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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
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.
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.
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.
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.
➢ 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.
➢ 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)
------------- (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
------------ (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
-------- (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
------- (9)
Therefore, substituting (4) into (9), we get
Dept. of ECE, CEC, Mangalore Page 4
MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]
➢ 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)
Dept. of ECE, CEC, Mangalore Page 5
MODULE-3 VI SEM DIGITAL COMMUNICATION [18EC61]
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°.
------------- (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
---------- (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.
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.
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.
❖ 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
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
Similarly, if there are two carrier frequencies, the basis functions are required are two.
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.
𝑇
s21 = ∫0 𝑏 s2(t) ø1(t) dt = -√𝐸𝑏
➢ The signal space diagram for binary PSK is shown in Fig. 2
➢ 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
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
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
➢ 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
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.
2
ø2(t) = √𝑇 sin 2πfct ; 0 ≤ t ≤ T
➢ Thus, QPSK is a two-dimensional constellation (N = 2) and four message points (M = 4). The signal
space diagram is shown in Fig. 5
➢ 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)
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.
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
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
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
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)
➢ 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.
➢ 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.
➢ 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.
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
WAVEFORM REPRESNTATION:
➢ The FSK waveform for the binary sequence 101100 is given in Fig. 11 where f1 > f2
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
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
➢ 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.
➢ 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
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}.
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)
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.
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.
--------------- (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.
➢ 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
----------- (8)
Equation (8) is an expression for SNR at the output of the matched filter.
➢ 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)
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.
----------- (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).
➢ 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
-------- (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)
➢ 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:
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
At this rate, there exists only one X(f) that results in Z(f) = T namely
which results in
or
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
Ideally,
---------- (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)
➢ At T = 1/2W, we obtain
𝟏 𝟏 −𝐣𝟐𝛑𝐟⁄
X(f) = 𝟐𝐖 + 𝐞 𝟐𝐖
𝟐𝐖
This pulse is called a duobinary signal pulse. x(t) along with its magnitude spectrum is as shown in Fig.
9
➢ 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
➢ Its spectrum is
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.
---------- (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)
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.
-------- (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
➢ 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,
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
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
➢ 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
➢ 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)|.
➢ 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)
Dept. of ECE, CEC, Mangalore Page 15
MODULE-4 VI SEM DIGITAL COMMUNICATION [18EC61]
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.
----------- (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
----------- (5)
➢ Thus, the necessary condition for MMSE is
--------- (6)
Hence, there are (2N + 1) linear equations for the equalizer coefficients.
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
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
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.
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.
➢ 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
➢ 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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MODULE-5 VI SEM DIGITAL COMMUNICATION [18EC61]
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. 3: Convolution of the spectra (a) data signal with the (b) PN code signal (c) Product
➢ 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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MODULE-5 VI SEM DIGITAL COMMUNICATION [18EC61]
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
➢ 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.
➢ 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.
➢ 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.
➢ 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.
where
𝑊⁄ is the Processing Gain
𝑅
𝑃𝐼
⁄𝑃 is the Interference Margin for FH spread spectrum signal
𝑠
➢ 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.
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.
➢ 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.
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.