal.

M 2 4 8 16 32 64
ρ (bits/(s/Hz)) 0.5 1 1.5 2 2.5 3
Table 2: Bandwidth efficiency of M – ary PSK signal

➢ Thus, as the number of states in M – ary PSK is increased, the bandwidth efficiency is improved at the
expense of error performance.

M – ary QUADRATURE AMPLITUDE MODULATION (M – QAM):


➢ In an M-ary PSK system, the in-phase and quadrature components of the modulated signal are
interrelated in such a way that the envelope is constrained to remain constant. However, if this constraint
is removed so as to permit the in-phase and quadrature components to be independent, we get a new
modulation scheme called M-ary QAM. The QAM is a hybrid form of modulation.

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

➢ In M-ary PAM, the signal-space diagram is one-dimensional whereas M-ary QAM is a two-dimensional
generalization of M-ary PAM. It involves two orthogonal passband basis functions:
2
ø1(t) = √𝑇 cos 2πfct ; 0 ≤ t ≤ T
---------------- (1)
2
ø2(t) = √𝑇 sin 2πfct ; 0 ≤ t ≤ T
➢ Let dmin denote the minimum distance between any two message points in the QAM constellation. Then,
the projections of the ith message point on the ø1 and ø2 axes are respectively defined by ai.dmin/2 and
bi.dmin/2, where i = 1, 2, …. M. With the separation between two message points in the signal-space
diagram being proportional to the square root of energy, we may therefore set
dmin/2 = √𝐸0 ---------(2)
where E0 is the energy of the message signal with lowest amplitude.
➢ M-ary QAM signal for symbol k in terms of E0 is defined as
2𝐸0 2𝐸0
sk(t) = √ ak cos (2πfct) - √ bk sin (2πfct) -------- (3)
𝑇 𝑇
where 0 ≤ t ≤ T and k = 0, ±1, ±2, …
➢ In M-ary QAM, the constellation of message points depends on the number of possible symbols, M. For
illustration, square constellations are considered for which the number of bits per symbol is even.

QAM SQUARE CONSTELLATIONS:


➢ With an even number of bits per symbol, we write
L = √𝑀 ; L is positive integer
➢ Under this condition, an M-ary QAM square constellation can always be viewed as the Cartesian product
of a one-dimensional L-ary PAM constellation with itself.
➢ The ordered pairs of coordinates form a square matrix as shown by

AVERAGE PROBABILITY OF ERROR:


➢ The average probability of error of the M-ary QAM can be determined by proceeding as follows:
• The probability of correct detection for M -ary QAM is written as
Pc = (1 − 𝑃𝑒′ )2 --------- (1)

where 𝑃𝑒 is the probability of symbol error for the L – ary PAM

• With L = √𝑀 , the probability of symbol error 𝑃𝑒′ is itself defined by


1 2𝐸
𝑃𝑒′ = 2 (1 − ) Q(√ 𝑁 0 ) -------- (2)
√𝑀 0

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

• The probability of symbol error for many M – ary QAM is given by


Pe = 1 - P c
= 1 - (1 − 𝑃𝑒′ )2
Pe ≈ 2 𝑃𝑒′ ------------ (3)
where it is assumed that 𝑃𝑒′ is small enough compared with unity to justify ignoring the quadratic term.
➢ Hence, using equations (1) and (2) in (3), we find that the probability of symbol error for M-ary QAM is
approximately given by
1 2𝐸
Pe ≈ 4 (1 − ) Q(√ 𝑁 0 ) ------- (4)
√𝑀 0

➢ The transmitted energy in M-ary QAM is variable, its instantaneous value depends on the particular
symbol transmitted. Therefore, Pe is expressed in terms of the average value of the transmitted energy
rather than E0.
1 2𝐸
𝑎𝑣
i.e., Pe = 4 (1 − ) Q(√(𝑀−1)𝑁 ) --------(5)
√𝑀 0

➢ For M = 4 in equation (5) and Eav = E, we find that the resulting formula for probability of symbol error
becomes identical to that in QPSK.

FREQUENCY SHIFT KEYING USING COHERENT DETECTION:


M - ary PSK and M - ary QAM are examples of linear modulation whereas FSK is a nonlinear modulation.

BINARY FSK (BFSK): (M = 2)


➢ In binary FSK, the signals s1(t) and s2(t) represent symbols 1 and 0 respectively are two sinusoidal
signals having two distinct frequencies i.e., for
2𝐸
Symbol 1 : s1(t) = √ 𝑇 𝑏 cos 2πf1t
𝑏

for 0 ≤ t ≤ T --------- (1)


2𝐸
Symbol 0 : s2(t) = √ 𝑇 𝑏 cos 2πf2t
𝑏

➢ In general, a typical pair of sinusoidal waves is described by


2𝐸𝑏
√ cos 2πf𝑖 t ; 0 ≤ 𝑡 ≤ 𝑇𝑏
𝑠𝑖 (t) = { 𝑇𝑏
0 ; 𝐸𝑙𝑠𝑒𝑤ℎ𝑒𝑟𝑒
𝑛𝑐 +1
where i = 1, 2 and Eb is transmitted signal energy per bit. Here, transmitted frequency is set at fi = 𝑇𝑏
for some fixed integer 𝑛𝑐 and i = 1, 2
➢ The FSK signal described here is known as Sunde’s FSK. It is a continuous-phase signal. The most
useful orthonormal basis function is described by
2
ø1(t) = √𝑇 cos 2πf1t ; 0 ≤ 𝑡 ≤ 𝑇𝑏
𝑏

2
ø2(t) = √ cos 2πf2t ; 0 ≤ 𝑡 ≤ 𝑇𝑏
𝑇𝑏
➢ The coefficients sij where i = 1, 2 and j = 1, 2 is defined by
𝑇
sij = ∫0 𝑏 𝑠𝑖 (𝑡) ø𝑗 (t) 𝑑𝑡
Dept. of ECE, CEC, Mangalore Page 15
MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

𝑇 2𝐸 2
sij = ∫0 𝑏 √ 𝑇 𝑏 cos 2πf𝑖 t √𝑇 cos 2πf𝑗 t 𝑑𝑡
𝑏 𝑏

which implies that sij = √𝐸𝑏 ; i = j


=0 ;i≠j

Hence, s1(t) = √𝐸𝑏 ø1(t) and s2(t) = √𝐸𝑏 ø2(t)


➢ It is seen that the signal space is two-dimensional space. The two message points are defined by the
vectors.
0
s1 = [√𝐸𝑏 ] and s2 = [ ]
0 √𝐸𝑏
➢ The Euclidean distance between two message points is equal to ǁs1 – s2ǁ which is equal to √2𝐸𝑏 . The
signal space diagram is as shown in Fig. 10

Fig. 10: Signal-space diagram for binary FSK system


➢ If the two symbols are equiprobable, then the decision boundary is a straight line passing through origin
and midpoint of the line joining the points s1 and s2. The decision regions are shown in Fig. 10.

WAVEFORM REPRESNTATION:
➢ The FSK waveform for the binary sequence 101100 is given in Fig. 11 where f1 > f2

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

Fig. 11: FSK waveform for sequence 101100


GENERATION AND COHERENT DETECTION OF BINARY FSK SIGNALS:
➢ The block diagram of Fig. 12 describes a scheme for generating the binary FSK signal. it consists of two
components:
1. On–off level encoder, the output of which is a constant amplitude of √𝐸𝑏 in response to input
symbol 1 and zero in response to input symbol 0.
2. Pair of oscillators, whose frequencies f1 and f2 differ by an integer multiple of the bit rate 1/Tb. The
lower oscillator with frequency f2 is preceded by an inverter. When in a signaling interval, the input
symbol is 1, the upper oscillator with frequency f1 is switched on and signal s1(t) is transmitted,
while the lower oscillator is switched off. On the other hand, when the input symbol is 0, the upper
oscillator is switched off, while the lower oscillator is switched on and signal s2(t) with frequency f2
is transmitted.

Fig. 12: Block diagram for binary FSK transmitter


COHERENT FSK DETECTOR:

Fig. 13: Block diagram for coherent binary FSK receiver


➢ The receiver consists of two correlators with a common input, which are supplied with locally generated
coherent reference signals ø1(t) and ø2(t).
➢ The correlator outputs are then subtracted, one from the other. the resulting difference y is then
compared with a threshold of zero.
➢ If y > 0, the receiver decides in favor of 1. On the other hand, if y < 0, it decides in favor of 0. If y is
exactly zero, the receiver makes a random guess in favor of 1 or 0.

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

ERROR PROBABILITY OF BINARY FSK:


STEP 1:
➢ The received signal x(t) = s(t) + w(t)
where x(t) = s1(t) + w(t) ; for symbol 1
= s2(t) + w(t) ; for symbol 0
➢ We define a Gaussian random variable y whose sample value y is equal to difference between x1 and x2.
i.e., y = x1 - x2
𝑇
where x1 = ∫0 𝑏 𝑥(t) ø1(t) dt
𝑇
x2 = ∫0 𝑏 𝑥(t) ø2(t) dt

STEP 2:
➢ Given that symbol 1 was sent, the Gaussian random variables X1 and X2, whose sample values are
denoted by x1 and x2 have mean values equal to √𝐸𝑏 and zero respectively.
i.e., E [y/1] = E [X1/1] - E [X2/1] = √𝐸𝑏
E [y/0] = E [X1/0] - E [X2/0] = −√𝐸𝑏
➢ var [y] = var [X1] + var [X2]
ρ2 = N0/2 + N0/2
= N0 [⸪ X1 and X2 are statistically independent]

STEP 3:
➢ Suppose we know that, the symbol 0 was sent. The conditional probability density function of random
variable Y is given by
1 −(𝑦+√𝐸𝑏 )2
fY(y/0) = exp [ ]
√2𝜋𝑁0 2𝑁0

STEP 4:
➢ The conditional probability of error given that symbol 0 was sent is

Pe (0) = ∫0 f𝑌 (y/0) dy
−(𝑦+√𝐸𝑏 )2
∞ 1 [ ]
2𝑁0
= ∫0 √2𝜋𝑁 𝑒 dy
0
2
1 1
(𝑦+√𝐸𝑏 ))
1 ∞ −(√𝑁0 2
= ∫ 𝑒 dy
√2𝜋𝑁0 0
1
➢ Let z = (𝑦 + √𝐸𝑏 )
√𝑁0
𝑑𝑦
⸫ dz = or dy = dz √𝑁0
√𝑁0
𝐸
y = 0 ; z = √𝑁𝑏
0

y=ꝏ;z=ꝏ
1 ∞ 2 /2
⸫Pe (0) = ∫ 𝐸 𝑒 −(𝑧) √𝑁0 dz
√2𝜋𝑁0 √ 𝑏
𝑁0

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

1 ∞ 2 /2
= ∫ 𝑒 −(𝑧) dz
√2𝜋 √𝐸𝑏
𝑁0

𝐸
Pe (0) = Q (√𝑁𝑏 )
0

𝐸
Similarly, Pe (1) = Q (√𝑁𝑏 )
0

STEP 5:
➢ Average probability of error is
1 𝐸 1 𝐸
Pe = Q (√ 𝑏 ) + Q (√ 𝑏 )
2 𝑁0 2 𝑁0

𝑬
Pe = Q (√𝑵𝒃 )
𝟎

NON – COHERENT ORTHOGONAL MODULATION TECHNIQUES:


➢ In Coherent demodulation scheme it is required to have a local reference signal at the receiver which is
in agreement with the phase of carrier.
➢ Some demodulation schemes that do not require a local reference signal that is in synchronized with the
received carrier. Such demodulation schemes are called Non – Coherent schemes. Non – Coherent
schemes are us when knowledge of phase carrier is not available at the receiver.
➢ In Non – Coherent systems, the structure of the receiver is simple compared to coherent systems. Under
large signal to noise ratios, the performance of non – coherent receiver’s approach that of coherent
receivers.
BINARY FREQUENCY SHIFT KEYING USING NON – COHERENT DETECTION:

Fig. 14: Generalized binary receiver for noncoherent orthogonal modulation


➢ FSK transmitted signal is given by
2𝐸𝑏
cos 2𝜋𝑓𝑖 𝑡 ; 0 ≤ 𝑡 ≤ 𝑇𝑏
si(t) = {√ 𝑇𝑏
0 ; 𝐸𝑙𝑠𝑒𝑤ℎ𝑒𝑟𝑒

➢ Here receiver consists of a pair of matched filters followed by Envelope detectors. The filter in the upper
path of the receiver is matched to cos 2𝜋𝑓1 𝑡 and the filter in the lower path is matched to cos 2𝜋𝑓2 𝑡 for the
signaling interval 0 ≤ t ≤ Tb.

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

➢ The resulting envelope detector outputs are sampled at t = Tb and their values are compared. The
envelope samples of the upper and lower paths are l1 and l2.
➢ The receiver decides in favour of symbol 1 if l1 > l2 and in favour of 0 if l1 < l2. If l1 = l2, the receiver
simply guesses randomly in favour of symbol 1 or 0.
➢ The Non-Coherent binary FSK here is a special case of non-coherent orthogonal modulation with T = Tb
and E = Eb where Eb is the signal energy per bit. Hence, the BER for non-coherent binary FSK is
𝟏 −𝐄
Pe = 𝟐 exp(𝟐𝑵𝒃 )
𝟎

DIFFERENTIAL PHASE SHIFT KEYING (DPSK):


➢ DPSK is noncoherent version of binary PSK. DPSK eliminates the need for synchronizing the receiver
to the transmitter by combining two basic operations at the transmitter:
• Differential encoding of the input binary sequence
• PSK of the encoded sequence
➢ Differential encoding starts with an arbitrary first bit, serving as the reference bit, to this end, symbol 1 is
used as the reference bit. Generation of the differentially encoded sequence then proceeds in accordance
with a two-part encoding rule as follows:
• If the new bit at the transmitter input is 1, leave the differentially encoded symbol unchanged with
respect to the current bit.
• If on the other hand, the input bit is 0, change the differentially encoded symbol with respect to the
current bit.
➢ The differentially encoded sequence, denoted by {dk}, is used to shift the sinusoidal carrier phase by
zero and 180˚, representing symbols 1 and 0, respectively. Thus, in terms of phase-shifts, the resulting
DPSK signal has following phases:
• To send symbol 1, the phase of the DPSK signal remains unchanged.
• To send symbol 0, the phase of the DPSK signal is shifted by 180°.

ERROR PROBABILITY OF DPSK:


➢ DPSK is an example of non-coherent orthogonal modulation when its behaviour is considered over
successive two-bit intervals i.e., 0 ≤ t ≤ 2Tb then
2𝐸
√ 𝑇 𝑏 cos 2𝜋𝑓𝑐 𝑡 ; 𝑠𝑦𝑚𝑏𝑜𝑙 1 𝑓𝑜𝑟 0 ≤ 𝑡 ≤ 𝑇𝑏
𝑏
s1(t) = --- (1)
2𝐸𝑏
√ cos 2𝜋𝑓𝑐 𝑡 ; 𝑠𝑦𝑚𝑏𝑜𝑙 1 𝑓𝑜𝑟 𝑇𝑏 ≤ 𝑡 ≤ 2𝑇𝑏
{ 𝑇𝑏
s1(t) is represented for successive two bits which are of same symbol i.e., no phase change
2𝐸
√ 𝑇 𝑏 cos 2𝜋𝑓𝑐 𝑡 ; 𝑠𝑦𝑚𝑏𝑜𝑙 1 𝑓𝑜𝑟 0 ≤ 𝑡 ≤ 𝑇𝑏
𝑏
s1(t) = --- (2)
2𝐸𝑏
√ cos(2𝜋𝑓𝑐 𝑡 + 𝜋) ; 𝑠𝑦𝑚𝑏𝑜𝑙 0 𝑓𝑜𝑟 𝑇𝑏 ≤ 𝑡 ≤ 2𝑇𝑏
{ 𝑇𝑏
1 −𝐸
➢ For DPSK, T = 2Tb and E = 2Eb Hence using Pe = 2 exp (2𝑁 ), The average probability of error for Non
0
– Coherent receiver of DPSK is
𝟏 −𝑬
Pe = 𝟐 𝐞𝐱𝐩 ( 𝑵 𝒃)
𝟎

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

GENERATION OF DPSK SIGNAL:


➢ Fig. 15 shows the block diagram of the DPSK transmitter. The transmitter consists of two functional
blocks:
• Logic network and one-bit delay (storage) element, which are interconnected so as to convert the
raw input binary sequence {bk} into the differentially encoded sequence {dk}.
• Binary PSK modulator, the output of which is the desired DPSK signal.

Fig. 15: Block diagram of a DPSK transmitter

OPTIMUM RECEIVER FOR THE DETECTION OF DPSK:


➢ The optimum receiver for the detection of binary DPSK is as shown in Fig. 16.

Fig. 16: Block diagram of a DPSK receiver

➢ The receiver is said to be optimum for two reasons:


• In structural terms, the receiver avoids the use of fancy delay lines that could be needed otherwise.
• In operational terms, the receiver makes the decoding analysis straightforward to handle, in that the
two signals to be considered are orthogonal over the interval [0, 2Tb].

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MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]

QUESTION BANK

1. Explain BPSK transmitter and receiver with neat block diagram and also derive the equation for
error probability with signal space diagram.
2. Explain the generation and coherent detection of QPSK signals. Draw the signal constellation
diagram for QPSK and derive its probability of error.
3. Explain generation and coherent detection of BFSK. Draw the signal space diagram for BFSK and
derive its probability of error.
4. With neat diagram and expressions, explain the generation and non-coherent detection of BFSK.
5. Explain M – ary PSK and illustrate its constellation diagram for the case M = 8.
6. Sketch QPSK waveform for binary sequence 1100101110.
7. Define M-ary QAM. Obtain the constellation of QAM for M=4 and draw the signal space diagram.
8. With neat block diagrams, explain DPSK transmitter and receiver. Illustrate the generation of
differentially encoded sequence and transmission of phase for binary input sequence
{0010011001110}.

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MODULE-4 VI SEM DIGITAL COMMUNICATION [18EC61]

MODULE - 4

SYLLABUS:
Communication through Band Limited Channels: Digital Transmission through Band limited channels:
Digital PAM Transmission through Band limited Channels, Signal design for Bandlimited Channels: Design
of band limited signals for zero ISI–The Nyquist Criterion (statement only), Design of band limited signals
with controlled ISI-Partial Response signals, Probability of error for detection of Digital PAM: Probability
of error for detection of Digital PAM with Zero ISI, Symbol–by–Symbol detection of data with controlled
ISI, Channel Equalization: Linear Equalizers (ZFE, MMSE)

TEXT, REFERENCE & ADDITIONAL REFERENCE BOOKS:

BOOK TITLE/AUTHORS/PUBLICATION
John G Proakis and Masoud Salehi, “Fundamentals of Communication Systems”,
T-1
2014 Edition, Pearson Education, ISBN 978-8-131-70573-5.

B.P.Lathi and Zhi Ding, “Modern Digital and Analog communication Systems”,
AR-1
Oxford University Press, 4th Edition, 2010, ISBN: 978-0-198-07380-2.

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INTRODUCTION:
➢ Till now, we considered digital communication over an additive white Gaussian noise (AWGN) channel
and evaluated the probability of error performance of the optimum receiver for several different types of
digital modulation techniques.
➢ In this model, we treat digital communication over a channel that is modeled as a linear filter with a
bandwidth limitation. The bandlimited channels most frequently encountered in practice are telephone
channels, microwave line-of-sight (LOS) radio channels, satellite channels and underwater acoustic
channels.
➢ We will see that a linear filter channel distorts the transmitted signal. The channel distortion results in
intersymbol interference (ISI) at the output of the demodulator and leads to an increase in the probability
of error at the detector. Devices or methods for correcting or undoing the channel distortion called
Channel Equalizers.

DIGITAL TRANSMISSION THROUGH BANDLIMITED CHANNELS:


➢ A bandlimited channel such as a telephone wireline is characterized as a linear filter with impulse
response c(t) and frequency response C(f) where

--------------- (1)
➢ If the channel is a baseband channel that is bandlimited to Bc Hz, then C(f) = 0 for |f| > Bc i.e., any
frequency components at the input to the channel that are higher than Bc Hz will not be passed by the
channel.
➢ Now we consider the design of signals for transmission through the channel that are bandlimited to W =
Bc Hz as shown in Fig. 1. Henceforth, ‘W’ will denote the bandwidth limitation of the signal and the
channel.
➢ Now if the input to a bandlimited channel is a signal waveform gT(t), where the subscript T denotes that
the signal waveform is the output of the transmitter. Then, the response of the channel is the convolution
of gT(t) with c(t), i.e.,

---------------(2)
or in frequency domain,
H(f) = C(f).GT(f) ------------- (3)
where GT(f) is the spectrum (Fourier transform) of the signal gT(t) and H(f) is the spectrum of h(t). Thus,
the channel alters or distorts the transmitted signal gT(t).
➢ Let us assume that the signal at the output of the channel is corrupted by AWGN. Then, the signal at the
input to the demodulator is of the form h(t) + n(t), where n(t) denotes the AWGN.

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Fig. 1: Magnitude and phase responses of bandlimited channel.

➢ The linear filter channel model is shown in Fig. 2.

Fig. 2: Linear filter model for a bandlimited channel.

➢ Now, let us pass the received signal r(t) = h(t) + n(t) through a filter that has a frequency response

------------- (4)
where t0 is some nominal time delay at which we sample the filter output and R denotes that the matched
filter is at the receiver.
➢ The signal component at the output of the matched filter at the sampling instant t = t0 is

----------- (5)
which is the energy in the channel output waveform h(t). The noise component at the output of the
matched filter has a zero mean and a power spectral density

-------------- (6)
➢ Hence, the noise power at the output of the matched filter has a variance

-------- (7)
➢ Then the SNR at the output of the matched filter is

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MODULE-4 VI SEM DIGITAL COMMUNICATION [18EC61]

----------- (8)
Equation (8) is an expression for SNR at the output of the matched filter.

DIGITAL PAM TRANSMISSION THROUGH BANDLIMITED BASEBAND CHANNELS:


➢ The functional block diagram of baseband pulse amplitude modulation (PAM) communication system is
shown in Fig. 3.

Fig. 3: Block diagram of a digital PAM system

➢ The system consists of a transmitting filter having an impulse response gT(t), the linear filter channel
with AWGN, a receiving filter with an impulse response gR(t), a sampler that periodically samples the
output of the receiving filter, and a symbol detector.
➢ The sampler requires the extraction of a timing signal from the received signal. This timing signal serves
as a clock that specifies the appropriate time instants for sampling the output of the receiving filter.
➢ Let us consider digital communication by means of M-ary PAM. Hence, the input binary data sequence
is subdivided into k-bit symbols, and each symbol is mapped into a corresponding amplitude level that
amplitude modulates the output of the transmitting filter. The baseband signal at the output of the
transmitting filter may be expressed as

----------- (1)
where T = k /Rb is the symbol interval (1 / T = Rb/ k is the symbol rate), Rb is the bit rate and {an} is a
sequence of amplitude levels corresponding to the sequence of k-bit blocks of information bits.
➢ The channel output, which is the received signal at the demodulator may be expressed as

------------ (2)
where h(t) is the impulse response of the cascade of the transmitting filter and the channel. Thus, h(t) =
c(t)*gT(t), c(t) is the impulse response of the channel and n(t) represent the AWGN.
➢ The received signal is passed through a linear receiving filter with the impulse response gR(t) and
frequency response GR(f). If gR(t) is matched to h(t), then its output SNR is maximum at the proper
sampling instant. The output of the receiving filter may be expressed as

----------- (3)

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where x(t) = h(t)*gR(t) = gT(t)*c(t)*gR(t) and w(t) = n(t)*gR(t) denotes the additive noise at the output of
the receiving filter.
➢ To recover the information symbols {an}, the output of the receiving filter is sampled periodically, every
T seconds. Thus, the sampler produces

------------ (4)
or equivalently,

------------- (5)
where xm = x(mT), wm = w(mT), and m = 0, ±1, ±2, ……
➢ The first term on the right-hand side (RHS) of Equation (6) is the desired symbol am, scaled by the gain
parameter x0. When the receiving filter is matched to the received signal h(t), the scale factor is

------------ (6)
The second term on the RHS of Equation (6) represents the effect of the other symbols at the sampling
instant t = mT, called the intersymbol interference.
➢ In general, ISI causes a degradation in the performance of the digital communication system. Finally, the
third term in equation (5) wm, which represents the additive noise, is a zero-mean Gaussian random
variable with variance

➢ By appropriately designing the transmitting and receiving filters, we can satisfy the condition xn = 0 for
n ≠ 0, so that the ISI term vanishes.

SIGNAL DESIGN FOR BANDLIMITED CHANNELS:


➢ Firstly, the design consideration is done for no channel distortion and later, the design of filter when the
channel distorts the transmitted signal. We know, H(f) = C(f).GT(f)
➢ For distortion-free transmission is that the frequency response characteristic C(f) of the channel must
have a constant magnitude and a linear phase over the bandwidth of the transmitted signal

----------- (1)
where W is the available channel bandwidth, t0 represents an arbitrary finite delay, which is set to zero
for convenience and C0 is a constant gain factor, which is set to unity for convenience. Thus, we have
H(f) = GT(f).

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➢ Consequently, the matched filter at the receiver has a frequency response GR(f) = GT*(f) and its output at
the periodic sampling times t = mT has the form

---------- (2)
OR

--------- (3)
where x(t) = gT(t)*gR(t) and w(t) is the output response of the matched filter to the input AWGN process
n(t).
➢ The middle term on the RHS of Equation (3) represents the ISI. The amount of ISI and noise that is
present in the received signal can be viewed on an oscilloscope. The resulting oscilloscope display is
called an eye pattern because of its resemblance to the human eye. Examples of two eye patterns, one for
binary PAM and the other for quaternary (M = 4) PAM, are illustrated in Fig. 4(a).
➢ The effect of ISI is to cause the eye to close, thereby reducing the margin for additive noise to cause
errors. Fig. 4(b) illustrates the effect of ISI in reducing the opening of the eye. ISI distorts the position of
the zero crossings and causes a reduction in the eye opening.

Fig. 4: Eye patterns (a) Examples of eye patterns for binary and quaternary PAM (b) effect of ISI on eye
opening

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MODULE-4 VI SEM DIGITAL COMMUNICATION [18EC61]

DESIGN OF BANDLIMITED SIGNALS FOR ZERO ISI - THE NYQUIST CRITERION:


➢ The Bandwidth of the channel which is the Fourier Transform of the signal at the output of the receiving
filter is given as

-------- (1)

where GT(f) and GR(f) denote the frequency responses of the transmitter and receiver filters and C(f)
denotes the frequency response of the channel.
➢ For convenience, we set C0 = 1 and t0 = 0. when output of the receiving filter is sampled periodically at t
= mT yields to an expression

Here, the first term on the RHS of the equation is the desired symbol, the second term constitutes the ISI
and the third term is the additive noise.
➢ To remove the effect of ISI, it is necessary and sufficient that x(mT - nT) = 0 for n ≠ m and x(0) ≠ 0,
where we can assume x(0) = 1. i.e., the overall communication system has to be designed such that

--------- (2)

NYQUIST CONDITION FOR ZERO ISI:


➢ A necessary and sufficient condition for x(t) to satisfy

and that its Fourier transform X(f) must satisfy

➢ Now, suppose that the channel has a bandwidth of W. Then, C(f) = 0 for |f| > W. Consequently, X(f) = 0
for |f| > W. Under this condition, we can have three cases:

1. T < 1/2W or 1/T > 2W

Fig. 5: Plot of Z(f) for the case T < 1/2W

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At this rate, there exists no pulse whose spectrum replicas add to form a flat spectrum and ISI is
inevitable at this rate.

2. T = 1/2W or 2W = 1/T

Fig. 6: Plot of Z(f) for the case T = 1/2W

At this rate, there exists only one X(f) that results in Z(f) = T namely

which results in

or

Difficulties with Sinc pulses:


• It is non-casual and therefore non realizable. A possible solution is delaying it until to so that it is
approximately zero, for t < 0. But of course, mT → mT + t0
• It decays as 1/t; a small mistiming error in sampling the output of matched at demodulator results in
an infinite series of ISI components. (The series does not coverage)

3. T > 1/2W or 2W > 1/T

Fig. 7: Plot of Z(f) for the case T > 1/2W

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At this rate, there exists an infinite number of choices for X(f) such that Z(f) = T. There exist numerous
pulses which satisfy the Zero-ISI criterion. A popular choice for x(t) at 1/T < 2W is the Raised-cosine
spectrum. It’s frequency characteristics are given as

where α is called the roll-off factor, which takes values in the range 0 ≤ α ≤ 1. The pulse x(t) having the
raised cosine spectrum is

Fig. 8: Pulses having a raised cosine spectrum


Benefit of raised cosine spectrum:
• It decays as 1/t3 for α > 0, so any sampling offset results in finite ISI.
• With larger α, the pulse decays faster but the bandwidth utilization is poorer.
• It is possible to design practical filters that implement the raised cosine pulse.
x(t) = Transmit Pulse * Channel * Reciever filter
= gT(t) * c(t) * gR(t)
• The overall impulse response of the system is then raised cosine. For the ideal channel, C(f) = 1, we have

• If the receiver filter is matched to the transmitter filter, then

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Ideally,

The transmit pulse is root raised cosine pulse.

DESIGN OF BANDLIMITED SIGNALS WITH CONTROLLED ISI - PARTIAL-RESPONSE


SIGNALS:
➢ For zero ISI-system design, it is necessary to reduce the symbol rate 1/T, below the Nyquist rate of 2W
symbols/sec in order to realize practical transmitting and receiving filters.
➢ Suppose if we relax the condition of zero ISI, allowing a symbol transmission at a rate of 2W
symbols/sec can be obtained for a controlled amount of ISI.
➢ To design the bandlimited signal to have controlled ISI can be obtained by allowing one additional non-
zero value in the samples {x(nT)}. The ISI that we introduce is deterministic or controlled which is taken
into account at the receiver.

---------- (1)
➢ We know that,
𝐦
Z(f) = ∑∞
𝒎= −∞ 𝒙 (𝐟 + 𝐓 ) -------------- (2)
Z(f) is periodic function with period 1/T. Z(f) expanded in terms of Fourier series coefficient {zn} as
Z(f) = ∑∞
𝒏= −∞ 𝒛𝒏 𝒆
−𝒋𝟐𝝅𝒇𝒏𝑻
------------ (3)

𝟏/𝟐𝑻
zn = T ∫−𝟏/𝟐𝑻 𝒆−𝒋𝟐𝝅𝒇𝒏𝑻 𝒅𝒇 = T x(-nT)

➢ From equation (3),

➢ At T = 1/2W, we obtain
𝟏 𝟏 −𝐣𝟐𝛑𝐟⁄
X(f) = 𝟐𝐖 + 𝐞 𝟐𝐖
𝟐𝐖

𝟏 𝐣𝟐𝛑𝐟⁄ −𝐣𝟐𝛑𝐟⁄ −𝐣𝟐𝛑𝐟⁄


= 𝟐𝐖 [𝐞 𝟐𝐖 +𝐞 𝟐𝐖 ] 𝐞 𝟐𝐖
𝐣𝟐𝛑𝐟⁄ −𝐣𝟐𝛑𝐟⁄
𝟏 −𝐣𝟐𝛑𝐟⁄ 𝐞 𝟐𝐖 +𝐞 𝟐𝐖
=𝐖 𝐞 𝟐𝐖 [ ]
𝟐

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➢ Taking IFT, we get

This pulse is called a duobinary signal pulse. x(t) along with its magnitude spectrum is as shown in Fig.
9

Fig. 9: Time-domain and frequency-domain characteristics of a duobinary signal

➢ We note that the spectrum decays to zero smoothly, which mean that physically realizable filters can be
designed. Another special case that leads to transmitting and receiving filters is

➢ The corresponding pulse x(t) is given as

➢ Its spectrum is

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Fig. 10: (a) Time-domain and (b) frequency-domain characteristics of a modified duobinary signal

➢ This pulse as illustrated in Fig. 10 is called a modified duobinary signal pulse. The spectrum of this
signal has a zero at f = 0, making it suitable for transmission over a channel that does not pass DC.

PROBABILITY OF ERROR FOR DETECTION OF DIGITAL PAM WITH ZERO ISI:


➢ Without ISI, the received signal sample at the output of the receiving matched filter has the form

---------- (1)
and

Therefore,

---------- (2)
where wm is Additive Gaussian Noise, Mean = 0 and variance

➢ Here the problem of evaluating the probability of error for digital PAM in a bandlimited Additive White
Gaussian Noise channel is identical to the evaluation of the error probability for M-ary PAM. Finally,

------------ (3)

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But

and

where ℰav is the average energy/symbol and ℰbav is the average energy/bit
➢ Hence, equation (3) written as

-------- (4)
➢ Equation (4) is exactly the form for the probability of error of M-ary PAM.
➢ Here, the transmitted signal pulses were designed to be bandlimited and to have zero ISI. And also, no
loss in error-rate performance results from the bandwidth constraint when the signal pulse is designed
for zero ISI and the channel does not distort the transmitted signal.

SYMBOL-BY-SYMBOL DETECTION OF DATA WITH CONTROLLED ISI:


➢ For the duobinary signal pulse, x(nT) = 1, for n = 0, 1 and zero otherwise. Hence, the samples at the
output of the receiving filter have the form

-------- (1)
where {am} is the transmitted sequence of amplitudes and {wm} is the sequence of additive Gaussian
noise samples.
➢ The errors from the additive noise tend to propagate. Error propagation can be avoided by precoding the
data at the transmitter instead of eliminating the controlled ISI by subtraction at the receiver.
➢ The precoding is performed on the binary data sequence prior to modulation. From the data sequence
{dn} of ones and zeros that is to be transmitted, a new sequence {pn} called the precoded sequence is
generated. For the duobinary signal, the precoded sequence is defined as

--------- (2)
where the symbol ө denotes modulo-2 subtraction
➢ The noise-free samples at the output of the receiving filter are given as

➢ Consequently

➢ Since

it follows that the data sequence dm is obtained from bm by using the relation

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MODULE-4 VI SEM DIGITAL COMMUNICATION [18EC61]

➢ Also,

and

➢ The received level for the mth transmission bm is directly related to dm, the data at the same transmission
time. Therefore, an error in reception of bm only affects the corresponding data dm, and no error
propagation occurs.

CHANNEL EQUALIZATION:
➢ Till now in the preceding section, we described the design of transmitting and receiving filters for digital
PAM transmission when the frequency-response characteristics of the channel are known.
➢ The design these filters for zero ISI at the sampling instants. This design methodology is appropriate
when the channel is precisely known and its characteristics do not change with time.
➢ But in practice, the frequency characteristics of channel are unknown or vary with time. Equations of
time-varying channels are radio channels. These channels are characterized by time-varying frequency
response characteristics.
➢ Under these circumstances, we may design the transmitting filter to have a square root raised cosine
frequency response i.e.,

and the receiving filter with frequency response GR(f) to be matched to GR(f). Therefore,

and due to channel distortion, the output of the receiving filter is

where x(t) = gT(t) * c(t) * gR(t)


➢ The filter output may be sampled periodically to produce the sequence

where xn = x(nT), n = 0, ±1, ±2, . . .. The middle term on the right-hand side of the above equation
represents the ISI.
➢ In order to detect the information sequence {am}, Maximum Likelihood Sequence detector (ML) is the
optimum detector. But with this detector, the computational complexity is high and becomes impractical

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MODULE-4 VI SEM DIGITAL COMMUNICATION [18EC61]

to design. Hence there are sub optional methods for which ML detector serves as a benchmark for the
comparison of its performance.

LINEAR EQUALIZERS:
➢ For channels whose frequency-response characteristics are unknown and time variant, linear equalizers
can be employed at the output of the receiving filter.
➢ The block diagram of an equalizer is shown in Fig. 11.

To Detector

Fig. 11: Block diagram of a system with equalizer


➢ Design characteristics for a linear equalizer from a frequency domain initially. The demodulator consists
of a receiving filter with the frequency response GR(f) followed by a channel equalizing filter that has a
frequency response GE(f).
➢ Since GR(f) and GT(f) are matched and satisfies equation

➢ GE (f) must compensate for the channel distortion. Hence, the equalizer frequency response must equal
the inverse of the channel response, i.e.,

where

and
θE(f) = -θc(f)
➢ In this case, the equalizer is said to be the inverse channel filter to the channel response. The inverse
channel filter completely eliminates ISI to zero at the sampling times t = nT and hence the equalizer is
said to be a zero-forcing equalizer.
➢ Hence, the input to the detector is of the form of

where wm is the noise component, which is zero-mean Gaussian with a variance

➢ In general, the noise variance at the output of zero forcing equalizer is higher than the noise variance at
the output of optimum receiving filter |GR(f)|.

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DESIGN OF LINEAR EQUALIZER IN TIME DOMAIN:


➢ In real channels, the ISI is limited to a finite number of samples, say L samples. Thereby, a channel
equalizer is approximated by a finite-duration impulse response (FIR) filter or transversal filter with
adjustable tap coefficients {cn} as illustrated in Fig. 12.

Fig. 12: Linear transversal filter

➢ The time delay τ between adjacent taps may be selected as large as T, the symbol interval in which case
the FIR equalizer is called a symbol-spaced equalizer.
➢ But in this case, the frequencies in the received signal above 1 / T are aliased and hence the equalizer
compensates for the aliased channel-distorted signal.
➢ When the time delay τ between adjacent taps is selected such that 1/τ ≥ 2W ≥ 1/T, no aliasing occurs and
the inverse channel equalizer compensates for the true channel distortion.
➢ For τ < T, the channel equalizer is said to have fractionally spaced taps and it is called a fractionally
spaced equalizer.
➢ The impulse response of the FIR equalizer is

-------- (1)
and the corresponding frequency response is

------------ (2)
where {cn} are the (2N + 1) equalizer coefficients and N is chosen sufficiently large i.e., 2N +1 ≥ L
➢ Since X(f) = GT(f) C(f) GR(f) and x(t) is the signal pulse corresponding to X(f), the equalized output
signal pulse is

---------- (3)
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At t = mt,

--------- (4)
➢ Since (2N + 1) coefficients are there, equation (4) can be written as

--------- (5)
which may be expressed in matrix form as Xc = q, where X is a (2N + 1) x (2N + 1) matrix with
elements {x(mT - nt)}, c is the (2N + 1) coefficient vector and q is the (2N + 1) column vector with one
nonzero element.
➢ FIR zero-forcing equalizer does not completely eliminate ISI because it has a finite length. But as N is
increased, the residual ISI can be reduced and as N → ꝏ, the ISI is completely eliminated.

MINIMUM MEAN SQUARE ERROR (MMSE):


➢ The drawback of zero-forcing equalizer is that it completely ignores the presence of additive noise. And
as a result, it leads to significant noise enhancement. Therefore, an alternative approach that the zero ISI
condition and select the channel equalizer characteristic such that the combined power in the residual ISI
and the additive noise at the output of the equalizer is minimized which is obtained using minimum
mean-square-error (MMSE) method.
➢ Let us consider the noise-corrupted output of the FIR equalizer be

----------- (1)
At t = mT,

---------- (2)
➢ The desired response sample at the output of the equalizer at t = mT is the transmitted symbol a m. The
mean square error between the actual output sample z(mT) and the desired values am is

------------- (3)
where

----------- (4)
and

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----------- (5)
➢ Thus, the necessary condition for MMSE is

--------- (6)
Hence, there are (2N + 1) linear equations for the equalizer coefficients.

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QUESTION BANK
1. Explain the process of Digital Transmission through Bandlimited Channels.
2. With a neat block diagram explain the digital PAM transmission through bandlimited baseband
channels and also obtain the expression for ISI.
3. Explain the two eye patterns Binary and Quaternary PAM with effect of ISI on eye opening.
4. State and prove Nyquist criteria for zero ISI.
5. With a neat block diagram explain the concept of equalization using a linear transversal filter.
6. Explain Duobinary signal pulse and modified duobinary signal pulse with related diagrams and
equations.
7. Write short note on:
a. Linear Equalizers
b. MSME

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MODULE - 5

SYLLABUS:
Principles of Spread Spectrum: Spread Spectrum Communication Systems: Model of a Spread Spectrum
Digital Communication System, Direct Sequence Spread Spectrum Systems, Effect of De-spreading on a
narrowband Interference, Probability of error (statement only), Some applications of DS Spread Spectrum
Signals, Generation of PN Sequences, Frequency Hopped Spread Spectrum, CDMA based on IS-95

TEXT, REFERENCE & ADDITIONAL REFERENCE BOOKS:

BOOK TITLE/AUTHORS/PUBLICATION
John G Proakis and Masoud Salehi, “Fundamentals of Communication Systems”,
T-1
2014 Edition, Pearson Education, ISBN 978-8-131-70573-5.

B.P.Lathi and Zhi Ding, “Modern Digital and Analog communication Systems”,
AR-1
Oxford University Press, 4th Edition, 2010, ISBN: 978-0-198-07380-2.

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INTRODUCTION:
➢ There are some applications like military communication where in a transmitted message is to be
received only by the receiver for which it is intended others should not be able to receive it.
➢ Further in order to make the communication reliable it should not be possible for anyone to jam the
transmitted signal.
➢ Spread spectrum systems are intended to provide such secure and reliable communication. As the name
itself suggests these systems spread the spectrum of the transmitted signal over a very wide bandwidth.
➢ This is achieved in these systems by modulating for a second time, an already modulated signal in such a
way as to spread the power of the transmitted spread spectrum signal over a very large bandwidth.
➢ Thus, the power spectral density of this signal is so low that any ordinary AM (or FM) receiver with its
10 kHz (or 200 kHz) front end bandwidth receives an amount of spread spectrum signal power that is
very much lower than the thermal noise power entering the receiver. Thus, unauthorized receiver will
not be able to receive the spread spectrum signal.
➢ The spread spectrum signal cannot easily be jammed. Thus, these systems can provide very secure and
reliable communication making them ideally suited for military communications.
➢ Spread spectrum signal enables an increase in the number of users over a given band, a feature that is
exploited for providing multiple access in satellite communications and for increasing the number of
subscribers using the same band, in the case of cellular mobile communications.
➢ Thus, spread spectrum communication provides:
• Protection against eaves dropping
• Resistance to intentional jamming
• Resistance to fading caused by multipath effects
• Multi-user facility over a given channel
• Ranging facility
➢ The definition of spread spectrum may be stated two parts as follows:
• Spread spectrum is a measure of transmission in which the data of interest occupies a bandwidth in
excess of minimum bandwidth necessary to send.
• The Spreading of Spectrum is accomplished before transmission through the use of code that is
independent of data sequence. The same code is used in the receiver (Operating Synchronized with
the transmitter) to despread the received signal so that the original data may be recovered.
➢ Here we discuss principles of spread spectrum modulation with Direct Sequence and Frequency
Hopping Technique.
MODEL OF A SPREAD SPECTRUM DIGITAL COMMUNICATION SYSTEM:
➢ The basic elements of a spread-spectrum digital communication system are as shown in Fig. 1.

Fig. 1: Model of a spread-spectrum digital communication system.

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➢ The channel encoder and decoder and the modulator and demodulator are the basic elements of a
conventional digital communication system.
➢ In addition, a spread-spectrum system employs two identical pseudorandom sequence generators, one
that interfaces with the modulator at the transmitting end and one that interfaces with the demodulator at
the receiving end.
➢ These two generators produce a pseudorandom or pseudonoise (PN) binary-valued sequence, which is
used to spread the transmitted signal at the modulator and to despread the received signal at the
demodulator.
➢ Time Synchronization of the PN sequence is achieved by transmitting a fixed PN bit pattern, which is
designed so that the receiver will detect it with high probability in the presence of interference.
➢ Interference is introduced in the transmission of the spread-spectrum signal through the channel. The
interference may be generally categorized as either broadband or narrowband relative to the bandwidth
of the information-bearing signal and either continuous in time or discontinuous (pulsed) in time.
➢ Two types of digital modulation are considered, namely, phase-shift keying (PSK) and frequency shift
keying (FSK). The PN sequence generated at the modulator is used in conjunction with the PSK
modulation to shift the phase of the PSK signal pseudorandomly at a rate that is an integer multiple of
the bit rate. The resulting modulated signal is called a direct sequence (DS) spread-spectrum signal.
➢ When used in conjunction with binary or M-ary (M > 2) FSK, the PN sequence is used to select the
frequency of the transmitted signal pseudorandomly. The resulting signal is called a frequency-hopped
(FH) spread-spectrum signal.
DIRECT SEQUENCE SPREAD-SPECTRUM SYSTEMS:
➢ The basic method for accomplishing the spreading is shown in Fig. 2(b). The information-bearing
baseband signal is denoted as v(t) and is expressed as

𝑣(𝑡) = ∑∞ 𝑛= −∞ 𝑎𝑛 𝑔𝑇 (𝑡 − 𝑛𝑇𝑏 ) ----------- (1)


where an = ±1, -ꝏ < n < ꝏ and gT(t) is a rectangular pulse of duration Tb.

➢ This signal is multiplied by the signal from the PN sequence generator, which may be expressed as
𝑐(𝑡) = ∑∞ 𝑛= −∞ 𝑐𝑛 𝑝(𝑡 − 𝑛𝑇𝑐 ) ----------- (2)

where {cn} represents the binary PN code sequence of ± l's and p(t) is a rectangular pulse of duration Tc,
as shown in Fig. 2(a).

➢ This multiplication operation of v(t).c(t) serves to spread the bandwidth of the information-bearing
signal into the wider bandwidth occupied by PN generator signal c(t) as shown in Fig. 2(c). The
spectrum spreading is shown in Fig. 3
➢ The product signal v(t).c(t) amplitude modulates the carrier Accos2πfct and generates the double-
sideband suppressed-carrier (DSB-SC) signal
u(t) = Ac v(t) c(t) cos2πfct ----------------- (3)

➢ Since v(t).c(t) = ±1 for any time t, equation (3) may also be expressed as
u(t) = Ac cos [2πfct + θ(t)] ------------ (4)

where θ(t) = 0 when v(t).c(t) = 1 and θ(t) = π when v(t).c(t) = -1. Therefore, the transmitted signal is a
binary PSK signal.
➢ The rectangular pulse p(t) used to represent the PN sequence is called Chip. The time duration of pulse
p(t) is called Chip Interval (Tc). The reciprocal of Tc is called Chip Rate (1/ Tc). The ratio of the bit
interval Tb to the chip interval Tc is selected to be an integer. We denote this ratio as Lc = Tb / Tc Here,
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Lc is the number of chips of the PN code sequence per information bit. Also, Lc represents the number of
possible 180° phase transitions in the transmitted signal during the bit interval Tb.

Fig. 2: Generation of a DS spread-spectrum signal

Fig. 3: Convolution of the spectra (a) data signal with the (b) PN code signal (c) Product

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DEMODULATION OF DS – SPREAD SPECTRUM SIGNAL:


➢ The block diagram is as shown in Fig. 3

Fig. 3: Demodulation of DS spread-spectrum signal

➢ The received signal is first multiplied by the waveform c(t) generated by the PN code sequence generator
at the receiver, which is synchronized to the PN code in the received signal. This operation is called
despreading.
➢ Thus, we have
b(t) = r(t) c(t)
= Ac v(t) c(t) cos2πfct c(t)
= Ac v(t) c2(t) cos2πfct
2
Since c (t) = 1 for all t
b(t) = Ac v(t) cos2πfct
➢ The resulting signal Ac v(t) cos2πfct occupies a bandwidth of R Hz, which is the bandwidth of the
information-bearing signal. Thus, the demodulator for the despread signal is simply the conventional
cross correlator or matched filter.
EFFECT OF DESPREADING ON A NARROWBAND INTERFERENCE:
➢ Suppose that the received signal is
r(t) = Ac v(t) c(t) cos2πfct + i(t)
where i(t) denotes the interference.
➢ The despreading operation at the receiver yields to
b(t) = r(t) c(t)
= [Ac v(t) c(t) cos2πfct + i(t)] c(t)
= Ac v(t) c2(t) cos2πfct + i(t) c(t)
Here, the effect of multiplying the interference i(t) with c(t) is to spread the bandwidth of i(t) to W Hz.
➢ For example, consider the sinusoidal interfering signal
i(t) = AI cos2πfIt
where fI is a frequency within the bandwidth of the transmitted signal.
➢ i(t). c(t) results in a wideband interference with power spectral density I0 = PI / W where PI = (AI)2 /2 is
the average power of the interference. Since the desired signal is demodulated by a matched filter that
has a bandwidth R, the total power in the interference at the output of the demodulator is
I0Rb = PI Rb / W = PI / (W/ Rb) = PI / (Tb/ Tc) = PI / Lc
➢ The power in the interfering signal is reduced by an amount equal to the bandwidth expansion factor
W/R. Here, R is the Bandwidth of the information bearing signal and W is the Bandwidth of the PN
Sequence.
➢ The factor W/R = Tb / Tc = Lc is called the processing gain of the spread-spectrum system. Processing
Gain gives the net effect in reducing the interfering power.
PROBABILITY OF ERROR:
➢ The probability of error for a direct sequence spread-spectrum system with binary PSK modulation is
easily obtained from the SNR at the detector i.e.,
(SNR)D = 2Eb / I0
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➢ The probability of error for the sinusoidal interference is


2𝐸𝑏
𝑃𝑒 = 𝑄 (√ )
𝐼𝑜
where 𝐼𝑜 is power spectral density of an equivalent broadband interference.

INTERFERENCE MARGIN:
➢ We may express Eb / I0 in Q function as
𝑃𝑆⁄ 𝑊
𝐸𝑏
=
𝑃𝑆 𝑇𝑏
= 𝑅 = ⁄𝑅
𝐼0 𝑃𝐼⁄ 𝑃𝐼⁄ 𝑃𝐼
𝑊 𝑊 ⁄𝑃
𝑆
➢ Using Logarithmic Scale for above equation, we express it as
𝑃𝐼 𝑊 𝐸𝑏
10 𝑙𝑜𝑔 ( ) = 10 𝑙𝑜𝑔 ( ) − 10 𝑙𝑜𝑔 ( )
𝑃𝑆 𝑅 𝐼0
𝑃𝐼 𝑊 𝐸𝑏
( ) = ( ) − ( )
𝑃𝑆 𝑑𝐵 𝑅 𝑑𝐵 𝐼0 𝑑𝐵
𝑃
The ratio of (𝑃 𝐼 ) is called the interference angle.
𝑆 𝑑𝐵

APPLICATIONS OF DS SPREAD-SPECTRUM SIGNALS:

➢ LOW-DETECTABILITY SIGNAL TRANSMISSION


• Here, the information-bearing signal is transmitted at a very low power level relative to the
background channel noise and thermal noise that is generated in the front end of a receiver.
• If the DS spread-spectrum signal occupies a bandwidth W and the power spectral density of the
additive noise is No W/Hz, the average noise power in the bandwidth W is PN = WNo.
• The average received signal power at the intended receiver is PR. Now, if the signal is transmitted at
a power level such that PR / PN << 1. The intended receiver can recover the weak information-bearing
signal from the background noise with the aid of the processing gain and the coding gain.

➢ CODE DIVISION MULTIPLE ACCESS


• The enhancement in performance obtained from a DS spread-spectrum signal through the processing
gain and the coding gain can enable many DS spread-spectrum signals to occupy the same channel
bandwidth provided that each signal has its own pseudorandom sequence.
• Thus, it is possible to have several users transmit messages simultaneously over the same channel
bandwidth. This type of digital communication, in which each transmitter-receiver user pair has its
own distinct signature code for transmitting over a common channel bandwidth, is called Code
Division Multiple Access (CDMA).
• A major advantage of CDMA is that a large number of users can be accommodated if each user
transmits messages for a short period of time. In such a multiple access system, it is relatively easy to
add new users or to decrease the number of users without reconfiguring the system.
• If there are Nu simultaneous users, the desired signal to noise interference power ratio at a given
receiver is
PS / PN = PS / (Nu – 1) PS = 1 / (Nu – 1)
From this relation, we can determine the number of users that can be accommodated simultaneously.
• CDMA provides digital cellular telephone service to mobile users.

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➢ COMMUNICATION OVER CHANNELS WITH MULTIPATH


• In mobile radio communication systems, multipath propagation occurs due to reflection from
buildings, trees and other obstacles located between the transmitter and the receiver.
• DS spread spectrum is a particularly effective way to generate a wideband signal for resolving
multipath signal components, we may also reduce the effects of fading.

➢ WIRELESS LAN’S
• Spread-spectrum signals have been used in the IEEE wireless LAN standards 802.11 and 802.11b,
which operate in the 2.4 GHz ISM (industrial, scientific, and medical) unlicensed frequency band.
• The available bandwidth is subdivided into 14 overlapping 22 MHz channels, although not all
channels are used in all countries. In the United States, only channels 1 through 11 are used.
• In the 802.11 standard, an 11 chip Barker sequence is modulated and transmitted at a chip rate of 11
MHz, i.e., the chip duration is 0.909 µsec. The 11 chip Barker sequence is {1, - 1, 1, 1, -1, 1, 1, 1, -1,
-1, -1}. The Barker sequence is modulated either with BPSK or QPSK. When BPSK is used with 11
chips per bit, a data rate of 1 Mbps is achieved. When QPSK modulation is used with 11 chips per
symbol (2 bits), a data rate of 2 Mbps is achieved.
• Direct sequence spread spectrum is also used in the higher speed IEEE 802.11b wireless LAN
standard, which operates in the same 2.4 GHz ISM band. In 802.11b, the 11 MHz chip rate is
maintained, but the Barker sequence is replaced by a set of 8 chip waveform sequences called
Complementary Code shift Keying (CCK), which can be viewed as direct-sequence spread-spectrum
modulation with multiple spreading sequences. The use of CCK modulation results in a data rate of
11 Mbps.

GENERATION OF PN SEQUENCES:
➢ A pseudorandom or pseudonoise (PN) sequence is a code sequence of 1's and 0's whose autocorrelation
has properties similar to those of white noise.
➢ The most widely known binary PN code sequences are the maximum-length shift register sequences or
m-sequence for has the length L = 2m - 1 bits and is generated by an m-stage shift register with linear
feedback as shown in Fig. 4. The sequence is periodic with period L. Each period contains 2m-1 ones and
2m-1 -1 zeros.

Fig. 4: General m-stage shift register with linear feedback

➢ Table 1 lists shift register connections for generating maximum-length sequences.

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Table 1: Shift-Register connections for generating maximum-length sequences

➢ In DS spread-spectrum applications, the binary sequence with elements {0, 1} is mapped into a
corresponding binary sequence with elements {-1, 1}. The equivalent sequence {Cn} with elements {-1,
1} a bipolar sequence.

IMPORTANT CHARACTERISTICS / PROPERTIES OF PN SEQUENCES:


➢ An important characteristic of a periodic PN sequence is its autocorrelation function, which is usually
defined in terms of the bipolar sequences {cn} as
𝑅𝑐 [𝑚] = ∑𝐿𝑛=1 𝑐𝑛 𝑐𝑛+𝑚 ; 0 ≤ m ≤ L-1
where L is the period of the sequence. Since the sequence {cn} is periodic with period L, the
autocorrelation sequence {Rc[m]} is also periodic with period L.
➢ The ideal autocorrelation sequence for {cn} is Rc[0] = L and Rc[m] = 0 for 1 ≤ m ≤ L-1. In the case of m-
sequences, the autocorrelation sequence is
𝐿, 𝑚=0
𝑅𝑐 [𝑚] = {
−1, 1 ≤𝑚 ≤𝐿−1
The ratio Rc[m]/ Rc[0] = - 1 / L is small.

DIFFERENT METHODS FOR GENERATING PN SEQUENCES:


➢ Methods for generating PN sequences with better periodic cross-correlation properties than m-sequences
have been developed by Gold and by Kasami.
➢ Gold sequences are constructed by taking a pair of specially selected m-sequences, called preferred m-
sequences and forming the modulo-2 sum of the two sequences for each of L cyclically shifted versions
of one sequence relative to the other sequence. L Gold sequences are generated as shown in Fig. 5.

Fig. 5: Generation of Gold sequences of length 31

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➢ For m odd, the maximum value of the cross-correlation function between any pair of Gold sequences is
Rmax = √2𝐿 and for m even, Rmax = √𝐿.
➢ Kasami described a method for constructing PN sequences by decimating an m-sequence. In Kasami's
method of construction, every 2m/2 + 1 bit of an m-sequence is selected. This method of construction
yields a smaller set of PN sequences compared with Gold sequences, but their maximum cross-
correlation value is Rmax = √𝐿
➢ Gold sequences and Kasami sequences are compared with the peak value of cross-correlation function.
Given a set of N sequences of period L, a lower bound on their maximum cross correlation is
𝑁−1
𝑅𝑚𝑎𝑥 = 𝐿 √
𝑁𝐿 − 1
➢ For large values of L and N, Rmax = √𝐿. Hence, Kasami sequences satisfy the lower bound and they are
optimal. On the other hand, Gold sequences with m odd have an Rmax = √2𝐿. Hence, they are slightly
suboptimal.

FREQUENCY-HOPPED SPREAD SPECTRUM:


➢ In FH spread spectrum, the available channel bandwidth W is subdivided into a large number of
nonoverlapping frequency slots. In any signaling interval, the transmitted signal occupies one or more of
the available frequency slots. The selection of the frequency slots in each signal interval is made
pseudorandomly according to the output from a PN generator.
➢ A block diagram of the transmitter and receiver for an FH spread-spectrum system is shown in Fig. 6.
➢ The modulation is either binary or M-ary FSK (MFSK). i.e., if binary FSK is employed, the modulator
selects one of two frequencies, such as f0 or f1, corresponding to the transmission of 0 or 1.
➢ The resulting binary FSK signal is translated in frequency by an amount determined by the output
sequence from a PN generator, which is used to select a frequency fc that is synthesized by the frequency
synthesizer.
➢ This frequency is mixed with the output of the FSK modulator and the resultant signal is transmitted
over the channel.

Fig. 6: Block diagram of an FH spread-spectrum system

➢ Fig. 7 illustrates an FH signal pattern.

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Fig. 7: An example of an FH pattern

➢ At the receiver, there is an identical PN sequence generator, which is synchronized with the received
signal and is used to control the output of the frequency synthesizer. Thus, the pseudorandom frequency
translation introduced at the transmitter is removed at the demodulator by mixing the synthesizer output
with the received signal.
➢ The resultant signal is then demodulated via an FSK demodulator. A signal for maintaining synchronism
of the PN sequence generator with the FH received signal is usually extracted from the received signal.

FREQUENCY HOPPING RATE (Rh):


➢ The frequency-hopping rate denoted as Rh, may be either equal to the symbol rate, lower than the
symbol rate or higher than the symbol rate.
➢ If Rh is equal to or lower than the symbol rate, the FH system is called a slow hopping system.
➢ If Rh is higher than the symbol rate i.e., there are multiple hops per symbol, the FH system is called a
fast-hopping system.

SLOW FREQUENCY-HOPPING SYSTEMS:


➢ Consider a slow frequency-hopping system in which the hop rate Rh = 1 hop per bit.
➢ Assuming that the interference on the channel is broadband and is characterized as AWGN with power
spectral density I0. Under these conditions, the probability of error for the detection of noncoherently
demodulated binary FSK is
1 −𝜌𝑏
⁄2
P2 = 2 𝑒
𝐸𝑏
where 𝜌𝑏 = is the SNR per bit.
𝐼0
➢ In the case of a DS spread-spectrum system, we observe that Eb, the energy per bit can be expressed as
𝑃 𝑃
𝐸𝑏 = 𝑃𝑠 𝑇𝑏 = 𝑠⁄𝑅 and 𝐼0 = 𝑊𝐼
where
Eb is the Energy per bit
Ps is the Average Transmitted Power
R is the Bit rate
PI is the Average power of Broadband Interference
W is the Available Channel Bandwidth
➢ The SNR 𝜌𝑏 is expressed as
𝐸𝑏 𝑊⁄
𝜌𝑏 = = 𝑅
𝐼0 𝑃𝐼
⁄𝑃
𝑠
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where
𝑊⁄ is the Processing Gain
𝑅
𝑃𝐼
⁄𝑃 is the Interference Margin for FH spread spectrum signal
𝑠

APPLICATIONS OF FH SPREAD SPECTRUM:


➢ CDMA systems based on FH spread-spectrum signals are particularly attractive for mobile (land, air,
sea) users because timing (synchronization) requirements are not as stringent as in a DS spread-spectrum
system.
➢ Larger processing gains are possible by FH, which more than offset the loss in performance inherent in
noncoherent detection of the FSK-type signals.

CDMA SYSTEM BASED ON IS-95:


➢ Direct sequence CDMA has been adopted as one multiple-access method for digital cellular voice
communications in North America. This first generation digital cellular (CDMA) communication system
was developed by Qualcomm, and it has been standardized and designated as IS-95 by the
Telecommunications Industry Association (TIA) for use in the 800 MHz and the 1900 MHz frequency
bands.
➢ The nominal bandwidth used for transmission from a base station to the mobile receivers (forward link)
is 1.25 MHz. A separate channel, also with a bandwidth of 1.25 MHz, is used for signal transmission
from mobile receivers to a base station (reverse link). The signals transmitted in both the forward and the
reverse links are DS spread spectrum signal and they have a chip rate of 1.2288 x 106 chips per second
(l.2288 Mchips/sec).
FORWARD LINK:
➢ A block diagram of the modulator for the signals transmitted from a base station to the mobile receivers
is shown in Fig. 8.
➢ The speech coder is a code-excited linear predictive (CELP) coder that generates data at the variable
rates of 9600, 4800, 2400 and 1200 bits/sec, where the data rate is a function of the user's speech activity
in frame intervals of 20 msec.
➢ The data from the speech coder is encoded by a rate 1/2, constraint length L = 9 convolutional code.
➢ The data rates are 4800, 2400, or 1200 bits/sec, the output symbols from the convolutional encoder are
repeated either twice, four times, or eight times to maintain a constant bit rate of 9600 bits/sec.
➢ The encoded bits for each frame are passed through a block interleaver which overcomes the effects of
burst errors that may occur in transmission through the channel.
➢ The data bits at the output of the block interleaver, which occur at a rate of 19.2 kbits/sec are scrambled
by multiplication with the output of a long code (period N = 242 - 1) whose output is decimated by a
factor of 64 to 19.2 kchips/sec. The long code uniquely identifies a call of a mobile station on the
forward and reverse links.
➢ Each channel user is assigned a Hadamard (also called a Walsh) sequence of length 64. There are 64
orthogonal Hadamard sequences assigned to each base station, thus there are 64 channels available. Two
Hadamard sequence one at transmitter other at receiver and one sequence is used for paging. Hence 61
channels are used for allocation to different users.
➢ Each user, using the assigned Hadamard sequence, multiplies the data sequence by the assigned
Hadamard sequence.
➢ The resulting binary sequence is now spread by multiplication with two PN sequences of length N = 215;
this creates In-phase (I) and Quadrature (Q) signal components.

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MODULE-5 VI SEM DIGITAL COMMUNICATION [18EC61]

➢ The I and Q components are filtered by baseband spectral-shaping filters. The signals for all 64 channels
are transmitted synchronously so that, in the absence of channel multipath distortion, other signals
received at any mobile receiver do not interfere because of the orthogonality of the Hadamard sequences.

Fig. 8: Block diagram of a IS-95 forward link

REVERSE LINK:
➢ A block diagram of the reverse link is as shown in Fig. 9.
➢ For lower speech activity, output bits from the convolutional encoder are repeated either two, four or
eight times. The coded bit rate is 28.8 kbits/sec.
➢ For each 20-msec frame, the 576 encoded bits are block-interleaved and passed to the modulator. The
data are modulated using an M = 64 orthogonal signal set of Hadamard sequences each of length 64.
Thus, a 6-bit block of data is mapped into one of the 64 Hadamard sequences.
➢ The signal is spread by the output of the long code PN generator, which is running at a rate of 1.2288
Mchips/sec.
➢ The resulting 1 .2288-Mchips/sec binary sequences of length N = 215 whose rate is also 1.2288
Mchips/sec, create I and Q signals (a QPSK signal) that are filtered by baseband spectral shaping filters
and then passed to quadrature mixers.

Fig. 9: Block diagram of the IS-95 reverse link

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MODULE-5 VI SEM DIGITAL COMMUNICATION [18EC61]

➢ The Q-channel signal is delayed in time by one-half PN chip relative to the I - channel signal prior to the
baseband filter. The output of the two baseband filters is an offset QPSK signal.
➢ The demodulator employs noncoherent demodulation of the M = 64 orthogonal Hadamard waveforms to
recover the encoded data bits.
➢ The output of the demodulator is then fed to the Viterbi decoder, whose output synthesizes the speech
signal.

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MODULE-5 VI SEM DIGITAL COMMUNICATION [18EC61]

QUESTION BANK
1. What is spread spectrum? Explain its advantages.
2. Explain the generation of direct sequence spread spectrum signal with relevant waveform and spectrums.
3. Explain the effect of despreading on narrowband interference.
4. Explain the properties of PN sequences.
5. Explain the generation of PN sequences.
6. Write a short note on application of spread spectrum:
a. Low Detectability Signal Transmission
b. Code Division Multiple Access
c. Wireless LAN’s.
7. Explain slow frequency hop spread spectrum.
8. Explain frequency hop spread spectrum technique with transmitter and receiver operations.
9. With a neat block diagram explain the CDMA system based on IS – 95.

Dept. of ECE, CEC, Mangalore Page 13

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