www.cambridge.org/Ha
HA
Providing the underlying principles of digital communication and the design techniques of
real-world systems, this textbook prepares senior undergraduate and graduate students for
the engineering practices required in industry. Covering the core concepts, including link
analysis, modulation, demodulation, spread spectrum, equalization, channel fading effects,
and channel coding, it provides step-by-step mathematical derivations to aid understanding
of background material. In addition to describing the basic theory, the principles of system
and subsystem design are introduced, enabling students to visualize the intricate connections between subsystems and understand how each aspect of the design supports the overall
goal of achieving reliable communications. Throughout the book, theories are linked to
practical applications with over 250 real-world examples, whilst 370 varied homework
problems in three levels of difculty enhance and extend the text material. With this
textbook, students can understand how digital communication systems operate in the real
world, learn how to design subsystems, and evaluate end-to-end performance with ease and
condence.
Tri T. Ha is a Professor in the Department of Electrical and Computer Engineering at the Naval
Postgraduate School (NPS), Monterey, California, a position he has held since 1989. Prior to
joining NPS he worked at Fairchild Industries and GTE, and was an Associate Professor at
Virginia Tech for four years. He is an IEEE Fellow who has written two previous textbooks,
and his current research interests are in wireless communications and cyber warfare.
World peace must develop from inner peace. Peace is not the absence of violence. Peace
is the manifestation of human compassion.
14th Dalai Lama of Tibet (Inscription on United States Congressional Medal)
Contents
Preface
Acknowledgements
List of symbols
List of abbreviations
1 Introduction
1.1 Brief overview
1.2 Scope
1.3 Summary
Bibliography
2.2
2.3
2.4
2.5
2.6
2.7
2.8
vii
Introduction
General description of deterministic signals
Continuous-time signals
Discrete-time signals
Periodic and aperiodic signals
Analog and digital signals
Power and energy
Orthogonal signals
Signal space
GramSchmidt procedure
Signal vectors
Linear time-invariant systems
Convolution
Fourier series of continuous-time periodic signals
Parseval relation for periodic signals
Unit step and unit impulse functions
Power spectral density
Bandwidth
Frequency shifting property
Response of an LTI system to a periodic signal
Fourier transform of continuous-time signals
Frequency shifting property
Parseval relation
Fourier transform of periodic signals
page xvii
xix
xx
xxx
1
1
3
8
9
10
10
10
11
11
12
13
16
21
27
27
28
34
35
37
39
39
41
41
42
45
47
50
52
53
viii
Contents
53
56
56
57
59
61
62
62
63
64
68
68
70
72
73
77
77
78
78
78
79
79
79
79
79
81
82
82
85
92
100
100
102
103
105
106
108
108
108
111
112
113
114
ix
Contents
3.6 Summary
Problems
Further reading
Bibliography
117
118
121
121
122
122
122
124
126
127
128
129
131
131
134
136
138
144
144
147
152
153
154
160
163
165
167
169
170
175
175
177
177
177
178
181
187
190
191
192
192
195
Contents
5.3
5.4
5.5
5.6
6 Modulation
6.1
6.2
6.3
6.4
6.5
6.6
6.7
6.8
6.9
6.10
6.11
6.12
Introduction
Review of double sideband-amplitude modulation (DSB-AM)
Digital modulation
Phase shift keying (PSK)
Differential phase shift keying (DPSK)
Amplitude shift keying (ASK)
Frequency shift keying (FSK)
Power spectral density
Minimum shift keying (MSK)
Power spectral density
Modulator
Gaussian minimum shift keying (GMSK)
The need for M-ary modulation
M-ary amplitude shift keying (MASK)
Signal space
Power spectral density
Modulator
M-ary phase shift keying (MPSK)
Signal space
Power spectral density
Modulator
Offset quadrature phase shift keying (OQPSK)
Differential M-ary phase shift keying (DMPSK)
Alternative differential encoding for DQPSK
Direct symbol mapping
Modulator
196
199
202
203
203
206
208
210
211
211
214
215
219
220
224
224
225
225
225
229
231
234
236
238
240
242
243
247
249
251
252
252
253
255
255
256
258
260
260
262
264
265
265
xi
Contents
7 Demodulation
7.1
7.2
7.3
7.4
7.5
Introduction
The matched lter
Time domain interpretation
Frequency domain interpretation
Output signal and noise waveforms
Decision variable
Summary
The correlator
The matched lterenvelope detector (noncoherent matched lter)
Output signal-to-noise ratio
Decision variable
Summary
The quadrature correlatorsquare law detector (noncoherent correlator)
The threshold detector
Optimum threshold
Maximum likelihood criterion
266
267
268
270
270
271
274
275
276
276
277
278
279
280
281
282
282
283
287
290
291
295
298
299
302
302
305
305
306
306
310
310
311
312
312
313
317
318
318
318
320
320
321
xii
Contents
322
323
323
325
326
328
329
331
332
337
339
345
345
348
350
351
353
355
355
358
360
362
366
371
375
379
380
383
384
386
395
397
401
402
403
409
410
411
412
414
416
422
422
xiii
Contents
Problems
Further reading
Bibliography
8 Spread spectrum
8.1
8.2
8.3
8.4
Introduction
Direct sequence modulation
Orthogonal covering
IS-95 forward link
Code division multiple access (CDMA)
IS-95 reverse link
Direct sequence demodulation
Quadrature orthogonal covering demodulation
Noncoherent demodulation of DS-CSK
CDMA: performance evaluation
Frequency hop spread spectrum
Partial-band jamming
Multi-tone jamming
Follower jamming
Summary
Problems
Further reading
Bibliography
9.3
9.4
9.5
9.6
9.7
9.8
9.9
Introduction
Intersymbol interference
Nyquist criterion for zero ISI
Optimum demodulator for bandlimited channel
Condition for maximum SNR0
Condition for zero ISI
Solution for |HT ( f )| and |HR ( f )|
Zero-forcing linear equalizer (ZF-LE)
Summary
Mean-square error linear equalizer (MSE-LE)
Zero-forcing decision-feedback equalizer (ZF-DFE)
Mean-square error decision-feedback equalizer (MSE-DFE)
Maximum likelihood sequence detection
Fractionally spaced equalizer (FSE)
Summary
Problems
Further reading
Bibliography
423
430
430
432
432
432
436
440
443
444
446
450
454
457
460
461
462
464
466
466
471
471
473
473
473
475
477
480
480
480
482
493
494
501
506
509
516
517
518
520
520
xiv
Contents
10 Fading channels
Introduction
10.1 Physical characterization of fading channels
Time-varying effect
Space-varying effect
Summary of fading characteristics
10.2 Mathematical representation of fading channels
Channel impulse response
Multipath autocorrelation and Doppler proles
ClarkeDoppler power spectrum
Generalized Doppler power spectrum
10.3 Coherent demodulation
Equivalent complex-valued demodulator
Rayleigh
Rice
Nakagami-m
Effect of channel tap error
10.4 Pilot symbol-aided decision-feedback demodulation
Differential and double-differential decision-feedback algorithms
10.5 OFDM
Low mobility
High mobility
10.6 Noncoherent demodulation
Rayleigh
Rice
Nakagami-m
Doppler tracking of MFSK
Doppler tracking of CSK
10.7 Pilot tone-aided demodulation of orthogonal covering signal
Complex spreading and despreading
Doppler analysis
10.8 Noncoherent demodulation of offset quadrature DS-CSK
10.9 Time diversity
Level crossing rate
Average fade duration
10.10 Maximal ratio combining (MRC)
Rayleigh
Rice
Nakagami-m
10.11 Selection combining (SC)
10.12 Equal gain combining (EGC)
Coherent EGC
Noncoherent EGC
522
522
524
524
526
527
528
528
530
532
534
536
536
539
540
541
543
547
549
552
556
557
558
558
558
559
559
562
566
566
567
572
574
575
577
578
579
581
581
583
585
585
586
xv
Contents
587
588
588
590
595
596
597
598
600
600
602
603
604
604
605
608
609
610
611
612
612
615
616
617
617
618
618
619
620
627
627
629
Preface
This book was written with two goals in mind: to provide the underlying principles of digital
communication and to study design techniques integrated with real world systems. The
ultimate aim of a communication system is to provide reliable transmission of information to
the user(s). This fundamental foundation was established in 1948 by Claude Shannon, the
founding father of information theory, and led eventually to the development of modern
digital communication. Analog communication is near extinction or at the very gate of it.
The full spectrum dominance of digital communication has arrived and new frontiers are
being established every decade; from cellular systems to wireless LAN and MAN, the bit
rates are being pushed ever higher for ubiquitous mobile applications.
Knowing the limit of digital transmission is vital to the design of future communication
systems, particularly mobile wireless systems, where both spectrum and power are precious
resources, and design techniques can be used to manipulate these two main resources to t
real world applications. No single technique can cover all the requirements of a modern
communication system, which makes it necessary for students to understand the intricate
web between subsystems, each designed to support others to achieve the common goal of
reliable communication.
The book contains more than 250 examples to help students achieve a rmer understanding of the subject. The problems at the end of each chapter follow closely the order of
the sections. They are designed for three levels: level one covers the straightforward
application of equations; level two requires patience and deep thinking; whilst level three
requires some research of the literature to assist in nding a solution. A solutions manual for
the instructor accompanies the book.
The book was written for both senior undergraduate and graduate students studying
communications at universities and colleges. The entire book is suitable for twosemester courses in digital communications. The rst course is typically a one-semester
senior course in digital communication, which may be taken by students new to
studying communications (the conventional wisdom is that students should learn analog
communication before learning digital communications) or after completing an introductory course in communication systems (one that is heavy in analog communication
systems such as AM and FM). The second course is a one-semester course for graduate
students who already have a rm background in random variables and processes. The
practical material included in this book (much of it focused on commercial and military
systems) will be helpful for practitioners and professionals in the digital communication
eld.
As in the learning of any subject, some prerequisites are required for the reading of this
book. A rst course in probability theory is necessary and exposures to random processes
xvii
xviii
Preface
would be helpful. Readers should also be familiar with linear system analysis. A knowledge
of analog communication is helpful but not required. For readers who do not have the
patience to go through all the design techniques but would appreciate the beauty of the
underlying principles, we recommend our favorite book, Principles of Digital
Communication, authored by the legendary Robert G. Gallager.
Acknowledgements
I would like to express my thanks to Dr. Phil Meyler of Cambridge University Press for his
enthusiastic support and for suggesting the title of the book to closely reect its coverage.
I would like to thank the anonymous reviewers for their valuable comments that helped in
improving the book. I am grateful for the support of my friends Nathan Beltz, Professor
Vicente Garcia, Professor Jeff Knorr, Professor Frank Kragh, Donna Miller, Rita Painter,
Professor Clark Robertson, Professor Wei Su, Dr. Jan Tighe, and Dr. Charlie Victory. I also
would like to thank my students over the years for their valuable suggestions to improve the
original notes. I am grateful to my parents for providing me the opportunity to go to the
United States for higher education, and to Thuy, Khanh, Tuan, Huong and Dennis, Thu, and
Hien for their help and support over the years. This book is dedicated to the people who
work tirelessly for world peace.
xix
Symbols
A
A
jjAjj2
A
Ad
Ae
A \ B; AB
A [ B; A B
a
a(hR)
af
ak
B
Bd
Bd free
C
C
Co
cDK
CN
CX t1 ; t2
C(z)
CovNj Nk
c
ci
c(t)
ct
cI t
cQ t
cxjHi
Di
xx
xxi
List of symbols
d
d
dB
d free
d free
di , d(i)
fdi g, fdig
dmin
d(u,v)
d x; si
E
Eb
Eb
E(d,t)
Eh
EIRP
E0
Es
Es
EX
EX 2
e
e
e(k)
e(t)
F
Ffxtg
F z
F 1=z
F 1 fX f g
2 F1
1
Fjhj2 po SNR
FX x
FXY x; y
fjhj x
fXY x; y
fX jY xjy
f
distance
data sequence
decibel
free distance of a convolutional code
Euclidean free distance of TCM
normalized bit or symbol amplitude, di 2 f1; 1, di 2 f0; 1
data sequence
minimum Hamming distance, minimum Euclidean distance
Hamming distance between two code words
Euclidean distance between two vectors
energy (with or without a subscript), smallest symbol energy in MQAM or
MASK, electric-eld wave, elevation angle
bit energy
diversity bit energy
free space E-eld at distance d from transmitter and time t
hop energy
effective isotropic radiated power
free space E-eld at distance d0 from transmitter
symbol energy
diversity symbol energy
expected value (mean value) of X
mean-square value of X 2
2.718
error word
error sequence of MSE-LE
error process
noise gure
Fourier transform of x(t)
minimum-phase transfer function, transfer function of a synthetic channel,
z-transform of the sequence f(k)
maximum-phase function
inverse Fourier transform of X(f )
Gauss hypergeometric function
outage signal-to-noise ratio
distribution function of X
joint distribution function of X and Y
density function of the channel tap magnitude jhj
joint density function of X and Y
conditional density function of X given Y
frequency, Doppler shift
xxii
List of symbols
fc
fD
fj
fL;i
fm
fs
fX x
fX x
G
G
G
GDC
GLNA
GR
GR =Ts
GS =Tsat
GT
g
g(t)
gn t
g(x)
H
H
Hi
H(f )
HFE f
jHf j
HT f
HR f
H(k)
H(X)
h
h
^
h
hi ; hi
~hi
hL t
h(n)
hR
carrier frequency
Doppler spread
instantaneous carrier frequency of FH
instantaneous local carrier frequency of a frequency synthesizer
maximum Doppler shift
sampling frequency
density function of X
density function of vector X
amplier gain, gain of a two-port network
generator matrix, space-time block code matrix
TCM asymptotic coding gain
gain of a downconverter
gain of a low-noise amplier
receiver antenna gain
antenna gain-to-noise temperature ratio of the earth station
antenna gain-to-noise temperature ratio of the satellite
transmitter antenna gain
parity
pulse
orthonormal pulse shapes in OFDM
code generator polynomial
Hadamard matrix (with or without a subscript), parity check matrix,
channel tap matrix
source entropy in bits/second
hypothesis
transfer function or frequency response
transfer function of the front-end lter of the receiver
magnitude response (amplitude response)
transfer function of transmit lter
transfer function of receive lter
N-point DFT of the sequence h(n)
entropy of the discrete random variable (discrete source) X
Planck constant, digital modulation index, complex channel tap
row vector of a Hadamard matrix
MMSE of the vector h
ith channel tap
complex channel tap
complex envelope of the impulse response h(t)
sequence used in OFDM
receive antenna height
xxiii
List of symbols
hT
h(t)
h(X)
h(X|Y)
h(X)
h(x)
I
Im
I
I(X)
Iui ; vj
IU ; V
I0
In
J
J0 , J0
K
K
k
k0
k/n
LC
L
L(f )
Lr
Ls
LTL
Lt
LdB
L; P1 ; :::; Pn
l
l
ln
ln X jy
log
log2
M
M(r | c)
m
xxiv
List of symbols
mA X 1
mG X
mn
m(t)
l
mij
^ l
m
ji
m2l
mX t
N
N
N
N
N
N f
ND
NH
Ni
Nn
N 0; 2
N0; 2
Nk
N0 =2
NV
n
nI ; N I
nQ ; NQ
(n,k)
n(t)
nL t
n0 t
P
Pc;h
Pd
Pe
P(f )
Pj
Pp
Pr(A)
Pr(A,B)
xxv
List of symbols
Pr(A|B)
Prc ! c 0
Pr(c | r)
Pr(r | c)
PT
P(z)
p
pout
pUV ui ; vj
pvj jui
pX xi
p(t)
Qa; b
Qx
Qz
q
R
Rb
Rc
R
Re
Rh ; t 0
Rhi t 0
Ri j
Rs
R(t)
RV
Rw
Rx
RX t1 ; t2
r
r
r2
re
S
S
S
S(f )
Sh ; f 0
Shi f 0
xxvi
List of symbols
s
sgn (x)
sign x
s(t)
fsi tg
fsi g
ksi k
sik
sL n
sL t
s0 t
sp
T
T
TA
Tb
Tc
Td
TDC
TDM
Te
Th
TLNA
Tn
T0
Tp
Tr(A)
Ts
t
U
M
S
Ai
i1
u(t)
u
V
VarX
Vrms
VT
v
v(t)
xxvii
List of symbols
W
fwn tg
X
X
X
X2
0
X ej2f
X f 2
X(f )
X(k)
Xn
XT f
X(z)
x
x
b xc
xn
x(t)
xI t
fxk tg
fxk g
xL t
xp t
xQ t
xs t
xT t
Y(k)
y(n)
y(t)
Z
Z(k)
Z0
zk t
bandwidth, watt
set of Walsh functions, n = 1, 2, . . . , M
random variable, discrete source, output variable of a Gaussian channel or
a matched lter
random vector, output vector of a Gaussian channel
mean value (expected value) of X
mean-square value of X 2
discrete-time Fourier transform of the sequence x(k)
energy spectral density of the energy signal x(t)
Fourier transform of x(t), folded spectrum
N-point DFT of the sequence x(n)
nth extension of the discrete source X
Fourier transform of xT t, 2T-truncation of x(t), T t T
z-transform of the sequence x(k), transfer function of the composite
channel in optimum MSE-LE
value assumed by a random variable X
value assumed by a random vector X
integer part of x
discrete-time signal, sequence used in OFDM
continuous-time signal (with or without a subscript)
in-phase component of the bandpass signal x(t)
set of L orthonormal basis functions, k = 1, 2, . . . , L
set of L orthonormal basis vectors, k = 1, 2, . . . , L
complex envelope (equivalent lowpass signal) of the bandpass signal x(t)
periodic signal
quadrature component of the bandpass signal x(t)
sampled function
2T-truncation of x(t), T t T
N-point DFT of the sequence y(n)
sequence
continuous-time function
pre-mapped vector at the input of the combiner
frequency samples of an OFDM signal (the IQ values of symbols of
OFDM subcarriers)
amplier transimpedance
complex envelope of the kth OFDM subcarrier
linear convolution
circular convolution
complex conjugate
arbitrary constant
xxviii
List of symbols
n
p
p
k;m
n
F
f
"l
^
"l1
"l ^
"l1
ij
t
"
"l
k
L
S
t
Xjy
X t
2
dB
2X
2s
i t
k
f
xxix
List of symbols
F0 f
ff k tg
C f
C0 f
Ct
O
Abbreviations
2G
3G
A/D
AGN
AMPS
APD
ASK
AWGN
BCH
BEC
BPA
BSC
CDM
CDMA
CDMA 2000
CP
CPM
CP-MFSK
CRC
CSI
CSIR
CSK
D/A
DD-DF
D-DF
DEMUX
DFS
DFT
DMC
DMPSK
DMQAM
DPSK
DQPSK
xxx
second generation
third generation
analog/digital conversion
additive Gaussian noise
advanced mobile phone system
avalanche photodiode
amplitude shift keying
additive white Gaussian noise
BoseChaudhuriHocquenghem code
binary erasure channel
belief propagation algorithm
binary symmetric channel
code division multiplexing
code division multiple access
3G CDMA
cyclic prex
continuous phase modulation
continuous phase M-ary frequency shift keying
cyclic redundancy check
channel side information
channel side information at the receiver
code shift keying
digital/analog conversion
double-differential decision-feedback algorithm
differential decision-feedback algorithm
demultiplexer
decision-feedback selection
discrete Fourier transform
discrete memoryless channel
differential M-ary phase shift keying
differential quadrature amplitude modulation
differential phase shift keying
differential quadrature phase shift keying
xxxi
List of abbreviations
DSB-AM
DS
DS-CSK
DS-PSK
DS-SS
DTFT
EGC
EIRP
ESN
ETACS
FCC
FDM
FDMA
FFH
FFT
FH
FIR
FM
FSE
FSK
fT
GEO
GMSK
GPS
GSM
2
jhj SNR
ICI
ICI
IDFT
IEEE
IFFT
iid
IIR
IPI
IS
ISI
ISI
JDC
JTACS
LDPC
xxxii
List of abbreviations
LFSR
LLR
LR
L-REC
L-RC
LSB
LTI
MAP
MASK
MFSK
MIMO
MIN
MIP
MISO
ML
MLSD
MMSE
MPA
MPSK
MQAM
MRC
MSC
MSE-DFE
MSE-LE
MSK
MUI
MUX
NAMPS
NRZ
NTACS
OFDM
OOK
OQPSK
PCS
PD
PDC
PDF
pdf
=4-DQPSK
PLL
xxxiii
List of abbreviations
PN
PSK
PSTN
QPSK
RS
SC
SCM
SFH
SIMO
SINR
SINR
SINR0
SIR
SIRi
SIR0
SJNR0
SJRi
SJRi; p
SJR0
SJR0; p
SNR
SNR
SNR0
SPA
TCM
TDMA
TIA/EIA
USB
USDC
VCO
WCDMA
WLAN
WMAN
WSCS
WSS
ZF-DFE
ZF-LE
pseudo-noise
phase shift keying (binary phase shift keying)
public switched telephone network
quadrature phase shift keying
ReedSolomon code
selection combining
station class mark
slow frequency hop
single-input multiple-output
signal-to-interference and noise ratio
path signal-to-interference-and-noise ratio
output signal-to-interference plus noise ratio
signal-to-interference ratio
input signal-to-interference ratio
output signal-to-interference ratio
output signal-to-jamming-plus-noise ratio
input signal-to-jamming ratio
input signal-to-pulse jamming ratio
output signal-to-jamming ratio
output signal-to-pulse jamming ratio
signal-to-noise ratio
diversity symbol signal-to-noise ratio
output signal-to-noise ratio
sum product algorithm
trellis coded modulation
time division multiple access
Telecommunication Industry Association/Electronic Industry Association
upper sideband
US digital cellular
voltage-controlled oscillator
wideband CDMA (3G CDMA)
wireless local area network
wireless metropolitan area network
wide-sense cyclostationary
wide-sense stationary
zero-forcing decision-feedback equalizer
zero-forcing linear equalizer
Introduction
If the mind is in tranquility, time and space cease to exist.
Essence of Buddhism
Introduction
message signal is to faithfully reproduce the baseband waveform from the modulated signal.
This can be done easily in the case of no noise and no equipment imperfections, but
otherwise the delity of the message signal may be reduced. Digital communication does
not involve the faithful reproduction of the baseband waveform in the presence of noise
and equipment imperfections. Digital communication operates instead with a nite set of
continuous-time modulation waveforms for transmission over a physical channel. This
implies that the message signal must be represented by a nite set of voltage baseband
waveforms. Mathematically, a nite set of waveforms can only represent a nite set
of alphabets, commonly referred to as symbols. A symbol consists of a xed number of
binary digits or bits. For example, the set of four distinct symbols {00, 01, 10, 11} can be
represented by four distinct waveforms fA cos 2fc t, A sin 2fc tg. The time separation of
consecutive waveforms that represent a symbol stream is called the symbol time, which is
the inverse of the symbol rate. If the waveforms are of nite duration then this duration is the
symbol time. This begs the question of how to obtain the bits or symbols that represent the
message. The process of converting a voltage baseband waveform that represents an audio
or video message into bits is referred to as the analog-to-digital conversion (or A/D). Text
messages generated by computers are inherently in bits, so with A/D conversion, audio,
video, text, and imagery can all be integrated into a single digital stream of bits. The process
of A/D, bit-symbol mapping, baseband waveform shaping, and modulation is referred to
as digital transmission. The process of demodulating the modulated signal, detecting the
symbol, symbolbit mapping, and digital-to-analog conversion (or D/A) is called digital
reception.
Digital communication makes no attempts to reproduce the nite set of voltage baseband
waveforms. Instead, the receiver detects the energy content of each baseband waveform in
the presence of noise and equipment imperfections, and then makes a best estimate of which
transmitted symbol was received. If the signal-to noise ratio per symbol is reasonably large,
a symbol will most likely be detected correctly with high probability. If not, a symbol error
may occur. This is the essence of digital communication. For a given signal-to-noise ratio, an
analog communication receiver attempts to reproduce the voltage baseband waveform with
certain subjective delity. On the other hand, for a given signal-to-noise ratio per symbol, a
digital communication receiver produces symbols with a quantitative error rate. It is
important to know in advance the lower bound of the signal-to-noise ratio per symbol for
a specied error rate irrespective of the type and size of the set of modulation waveforms. In
1948 Claude Shannon established this lower bound and also provided the channel capacity
for reliable transmission [1]. Shannons work gives the designers of digital communication
systems the freedom to choose the set of modulation waveforms that achieve either the
best power or bandwidth efciency, or a trade-off combination of both. As long as the
transmission rate is below the channel capacity and the signal-to-noise ratio per symbol is
above the Shannon limit, reliable communication is possible with an arbitrarily small error
rate. Guided by the Shannon channel capacity theorem (main theorem), the designer can
further integrate error-correction codes with modulation techniques to lower the signal-tonoise ratio per symbol to achieve a specied error rate. The rst error-correction code, the
Hamming code, was discovered by Richard W. Hamming in 1950, two years after Shannon
published his landmark work [2]. In addition to the main theorem, the Shannon rst theorem
1.2 Scope
Analog
message
A/D
compression
encoding
bit-symbol
mapping
Baseband
waveform
shaping
Modulation
Channel
Analog
message
Figure 1.1
Decoding
decompression
D/A
Symbol-bit
mapping
Equalization
detection
Demodulation
signal
processing
1.2 Scope
Chapter 2 provides a general study of deterministic signals that can be analyzed with Fourier
transform and Fourier series. Simple classication of signals and the concept of power and
energy are reviewed. One important class of signal, namely orthogonal signals, such as the
Walsh functions employed in IS-95, CDMA 2000, and WCDMA, is discussed in detail.
The majority of continuous-time and nite-energy signals in practice can be conveniently
analyzed via their signal spaces. These signal spaces are displays of the signal vectors in
their respective constellations. The signal vectors which can be viewed as the A/D versions
of a signal set contain all the information about the signal set. Practical communication
systems are inherently linear time-invariant systems operating in the small-signal range.
They can be analyzed by Fourier series and Fourier transform to provide a frequencydomain snapshot of the signal bandwidth. The concept of autocorrelation and its relationship
with energy or power spectral density are discussed for linear time-invariant systems.
The sampling theorem that governs the A/D conversion of an analog signal and the
Introduction
1.2 Scope
networks modeling a receiver. The system noise temperature allows the evaluation of noise
power in the receiver bandwidth, and hence the system signal-to-noise ratio. The physical
channels investigated in this chapter are the cellular and satellite channels. For the cellular
channel, we adopt the well-known Hata model to estimate the median path loss between
the transmitter and receiver. The presence of co-channel interference between cells is also
taken into account. Both narrowband cellular systems (IS-136, GSM) and wideband CDMA
cellular systems (IS-95, CDMA-2000, WCDMA) are covered. For a satellite channel,
the communication link is a point-to-point link, consisting of up- and downlinks. The
Friis formula for free-space attenuation is employed to establish the uplink or downlink
attenuation.
Chapter 6 presents modulation techniques for transmitting information over the physical
channel. The chapter essentially has two parts, namely binary modulation and M-ary
modulation. The structure of each modulation technique is studied via the signal waveform,
the power spectral density, and the modulator. For binary modulation we investigate phase
shift keying (PSK), differential phase shift keying (DPSK), amplitude shift keying (ASK)
(commonly referred to as intensity-modulated onoff keying (OOK), a technique used in
ber optic communication), frequency shift keying (FSK), minimum shift keying (MSK),
and Gaussian MSK employed by the GSM cellular standard. Many practical applications
require either the higher spectral efciency or higher power efciency that binary modulation techniques can provide; M-ary modulation can accommodate both. The second part of
this chapter covers M-ary amplitude shift keying (MASK), M-ary phase shift keying
(MPSK), offset quadrature phase shift keying (OQPSK), differential M-ary phase shift
keying (DMPSK), =4 shifted differential quadrature phase shift keying (=4-DQPSK),
M-ary quadrature amplitude modulation (MQAM), code shift keying (CSK), M-ary frequency shift keying (MFSK), and continuous phase modulation (CPM). The chapter continues with a treatment of the dominant multiplexing-modulation technique, namely
orthogonal frequency division multiplexing (OFDM), which is used in many wireless
standards. The chapter ends with a look at trellis coded modulation (TCM) for bandlimited
channels. Both Ungerboeck and pragmatic TCM are investigated.
Chapter 7 provides a treatment of digital demodulation. A generic digital demodulator
consists of two major subsystems, namely the signal processor and the detector. There are
four types of signal processor: the matched lter, the correlator, the noncoherent matched
lter, and the noncoherent correlator. The rst two types are employed in coherent demodulation while the last two types are used in noncoherent demodulation. For binary
demodulation the two fundamental detectors are threshold and maximum detectors. For
M-ary demodulation the two fundamental detectors are the minimum Euclidean distance
detector and the M-ary maximum detector. Combining the signal processor(s) and the
detector in that order produces an L-path demodulator for the set of digital signals with
L orthonormal basis functions and an M-path demodulator for the set of M orthogonal
signals. The bit error probability analysis is carried out for binary modulation techniques
such as coherent PSK, coherent DPSK, direct-detection ASK (for ber optic communication), coherent FSK, coherent MSK, precoded MSK and GMSK, noncoherent FSK and
MSK, and noncoherent DPSK. For M-ary demodulation, the bit error probability analysis is
carried out for coherent MASK, coherent MPSK, coherent DMPSK, noncoherent DMPSK,
Introduction
coherent MQAM and DMQAM, coherent CSK and MFSK, noncoherent CSK and MFSK,
coherent CPM with sequence detection, coherent CPM with symbol-by-symbol detection,
and noncoherent CPM. The chapter continues with OFDM demodulation, with emphasis on
the IEEE 802.11a,g standards. Finally, the demodulation and decoding of TCM are studied
and performance analysis is investigated. The Viterbi algorithm is again used to illustrate the
decoding process.
Chapter 8 investigates two major spread spectrum communication techniques for both
commercial and military applications: direct sequence (DS) and frequency hop (FH). The
chapter begins with a presentation of the pseudo-noise (PN) sequences needed for spreading
the modulated signal. Next the concept of quadrature orthogonal covering using Walsh
functions of the same length for multiplexing DS signals with an identical symbol rate is
discussed. This concept is then extended to variable-length orthogonal covering for variable
symbol rates. IS-95 is used as a real life example for the study of the direct sequence
spread spectrum. The demodulation of DS signals in the presence of tone jamming, broadband jamming, and pulse jamming is analyzed. Demodulation of quadrature orthogonal
covering (IS-95 forward link) as well as noncoherent DS-CSK (IS-95 reverse link) is
presented. The analysis of code division multiple access (CDMA) with random spreading
sequences is presented together with a closed form expression and a tight upper bound for bit
error probability. For frequency hop signals, three jamming strategies are studied: partialband jamming, multi-tone jamming, and follower jamming. Both slow and fast hops are
considered for follower jamming.
Chapter 9 deals with intersymbol interference (ISI) in a bandlimited channel. The Nyquist
criterion for zero ISI is stated together with the corresponding pulse shapes that satisfy it.
The design of an optimum demodulator for a bandlimited channel with Gaussian noise is
carried out. The optimum demodulator relies on the signal pulse shape implemented at the
modulator. The channel is converted to an ideal channel via an equalizer implemented at the
modulator (the equalizer is a lter with a transfer function equal to the inverse transfer
function of the channel). At the demodulator, a matched lter matched to the signal pulse
shape simultaneously achieves both the maximum signal-to-noise ratio and zero ISI as long
as the pulse shape at the matched lter output satises the Nyquist criterion of zero ISI.
In practice, because the channel transfer function is not known or varies with time, ISI
removal is instead implemented at the demodulator. The equalizer implemented at the
demodulator can be classied into two types: linear and nonlinear. The treatment of linear
equalizers covers zero-forcing and mean-square error equalizers. The latter alleviates the
noise enhancement effect that severely degrades the former in channels with deep attenuation in the passband. Nonlinear equalizers such as zero-forcing decision-feedback and
mean-square error decision-feedback can avoid the noise enhancement effect altogether,
although in channels with severe distortion the error propagation due to decision feedback
could worsen the performance. To obtain optimum performance, maximum likelihood
sequence detection may be employed to mitigate the ISI. The motivation behind sequence
detection is to use the symbol energy that resides in the ISI portion of the symbol to aid the
detection instead of throwing it away. The Viterbi algorithm is employed in practice for
sequence detection. Finally, a fractionally spaced equalizer that can mitigate timing error
is presented.
1.2 Scope
Chapter 10 studies the transmission and reception of a digitally modulated signal over a
fading channel. Unlike the AWGN channel, where noise is the only problem, a fading
channel is a greater challenge, as it may cause signal envelope variations, phase errors, and
intersymbol interference, all of which are detrimental to the performance of the signal. Thus,
it is necessary to understand the mechanism that causes these unwanted effects and nd
ways to mitigate them. A fading channel arises from the movement of the transmitter and
receiver, commonly referred to as the time-varying effect or Doppler effect. This may cause
random amplitude attenuation and random phase rotation of the signal space. In underwater
acoustic communication, the medium also changes over time, further compounding the
problem. A fading channel also exhibits a space-varying effect, where the locations of the
transmitter and receiver and the physical structures in the environment dictate the paths that
the signal may travel. A transmitted signal representing an arbitrary symbol may arrive at the
receiver via multiple paths. Depending on the time delay between paths, signal echoes of a
symbol may overlap the next several symbols causing intersymbol interference. Both timevarying and space-varying effects can be classied into four fading characteristics: slow
fading with random amplitude attenuation and negligible phase error, fast fading with
random amplitude attenuation and large phase error, at fading with random amplitude
attenuation and negligible intersymbol interference, and frequency-selective fading with
random amplitude attenuation and intersymbol interference. A mobile wireless channel may
have two of these four characteristics where the random amplitude attenuation is described
by a special distribution (Rayleigh, Rice, and Nakagami-m). For analysis, it is convenient to
model a fading channel with a channel impulse response that includes both time- and spacevarying effects. From the channel impulse response, the multipath autocorrelation and
Doppler proles are derived, which lead to the concept of the Doppler power spectrum.
ClarkeDoppler and AulinDoppler spectra are studied as examples. Using a mathematical
model, the performance of a modulated signal in a fading channel is analysed. First, ideal
coherent demodulation (assuming the carrier phase is always available for symbol-bysymbol detection) is investigated and the fading channel is assumed to produce only random
amplitude attenuation. Channel tap estimation as well as the channel tap error effect is
studied to reect real world situations.
Next, the slow fading channel with random amplitude attenuation is investigated for pilot
symbol-aided demodulation (the pilot symbols are periodically transmitted in the symbol
stream; this represents a less desirable situation than ideal coherent demodulation but
remains in line with practical applications). These investigations are extended to OFDM
(the major waveform that was adopted by IEEE 802.11a-g, 802.16 to name a few) where a
slow and frequency-selective fading channel is assumed. The fundamentals of coherent
demodulation are extended to noncoherent demodulation, where the Doppler tracking of
orthogonal signals is investigated. The next discussion centers on another major waveform
that was adopted by IS-95, CDMA 2000, and WCDMA (for use either in the forward
channel or reverse channel or both) for their respective cellular systems, namely, orthogonal
covering and spread spectrum signals. Complex spreading and despreading as well as
Doppler analysis and tracking are presented. For completeness the demodulation of the
signal used in the reverse channel of IS-95 is also presented. Once the Doppler phase error
resulting from either slow or fast fading is corrected via Doppler tracking, and the ISI
Introduction
resulting from frequency-selective fading is mitigated via OFDM and equalization, only
random amplitude attenuation remains to be dealt with. This particular effect can be
effectively alleviated via the use of time, frequency, antenna, or multipath delay diversity.
Diversity is a signal combining method that makes use of uncorrelated signal redundancy
for both transmission and reception to enhance symbol detection in the presence of a
deep fade which may destroy a non-diversity symbol. Diversity can be achieved via
redundant symbol interleaving for time diversity, or via uncorrelated subcarrier combining in OFDM for frequency diversity. It can also be achieved via multiple transmit
antennas for transmit antenna diversity, and via multiple receive antennas for receive
antenna diversity or using a Rake receiver for multipath delay diversity. Combinations of
these methods are also possible. Three main signal combining methods are studied:
maximal ratio combining (MRC), selection combining (SC), and equal gain combining
(EGC). MRC is the optimum combining scheme for coherent or pilot symbol-aided
demodulation in AWGN and is the most commonly used method for wireless LAN,
MAN, WAN, and cellular systems. It is superior to SC and EGC, although it cannot be
used for noncoherent demodulation, unlike the other two schemes. Wireless communication in a fading channel favors the use of multiple transmit antennas for performance
enhancement. This type of antenna diversity employs orthogonal space-time block codes
with rates of , , and 1 with MRC at the receiver. The Alamouti code with unity rate
was recommended for the IEEE 802.16 family. Integrating both transmit and receive
antenna diversity provides the receiver with a powerful method to combat random
amplitude attenuation.
The remaining parts of this chapter investigate the capacity of a fading channel. Since
a fade causes an outage that may drive the instantaneous channel capacity to zero, the
average channel capacity dened for an AWGN channel does not exist for a slow fading
channel. Therefore, the outage channel capacity is dened instead. The evaluation of
outage capacity for a slow fading channel, as well as slow fading channels with receive
antenna diversity (SIMO), transmit antenna diversity (MISO), both receive and transmit
antenna diversity (MIMO), and OFDM are presented. For a fast fading channel, the
average channel capacity is well dened since symbols fade independently and there
exists a coding system that ensures a maximum reliable rate, which is the ensemble
average rate. Fast fading is less detrimental than slow fading from the capacity point of
view but requires more complex channel coding to deal with both Doppler phase error
(virtually eliminated via Doppler tracking in slow fading) and random amplitude
attenuation.
1.3 Summary
The structure of the book can be summarized as follows:
*
*
Bibliography
*
*
*
*
*
Measuring and establishing the required signal-to-noise ratio for a given communication coverage:
Chapter 5.
Sending the message based on the required signal-to-noise ratio and bandwidth: Chapter 6.
Receiving the message and providing the best detection: Chapter 7.
Enhancing the survivability of narrowband modulation in the presence of interference via bandwidth
spreading, i.e., spread spectrum modulation: Chapter 8.
Pulse shaping and equalizing the effect of a bandlimited channel for modulated signals: Chapter 9.
Dealing with sending and receiving signals over a mobile channel for previously discussed modulation
techniques: Chapter 10.
Bibliography
1. C. E. Shannon, A mathematical theory of communication, Bell Syst. Tech. J., Vol. 27,
pp. 379423, pp. 62356, 1948. Also available at http://cm.bell-labs.com/cm/ms/what/
shannonday/paper.html.
2. R. W. Hamming, Error detecting and error correcting codes, Bell Syst. Tech. J.,
Vol. 29, pp. 14760, 1950.
3. R. W. Hamming, Coding and Information Theory, 2nd edition, Hemel Hempstead:
Prentice-Hall, 1986.
4. R. G. Gallager, Principles of Digital Communications, Cambridge: Cambridge
University Press, 2008.
Introduction
In this chapter we lay the foundation for the analysis and design of communication systems,
and digital communication systems in particular. We employ deterministic signals to carry
information from the transmitter to the receiver. These deterministic signals contain certain a
priori features sufciently adequate for the receiver to retrieve the information. Note that the
information always appears random to the receiver, that is, it does not know which data it
will receive; otherwise, communications would not be needed. Deterministic signals form
a very broad class of signals; therefore, the rst step is to categorize them so that their
characterization can be fully exploited. The categorization leads to the labels continuoustime, discrete-time, periodic, aperiodic, analog, digital, energy, and power signals. Further
study leads us to orthogonal signals and the use of signal space to represent digital signals as
vectors. We also review linear time-invariant (LTI) systems and the important convolution
operation that relates the inputs and outputs of an LTI system.
We then investigate Fourier series representation of continuous-time periodic signals,
and Fourier transform of continuous-time aperiodic signals. The Fourier transform is
indispensable in the analysis and design of LTI systems. The energy spectral density of an
energy signal and the power spectral density of a power signal are studied. From here the
autocorrelation functions of both energy and power signals are examined.
The process of representing a continuous-time signal by its samples is then studied using
the sampling theorem. We also discuss the process of recovering a continuous-time signal
from its samples. Finally, we study various representations of bandpass signals, which are
commonly used in the analysis of communication systems.
11
x(t)
1
f0
A
Figure 2.1
Sinusoidal signal.
x(n) = A cos
2n
12
Figure 2.2
Continuous-time signals
A continuous-time signal x(t) is a signal of the real variable t. For example, the sinusoidal
signal
xt A cos 2f0 t
(2:1)
shown in Figure 2.1 is a function of the time variable t, with amplitude A and frequency f0.
Discrete-time signals
A discrete-time signal xn is a sequence where the values of the index n are integers. For
example, the discrete-time sinusoidal signal
xn A cos 2f0 n
with f0 1=12 is shown in Figure 2.2.
(2:2)
12
(2:3)
for all time t, where T0 is the period. A signal x(t) that is not periodic is referred to as an
aperiodic or nonperiodic signal.
(2:4)
(2:5)
(2:6)
This implies that 2f0 N0 must be a multiple of 2. Thus, for an arbitrary integer m
2f0 N0 2m
(2:7)
f0 m=N0
(2:8)
or equivalently
13
iTb
s1(t)
(i + 1) Tb
iTb
A
Figure 2.3
(i + 1) Tb
s 2(t)
Antipodal signals.
1
Figure 2.4
14
s1(t)
(i + 1)Tb
iTb
Figure 2.5
(i + 1)Tb
iTb
Figure 2.6
s2(t)
A generalization of PSK is M-ary phase shift keying (MPSK) where M data symbols are
represented by M signals with distinct phase shifts. Each symbol has k bits, hence, M 2k .
The symbol time (symbol interval, symbol duration) is Ts. The parameter Rs 1=Ts is
referred to as the symbol rate. Note that a symbol has k bits; therefore, a symbol time consists
of k bit times, that is, Ts kTb . Thus, the symbol rate is 1=k times the bit rate, that is,
Rs Rb =k 1=kTb .
The MPSK signals can be written in an arbitrary symbol interval iTs t < (i + 1)Ts, where
i is an integer, as follows:
sn t A cos 2fc t 2n 1
; n 1; 2; :::; M
(2:9)
M
where the frequency fc > 1/Ts.
15
00
01
Figure 2.7
10
11
In the next section we will investigate two special types of deterministic signals, namely,
power and energy signals, and their relationship.
16
2
Vrms
R
(2:10)
1
T !1 2T
lim
T
1=2
v2 t dt
(2:11)
1 T
v2 t dt
P lim
T!1 2T T
(2:12)
(2:13)
A signal v(t) is called a power signal if and only if the average power is non-zero and nite,
that is,
05 P 5 1
(2:14)
1 T
P lim
A cos 2f0 t 2 dt
T !1 2T T
A2 T
lim
1 cos 4f0 t dt
T !1 4T T
(2:15)
A2
A2 sin 4f0 T sin 4f0 T
lim
2 T !1 4T
4f0
4f0
A2 A2
sin 4f0 T A2
lim
2
2 T !1 4f0 T
2
17
1
1
jvtj2 dt
(2:16)
A signal v(t) is called an energy signal if and only if its energy is non-zero and nite,
that is,
05 E 5 1
(2:17)
1
1
A cos 2f0 t 2 dt
(2:18)
A2
1 cos 4f0 t dt 1
1 2
The above two examples show that a power signal has innite energy. In the next two
examples we investigate some additional energy signals.
Eb
1
1
A2 dt
i 1Tb
A2 dt A2 Tb
(2:19)
iTb
The energy of s1(t) and the energy of s2(t) satisfy (2.17), hence both signals are energy
signals. However, the average power of these two signals is zero, as shown below:
1
T !1 2T
P lim
T
T
1 2
A Tb 0
lim
T !1 2T
1
T !1 2T
A2 dt lim
i 1Tb
iTb
A2 dt
(2:20)
18
i1Tb 2
A
A2
cos 4fc t dt
2
2
iTb
2
2
4fc Tb
4fc Tb
(2:21)
In practice, the frequency fc is normally chosen to be much greater than 1/Tb. Thus, the
second term on the right-hand side of (2.21) is very small compared to the rst term, and Eb
is approximately equal to
Eb
A 2 Tb
2
(2:22)
(2:23)
where k is a positive integer, then, the second term on the right-hand side of (2.21) is zero and
A 2 Tb
(2:24)
2
It is noted that Eb in (2.21) satises (2.17), hence s1(t) and s2(t) are both energy signals. It can
be shown that the average power of both s1(t) and s2(t) is zero.
Eb
From the above examples we observe that a power signal has innite energy and an energy
signal has zero average power. This is true for practical signals employed in both digital and
analog communication systems. It is quite obvious that any transmitted signal must be a power
signal because the transmitter can only have nite power. But, over an innite time, such a
power signal must have innite energy since energy is a product of power and time.
In a digital communication system, data bits or symbols represented by associated digital
signals are transmitted sequentially in time. The sequence of digital signals must therefore
be a power signal due to the nite transmitted power. Although a single digital signal must
have zero average power, an innite sequence of signals turns out to have nite power. The
next example discusses the energypower relationship.
19
p(t iTb)
(i + 1)Tb
iTb
Figure 2.8
st
xt iTb
i1
1
X
(2:25)
i
i1
where p(t iTb) is a pulse of unit amplitude and duration Tb, as shown in Figure 2.8
pt iTb
1;
0;
iTb t 5i 1Tb
otherwise
1 T 2
s t dt
P lim
T!1 2T T
"
#2
1
1 T 2 X
i
lim
A
1 pt iTb cos 2fc t dt
T !1 2T T
i1
(2:26)
(2:27)
Since the pulses p(t hTb) and p(t iTb) do not overlap for i h, the cross-terms that involve
the product p(t hTb) p(t iTb) are all zeros, and (2.27) reduces to
#
"X
1
A2 T
p2 t iT cos2 2fc t dt
P lim
T!1 2T T
i1
A2
lim
T!1 2T
A2
lim
T!1 4T
T
T
T
cos2 2fc t dt
(2:28)
1 cos 4fc t dt
A2
2
The last two lines of (2.28) follow (2.15). From (2.22) and (2.24), we have the following
relationship between the energy of the digital signals s1(t) and s2(t) and their associated
sequence s(t) in (2.25):
20
Eb PTb
(2:29)
Eb PTb
(2:30)
or
Example 2.11 shows that although the innite sequence s(t) of digital signals has nite
average power, its component digital signal must have zero average power. If its component
digital signal has nite power, an innite number of them would result in a sequence s(t)
possessing innite power. The concept can be made simpler by the following example.
T
T
v2 t dt
(2:31)
Now consider the sequence s(t) in (2.25). For simplicity let fc = k/Tb, where k is a positive
integer. By using the denition of power in (2.31) and choosing T = nTb, where n is a
positive integer, we can calculate the power of s(t) as follows:
A2 A2 sin 4fc T
A2 A2 sin 4nk
P T
4fc T
4nk
2
2
2
2
(2:32)
A2
2
Similarly, the power of the digital signal s1(t) = Ap(t iTb) cos 2fct or s2(t) = Ap(t iTb)
cos 2fct averaged over 2T = 2nTb > 4(i + 1)Tb is
P s T
1
2T
T
T
A2 p2 t iTb cos2 2fc t dt
A2 i1Tb
1 cos 4fc t dt
4T iTb
(2:33)
A 2 Tb
4T
A2
4n
(2:34)
21
Since there are 2n non-overlapping signals s1(t) and s2(t) in the interval 2T, it is obvious that
PT 2nPs T
(2:35)
where y*(t) is the complex conjugate of y(t). If, in addition, the energy of each signal dened
over the interval (a, b) is unity, that is,
Ex
jxtj dt 1; Ey
jytj2 dt 1
(2:37)
then the two signals are said to be orthonormal. A set of signals {xi (t)} is said to be
orthogonal (orthonormal) if each pair of signals in the set is orthogonal (orthonormal).
Conditions (2.36) and (2.37) can be relaxed to include quasi-orthogonal (quasiorthonormal) signals. Two signals x(t) and y(t) are said to be quasi-orthogonal over
the interval (a, b) if their inner product is much less than the energy of each signal, that is,
b
xty t dt
Ex Ey
(2:38)
In addition, if
Ex Ey 1
(2:39)
then the two signals are said to be quasi-orthonormal. A set of signals {xi (t)} is said to be
quasi-orthogonal (quasi-orthonormal) if each pair of signals in the set is quasi-orthogonal
(quasi-orthonormal).
22
T0
0
2
1
cos 2nf0 t sin 2nf0 t dt
T0
T0
T0
sin 4nf0 t dt
0
1
cos 4nf0 t T0
0
T0
4nf0
0
(2:40)
Furthermore, applying (2.37) we obtain the energy of xt and yt, dened over (0, T0 ) as
follows:
T0
T0
2
2
cos2 2nf0 t dt
sin2 2nf0 t dt
0 T0
0 T0
1 T0
1 T0
1 cos 4nf0 t dt
1 cos 4nf0 t dt
T0 0
T0 0
(2:41)
1
Thus, the two signals xt and yt are orthonormal over (0, T0 ).
The above example shows that two sinusoidal signals with a relative phase difference of
=2 radians are orthogonal over an integer multiple of their period 1=nf0 T0 =n: In
practice, the frequency of the sinusoidal signals might not be an integer multiple of 1=T0 .
In other words, n is not a positive integer but a positive real number. In this case the energy
Ex of xt and the energy Ey of yt as calculated in (2.41) become
Ex 1
sin 4n
1
4n
(2:42a)
Ey 1
sin 4n
1
4n
(2:42b)
T0
2
1 T0
cos 2nf0 t sin 2nf0 t dt
sin 4nf0 t dt
T0 0
0 T0
1
cos 4nf0 t T0
T0
4nf0
0
1 cos 4n
4n
(2:43)
Thus, the two signals xt and yt have near unity energy and their inner product is much
smaller than unity. In this case they are quasi-orthonormal.
23
Tb
Tb
Figure 2.9
T0
1
f0
(2:44)
Substituting a pair of signals xk t and xm t with k 6 m into (2.36) and setting a 0 and
b T0 yields
T0
0
T0
2
cos2kf0 t k cos2mf0 t m dt
T
0
0
1 T0
fcos2k mf0 t k m
T0 0
xk txm t dt
(2:45)
The signals in (2.45) are often employed to represent data bits or symbols. For example,
two adjacent signals xn t and xn1 t can be used to represent bit 0 and bit 1, respectively.
The parameter T0 is set to be the bit time Tb . Note that the frequencies of the two signals
are separated by f0 1=T0 1=Tb, which is also the bit rate. This is the minimum
frequency spacing between xn t and xn1 t. This modulation technique is referred to
as binary frequency shift keying or simply FSK (also referred to as BFSK or 2FSK in the
literature). In another example, M 2k adjacent signals, xn t, xn1 t, . . . , xnM t, can
be used to represent M symbols. In this case the parameter T0 is set to be the symbol
time Ts . The adjacent frequencies are separated by the minimum frequency spacing
f0 1=T0 1=Ts , which is also the symbol rate. This is called M-ary frequency shift
keying or simply MFSK. Figure 2.9 illustrates an FSK representation of two consecutive
bits 0 and 1.
24
r
2
xn;i t
cos nf0 t iT0 n;i
T0
(2:46)
r
2
xn1;i t
cos n 1f0 t iT0 n1;i
(2:47)
T0
where n 2; 3; :::, T0 1=f0 , and n;i and n1;i are the initial phases of xn;i t and xn1;i t at
time t iT0 , respectively. Note that the minimum frequency spacing between these two signals
is f0 =2. Applying these two signals to (2.36) over the interval iT0 ; i 1T0 we get
i1T0
xn;i txn1;i t dt
iT0
1
T0
i1T0
iT0
T0
f0
2n 1f0
iT0
1 sin2 i n1;i n;i sin2 i n1;i n;i
T0
f0
sin2n 1 n;i n1;i sinn;i n1;i
2n 1f0
(2:48)
The result in (2.48) shows that the two signals are not orthogonal unless the following
conditions are satised:
n1;i n:i k
(2:49a)
n1;i n:i m
(2:49b)
where k and m are integers. This implies that n;i and n1;i must be an integer multiple of
radians. In practice, it is difcult to force the phases of two sinusoidal signals at different
frequencies to satisfy (2.49). It can be seen that the second term of the last equation in (2.48)
is much smaller than the rst term. Thus, when only (2.49a) is required to be satised, then
the rst term reduces zero. Furthermore, for a large n, the second term is much smaller than
unity. Condition (2.49a) can be implemented in practice since it requires only the phase
difference to be an integer multiple of radians. It is easy to verify that the energy of the
signals is unity via (2.37). In this case, the signals xn;i t and xn1;i t are said to be quasiorthonormal over the interval iT0 ; i 1T0 .
Now consider two signals xn;i1 t and xn1;i t in two consecutive intervals
i 1T0 ; iT0 and iT0 ; i 1T0 . At time t iT0 the phase of xn;i1 t is n n;i1 ,
and the phase of xn1;i t is n1;i . It is seen that if the phases of the two signals are forced to
be continuous at the transition time t iT0 , that is, n n;i1 n1;i , then the phase
difference n1;i n;i1 must be an integer multiple of radians. This is the same condition
as (2.49a). The same result holds if the roles of the signals are reversed.
25
Tb
Tb
Figure 2.10
Equations (2.46) and (2.47) can be used to represent binary data by selecting T0 to be the
bit interval Tb and n;i n1;i . This signaling or modulation technique is called minimum
shift keying or simply MSK. It has the minimum frequency spacing of f0 =2 1=2Tb , which
is half the bit rate. Furthermore, the phase at the bit transition time is continuous and the
signals are orthonormal. Figure 2.10 shows the MSK representation of two consecutive bits
0 and 1.
In the three examples above, orthogonal sinusoids were investigated. In the following
discussion other orthogonals signals called Walsh functions will be studied. The Walsh
functions can be obtained from the Hadamard matrix, which can be generated by means of
the following recursive procedure:
H1 1
1
H2
1
1
1
HM =2
HM
HM =2
HM =2
HM =2
(2:50)
(2:51)
Note that the row vectors of the Hadamard matrix are mutually orthogonal, that is, hi htj 0
where hi is the ith row vector and htj is the jth column vector of HM .
The Walsh functions of M chips are given as follows:
wn t
M
X
(2:52)
i1
where Tc is the chip time, pt i 1Tc is a pulse of unit amplitude and duration Tc ,
i 1Tc t 5 iTc , and hni is the element n; i of HM . Since Walsh functions are formed
from the orthogonal row vectors of the Hadamard matrix, they are also mutually orthogp
onal. A normalization constant 1= MTc can be used in the Walsh functions to form an
orthonormal set. The orthonormality of Walsh functions can be veried by using (2.36)
and (2.37).
26
w1 (t)
w3(t)
4Tc
Figure 2.11
w4(t)
w2 (t)
4Tc
2Tc
Tc
2Tc
3Tc
4Tc
t
0
Tc
3Tc
4Tc
Another extension of the Walsh functions referred to as the sinusoidal Walsh functions is
dened as follows:
M
X
wn t
hni pt i 1Tc cos 2fc t; n 1; 2; :::; 2n
(2:54)
i1
where fc m=Tc and m is a positive integer. It can be veried that the sinusoidal
Walsh
p
functions form an orthogonal set by using (2.36). A normalization constant 1= MTc =2 can
be used in the sinusoidal Walsh functions to form an orthonormal set.
The Walsh functions and the sinusoidal Walsh functions can be used to represent M 2k
k-bit symbols. The cellular standard IS-95 employs 64-chip Walsh functions in both
forward (base station-to-mobile) and reverse (mobile-to-base station) channels.
27
w1 (t)
w3 (t)
4Tc
w2 (t)
w4 (t)
4Tc
Figure 2.12
4Tc
4Tc
GramSchmidt procedure
Let si(t), i = 1, 2, . . . , M be a set of real-valued signals with corresponding nite
energies Ei. In terms of representation, each signal in the set can be expressed as a linear
combination of L orthonormal basis functions xk(t), k = 1, 2, . . . , L, where L M .
When the signals si(t), i = 1, 2, . . . , M form a linearly independent set, then L = M. We
have
si t
L
X
sik xk t
(2:55)
k1
(2:56)
28
The orthonormal basis functions can be obtained by means of the following recursive
procedure:
s1 t
s1 t
p
(2:57)
x1 t q
1 2
E1
1 s1 t dt
f2 t
;
x2 t q
1 2
f
tdt
1 2
fk t
;
xk t q
1 2
f
tdt
1 k
f2 t s2 t s21 x1 t
fk t sk t
k1
X
skm xm t;
(2:58)
kL
(2:59)
m1
Signal vectors
From the GramSchmidt procedure we observe that for each orthonormal basis function
xk(t) there is a corresponding orthonormal basis vector xk:
2 3
2 3
2 3
1
0
0
607
617
607
6 7
6 7
6 7
6:7
607
6:7
7
6 7
6 7
(2:60)
x1 6
6 : 7; x2 6 : 7; . . .; xL 6 : 7
6 7
6 7
6 7
4:5
4:5
4:5
0
0
1
The set of orthonormal basis vectors xk spans an L-dimentional Euclidean signal space. In this
signal space we observe that each signal si(t) is completely determined by the signal vector si:
2 3
si1
6 si2 7
6 7
L
X
6 : 7
7
si 6
sik xk ;
i 1; 2; . . . ; M
(2:61)
6 : 7
6 7
k1
4 : 5
siL
We may visualize the set of M signal vectors as dening a set of M points in the signal space
with L mutually perpendicular axes labeled x1, x2, . . . , xL. The signal vector contains all the
information such as energy and relative phase shift or orthogonality needed to analyze the
performance of the signal.
The energy of a signal turns out to be the squared length or squared norm of its signal
vector as shown below. First consider the length or norm of a signal vector si of the signal
si(t), dened as
v
u L
uX
(2:62)
s2ik
k si k t
k1
29
Now consider the signal energy Ei, evaluated with the help of (2.55):
Ei
1
1
s2i t dt
L X
L
X
1 X
L X
L
1 k1 m1
1
sik sim
k1 m1
1
xk txm tdt
L
X
(2:63)
s2ik
k1
The last expression on the right-hand side of (2.63) results from the fact that xk(t) and xm(t)
form a pair of orthonormal functions, and therefore
1
1; k m
xk txm tdt
(2:64)
0; k 6 m
1
Comparing (2.63) and (2.62) we obtain
E i k si k 2
(2:65)
Another useful quantity for signal analysis is the Euclidean distance between the two signal
vectors:
v
u L
uX
s i sj t
(2:66)
sik sjk 2
k1
The Euclidean distance indicates the degree of separation between two signals in the signal
space.
(2:67)
Furthermore,
s11
s1 tx1 tdt
1
p
A Tb
s2 tx1 tdt
Apt iTb p pt iTb dt
Tb
1
1
p
A Tb
1
s21
1
Apt iTb p pt iTb dt
Tb
1
(2:68)
(2:69)
30
Eb
Figure 2.13
Eb
x1
p
A Tb
x2 t Apt iTb p pt iTb 0
Tb
(2:70)
In addition, we also get s12 s22 0. Thus, the signal space is a one-dimensional space.
The signal vectors are given by
p
p
p
s1 A Tb Eb Eb x1
(2:71)
p
p
p
s2 A Tb Eb Eb x1
(2:72)
It is seen that the signal energy is equal to the squared length of the signal vector. The
Euclidean distance between the two signal vectors is given by
p
(2:73)
ks1 s2 k 2 Eb
The signal space for the set of antipodal signals is given in Figure 2.13.
(2:74)
x2 t 0
(2:75)
By using (2.56) the coefcients of the signal vector can be calculated as follows:
s11
r
Tb
A
;
2
s12 0 ;
s21
r
Tb
A
;
2
s22 0
(2:76)
Thus the signal space is a one-dimensional space. The signal vectors are given by
s1 A
p
p
p
Tb =2 Eb Eb x1
p
p
p
s2 A Tb =2 Eb Eb x1
(2:77)
(2:78)
31
It is seen that the signal energy is equal to the squared length of the signal vector. The
Euclidean distance between the two signal vectors is given by
k s1 s 2 k 2
p
Eb
(2:79)
The signal space of PSK signals is therefore identical with that of the antipodal signals given
in Figure 2.13, although their waveforms differ.
For many digital signal sets, the orthonormal basis functions and the signal vectors can be
found by inspection without going through the GramSchmidt procedure. The following
example illustrates this.
p
Tb
2
Tb
s1 t A
cos 2f1 t A
x1 t Eb x1 t
Tb
2
2
r r
r
p
Tb
2
Tb
cos 2f2 t A
s2 t A
x2 t Eb x2 t
Tb
2
2
(2:80)
(2:81)
p
2
where Eb Ap
T
b =2 is
the signal energy, and the two functions x1 t 2=Tb cos 2f1 t
and x2 t 2=Tb cos 2f2 t form a set of orthonormal functions over the
interval (iTb ; i 1Tb ). The signal vectors are displayed in Figure 2.14 and given as
follows:
x2
Eb
Figure 2.14
Eb
x1
32
p
Eb
Eb x1
0
(2:82)
p
0
s2 p Eb x2
Eb
(2:83)
s1
p
2Eb
(2:84)
(2:86)
r
r
h
Ts
i
2
cos 2fc t
cos 2n 1
M
Ts
2
r h
r
i
Ts
2
(2:87)
sin 2n 1
sin 2fc t
A
2
M
Ts
h
p
p h
i
i
x1 t Es sin 2n 1
x2 t
Es cos 2n 1
M
M
p
Ts =2
is the signal energy, and the two functions x1 t 2=Ts cos 2fc t
where Es A2p
and x2 t 2=Ts sin 2fc t form a set of orthonormal functions over the symbol time
(iTs ; i 1Ts ). The signal vectors are given as follows:
sn t A
2 p
h
i3
Es cos 2n 1
6
Mi 7
h
sn 4 p
5
Es sin 2n 1
M
h
h
p
p
i
i
x1 Es sin 2n 1
x2
Es cos 2n 1
M
M
(2:88)
Thus the MPSK signal space is a two-dimensional space. The Euclidean distance between
the two adjacent MPSK signals is given by
33
x2
Es
Figure 2.15
x1
i i2 1=2
Es sin 2k 1
Es sin 2m 1
M
M
h h
h
i
i
2Es 2Es cos 2k 1
cos 2m 1
M
M
h
i
h
ii1=2
sin 2k 1
sin 2m 1
M
M
(2:89)
Consequently,
s
2
ksk sm k 2Es 1 cosk m
M
(2:90)
The Euclidean distance between two adjacent MPSK signal vectors corresponds to the case
of jk mj 1, that is,
s
p
2
(2:91)
2 Es sin
ksk sk1 k ksk sk1 k 2Es 1 cos
M
M
Figures 2.15 and 2.16 show the signal spaces of QPSK and 8PSK, respectively.
From Examples 2.18 and 2.19 we observe that antipodal signals and PSK signals have the
identical signal space. Therefore, these two sets of signals must have the same noise
performance. From Examples 2.19 and 2.20 it is seen that PSK has a larger Euclidean
distance than FSK. It turns out that a larger Euclidean distance corresponds to a better noise
performance. A PSK signal is less likely to be mistaken by the other PSK signal in noise than
its FSK counterpart.
34
x2
Figure 2.16
Es
x1
(2:92)
Now consider a second input obtained by shifting i(t) by t0, that is, it t0 . The corresponding output v1(t) is
v1 t Rit t0 vt t0
(2:93)
(2:94)
35
2.6 Convolution
(2:95)
(2:96)
(2:97)
yn n0 n n0 xn n0
(2:98)
^yn 6 yn n0
(2:99)
Thus
(2:100)
is
y
a k y k a1 y 1 a 2 y 2
(2:101)
where yk, k = 1, 2, . . . , is the response to xk, k = 1, 2, . . . . Equations (2.100) and (2.101) hold
for both continuous-time and discrete-time systems.
2.6 Convolution
The output of a continuous-time LTI system can be expressed in terms of the input x(t) and
the impulse response h(t) as a convolution integral:
y t x t h t
1
1
1
1
x ht d
(2:102)
h xt d ht xt
36
x()
h()
1
0
(a)
T
(c)
h(t )
T t < 2T
t<0
h(t )
0t<T
tT 0 t
1
0 tT
(d)
Figure 2.17
T
(b)
h(t )
1
h( )
tT
(e)
t 0
(f)
For a discrete-time system, the convolution integral in (2.102) becomes the convolution
sum
yn xn hn
1
X
xkhn k
k1
1
X
hkxn k
(2:103)
k1
1
1
xt h d ht xt
(2:104)
1; 0 t 5T
0; otherwise
(2:105)
Figure 2.17 depicts the evaluation of the output y(t) as a convolution of x(t) and h(t). It is
seen that the product of x and ht is zero for t < 0, and for t > 2T. For 0 t 5T the
value of the product is 1 in 0 t:
37
y(t)
T
t
0
Figure 2.18
2T
Response y(t).
y t
x ht d
d t;
0 t 5T
(2:106)
(2:107)
tT
(2:108)
It is easy to see that x1(t + T0) = x1(t), where T0 = 1/f0, as shown below:
x1 t T0 ej2f0 tT0 ej2f0 tj2
ej2f0 t x1 t
(2:109)
The period of x1(t) is T0. Associated with x1(t) is an innite set of periodic harmonic
components:
xk t ej2kf0 t ;
k 0; 1; 2; . . .
(2:110)
The period of the kth harmonic xk(t) is T0/k and its frequency is kf0. It is obvious that xk(t) is
also periodic with period T0, as seen below:
38
x(t)
T0
Figure 2.19
Tp 0 T p
T0
(2:111)
ej2kf0 t xk t
Furthermore, a linear combination x(t) of the harmonics xk(t) is also periodic with
period T0:
x t
1
X
ak ej2kf0 t
(2:112)
xt ej2kf0 t dt
(2:113)
k1
1
ak
T0
T0
0
The representation of a periodic signal x(t) in the form of (2.112) is termed the Fourier series
representation. By multiplying both sides of (2.112) by ej2kf0 t and integrating from 0 to T0
we obtain the complex Fourier series coefcient ak in (2.113).
We note that the coefcient a0 is given by
a0
1
T0
T0
xt dt
(2:114)
Thus, a0 represents the average value (dc value) of x(t) averaged over one period T0. The
other ak coefcients represent the portion of x(t) in each harmonic component. Specically,
the magnitude |ak| of the complex Fourier series coefcient ak is the amplitude of the kth
harmonic component of x(t).
1
T0
T0
0
A dt
1
T0
T0 =2
T0 =2
A dt
1
T0
Tp =2
Tp =2
A dt
ATp
T0
(2:115)
39
ak
k
T0
01 1 2
2 1
Tp Tp
T0 Tp Tp
Figure 2.20
ak
1
T0
T0 =2
T0 =2
A ej2kt=T0 dt
1
T0
Tp =2
Tp =2
A ej2kt=T0 dt
sinkTp =T0 ATp sinkTp =T0
;
A
T0
k
kTp =T0
i
A h jkTp =T0
ejkTp =T0
e
j2k
k 6 0
(2:116)
The Fourier series coefcients in this case are real and assume the form of the well-known
function sin u=u. Figure 2.20 shows a plot of ak as a function of the discrete-frequency
kf0 k=T0 , together with their envelope, for T0 2Tp :
1 nT0 =2
n T0 =2
1 T0 =2
jxtj2 dt lim
jxtj2 dt
jxtj2 dt
n!1 nT0 nT =2
n!1 nT0 T =2
T
0
T
=2
0
0
0
T0
T0
1
1
X
X
1
1
1 T0
xtx t dt
xt
ak ej2kf0 t dt
ak
xtej2kf0 t dt
T0 0
T0 0
T
0
0
k1
k1
P lim
1
X
k1
ak ak
1
X
jak j2
(2:117)
k1
Note that jak j2 is the average power in the kth harmonic component ak ej2kf0 t of x(t).
40
u(t)
Figure 2.21
Figure 2.22
(t t0)
t0
0; t 50
1; t 40
(2:118)
1; z 0
0; z
6 0
(2:119)
ut
and is shown in Figure 2.21.
The unit impulse function is dened as
z
and
1
1
z dz 1
(2:120)
The variable z can be time or frequency depending on the application. The unit impulse
function has the following properties:
1
xzz dz x0
(2:121)
1
xzz z0 dz xz0
(2:122)
xzz z0 xz0 z z0
(2:123)
1
ut
d ut
dt
t
1
z dz
(2:124)
(2:125)
Figure 2.22 shows the unit impulse as a function of time at location t 0 and t t0 :
41
Sx ( f )
Figure 2.23
2
1 01 1 2
Tp Tp T0 Tp Tp
1
1
Sx f df
(2:126)
Combining (2.117) and (2.126) we obtain the power spectral density as an innite series of
impulses occurring at the harmonically related frequencies, and for which the area of the
impulse at the kth harmonic frequency kf0 is jak j2 :
1
1
X
X
k
Sx f
(2:127)
jak j2 f kf0
jak j2 f
T0
k1
k1
Figure 2.23 shows the power spectral density Sx f , and its envelope, of the periodic pulse
signal in Example 2.26.
From (2.127) we observe that the power spectral density of a real-valued periodic signal is
a nonnegative real-valued and even function of frequency, that is,
Sx f 40;
for all f
Sx f Sx f
(2:128)
(2:129)
A signal with a power spectral density concentrated around the zero frequency is referred to
as a baseband or lowpass signal. This denition applies to power signals that include
periodic signals. The periodic pulse signal in Example 2.26 is a baseband signal.
Bandwidth
The power spectral density allows us to determine the frequency range that contains most of
the power of the signal. The positive part of this frequency range denoted as B is loosely
dened as the bandwidth of the signal. For example, the frequency range B f B that
42
contains most of the power of the periodic pulse signal shown in Figure 2.23 is the main
lobe. This is called the rst-null bandwidth, and is equal to B 1=Tp . It is seen that a
narrower pulse width Tp results in a wider rst-null bandwidth.
(2:130)
where the shifting frequency fc nf0, and n is a positive integer. We observe that
yt T0 xt T0 ej2nf0 tT0
xt ej2nf0 t yt
(2:131)
Therefore, y(t) is also periodic with period T0 . The Fourier series representation of y(t) is
given by
y t
1
X
bk ej2kf0 t
(2:132)
yt ej2kf0 t dt
(2:133)
k1
bk
1
T0
T0
0
1 T0
bk
xt ej2knf0 t dt
T0 0
(2:134)
The right-hand side of (2.134) is just akn as viewed from (2.113). Thus
bk akn
(2:135)
In other words, bn1 a1 ; bn1 a1 ; bn a0 , and so on. Hence, the harmonic components of y(t) are simply the harmonic components of x(t) shifted in frequency by fc nf0.
Similarly, the Fourier series coefcients of the signal
yt xt ej2fc t xt ej2nf0 t
(2:136)
bk akn
(2:137)
is given by
43
y(t)
T0
Tp
2
Tp
2
T0
Figure 2.24
(2:138)
It is seen that y(t) is periodic with period T0 . The Fourier series representation of y(t) is given
by (2.132) and (2.133) with the following coefcients:
1
bk akn akn
2
(2:139)
ATp sink nf0 Tp sink nf0 Tp
k nf0 Tp
k nf0 Tp
2T0
(2:140)
44
bk
Figure 2.25
fc
fc
k
T0
1
X
jbk j2
(2:141)
k1
1
X
jbk j2 f kf0
(2:142)
k1
1
1 X
jakn akn j2 f kf0
4 k1
1 h
i
1 X
jakn j2 Reakn akn f kf0
4 k1
1 h
i
1 X
jai j2 Reai ai2n f i nf0
4 i1
1 h
i
1 X
jakn j2 Reakn akn f kf0
4 k1
(2:143)
1 h
i
1 X
jam j2 Ream2n am f m nf0
4 m1
1 h
i
1 X
jai j2 Reai ai2n f fc if0
4 i1
1 h
i
1 X
jam j2 Ream2n am f fc mf0
4 m1
45
Sy( f )
fc
fc
Figure 2.26
1
Tp
fc
fc +
1
Tp
Sy f
1
1
1 X
1 X
jai j2 f fc if0
jam j2 f fc mf0
4 i1
4 m1
(2:144)
1
Sx f fc Sx f fc
4
The power spectral density of y(t) is shown in Figure 2.26 for T0 2Tp.
The bandwidth of y(t) is dened loosely as the positive frequency range around fc that
contains most of the power of y(t). The null-to-null bandwidth B is the frequency range
from fc 1=Tp to fc 1=Tp , that is, B 2=Tp . The signal y(t) is referred to as a bandpass
or passband signal. It is seen that the power spectral density of y(t) is one-fourth the
power spectral density of x(t) shifted to the positive frequency range by fc and one-fourth
the power spectral density of x(t) shifted to the negative frequency range by fc for
fc 1=Tp . The periodic signal x(t) is referred to as the equivalent baseband or lowpass
signal of y(t).
The above study shows that Fourier series representation is an indispensable tool for
analyzing periodic signals in the frequency domain via their harmonic components or their
power spectral density. The frequency domain approach allows us to determine the frequency range that contains most of the signal power. This frequency range is dened loosely
as the bandwidth of the signal. In the next section we will investigate the response of an LTI
system to a periodic input signal.
1
1
x ht d
1
1
h xt d
(2:145)
46
Let x(t) be a periodic signal with the Fourier series representation given in (2.112).
Substituting (2.112) into (2.145) we obtain
yt
1
h
1
1
X
k1
1
X
ak e
j2kf0 t
ak ej2kf0 t d
1
(2:146)
h e
j2kf0
1
k1
(2:147)
1
Note that the transfer function is a function of the frequency f. By using (2.147) in (2.146)
we get
yt
1
X
Hkf0 ak ej2kf0 t
(2:148)
k1
From (2.148) we note that the Fourier series coefcients of y(t) are the Fourier coefcients of
x(t) weighted by the values of the transfer function at the harmonic frequencies.
Consequently, the power spectral density function of y(t) is given by
Sy f
1
X
(2:149)
k1
Substituting (2.127) into (2.149), and with the help of (2.123), we obtain the following
relation between the power spectral density Sy f of the output signal y(t) of an LTI system
and the power spectral density Sx f of the periodic input signal x(t):
Sy f jHf j2 Sx f
(2:150)
The result in (2.150) is very important because it enables the design of LTI lters to pass
certain desirable harmonic components and to eliminate undesirable ones.
Example 2.29 Periodic pulse signals and the ideal lowpass lter
Let the periodic pulse signal x(t) in Example 2.26 be the input to an ideal lowpass lter with
the following transfer function:
(
1; T1p f T1p
Hf
(2:151)
0;
otherwise
A plot of H(f ) is shown in Figure 2.27(a). The ideal lowpass lter passes all the harmonic
components of x(t) within the main lobe. Figure 2.27(b) shows the power spectral density of
y(t) and its envelope, for T0 2Tp :
47
Sy( f )
H (f)
1
1
Tp
1 01 1
Tp T0 Tp
1
Tp
(a)
Figure 2.27
(b)
(a) Ideal lowpass lter; (b) power spectral density of output signal.
x(t)
t
0
(a)
xp(t)
T0
T0
(b)
Figure 2.28
(a) Aperiodic signal x(t); (b) periodic signal xP(t), constructed to be equal to x(t) over one period T0.
48
1
X
xp t
ak ej2kf0 t
(2:152)
xp t ej2kf0 t dt
(2:153)
k1
1
ak
T0
T0 =2
T0 =2
T0 =2
T0 =2
xp t ej2kf0 t dt
(2:154)
xp t
X kf0 ej2kf0 t f0
(2:155)
k1
X f
xtej2ft dt
(2:157)
1
The above two equations are referred to as the Fourier transform pair. The function X f is
called the Fourier transform of the signal xt, and xt is the inverse Fourier transform of
X f . The following notations are often used:
xt !X f
(2:158)
X f Ffxtg
(2:159)
xt F 1 fX f g
(2:160)
jxtj dt 51:
2. x(t) has a nite number of maxima and minima within any nite interval.
3. x(t) has a nite number of nite discontinuities within any nite interval.
Another sufcient condition for the existence of the Fourier transform is that the signal x(t) is
a nite energy signal, that is, it is square integrable, so that
49
X( f )
x(t)
ATp
Tp
0 1
1
Tp
Tp
Tp
2
2
(a)
Figure 2.29
(b)
jxtj2 dt 51:
Tp
2
A;
0;
otherwise
t
Tp
2
(2:161)
as shown in Figure 2.29(a). Applying (2.157) we obtain the Fourier transform of x(t) as
X f
Tp =2
Tp =2
ATp
Aej2ft dt A
ejfTp ejfTp
j2f
sin fTp
fTp
(2:162)
The plot of X(f) is shown in Figure 2.29(b). Note that T0 X f is exactly the
envelope of the Fourier series coefcients of the periodic pulse signal x(t) shown in
Example 2.26.
50
x(t)
X( f )
2B
0 1
1
2B
2B
(b)
(a)
Figure 2.30
xt
B
B
ej2ft df
ej2Bt ej2Bt
j2t
sin 2Bt
2B
2Bt
(2:164)
(2:165)
1
1
1
yt ej2ft dt
xt ej2f fc t dt
(2:166)
1
X f fc
Similarly, the Fourier transform of the signal
yt xt ej2fc t
(2:167)
Y f X f fc
(2:168)
is given by
51
y(t)
Tp
Tp
2
2
A
(a)
Y( f )
ATp
2
fc
fc
fc
1
Tp
fc +
1
Tp
(b)
Figure 2.31
A cos 2fc t;
0;
Tp
2
t
Tp
2
(2:169)
otherwise
as shown in Figure 2.31(a). We note that y(t) is simply the product of x(t) in (2.161) and
cos 2fc t, that is,
yt xt cos 2fc t
1
xt ej2fc t xt ej2fc t
2
(2:170)
The Fourier transform of y(t) as sketched in Figure 2.31(b) can be obtained from (2.166) and
(2.168) as
52
X (f)
Figure 2.32
2
1 0
Tp Tp
f
1 2
Tp Tp
(2:171)
Parseval relation
For aperiodic signals such as energy signals, the Parseval relation is the direct counterpart
of the Parseval relation for periodic signals. It expresses the signal energy in terms of the
Fourier transform of the signal. If we consider an energy signal x(t), its energy is given by
1
1
2
E
xtx t dt
jxtj dt
1
1
1
1
j2ft
xt
X f e
df dt
(2:172)
1
1
1
1
X f
xtej2ft dt df
1
1
The bracketed term in the last line is the Fourier transform of x(t), thus
1
1
2
E
jxtj dt
jX f j2 df
1
(2:173)
1
The quantity jX f j2 is referred to as the energy spectral density or energy spectrum of the
signal x(t), and displays the distribution of the energy in the frequency domain. The energy
spectral density of the pulse signal x(t) in Example 2.30 is shown in Figure 2.32. This is a
baseband signal with the rst-null bandwidth equal to 1=Tp .
The energy spectral density of the sinusoidal pulse signal in Example 2.32 is obtained
from (2.171) as follows:
1
jY f j2 jX f fc X f fc j2
4
1
jX f fc j2 jX f fc j2
4
(2:174)
53
Y( f )
fc
fc
fc 1
Tp
Figure 2.33
fc +
1
Tp
1 k1
1
1
X
ak
k1
(2:175)
e
j2kf0 t j2ft
dt
1
(2:176)
1
Note that the left-hand side of (2.176) is simply the inverse Fourier transform of f kf0 ,
therefore we conclude that the integral inside the summation on the right side of (2.175) is
just the Fourier transform of ej2kf0 t , which is precisely f kf0 . Thus
X f
1
X
ak f kf0
(2:177)
k1
54
yt
1
1
xht d
(2:178)
1 1
xht d ej2ft dt
Y f
1
1
1
1
j2ft
x
ht e
dt d
1
1
1
1
0 j2f t 0
0
x
ht e
dt d
1
1
1
1
j2f
0 j2ft 0
0
x e
ht e
dt d
1
1
1
1
0
ht 0 ej2ft dt 0
x ej2f d
1
(2:179)
1
The above integrals are just the Fourier transforms H(f ) and X(f ) of h(t) and x(t), respectively. Note that H(f ) is simply the transfer function of the LTI system as dened in (2.147).
Thus
Y f Hf X f
(2:180)
yt ht xt
(2:181)
Equation (2.180) provides a convenient means to evaluate the frequency response of an LTI
system. It also allows the design of LTI lters to shape the output signal.
Y( f )
ATp
Figure 2.34
1 0
Tp
f
1
Tp
Fourier transform of the output signal of an ideal lowpass lter with a pulse signal input.
55
Y f Hf
ak f kf0
k1
1
X
(2:182)
Hf ak f kf0
k1
1
X
Hkf0 ak f kf0
(2:183)
k1
Now taking the inverse Fourier transform of both sides of the above equation and noting that
the inverse Fourier transform of f kf0 is ej2kf0 t as given in (2.176) we obtain
yt
1
X
Hkf0 ak ej2kf0 t
(2:184)
k1
This is exactly the same expression for y(t) given in (2.148) by the convolution operation.
(2:185)
where t0 is the time delay through the system and A is an arbitrary constant. Taking the
Fourier transform of (2.185) we obtain
1
Axt t0 ej2ft dt
Ax ej2f t0 d
1
1
1
j2ft0
x ej2f d A ej2ft0 X f
Ae
Y f
(2:186)
1
Comparing (2.180) and (2.186) we observe that a distortionless transmission occurs if and
only if
Hf jHf j ejf A ej2ft0
(2:187)
56
Signal
Fourier transform
Conjugate symmetry
Linearity
xt real
axt byt
Time scaling
xat
Time reversal
Conjugation
Time shifting
Frequency shifting
Modulation
Convolution
Multiplication
Duality
xt
x t
xt t0
xtej2fc t
xt cos 2fc t
xt ht
xtyt
X t
dn xt
dtn
t n xt
t
1 xt dt
X f X f
aX f
bY f
1
f
X
a
jaj
X f
X f
X f ej2ft0
X f
fc
1
2 X f fc X f fc
X f Hf
X f Y f
xf
Differentiation
Differentiation in frequency
Integration
j2f n X f
dn X f
j2n
n
X f 1 df
X 0f
j2f
2
where jHf j and f are the magnitude and phase of the transfer function H(f ), respectively. Equation (2.187) indicates that the transfer function magnitude must be a constant,
and the transfer function phase must be a linear function of frequency, that is,
jHf j A
(2:188)
f 2t0 f
(2:189)
2.9 Autocorrelation
We have seen the important role of energy spectral density and power spectral density in
determining the bandwidth of energy signals and power signals, respectively. In this section
we investigate the counterparts of these functions in the time domain.
57
2.9 Autocorrelation
Fourier transform
Unit impulse
Unit step
t
ut
Unit impulse at t t0
Constant
t t0
1
1; t 40
sgnt
1; t 50
Signum
Hilbert transformer
Impulse train
Complex exponential
Cosine
Sine
Gaussian pulse
Rectangular pulse
(sinz)/z pulse
One-sided exponential
Two-sided exponential
Parseval formula:
Poisson sum formula
e
f
1
jf
jsgnf
1
P
1
f Tk
T
1
t
t kT
k1
1
P
Periodic signal
1
1
j2f 2 f
j2ft0
k1
1
P
ak ej2kf0 t
k1
j2f0 t
e
cos 2fc t
sin 2fc t
2
et=T
A;
2p t
0;
otherwise
sin 2Bt
2B
2Bt
eat ut; a40
xt
ak f kf0
k1
f f0
1
2j f fc f
1
2j f 2fc f
T efT
sin fTp
ATp
fT
p
1;
X f
0;
Tp
2
fc
fc
B f B
otherwise
1
aj2f
2a
a2 2f 2
1
aj2f k
1
1
1
1
X f X f ej2f df
1
1
1
X f
xt
X f ej2f t df dt
xt ej2ft dt ej2f df
1
(2:190)
1
or
Rx
1
1
xtxt dt
(2:191)
58
The function Rx is referred to as the autocorrelation of the energy signal x(t). It provides a
measure of the similarity between the signal and its delayed version. An important property
of the autocorrelation is that the energy E of the signal x(t) is equal to the value of Rx at the
delay time 0:
1
x2 t dt Rx 0
(2:192)
E
1
Thus, the autocorrelation of a real-valued energy signal is a real-valued even function, that
is,
Rx Rx
(2:194)
Furthermore, the maximum value of the autocorrelation occurs at the origin, that is,
jRx j Rx 0 for all
(2:195)
The inequality in (2.195) holds because the similarity between the signal x(t) and its delayed
version xt is reduced as is increased. We can prove (2.195) by considering the
following inequality:
1
xt xt 2 dt 0
for all
(2:196)
1
Equivalently, we have
1
1
x2 t dt
1
1
x2 t dt 2
1
1
xtxt dt
1
xtxt dt for all
2
(2:197)
1
The left-hand side of (2.197) is simply the sum of Rx 0 and Rx 0 and the right-hand side is
just 2jRx j. This proves (2.195).
Now consider an LTI system with input x(t), transfer function H(f ), and corresponding
real-valued impulse response h(t). We are interested in the energy spectral density of the
output signal y(t) whose Fourier transform is Y(f ). Let the input signal x(t) be an energy
signal with the Fourier transform X(f ). From (2.180) we obtain
jY f j2 jHf j2 jX f j2
(2:198)
Taking the inverse Fourier transform of both sides of (2.198), and noting that
jHf j2 H f Hf , we have
Ry h h Rx
(2:199)
where Ry and Rx are the autocorrelation functions of y(t) and x(t), respectively.
59
2.9 Autocorrelation
1
X
jak j2 f kf0
(2:200)
k1
1
X
jak j2 ej2kf0
(2:201)
k1
We observe that the power P of the periodic signal x(t) is equal to Rx 0 via (2.117), that is,
P Rx 0
1
X
jak j2
(2:202)
k1
For a real-valued signal x(t) that is not periodic, the power spectral density must be derived
separately. From (2.13) the average power of x(t) is given by
1
T !1 2T
P lim
T
T
1
T !1 2T
x2 t dt lim
1
1
T t T
otherwise
x2T t dt
(2:203)
(2:204)
Although the power signal x(t) might not have a Fourier transform because its energy is
innite, its truncated version xT t has nite energy and hence possesses a Fourier transform
XT f . By using the Parseval relation for (2.203) we have
1 1
(2:205)
P lim
jXT f j2 df
T !1 2T 1
Since the average power P is nite, we can exchange the order in which the limit and the
integration are performed:
1
1
P
lim
(2:206)
jXT f j2 df
T!1
2T
1
If we let Sx f be the power spectral density of the power signal x(t), by denition we have
1
1
1
Sx f df
lim
(2:207)
P
jXT f j2 df
T
!1
2T
1
1
60
Sx f lim
T!1 2T
jXT f j2
(2:208)
The function jXT f j2 =2T is commonly referred to by statisticians as the periodogram of the
power signal.
If we denote the inverse Fourier transform of Sx f as Rx , we have
1
1
lim
Rx F 1 fSx f g
jXT f j2 ej2f df
1 T !1 2T
(2:209)
1 1
lim
jXT f j2 ej2f df
T !1 2T 1
Using the relation that leads to (2.190) we obtain
1 1
Rx lim
xT txT t dt
T !1 2T 1
(2:210)
Rx lim
T
T
xtxt dt
(2:211)
The function Rx in (2.211) is referred to as the autocorrelation of the power signal x(t).
Note that the autocorrelation of the periodic signal given in (2.201) is a special case of
(2.211). In fact, (2.201) can be obtained by substituting (2.112) directly into (2.211). We also
observe that the average power P of the power signal x(t) is equal to Rx 0 by virtue of
(2.203) and (2.211). Thus, the average power of the power signal x(t) is equal to the
autocorrelation at the origin, that is,
P Rx 0
(2:212)
Furthermore, the autocorrelation of a power signal also satises (2.194) and (2.195).
The autocorrelation reects the degree of coupling between a signal and its delayed
version. If the signal xt varies slowly, the signals delayed version xt looks almost
like the signal as the delayed interval gets larger. In the extreme case of a dc (constant)
signal, the autocorrelation is a constant, meaning that the power spectral density is an
impulse function at zero frequency. This is expected since all power must be concentrated
at zero frequency. When xt varies rapidly then its delayed version xt can be very
much different to xt within a short delayed interval , and the autocorrelation can go down
to zero. In this case the signal may contain many harmonics at various frequencies. The
high-frequency harmonics are the rapidly time-varying signals and may contain a large
amount of power. In this case, the power spectral density may have high weighting at
the high frequencies, which means that the signal has a large bandwidth. In the extreme case,
the signal varies so rapidly that its autocorrelation is an impulse function at time zero. This
implies that the power spectral density is a constant, and the signal has innite bandwidth
and innite power.
61
2.9 Autocorrelation
Ry lim
T
T
ytyt dt
(2:213)
T 1
T
1
huxt u du
1
1
hvxt v dv dt
1 1
1 T
hu hv lim
x x u v d du dv
Ry
T !1 2T T
1 1
(2:214)
(2:215)
Note that the term in the bracket is simply the autocorrelation of x(t) with the argument
u v, that is, Rx u v, thus (2.215) can be expressed as
1 1
Ry
hu hv Rx u vdu dv
1
1
hu
1
1
1
hv Rx u v dv du
(2:216)
huhu Rx u du
(2:217)
1
(2:218)
The power spectral density of the output signal y(t) can be obtained from (2.218) by
taking the Fourier transform of Ry . Using the convolution, conjugate symmetry, and real
signal properties in Table 2.2 we have
Sy f H f Hf Sx f
or, equivalently,
(2:219)
62
Sy f jHf j2 Sx f
(2:220)
This is exactly the same expression that we derived for periodic signals in (2.150).
2.10 Sampling
Sampling is the rst step in the process of converting a continuous-time analog signal into a
set of digital signals that represent bits or symbols. Sampling produces values or samples of
the continuous-time signal at points equally spaced in time. Sampling is useful only when
the continuous-time signal can be reconstructed from its samples. In general, many signals
can generate a given set of samples. Therefore, the samples can uniquely specify the signal if
and only if additional conditions are obeyed. These conditions are stated in a well-known
theorem called the sampling theorem.
Sampling theorem
Let x(t) be a bandlimited signal with a Fourier transform X(f ) such that X f 0 for j f j4B,
where B is the absolute signal bandwidth (highest frequency of the signal). Then, x(t) is
uniquely specied by its samples xkT ; k 0; 1; 2; . . ., equally spaced in time
T 1=fs . The parameter T is referred to as the sampling period, and the sampling frequency
fs must be greater than twice the absolute signal bandwidth B, that is,
fs 42B
(2:221)
(2:222)
H( f )
p(t) = (t kT)
k =
xs(t)
x(t)
H( f )
xr (t)
(a)
Figure 2.35
(b)
63
2.10 Sampling
The sampling theorem is summarized in Figure 2.35, where p(t) is the unit impulse-train,
xs t is the sampled signal, xr t is the reconstructed signal, and H(f ) is the transfer function
of the ideal lowpass lter, given by
(
Hf
T; j f j W
0; otherwise
(2:223)
Impulse-train sampling
The sampled signal xs t is generated by multiplying the continuous-time signal x(t) with a
unit impulse train pt:
xs t xtpt
(2:224)
where
pt
1
X
t kT
(2:225)
k1
Thus,
xs t xt
1
X
t kT
k1
1
X
xtt kT
(2:226)
k1
1
X
xkT t kT
(2:227)
k1
X( f )
2B
1 0 1
2B
2B
(a)
Figure 2.36
(b)
64
xs(t)
2B
Figure 2.37
t
1
4B
1
4B
(2:228)
1
1
fs 4B
1
X
k1
2B
sin k=2
k
t
k=2
4B
(2:229)
(2:230)
(2:231)
(2:232)
where X(f ) and P(f ) are the Fourier transforms of x(t) and p(t), respectively. Using the
Fourier transform of the impulse train in Table 2.2 we obtain the Fourier transform P(f ) of
p(t) as follows:
Pf
1
1 X
f kfs
T k1
(2:233)
65
2.10 Sampling
Substituting (2.233) into (2.232) and performing the convolution operation we get
Xs f
PuX f u du
1
1
1
T
1
T k1
1
X
1 k1
1 1
X
1
u kfs X f udu
(2:234)
u kfs X f u du
Applying (2.122) to (2.234) we arrive at the following expression for the Fourier transform
of the sampled signal:
Xs f
1
1 X
X f kfs
T k1
(2:235)
It can be seen that the Fourier transform of the sampled signal is the sum of shifted replicas
of X(f ) scaled by the factor 1/T. Furthermore, Xs f is a periodic function of frequency f with
period fs . When the sampling frequency fs 42B, the adjacent shifted replicas of X(f ) do not
overlap. By passing the sampled signal through an ideal lowpass lter, with a transfer
function given in (2.223) that satises the constraint in (2.222), we can recover the original
signal x(t). The recovered signal xr t can be described in the frequency domain by its
Fourier transform Xr f as follows:
Xr f Hf Xs f
8 1
X
>
>
<
X f kfs ;
k1
>
>
: 0;
j f j 4W
jfj W
(2:236)
X f
Thus, the original signal x(t) is completely recovered. The next example illustrates the
reconstruction process.
66
Xs( f )
H( f )
1/T
fs
Figure 2.38
f
BW
fs
(2:237)
1
1
X
1 k1
1
X
xkT kT ht d
(2:238)
xkT ht kT
k1
Given H(f) in Figure 2.35(b) the corresponding h(t) can be obtained from Table 2.2 as
ht 2WT
sin 2Wt
2Wt
(2:239)
Combining (2.238) and (2.239) we can write the reconstructed signal as follows:
xr t
sin 2W t kT
xkT 2WT
2W t kT
k1
1
X
(2:240)
The left-hand side of (2.240) is just the inverse Fourier transform of the left-hand side of
(2.236), which is exactly x(t). Equation (2.240) provides a convenient way of representing a bandlimited signal as a series of time-shifted (sin x)/x pulses. The coefcients
of the series are the samples of the signal. It turns out that the above series is also
an orthogonal series when 2WT 1 or 2W 1=T fs : this is proven below in
Example 2.38.
sin fs t kT
; where k is an integer
fs t kT
(2:241)
67
2.10 Sampling
where fs 1=T . We need to show that this is a set of orthogonal functions. We rst establish
the inner product of two distinct functions xk t and xm t as indicated by (2.36) and make
use of the Parseval relation in Table 2.2:
1
1
xk txm t dt
1
1
Xk f Xm f df
(2:242)
where Xk f and Xm f are the Fourier transforms of xk t and xm t, respectively. Note that
xk t x0 t kT and xm t x0 t mT . Hence, by applying the time shifting property
in Table 2.1 we get
Xk f X0 f ej2kTf
(2:243)
Xm f X0 f ej2mTf
(2:244)
where X0 f is the Fourier transform of x0 t. Substituting (2.243) and (2.244) into (2.242)
we obtain
1
1
xk txm t dt
(2:245)
jX0 f j2 ej2kmTf df
1
1
Note that X0 f is given in Figure 2.36(b) with fs replacing 2B, and its amplitude is 1=fs .
Equation (2.245) can now be rewritten as
1
1
xk txm t dt
fs =2
1 j2kmTf
e
df 0
2
f
fs =2 s
(2:246)
Thus, the set of functions in (2.241) is an orthogonal set. This example shows that a
continuous-time nite-energy signal x(t), bandlimited to a bandwidth B, can be represented
by an orthogonal series of the form
xt
1
X
k1
xkT
sin fs t kT
fs t kT
(2:247)
where fs 1=T 2B. The above expression is referred to as the NyquistShannon interpolation formula.
68
Xs( f )
Figure 2.39
f
B fs
Representations
A bandpass signal can be described mainly by three representations: envelope-phase, inphase and quadrature or I-Q, and complex envelope. In the envelope-phase representation, a
bandpass signal x(t) can be expressed as
xt At cos2fc t t
(2:248)
where fc is the carrier frequency, A(t) is the time-varying amplitude, and t is the timevarying phase. When a bandpass signal is employed to carry information, the amplitude A(t)
69
and/or the phase t may vary according to the information, resulting in various types of
modulation techniques.
In the I-Q representation, the bandpass signal x(t) can be expressed as follows:
xt xI t cos 2fc t xQ t sin 2fc t
(2:249)
The components xI t and xQ t are referred to as the in-phase and quadrature components
of x(t), respectively. The term quadrature results from the fact that sin 2fc t is in phasequadrature with respect to cos 2fc t, that is, =2 radians or 90 out-of-phase.
For the complex envelope representation, the bandpass signal x(t) can be written as
xt RefxL t ej2fc t g
(2:250)
where the complex envelope xL t of x(t) is often referred to as the equivalent lowpass
signal. The three representations of a bandpass signal are equivalent and their relations are
shown below:
q
(2:251)
At x2I t x2Q t jxL tj
xL t xI t jxQ t At ejt
t tan1
xQ t
xI t
(2:252)
(2:253)
xI t RefxL tg At cos t
(2:254)
xQ t ImfxL tg At sin t
(2:255)
1
XL f fc XL f fc
2
(2:257)
where XL f is the Fourier transform of the equivalent lowpass signal xL t of x(t). Thus, the
characteristics of X f are all contained in XL f .
70
(2:258)
where the function p(t) is dened in (2.26). The envelope A(t) and the phase t are
given by
At pt
t
(2:259)
(2:260)
2
2
pt cos 2fc t
pt sin 2fc t
2
2
(2:261)
(2:262)
(2:263)
(2:264)
(2:265)
71
where the complex envelope hL t is called the impulse response of the equivalent lowpass
system. Following the derivation of (2.257) we arrive at the Fourier transform Hf of
h(t) as
Hf
1
HL f fc HL f fc
2
(2:266)
where HL f is the Fourier transform of hL t. Now assume that the input signal x(t) has a
Fourier transform X f , the output signal y(t) of the bandpass system is
yt ht xt
(2:267)
(2:268)
Y f
(2:269)
We arrive at the above equation with the assumption that HL f fc and XL f fc do not
overlap, and neither do HL f fc and XL f fc . Thus, we have the relations
HL f fc XL f fc 0 and HL f fc XL f fc 0. Dene the following
parameter
1
YL f HL f XL f
2
(2:270)
1
YL f fc YL f fc
2
(2:271)
Then, we have
Y f
Taking the inverse Fourier transform of Y f yields the output response y(t):
1
yL t ej2fc t yL t ej2fc t
2
RefyL t ej2fc t g
yt
(2:272)
where the complex envelope yL t of y(t) is the inverse Fourier transform of YL f in (2.270)
and is given by
1
yL t hL t xL t
2
(2:273)
In summary, when the bandwidths of both bandpass signal and bandpass system are much
smaller than the carrier frequency such that the assumption leading to (2.269) holds, we can
obtain the response y(t) via (2.267) or via the equivalent lowpass operation given in (2.272)
and (2.273).
72
2.12 Summary
In this chapter we have provided a fairly detailed study of deterministic signals. We have placed the emphasis
on continuous-time signals that are often employed in practical communications systems. The concept of
continuous-time digital signals was introduced via examples such as PSK and ASK.
We then introduced the concept of power and energy signals. We showed that a power signal has innite
energy and an energy signal has zero average power. It was noted that the digital signals employed in practice
are energy signals and an innite sequence of them forms a power signal.
The study of energy signals has led to a special class of important digital signals, namely, orthogonal signals,
that are widely used in commercial and military systems. The most well-known pair of orthogonal signals are
the cosine and sine functions with identical frequency. We also studied the well-known orthogonal Walsh
functions, which are derived from the Hadamard matrix. The Walsh functions are employed in the cellular
standard IS-95.
Further study of nite-energy signals has led to the concept of signal space. The GramSchmidt procedure
enables us to express continuous-time nite-energy signals as vectors in the signal space, thus simplifying the
analysis.
A review of linear time-invariant systems was provided together with the important convolution operation. It
is worth mentioning that the majority of communication systems are LTI systems. Even nonlinear communication
systems contain many subsystems that are linear time-invariant.
Two great techniques that are indispensable in the analysis of communications systems were presented. They
are Fourier series for periodic signals and Fourier transform for aperiodic signals.
The study of Fourier series has revealed a number of important properties. The Parseval relation states that
the average power in a periodic signal equals the sum of the average power in all of its harmonics components.
The power in a harmonic component is equal to the squared magnitude of the corresponding Fourier series
coefcient. This relation leads to the study of power spectral density, which consists of an innite series of
weighted impulses at harmonic frequencies. As the signal propagates through an LTI system, its power spectral
density is shaped by the transfer function of the system. The output power spectral density function is
equal to the product of the squared magnitude of the systems transfer function and the input power spectral
density.
The study of Fourier transform of aperiodic nite energy signals has provided a number of interesting results.
The energy of a signal is equal to the total area under the energy spectral density function, which is simply the
squared magnitude of the Fourier transform of the signal. This is the Parseval relation for an energy signal. One
surprising result is that a continuous-time periodic signal has a Fourier transform that is an innite series of
impulses at harmonic frequencies. The coefcients of the impulses are simply the Fourier series coefcients. As
an energy signal propagates through an LTI system, its Fourier transform is shaped by the transfer function of
the system. The Fourier transform of the output signal is simply the product of the systems transfer function
and the Fourier transform of the input signal.
It was shown that the autocorrelation and the energy spectral density of an energy signal form a Fourier
transform pair. For a power signal, the autocorrelation and the power spectral density also form a Fourier
transform pair. The spectral density of the output signal of an LTI system is equal to the product of the squared
73
Problems
magnitude of the systems transfer function and the spectral density of the input signal. This applies to both
energy and power signals.
Again using the Fourier transform we have studied the sampling of a bandlimited signal. It was shown that
the original signal can be recovered from the sampled signal by means of an ideal lowpass lter provided that
the sampling rate exceeds the Nyquist rate The Nyquist rate is exactly twice the absolute bandwidth or highest
frequency of the original signal. If the sampling rate is smaller than the Nyquist rate, then aliasing results, and
the original signal cannot be recovered from the sampling signal.
In the last section we have investigated the representations of bandpass signals which are employed in
practical communication systems. Three representations were considered, namely, envelope-phase, in-phase and
quadrature (I-Q), and complex envelope representations. The relationship between a bandpass signal and its
equivalent lowpass signal was studied. The equivalent lowpass signal is simply the complex envelope of the
bandpass signal.
Problems
1.
2.
3.
4.
5.
6.
7.
8.
9.
10.
Examine whether the following signals are periodic: st sin 6t cos 7t,
st 5 cos14t 7 sin42t 6. Specify the smallest period.
Write the expression for the PSK waveform s(t) that represents four consecutive bits
1011, assuming that the leftmost bit starts at time t = 0 and the carrier frequency is an
integer multiple of 1=Tb , where Tb is the bit time.
Write the expression for the QPSK waveform s(t) that represents four consecutive symbols 10 01 11 00, assuming that the leftmost symbol starts at time t =
0 and the carrier frequency is an integer multiple of 1=Ts , where Ts is the
symbol time.
Write the expression for the 4ASK waveform s(t) that represents four consecutive
symbols 00 01 10 11, assuming that the leftmost symbol starts at time t = 0 and the
carrier frequency is an integer multiple of 1=Ts , where Ts is the symbol time.
Find the energy and power of the following signals: st10sin6t3cos7tV;
stsin5t 2cos7tfmV;andst20cos15tcos25tfV:
Find the energy of the unit impulse function. Is it a power signal?
Find the energy and power of the unit step function.
2Bt
Find the energy of the signal st 2B sin2Bt
.
Find the PSK signal energy assuming the bit time Tb 104 s and the amplitude
A = 100 mV. Find the QPSK signal energy assuming the symbol time Ts 2 104 s
and the amplitude A = 100 mV.
Consider the following two signals for 0 t 5Tb :
s1 t A cos2f1 t 1
s2 t A cos2f2 t 2
where for a positive integer 41 and fc =Tb we have
74
1
2Tb
1
f2 fc
2Tb
f1 fc
11.
12.
1
s1 t A cos 2 fc
t n
4Tb
1
s2 t A cos 2 fc
t n
4Tb
where n 2 f0; gmod 2 . Show that the two signals are quasi-orthogonal assuming that the frequency fc is an integer multiple of 1=Tb .
Consider the following two signals in the interval 0 t 5Tb :
s1 t A cos2ft
s2 t A cos2f f t
13.
14.
15.
16.
17.
18.
Let E be the smallest signal energy. Find the basis function and express each signal in
the set in terms of the basis function and E. Plot the signal space.
Determine if the following systems are LTI:
(a) yt dtd 3xt xt 7
7t
(b) yt 0 xd
(c) yt xt sin 300t
(d) yt x2 t
(e) yn 2n 3xn xn 3
Find the convolution of the waveform xt pt pt T pt T with pt,
where pt is a unit amplitude pulse of duration T.
75
Problems
x(t)
1
NT
1/N
Figure 2.40
19.
20.
21.
22.
23.
24.
25.
26.
27.
28.
29.
1 1
1
ak
mt ej2kt=T dt M k=T
T 1
T
where M k=T is the Fourier transform of m(t) evaluated at f = k/T.
P
j2kt=T
(b) Show that xt T1 1
.
k1 M k=T e
Consider a Hilbert transformer with impulse response ht 1=t. Let xt be the
input signal. Write the expression for the output signal yt ht xt, that is, the
Hilbert transform of xt. Find yt when xt sin nt.
^ t sin 2fc t , where mt
^ mt 1=t
Given the signal xt A2 mt cos 2fc t m
is the Hilbert transform of m(t), nd the Fourier transform of x(t).
Consider the inner product of two following signals x1 t pt cos 2fc t and
x2 t pt sin 2fc t given the Fourier transform Pf of p(t). We have
1
x1 tx2 t dt
hx1 ; x2 i
1
76
30.
31.
32.
33.
34.
35.
36.
37.
38.
39.
40.
77
Bibliography
41.
42.
Further reading
For a thorough analysis of Fourier series, Fourier transforms, discrete-time Fourier transforms, and z-transforms we recommend Oppenheim, Willsky, and Nawab [1]. Advanced
materials in signal analysis can be found in Bracewell [2].
Bibliography
1. A. V. Oppenheim and A. S. Willsky, with S. Hamid Nawab, Signals and Systems, 2nd
edition, Upper Saddle River, NJ: Prentice Hall, 1997.
2. R. N. Bracewell, The Fourier Transform and its Applications, 2nd edition, revised,
New York: McGraw-Hill, 1986.
3. L. W. Couch II, Digital and Analog Communication Systems, 7th edition, Harlow:
Prentice Hall, 2007.
4. S. Haykin, Communication Systems, 3rd edition, Chichester: John Wiley & Sons, 1994.
5. R. E. Ziemer and W. H. Tranter, Principles of Communications, 4th edition, Chichester:
John Wiley & Sons, 1995.
6. J. G. Proakis and M. Salehi, Fundamentals of Communication Systems, Hallow:
Pearson Prentice Hall, 2005.
7. A. Bruce Carlson, P. B. Crilly, and J. C. Rutledge, Communication Systems, 4th edition,
New York: McGraw-Hill, 2002.
Introduction
So far we have studied deterministic signals that are used to carry information. As far as
the receiver is concerned, the stream of transmitted digital waveforms that carries information symbols is a random signal because the information is random. Also, as the transmitted
signal travels through the channel, it is modied by noise, which is a random signal and is
often referred to as a random process. Therefore, the receiver receives the transmitted signal
plus noise. Such a channel is called an additive noise channel. Furthermore, wireless signals
such as cellular signals and wireless LAN and MAN signals always travel through a timevarying multipath fading channel, which causes the signal envelopes to vary randomly.
The time-varying phenomenon arises from Doppler shift, which is the result of the motion of
the transmitters and/or receivers. The multipath fading is the result of the destructive
interference of signal rays that travel via randomly delayed and attenuated paths.
Therefore, the received signal itself becomes a random signal. To analyze random signals
in communication receivers we need to know their statistics. In this section we explore some
tools necessary for such a task. We divide the discussion into reviews of probability theory
and random variables, and study of random processes and their applicability to communication theory.
79
n
X
PrBjAi PrAi
(3:1)
i1
where PrBjAi is the conditional probability of the event B given that the event Ai
occurred.
Bayes theorem
The a-posteriori probability PrAi jB is expressed in terms of the a-priori probability PrAi ,
the conditional probability PrBjAi , and the total probability Pr(B) as follows:
PrBjAi PrAi
PrB; Ai
PrAi jB P
P
n
n
PrBjAi PrAi
PrBjAi PrAi
i1
(3:2)
i1
Independence
The concept of independence plays an important role in signal analysis. Two events A and B
are independent if
PrAB PrA PrB
(3:3)
Union bound
Let
n
S
i1
Ai be the union of n events Ai, then, we have the following union bound:
!
n
n
X
[
Pr
Ai
PrAi
i1
(3:4)
i1
In the following discussion we study random variables that are essential to signal analysis
where they represent decision voltage samples at the receiver
80
signal-to-noise ratio and then sampled to obtain a decision sample for the detector to decide
which symbol had been transmitted. The decision sample is referred to mathematically as a
random variable.
A random variable is a function dened on the sample space. We use a capital letter such
as X for a random variable and the lower-case letter x for the value it assumes. Depending on
the experiment, the random variable X can be discrete or continuous. Let X be a discrete
random variable and let xi, i = 1, 2, . . . , n, be the values that X assumes. For each event
fX xi g and its probability PrfX xi g the function
pX xi PrfX xi g
i 1; 2; . . . ; n
(3:5)
X is then dened as
15 x51
FX x PrX x
(3:6)
dFX x
dx
(3:7)
x
1
f X y dy
(3:8)
(3:9)
x1
(3:10)
2 FXY x; y
x y
(3:11)
Thus, the density function of either X or Y can be found by integrating its joint density
function with respect to the other variable. We have
1
fXY x; y dy
(3:12)
fX x
1
81
fY y
fXY x; y dx
fY jX yjxfX x
fXY x; y
fY jX yjxfX xdx
fY jX yjxfX xdx
fX jY xjy
(3:13)
where fX jY xj y is the conditional density function of X given Y and the joint density is
given by
fXY x; y fY jX yjxfX x fX jY xjyfY y
(3:14)
If X and Y are independent random variables, then the following relationship holds:
fX x; y fX xfY y
(3:15)
fY y fY jX yjx fX x dx
(3:16)
X 2 EX 2
x2i pX xi
X 2 EX 2
1
x2 fX x dx
2X VarX EfX X 2 g EX 2 X 2
(3:18)
(3:19)
The quantity X is called the standard deviation of the random variable X. The mean and
mean-square values of a random variable are also referred to as its rst and second moments.
The nth moment of the random variable X is dened as
1
X
n
n
n
n
n
X EX
xi pX xi
X EX
xn fX x dx
(3:20)
1
The above results can be generalized to a function g(X). For example, the expected value of
g(X) is given by
1
X
EfgX g
gxi pX xi
EfgX g
gxfX x dx
(3:21)
i
1
82
In the following discussion we present the application of probability theory and random
variables to the analysis of digital communication signals from the standpoint of random
processes.
(3:22)
The autocovariance is simply the autocorrelation of a random process minus its mean value,
that is,
CX t1 ; t2 EfX t1 EfX t1 gX t2 EfX t2 gg
(3:23)
83
Therefore, the autocorrelation and autocovariance are related by the following expression:
CX t1 ; t2 RX t1 ; t2 EfX t1 gEfX t2 g
(3:24)
i1
ent and equally likely bits f0; 1g, where p(t) is the squared pulse shape of unit amplitude and
duration Tl, which is a bit time, and di 2 f1; 1 represents the normalized signal amplitudes. The mapping 0 ! 1 and 1 ! 1 maps bit 0 to amplitude 1 and bit 1 to amplitude
1 (the mapping 0 ! 1 and 1 ! 1 maps bit 0 to amplitude 1 and bit 1 to amplitude 1
would give the same result). Since the amplitudes are equally likely, the mean value of this
random process is zero. Consider the times t1 t and t2 t , where is an arbitrary
time interval. We have
RX t; t EfX tX t g EfX tgEfX t g 0; jj 4Tb
(3:25)
since the process has zero mean and the bits are independent. For jj 5Tb, and with an
arbitrary time index i = 0, let us assume that the time origin t of the zeroth bit is uniformly
distributed in the bit interval Tb (without any knowledge about the location of the time origin
the uniform distribution is the worst-case assumption), we have
RX t; t EfX tX t g ; 0 5 t Tb
8 T
< 0 b 12 T1b dt TbT
1 Tb ;
0 Tb
b
: Tb 12 1 dt Tb 1 ;
Tb 0
Tb
Tb
Tb
1
(3:26)
jj
; jj Tb RX
Tb
RX()
1
4Tb
Tb
Tb
2Tb
3Tb
(a)
Figure 3.1
Tb
(b)
Tb
84
Thus, the random sequence of antipodal signals with a uniform time origin has a triangular
autocorrelation, which depends only on the time difference jj jt2 t1 j , that is,
RX t; t RX . Note that RX 0 1 as expected. Figure 3.1 shows the representative antipodal signal sequence and its autocorrelation.
In the above analysis we assume that the time origin of the zeroth bit is uniformly
distributed in the bit time Tb. This is equivalent to shifting the process X(t) by a time
delay uniformly distributed in a bit time and independent of X(t).
1
P
i1
and equally likely bits {0,1}, where p(t) is the squared pulse shape of unit amplitude and
duration Tb, which is a bit time, and di 2 f1; 0g represents the normalized signal amplitudes.
The mapping 0 ! 1 and 1 ! 0 maps bit 0 to amplitude 1 and bit 1 to amplitude 0 (the
mapping 0 ! 0 and 1 ! 1 maps bit 0 to amplitude 0 and bit 1 to amplitude 1 would give
the same result). We assume that the time origin is uniform in a bit time Tb. The mean value
of this random process is
EfX tg
1
X
i1
0:5
1 iTb Tb
X
i1 iTb
(3:27)
1
1 dt 0:5
Tb
The process Y t X t 0:5 therefore represents an antipodal signal sequence of amplitudes 0.5 and 0:5. Since X t Y t 0:5 the autocorrelation of X(t) can be obtained
from that of Y(t), which has been derived in the previous example. Consider the times
t1 t and t2 t , where is an arbitrary time interval. We have
RY t; t EfY tY t g ; 0 5 t Tb
1
jj
1
;
jj Tb
4
Tb
(3:28)
And consequently the autocorrelation of the OOK signal sequence with a uniform time
origin is given as
RX t; t RY t; t 0:52
8
< 14 1 Tjj 14 ;
b
: 1
4;
RX
jj Tb
(3:29)
jj4Tb
85
RX()
0.5
0.25
Tb
Tb
2Tb
3Tb
t
4Tb
(a)
Figure 3.2
Tb
Tb
(b)
Denition 3.3 A random process is wide-sense stationary (WSS) if its mean value is
constant and its autocorrelation is a function of the time difference, that is,
RX t1 ; t2 RX t2 t1 , or, for notational convenience, RX t; t RX :
A stationary process is obviously wide-sense stationary but not necessarily vice versa.
The antipodal signal and OOK signal sequences with uniform time origin are both WSS
processes. For a WSS process we have
RX 0 EfX 2 tg; RX RX ; jRX j RX 0
(3:30)
Denition 3.5 A random process is wide-sense cyclostationary (WSCS) if its mean value
and its autocorrelation are periodic in time t with some period T, that is, they are invariant to
86
1
P
i1
equally likely bits [0,1] where di 2 f1; 1g represents the normalized signal amplitudes.
The mapping 0 ! 1 and 1 ! 1 maps bit 0 to amplitude 1 and bit 1 to amplitude 1 (the
mapping 0 ! 1 and 1 ! 1 maps bit 0 to amplitude 1 and bit 1 to amplitude 1 would
give the same result). Also, p(t) is the half-squared pulse shape of unit amplitude
and duration Tb, which is a bit time, and is dened as pt 1; 0 t 5Tb =2 and
pt 0; Tb =2 t 5Tb . A sample function of this signal sequence is shown in Figure 3.3.
We wish to evaluate the mean value and autocorrelation at different xed time origins t.
We consider two cases with a common time delay Tb =4 and two time origins t = 0 and
t = 3Tb/4 as follows:
EfX 0g 0
(3:31)
(3:32)
Since the autocorrelation depends on the time origin the process is not WSS. Let us examine
the mean value and autocorrelation of this random process in general. We have
(
)
1
X
X
di pt iTb
Edi pt iTb
EfX tg E
i1
i
(3:33)
X
Edi
pt iTb
i
p(t)
X(t)
t
Tb
2
(a)
Figure 3.3
Tb
Tb
2
Tb
3Tb
2
(b)
87
(
RX t; t EfX tX t g E
XX
i
1
X
di pt iTb
i1
1
X
)
dj pt jTb
j1
(3:34)
Using the fact that Edi 0, and Edi dj 1; i j and Edi dj 0; i 6 j , we obtain
P
EfX tg 0, and RX t; t pt iTb pt iTb is time-varying but periodic
with period Tb. Thus, for a xed timei origin, this random process is a WSCS process.
Digital signal sequences that are often found in practice are wide-sense cyclostationary
processes in nature. This property implies that the sequences mean value and autocorrelation are periodic in T (bit or symbol time). Therefore, these statistical averages possess
Fourier series with a harmonic at frequency 1/T, the data clock rate. This provides a means
for designing timing recovery circuits to extract the data clock signal, for the purpose of
sampling the received signal to obtain a decision sample for bit or symbol detection to
recover the transmitted information. The following theorem shows how a WSS process can
be derived from a WSCS process.
Theorem 3.6 (Conversion of a WSCS process into a WSS process) If X(t) is a WSCS process
with periodic mean value mX t and autocorrelation RX t; t of period T , then the shifted
process Y t X t , where is a uniform random variable in (0, T) and independent of
X (t), is a WSS process with constant mean and t-invariant autocorrelation given by
1 T
mY
mX t dt
T 0
(3:35)
1 T
RX t; t dt
RY
T 0
Proof With the time delay independent of the process X(t) and mX t periodic with
period T, the mean value of Y(t) is given by
mY EfX t g EfEX t jg EfmX t g
1 T
1 T
mX t d
mX t dt
T 0
T 0
(3:36)
1 T
1 T
RX t ; t d
RX t; t dt
T 0
T 0
(3:37)
88
The above theorem enables us to calculate the time-averages of the mean value and
autocorrelation of a WSCS process. This will prove to be very useful in dealing with the
power spectral density of the process, which will be covered later. In fact, since the
autocorrelation function of a WSCS process is a function of both t and , its power spectral
density is not dened. By doing time-averaging we can convert a WSCS process into a WSS
process so that its power spectral density can be calculated.
So far we have presented examples of WSCS and WSS processes that are of lowpass or
baseband types, such as the antipodal and OOK signal sequences. The waveform that
represents a bit is a lowpass waveform (such as the squared pulse shape) whose amplitude
spectrum (the magnitude of its Fourier transform) centers on the zero frequency. In the
following examples we present two cases of digital signals that are of bandpass or modulated types, used to carry information over high-frequency channels.
assume the carrier frequency is integer multiples of the bit rate 1/Tb. The mean and
autocorrelation of this bandpass process are given by
(
)
1
X
X
EfX tg E
di pt iTb cos 2fc t
Edi pt iTb cos 2fc t
i1
Edi
(3:38)
and
RX t; t EfX tX t g
(
)
1
1
X
X
E
di pt iTb cos 2fc t
dj pt jTb cos 2fc t
i1
j1
(3:39)
1XX
Edi dj pt iTb pt jTb cos 2fc cos 4fc t 2fc
2 i j
Using the fact that Edi 0, and Edi dj 1; i j and Edi dj 0; i 6 j, the mean
value of the process is zero, that is, EfX tg 0, and the autocorrelation RX t; t
P
1
pt iTb pt iTb cos 2fc cos 4fc t 2fc is time-varying but periodic
2
i
with period Tb. Thus for a xed time origin, this random process is a WSCS process.
To convert this WSCS process into a WSS we need to time-average the autocorrelation
function using the same procedure as used in the case of the antipodal signal sequence.
By introducing a uniform time shift into the data sequence but not the carrier we obtain a
WSS PSK signal sequence. Therefore, the time-averaged autocorrelation function is given
as follows:
89
Tb jj
1
jj
1
cos 2fc
cos 2fc ; jj Tb
Tb
2
Tb
0; jj 4 Tb
1
2
RX
(3:40)
1
X
(3:41)
i1
where Ts is the symbol time, which is twice the bit time, and the carrier frequency is taken to
be integer multiples of the symbol rate 1/Ts. We observe that the QPSK signal sequence is
the sum of two orthogonal PSK signal sequences at the symbol rate, and that both are WSCS
processes. Thus, the QPSK signal sequence is itself a WSCS process. If we convert this
WCCS process into a WSS process by assuming a uniform time origin over (0, Ts), we
obtain a mean value of zero and a time-averaged autocorrelation identical to the sum of the
autocorrelations of two orthogonal
PSK signal sequences with Tb replaced by Ts and
p
amplitude 1 replaced by 2=2, as follows:
1
jj
1
cos 2fc ; jj Ts
RX
2
Ts
(3:42)
0;
jj 4 Ts
The QPSK signal sequence can be represented in complex notation as
X t XI t cos 2fc t XQ t sin 2fc t
RefXL t e j2fc t g
(3:43)
Input bits
11
01
00
10
I
p
p2=2
p2=2
p2=2
2=2
Q
p
p2=2
p2=2
p2=2
2=2
90
where XL(t) is the complex envelope (equivalent lowpass process, or baseband process) of
the bandpass process X(t) and is given by
XL t XI t jXQ t
1
1
X
X
Ii pt iTs ; XQ t
Qi pt iTs
XI t
i1
(3:44)
i1
(3:45)
Because the amplitudes Ii and Qi are iid, the in-phase process XI t and the quadrature
process XQ t, both with zero mean, are independent and therefore their cross-correlations
RXI XQ EfXI tXQ t g and RXQ XI EfXQ tXI t g are both zero. Note that
the time-averaged autocorrelations of XI t and XQ t are identical and are given by
RXI RXQ
1
jj
1
;
2
Ts
0;
jj Ts
(3:46)
jj 4 Ts
The time-averaged autocorrelation of the complex envelope XL t is simply the sum of two
time-averaged autocorrelations RXI and RXQ , thus, RXL RXI RXQ . On the
other hand, the time-averaged autocorrelation of the bandpass process X(t) is given by
RX 12RXI RXQ cos 2fc RXI cos 2fc 12RXL cos 2fc :
(3:47)
If, in addition, the in-phase and quadrature processes are uncorrelated, that is, RXI XQ
EfXI tgEfXQ t g 0 for all (this also implies that they are orthogonal), then the
time-averaged autocorrelation of X(t) is reduced to the following expression:
91
1
RX RXI cos 2fc RXL cos 2fc
2
(3:48)
where XL t XI t jXQ t is the equivalent lowpass process of X(t) and its autocorrelation is RXL EfXL tgEfXL t g.
So far we have dealt with ensemble averages (statistical averages) of a random
process. This can be achieved if a large number of sample functions (realizations) of the
random process are available. In practice, we perform measurements on a single sample of a
random process to obtain time averages instead. This results in the estimation of the
ensemble averages if the length of the available sample is very long. In that case, we expect
the sample function to be typical of all realizations of the random process. This property is
referred to as ergodicity. The following theorems establish the criteria for a random process
to be ergodic.
Theorem 3.8 (Slutsky theorem) A WSS process X(t) is said to be mean-ergodic, that is,
1
2T
T
T
X t dt ! EfX tg
(3:49)
T!1
if and only if
1
T
T
0
CX d ! 0
(3:50)
T!1
T
T
(3:51)
if and only if
1
T
T
0
CY d ! 0
T !1
(3:52)
(3:53)
The result also applies to autocorrelation-ergodic processes if we replace the autocovariance with autocorrelation. Also the process is variance-ergodic if we set 0 in the
autocovariance CY .
92
1 T
1 T
jj
CX d
1
cos 2fc d; jj Ts
T 0
2T 0
Ts
(3:54)
0; elsewhere
The integral approaches zero as T ! 1, and when fc is integer multiples of Ts the integral is
zero independent of T. Therefore the sequence is mean-ergodic.
CX
CX 0
(3:55)
We note that as the delay time increases the process decorrelates, that is, it loses its
correlation. Thus, it makes sense to dene a delay spread to reect this characteristic of the
process.
Denition 3.11 The delay spread d of a WSS process is the delay time beyond which the
normalized autocovariance remains below 0.1, that is,
jX j 0:1 for all 4 d
(3:56)
The normalized autocovariance of the QPSK signal sequence with uniform origin is a
scaled version of its time-averaged autocorrelation. The delay spread is d 0:9Ts . By
observation we see that the autocorrelation is zero for jj 4 Ts, that is, the sequence looses
its correlation after one symbol time. Hence the delay spread gives a fairly accurate
prediction of the decorrelation. We note that the inequality (3.56) is purely empirical and
can be adjusted to t any decorrelation requirement.
Denition 3.12 The power spectral density SX f and the autocorrelation RX of a WSS
process X(t) form a Fourier transform pair known as the EinsteinWienerKhinchine relation:
1
SX f
RX ej2f d FRX
(3:57)
1
93
RX
1
1
SX f ej2f df F 1 SX f
(3:58)
SX f lim
1
1
RX ej2f d
(3:59)
jRX j d 5 1
(3:60)
where XT t is a 2T-truncation of a single sample of the random process X(t) and FXT t is
its Fourier transform.
Proof We consider the truncated version of a sample function for the reason that it is not an
energy signal and may not have a Fourier transform. We have
FXT t
1
1
XT t e
j2ft
dt
T
T
X t ej2ft dt
(3:61)
Therefore,
T
T
o
1 n
1
2
j2ft1
j2ft2
E jFXT tj
E
X t1 e
dt1
X t2 e
dt2
2T
2T
T
T
T T
1
(3:62)
T
1 0
1 2T T
j2f
RX e
dt2 d
RX ej2f dt2 d
2T 2T T
2T 0 T
1 0
1 2T
2T RX ej2f d
2T RX ej2f d
2T 2T
2T 0
2T
1 2T
jj
RX ej2f d
2T jjRX ej2f d
1
2T 2T
2T
2T
(3:63)
94
The periodogram is a nonnegative function for all frequencies, consequently, the power
spectral density is also a nonnegative function for all frequencies, that is,
SX f 0 for all f
(3:64)
Furthermore, since the autocorrelation is real and even, the power spectral density is an even
function, hence
SX f SX f
(3:65)
1
SX f df
(3:66)
Example 3.7 Power spectral densities of digital signal sequences with squared pulse shape
The power spectral density of a digital signal sequence of uniform time origin is the Fourier
transform of its autocorrelations. We have added the amplitude A to the original sequence,
which also has the amplitude 1 to make the results more general.
sin fTb 2
Antipodal: SX f A Tb
fTb
(3:67)
1
sin fTb 2 1 2
OOK: SX f A2 Tb
A f
4
4
fTb
(3:68)
A 2 Tb
PSK: SX f
4
"
#
sin f fc Tb 2
sin f fc Tb 2
f fc Tb
f fc Tb
A 2 Ts
QPSK: SX f
4
"
#
sin f fc Ts 2
sin f fc Ts 2
f fc Ts
f fc Ts
(3:69)
(3:70)
(3:71)
95
SX( f )
SX( f )
(f)
2
Tb
1
Tb
1
Tb
2
Tb
2
Tb
(a)
1
Tb
0
(b)
1
Tb
2
Tb
SX ( f )
fc
(c)
f
fc 1 c
Tb
fc + 1
Tb
SX( f )
fc
(d)
Figure 3.4
fc 1
Ts
fc
fc + 1
Ts
Power spectral densities of various random processes: (a) antipodal; (b) OOK; (c) PSK; (d) QPSK.
Figure 3.4 shows the power spectral densities of the antipodal, OOK, PSK, and QPSK
signals.
Linear lters are used in communication systems for signal processing purposes,
such as limiting noise within a prescribed bandwidth, shaping a signal spectrum to meet
the FCC requirement, rejecting the out-of-band interference, and so forth. Therefore,
it is necessary to know the response of linear lters to random processes. We will
consider a linear time-variant lter with impulse response h(t) and corresponding Fourier
transform H(f ).
Theorem 3.14 Let the input X(t) of a linear time-invariant lter with impulse response h(t)
and corresponding Fourier transform H(f ) be a WSS process with power spectral density
SX f and autocorrelation RX . The output Y(t) of the lter is also a WSS process with
power spectral density SY f and autocorrelation RY given by
SY f jHf j2 SX f
(3:72)
RY h h RX
(3:73)
96
Proof Let XT t be a 2T-truncation of a single sample of the random process X(t) and
FXT t be its Fourier transform. Also let YT t be the response of the lter to the input
XT t. We have
o
o
1 n
1 n
E jFYT tj2
E jHf FXT tj2
2T
2T
o
1 n
E jFXT tj2
jHf j2
2T
(3:74)
(3:75)
SY f Hf H f SX f
(3:76)
SY f lim
We also have
Taking the inverse Fourier transform of SY f and realizing that the inverse Fourier transform of H f is h we obtain
RY h h RX
(3:77)
Example 3.8 Power spectral density of a digital signal sequence with arbitrary pulse shape
In Example 3.7 we calculated the power spectral densities of digital signal sequences
employing a squared pulse shape. This particular pulse shape yields a sin x=x2 power
spectral density that requires an innite bandwidth, although about 90% of the sequence
power is contained within the null-to-null bandwidth. In practice, other pulse shapes that
yield nite bandwidth may be used such as the raised-cosine pulse shape. We wish to derive
the general expression for power spectral density for an arbitrary pulse shape in this
1
P
example. We consider a WSCS digital signal sequence X t
di pt iT and coni1
vert it into a WSS process by assuming a uniform time origin over (0, T), where T is a bit time
or a symbol time depending on the type of digital signal. The data di can be real or complex. If
it is complex, the process represents the envelope of another bandpass process. The pulse
shape p(t) is assumed to have a Fourier transform P(f ). The autocorrelation of the sequence is
(
)
1
1
X
X
di p t iT
dj pt jT
RX t; t EfX tX t g E
XX
i
i1
Edi dj Ep t
iT pt jT
j1
(3:78)
97
The term Edi dj is the autocorrelation of the data and is denoted as Rd i j. Hence
T
1 XX
Rd i j p t iT pt jT dt
RX t; t
T i j
0
X iT T
1X
Rd m
p p mT d
T m
iT
i
1
(3:79)
1X
Rd m
p p mT d
T m
1
1X
Rd mp mT p
T m
RX
Taking the Fourier transform of the autocorrelation we obtain
1X
Rd mPf ej2mfT P f
T m
X
1
Rd m ej2mfT
jPf j2
T
m
SX f
(3:80)
1
jPf j2 Sd ej2fT
T
Denition 3.15 The coherence bandwidth Bc of a WSS lowpass process is dened as onehalf the inverse of its delay spread d :
Lowpass:
Bc
1
2 d
(3:82)
Furthermore, the coherence bandwidth of a WSS bandpass process is the inverse of the
delay spread of its in-phase or quadrature process:
98
Bandpass:
Bc
1
d
(3:83)
Denition 3.16
The WSS process n(t) is called white noise if it has the following power
spectral density and autocorrelation:
Sn f
N0
2
Rn
N0
(3:84)
(3:85)
By direct substitution, the lowpass in-phase and quadrature processes can be expressed in
terms of the bandpass process and its Hilbert transform X^ t 1=t X t as follows [1,2]:
XI t X t cos 2fc t X^ t sin 2fc t
XQ t X^ t cos 2fc t X t sin 2fc t
(3:86)
Furthermore, it can be shown that the autocorrelations of both XI t and XQ t are given by
^ X sin 2fc
RXI RXQ RX cos 2fc R
(3:87)
(3:88)
99
The Fourier transforms of the crosscorrelations RXI XQ and RXQ XI are referred to as their
cross-spectral densities. From (3.88) we have
j
j
SXI XQ f SXQ XI f 1 sgnf fc SX f fc 1 sgnf fc SX f fc
2
2
(3:90)
(3:91)
In addition the cross-spectral densities of the in-phase and quadrature processes become
SXI XQ f SXQ XI f jSX f fc SX f fc ; jf j B
0; jf j 4 B
(3:92)
When the power spectral density of the WSS bandpass process is symmetric around the
carrier frequency fc then
SX f fc SX f fc ; jf j B
(3:93)
(3:94)
Consequently,
And the in-phase and quadrature processes are uncorrelated. Figure 3.5 illustrates the power
spectral densities of strictly bandlimited white noise n(t).
SnI( f ) = SnQ( f )
Sn(f)
N0
N0
2
fc B f
c
fc + B
fc B f
c
(a)
Figure 3.5
fc + B
(b)
(a) Power spectral density of bandlimited white noise process. (b) Power spectral density of in-phase and quadrature
noise processes.
100
Denition 3.17 The Gaussian process X(t) is characterized by a set of n jointly Gaussian
random variables X1 X t1 ; . . . ; Xn X tn for all n and tn . The joint density function of
the random variables is given by
t
fX x
exX K
1
xX=2
2n=2 jKj1=2
(3:95)
Linear transformation
In a communication system Gaussian noise is processed by linear lters of various kinds
such as the matched lter and integrator for signal processing. Linear ltering does not
change the Gaussian nature of noise as we will subsequently show. The following theorem
shows that a linear transformation of a sequence of jointly Gaussian random variables
preserves the Gaussian property.
Theorem 3.18 The linear transformation of a jointly Gaussian vector is another jointly
Gaussian vector.
101
Proof Consider a jointly Gaussian vector X X1 X2 . . . Xn t with mean X and covariance matrix K. We now consider the following linear transformation:
Y AX
(3:97)
(3:98)
eA
1
eyY A
1 t
K 1 A1 yY=2
(3:99)
X d
(3:100)
n
X
X j t j t
(3:101)
j1
102
n
X
ht j x j j
(3:103)
j1
The above examples show that the Gaussian process is closed under a linear operation.
That is, any linear operation on a Gaussian process produces another Gaussian process.
This forms the basic foundation for detection of random signals in Gaussian noise that are
processed by a linear system. Only the impulse response of the system, the mean, and
autocorrelation of the input process are required to completely characterize the density
function of the output process and hence the calculation of the probability of detection or
error of any output sample.
Theorem 3.19 The response of a linear system to an input Gaussian process is also a
Gaussian process.
Sampling
The sampling theorem plays an important role in the processing of deterministic signals. In
the following discussion we show that the same sampling theorem can be applied to a strictly
bandlimited white Gaussian process to produce independent Gaussian samples. This provides an alternative means of analyzing a continuous-time signal as a discrete-time sequence
and the derivation of the information capacity of a strictly bandlimited white Gaussian
channel.
Theorem 3.20 (Sampling Theorem) The samples of a strictly bandlimited zero mean
white Gaussian process taken at the Nyquist rate are independent Gaussian random
variables.
103
Proof Let us consider the white noise process n(t) with the following power spectral
density
N0
; jf j B
2
0;
otherwise
Sn f
(3:104)
The corresponding autocorrelation is the inverse Fourier transform of the power spectral
density and is given as
sin 2B
Rn N0 B
(3:105)
2B
Consider two Nyquist samples taken at time t1 iT and t2 jT for integers i and j, where
T 1=2B. Since the mean of the process is zero, the autocovariance is the same as the
autocorrelation, hence
sin 2Bj iT
Cn t1 ; t2 Rn t1 ; t2 Rn j iT N0 B
(3:106)
2Bj iT
which is zero when i 6 j for any i and j. This proves that the Gaussian samples are
uncorrelated and hence independent.
L
X
sik xk t
(3:107)
si txk t dt
(3:108)
k1
1
1
The coefcients sik contain all the information about the transmitted signal si t. Now
consider an AWGN channel with noise n(t) having power spectral density N0 =2. There is
no constraint on the bandwidth of noise so that its autocorrelation is Rn N0 =2 .
104
The signal plus noise at the receiver input is modeled as the sum of a transmitted signal si t
and noise n(t) as
X t si t nt
(3:109)
How do we extract the information in the transmitted signal? By using (3.108) we simply
multiply the received signal X(t) with the known orthonormal basis functions
x1 t; . . . ; xL t to obtain the set of L samples
Xk sik Nk
k 1; 2; . . . ; L
(3:110)
where the noise samples Nk are the projection of noise n(t) onto the orthonormal basis
functions and are given by
1
Nk
ntxk t dt k 1; 2; . . . ; L
(3:111)
1
Note that these noise samples are zero mean Gaussian random variables with covariance
given by
CovNj Nk ENj Nk
1 1
1 1
E
ntnxj txk dt d
Entnxj txk dt d
1
1 1
1
1
1
1
Rn t xj txk dt d
1 1
1
1
N0
t xj txk dt d
1 2
N0
xj txk t dt
2 1
(
N0 =2; j k
0;
(3:112)
j 6 k
Thus the noise samples are uncorrelated for j 6 k and since they are Gaussian they are also
independent.
The received signal can be represented by
X t
L
X
Xk xk t et
(3:113)
k1
(3:114)
This process depends only on the noise, not on the signal, and it is also a zero mean Gaussian
process. Furthermore, the following derivation shows that the Gaussian process e(t) is
independent of the received samples Xk sik Nk ; k 1; 2; . . . ; L and hence irrelevant
in the detection of X(t):
105
(
EetXj E nt
)
Nk xk tsij Nj
X
1
nxj d
ENk Nj xk t
E nt
1
(3:115)
N0
N0
xj t xj t 0
2
2
KarhunenLoeve expansion
Again we consider a zero mean Gaussian noise process n(t) with autocorrelation Rn that
accompanies a transmitted signal si t; 0 t 5T , for arbitrary T. The following theorem
veries the existence of a set of orthonormal eigenfunctions f k t of the noise autocorrelation. This set represents the noise process n(t) as a series expansion with coefcients nk
forming a set of independent Gaussian random variables with zero mean and variance equal
to the eigenvalues of the noise autocorrelation function. In other words we have
1
X
nt
nk f k t
(3:116)
k1
f i tf j t dt
1 ij
0i
6 j
(3:117)
(3:118)
where
nk
f k tnt dt
(3:119)
106
Rn t f k d k f k t
(3:120)
where Rn is the autocorrelation of the zero mean random process n(t). Then the
1
P
nk f k t are uncorrelated and have
expansion coefcients nk of the series nt
k1
variances k En2k . Furthermore, if n(t) is Gaussian, then the coefcients nk are independent Gaussian random variables.
Proof We establish the following relation to show that the series coefcients are uncorrelated:
Eni nj E
f i t
ntf i t dt
nf j d
Entnf j ddt
f i t
Rn t f j ddt
0
T
j f i t f j t dt
0
j i j
0 i 6 j
(3:121)
Since the random variables nk are both Gaussian and uncorrelated, they are also independent.
The eigenfunctions f k t form a complete set of orthonormal basis functions for the niteenergy signals si t; 0 t 5T , when the noise autocorrelation is a positive-denite function. The signals can then be represented by the series
X
si t
sik f k t i 1; 2; . . . ; M
(3:122)
k
where
sik
1
1
si tf k t dt
(3:123)
The receiver for this type of channel will not be practical since the samples of noise and
signal form a countable set and are hence innitely dimensional. In the next section we
discuss a method to transform non-white noise into white noise so that the result of the
previous section applies.
Whitening filter
The power spectral density of noise at the receiver is shaped by the receiver transfer function
Hf whose corresponding impulse response is h(t). For example, if the input noise is white
107
and has power spectral density N0 =2, then as the noise propagates through the receiver the
noise at the detector ends up with a new power spectral density Sn f jHf j2 N0 =2 and
becomes non-white. In another scenario, the noise at the input of the receiver could already
be non-white with arbitrary power spectral density Sn f , and the receiver is wide-band such
that its transfer function can be considered constant over a bandwidth of the signal. In either
case we canpwhiten
the input noise by using a lter whose transfer function satises
jGf j 1= Sn f . Such a lter reproduces white noise at its output. In the rst scenario
the signals yi t; i 1; 2; . . . ; M , at the output of the whitening lter are
yi t si t ht gt, where si t; i 1; 2; . . . ; M, are the original transmitted signals,
and g(t) is the inverse Fourier transform of G(f ). In the second scenario the output of the
whitening lter is yi t si t gt.
Af 2
Bf 2
(3:124)
As2
Bs2
(3:125)
The roots of Sn(s) which are either zeros or poles must be either real or complex conjugate.
Furthermore, if zk is a root, then zk is also a root. Therefore, we can factor Sn(s) into
products of polynomials with roots in the left-half s-plane Re s50 and roots in the righthalf s-plane Re s40:
Sn s
CsCs
DsDs
(3:126)
The minimum-phase factor G(s) is the ratio of two polynomials formed with roots in the lefthalf s-plane:
Gs
Cs
Ds
(3:127)
a
b2 2f 2
(3:128)
We have
Sn s
b2
a
a
2
b sb s
s
(3:129)
108
(3:130)
We now conclude the study of Gaussian processes with the following important theorem.
(3:131)
(3:132)
109
The envelope R(t) and phase Ct of the noise process are related to the in-phase noise nI t
and quadrature noise nQ t as follows:
nQ t
Rt n2I t n2Q t1=2
Ct tan1
(3:133)
nI t
nI t Rt cos Ct
nQ t Rt sin Ct
(3:134)
Both in-phase and quadrature noise components of n(t) are uncorrelated. Since they
are Gaussian processes, they are also independent. Let us consider a sample of a realization of the envelope R(t) at a xed time t. We represent this sample via the random
variable R. Similarly, we obtain a sample of the phase process at time t, which can be
represented by the random variable C. At the xed time t, we also obtain the in-phase and
quadrature noise samples nI and nQ , which are independent zero mean Gaussian random
variables of the same variance as the variance of n(t), that is, VarnI VarnQ
Varn En2 2 .
Applying the theory of transformation of random variables we obtain the probability
density functions of both random variables R and C:
fR r
r r2 =22
e
2
fC
1
2
r0
(3:135)
0 2
(3:136)
Note that the probability density function of R is independent of C, hence, these two random
variables are independent when they are taken at the same time. This does not mean that the
two random processes R(t) and Ct are independent. The density function of the noise
envelope is the well-known Rayleigh density function. The density function of the phase is
the uniform density function. Note that ER2 2 2 .
(3:137)
where the complex envelope is nL t nI t jnQ t. The envelope is simply the absolute
value of nL t, which is Rt n2I t n2Q t1=2 :
Now let us return to the second case where we attempt to detect the envelope of the signal
plus noise. The received signal for bit 0 plus noise can be written as
X t A cos2fc t nt
A cos nI t cos 2fc t A sin nQ t sin 2fc t
(3:138)
110
(3:139)
(3:140)
1
The Gaussian processes n0I t A cos nI t and n0Q t A sin nQ t are independent with mean A cos and A sin , respectively, and variance 2 , the variance of n(t).
Applying the theory of transformation of random variables we obtain the following wellknown Rice density function for the envelope
r r2 A2 =22
Ar
I0 2
r0
(3:141)
fR r 2 e
The function I0 x is the modied Bessel function of the rst kind of zero order. When there
is no signal, that is, A = 0 and I0 0 1, the Rice density function reduces to the Rayleigh
density function. Note that ER2 A2 2 2 . The ratio A2 =2 2 is the familiar signal-tonoise ratio.
(3:142)
XL t A cos nI t jA sin nQ t
(3:143)
The envelope of X(t) is the absolute value of XL t, which is R(t) given the above.
111
X t XI t cos 2fc t XQ t sin 2fc t , where both in-phase and quadrature processes
are independent. The received signal is a sum of attenuated and delayed-shifted and phaseshifted versions of X(t). For n independent paths we have
Y t
n
X
k1
n
X
k1
(3:144)
n
P
n
P
k1
k1
YL t YI t kYQ t
(3:145)
The path components XI ;k t and XQ;k t are functions of path attenuations, path delays,
and path phase-shifts. We are interested in the envelope R(t) of the received multipath
fading signal Y(t) at a xed time t. We observe that as the number of paths n approaches
innity, the central limit theorem dictates that both YI t and YQ t approach independent
Gaussian random variables with common variance 2s , which is also the variance of Y(t). If
there is no direct path then the mean of these in-phase and quadrature components is zero
and the density function of the envelope R(t) of Y(t) at a xed time t is a Rayleigh density
function. Note that ER2 2 2s is the power of all non-direct (diffuse) paths. If there is a
direct path with amplitude A then the density function of the envelope R(t) of Y(t) at a xed
time t is a Rice density function. Note that ER2 A2 2 2s is the power of the direct
path and all diffuse paths. The ratio A2 =2 2s is referred to as the direct signal-to-diffuse
signal ratio.
Squared envelope
The detection of the envelope of signal plus noise is the same as the detection of its
squared envelope since both quantities are nonnegative. For some applications it is more
mathematically tractable to deal with the squared envelope. The density function of the
squared envelope of noise Zt R2 t at a xed time t is the exponential density function
given by
fZ z
1 z=22
e
2 2
z0
(3:146)
On the other hand, the density function of the squared envelope of signal plus noise at a xed
time t is
p
1
A z
2
2
z0
(3:147)
fZ z 2 eA z=2 I0
2
2
112
L
X
k1
L
X
L
X
k1
(3:148)
k1
where L is the diversity order, and YL;k t and nL;k t are the complex envelopes of the signal
and noise in the kth diversity channel, respectively. We are interested in the density function
of U(t).
L
X
n2I;k t n2Q;k t
(3:149)
k1
In this case we have the sum of 2L squared Gaussian processes. These Gaussian processes are
mutually independent and each has zero mean and common variance 2 . The density function
of U(t) at a xed time t is commonly referred to as the 2 -density function (English pronunciation: chi-squared; Greek pronunciation: hee-squared) with n degrees of freedom, given by
fU u
1
2 2 L L
uL1 eu=2 u 0
2
1!
(3:150)
113
U t
L
X
(3:151)
k1
At a xed time t, the Gaussian random variables UI ;k t YI;k t nI;k t and the Gaussian
random variables UQ;k t YQ;k t nQ;k t are independent and have zero mean and
common variance 2k 2s 2 , where 2s is the power of the diffuse paths. The density
function of U(t) is the 2 -density function with 2L degrees of freedom, given by
fU u
1
2 2k L L 1!
uL1 eu=2 k u 0
2
(3:152)
L
X
(3:153)
k1
At a xed time t, the Gaussian random variables UI ;k t YI;k t nI;k t and the Gaussian
random variables UQ;k t YQ;k t nQ;k t are independent and have non-zero mean
Ak cos k and Ak sin k , respectively, and common variance 2k 2s 2 . The
density function of U(t) is the non-central 2 -density function with 2L degrees of freedom,
given by
p
1 u L1=2 A2 u=22
A u
k
fU u 2 2
e
IL1
u0
(3:154)
2k
2 k A
where the non-central parameter A2 is dened as
A2
L
X
k1
L
X
A2k
(3:155)
k1
114
fR r
(3:157)
where
O ER2
m
O2
ER2
O
1
2
(3:158)
(3:159)
The Rayleigh density function is a special case of the Nakagami-m density function where
m = 1 and O 2 2 . The Nakagami-m density function only approximates the exact density
function of the envelope of the sum of random vectors whose amplitudes and phases vary
according to certain statistical laws. However, the approximation is sufciently good
for many engineering applications. This fact has been veried experimentally in the
ionospheric and troposheric propagations.
(3:160)
We can transform the log-normal random variable X with decibel unit to ratio via the
following relation:
Y 10X =10 $ X 10 log Y
(3:161)
; y40
(3:162)
115
1 1 x2 =2
Qa p
e
dx
2 a
(3:164)
The outage probability is dened as the probability that the received power falls below the
threshold . We have
!
Pr
pout r PrPr Q
(3:165)
dB
For numerical illustration let us x the outage probability as pout r 0:01. The argument of
the Q-function is given by Pr = dB 2:32. Thus, we get Pr 2:32 dB . For
example, with dB 10 dB the mean value of the received power of the mobile has to be
23.2 dB above the threshold value to achieve this outage probability. Normally one would
like to achieve this benchmark at the cell boundary, where r R. In practice, the threshold
can be calculated given a threshold signal-to-noise ratio SNR (dB) at the cell boundary. If
the noise power N (dBW) is known as described in Chapter 5, the threshold signal power at
the cell boundary can be calculated as SNR N . For the above numerical example we
can then calculate the received power of the mobile at the cell boundary as
PR 2:32 dB SNR N 2:32 dB . Since PR PT LdB R the base station
can adjust the transmit power PT to meet the required threshold SNR if the path loss is known
either by measurements or by using an empirical model such as the Hata model (see
Chapter 5). The Hata model estimates the median path loss (dB) given the carrier frequency,
the transmit and receive antenna heights, the path loss exponent, and the distance between
base station and mobile for various environments. For the log-normal shadowing path loss
(dB), the median is the same as the mean value (the mean value of the Gaussian random
variable is also its median).
116
probability that the received signal exceeds the threshold (the fractional coverage area) is
given by
X
PrPri 4jAi PrAi
(3:166)
i
1 X
PrPri 4jAi Ai
R2 i
(3:167)
In the limit the above expression becomes the integral over the cell area. We have
1
P 2
PrPr4 dA
R Area
(3:168)
1 2 R
PrPr4 r dr d
2
R 0 0
Substituting (3.163) into (3.168) we obtain the fractional coverage area, as follows:
!
1 2 R
Pr
P 2
r dr d
(3:169)
Q
R 0 0
dB
Since Pr PT LdB r we can evaluate the path loss at distance r based on measurements or on the Hata model.
1. Measurement model: the path loss is given by LdB r LdB r0 10n logr=r0 ,
where r0 is a reference distance close to the base station but in the far-eld or
Fraunhofer region of its antenna (the far-eld distance is rf 2D2 =, where D is the
largest physical dimension of the antenna and is the signal wavelength). The reference
path loss LdB r0 can be obtained via measurement or it can be calculated using the Friis
formula for free space loss LdB r0 10 log4r0 =2 . The path loss exponent n is
obtained from measurements. Note that in ratio form the path loss is expressed as the
power-law Lr Lr0 r=r0 n. For free space (n = 2), the power-law expression
becomes the Friis formula that applies for any distance r.
2. Hata model: the path loss in the Hata model is the median path loss and is given by
LdB r 10n log r, where the path loss exponent n is given explicitly as a function
of the base station antenna height hT , n 4:49 0:655 log hT , 30 m hT 200 m
(see Chapter 5). The constant depends on the carrier frequency, and the base station and
mobile antenna heights.
In both cases the path loss can be expressed in general as LdB r 10n logr=R, where
LdB r0 10n logR=r0 for the measurement model and 10n log R for the
Hata model. The received power is Pr PT LdB r PT 10n logr=R. Let
a
PT PR
;
dB
dB
10n
dB
Then we have the following expression for the fractional coverage area:
(3:170)
117
3.6 Summary
2
P 2
R
R
r
Q a b log r dr
R
0
(3:171)
(3:172)
3.6 Summary
In this chapter we have presented a fairly detailed study of the random signals often encountered in
communications. These signals belong to two broad classes of random processes, namely, wide-sense cyclostationary (WSCS) and wide-sense stationary (WSS). The sequence of digital signals that carries random information
symbols is itself a WSCS process. Noise that contaminates the signals is a WSS process. Furthermore, in a
multipath fading channel, the envelope of the signal can be represented by special Rayleigh and Rice processes.
A WSS process is characterized by a constant mean and an autocorrelation that is a function of the time
difference. On the other hand, a WSCS process is characterized by periodic mean and autocorrelation.
A WSS process possesses a power spectral density function that is the Fourier transform of its autocorrelation.
A WSCS process with a uniform time shift becomes a WSS process. This procedure allows the calculation of the
power spectral densities of digital signal sequences. White noise has constant power spectral density. The most
important process in communication theory is the Gaussian process. We learned that the processing of a Gaussian
process by a linear lter also produces a Gaussian process. Noise that is encountered in communications is
modeled as white Gaussian noise. Signal processing in an additive white Gaussian noise channel using a
complete set of L orthonormal basis functions has sufcient statistics to be represented by exactly L samples for
detection. If the noise is non-white, a whitening lter can be employed at the receiver to obtain white noise for
subsequent signal processing. There are important Gaussian-derived processes that play an important role in
noncoherent demodulation of digital signals in noise and in analyzing their performance in a multipath fading
environment. They are the Rayleigh and Rice processes, which represent the envelope of noise and that of signal
plus noise in noncoherent demodulation. In a fading channel both processes represent the envelope of the sum
of diffuse signal vectors and that of the sum of direct and diffuse signal vectors, respectively. We also present the
Nakagami-m density function of the envelope of the sum of random signal vectors and the log-normal density
function for signals that encounter the shadowing effect.
118
Problems
1.
1 1 y2 =2
e
dy
p
2 x
2.
p
2
5ex =2 = 2 x
Using integration by parts show that Q(x)
is
upper
bounded
by
Qx
The characteristic function FX E ejX of the random variable X is dened as
1
fX xejx dx
FX
1
3.
(a) Find FX s of a Gaussian random variable X with zero mean and unit variance.
(b) Find FX s of a Gaussian random variable X with mean m and variance 2
The Chernoff bound is useful in establishing the upper bound for the distribution
function of a random variable X with corresponding moment generating function
FX s EesX . We have the following three forms of the Chernoff bound for real s:
1 FX x PrX 4x esx Fs; s40
FX x PrX x esx Fs; s40
FX s
Pr esX 4a
; a4 0
a
(a) Verify all three forms of the Chernoff bound.
(b) For a zero mean, unit variance Gaussian random variable, nd the Chernoff bound
x2 =2
:
for the complimentary distribution function, that
e
is,
pQx
40 show that E Q X
(c) For
a
random
variable
X
has
the
upper bound
p
E Q X 5FX 1=2, where
1 1 y2 =2
e
dy
Qx p
2 x
4.
Consider the complex Gaussian random variable X X1 jX2 , where X1 and X2 are
independent Gaussian random variables with zero mean and identical variance 2 .
The mean value and variance of X are
X EX EX1 jEX2
119
Problems
h
2 i
VarX E X X
h
h
i
2 i
We have X 0 and VarX E X X E j X j2 2 2 . Now consider the
5.
6.
7.
8.
9.
10.
11.
random variable Y ej X for arbitrary . Show that both Gaussian random variables
Y and X have the same mean and variance. (This property is called circular symmetry.)
A complex Gaussian random variable X X1 jX2 , where X1 and X2 are independent Gaussian random variables with zero mean and identical variance 2 , can be
represented in vector form as X X1 X2 t . The mean value of X is X 0, and the
variance is EX t X 2 2 . Express the complex random variable Y ej X for
arbitrary in vector form Y and calculate the covariance matrix K of Y.
Consider the process X t A sin2ft , where A is a constant and is a
random variable. Is X(t) wide-sense stationary? Find the conditions for which the
process is wide-sense stationary.
Consider the process X t A sin2ft , where is a discrete random variable
with probability distribution Pr 0 1=2; Pr 1=2 . Shows that X(t)
is not stationary by demonstrating that two random variables X(0) and X(a) for a
selected time t = a have different distributions. Is it wide-sense stationary?
Find the autocorrelation of the process X t A cos2ft , where is uniformly
distributed on 0; 2 and A is a Gaussian random variable with zero mean and unit
variance and is independent of .
Find the autocorrelation of the process Y t X t cos2ft , where is uniformly distributed on 0; 2 and X(t) is a wide-sense stationary process.
Consider a Gaussian vector X X1 X2 Xn whose independent elements have
zero mean and identical variance 2 . Find the moment generating function of the
random variable Y XX t
Derive the general expression for the autocorrelation of the random process
1
P
di pt iT with arbitrary pulse shape p(t) and independent bits
X t
i1
di ; assuming that the time delay is uniformly distributed in (0, T) and independent
of di , where T is the pulse time. Show that the power spectral density of the process
X(t) is given by
SX f
12.
13.
jPf j2
Sd ej2fT ;
T
Sd ej2fT
Rd kej2fkT
pendent and equally likely bits [0,1], where p(t) is the half-squared pulse shape of unit
amplitude and duration Tb, which is a bit time (Figure 3.3(a)), and di 2 f1; 0g
represents the normalized signal amplitudes. The mapping 0 ! 1 and 1 ! 0
maps bit 0 to amplitude 1 and bit 1 to amplitude 0 (the mapping 0 ! 0 and 1 ! 1
120
14.
15.
16.
17.
18.
maps bit 0 to amplitude 0 and bit 1 to amplitude 1 would give the same result).
Assuming the time origin is uniform in a bit time Tb, nd the power spectral density
function of X(t).
Consider the alternative mark inversion (AMI) signaling, where bit 1 is represented
by alternate positive and negative half-squared pulses and bit 0 is represented by a
zero voltage level. Assuming the time origin is uniform in a bit time Tb, nd the power
spectral density function of X(t).
Consider the Manchester signaling, where bit 1 is represented by a positive halfsquared pulse followed by a negative half-squared pulse and bit 0 is represented by a
negative half-squared pulse followed by a positive half-squared pulse. Assuming the
time origin is uniform in a bit time Tb, nd the power spectral density function of X(t).
Let X(t) be a WSS process and Y t X t T X t T . Find the autocorrelation
and power spectral density function of Y(t).
Let g(t) be a deterministic periodic function and X(t) is a zero mean ergodic process.
Find the autocorrelation and power spectral density of Y t gt X t
1
P
Consider the process X t
di pt iTb with uncorrelated and equally likely
i1
19.
20.
21.
22.
23.
24.
25.
26.
121
Bibliography
27.
Further reading
For a thorough review of probability theory, random variables, and random processes, we
recommend Papoulis [3], Helstrom [4], and Leon-Garcia [5]. Advanced materials can be
found in Feller [6]. More materials on cyclostationary processes can be found in Gardner [7].
Bibliography
1. R. N. McDonough and A. D. Whalen, Detection of Signals in Noise, 2nd edition,
London: Academic Press, 1995.
2. S. Haykin, Communications Systems, 2nd edition, Chichester: John Wiley & Sons,
1983.
3. A. Papoulis, Probability, Random Variables, and Stochastic Processes, 3rd edition,
New York: McGraw-Hill, 1991.
4. C. W. Helstrom, Probability and Stochastic Processes for Engineers, 2nd edition,
New York: Macmillan, 1991.
5. A. Leon-Garcia, Probability and Random Processes for Electrical Engineering,
2nd edition, Reading, MA: Addison-Wesley, 1994.
6. W. Feller, An Introduction to Probability Theory and its Applications, Vol. I
(3rd edition), Vol. II (2nd edition), Chichester: John Wiley & Sons, 1968, 1971.
7. W. A. Gardner, Introduction to Random Processes, 2nd edition, New York:
McGraw-Hill, 1990.
8. J. R. Cogdell, Modeling Random Systems, Harlow: Pearson Prentice-Hall, 2004.
9. M. Nakagami, The m-distribution-A general formula of intensity distribution of rapid
fading, in Statistical Methods of Radio Wave Propagation, edited by W. C. Hoffman,
Oxford: Pergamon Press, pp. 336, 1960.
10. P. Z. Peebles, Jr., Probability, Random Variables, and Random Signal Principles, 3rd
edition, New York: McGraw-Hill, 1993.
11. S. O. Rice, Mathematical analysis of random noise, Bell Syst. Tech. J., Vol. 23,
pp. 282332, 1944, Vol. 24, pp. 46156, 1945.
12. J. B. Thomas, An Introduction to Statistical Communication Theory, Chichester: John
Wiley & Sons, 1969.
Introduction
In this chapter we present the applicability of probability theory and random variables to the
formulation of information theory pioneered by Claude Shannon in the late 1940s [1,2].
Information theory introduces the general idea of source coding and channel coding. The
purpose of source coding is to minimize the bit rate required to represent the source
(represented mathematically by a discrete random variable) with a specied efciency at
the output of the source coder. On the other hand, the goal of channel coding is to maximize the bit rate at the input of the channel encoder so that code words can be transmitted
through the channel with a specied reliability. Both source coding and channel coding can
be achieved with the knowledge of the statistics of both the source and channel.
(4:1)
It is seen that the smaller the probability of the outcome xi, the larger its self-information.
Thus, a certain event has zero self-information and an impossible event has innite selfinformation. The average information in the source X with a given distribution is the
expected value of I(X) and is formally denoted as the entropy of X with the designated
notation H(X). Using (3.17) we have
HX EfIX g
M
X
i1
122
pX xi log2 pX xi
(4:2)
123
The use of the base-two logarithm indicates that the unit of entropy is the bit.
(4:3)
Let p = 0.1, then H(X) = 0.47 bits. This is the average number of bits needed per source
symbol. The entropy is maximum, that is, maxp HX 1 bit, when p = 0.5. This indicates
that the entropy is at maximum when the distribution is uniform or equally likely. Since the
average symbol length is 1 bit and the source is binary, the source coder transmits the symbol
unaltered.
(4:4)
The equality is achieved when the distribution is uniform (equally likely symbols), that is,
pX xi 1=M for all i.
M
P
i1
expression:
HX log2 M
pX xi log2 pX xi log2 M
pX xi log2 fMpX xi g
pX xi
pX xi
pX xi log2
ln MpX1xi
1
MpX xi
(4:5)
ln 2
X
1
1
1
pX xi
ln 2 i
MpX xi
i
Therefore,
1 X 1
pX xi
ln 2 i M
"
#
1 X1 X
pX xi
ln 2 i M
i
HX log2 M
1
1 1
ln 2
0
(4:6)
124
When pX xi 1=M for all i, that is, the distribution is uniform, we have
P
P
HX 1=M log2 1=M 1=M log2 1=M 1 log2 M .
i
i
The next discussion centers around source encoding, that is, what kind of code should one
use to represent source symbols with a given distribution such that the bit rate at the source
coder output is minimized? Intuitively one would think about a code with the average code
word length (each source symbol is represented by a code word whose length is in bits)
equal to the source entropy. After all, the source entropy, which is the average selfinformation of the source, is its best snapshot. We will present the best code that can
approach the prescribed goal. Furthermore, the average code word length is of interest
because if we transmit a long string of n symbols with a total length Ln bits, then the law of
large numbers dictates that the number of bits per symbols Ln =n approaches the average
code word length with a probability close to one.
x2 10;
x3 110;
x4 111
(4:7)
(4:8)
Thus, in a prex code each symbol can be decoded without looking to the right of it. The
prex codes are examples of uniquely decodable codes whose code words can be unambiguously reconstructed from the bit sequence. Here, it is assumed that synchronization is
established before decoding starts. The existence of a prex code can be veried by the Kraft
inequality, as stated in the following theorem.
Theorem 4.2 (Kraft inequality) The necessary and sufcient condition for the existence of a
prex code of M symbols with code words of length lm ; m 1; 2; :::; M , is
M
X
1
1
2l m
m1
(4:9)
125
The code in (4.7) satises the Kraft inequality with an equals sign. The following theorem
shows a relationship between the average code word length and the entropy.
(4:10)
Proof The rst step is to establish the expression HX l and show that it is less than or
equal to zero. We use the following identities:
log2 x log2 e ln x; ln x x 1
X
pX xi log2 pX xi
pX xi li
i
i
X
X
1
2li
li
log2 2
pX xi log2
pX xi log2
pX xi
pX xi
i
i
li
X
2
pX xi ln
log2 e
p
X xi
i
l
X
2 i
1
pX xi
log2 e
pX xi
i
!
X
X
2li
pX xi log2 e 1 1 0
log2 e
HX l
(4:11)
(4:12)
Then, the average code word length satises the following inequalities:
HX l HX 1
(4:13)
1.
Since
the
Kraft
inequality
is
satised
there
exists
a
prex code
X
i
i
i
with the above selected lengths such that the left-hand side of (4.13) is satised per Theorem
4.3. From the right-hand side of (4.12) we have
X
X
X
l
p x l
p x log2 pX xi
p x HX 1 (4:14)
i X i i
i X i
i X i
Proof The
li
126
(4:15)
l5 2
(4:16)
5
X
pX xi li 2:8 4 HX
i1
5
X
pX xi log2 pX xi 2:12
(4:17)
i1
The above theorem shows that one can do no worse than H(X) + 1 with ShannonFano
codes. The question is, can one do better? In the following discussion we present the Huffman
codes, which are optimum in the sense that they are the prex codes that approach the entropy
in average lengths. Furthermore, no other prex codes have smaller average code word lengths.
Huffman codes
A Huffman code is constructed as follows:
1. The source symbols are listed in order of decreasing probability.
2. A tree is constructed to the right with the two source symbols of the smallest probabilities
combined to yield a new symbol with the probability equal to the sum of the two previous
probabilities.
3. The nal tree is labeled with 0 on the lower branch (upper branch) and 1 on the upper
branch (lower branch).
(4:18)
We construct the two trees shown in Figure 4.1. Both Huffman codes have the following
average length in bits:
l 0:14 0:14 0:23 0:22 0:41 2:2
l 0:13 0:13 0:22 0:22 0:42 2:2
(4:19)
The source entropy is H(X) = 2.12. If we compare the Huffman code with the ShannonFano
code in the previous example we conclude that the former is more efcient.
127
x5
0.4
1
x4
x3
x2
0.2
1
0.6
0.2
1
0.1
x5
0.1
0.4
0.4
1
0.2
x1
0
(a)
1
0.6
x4
x3
x1 = 0000
x2 = 0001
x3 = 001
x4 = 01
x5 = 1
0.2
x1 = 000
x2 = 001
x3 = 01
x4 = 10
x5 = 11
0.2
1
0
0.4
x2
0.1
1
0.2
x1
0.1
0
(b)
Figure 4.1
Huffman codes.
(4:20)
Proof Let us consider the nth extension of the source X called X n , which consists of
M n extended symbols. Each extended symbol consists of n concatenated original source
symbols and has a probability equal to the product of n probabilities. Since the log of the
product equals the sum of logs, the entropy of the extended source is n times the entropy of
the original source, that is,
HX n nHX
(4:21)
Applying the result of Theorem 4.4 to the extended source we obtain the following bounds
on L, the average code word length of X n :
128
HX n L HX n 1
(4:22)
L
1
HX
n
n
(4:23)
Note that l , the average code word length of the original source X, is equal to L=n since the
extended symbol has n original symbols. Therefore,
HX l HX
1
n
(4:24)
In the subsequent discussion we introduce the concept of channel coding to maximize the
rate of reliable transmission over a noisy channel. This necessitates the evaluation of the
Shannon channel capacity, which cannot be exceeded if reliable transmission is to be
accomplished. We investigate information capacities of discrete and Gaussian channels.
mi
Figure 4.2
Channel
encoder
ui U
Discrete channel
{p(vj ui)}
vj V
Channel
decoder
m i
129
Antenna
Channel
encoder
Modulator
Up converter
High-power
amplifier
Antenna
Channel
decoder
Demodulator
Down
converter
Low-noise
amplifier
Discrete channel
Figure 4.3
A physical channel.
code the message block consists of any k source symbols, and hence there are M 2k
distinct message blocks. The channel encoder may encode each message block into an n-bit
input symbol (channel code word) ui , where n > k. Thus, there are a total of M 2k input
symbols. Given the n-bit input symbol ui ui1 ; :::; uin the probability of observing the
n-bit output symbol vj vj1 ; :::; vjn is the following transition probability:
pvj jui
n
Y
pvjl juil
(4:25)
l1
Thus throughout our discussion we assume that the channel is characterized by the conditional probabilities pvjl juil ; i 2 U ; j 2 V , and hence pvj jui :
Mutual information
The concept of mutual information is intuitive. Suppose a symbol ui was sent by the channel
encoder with probability pU ui . This is often called the a-priori probability of the input
symbol ui . The probability that ui was the sent symbol after the observation of the output
symbol vj becomes pui jvj . This is often referred to as the a-posteriori probability of ui . If the
channel is noiseless then after receiving vj one would know exactly that ui was sent and the
a-posteriori probability is 1. Now consider the case of a very noisy channel. The reception of vj
will not help at all in determining what input symbol in the set U was sent. This means that the
a-posteriori probability is the same as the a-priori probability. If the channel is somewhere
between these two extreme cases, then the a-posteriori probability is larger than the a-priori
probability. Hence, one obtains an information gain from the reception of the output symbol.
This can be expressed quantitatively as the difference between the self-information of ui and
the self-information of ui after vj was observed, and is called the pair-wise mutual information:
1
1
log2
pU ui
pU ui jvj
pU ui jvj
pV vj jui
log2
log2
pU ui
pV vj
Iui ; vj log2
(4:26)
130
XX
j
XX
i
pUV ui ; vj log2
pUV ui ; vj log2
pU ui jvj
pU ui
(4:27)
pV vj jui
pV vj
pU ui log2 pU ui
XX
!
pUV ui ; vj log pU ui jvj
HU HU jV
(4:28)
where the entropy H(U) and the conditional entropy H(U|V) are given by
X
pU ui log2 pU ui
HU
(4:29)
HU jV
pV vj HU jvj
XX
XX
i
(4:30)
(4:31)
(4:32)
HV jU
pU ui HV jui
XX
XX
i
(4:33)
131
Channel capacity
The information channel capacity is the maximum of the mutual information over all
possible input symbol distributions
C max IU ; V
pU ui
(4:34)
To see why the maximum is carried out over pU ui only, we need to examine (4.32) and
P
(4.33). Applying the total probability theorem we have pV vj i pV vj jui pU ui .
Furthermore, pUV ui ; vj pV vj jui pU ui , thus I(U,V) is a function of pU ui and
pV vj jui . The channel transition probabilities pV vj jui which completely describe the
channel are xed once the channel model is specied. This leaves the designer with only
the choice of nding a suitable pU ui to maximize the mutual information. Thus, the
channel capacity depends only on the channel transition probabilities. In practice, nding
a suitable input symbol (code word) distribution is equivalent to nding a suitable channel
code to approach the channel capacity dened in (4.34). A suitable code must apparently
have code words that contain long message blocks to be efcient and the code must be able
to correct as many errors as possible in the message blocks given the channel model, that is,
its transition probabilities. The smaller the specied bit error probability, the larger the size
of the code words in order to average out the channel errors (by the law of large numbers). In
other words, there exist codes that allow one to arbitrarily approach channel capacity via the
optimization in (4.34) and at the same time arbitrarily approach a specied error rate via
the law of large numbers. This is the essence of Shannon noisy channel coding. If the
information rate exceeds the Shannon channel capacity, then the error rate cannot be kept
arbitrarily small.
The channel capacity unit is coded bits/input symbol since this is the unit of entropy. If the
unit is in coded bits/second, then we need to multiply the channel capacity in coded bits/
input symbol by the input symbol rate in input symbols/second to have the channel capacity
in coded bits/second. We can restate the Shannon channel capacity in an alternate way
using the Shannon main theorem, described next.
132
The meaning of the Shannon main theorem can be interpreted in the following manner.
Let X be a discrete source with M source symbols and entropy H(X) in bits/source symbol.
The source symbols are encoded into M channel input symbols (code words). For example,
if the source symbol (message block) has k bits and the input symbol (code word) has n > k
coded bits, then the code rate is r = k/n. Assume that the source transmits R source symbols/
second, then the source entropy in bits/second is H = RH(X) bits/second. The channel
capacity C is given in coded bits/input symbol. The channel capacity is C = RC/r coded bits/
second. If H C then we have RH X RC=r. This implies r C=HX . The key to
designing a good communication system is to nd a code with rate r close to C=HX that
can meet a specied error rate.
The converse of the Shannon noisy channel coding theorem is if H > C it is not possible to
transmit the output of the source over the channel with an arbitrarily small frequency of errors.
With the publication of his work [1,2], Shannon opened the eld of coding theory and set
the course for coding theorists to discover many types of efcient codes [4,5] capable of
achieving what Shannon stated in his famous main theorem.
(4:35)
Using (4.31) and (4.34) we obtain the capacity of the binary symmetric channel as follows:
C 1 p log2 p 1 p log2 1 p
(4:36)
0
p
p
Figure 4.4
1p
133
small error probability, then according to the Shannon main theorem there exist codes with
rate r C=HX 0:919=1 0:919 that can accomplish this task. For simplicity we
consider the repetition code (n,1) with code rate r = 1/n, where an information bit is encoded
by repetition into a code word (channel input symbol) of n identical bits. For example, in a
(3,1) repetition code the following encoding is carried out: 0 ! 000; 1 ! 111. In
practice, the code word length n is commonly chosen to be an odd integer. Thus, if a
received code word has more 0s than 1s, the decoded bit is bit 0. Otherwise, the decoded bit
is bit 1. An error occurs when bn 1=2c or more coded bits out of n bits are received
incorrectly, where bn 1=2c is the largest integer no greater than n 1=2. Thus a (n,1)
repetition code can correct t = bn 1=2c errors in a code word. The bit error probability,
which is the same as the code word error probability (a code word carries only one
information bit), for the coded system is given by
n
X
n i
Pe
(4:37)
p 1 pni
i
it1
Numerically we have r 1=3 yields Pe 3 104 and r 1=5 yields Pe 106 . Thus
we can achieve arbitrarily small error probability by using longer codes. For the same
transmission rate a repetition code decreases the channel throughput by a factor n. This also
implies that the transmission rate over the channel will be n times larger for a constant
throughput, and hence the physical channel must have a larger bandwidth to accommodate
this higher transmission rate. In the next example we introduce a more efcient code to
reduce the transmission rate as compared with the repetition code.
n 2m 1
k 2m 1 m
nk m
t=1
dmin 3
The message block is the number of information bits; the parity-check block is unique to
the code and is a linear sum of the information bits. The Hamming distance between two
different code words is the number of positions in which they differ. The error-correcting
capability t is related to the minimum Hamming distance as t bdmin 1=2c. In other
words, we also have dmin 2t 1. Note that this relationship holds for repetition code (n,1)
for any n (even or odd). To prove this relationship, which is true for any linear block code, let
t be a positive integer such that the following inequalities hold:
134
Code word
Message block
Code word
0000000
1010001
1110010
0100011
0110100
1100101
1000110
0010111
1000
1001
1010
1011
1100
1101
1110
1111
1101000
0111001
0011010
1001011
1011100
0001101
0101110
1111111
2t 1 dmin 2t 2
(4:38)
Let u, v, and w be the transmitted code word, the received word, and an arbitrary code word in the
code. Applying the triangle inequality to the Hamming distances among u, v, and w, we obtain
du; v dw; v du; w
(4:39)
Note that du; w dmin 2t 1, therefore any error pattern which has less than or equal
to t errors, that is, du; v t, results in the following inequality for an arbitrary w:
dw; v 2t 1 t t 1
(4:40)
The decoder selects u over all ws in the code since u is closer in Hamming distance to v.
Table 4.1 shows the 16 code words of the (7,4) Hamming code together with the
corresponding message block. The rst three bits are the parity-check bits and the last
four bits are the message bits.
The code word error probability is given in (4.37). Using the approximation that for an
error pattern of i > t errors, at most i + t errors can occur if all i channel errors occur in
the information bits and t correct bits are changed by the decoder. Thus, there is a fraction
(i + t)/n of k information bits to be decoded erroneously, and hence the bit error probability
of the coded system is given by [5]
n
X
it n i
Pe
(4:41)
p 1 pni
i
n
it1
For a BSC with p 102 we obtain the bit error probability Pe 6 104. Therefore, the
(7,4) Hamming code increases the transmission rate by a factor 7/4.
135
N(0, 2)
N(0, 2)
(a)
Figure 4.5
(b)
(4:42)
The noise variable N is a Gaussian random variable with zero mean and variance 2 . It is a
common practice to write the Gaussian noise variable as N 0; 2 . The probability density
function of N 0; 2 is given by
1
2
2
fN p e =2
2
(4:43)
This channel model is usually referred to as an additive Gaussian noise (AGN) channel. If
the noise has a constant power spectral density function (called white noise) then the channel
is referred to as an additive white Gaussian noise (AWGN) channel. Figure 4.5(b) depicts
the case of a multiple input-multiple output Gaussian channel (vector Gaussian channel),
where the continuous output random vector X is the sum of the input random vector S and
the independent continuous channel noise vector N:
X SN
(4:44)
The components of N are independent and identically distributed (iid) Gaussian random
variables, each with zero mean and common variance 2 . For the vector Gaussian channel,
the probability density function of the noise vector N0; 2 is simply the joint density of its
n independent components and is given by
n
P
2i =2 2
1
fN h p n e i1
2
(4:45)
Figure 4.6 illustrates the Gaussian channel model of a coded communication system.
The mapping ui ! S i represents the equivalent lowpass model of the modulator. It maps
the code word ui into the voltage vector S i. For example, the code word ui = (010) is
p p p
mapped into S i Eb ; Eb ; Eb , which is the equivalent lowpass vector of the
set of three PSK signals A cos 2fc t; A cos 2fc t; A cos 2fc t, each dened in their
time interval. The mapping X ! vj represents the equivalent lowpass model of the
demodulator. It maps the received voltage vector X into the received word vj. The two
mappings and the Gaussian channel model constitute the discrete channel shown in
Figure 4.2.
136
N(0, 2)
mi
Channel
encoder
ui U
Mapping
ui Si
Si S
Mapping
X vj
vj V
Channel
decoder
m i
Gaussian
channel
Figure 4.6
Since the Gaussian random variable is a continuous random variable, the denition of
entropy for a discrete random variable does not apply in a straightforward manner. To
illustrate this case let us approximate the probability density function fX x by a probability
distribution function pi , where pi fX xi x is the area under fX x in the ith interval of
width x. Note that in the limit we have
X
pi
x!0
fX xi x !
fx xdx 1
(4:46)
1
pi log2 pi
fX xi x log2 fX xi x
fX xi log2 fX xi x log2 x
(4:47)
fX xi x
fX xi log2 fX xi x log2 x
In the limit as x ! 0 the second term becomes innite and renders the entropy of a
continuous random variable innite. But since the second term of H is common to any
two density functions fX x and gX x, the difference in their entropies is equal to the
difference of the rst terms and is nite as long as their rst terms are. The rst term of H is
commonly referred to as the differential entropy.
Differential entropy
Denition 4.6 The differential entropy h(X) of a continuous random variable X with a given
density function fX x is dened as
1
hX
fX x log2 fX x dx
(4:48)
1
137
Denition 4.7 The differential entropy h(X) of a continuous n-dimensional random vector X
with a given density function fX x is dened as
hX fX x log2 fX x dx
(4:49)
1
1
xX 2 =2 2
fX x log2 p e
dx
hX
2
1
"
#
1
p
x X 2
fX x log2 2
log2 e dx
2 2
1
p
log2 e
log2 2
EfX X 2 g
2 2
(4:50)
1
1
log2 2 2 log2 e
2
2
1
log2 2e 2
2
Therefore, the differential entropy of a Gaussian random variable is independent of
the mean.
(4:51)
Theorem 4.8 (Maximum differential entropy) The Gaussian random variable X with
arbitrary mean X and variance 2 has the largest differential entropy of all random
variables of the same mean and variance.
138
Proof
(4:52)
we obtain, for arbitrary density functions fX x and gx with the same mean X and variance 2 ,
1
gX
gx
E log2
dx
fX x log2
fX X
f
X x
1
1
gX
gx
(4:53)
log2 E
dx
fX x
log2
fX X
f
X x
1
1
log2
gx dx log2 1 0
1
Hence, the upper bound of the differential entropy h(X) of the random variable X with
arbitrary density function fX x is given by
1
1
fX x log2 fX x dx
fX x log2 gx dx
(4:54)
hX
1
1
1
1
xX 2 =2 2
hX
fX x log2 p e
dx
2
1
(4:55)
1
log2 2e 2
2
The right-hand side of the above inequality is the differential entropy of a Gaussian random
variable with variance 2 . Thus the Gaussian random variable attains the largest differential
entropy. In summary we have
1
maxEfX X 2 g 2 hX log2 2e 2
2
(4:56)
The above result can be generalized to the case of a random vector X with iid components as
follows:
n
maxK hX log2 2e 2
2
(4:57)
139
Denition 4.9 The mutual information I(X,Y) between two continuous random variables X
and Y with a given joint density function fXY x; y fY jX yjxfX x fX jY xjyfY y, is
dened as
1 1
fXY x; y
fXY x; y log2
IX ; Y
dx dy
f
X xfY y
1 1
1 1
1 1
1 1
1
1
fXY x; y log2
fY jX yjx
dx dy
fY y
fXY x; y log2
fX jY xjy
dx dy
fX x
(4:58)
Denition 4.10 Let X and Y be two continuous random variables with the joint
density function fXY x; y. The conditional differential entropies h(X|Y) and h(Y|X) are
dened as
1
1
hX jY
fY y
fX jY xjy log2 fX jY xjy dx dy
1
1 1
1 1
1
(4:59)
and
hY jX
1
1
fX x
1 1
1
1
1
1
From the above denitions we obtain the mutual information in terms of the differential
entropies as follows:
IX ; Y hX hX jY hY hY jX
(4:61)
(4:62)
140
are independent zero mean Gaussian random variables, and variance 2i , i = 1,2, . . . , n, is
given in bits/vector transmission (bits/vector channel use) as
C maxESS 0 P IS; X maxESS 0 P
n
X
1
i1
log2 1 SNRi
(4:63)
P
where SNRi Pi = 2i and ni1 Pi P.
The calculation of the channel capacity follows directly from the denition of mutual
information. We have
IS; X hX hX jS
hX hS N jS
hX hN jS
(4:64)
hX hN
1
hX log2 2e 2
2
Furthermore, EX 2 EfS N 2 g ES 2 2SN N 2 P 2 and hence the
entropy of X is upper bounded by the entropy of a Gaussian random variable with variance
P 2 , which is 12 log2 2eP 2 . Therefore, we obtain
1
P
1
IS; X log2 1 2 log2 1 SNR
2
(4:65)
The information capacity of the Gaussian channel is the maximum of I(S, X) and is achieved
when input S is a Gaussian random variable with zero mean and variance P.
The result for a vector Gaussian channel can be investigated via the constraint optimization problem, which for this case can be solved using Lagrange multipliers. Consider the
Lagrangian
n
n
X
X
1
Pi
Pi
(4:66)
log2 1 2 l
Ll; P1 ; :::; Pn
2
i
i1
i1
where l is the Lagrange multiplier. The KuhnTucker condition for the optimality of a
power allocation is given by
L 0 if Pi 4 0
(4:67)
Pi 0 if Pi 0
And hence the power constraint optimization problem yields the following power
allocation:
1
2i
(4:68)
Pi max 0;
2l
P
where the Lagrange multiplier is selected such that ni1 Pi P.
141
In the special case where the noise vector has iid components with zero mean and
common variance 2 , the power allocation strategy allocates equal signal power P to each
subchannel, that is, P P=n. Therefore,
n
C maxESS t P I S; X log2 1 SNR
2
(4:69)
i2
P9 = 0
P7 = 0
P1 = 0
1
2
P2
P3
P4
Figure 4.7
Subchannel
142
For numerical illustration consider the case of three subchannels with noise powers
21 5; 22 1; 23 8 power units. The total signal power for allocating to the three
subchannels is P 10 power units. We have
1
1
2
P1 max 0;
1 max 0;
5
2l
2l
1
1
(4:70)
P2 max 0;
22 max 0;
1
2l
2l
1
1
2
3 max 0;
8
P3 max 0;
2l
2l
Pn
subject to
i1 Pi P 10. Solving for the subchannel signal power we obtain
P1 3; P2 7, and P3 0 with 1=2l 8.
Using the above background fundamentals we arrive at the following information
capacity theorem for a Gaussian channel with a power constraint.
Theorem 4.12
2
For a rate less than C, there exists a coding system such that the code words can be
transmitted over the channel with an arbitrarily small frequency of errors.
pThe
nP 2 . Since each decoding sphere is associated with a code word, the more
143
decoding spheres that can be packed within the larger sphere, the more code words that can
be sent and the higher the channel capacity. The volume of an n-dimensional sphere is
proportional to the nth power of its radius. Thus, the maximum number of decoding spheres
that can be packed into the receiving sphere is at most equal to
nP 2 n=2
n 2 n=2
n
2
P n=2
1 2
22 log2 1P=
(4:72)
This number is also the maximum number of code words that can be sent. Hence, the
number of bits that can be reliably sent over the Gaussian channel per code word is at most
n
1
2
2
2 log2 1 P= , and the number of bits per transmission is 2 log2 1 P= .
1 : A cos 2fc t; 0 t 5T ! S P.
Here T represents the transmission time of a PSK waveform (bit time) and ES 2 P. The
bit errorp
probability for optimum detection of PSK signals is given in Chapter 7 as
p Q 2SNR, where Q is the Gaussian integral function dened in (3.164) and
repeated here for convenience:
1 1 x2 =2
Qa p
e
dx
(4:73)
2 a
Thus, we can convert the Gaussian channel with PSK signaling into a binary symmetric
channel with crossover probability p and capacity C 1 p log2 p 1 p log2 1 p.
The maximum transmission rate of a PSK signaling is 1 bit per transmission and this
happens when p = 0 (noiseless channel or innite SNR). For numerical illustration consider
the case of SNR = 2.7 or 4.3 dB, which results in p = 102. The capacity of the BSC is C =
0.919 bits. For the same SNR the information capacity of a Gaussian channel with Gaussian
input is given by C 12 log2 1 SNR 12 log2 1 2:7 0:94 bits. Thus, at p = 102, the
capacity of a BSC is about 3% below that of a Gaussian channel with Gaussian input.
Now let us consider the case of SNR = 9.1 or 9.6 dB. This results in a PSK bit error probability
of p = 105. The information capacity of the BSC is very close to 1 bit/transmission. On the other
hand, to achieve the information capacity of 1 bit/transmission, the Gaussian channel with
Gaussian input requires SNR = 3 or 4.8 dB. Thus, there is a potential power gain of 9.54.8 =
4.7 dB for communication over the BSC at 1 bit/transmission at a bit error rate of 105.
144
2
Since there are 2B samples each second, the capacity can be expressed in bits/second as follows:
P
C B log2 1 2 B log2 1 SNR bits=second
(4:75)
M
X
fXjs xjsi
C max
dx
pi fXjsi xjsi log2 i
pi
fX x
i1
M
(4:76)
X
fXjs xjsi
dx
pi fXjsi xjsi log2 M i
max
pi
P
i1
pj fXjsj xjsj
j1
145
where the integral is an L-fold integral and the L-dimensional Gaussian density function
fXjsi xjsi is
PL
2
1
1
fXjsi xjsi p L e22 k1 xk sik
2
2
1
2
(4:77)
p L ekxsi k =2
2
For digital communications all symbols are equally likely, therefore the signal vector
distribution pi ; i 1; 2; :::; M , is constrained to pi 1=M ; i 1; 2; :::; M . Thus the channel capacity given a set of si ; i 1:2; :::; M is
C
M
X
fXjs xjsi
1
fXjsi xjsi log2 M i
dx
P
M
i1
pj fXjsj xjsj
j1
M
X
1
1
kxsi k2 =2 2
log2
p L e
M
2
i1
ekxsi k =2
dx
M
P
2
2
kxs
k
=2
1
j
M
e
2
(4:78)
j1
M
X
1
1
kyk2 =2
log2
pL e
M
2
i1
ekyk =2
dy
M
P
2
M 1 ekyvi vj k =2
j1
M
M
X
X
2
2
1
1
kyk2 =2
log2 M
log2
ekyvi vj k =2 ekyk =2 dy
pL e
M
2
i1
j1
2
2
t
Since y vi vj vi vj 2 vi vj y kyk2 , we obtain
(4:79)
M
M
X
X
2
2
1
1
kyk2 =2
log2
ekyvi vj k =2 ekyk =2 dy
C log2 M
pL e
M
2
i1
j1
M
M
X
X
2
t
1
1
kyk2 =2
log2 M
log2
ekvi vj k =2 vi vj y dy
pL e
M
2
i1
j1
(4:80)
For one-dimensional (L = 1) signal vectors such as PSK and ASK and two-dimensional (L =
2) signal vectors such as MPSK and MQAM (Chapter 6) we note that the average energy
P
2
per dimension is 1=LM M
is 2 , therefore
i1 ksi k , and the noise variance per dimensionP
2
the average signal-to-noise ratio per dimension is SNR 1=LM M
i1 ksi =k
PM
2
1=LM i1 kvi k . Thus, the channel capacity for these digital signals can be calculated
as a function of the average signal-to-noise ratio per dimension. For a white Gaussian noise
146
channel with power spectral density N0 =2, the variance of a component of the noise vector N
is 2 N0 =2 (Chapter 7). The squared length (squared norm) of the signal vector si is the
signal energy Ei, that is, Ei ksi k2 . This is the result that arises from the use of orthonormal
basis functions to obtain the signal vectors (Chapters 2 and 6). Furthermore, the noise vector
provides sufcient statistics for the detection of signal vectors (Chapter 3). Hence, the
average signal-to-noise ratio per dimension is SNR 2Es =LN0 , where the average signal
P
(symbol) energy is Es and is given by Es 1=M M
i1 Ei. Since vi si =, we get
be expressed in term of the
kvi k2 2Ei =N0 . The normalized signal vector vi can therefore
p
average symbol energy-to-noise density ratio Es =N0 as vi Es =N0 ci , where ci is a xed
constant vector characterizing a given signal set. The channel capacity can then be calculated as a function of Es =N0 and is shown in Figure 4.8 for some digital modulation
techniques [12,14,28]. These are two-dimensional (L = 2) signaling techniques, therefore
the integral in (4.80) is a double-integral. The capacity of the vector Gaussian channel is
C L=2 log2 1 SNR log2 1 SNR bits/symbol. The average signal-to-noise
ratio per dimension is SNR Es =N0 . The potential coding gain (reduction in SNR at
the same bit error probability) is about 7.58.5 dB if channel coding is employed. For
example, QPSK requires 13.5 dB in SNR to achieve a bit error probability of 106 and a
capacity of 2 bits/symbol. On the other hand, the channel capacity theorem says that the
required SNR is only 5 dB for communication at 2 bits/symbol. Furthermore, channel coding
does not need to increase the bandwidth as with repetition codes or Hamming code. Instead,
5
32QAM
4.5
4
16QAM
C, bits/symbol
3.5
3
8PSK
2.5
2
4PSK
1.5
1
PSK
0.5
0
5
Figure 4.8
10
Es/N0, dB
15
20
Capacity of digital modulation techniques versus the Shannon limit log2 1 SNR; where SNR Es =N0 .
25
147
one can employ an expand signal space, for example, from 16QAM to 64QAM, to
compensate for the increase in the channel bit rate due to the redundancy of the code. The
use of 64QAM (capacity is 6 bits/symbol) reduces the bandwidth by a factor of 6/4 as
compared to 16QAM (capacity is 4 bits/symbol). This fact permits the use of codes with rate
4/6 with 64QAM to achieve the same bandwidth as uncoded 16QAM (thus, both uncoded
16QAM and coded 64QAM signals provide a capacity of 4 bits/symbol). Furthermore, at
4 bits/symbol, 64QAM requires about 8 dB less in SNR as compared to 16QAM, thus a
potential coding gain of 8 dB can be achieved with codes of rate 4/6.
(4:81)
The generator matrix G can be put into the following forms which produce linear systematic
block codes:
G Ik
P or G P
Ik
(4:82)
The matrix Ik is the k k identity matrix. For example, the following generator matrices can
produce the set of 16 Hamming code words:
2
1
60
G6
40
0
0
1
0
0
0
0
1
0
0
0
0
1
1
1
1
0
0
1
1
1
3
2
1
1
60
17
7and G 6
41
05
1
1
1
1
1
0
0
1
1
1
1
0
0
0
0
1
0
0
0
0
1
0
3
0
07
7
05
1
(4:83)
Note that for systematic codes generated by G P I k the last k bits of the code words
are identical to the information bits. On the other hand, if G Ik P is used, the rst k
bits of the code words are identical to the information bits. Note that the rows of G are
linearly independent.
Associated with a generator matrix is an n k n parity check matrix H that satises
the relationship
HGt 0 or GHt 0
(4:84)
148
where 0 is a null matrix. For a systematic code the parity-check matrix assumes the
following forms
H Pt I nk or H Ink Pt
(4:85)
It is seen that c is a code word if and only if the following relationship holds:
cHt 0
The parity-check matrices for the above generator matrices are given by
2
3
2
3
1 0 0 1 0 1 1
1 1 1 0 1 0 0
H 4 0 1 1 1 0 1 0 5 and H 4 0 1 0 1 1 1 0 5
1 1 0 1 0 0 1
0 0 1 0 1 1 1
(4:86)
(4:87)
The symmetry relationship in (4.84) shows that H can be used as a generator matrix for an
n; n k linear block code and G as a parity-check matrix. The rows of H are linearly
independent. The parity-check matrix therefore performs n k parity checks on n bits of the
received word.
The decoding of simple block codes can be carried out via syndrome decoding. Let the
code word c be transmitted over a noisy channel, and hence the received word r may contain
errors as follows:
rce
(4:88)
where e is the error word or error pattern, which is the zero vector when there is no error.
The syndrome vector s is dened as
s rHt
c eHt
eH
(4:89)
When there is no error, the syndrome is a zero vector. For block codes that can correct a
single error in the code words like Hamming codes, the syndrome is the column of H that
corresponds to the error position in the error vector e. Because all columns of H are distinct,
it is simple to locate the column of H that is identical to the syndrome s and correct the error.
149
This syndrome matches the second column of H and indicates an error in the second element
of the vector r, which is 1 and can be corrected to 0.
In general, syndrome decoding is not effective for long linear block codes with multiple
errors correcting capability. In the following discussion we present some important linear
block codes.
Cyclic codes
A linear block code is cyclic if a cyclic shift of a code word is also a code word. For example,
if c0 ; c1 ; :::; cn1 is a code word, then cn1 ; c0 ; :::; cn2 , . . . , c1 ; c2 ; :::; cn1 ; c0 are also
code words. A cyclic linear block code (n, k) is characterized by its generator polynomial
gx 1 g1 X gnk1 X nk1 X nk , which yields the following generator
matrix:
2
1
60
6
6
G 60
6 ..
4.
g1
1
0
..
.
0 0
g2
g1
1
..
.
:::
:::
g2
g1
..
.
gnk1
:::
g2
..
.
1
gnk1
:::
..
.
0
1
gnk1
..
.
g1
g2
0
0
1
..
.
0
0
0
..
.
::: gnk2
:::
:::
:::
..
.
gnk1
3
0
07
7
07
7 (4:91)
.. 7
.5
1
Note that the generator matrix of a cyclic code is not in systematic form. For example, the
(7,4) cyclic Hamming code with the generator polynomial gx 1 X X 3 is given by
2
1
60
G6
40
0
1
1
0
0
0
1
1
0
1
0
1
1
0
1
0
1
0
0
1
0
3
0
07
7
05
1
(4:92)
Golay codes
The Golay code is a (23,12) cyclic code with an error-correcting capability of t = 3 and a
minimum Hamming distance dmin 7. The two generator polynomials for Golay codes are
gx 1 X 2 X 4 X 5 X 6 X 10 X 11 , and gx 1 X X 5 X 6 X 7
X 9 X 11 .
150
n 2m 1
Parity-check block
n k mt
Error-correcting capability
dmin 2t 1
n 2m 1
Parity-check block
n k 2t
Error-correcting capability
dmin 2t 1
Since RS codes can correct random symbol errors which are equivalent to random bursts of
m bit errors, RS codes are suitable for wireless applications in fading channels. The IEEE
802.16e employs RS (255,239) code with 8 bits/symbol in GF(28). The bit error probability
of an RS code is given by a modied form of (4.41) as follows:
Pe
n
2m1 X
it n i
ps 1 ps ni
i
2m 1 it1 n
(4:93)
where ps is the symbol error probability at the decoder input and 2m1 =2m 1 is the
average number of bit errors per symbol error.
23
X
i 3 23
i4
23
(4:94)
151
At this bit error probability an uncoded PSK system would require SNR = 12.3 dB. Thus a
coding gain of 12.3 9.6 = 2.7 dB is realized. Note that this coding gain is achieved at the
expense of bandwidth, which increases by a factor 1/r = 23/12 = 1.92.
Concatenated codes
To achieve the large error-correcting capability of a code with long block length, two shorter
codes connected in concatenation can be employed. The rst code is the outer code no ; ko
with code rate ro ko =no and the second code is the inner code ni ; ki with code rate
ri ki =ni . For example, the outer code can be an RS code with ki -bit symbols. Each RS code
symbol is then encoded by the inner code to form an inner code word of length ni . The
overall code rate is r ro ri . The bit error probability of concatenated codes can be evaluated
by rst calculating the bit error probability of the inner code using (4.41), for example, then
nding the bit error probability of the outer code using (4.93).
7
X
i1 7
i2
(4:95)
152
15
241 X
i3
24 1 i4 15
15 i
ps 1 ps 15i 9 1010
i
(4:96)
At this bit error probability an uncoded PSK system would require SNR = 12.6 dB. Thus a
coding gain of 12.6 9.6 = 3 dB is realized. Note that this coding gain is achieved at the
expense of bandwidth, which increases by a factor 1/r = 2.92. The advantage is the ability of
the RS outer code to correct an error burst of up to 4 consecutive bits.
1
61
6
H6
60
40
0
1
0
0
1
0
1
0
1
0
0
1
0
0
0
1
0
1
0
0
1
0
1
0
1
0
0
1
1
0
0
0
0
0
1
1
0
0
1
0
1
3
0
07
7
17
7
15
0
(4:97)
153
1
61
6
60
H6
61
6
40
0
1
1
0
0
1
0
1
1
0
0
0
1
0
1
0
1
1
0
0
1
0
0
1
1
1
0
1
0
0
1
1
0
1
0
1
0
0
0
1
1
1
0
0
0
0
1
1
1
0
0
1
1
0
1
1
0
1
0
0
1
3
0
17
7
17
7
17
7
05
0
(4:98)
Tanner graph
The Tanner graph [27] visually describes a block code via its parity-check matrix. Since the
parity-check matrix performs n k parity checks on n bits of the received word, one can
represent the parity-check operation by two separate sets of nodes. The left set of nodes
consists of n bit nodes (circle), one node for each bit in the code word, with no direct
connection between any two bit nodes. The right set of nodes consists of n k check nodes
(square), one node for each parity check, with no direct connection between any two check
nodes. The check nodes are connected to the bit nodes they check via branches, that is, the
check node i is connected to the bit node j if and only if hij 1. Thus, the n k rows of
H specify the check nodes connections and the n columns of H specify the bit nodes
connections. The Tanner graph is a bipartite graph whose nodes are separated into two
distinct types with connections only between nodes of different types. Figure 4.9 shows the
Tanner graph of the regular LDPC code of (4.98).
Bit nodes
Figure 4.9
Check nodes
154
The cycle in a Tanner graph is a path that closes on itself. The shortest cycle of the
Tanner graph has four branches. Cycles degrade the performance of decoding algorithms
that are based on iterative techniques. The Tanner graph in Figure 4.9 has the shortest
cycle.
Example 4.17 Conditional a-posteriori probability, likelihood ratio, and ln-likelihood ratio of
equiprobable binary random variable
The conditional a-posteriori probabilities of an equiprobable binary random variable X
given the observed vector y y1 y2 yn of another random vector Y are denoted as
PrX 0jy and PrX 1jy. The conditional likelihood ratio (LR) LX jy of X is dened
via Bayes rule, PrX ijy fYjX yjX i PrX i=fY y; i 0; 1, as
LX jy
(4:99)
PrX 0jy
PrX 1jy
(4:100)
155
PrX 0jy
PrX 1jy
n
Q
PrX 0jyk
k1
ln Q
n
PrX 1jyk
(4:101)
k1
n
X
ln LX jyk
k1
1p
p
p
L0j1
1p
(4:102)
(4:103)
(4:104)
(4:105)
156
fY jX yjx PrX x
fY y
(4:106)
The conditional Gaussian density function fY jX yjx of the received variable Y can be
expressed as follows:
2
1
2
fY jX yjx p eyx =2
2
(4:107)
fY jX yjA
PrX Ajy
eyA =2
2
LX jy
e2Ay=
(4:108)
2Ay
2
(4:109)
The messages for the check nodes are dened and calculated according to the paritycheck equation cHt 0. Each check node sums (modulo-2 addition) the bits from adjacent
bit nodes via the parity
g chtj
n
X
ci hji
i1
n
X
xi
(4:110)
i1
Prxi 0jyi
Prxi 0jyi
(4:111)
Lxi jyi
Lxi jyi 1
(4:112)
157
Consequently we have
Lxi jyi 1
Lxi jyi 1
1
tanh ln Lxi jyi
2
2 Prxi 0jyi 1
(4:113)
Furthermore, from (4.110) it is seen that the parity-check g is 0 when the number of ones in
the set of bits fxi g is even and the parity-check g is 1 when the number of ones in fxi g is odd.
If all bits xi are independent and occur with conditional probabilities Prxi 0jyi and
Prxi 1jyi , then the conditional probability Prg 0jy that the set of bits fxi g contains
an even number of ones is given by [8]
n
1 1Y
Prg 0jy
2 Prxi 0jyi 1
2 2 i1
(4:114)
The right-hand side of (4.114) can be veried via the following proof.
2
Y
2 Prxi 0jyi 1
i1
(4:115)
Similarly,
2 Prx1 x2 x3 0jy1 ; y2 ; y3 1 2 Prx1 x2 x3 0jy1 ; y2 ; y3 1
2 Prx1 x2 0jy1 ; y2 1
2 Prx3 0jy3 1
3
Y
2 Prxi 0jyi 1
i1
(4:116)
158
n
Y
2 Prxi 0jyi 1
i1
1
tanh ln Lxi jyi
2
i1
n
Y
(4:117)
Prg 0jy
Prg 1jy
1
1
n
Q
i1
n
Q
i1
tanh 12 ln Lxi jyi
(4:118)
tanh 12 ln Lxi jyi
and
2
3
tanh 12 ln Lxi jyi
6
7
i1
7
ln Lgjy ln6
n
4
1
5
Q
1 tanh 2 ln Lxi jyi
1
n
Q
(4:119)
i1
The implementation of MPA is based on the recursive calculations of (4.101) for the bit node
l
LLR message and (4.119) for the check node LLR message. Denote mij as the message sent
^ l
by the bit node i to check node j at the l th iteration, and m
ji as the message sent by the check
th
node j to bit node i at the l iteration. Note that for iteration l = 0, that is, the initialization, the
l
0
message mij is independent of all the check nodes and is denoted as mi . For example,
0
0
mi 2Ayi = 2 , i = 1,2, . . . , n for an AWGN channel and mi ln1 p ln p, i =
1,2, . . . , n for a BSC. The MPA is summarized as follows:
1. Initialization
* m
^ 0
ji 0 for all i and j.
0
(4:120)
where Bj is the set of bit nodes adjacent to check node j and Bj ni is the set of bit nodes
adjacent to check node j excluding bit node i.
Bit node update via (4.101)
X
l
0
^ l
m
(4:121)
mij mi
k2C n j ki
i
159
*
*
where Ci is the set of check nodes adjacent to bit node i and Ci n j is the set of check
nodes adjacent to bit node i excluding check node j.
(
l
0 if mij 0
For i = 1,2, . . . , n set ^ci
l
1 if mij 50
t
Stop when ^cH 0 or l lmax.
The check node message requires the calculation of a product of tanh-terms. From the
implementation point of view addition is preferable to multiplication. Therefore, it would be
advantageous to simplify the LLR of a check node. Returning to (4.118) and solving for the
product of tanh-terms we get
1
Lgjy 1
tanh ln Lxi jyi
2
Lgjy 1
i1
n
Y
eln Lgjy 1
eln Lgjy 1
1
tanh ln Lgjy
2
(4:122)
Consequently we have
ln Lgjy 2 tanh
1
n
Y
1
tanh ln Lxi jyi
2
i1
(4:123)
(4:124)
n
Y
#"
signln Lxi jyi
i1
"
n
Y
i1
n
Y
1
2 tanh1 ln1 ln
tanh jln Lxi jyi j
2
i1
#"
signln Lxi jyi
2 tanh
1
ln
1
#
#
n
X
1
ln tanh jln Lxi jyi j
2
i1
(4:125)
160
f z lntanhz=2 ln
ez 1
ez 1
ln
z
e 1
ez 1
(4:126)
e
1
ln f z
1
e
(4:127)
f f z
(4:128)
i1
Applying this result to (4.120) gives an alternative method of updating the check node
message, as follows:
i
hY
h
iih X
l1
l1
^ l
f
m
(4:129)
sign
m
f
m
ji
kj
kj
k2B ni
k2B ni
j
Syndrome decoding discussed in Section 4.5 is the rst iteration of MPA. By recursively
updating the LLRs, the decoder improves the detection probability of a code word. MPA is a
soft decoding technique and achieves a higher coding gain (about 2 dB) at low bit error
probability than predicted by (4.41) for a hard decoder.
161
c1,l
il
il1
il2
c1(x)
x2i(x)
i(x)
xi(x)
c2(x)
c2,l
Figure 4.10
The convolutional encoder can also be described by its generator polynomials that
represent the connection between the shift registers (SR) and the modulo-2 adders. For
example, the two generator polynomials for the convolutional code in Figure 4.10 are
g1 x x2 1 and g2 x x2 x 1. They represent the upper and lower connections,
respectively, where the lowest-order coefcients represent the connection to the leftmost SR
stage, and the highest-order coefcients represent the connection to the rightmost SR stage.
Given an input sequence represented by the polynomial i(x), the two output code sequences
are c1 x ixg1 x and c2 x ixg2 x, where the polynomial multiplication is carried
out in GF(2).
The number of states of the rate k/n convolutional encoder is 2v, where v is given by
P
v ki1 maxj degree gij x, j =1, 2, . . . , n, and gij x are the generator polynomials. The
constraint length is K = v + 1. The code polynomial vector is cx ixGx, where
cx c1 x c2 x cn x, ix i1 x i2 x ik x, and Gx gij x is the
k n generator polynomial matrix.
162
Figure 4.11
10/1
00/0
00
11/1
00/1
10
01
01/0
11
01/1
10/0
11/0
Figure 4.12
The convolutional encoder is a nite-state machine, and therefore can be described by a state
diagram depicting all 2v states and the corresponding transitions between states. Figure 4.12
shows the state diagram for the convolutional encoder in Figure 4.10, where the label yz/x
163
State
00
11
00
00
11
11
11
01
10
00
00
11
11
11
11
00
00
00
01
01
01
10
10
00
10
10
10
01
10
10
11
01
Figure 4.13
01
01
Hard decodingBSC
The error-correcting capability of a convolutional code is characterized by its free distance
dfree , which is the smallest Hamming distance between any two distinct code sequences.
Since convolutional codes are linear codes, the all-zero code sequence is a legitimate code
sequence. Thus, dfree is the Hamming weight (the total number of 1s) of the minimumweight path through the trellis that originates from the all-zero state and ends in the all-zero
state after a nite time. For the trellis in Figure 4.13, the minimum-weight path includes the
set of states {00, 01, 10, 00} and corresponds to the code sequence (11, 01, 11), which has
the Hamming weight of 5, hence dfree 5.
The decoding at the receiver can be performed on the trellis via the well-known Viterbi
algorithm, which efciently implements the maximum likelihood (ML) sequence decoding
scheme. In practice the rate k/n encoder takes a nite-length input sequence of kL bits and
produces a nite-length output code sequence of nL bits. The decoding is carried out on each
received code sequence independently. To implement this scheme, the encoder inserts tail
bits at the end of each input sequence. If the constraint length of the code is K, the number of
tail bits is k(K 1). The tail bits return the encoder to the initial state, which is normally the
all-zero state. This enables the decoder to reset the Viterbi algorithm to the all-zero state for
the next code sequence decoding. Hence, in practice the input sequence to the encoder can
164
be represented by a k(L + K 1)-vector i and the output code sequence by a n(L + K 1)vector c. Let us consider a discrete memoryless channel (DMC) with binary input and Q-ary
output that presents the decoder a vector r, which is the transmitted code vector c that may
contain errors in its components. Using Bayes theorem we have
Prcjr
Prrjc Prc
Prr
(4:130)
Since convolutional codes are linear, all code vectors of the same length are equally likely;
P
therefore, Pr(c) is identical for all possible values of c. Furthermore, Prr Prrjc Prc
is independent of whichever code vector was transmitted. The decoder that maximizes the
a-posteriori probability Prcjr, that is, maximizing the probability of selecting the code vector
correctly given the received vector r, is called an ML decoder. Consequently, the ML
decoder also maximizes the probability Prrjc, the likelihood of the transmitted code vector
c. Since log x is a monotonically increasing function of x, the maximization of Prrjc is
equivalent to the maximization of the log-likelihood function Mrjc log Prrjc, commonly referred to as the path metric in the Viterbi algorithm.
Qc
Mrjc log Prrjc log
Prri jci
nc
P
i1
i1
nc
P
(4:131)
Mri jci
i1
where nc is the length of the binary vectors r and c, and Mri jci log Prri jci represents
the bit metric. For a BSC we have
p; ri 6 ci
(4:132)
Prri jci
1 p; ri ci
Let d dr; c be the Hamming distance between the two binary vectors r and c, we obtain
Prrjc pd 1 pnc d and consequently
Mrjc d log p nc d log1 p
p
nc log1 p
d log
1p
(4:133)
The term nc log1 p is common for all code vectors c. Furthermore, we have
logp=1 p50 for p < 0.5. Therefore, the Viterbi algorithm chooses a path in the trellis
that maximizes Mrjc log Prrjc or, equivalently, chooses a code vector that minimizes
the Hamming distance d dr; c. This also means choosing a code vector that differs from
the received vector in the fewest places. The decoder that implements the Viterbi algorithm
for a BSC is commonly referred to as a hard decoder.
165
10
00
00
00
11
State
2, 3
00
11
11
11
01
10
Figure 4.14
2, 4
11
2, 3
2, 5
+ 3, 4
11
00
01
01
01
3, 4
10
2, 5
01
10
10
10
10
3, 4
00
11
+ 2, 4
11
00
00
10
01
00
00+
10
00
00
10
3, 4
3, 4
01
01 +
3, 4 11
Soft decodingDMC
When the components of the received vector r are voltages, such as the sample values at the
output of the matched lter in the receiver (Chapter 7), the metric can no longer be expressed
in terms of the Hamming distance. In practice, the received components of r are quantized to
Q-levels (16 to 64 levels are sufcient in practice) to form Q-ary outputs with a set of
transition probabilities Prri jjci . Thus, Prri jjci is the probability that the output
166
Table 4.2 The bit metric M ri jjci log Prri jjci for a 4-ary channel
M ri jj0
M ri jj1
ri(0) = 00
ri(1) = 01
ri(2) = 11
ri(3) = 10
0.4
1
0.5
0.7
0.7
0.5
1
0.4
Table 4.3 The positive bit metric M ri jjci 17:3 17:3 log Prri jjci for a 4-ary channel
M ri jj0
M ri jj1
ri(0) = 00
ri(1) = 01
ri(2) = 11
ri(3) = 10
10
0
8
5
5
8
0
10
Table 4.4 The uniform branch metric M ri jj1 j and M ri jj0 3 j, 0 j 3, for a
4-ary channel
M ri jj0
M ri jj1
ri(0) = 00
ri(1) = 01
ri(2) = 11
ri(3) = 10
3
0
2
1
1
2
0
3
voltage of the ith bit falls in the jth quantized level given the transmitted coded bit ci . For
example, with Q = 4, we have the following assignments: ri 0 00 ; ri 1 01 ;
ri 2 11 ; ri 3 10 . The quantized levels 00 ; 01 ; 11 ; and 10 refer to strong zero, weak
zero, weak one, and strong one, respectively. The transition probabilities depend on the
knowledge of the received signal-to-noise ratio. A common practice is to set the branch or
bit metric in some convenient positive range without affecting the performance of the Viterbi
algorithm by using the following expression:
M ri jjci A B log Prri jjci
(4:134)
where the constants A and B are chosen accordingly. Table 4.2 shows the branch metrics
M ri jjci log Prri jjci for 4-ary channels and Table 4.3 shows the corresponding
positive branch metrics. The knowledge of the transition probabilities is required for the
computation of the metric. Furthermore, as the signal-to-noise varies in a wireless fading
channel, the metrics must be adjusted accordingly. Therefore, a xed set of metrics that can
be easily specied and represents a good compromise over a large range of signal-to-noise
ratios is desirable. For Q-level quantization, the branch metrics are M ri jj1 j and
M ri jj0 Q 1 j, 0 j Q 1. Table 4.4 shows the uniform branch metric
assignment. Simulation shows a small degradation in performance over a broad range of
signal-to-noise ratios. The decoder that implements the Viterbi algorithm for a DMC is
commonly referred to as soft decoder.
167
0 1 00
1 0 01
00
0 0 00
1 1 00
00
11
11
11
10
10
10
+
4,13
01
Figure 4.15
01
8, 9
+ 17, 20
01
10
11
00
10
11, 14 10
+ 12, 13
01
00
00
01
01
14, 23
+ 11, 19 11
11
00
11
+ 9,10
11
13, 17
8, 11
State
00
00
00
0 0 00
01
16, 22 10
10
01
16, 22
11
fR rjc Prc
fR r
(4:135)
168
The conditional Gaussian density function fR rjc of the received vector r can be expressed
as follows:
Y
2
1
2
p erk sk =2
2
k
2
1
2
p n ekrsk =2
2
fR rjc
(4:136)
Since convolutional codes are linear, all code vectors of the same length are equally likely;
P
therefore, Pr(c) is identical for all possible values of c. Furthermore, fR r fR rjc Prc
is independent of whichever code vector was transmitted. The decoder that maximizes the
a-posteriori probability Prcjr, that is, maximizes the probability of selecting the transmitted
code vector correctly given the received vector r, is the ML decoder. Consequently, the ML
decoder also maximizes the likelihood fR rjc of the transmitted code vector c. Since log x is
a monotonically increasing function of x, the maximization of fR rjc is equivalent to the
maximization
of the ln-likelihood function Mrjc ln fR rjc kr sk2 =2 2
p
n ln 2 , or the path metric in the Viterbi algorithm. This in turn is equivalent to
minimizing the squared Euclidean distance d 2 kr sk2 (or simply the Euclidean
distance) between the received vector r and the voltage vector s that represents the transmitted code vector c. Therefore, the Viterbi algorithm chooses a path in the trellis that
maximizes Mrjc ln fR rjc, or equivalently, chooses a code vector that has the smallest
squared Euclidean distance to the received vector. The decoder that implements the Viterbi
algorithm for a Gaussian channel is referred to as the unquantized soft decoder. If the
received Gaussian vector r is quantized to Q levels, then the Gaussian channel becomes
a DMC.
169
4.8 Summary
1, 1
00
0.5, 1
0.5, 1
1, 1
00
00
00
10.3
11
11
11
+ 6.5
11
+
00
01
01
01
10
6.5
10
12.5
01
10.5 10
12.5
10.5
10
12.5
10
10
6.3
Figure 4.16
00
10
00
00 +
8.3
01
6.5
11
8.5
01
10
6.5
11
11
State
00
6.5
43
11
1, 1
01
01
10.5
11
Convolutional codes with large constraint length are very powerful. Furthermore, the
concatenated outer RS code-inner convolutional code (adopted by IEEE 802.16e) can
provide a very large coding gain that may come within 2 dB of the Shannon limit. This is
perhaps the reason why LDPC codes were forgotten for three decades.
4.8 Summary
We have provided an introductory study of information theory pioneered by Shannon that centers on source
coding and channel coding. We studied the Shannon noiseless source coding theorem and the famous
Huffman code, which is the optimum prex code (no other prex codes have smaller average code word
lengths). We stated without proof the Shannon noisy channel coding theorem and derived the information
capacities of both discrete and Gaussian channels. For Gaussian channels, we considered those with no constraint
on bandwidth and inputs, those with bandwidth constraints, and those with input constraints, especially
equiprobable discrete inputs. The channel coding is carried out by error-correcting codes such as block codes
and convolutional codes. The latter are widely used in IEEE 802.11 and 802.16 standards and in 2G and 3G
cellular communications.
170
Problems
1.
2.
3.
2
1
1.5
0.5
0.5
1.5
1
2
Figure 4.17
Quantizer
4.
5.
6.
7.
8.
9.
10.
171
Problems
0
0
0
1
1
Figure 4.18
13.
14.
The code {0, 11, 110, 111} is not a prex code. Does it satisfy the Kraft inequality?
Show that the prex codes have the binary tree shown in Figure 4.18.
(a) If we let l 4 max lm , how many nodes exist at level l?
(b) When a code word of length lm is chosen, how many nodes on level l are blocked
off?
(c) Show that a prex code satises the Kraft inequality.
The Kraft inequality determines whether there is a corresponding prex code for a
given set of code word lengths. Find a non-prex code whose code word lengths [1, 2,
3, 3] satisfy the Kraft inequality.
In this problem we wish to show that the Kraft inequality applies not just to prex
codes but to all uniquely decodable codes. Show the inequality
M
X
1
2lm
m1
15.
16.
17.
!n
nl
where n is an arbitrary positive integer and l is the longest code word length. From this
inequality show that Kraft inequality applies to all uniquely decodable codes.
Given a discrete source X with distribution pX xi , show that a prex code for this
source whose average code word length is minimum has the following properties:
(a) If pX xi 4pX xj , then li lj .
(b) Two symbols with smallest probabilities have code words of equal length.
(c) If there are two or more code words of equal length, then two of these code words
agree in all places except the last place.
Construct a Huffman code for the random variable X with the distribution {1/16, 3/16,
3/4}. Compare the average code word length to the entropy of X.
Construct a Huffman code for the random variable X with the distribution {1/32,
1/32, 1/16, 1/8, 3/16, 3/16, 3/8}. Compare the average code word length to the
entropy of X.
172
1p
p
E
p
Figure 4.19
1p
19.
20.
21.
22.
23.
24.
25.
26.
27.
28.
29.
30.
31.
Consider a discrete source of three symbols fA; B; Cg with the distribution {1/16,
3/16, 3/4}. Construct a Huffman code for the extended source
fAA; AB; AC; BA; BB; BC; CA; CB; CCg. Compare the code rate to the entropy of
the original source. Calculate the variance of the code.
Show that the entropy of the extended source is n times the entropy of the original
source as described in (4.21).
Consider a binary erasure channel (BEC) with two equiprobable inputs 0 and 1 and
three outputs 0, 1, and E, where a fraction p of the bits is erased rather than corrupted.
The receiver is supposed to know which bits are erased. The BEC is shown in
Figure 4.19. Find the channel capacity.
Verify the BSC capacity in (4.36).
Find the mutual information I(0,V) and I(1,V) of a BSC. Calculate the channel
capacity using these results.
Find the mutual information I(0,V) and I(1,V) of the BEC in Problem 20. Calculate the
channel capacity using these results.
Consider a BSC with p 0:5, that is, C = 0. Show that there are no repetition codes
that can meet a specied bit error rate less than 0.5.
Find the channel capacity of a series connection of n identical BSCs.
Consider a BSC with error probability p = 0.14.
(a) What is the BSC capacity?
(b) Is there a code that can be used to achieve a specied error rate? What is the code
rate?
(c) Assume that Hamming code is used to achieve a bit error probability equal to 0.1.
Can the goal be achieved? Provide two solutions.
Find the differential entropy of the random variable Y = X + c, where X is an arbitrary
continuous random variable and c is a real constant.
Find the differential entropy of Y = cX, where X is an arbitrary continuous random
variable and c is a real constant.
Consider a random variable X uniformly distributed over (a, b). Find the differential
entropy of X. Can the differential entropy be negative?
Derive (4.54) using the inequality ln z z 1 instead of the Jansen inequality.
Consider the water-lling strategy for a vector Gaussian channel with four subchannels. The noise variances of the subchannels are 21 2; 22 4; 23 1; 24 8.
The total power for allocating to the subchannels is nine power units. Calculate the
power allocated to each subchannel.
173
Problems
32.
33.
34.
35.
Consider the water-lling strategy for a vector Gaussian channel with ve subchannels. The noise variances of the subchannels are 21 1; 22 5; 23 3;
24 9; 25 8. The total power for allocating to the subchannels is 20 power units.
(a) Calculate the power allocated to each subchannel.
(b) Suppose the total power is increased to 30 power units, calculate the power
allocated to each subchannel.
Consider the conversion of a Gaussian channel with PSK signaling into a BSC
channel with a crossover probability of p 106 .
(a) Calculate the BSC capacity.
(b) Calculate the required SNR for a Gaussian channel with Gaussian inputs to have
the same capacity as the BSC.
(c) Calculate the potential coding gain.
Consider a (8,4) block code whose generator matrix is given by
2
3
1 0 0 0 0 1 1 1
60 1 0 0 1 1 1 07
7
G6
40 0 1 0 1 1 0 15
0 0 0 1 1 0 1 1
(a) Find the code word for the input sequence (1011).
(b) Is the word (10111101) a valid code word?
(c) Use the syndrome decoding to decode the received word (11100101) with one
error in it.
Consider a sphere of radius t around a code word c of length n and rate r = k/n. This
sphere contains the set of all code words of Hamming distance less than or equal to t
from the code word c.
(a) Calculate the number of code words in the sphere.
(b) Calculate the number of code words in all legitimate spheres of radius t.
(c) Show the following bound:
"
#
t
X
1
n
1 r log2
i
n
i0
36.
37.
38.
Verify (4.99).
Consider two independent binary random variables x1 and x2 with corresponding
probabilities Prx1 0 p1 , Prx1 1 1 p1 , Prx2 0 p2 , and
Prx2 1 1 p2 .
(a) Evaluate the probabilities of x1 x2 .
(b) Evaluate the LLR of x1 x2 in terms of the LLRs of x1 and x2 .
(c) Generalize the result to n independent binary random variables.
Plot the function f z lntanhz=2 and show that the smallest value of z
dominates. This means that the check node LLR is dominated by a bit that is least
certain, that is, likely to be zero or one. In this case the parity is also likely to be either
zero or one irrespective of other bits. Modify the check node LLR to reect this
observation for less complex calculation.
174
State
00
11
00
00
11
11
11
01
10
00
11
11
11
11
00
00
00
00
01
01
01
10
10
00
10
10
10
01
10
10
11
01
Figure 4.20
01
01
39.
40.
41.
42.
43.
44.
45.
46.
Implement the MPA for the Hamming code (7,4) and plot the bit error probability
versus Eb =N0 (dB).
Consider a rate 2=3 convolutional code with the following generator matrix:
1x 1x 1
Gx
0
x
1x
(a) Design the encoder.
(b) Calculate the constraint length.
Consider the convolutional encoder specied by the generator polynomials
g1 x 1 x3 and g2 x x x2 x3 .
(a) Draw the encoder.
(b) For the input sequence (10111), add the smallest number of tail bits and nd the
output sequence.
(c) Draw the state diagram.
Perform the Viterbi algorithm for the convolutional code with the trellis shown in
Figure 4.20, assuming the received vector r 0101011111.
Perform the Viterbi algorithm for the convolutional code with the trellis shown in
Figure 4.20, assuming the received vector r 1000100001.
Perform the Viterbi algorithm for the convolutional code with the trellis shown in
Figure 4.20, assuming the received vector r 01 01 10 00 00 11 00 11 11 00 .
Perform the Viterbi algorithm for the convolutional code with the trellis shown in
Figure 4.20, assuming the received vector r 11 01 10 00 11 11 00 10 11 01 .
Perform the Viterbi algorithm for the convolutional code with the trellis shown in
Figure 4.20, assuming the received vector r 0:8; 0:6;1; 0:7; 0:7; 0:6; 0:5;
0:9; 0:8; 0:6.
175
Bibliography
47.
Perform the Viterbi algorithm for the convolutional code with the trellis shown in
Figure 4.20, assuming the received vector r 0:7; 0:9;1; 0:7; 0:7; 0:6; 0:5;
0:9; 0:8; 0:6
Further reading
The foundation of information theory was laid out by Shannon in [1,2]. For an introduction
to information theory and coding we recommend Hamming [6,9]. The serious reader can
explore [4,1016] for advanced materials.
Bibliography
1. C. E. Shannon, A mathematical theory of communication, Bell Syst. Tech. J., Vol. 27,
pp. 379423, pp. 623656, 1948.
2. C. E. Shannon, Communication in the presence of noise, Proc. IRE, Vol. 37,
pp. 1021, 1949.
3. R. V. L. Hartley, The transmission of information, Bell Syst. Tech. J., Vol. 7, pp. 535
563, 1928.
4. S. Lin and D. J. Costello, Jr., Error Control Coding, 2nd edition, Englewood Cliffs, NJ:
Pearson Prentice-Hall, 2004.
5. G. C. Clark, Jr. and J. B. Cain, Error-Correction Coding for Digital Communications,
New York: Plenum Press, 1981.
6. R. W. Hamming, Error detecting and error correcting codes, Bell Syst. Tech. J., Vol.
29, pp. 147160, 1950.
7. R. Gallager, Low-density parity-check codes, IRE Trans. Inform. Theory, pp. 2128,
January 1962.
8. R. Gallager, Low-Density Parity-Check Codes, Cambridge, MA: MIT Press, 1963.
9. R. W. Hamming, Coding and Information Theory, 2nd edition, London: Prentice-Hall,
1986.
10. N. Abramson, Information Theory and Coding, New York: McGraw-Hill, 1963.
11. R. G. Gallager, Information Theory and Reliable Communication, New York: Wiley,
1968.
12. R. E. Blahut, Principles and Practice of Information Theory, Reading, MA: AddisonWesley, 1987.
13. T. M. Cover and J. A. Thomas, Elements of Information Theory, 2nd edition,
Chichester: John Wiley & Sons, 2006.
14. J. R. Barry, E. A. Lee, and D. G. Messerschmitt, Digital Communication, 3rd edition,
Amsterdam: Kluwer Academic Publishers, 2004.
15. R. G. Gallager, Principles of Digital Communications, Cambridge: Cambridge
University Press, 2008.
176
Introduction
In this chapter, we provide the foundation for analyzing a wireless communication link. The
purpose is to evaluate the signal-to-noise ratio at the receiver to assess the link performance.
Evaluation of the signal power and noise power requires the path loss and receiver system
noise temperature, respectively. The receiver consists of an antenna, a low-noise amplier, a
downconverter, and a demodulator. The concept of the thermal noise source and its noise
temperature, as well as the antenna noise temperature, is discussed. We also introduce the
effective noise temperature and noise gure of a two-port network such as the low-noise
amplier, downconverter, or demodulator. The effective noise temperature and noise gure
of a cascade or series connection of two-port networks are derived. This leads to the
evaluation of the receiver system noise temperature.
For free space links, such as satellite communications links, we introduce the Friis
equation to calculate the path loss. For cellular communications links, we present the
well-known Hata model. Many important aspects of cellular systems are also discussed,
such as the frequency spectrum, standards, and the co-channel interference.
178
Transmitted
information
Modulator
Upconverter
High-power
amplifier
Antenna
Channel
Received
information
Figure 5.1
Demodulator
Downconverter
Antenna
Noisy
resistor
Figure 5.2
Low-noise
amplifier
Noiseless
resistor
Noise
source
~ v(t)
noise power generated by the transmitters subsystems; therefore, the effect of the thermal
noise is negligible. The signal is attenuated as it travels through the channel. When the signal
arrives at the receiver, its power can diminish to a level so that the effect of noise becomes
relevant. The receiver antenna receives the signal contaminated by sky noise. At each
subsystem of the receiver, the signal is processed and thermal noise is added to the signal.
The purpose of link analysis is to obtain the signal-to-noise ratio at the receiver that can
provide a qualitative assessment of the performance of the entire communication system.
Two quantities need to be evaluated, namely, the receiver noise power and the received
signal power. We rst study the effect of thermal noise in the receiver.
Thermal noise
Thermal noise is generated by random motion of electrons in a resistor whose temperature is
above absolute zero kelvin. A noise voltage is generated across the terminal of the resistor,
which can be modeled as a noiseless resistor with resistance R ohms in series with a noise
voltage source v(t) as in Figure 5.2.
The power spectral density of the thermal noise source v(t), which is normalized to
R = 1 , is given in W/Hz as follows:
Sv f 2
hj f j
hj f j
hj f j=kT 1 W=Hz
n
2
e
(5:1)
179
R
R
v(t)
Figure 5.3
where h 6:6 1034 Js is the Planck constant, k 1:38 1023 J/K is the Boltzmann
constant, Tn is the physical temperature of the resistor (also referred to as the noise temperature of the source) in kelvin, and f is the frequency in Hz.
For frequency f 51000 GHz, the power spectral density of a thermal noise source can be
very well approximated by the following constant:
Sv f 2kTn
(5:2)
Noise with a constant power spectral density is called white noise. In a communication
system, all subsystems are designed to have matched impedances at both input and output.
For example, the output impedance of the antenna is matched with the input impedance of
the low-noise amplier. In practice, these input and output impedances are 50 . In our
discussion, we assume a normalized impedance of R = 1 . Matched impedances are
employed to provide the maximum power transfer from one subsystem to the next one.
Unfortunately, this also allows the maximum power transfer of noise.
When a matched load is connected across the thermal noise source as shown in Figure 5.3,
the available power spectral density in W/Hz across the load is
S f j H f j 2 Sv f
(5:3)
where H(f ) is the transfer function of the resistor divider network and is given by
H f
R
1
RR 2
(5:4)
kTn
2
(5:5)
Note that the available power spectral density of a thermal noise source is independent of the
value of R.
The available noise power, in watts, delivered by the thermal noise source v(t) to the
matched load in a bandwidth of B Hz is therefore given by
N
B
B
S f df
kTn
df kBTn
B 2
(5:6)
180
S( f )
kTn/2
N = kBTn
B
Figure 5.4
kTn
2 1021 W=Hz
2
Assuming that the operating bandwidth is B = 1 MHz, the available noise power is
N kBTn 4 1015 W
In summary, a thermal noise source is specied by its noise temperature Tn, which is also
its physical temperature. Returning to Figure 5.1, we observe that the receiver antenna
receives the signal plus sky noise. The term sky noise is used to include noise signals from
emittingabsorbing objects in space. High-temperature-emitting objects such as the stars
and the Sun produce electromagnetic radiation with a white noise power spectral density in
the practical frequency range that wireless communications operate in. The antenna also
receives noise signals from absorbing objects, such as the Earth and the Moon. A blackbody
that absorbs electromagnetic radiation also acts as a resistor above absolute zero kelvin and
hence also radiates noise. The composite noise signal received by the receiver antenna can
be modeled as a white noise source of temperature TA called the antenna noise temperature.
This equivalent noise source delivers an available noise power of kTAB. In other words, the
noise that the antenna receives can be taken into account by assigning a noise temperature
TA to the antenna radiation resistance. The antenna noise temperature is a function of
frequency and antenna beam angle, and includes contributions from sources of radiation
in the main beam as well as sources of radiation in all directions in proportion to the antenna
pattern.
181
Resistor
at TA
Receiver
antenna
Noiseless
resistor
Noise
source
Sky temperature
TA
(a)
Figure 5.5
(b)
(a) Antenna with entire pattern sees the sky at temperature TA; (b) resistor at temperature TA. The same noise power
spectral density is available at the output terminals in both cases.
Sky contribution :
Ground contribution :
0:9 10 K 9 K
0:1 290 K 29 K
Thus,
TA 9 K 29 K 38 K
The noise power spectral density available at the antenna output is given by (5.5) as
S f
kTA
2:6 1022 W=Hz
2
In summary, the noise that the antenna receives can be taken into account by assigning a
noise temperature TA to the antenna radiation resistance. Thus, if the antenna is replaced by a
resistor of the same radiation resistance at temperature TA, the noise power spectral density
available at the output terminals is unchanged, and so is the noise power available within a
bandwidth B. Figure 5.5 illustrates the modeling of the noise received by the receiver
antenna.
Returning to Figure 5.1, the signal and sky noise received by the receiver antenna are
delivered to the low-noise amplier. The low-noise amplier amplies both signal and sky
noise and also adds its own thermal noise to the signal. More thermal noise is added as the
signal is downconverted from the carrier frequency to an intermediate frequency by the
downconverter and, nally, the demodulator adds more noise as it detects the information
from the modulated signal.
We assume that the gains of both the low-noise amplier and downconverter are constant
over the operating frequency range. This is the case in practice to avoid frequency-selective
distortion of the signal spectrum, which in return can cause intersymbol interference and/or
loss of the signal-to-noise ratio.
182
(R, Tn)
Noisy
two-port
gain = G
(R, Tn + Te)
(a)
Figure 5.6
Noiseless
two-port
gain = G
(b)
(a) Noisy two-port network with input noise source at temperature Tn ; (b) model with the effect of noise temperature
Te added to noise source temperature.
as shown in Figure 5.6(a). The input and output of the network are matched to R ohms. The
network has a constant gain G and is driven by a thermal noise source at temperature Tn.
The noise power available in a bandwidth B at the input of the network is kBTn as indicated
by (5.6). The noise power available in B at the output of the network due to the input noise
source is simply kBGTn. Let NI denote the noise power available in the bandwidth B at the
output port due to the internal noise sources of the network. Then, the total available noise
power at the output port is given by
N kBGTn NI
NI
kBG Tn
kBG
(5:7)
kBGTn Te
where
Te
NI
kBG
(5:8)
is dened as the effective noise temperature of the two-port network and is incorporated in
the model shown in Figure 5.6(b). In this model, the noisy two-port network is replaced by a
noiseless two-port network with the same gain G, and the input noise source temperature is
changed from Tn to Tn + Te. Note that the noise power available at the input of the noiseless
two-port network is kB(Tn + Te), as compared to kBTn at the input of the noisy two-port
network. Nevertheless, they both yield the same available output noise power given in (5.7).
In summary, the effective noise temperature of a noisy two-port network is the additional
temperature that a noise source requires to produce the same available noise power at the
output of the equivalent noiseless two-port network (the noisy two-port network with the
internal noise removed).
Denition 5.1 The system noise temperature of a noisy two-port network is dened by
Ts Tn Te
(5:9)
(5:10)
183
(R, Ts)
s(t) with
power Pi
Figure 5.7
SNR =
Pi/kBTs
Noiseless
two-port
network
System signal-to-noise ratio where the input signal s(t) has power Pi, and the system noise n(t) has power kBTs.
Denition 5.3 The available output noise power of a two-port network is dened by
N kBGTs
(5:11)
Denition 5.4 The system signal-to-noise ratio of a noisy two-port network of effective
noise temperature Te, driven by an input noise source of temperature Tn and an input signal
source of power Pi , is dened by
SNR
Pi
Pi
Ni kBTs
(5:12)
(5:13)
It is seen that the noise gure is dened for a reference noise source temperature T0 . The
reference room temperature adopted by the IEEE is T0 = 290 K, which yields
F 1
Te
290
(5:14)
Therefore, the effective noise temperature in kelvin of the noisy two-port network can be
related to its noise gure as
Te 290F 1
(5:15)
184
120
1:41 1:5 dB
290
1012
402:6 26 dB
1:38 1023 106 180
Returning to Figure 5.1, we note that the antenna output might be connected to the lownoise amplier via a transmission line, such as a waveguide. To account for the loss of the
transmission line, we return to Figure 5.6 and consider the transmission line as a reciprocal
two-port network with loss L, or equivalently with gain
G
1
L
(5:16)
Let the characteristic impedance of the transmission line be R ohms, so that both input and
output are matched to R ohms. Also, let T0 be the physical temperature of the transmission
line. Thus, looking into the output of the transmission line, the load sees a resistor R at
temperature T0 . Therefore, the available output noise power at the load in a bandwidth of B
Hz is kBT0 . But this is also equal to the quantity on the right-hand side of (5.7), that is,
kBGT0 Te kBT0
(5:17)
This yields the effective noise temperature of the transmission line as follows:
Te T0 L 1
(5:18)
Te 290L 1
(5:19)
185
Comparing (5.15) and (5.19), we obtain the noise gure of the transmission line of loss L as
FL
(5:20)
From the link analysis point of view, the effective noise temperature of the transmission line
can be evaluated by (5.18) knowing its loss L and its physical temperature T0 . Note that the
loss L = 1 or gain G =1 corresponds to a lossless transmission line with Te = 0.
Returning to Figure 5.1, we also need to account for the internal noise generated by the
downconverter and the demodulator. These two systems are connected in series with the
low-noise amplier. To account for their noise effects, we consider the model shown in
Figure 5.8, which consists of two noisy two-port networks. Each two-port network has an
effective noise temperature Tek and a gain Gk, k = 1, 2. The input noise source has a
temperature Tn. We assume that the two-port networks are matched to R ohms at their inputs
and outputs. Our purpose is to evaluate the effective noise temperature of the series
connection, which obviously has a gain G1G2.
We note that by using (5.9) and (5.11), we obtain the available noise power at the output of
the rst two-port network as
N1 kBG1 Tn Te1
(R, Tn)
Noisy
two-port
network
Te2, G2
Noisy
two-port
network
Te1, G1
Te , G
Figure 5.8
(5:21)
186
The available noise power N at the output of the second two-port network consists of G2N1
and the power delivered by the internal noise sources of the second two-port network. Using
(5.8) we have
N G2 N1 kBG2 Te2
kBG1 G2 Tn Te1 kBG2 Te2
Te2
kBG1 G2 Tn Te1
G1
(5:22)
From (5.9) and (5.11), we conclude that the effective noise temperature Te of the series
connection is Te1 Te2 =G1 . Now, if we connect a third two-port network with effective
noise temperature Te3 and gain G3 in series with the second two-port network, we end
up with a series connection of two networks, one with Te1 Te2 =G1 and G1G2 and one
with Te3 and G3. Applying (5.22) with Te1 replaced by Te1 Te2 =G1, where Te2 is
replaced by Te3 and G1 is replaced by G1G2, the effective noise temperature of the
series connection of three two-port networks is Te1 Te2 =G1 Te3 =G1 G2 and its gain is
G1G2G3. Continuing this way, we can generalize the result to a series connection of
n two-port networks with gain G = G1G2 . . . Gn and effective noise temperature
Te given by
Te Te1
Te2
Te3
Ten
G1 G1 G2
G1 G2 Gn1
(5:23)
Applying (5.15) to (5.23) yields the noise gure F of a series connection of n two-port
networks, each with noise gure Fi, i = 1, 2, . . . , n:
F F1
F2 1 F3 1
Fn 1
G1
G1 G2
G1 G2 Gn1
(5:24)
Observing (5.23), we come to the following conclusion. If the gain G1 of the rst two-port
network is large, then the noise contribution of subsequent two-port networks becomes
negligible.
2 104
105
105
105 102
187
103
103 102
120 20 1 141 K
Te 120
Example 5.6 System noise temperature at the input of the transmission line
To evaluate the system noise temperature at the input of the transmission line, we employ (5.9)
and the model shown in Figure 5.6. We note that Tn = TA. The effective noise temperature Te of
the series connection of the transmission line, the low-noise amplier, the downconverter, and
the demodulator is calculated via (5.18) and (5.23) and is given as follows:
Te T0 LTL 1 LTL TLNA
LTL TDC
LTL TDM
GLNA
GLNA GDC
(5:25)
(5:26)
LTL TDC
LTL TDM
GLNA
GLNA GDC
(5:27)
(R, TA)
Figure 5.9
Transmission
line
Low-noise
amplifier
Downconverter
Demodulator
LTL , T0
GLNA, TLNA
GDC , TDC
TDM
188
Example 5.7 System noise temperature at the input of the low-noise amplier
We can model the antenna and the transmission line as a combined noise source at temperature Tn0 . First we evaluate the available noise power N 0 at the output of the transmission line
via (5.7) and (5.18):
N0
kB
TA T0 LTL 1
LTL
(5:28)
1
TA T0 LTL 1
LTL
(5:29)
The effective noise temperature Te0 of the series connection of the low-noise amplier, the
downconverter, and the demodulator is calculated via (5.23) as
Te0 TLNA
TDC
TDM
(5:30)
TA T0 LTL 1 TLNA
(5:31)
Ts
LTL
(5:32)
kBGLNA
TA T0 LTL 1 LTL TLNA
LTL
(5:33)
GLNA
TA T0 LTL 1 LTL TLNA
LTL
(5:34)
189
The effective noise temperature T00e of the series connection of the downconverter and the
demodulator can be calculated via (5.23) as
Te00 TDC
TDM
GDC
(5:35)
(5:36)
GLNA Ts
LTL
(5:37)
kBGLNA GDC
LTL TDC
N 000
TA T0 LTL 1 LTL TLNA
(5:38)
LTL
GLNA
Setting N 000 kBTn000 to obtain the effective noise temperature of the combined noise source,
we get
GLNA GDC
LTL TDC
000
TA T0 LTL 1 LTL TLNA
Tn
(5:39)
LTL
GLNA
Using (5.9), we obtain the system noise temperature as follows:
Ts000 Tn000 TDM
GLNA GDC
LTL TDC
(5:40)
GLNA GDC Ts
LTL
(5:41)
190
LTL PR00
LTL PR000
PT GT GR
GLNA
GLNA GDC
LC
(5:42)
Using (5.42), (5.32), (5.37), and (5.41), we obtain the system signal-to-noise ratio (SNR) as
follows:
SNR
PR
P0
P00
P000
R 0 R 00 R 000
kBTs kBTs
kBTs
kBTs
(5:43)
It is seen that the system signal-to-noise ratio is invariant with respect to any reference input
and is given by
SNR
PT GT GR
LC kBTs
(5:44)
where Ts is given in (5.27). The product PTGT is called the effective isotropic radiated power
(EIRP) of the transmitter, and the ratio GR/Ts is called the receiver gure-of-merit.
Te
104
200 120 4 321 K
GLNA
10
The gain of the low-noise amplier is large; therefore, the contribution of noise from the rest
of the receiver is only 1 K. From (5.44), we obtain the channel loss LC as
191
EIRPGR
kBTs SNR
103 102
2:26 1016
23
1:38 10 107 321 102
LC
(5:45)
where d is the distance between the transmitter and receiver and is the signal wavelength
given by
l
c 3 108 m=s
f
f
(5:46)
192
MSC
Figure 5.10
PSTN
990
991 1023
Forward channel
779
824849 MHz
Figure 5.11
779
869894 MHz
Frequency spectrum
In the United States, the Federal Communications Commission (FCC) allocated a frequency
spectrum of 50 MHz to cellular communication in the 800 MHz band as shown in
Figure 5.11. The frequency spectrum consists of two bands. The forward channel band of
25 MHz (869894 MHz) is used for base station-to-mobile communication. The reverse
channel band of 25 MHz (824849 MHz) is used for mobile-to-base station communication. Each band consists of 832 channels (channel 990 is not used) and each channel has a
bandwidth of 30 kHz. Two channels of the same number, one in each band, form a 60 kHz
full duplex channel, which allows simultaneous two-way communication between the
base station and the mobile. To encourage competition, the Federal Communications
Commission licensed the frequency spectrum to two competing service providers in each
geographical area. Block A channels were licensed to a non-wireline company, which does
not provide a telephone service. Block B channels were licensed to a wireline company that
provides a telephone service. Table 5.1 shows the channel allocation for both providers A
and B, and Table 5.2 shows the center frequency of the channel.
193
Bandwidth (MHz)
Number of channels
Not used
A
B
Channel number
990
1
10
1.5
33
333
50
991 to 1023
1 to 333
667 to 716
10
2.5
333
83
334 to 666
717 to 799
Channel number
1 N 799
990 N 1023
1 N 799
990 N 1023
0.03 N + 825
0.03 (N 1023) + 825
0.03 N + 870
0.03 (N 1023) + 870
(AMPS), established in 1983, and the narrowband advanced mobile phone system
(NAMPS), established in 1992 [1]. Each AMPS provider has 416 duplex channels (416
reverse channels and 416 forward channels). Each FM channel (reverse or forward) has a
channel bandwidth of 30 kHz and a peak frequency deviation of 12 kHz. Out of 416
channels, there are 395 voice channels and 21 control channels (channels 313 to 333 for
provider A and channels 334 to 354 for provider B).
As the number of mobile users increased, more channels were needed to meet the
demand. NAMPS was designed to provide three times as many channels as AMPS and at
the same time be backward compatible with it. An FM channel of NAMPS (reverse or
forward) has a bandwidth of 10 kHz and a peak frequency deviation of 5 kHz. Thus, a
provider can gradually replace an AMPS channel with three NAMPS channels to phase in a
new system. The frequency spectrum of 50 MHz can accommodate 2496 duplex NAMPS
channels. Both AMPS and NAMPS employ frequency division multiple access (FDMA) to
provide duplex channels to mobiles. In FDMA, the frequency spectrum is divided into
channels that occupy distinct frequency bands. Users are assigned separate channels for
access to the frequency spectrum.
The rst generation analog cellular system in Europe was deployed in 1985 and was
named the extended total access cellular system (ETACS). ETACS employs a frequency
spectrum of 62 MHz (917933 MHz and 935960 MHz for forward channels; 872
888 MHz and 890915 MHz for reverse channels), which can accommodate 1000 fullduplex FM channels. Each FM channel (reverse or forward) has a bandwidth of 25 kHz and
a peak frequency deviation of 9.5 kHz.
The Japanese total access communication system (JTACS) was deployed in 1988 in the
800 MHz band. JTACS provides full-duplex FM channels. Each channel has a 25 kHz
194
40 ms or 1944 bits
Slot 1
Figure 5.12
Slot 2
Slot 3
Slot 4
Slot 5
Slot 6
bandwidth with a peak frequency deviation of 9.5 kHz. In 1993, Japan introduced narrowband total access communication systems (NTACS) to double the capacity of JTACS. An
NTACS FM channel has a bandwidth of 12.5 kHz and a peak frequency deviation of 5 kHz.
The total number of duplex channels is 1040. Both JTACS and ETACS employ FDMA to
provide access for mobiles.
The second generation (2G) cellular system was digital and was developed to provide a
larger capacity than the rst generation systems. Two major digital standards were deployed
in the United States: the Telecommunication Industry Association/Electronic Industry
Association (TIA/EIA) Interim Standard IS-54 [2], later became IS-136, also known as
US digital cellular (USDC) in 1990, and the TIA/EIA Interim Standard IS-95 in 1993 [3].
IS-136 operates in the same frequency spectrum as AMPS. Also, each IS-136 channel has a
bandwidth of 30 kHz, but can support three mobiles using full-rate speech coding of
7.95 kbps or six mobiles using half-rate via time division multiple access (TDMA), as
shown in Figure 5.12.
The base station or the mobile transmits a /4-DQPSK signal, which carries data at a
TDMA burst rate of 48.6 kbps. The /4-DQPSK signal occupies a bandwidth of 30 kHz (one
AMPS forward or reverse channel). When full-rate speech (high quality voice) is employed,
the channel can accommodate three mobiles simultaneously in each TDMA frame as
follows: time slots 1 and 4 for mobile 1, time slots 2 and 5 for mobile 2, and time slots 3
and 6 for mobile 3. For half-rate speech, the channel can accommodate six mobiles
simultaneously with one time slot for each mobile per TDMA frame. By allowing mobiles
to access the 30 kHz channel sequentially on a time division basis, IS-136 can provide three
times the capacity of AMPS with full-rate speech or six times with half-rate speech. Note that
a 30 kHz AMPS channel allows access to only one user. Since IS-136 employs the same
frequency reuse plan as AMPS, the backward compatibility with AMPS allows providers to
gradually replace one AMPS channel by three or six IS-136 channels to meet the growing
demand for capacity. IS-136 can provide 2496 full duplex TDMA channels with full-rate
speech or 4992 full-duplex TDMA channels with half-rate speech in a bandwidth of
50 MHz.
AMPS, NAMPS, and IS-136 employ a frequency reuse plan of seven cells per cluster.
Therefore, each AMPS cell has about 416/7 59 full-duplex channels, while an NAMPS
cell can have 178 full-duplex channels and an IS-136 cell can have 178 full-duplex TDMA
channels for full-rate speech or 356 full-duplex TDMA channels for half-rate speech.
In 1993, the IS-95 system was deployed as a competing digital cellular system. IS-95
employs code division multiple access (CDMA) to provide mobile access to a spread
spectrum channel of 1.25 MHz. All mobiles in a spread spectrum channel employ distinct
quasi-orthogonal pseudo-noise (PN) codes to separate themselves from each other and to
195
Figure 5.13
Slot 1
Slot 2
Slot 3
Slot 4
Slot 5
Slot 6
Slot 7
Cell connection
In a cellular system, a number of channels are designated as control channels that are
employed specically for transmission of information concerning call setup, call request,
196
call initiation, and other control purposes. In AMPS, there are 21 forward control channels
out of 416 channels on the forward link, and the same number on the reverse link.
When a mobile initiates a call, it sends a request on the reverse control channel to the cell
base station. Along with the request, the mobile submits its telephone number, called mobile
identication number (MIN), its electronic serial number (ESN), and the telephone number
of the called party. A station class mark (SCM) indicating the nominal EIRP of the mobile is
also transmitted to the base station. The cell base station receives this information and passes
it to the MSC, which veries the valid MIN and ESN. The ESN of a stolen phone will be
rejected by the MSC, and the call request is invalidated. The MSC instructs the cell base
station to notify the mobile via the forward control channel to tune to a full-duplex voice
channel. If the called party is also another mobile, the MSC then instructs all base stations to
broadcast a paging message for the called mobile over all forward control channels
throughout the home area. The called mobile receives the paging message from the forward
control channel of the base station that it monitors, and acknowledges over the reverse
control channel with its MIN, ESN, and SCM. The base station passes the data to the MSC to
validate the ESN. Then, the MSC instructs the base station to notify the called mobile via the
forward control channel to tune to a full-duplex channel and to ring the phone. If the called
party is a landline subscriber, the connection is established by the MSC via the PSTN. The
connection is similar when a landline subscriber initiates a call via the PSTN to the MSC.
When a mobile moves to another cell during a call, the MSC performs a handoff to
transfer the call to a new full-duplex channel of the new cell. The handoff is initiated when
the received power at the present base station falls below a prexed threshold for a predetermined dwell time, but still above a minimum usable level. Handoff is made to a new
base station once the dwell time expires and must be done before the received power at the
present base station drops below the minimum usable level.
To provide services to mobiles that travel to a new coverage area, all cellular systems
require roaming registration. The roaming mobile monitors a forward control channel and
reports its MIN and ESN over the reverse control channel upon receiving a command issued
periodically by the MSC. Unregistered mobiles are not allowed to receive or initiate calls
from the new area. Billing for roaming mobiles is forwarded to the home provider.
197
d1
d2
hT
Direct wave
Reflected
wave
hR
hR
Figure 5.14
Let be the ground reection coefcient and be the relative phase shift between E1 and E2.
Then, with phasor notation we have
E2 E1 ej
(5:47)
E E1 E2
1 ej E1
(5:48)
The phase shift can be related to the differential distance between the direct wave distance
d1 and the ground reected wave d2 as
2
d2 d 1
l
(5:49)
where is the signal wavelength. For the mobile environment, the phase shift is much
smaller than one radian, and assuming perfect ground reection, that is, = 1, we obtain
E jE1
(5:50)
The total receiver power PR at the antenna output is proportional to the square of the eld
strength, that is,
Ae
j E j2
120
Ae
jj2
jE1 j2
120
PR
(5:51)
where Ae is the effective aperture area of the receiver antenna, and the intrinsic impedance of
the free space is 120 ohms. But Ae =120 jE1 j2 is simply the received power due to the
direct wave and can be calculated via (5.42) and (5.45) as
Ae
l 2
2
jE1 j PT GT GR
(5:52)
4d1
120
198
where PT is the transmitter power, GT is the transmitter antenna gain, and GR is the receiver
antenna gain. Substituting (5.52) into (5.51) yields the received power as
l 2
2
PR jj PT GT GR
(5:53)
4d1
To calculate PR in (5.53), we need to evaluate jj2 . We note that
s
q
hT hR 2
d1 hT hR 2 d 2 d 1
d2
(5:54)
and
q
d2 hT hR 2 d 2 d
s
hT hR 2
1
d2
(5:55)
Using the fact that in a mobile environment hT hR 55d, we thus have hT hR 2 =d 2 551
and hT hR 2 =d 2 551. Applying the following series expansion:
p
1
1 2
13 3
x
x ;
1x1 x
2
24
246
15x 1
(5:56)
#
1 hT hR 2
1 hT hR 2
d1 d 1
d
2
2
d2
d
#
1 hT hR 2
1 hT hR 2
d
d2 d 1
d2
d
2
2
(5:57)
"
(5:58)
4hT hR
ld
(5:59)
Combining (5.51), (5.52), and (5.59) we obtain the received power at the receiver antenna
output as follows:
PR
(5:60)
d4
h2T h2R
(5:61)
Note that the role of hT and hR in (5.61) can be interchanged. Therefore, the path loss
calculation applies to both forward and reverse channels. Equations (5.60) and (5.61) show
199
that, at large distance, the received power is inversely proportional to the fourth power of
distance for a two-ray model. The resulting path loss is much larger than the free space loss.
Furthermore, it becomes (nearly) independent of the signal frequency. The two-ray model is
applicable to an open area with at ground.
PR LC
GT GR
(5:62)
PT
4:1 103 W
Hata model
The two-ray ground reection model is too simple to represent propagation mechanisms in
urban areas, where signals arrive at the receiver via reection, diffraction, and scattering.
Empirical formulas of the path loss for different environments have been developed. A
popular model is the Hata model, which is valid from 150 MHz to 1500 MHz [5]. The
median (50 percentile) path loss in urban areas is given in dB by
200
(5:63)
where fc is the carrier frequency in MHz, hT is the effective base station antenna height in
meters 30 m hT 200 m, hR is the effective mobile antenna height in meters
1 m hR 10 m, d is the base stationmobile separation distance in km, and a(hR) is
the correction factor for hR and is a function of the coverage area. For a small to mediumsized city:
ahR 1:1 log fc 0:7hR 1:56 log fc 0:8 dB
(5:64)
f 300 MHz
(5:65)
f 300 MHz
(5:66)
2
fc
5:4 dB
LC suburban LC urban 2 log
28
(5:67)
(5:68)
hR 2 m;
GT 0 dB;
GR 0 dB
d 10 km;
fc 894 MHz
Mobile station:
SNR 25 dB;
Ts 500 K;
B 30 kHz
Calculate the median path loss using the Hata model and the required transmitted power per
channel for the base station.
From (5.66) and (5.63), we get
ahR 3:2log11:75 22 4:97 1:05 dB
LC urban 69:55 26:16 log894 13:82 log200 1:05
44:9 6:55 log 200 log10
143:74 dB
201
Applying (5.68), we obtain the median path loss for the open area as follows:
LC open area 143:74 4:78log8942
18:33 log894 40:94
115:26 dB
If we use (5.64) instead of (5.66), we get
ahR 1:1 log894 0:72 1:56 log894 0:8
1:29 dB
We will use a(hR) = 1.05 dB for a more conservative estimate of the path loss. Substituting
GT = 1, GR = 1, LC (open area) = 3.36 1011, and PR = 6.54 1014 W as calculated in
Example 5.11 into (5.62), we obtain the transmitted power for the base station per channel as
PT 6:54 1014 3:36 1011 22 103 W
hR 2 m;
fc 894 MHz;
GT 0 dB;
Ts 500 K;
B 30 kHz
GR 0 dB
Mobile station:
SNR 25 dB;
Base station:
maximum transmitted power per channel PT 4 W
Calculate the base stationmobile separation distance d.
Using PR = 6.54 1014 W as calculated in Example 5.11, PT = 4 W, and GT = GR = 1 in
(5.62), we obtain the corresponding median path loss as follows:
PT GT GR
4
6:1 1013
6:54 1014
PR
137:9 dB
LC urban
202
Example 5.14 Modied Hata model for large cities (15002000 MHz)
Consider a cell situated in a large city with the following parameters:
hT 200 m;
hR 2 m;
fc 1950 MHz;
SNR 15 dB;
Ts 500 K;
GT 0 dB;
GR 0 dB
Mobile station:
B 200 kHz
Base station:
maximum transmit power per 200 kHz channel PT 4 W
Calculate the base stationmobile separation distance d.
The received power PR at the mobile station can be obtained from (5.43) as
PR kBTs SNR
1:38 1023 J=K 200 103 Hz 500 K31:62
4:36 1014 W
(5:69)
203
0:92 1014
4:36 1014
PR
139:6 dB
LC urban
In summary, the cellular link analysis enables us to determine the cell size given a specic
signal-to-noise ratio, receiver system noise temperature, and transmitter power. In the next
section, we will discuss the limitation imposed on a link by the cell architecture.
204
EE
DD
CC
FF
C
B
BB
GG
D
O
A
AA
HH
E
F
LL
II
JJ
KK
Figure 5.15
Mobile location
First- and second-tier co-channel cells for a seven-cell per cluster system with omni-directional antennas.
the forward link (base station-to-mobile) since this is the link that dictates the design of
cellular architecture.
Consider the forward link from the base station at the center of the reference cell O to the
mobile located at the corner of the hexagonal cell. The received signal power at the mobile is
given by (5.42) as
PR
PT GT GR
LO
(5:70)
where PT and GT are the transmitter power and the antenna gain of the base station O,
respectively, and GR is the mobile receiver antenna gain. The parameter LO represents the
path loss between the base station O and the mobile. Assume that all base stations have
identical transmitter power and antenna gain, and all mobiles have identical receiver antenna
gain. Then, the total co-channel interference power I, received by the mobile due to the rsttier base stations A, B, C, D, E, and F is
1
1
1
1
1
1
(5:71)
I PT G T G R
LA LB LC LD LE LF
where Li, i = A, B, C, D, E, F represents the path loss between the base station i and the
mobile. The co-channel interference generated by the second-tier co-channel cells is small
compared to that of the rst-tier cells due to much larger path losses and can therefore be
ignored.
The signal-to-interference ratio at the mobile is given by
SIR
PR
L1
O
1
1
1
1
1
I
LA LB LC L1
D LE LF
(5:72)
205
We know via the Friis equation that the free-space path loss is proportional to the squared
distance. For non-free-space environments, we modify the Friis equation to have a channel
path loss Lc as
Lc d n
(5:73)
where d is the distance between the transmitter and receiver, n is the path loss exponent, and
is a proportionality constant that is a function of the antenna heights of both transmitter and
receiver and the carrier frequency. Note that the median path loss, in dB, given by the Hata
model in (5.63), (5.67)(5.69) can be put in the dimensionless form of (5.73).
Let the cell radius R be the distance from the center of the cell to any of the six vertices of
the cell. Applying (5.73) to (5.72), we obtain
SIR
no
1
0 R
p nc 1 p nD
nA
nB
31R
28R
1
1
1
D
A 4R
B 5R
c
p nE
p nF
1
1
19R
13R
E
F
(5:74)
where ni and i, i = 0, A, B, C, D, E, F, are the path loss exponent and proportionality constant
between the base station i and the mobile, respectively. If all values of ni are identical and all
values of i are also identical, (5.74) reduces to the following equation:
SIR
1
pn pn pn pn
4n 5n 31 28 19 13
(5:75)
where n is the common path loss exponent for all seven clusters under consideration.
Table 5.3 shows the signal-to-interference ratio for a seven-cell per cluster cellular system
for various path loss exponents n. The result in Table 5.3 shows that the signal-tointerference ratio increases as the path loss exponent rises. This happens because the cochannel interference signals suffer more path loss in an environment with a larger path loss
exponent. This explains why, in the case of free space loss with n = 2, the SIR is very low. In
this situation, there are six direct line-of-sight co-channel signals that add up to a very strong
interference.
From the Hata model, the path loss exponent n can be obtained from (5.63) or (5.69) as
follows:
SIR (dB)
5.2
11.6
14.7
17.8
20.9
25
206
(5:76)
Here hT is the effective antenna height of the base station, and ranges from 30 m to 200 m.
According to the Hata model, the path loss exponent for a cellular environment seems to fall
between 3 and 3.5 depending on the antenna heights. Therefore, according to Table 5.3, the
signal-to-interference ratio ranges from 11.6 dB for hT = 200 m to 14.7 dB for hT = 30 m.
From the above results, we conclude that a higher base station antenna actually reduces the
signal-to-interference ratio because of stronger co-channel interference. But, also according
to the Hata model, a higher base station antenna reduces the path loss and thus increases the
signal-to-noise ratio. Therefore, a compromise in the base station antenna height to achieve
both the specied signal-to-interference ratio and the signal-to-noise ratio should be carried
out in any design.
For the reverse link, SIR can be calculated in a similar way by considering the worst case
of a mobile station located at the corner of the center cell of a seven-cell cluster. There are six
interfering mobile stations from the corresponding rst-tier co-channel cells. We have
SIR
1
64n
(5:77)
For example, letting n = 4, we obtain SIR = 16.3 dB. This is slightly smaller than SIR
= 17.8 dB of the forward channel.
PR
I N
(5:78)
where PR is the received signal power, I is the received co-channel power, and N is the
received noise power. Furthermore,
SINR
1
1
PR =I PR =N 1
(5:79)
or, equivalently,
SINR
1
1
SIR SNR
1
(5:80)
207
Example 5.15 Signal-to-interference and noise ratio for a seven-cell per cluster system
Consider a system in a large city that employs a frequency reuse plan of seven-cells per
cluster. The system has the following parameters:
hR 2 m;
fc 894 MHz;
GT 0 dB;
GR 0 dB;
B 30 kHz;
d 6:35 km
Base station:
transmitted power per channel 4 W
Mobile:
system noise temperature 500 K
Calculate the signal-to-interference and noise ratio in (5.80) using the Hata model for large
cities for two base station antenna heights hT = 200 m and 100 m.
(a) hT = 200 m
Using (5.63) and (5.66) we have the following path loss:
LC urban 69:55 26:16 log 894 13:82 log 200 1:05
44:9 6:55 log 200 log 6:35 137:9 dB
The received power at the mobile is given by
PR
PT GT GR
411
6:54 1014 W
LC urban
1013:79
PR
6:54 1014 W
316 25 dB
23
kTB 1:38 10 J=K500 K30 103 Hz
From (5.76), the path loss exponent is n = 3. Using Table 5.3, the signal-to-interference
ratio SIR for hT = 200 m is 11.6 dB. Applying (5.80), we obtain the signal-tointerference and noise ratio as
SINR
1
1
2:5
1:16
10
10
1
13:76 11:4 dB
The above result shows that SINR SIR, that is, the link is interference-limited, and
thermal noise has negligible effect.
(b) hT = 100 m
Again, using (5.63) and (5.66), we get the following path loss:
LC urban 69:55 26:16 log 894 13:82 log 100 1:05
44:9 6:55 log 100 log6:35 143:6 dB
208
PT GT GR
4 1 1
1:75 1014 W
LC urban
1014:36
PR
1:75 1014 W
84:35
23
kTs B 1:38 10 J=K500 K30 103 Hz
19:3 dB
From (5.76), the path loss exponent is n = 3.18. Substituting n = 3.18 into (5.75), we
have SIR = 18.65 = 12.7 dB. Using (5.80) we obtain the signal-to-interference and noise
ratio as follows:
1
1
1
SINR
15:27 11:84 dB
18:65 84:35
The above results show that in a cellular environment, taller base station antennas do not
necessarily provide a better signal-to-interference and noise ratio. For narrowband cellular
systems, fairly good voice quality can be provided with a signal-to-interference and noise
ratio of about 14 dB. This requirement cannot be met with a cell radius R = 6.67 km, at least
from the worst-case point of view. One could use lower base station antenna heights to
reduce the co-channel interference, that is, to increase SIR; for example, hT = 30 m would
yield SIR = 14.7 dB. Also, to increase SNR, the radius of the cell should be made smaller.
From Example 5.15, a cell radius R = 3 km would yield a worst-case path loss for a mobile at
a distance d = R = 3 km to be LC (urban) = 141.5.1 dB for hT = 30 m and a corresponding
SNR = 21.34 dB. This would yield a signal-to-interference and noise ratio SINR = 13.85 dB,
which meets the worst-case performance. This method increases the number of clusters per
coverage area, and hence provides a larger capacity at the expense of more base stations.
Another option is to increase the number of cells per cluster to reduce SIR while keeping the
cell radius xed. This method reduces the number of clusters per coverage area and hence
decreases the system capacity.
Sectoring
Another popular method to reduce co-channel interference is sectoring. Sectoring is the
replacement of the base station omni-directional antenna with several directional antennas,
each transmitting to a certain sector of the cell. Sectoring patterns are normally implemented
with three 120 antennas or six 60 antennas. In sectoring, the total number of channels per
cell is divided equally among sectors. Thus, an idle channel in one sector cannot be assigned
to a mobile in another sector of the same cell, hence sectoring increases the call blocking
probability. Note that, in a conventional system, an idle channel in a cell can be assigned to
any mobile in that cell. Therefore, sectoring reduces the capacity of a cellular system since
209
SIR (dB)
2
3
3.5
4
4.5
5
8.7
15.3
18.6
21.8
25.1
28.3
DD
EE
FF
CC
B
C
GG
BB
A
O
D
HH
AA
F
E
II
LL
KK
Figure 5.16
JJ
Mobile location
First- and second-tier co-channel cells for a seven-cell per cluster system and 120 sectoring.
the channels in a cell cannot be pooled together and assigned to mobiles on demand. This
trunking efciency reduction increases with the number of sectors per cell. Another
disadvantage of sectoring is the frequent handoffs when a mobile moves from one sector
to another sector. Modern base stations are normally equipped to handle sectoring handoffs
without requiring the assistance of the MSC, and therefore this is not a major concern.
Let us consider rst the forward link. Figure 5.16 shows a seven-cell per cluster system
with 120-sectoring. The worst-case mobile location is at the right-hand lower corner of cell
O. The mobile is interfered by the three left-looking sectors of cells B, C, and D. The signalto-interference ratio can be calculated as
SIR
1
pn
31 4n
5n
(5:81)
Table 5.4 tabulates the signal-to-interference ratio for various path loss exponents. A look at
Table 5.4 reveals that SIRs improve by almost 4 dB for n in the range of 3 to 3.5.
210
5n
1
pn
31
(5:82)
Example 5.16 Signal-to-interference and noise ratio for a seven-cell per cluster system with
120o-sectoring
Consider the system in Example 5.15 with base station antenna height hT = 100 m. The path
loss exponent obtained from the Hata model is n = 3.18, which yields SIR = 16.5 dB. For a
hexagon cell with radius R = 6.67 km, the worst case mobile location d = R = 6.67 km yields
a SNR = 19.2 dB assuming the same antenna gain. The combined signal-to-interference and
noise ratio is SINR = 14.6 dB as compared to 11.82 dB in the no-sectoring case. In practice,
the sectoring antenna gain is larger than the omni-directional antenna gain and SINR could
be better. For example, if GT 2, then SNR = 22.2 dB and SINR = 15.5 dB.
For the reverse link, with the mobile station at the corner of the cell, the signal-tointerference ratio for the 120o-sectoring is given by
SIR
5n
1
pn
19 4n
(5:83)
1
pn
19
(5:84)
5n
Microcell-zoning
Sectoring requires the allocation of a xed set of channels to each sector. Thus, when the
mobile moves from one sector to another sector in the same cell, handoff must be performed
to a new channel in the new sector. Microcell-zoning, as illustrated in Figure 5.17, can be
used to avoid too many handoffs in the same cell. The cell is divided into three micro-zones
Zone selection
Figure 5.17
211
with directional antennas aiming toward the center of the cell. The three antennas are
connected via terrestrial lines to a zone-selection switch, which in turn is connected to the
base station of the cell. The mobile station signal is received by all three zone antennas, and
the switch selects the strongest signal. The base station communicates with the mobile via
the corresponding zone antenna that produces the strongest signal. In this manner, the
mobile station can move all over the cell without handoffs. For a seven-cell per cluster
system, the signal-to-interference ratio for the forward link assuming the mobile station is at
the zone center is given by the following expression:
1
SIR pn pn pn pn pn pn
112 109 103 64 79 91
(5:85)
Forward link
Consider seven hexagonal cells in Figure 5.18 arranged with the interested cell at the center
surrounded by six adjacent co-channel cells. The base stations have omni-directional
antennas and are located at cell centers. The worst case signal-to-interference ratio occurs
Figure 5.18
CDMA cells with base stations at cell centers and omni-directional antennas.
212
when the mobile station is located at the corner of the cell boundary. We make use of the
results derived for CDMA in Chapter 8 for the following link analysis. The spread factor of
the orthogonal covering forward channel is denoted as N. To be modulation-independent
and PN sequence-independent, we use a conservative approach by accepting the interference power in the despread signal bandwidth as PI =N , where PI is the power of the
interference signal. Assuming each base station transmits the same power and their forward
links contain identical number of user channels, K, where K N , we obtain the following
worst case despread input signal-to-interference ratio:
N =K
SIR pn
2 7 22n 2
(5:86)
where n is the path loss exponent (assuming to be identical for all cells), and is the voice
activity factor or data duty cycle. The despread input signal-to-interference-and-noise ratio
h
i1
is SINR SIR1 SNR1 .
9:4 1014
4:26 1015
PR
149:7 dB
LC urban
The distance d can be evaluated via the modied Hata model. We have
213
ahR 1:05 dB
LC urban 46:3 33:9 log 1950 13:82 log 200
1:05 44:9 6:55 log 200 log d 3
149:7 dB
Solving for d we get d = 5.35 km.
Now consider the co-channel interference for this forward link. Let us assume that the
path loss exponent n = 4, the voice activity factor 0:4, and a required despread input
signal-to-interference ratio SIR 7 dB. Then, the number of transmitted user channels per
base station is K = 15. This is the maximum number of user channels plus overhead channels
(pilot channel for example) that the forward link can carry at any instant of time. It is smaller
than the total number of user channels, which is 64 for IS-95. Using the input signal-to-noise
ratio SNR 15 dB, we get SINR 6:4 dB.
In narrowband cellular systems, sectoring is employed to reduce the co-channel interference in order to improve link reliability. The allocated channels in a sector cannot be used in
other sectors since the frequency plan is xed. On the other hand, CDMA cellular systems
operate with the entire allocated frequency spectrum for each cell. Therefore, a CDMA
forward link is used for the entire cell irrespective of the sectors. This is called sector
frequency reuse and it increases the cell capacity by the number of sectors. Thus, a mobile
station, at the cell corner as shown in Figure 5.19, experiences all six CCI forward links with
the same frequency. Hence, the SINR of sectoring CDMA remains the same as in the case of
no sectoring, as is the number of user channels per sector. But the number of total user
channels is now 3K for 120 -sectoring and 6K for 60 -sectoring. From Example 5.17, the
maximum number of user channels that can be accommodated by a forward link is 45 with
120 -sectoring, and 64 with 60 -sectoring. For the US cellular band, the allocated frequency
spectrum for the forward channel is 12.5 MHz, which accommodates ten 1.25 MHz-CDMA
carriers. Thus, a CDMA cell can have 10K user channels for no sectoring, 30K user channels
for 120 -sectoring, and max[60K, 10N] user channels for 60 -sectoring.
Figure 5.19
CDMA cells with base stations at cell centers and 120 -sectoring antennas.
214
Reverse link
The reverse link of a CDMA cellular system is a multiple access channel with mobile
stations in the cell asynchronously transmitting to the base station using the same carrier
frequency but with different PN sequences. The user rates may be identical or they may be
different (integer multiple of the basic rate). With power control, all mobile station signals
arrive at the base station with the same power. Both MUI and CCI must be accounted for.
The despread input signal-to-interference-and-noise ratio SINR of the user channel consists
of the despread input signal-to-MUI ratio, the despread input signal-to-CCI ratio,
h and the
despread input signal-to-noise ratio. Therefore, we have SINR SIR1
MUI
1 1
SNR
.
With
power
control
we
have
SIR
N
=K,
where
K
is
the
SIR1
MUI
CCI
number of multi-access users and is the voice activity factor or data duty cycle. The
CCI depends on the distribution of mobile stations in the adjacent cells. In each adjacent
cell, only a small fraction of interfering mobile stations, which are closer to the base
station in the center cell, contributes to CCI. For example, assuming hexagonal cells
and a path loss exponent n = 4, the interfering mobile station signals at the distance of 1.5
times the cell radius from the receiving base station in the center cell are attenuated by 7 dB.
If the distance is increased to twice the cell radius, the attenuation is 12 dB. Thus, for no
sectoring, it is reasonable to model the CCI of all six adjacent cells by SIRCCI N =K,
where is a factor that reects the real world situation. For 120 -sectoring, only two
adjacent cells facing the intended sector contribute to CCI, hence, we have
SIR120
CCI 3N =K. The number of simultaneous users per sector is K; the number of
simultaneous users for a 120 -sectoring cell is 3K per CDMA carrier. For 60 -sectoring,
only one adjacent cell facing the intended sector contributes to CCI; hence, we have
SIR60
CCI 6N =K. The number of simultaneous users per sector is K; the number of
simultaneous users for a 60 -sectoring cell is 6K per CDMA carrier. For the US cellular
band, the allocated frequency spectrum for the reverse channel is 12.5 MHz, which accommodates ten 1.25 MHz-CDMA carriers. Thus, a CDMA cell can have 10K user channels for
no sectoring, 30K user channels for 120 -sectoring, and max[60K, 10N] user channels for
60 -sectoring.
10:67
10:3
dB.
Furthermore,
set
SIR
3N
=K 28:8 14:6 dB
SIR120
MUI
CCI
with 0:9. Thus, SINR = 7.9 (9 dB) as compared to 6.4 dB for the non-sectoring case.
215
This is possible because the CCI power is greatly reduced. The number of users per CDMA
carrier is 45 or 450 for the entire cell. For 60 -sectoring, the number of users per CDMA
carrier is 64 or 640 for the entire cell.
(5:87)
Here S is the GEO satellite longitude, and L and are the Earth station longitude and
latitude, respectively.
The elevation angle E can be calculated as follows:
r Re cos cosjS L j
1
E tan
Re sincos1 cos cosjS L j
(5:88)
1
cos cos cosjS L j
where r = 42 164 km and Re = 6378 km is the Earth radius.
The distance between the Earth station and the GEO satellite, called the slant range, is
given by
s
1 Re
2
2
d r Re 2Re r sin E sin
cos E
(5:89)
r
216
Antenna
Low-noise
amplifier
Figure 5.20
Bandpass
filter
Downconverter
Power
amplifier
Repeater transponder.
4d
l
2
(5:90)
where = c/f is the signal wavelength, c 3 108 m=s is the speed of light, and f is the
carrier frequency.
Commercial GEO satellites operate mainly in the C-band (uplink: 6 GHz, downlink:
4 GHz) and the Ku-band (uplink: 14 GHz, downlink: 12 GHz). In each band, the higher
frequency is used for the uplink (Earth station to satellite) and the lower frequency is used for
the downlink (satellite to Earth station). The available bandwidth for the uplink or downlink
of each band is about 500 MHz. A GEO satellite carries many transponders. Each transponder consists of a transreceiver that receives the uplink signal (6 or 14 GHz) in a particular
bandwidth, then downconverts it to the downlink frequency (4 or 12 GHz), amplies the
signal and retransmits it back to Earth. This type of transponder is called a repeater and is
shown schematically in Figure 5.20.
217
Downlink
Uplink
6 or 14 GHz
Figure 5.21
4 or 12 GHz
Satellite link.
A satellite link consists of both uplink and downlink, as shown in Figure 5.21. The uplink
signal power at the satellite receiver input is
Pu
PT GT GS
Lu
(5:91)
where:
PT = transmitter power of the uplink Earth station
GT = antenna gain of uplink Earth station
GS = antenna gain of satellite receiver
Lu = uplink path loss
The noise power in the satellite receiver is
Nu kTsat B
(5:92)
where Tsat is the satellite system noise temperature, B is the channel bandwidth, and k is
Boltzmann constant.
The Earth station on the downlink receives the satellite signal and accompanied noise
with respective powers
Pd
Pu GGR
Ld
(5:93)
Nu GGR
Ld
(5:94)
where:
G = total transponder gain, which includes satellite transmit antenna gain
GR = antenna gain of downlink Earth station
Ld = downlink path loss
The noise power in the downlink Earth station receiver is
Nd kTs B
(5:95)
Here Ts is the downlink Earth stations system noise temperature. The combined signal-tonoise ratio at the input of the downlink Earth stations receiver is given by
218
SNR
Pd
1
1
1
1 1 1
1
N Nd
SNRu SNR1
Pd 1
Pd
Pu
Pd
d
N
Nd
Nu
Nd
(5:96)
The ratio SNRu Pu =Nu is the uplink signal-to-noise ratio and the ratio SNRd Pd =Nd is
the downlink signal-to-noise ratio. They are given by
SNRu
Pu PT GT GS EIRPGS
Nu Lu kTsat B
Lu kTsat B
(5:97)
and
Pd Pu GGR EIRPS GR
(5:98)
Nd Ld kTs B
Ld kTs B
The quantities EIRP PT GT and EIRPS Pu G are called the effective isotropic radiated
powers of the uplink Earth station and satellite, respectively. The quantities GS =Tsat and
GR =Ts are called the antenna gain-to-noise temperature ratios of the satellite and downlink
Earth station, respectively.
SNRd
SNR
102:77
219
5.6 Summary
5.6 Summary
In this chapter we have provided a detailed study of wireless communication links in free space and in cellular
environments. The purpose of a link analysis is to nd the system signal-to-noise ratio at the receiver.
The noise power is determined by the receiver system noise temperature, which is a function of the antenna
noise temperature and the effective noise temperatures of various subsystems in the receiver, such as the lownoise amplier, the downconverter, and the demodulator. We have shown that if the gain of the low-noise
amplier is large, then the noise contribution of the rest of the receiver can be made negligible.
The signal power in free space is a function of the path loss and can be evaluated by the Friis equation. We
have also shown that the system signal-to-noise ratio is invariant with respect to the reference inputs at the
receiver.
We have introduced the concept of frequency reuse in cellular communications. Each coverage area is divided
into clusters; each cluster has a xed number of cells. Each cell in a cluster employs a separate portion of the
frequency spectrum to avoid inter-cell interference. Each cluster reuses the same frequency spectrum. If the
number of cells per cluster is large, the co-channel interference between co-channel cells in adjacent clusters can
be kept low.
A review of major cellular standards, such as AMPS, NAMPS, IS-136, IS-95, and GSM, was provided together
with a discussion of call connection and handoff.
The study of the two-ray ground reection path loss model in a cellular environment has shown that the path
loss is proportional to the fourth power of the distance as compared to the square of the distance in free space
loss.
The well-known empirical Hata model was introduced. This model can calculate the median path loss in
small, medium-sized, and large cities as well as suburban and rural areas. The model covers the 150
2000 MHz frequency band. The Hata model was employed in the cellular link analysis, for example, to
determine the cell size given a specied signal-to-noise ratio, receiver system noise temperature, and transmitter
power.
The co-channel interference imposed on a cellular link was shown to be architecture-specic. Practical
narrowband cellular systems, such as AMPS, GSM, and IS-136, normally employ seven cells per cluster. The
signal-to-interference ratio for the worst-case mobile location was presented. Wideband cellular systems, such as
IS-95, CDMA 2000, and WCDMA, employ one cell per cluster and rely on spread spectrum processing gain to
reduce the interference effect.
The link analysis and co-channel interference analysis were then integrated to calculate the combined signalto-interference and noise ratio. In order to reduce the co-channel interference and improve performance, cellular
systems normally use sectoring at the base station. Sectoring patterns are commonly implemented with three
120-sectoring antennas or six 60 -sectoring antennas.
Finally, the communication link for a geostationary satellite was studied. The azimuth and elevation angles are
needed to point the Earth station antenna toward the satellite for both uplink transmission and downlink
reception. The slant range between an Earth station and a geostationary satellite was shown to be a function of
the elevation angle. The satellite path loss can be calculated via the Friis equation. The uplink and downlink
signal-to-noise ratios were combined to form the total link signal-to-noise ratio.
220
Problems
1.
2.
3.
4.
5.
221
Problems
6.
7.
8.
9.
10.
11.
12.
13.
14.
Show that the median path loss, in dB, given by the Hata model in (5.63), (5.67)
(5.69) can be put into the dimensionless form of (5.73) with the path loss exponent
given by (5.76).
Consider a cell in an open rural area. The base station antenna has a height of 200 m,
and the mobile antenna height is 2 m. The base station antenna gain is 3 dB and the
mobile employs an omni-directional antenna with 0 dB gain. A forward channel has a
bandwidth of 200 kHz. The cell has a radius of 10 km. Assume that the received
signal-to-noise ratio at the mobile station must be at least 15 dB for the forward link
and that the system noise temperature of the mobile station receiver is 700 K.
Calculate the received power at the mobile station and the required transmitted
power for 60 channels at the base station assuming all mobiles are located at the
cell boundary. Use the two-ray ground reection model for open rural areas.
Consider a cell in an open rural area. The base station antenna has a height of 200 m, and
the mobile antenna height is 2 m. The base station antenna gain is 3 dB and the mobile
employs an omni-directional antenna with 0 dB gain. A forward channel has a bandwidth of 200 kHz. The cell has a radius of 10 km. Assume that the received signal-tonoise ratio at the mobile station must be at least 15 dB for the forward link and that the
system noise temperature of the mobile station receiver is 700 K. Calculate the received
power at the mobile station and the required transmitted power for 60 channels at the
base station assuming all mobiles are located at the cell boundary. Assume that the
nominal carrier frequency is 894 MHz. Use the Hata model for open rural areas.
Consider a cell in a large city that is allocated 60 GSM channels. The base station
antenna has a height of 200 m, and the mobile antenna height is 2 m. The base station
antenna gain is 3 dB and the mobile employs an omni-directional antenna with 0 dB
gain. A forward GSM channel has a bandwidth of 200 kHz. Assume that the received
signal-to-interference and noise ratio at the mobile station must be at least 9 dB for the
forward link and that the system noise temperature of the mobile station receiver is
700 K. The maximum transmitted power for the base station is 240 W. Assume that the
nominal carrier frequency is 894 MHz. Calculate the maximum radius of the cell with
the Hata model for large cities and include co-channel interference.
Repeat Problem 9 for a cell in a large city with nominal carrier frequency of 1950 MHz
and receiver noise temperature of 500 K.
Repeat Problem 9 with 120 -sectoring and 60 -sectoring.
Repeat Problem 10 with 120 -sectoring and 60 -sectoring.
For a three-cell per cluster system using microcell-zoning, derive the signal-tointerference ratio for the forward link, assuming that the mobile station is at the zone
center.
Consider a forward CDMA link situated in a large city with the following parameters:
hT 100 m; hR 2 m; fc 1950 MHz; GT 0 dB; GR 0 dB;
BCDMA 1250 kHz; N 64:
Mobile station:
Ts 500 K
222
Base station:
maximum transmit power per Walsh channel PT 4 W
15.
16.
Assume that the path loss exponent n = 4, and the voice activity factor 0:4. The
required despread input signal-to-interference-and-noise ratio is 8 dB. Given a cell
radius of 5 km, calculate maximum number of user channels plus overhead channels
(pilot channel for example) that the forward link can carry at any instant of time:
(a) without sectoring;
(b) with 120 -sectoring;
(c) with 60 -sectoring.
Ten forward CDMA links carry four types of users with bit rates kRb , k = 1, 2, 3, 4. The
percentage of users with bit rate Rb is 40%, the percentage of users with bit rate 2Rb is
25%, the percentage of users with bit rate 3Rb is 20%, and the percentage of users with
bit rate 4Rb is 15%. A forward link carries a maximum of 60 user channels with basic
bit rate Rb . Calculate the number of users for each type for the following cases:
(a) without sectoring;
(b) with 120 -sectoring;
(c) with 60 -sectoring.
Consider a reverse CDMA link situated in a large city with the following parameters:
hT 100 m; hR 2 m; fc 1850 MHz; GT 0 dB; GR 0 dB;
BCDMA 1250 kHz; N 64
Base station:
Ts 500 K
Mobile station:
maximum transmit power per user channel PT 4 W
17.
18.
Assume that the path loss exponent n = 4, and the voice activity factor 0:4. The
required despread input signal-to-interference-and-noise ratio is 8 dB. Given a cell
radius of 5 km, calculate maximum number of user channels plus overhead channels
(pilot channel for example) that the reverse link can carry at any instant of time:
(a) without sectoring;
(b) with 120 -sectoring;
(c) with 60 -sectoring.
Consider a 1.25 MHz reverse CDMA link with a required SINR = 8 dB. Given N = 64,
0:5, and SNR = 20 dB. The maximum despread input signal-to-CCI ratio without
sectoring is SIRCCI 10 dB. The allocated spectrum is 12.5 MHz.
(a) Find the total number of simultaneous users in the cell without sectoring.
(b) Find the total number of simultaneous users in the cell with 120 -sectoring.
(c) Find the total number of simultaneous users in the cell with 60 -sectoring.
Consider a forward CDMA link with two types of users of bit rates Rb and 2Rb . The
users with the highest bit rate (30% of all users) have a spread factor of N and the users
with the lowest bit rate (70% of all users) have a spread factor of 2N. Thus, all
223
Problems
19.
transmitted user signals have the same bandwidth. The available power for the link is
P. Find the power allocated to a user of each type, so that the SINR is the same.
Consider a C-band satellite link (uplink frequency: 6 GHz, downlink frequency:
4 GHz). The required signal-to-noise ratio for the combined link must be at least
25 dB. The following parameters are given for the satellite system:
Satellite:
B 36 MHz:
Transmit Earth station: uplink slant range du 37 300 km:
Receive Earth station:
20.
21.
22.
23.
(a) Calculate the downlink signal wavelength ld and the uplink signal wavelength lu .
(b) Calculate the downlink path loss Ld (dB) and the uplink path loss Lu (dB).
(c) Calculate the downlink signal-to-noise ratio.
(d) Calculate the required uplink signal-to-noise ratio.
(e) Calculate the minimum EIRP of the transmit Earth station.
Consider a Ku-band (14/12 GHz) satellite link with the following parameters: bandwidth = 36 MHz, satellite antenna gain-to-noise temperature ratio = 1.6 dB/K, satellite
EIRP = 44 dBW, transmit Earth station antenna gain = 57.6 dB, signal power from
transmit Earth station = 174 W, receive Earth station antenna gain = 56.3 dB, receive
Earth station system noise temperature = 160 K, uplink slant range = downlink slant
range = 37 506 km, tracking loss = 1.2 dB (uplink) and 0.9 dB (downlink). Calculate
the combined SNR (dB) of the entire satellite link.
The power ux density at the satellite induced by a transmitted signal with a given
EIRP is dened as EIRP=4du2 , where du is the uplink slant range. Express the
uplink signal-to-noise ratio as a function of O, satellite antenna gain-to-noise temperature ratio, uplink frequency fu, and other suitable parameters, such as Boltzmann
constant k, signal bandwidth B, and speed of light c.
Consider a C-band (6/4 GHz) satellite transponder that accommodates 200 distinct
carriers with identical bandwidth. The carrier bandwidth is 40 kHz. The satellite
antenna gain-to-noise temperature ratio is 7 dB/K, and its transponder EIRP is
36 dBW. To operate the transponder in a linear mode, the transponder amplier output
power is backed off by 6 dB. The power ux density at the satellite induced by
all 200 carriers is 91 dBW/m2. The receive Earth station has an antenna gain-tonoise temperature ratio of 22 dB/K. The downlink slant range is 37 506 km. Calculate
the combined SNR (dB) of the entire satellite link
A point-to-point communication system has the following parameters:
PT 0:5 W; GT 10 dB; LC 150 dB; GR 10 dB; TA 160 K;
GLNA 40 dB; TLNA 200 K; GDC 10 dB; TDC 5000 K;
TDEM 4000 K; B 50 KHz
(a) Find the system noise temperature.
(b) Find the system SNR at the LNA input.
(c) Find the noise power at the demodulator input in dBW.
224
Further reading
For anyone who wants to study more about cellular systems we recommend the indispensable book by Rappaport [1]. The material on WCDMA and CDMA-2000 is readily
available on the Internet, or the reader could consult the book by Garg [4]. For a detailed
study of satellite communication, we recommend the book by Ha [6].
Bibliography
1. T. S. Rappaport, Wireless Communications, 2nd edition, Englewood Cliffs, NJ: PrenticeHall, 2002.
2. EIA/TIA Interim Standard, Cellular system dual mode mobile stationland station
compatibility specications, IS-54, Electronic Industries Association, May 1990.
3. TIA/EIA Interim Standard-95, Mobile stationbase station compatibility standard for
dual mode wideband spread spectrum cellular system, July 1993.
4. V. K. Garg, Wireless Network Evolution: 2G to 3G, Englewood Cliffs, NJ: Prentice Hall,
2002.
5. M. Hata, Empirical formula for propagation loss in land mobile radio services, IEEE
Trans. Vehicular Technol., Vol. VT-29, No. 3, pp. 317325, 1980.
6. T. T. Ha, Digital Satellite Communications, 2nd edition, New York: McGraw-Hill, 1990.
Modulation
Introduction
The purpose of digital modulation is to transmit information symbols via the amplitude and/
or phase and/or frequency of a sinusoidal carrier. The symbol is transmitted over a symbol
time which is the inverse of the symbol rate. For binary modulation, a bit is the same as the
symbol. For higher order M-ary modulation (M 2k ), a symbol consists of k log2 M
bits. For binary modulation we investigate phase shift keying (PSK), differential phase shift
keying (DPSK), amplitude shift keying (ASK), frequency shift keying (FSK), minimum shift
keying (MSK), and Gaussian minimum shift keying (GMSK). For M-ary modulation we
provide a detailed study of M-ary amplitude shift keying (MASK), M-ary phase shift keying
(MPSK), offset quadrature phase shift keying (OQPSK), differential M-ary phase shift
keying (DMPSK), M-ary quadrature amplitude modulation (MQAM), differential M-ary
quadrature amplitude modulation (DMQAM), code shift keying (CSK), M-ary frequency
shift keying (MFSK), continuous phase modulation (CPM), and orthogonal frequency
division multiplexing (OFDM). For each modulation technique we will provide the signal
space, power spectral density, and modulator.
(6:1)
Note that the amplitude of s(t) is Am(t), which varies according to the message signal m(t),
hence, the name amplitude modulation. In other words, the message signal is now being
carried in the amplitude of the DSB-AM signal. The generation of s(t) is achieved by the
modulator shown in Figure 6.1
The voltage multiplier shown in Figure 6.1 is commonly referred to as the mixer.
225
226
Modulation
m(t)
A cos 2fc t
Figure 6.1
DSB-AM modulator.
m(t)
s(t)
Figure 6.2
(6:2)
A plot of s(t) in (6.2) is shown in Figure 6.2. Note that a phase reversal occurs at the point
where m(t) goes negative.
When the message signal m(t) is deterministic and has the Fourier transform M( f ), then
the Fourier transform S( f ) of the DSB-AM signal s(t) in (6.1) can be evaluated using the
modulation property in Table 2.1:
S f
A
M f fc M f fc
2
(6:3)
227
m(t)
3
2
1
0
2T
3T
4T
2
3
(a)
00
01
10
11
(b)
Figure 6.3
Figure 6.4 illustrates the concept of DSB-AM in the frequency domain, and shows that M( f )
is shifted to fc :
Note that S( f ) consists of two sidebands: the upper sideband (USB) and the lower sideband
(LSB). The bandwidth of a DSB-AM signal is 2B, which is twice that of the message
signal.
(6:4)
which is shown in Figure 6.5. The USB consists of two impulse functions at fc f0 and
fc f0 , and the LSB consists of two impulse function at fc f0 and fc f0 .
When m(t) is a random signal (hence, its Fourier transform does not exist) we can look at
the distribution of its power as a function of frequency via the power spectral density Sm( f ).
228
Modulation
M( f )
(a)
S( f )
A/2
LSB
USB
USB
LSB
f
fc B
fc
fc + B
fc B
fc
fc + B
(b)
Figure 6.4
(a) Fourier transform of a message signal m(t); (b) Fourier transform of DSB-AM signal s(t).
S(f)
A/4
f
fc f0
Figure 6.5
fc
fc + f0
fc f0
fc
fc + f0
Fourier transform of a DSB-AM signal s(t) with a sinusoidal message signal m(t).
Consequently, the power spectral density Ss( f ) of the DSB-AM signal s(t) in (6.1) can be
evaluated via (3.71) as follows:
Ss f
A2
Sm f fc Sm f fc
4
(6:5)
229
Sm( f )
Tb
2
Tb
1
Tb
1
Tb
2
Tb
(a)
Ss( f)
A2Tb
4
fc
1
Tb
fc
fc+
1
Tb
fc
1
Tb
fc
fc+
1
Tb
(b)
Figure 6.6
Ss f
(6:7)
4
f fc Tb
f fc Tb
Both Sm( f ) and Ss( f ) are shown in Figure 6.6.
From Figure 6.4 or Figure 6.5, we observe that the message signal m(t) can be extracted
from the DSB-AM signal by simply retranslating its spectrum to the zero frequency.
230
Modulation
r R=B
(6:10)
An interesting observation on (6.9) reveals that, in the case of innite bandwidth, the
quantity r R=B in bits/s/Hz, commonly referred to as the spectral efciency, approaches
zero. On the other hand, the limit of Eb =N0 can be found by LHpitals rule as
Eb
2r 1
d=dr2r 1
4 lim
lim
lim 2r ln 2 ln 2 0:693
r!0
r!0
d=drr
N0 r!0 r
(6:11)
Or, in dB,
Eb
4 1:59 dB
N0
Shannon limit
(6:12)
It is seen that the ratio Eb =N0 cannot be less than 1.59 dB, even with innite bandwidth,
for reliable transmission. A plot of r versus Eb =N0 is shown in Figure 6.7.
The Shannon theorem provides a trade-off between the signal-to-noise ratio per bit
Eb =N0 , which dictates the power efciency of a digital modulation, and the spectral
efciency r, which dictates the bandwidth efciency. Note that the Shannon theorem says
nothing about the degree of reliability of the transmission, that is, about the bit error rate of a
digitally modulated signal. Given an Eb =N0 that is larger than 1.59 dB, and a spectral
efciency r such that the data rate is less than the channel capacity, one can always nd a
digital modulation and a channel coding scheme to achieve a desired bit error rate. A low
error rate certainly requires a higher system complexity.
231
5
4.5
4
r (bits/s/Hz)
3.5
3
2.5
2
1.5
1
0.5
0
Figure 6.7
4
Eb /N0
232
Modulation
shape p(t) can be a pulse of unit amplitude and duration Tb per (2.26), or a raised cosine
pulse, which is commonly used in practice to shape the PSK spectrum to a required
bandwidth and to eliminate intersymbol interference. Note that s1(t) and s2(t) in (6.13) can
be combined into one for an innite sequence of bits as follows:
"
#
1
X
2di 1 pt iTb cos 2fc t
(6:14)
st A
i1
where di 2 f0; 1g represents the ith data bit. The complex envelope or equivalent lowpass
signal of s(t) is given by
m t
1
X
2di 1 pt iTb
(6:15)
i1
Thus, PSK is simply a DSB-AM signal with the message signal m(t). Note that m(t) is the
antipodal signal described in Example 2.3, where p(t) is a unit amplitude squared pulse.
Figures 2.4 and 2.6 show the complex envelope and the PSK signal itself for the sequence of
bits {1 0 1 1} with squared pulse shape.
1
Tb
2Tb
Figure 6.8
3Tb
Tb
2Tb
3Tb
233
8
>
>
>
<
Tb ;
Tb
Tb
1
P f
1 cos
;
jfj
>
2
2Tb
>
>
:
0;
0 j f j5
1
2Tb
1
1
jfj
2Tb
2Tb
j f j4
(6:17)
1
2Tb
where is the lter roll-off factor. Applying the Parseval formula in Table 2.2 to (6.16), we
have
1
1
S12 f df
S22 f df
(6:18)
Eb
1
1
where S1( f ) and S2( f ) are the energy spectra of s1(t) and s2(t) in (6.13), respectively. Using
the frequency shifting property in Table 2.1 on (6.13), we get
S1 f S2 f
A
P f f c P f f c
2
(6:19)
1
2
P2 f df
(6:20)
Hence,
2
Tb2
Tb
1
df
1 cos
f
2Tb
4
0
1=2Tb
A2 Tb
A2 Tb2 1=2Tb
Tb
1
Tb
1
12 cos
cos2
df
1
f
f
2Tb
2Tb
2
4 1=2Tb
A2 Tb
A2 Tb2
A2 T b
1
1
Tb 2Tb
4
2
4
2
Eb A2
1=2Tb
Tb2 df A2
1=2Tb
(6:21)
j P f j 2
Tb
(6:22)
234
Modulation
Ss( f )
A2Tb
4
f
1+
fc
2Tb
Figure 6.9
fc
1+
fc+
2Tb
1+
fc
2Tb
1+
fc+
2Tb
fc
di {0, 1}
Pulse
shaping
m (t)
PSK
A cos 2 fc t
Figure 6.10
PSK modulator.
Note that the raised-cosine pulse shape limits the bandwidth to 1 =Tb , which is
between 1=Tb and 2=Tb as shown in Figure 6.9. On the other hand, the squared pulse
shape produces a power spectral density with an innite bandwidth, although most of
the PSK signal power is contained in the null-to-null bandwidth of 2=Tb as shown in
Figure 6.6. The spectral efciency of PSK is in the range 0.51 bits/s/Hz.
The power of the PSK signal in (6.15) with a raised-cosine pulse shape is given by
1
1
2
P
Ss f df A
Sm f df
1
0
(6:23)
1 2
P f
A2 1 2
A2
2
1
df
P f df
A
Tb 0
2
Tb
4
0
Note that the bit energy Eb is equal to PTb as expected.
The PSK signal in (6.13) or (6.14) can be generated by the modulator shown in Figure 6.10.
The PSK signal space is the same as that of the antipodal signal and is shown in
Figure 2.13. Note that the basis function for the signal set in (6.13) is
p
x1 t A= Eb pt cos 2fc t, where Eb is given in (6.16). Thus, the signal vectors are
p
p
still given by (2.77) and (2.78) as s1 Eb x1 and s2 Eb x1 , where x1 1.
235
ci
di {0, 1}
PSK
modulator
DPSK
ci 1
1-bit
delay
Differential encoder
Figure 6.11
DPSK modulator.
si t Apt cos2fc t i
(6:24)
i i1 di mod 2
(6:25)
where p(t) is the pulse shape, di 2 f0; 1g represents the ith data bit, and i and i1 are the
phases of the ith bit and the previous (i 1)th bit, respectively. The recursive phase encoding
rule can be stated as follows:
(a) di 0, i i1 : no phase change between two successive waveforms when the
current data bit is 0.
(b) di 1, i i1 : a phase change of radians from the previous waveform to the
current waveform when the current data bit is 1.
Thus, in DPSK the previous data bit serves as a reference for the current data bit. The
recursive phase encoding rule in (6.25) can be implemented via a differential encoder shown
in Figure 6.11 as part of the DPSK modulator.
The data bit di is modulo-2 added to the previously differentially encoded bit ci1 to
provide the current differentially encoded bit ci, which is then PSK modulated. Hence,
ci di ci1
(6:26)
The operations in (6.25) and (6.26) are equivalent, that is, they produce the same DPSK
signal for a given data bit di, as seen in the following example.
di
ci
si(t)
0
1
2
3
4
5
6
0
0
2
2
3
3
1
0
1
1
0
1
0
1
1
0
1
1
0
0
236
Modulation
Note that bit c0 = d0 at the time index i = 0 is used as the initial reference bit.
"
1
X
(6:27)
i1
It is seen that DPSK is essentially PSK with the data bits di replaced by the differentially
encoded bits ci. Therefore, the power spectral density of a DPSK signal is the same as that of
the corresponding PSK signal.
A natural question arises concerning the use of differential encoding in PSK. In practice,
coherent detection of PSK requires the demodulator to extract the carrier from the PSK
signal. This process is done via a carrier recovery circuit that inherently introduces a phase
ambiguity of radians (180). That is, the recovered carrier is either cos 2fc t or
cos2fc t cos 2fc t. If data are demodulated with cos 2fc t, bit 0 becomes 1,
and vice versa. The use of differential encoding can resolve this phase ambiguity. Inverted
data bits can be corrected via a differential decoder at the output of the demodulator. In
addition to phase ambiguity resolution, differential encoding also allows DPSK to be
demodulated noncoherently without a reference carrier, a feature that is attractive in
applications where coherent demodulation is not feasible. Both phase ambiguity resolution
and noncoherent demodulation of DPSK will be examined in the next chapter.
1
0
Figure 6.12
Electrical
signal
Light
source
Optical ASK
237
current is biased to be near the threshold value for bit 0 to nearly turn off the emitted light,
and signicantly above the threshold for bit 1. It is customary not to bias the laser diode so
that the light during bit 0 is completely turned off to avoid the effect called transient chirp
that can appear as crosstalk to adjacent channels in a multi-channel system. The ratio of the
light energy for bit 0 to that for bit 1 is called the extinction ratio re . Note that 0 re 51.
The ASK signal that represents bit 1 or bit 0 over an arbitrary bit time Tb is given by
8
1
>
>
Apt cos 2fc t : bit 1
> s1 t p
<
1 re
(6:28)
st
r
>
re
>
>
: s2 t
Apt cos 2fc t : bit 0
1 re
where p(t) is the pulse shape. For an innite sequence of bits, the ASK signal can be written
as
"
#
p p
1
X
di 1 re re
p
st A
pt iTb cos 2fc t
(6:29)
1 re
i1
where di 2 f0; 1g represents the ith data bit. The complex envelope or equivalent lowpass
signal for the ASK signal is
p p
1
X
di 1 r e r e
p
(6:30)
m t
pt iTb
1 re
i1
For the case of zero extinction ratio and squared pulse shape, the power spectral density of
m(t) is that of the unipolar signaling given by (3.68) as
Sm f
Tb sin fTb 2
f
1
4
fTb
Tb
(6:31)
The power spectral density of the ASK signal can be obtained by substituting (6.31) into
(6.5). Again, note that the ASK signal is simply a DSB-AM signal with the message m(t) in
(6.30). The null-to-null bandwidth of the ASK signal with squared pulse is 2=Tb . A spectral
line represented by the ( f ) function in (6.31) indicated that a dc component exists at the
carrier frequency fc in its power spectral density.
Tb
0
s22 dt
A2 Tb
re
2 1 re
(6:33)
238
Modulation
x1
0
Figure 6.13
E1
E2
Eb
1
4
4
(6:34)
(6:35)
(6:36)
f1 fc
(6:37)
Both frequencies f1 and f2 are selected to be an integer multiple of the bit rate 1=Tb , and the
minimum frequency spacing of 1=Tb is required to preserve orthogonality between s1 t and
s2 t . The peak frequency deviation is f 1=2Tb .
Pulse shapes other than the squared pulse can be chosen to shape the power spectral
density but care must be exercised to ensure orthogonality between the two signals s1 t and
239
s2 t . From (6.36) and (6.37), the FSK signals for bits 1 and 0 can be combined into one
equation for an innite sequence of bits up to the nth bit as
"
#
t X
n
1
2di 1 p iTb d
(6:38)
st A cos 2fc t 2
2Tb 1 i1
where di 2 f0; 1g represents the ith data bit and pt iTb is the squared pulse shape of the
ith bit given in (2.26). The phase at the bit transition is required to be continuous, that is, at
the bit transition the phase of the present bit is required to be the same as that of the previous
bit. FSK is a special form of FM with the message signal m(t) given by
mt
n
X
2di 1 pt iTb
(6:39)
i1
1
2Tb
(6:40)
Note that the message signal m(t) is an antipodal signal with an innite bandwidth.
Therefore, it is not appropriate to use Carsons rule bandwidth or the FM modulation
index [26] unless we dene a bandwidth B for m(t) that contains, say over 90% of the
power. This yields B 1=Tb , which is the rst null bandwidth.
Carsons rule bandwidth can then be estimated as follows:
1
1
1
3
BC 2 f
(6:41)
2
Tb
2Tb Tb
Tb
The modulation index of the FSK signal can be evaluated as f =B f =1=Tb 0:5.
A digital modulation index h 2Tb f 1 is commonly used because it is independent of
the bandwidth of m(t).
Figures 6.14 and 6.15 show the FSK modulators for both discontinuous phase and
continuous phase at the bit transitions.
Switch
di {0,1}
FSK
A cos (2 f1t + 1)
A cos (2 f2t + 2)
Figure 6.14
240
Modulation
Antipodal
signal
generator
di {0,1}
Figure 6.15
FM
modulator
f = 1
2Tb
FSK
Modulator for continuous phase FSK with a frequency spacing equal to the data rate.
n1
X
t
2di 1
2dn 1 p nTb d
st A cos 2fc t
Tb nTb
i1
(6:42)
t
A cos 2fc t 2dn 1 n
Tb
where n is the phase of s(t) dened as
n
n1
X
2di 1 2dn 1n
(6:43)
i1
The requirement for continuous phase at the nth bit transition t nTb dictates that
2fc nTb 2dn 1n n 2fc nTb 2dn1 1n n1
(6:44)
In other words,
n n1 2dn1 dn n
n1 mod 2
(6:45)
The above recursive relation also dictates that n n1 1 mod 2 , where
1 is the initial phase at time t 1. This proves that n is independent of the data bits,
and hence, without loss of generality, we set 1 0 to yield n 0 mod 2 . Using this
result in (6.42) we obtain, for nTb t 5n 1Tb,
t
st A cos 2fc t 2dn 1
Tb
(6:46)
t
t
A cos 2dn 1
cos 2fc t A sin 2dn 1
sin 2fc t
Tb
Tb
Since dn 2 f0; 1g, hence 2dn 1 2 f1; 1g, and (6.46) reduces to
t
t
cos 2fc t A2dn 1 sin
sin 2fc t
st A cos
Tb
Tb
A Re sL t ej2fc t
(6:47)
241
where sL t is the complex envelope of s(t) and is given by, for nTb t 5n 1Tb,
t
t
j2dn 1 sin
(6:48)
sL t cos
Tb
Tb
The power spectral density S( f ) of s(t) can be evaluated as follows:
S f
A2
SL f f c SL f f c
4
(6:49)
where SL f is the power spectral density of sL t. Note that the in-phase (real) and
quadrature (imaginary) components of s(t) are independent, so SL f is the sum of the
power spectral densities of cost=Tb and 2dn 1 sint=Tb , respectively. Note that
cost=Tb is a periodic signal for 15t 5n 1Tb since it is independent of the data
bit, while 2dn 1 sint=Tb is an antipodal signal with a pulse shape sint=Tb of
duration Tb. The power spectrum of a periodic signal expressed in (2.112) is given in
(2.127). Thus, for the periodic in-phase signal cost=Tb with frequency 1=2Tb, the
power spectral density is
1
1
1
SL;I f f
f
(6:50)
4
2Tb
2Tb
The power spectral density of the antipodal quadrature signal 2dn 1 sint=Tb is
given by
2
1
t
(6:51)
pt nTb
SL;Q f F sin
Tb
Tb
where pt nTb is a unit amplitude pulse with duration Tb from nTb to (n + 1)Tb.
(6:52)
p t
2
2
:
2
0;
otherwise
Therefore, from Table 2.2, we get
2
2
sin Tb f
F cos t p t Tb
1 f 1 f 1
Tb
2
Tb
2Tb
2Tb
Tb f
2
Tb sin Tb f 1=2Tb sin Tb f 1=2Tb 2
Tb f 1=2Tb
Tb f 1=2Tb
2
"
#2
2T
b cos Tb f
(6:53)
2
2Tb f 1
242
Modulation
SL ( f )
2Tb
Figure 6.16
2Tb
1 0 1
2Tb 2Tb
f
3
2Tb
5
2Tb
n1=2
T
b
F sin
e
p
t
nT
b
Tb
2Tb f 2 1
(6:54)
By substituting (6.54) into (6.51), the power spectral density SL;Q f is given by
"
#2
4Tb cos Tb f
SL;Q f 2
(6:55)
2Tb f 2 1
From (6.50) and (6.51), we obtain the power spectral density of the complex envelope sL t
of the continuous phase FSK signal s(t), as follows:
"
#2
1
1
1
4Tb cos Tb f
(6:56)
SL f f
f
2
4
2Tb
2Tb
2Tb f 2 1
A plot of SL f is shown in Figure 6.16 where the rst null bandwidth is equal to 1:5=Tb .
The null-to-null bandwidth of a FSK signal is therefore equal to 3=Tb . This is the same as
Carsons rule bandwidth in (6.41). In practice, the bandwidth is normally chosen to be
between 1:25=Tb , which contains about 90% of the total power, and 2:1=Tb , which contains
about 99% of the signal power. The spectral efciency of FSK is, therefore, about
0.50.8 bits/s/Hz.
The FSK signal space is illustrated in Figure 2.14. Note that the Euclidean distance
p
p
between the two signals is 2Eb as compared to 2 Eb for PSK. The shorter Euclidean
distance results in an inferior performance in noise than PSK, as will be studied in Chapter 7.
243
n1
X
t
(6:57)
2di 1
2di 1p nTb d
A cos 2fc t
2 i1
2Tb nTb
t
A cos 2fc t 2dn 1
n
2Tb
where p(t) is a squared pulse shape of unit amplitude and duration Tb, and n is given by
"
#
n1
X
n
n
2di 1 2dn 1
(6:58)
2 i1
2
For the phase to be continuous at the nth bits transition t nTb , we must have
n
n
2fc nTb 2dn 1 n 2fc nTb 2dn1 1 n1
2
2
or
n n1 dn1 dn n
(6:59)
Thus, we get n n1 mod 2 for dn dn1, and n n1 nmod 2 for
dn 6 dn1 . Without loss of generality, we assume the initial phase at time t 1 is 0,
that is, 1 0. Then, n 2 f0; gmod 2 . By observing (6.57), we can represent the
data bit dn 2 f0; 1g as follows:
1
t n ;
s1 t A cos 2 fc
4Tb
1
s2 t A cos 2 fc
t n ;
4Tb
dn 1
(6:60)
dn 0
It is obvious that the peak frequency deviation is f 1=4Tb , which is half that of FSK.
This explains the name minimum shift keying. The digital modulation index is
k 2Tb f 0:5. Note that s1 t and s2 t are orthogonal if fc is an integer multiple of
1=Tb as indicated by (2.49), and they are quasi-orthogonal if fc 1=Tb .
t
t
st A cos n cos 2dn 1
cos 2fc t A cos n sin 2dn 1
sin 2fc t
2Tb
2Tb
A Re sL tej2fc t
(6:61)
244
Modulation
where cos n 2 f1; 1g, and sL(t) is the complex envelope of s(t) dened for nTb t
5n 1Tb as
t
t
sL t cos n cos 2dn 1
j cos n sin 2dn 1
2Tb
2Tb
(6:62)
t
t
j2dn 1 cos n sin
cos n cos
2Tb
2Tb
It is obvious that cos n and 2dn 1 cos n represent the data for the in-phase (real) and
quadrature (imaginary) parts of sL(t), respectively. From (6.59), we observe that
cos n cos n1 ;
cos n
cos n1;
cos n1;
dn dn1
even n
;
odd n
(6:63)
dn 6 dn1
(6:64)
Thus, cos n remains constant over a two-bit interval of 2Tb and can only change sign when
n is odd. Also,
2dn 1 cos n 2dn1 1 cos n1 ; dn dn1
2dn1 1 cos n1 :
even n
2dn 1 cos n
;
2dn1 1 cos n1 :
odd n
dn 6 dn1
(6:65)
Thus 2dn 1 cos n remains constant over a two-bit interval of 2Tb and can only change
sign when n is even.
The in-phase data cos n and quadrature data 2dn 1 cos n , for all n, therefore
represent two antipodal data streams of period 2Tb; one is offset from the other by Tb
duration.
dn
n mod 2
cos n
0
1
2
3
4
5
6
7
8
1
0
0
1
0
1
0
1
0
0
0
0
0
1
1
1
1
1
1
1
1
1
2dn 1 cos n
1
1
1
1
1
1
1
1
1
For convenience, the initial phase 0 is assumed to be 0. Note that the in-phase data stream
cos n is always constant during a 2Tb interval starting with odd n = 1, 3, 5, 7. On the
245
2dn1
Tb
(a)
cosn
Tb
(b)
(2dn1)cosn
t
0
Tb
(c)
t
cosn cos
2Tb
t
0
Tb
(d)
t
(2dn1)cosnsin
2Tb
t
0
Tb
(e)
Figure 6.17
(a) Input data in antipodal format of period Tb; (b) in-phase antipodal data of period 2Tb; (c) quadrature antipodal
data of period 2Tb; (d) in-phase antipodal signal of pulse shape cos(t/2Tb); (e) quadrature antipodal signal of pulse
shape sin (t/2Tb).
other hand, the quadrature data stream 2dn 1 cos n is constant over a 2Tb interval
starting with even n = 0, 2, 4, 6, 8. The two data streams are staggered by one bit interval.
Figure 6.17 illustrates the in-phase and quadrature antipodal data streams cos n and
2dn 1 cos n .
246
Modulation
The above discussion also concludes that cos n cost=2Tb is the in-phase antipodal
signal of pulse shape cost=2Tb of 2Tb seconds. Also, 2dn 1 cos n sint=2Tb is the
quadrature antipodal signal of pulse shape sint=2Tb of 2Tb seconds. These two signals
are staggered by Tb seconds. Note that sint=2Tb is simply cost=2Tb shifted by Tb
seconds.
The power spectral density SL( f ) of the complex envelope sL(t) in (6.62) is the sum of two
identical power spectral densities SL,I( f ) and SL,Q( f ) of the in-phase and quadrature
components, respectively. Following the analysis in Example 6.22, and noting that both
in-phase and quadrature components have a period of 2Tb seconds, we have
2
1
t
F cos
p2Tb t
SL;I f SL;Q f
2Tb
2Tb
where p2Tb t is a squared pulse dened as follows:
1; Tb t 5Tb
p2Tb t
0; otherwise
(6:66)
(6:67)
sin 2fTb
F cos t p2T t 1 f 1 f 1
2Tb
b
2
2fTb
2Tb
4Tb
4Tb
sin 2Tb f 1=4Tb sin 2Tb f 1=4Tb 2
T b
2Tb f 1=4Tb
2Tb f 1=4Tb (6:68)
"
#
4T cos 2T f 2
b
b
4Tb f 2 1
Substituting (6.68) into (6.66) and using the fact that SL f 2SL;I f 2SL;Q f , we obtain
"
#2
16Tb cos 2Tb f
SL f 2
4Tb f 2 1
(6:69)
Hence, the power spectral density of the MSK signal s(t) in (6.61) can be evaluated as
S f
A2
SL f fc SL f fc
4
(6:70)
Figure 6.18 shows the power spectral density of the complex envelope of an MSK signal.
The rst null bandwidth is 0.75/Tb. The 99% power bandwidth of the complex envelope is
about 1.2/Tb. On the other hand, 90% of the signal power is contained in the bandwidth
approximately equal to 0.78/Tb. Thus, the spectral efciency of MSK is between 0.8 and
1.3 bits/s/Hz, which is far superior than that of FSK. Also, for FSK, half the power is in the
two harmonics of fc 1=Tb . This results in a tremendous waste of power since these two
harmonics carry no information.
247
Figure 6.18
1.25
Tb
0.75
Tb
0.75
Tb
1.25
Tb
di {0,1}
Figure 6.19
FM
modulator
f = 1
4Tb
MSK
Modulator
The MSK signal in (6.57) can be implemented by an FM modulator with a peak frequency
deviation f 1=4Tb , as shown in Figure 6.19. This is referred to as the analog implementation of MSK.
The digital implementation of MSK can be carried out as follows. First, we observe that
the in-phase data cos n and the quadrature data 2dn 1 cos n can be written as one data
stream n as
cos n ;
n : odd
(6:71)
n
2dn 1 cos n ; n : even
In other word, the in-phase and quadrature data are odd and even splits of the data fn g.
Example 6.24 illustrates the above observation.
dn
2dn1
cos n
(2dn1)cos n
cn
0
1
2
3
4
5
6
7
8
1
0
0
1
0
1
0
1
0
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
1
0
1
1
0
0
1
1
0
248
Modulation
Tb
(a)
2n1
Tb
(b)
2n
t
0
Tb
(c)
Figure 6.20
(a) Data n; (b) odd split 2n1; (c) even split 2n.
The antipodal data fn g, and its even f2n g and odd f2n1 g splits are shown in Figure 6.20.
Note that f2n g is identical to f2dn 1 cos n g and f2n1 g is identical to fcos n g as seen
in Figure 6.17.
We note that (6.71) can be rewritten in a more compact form, that is,
n 2dn 1n1
(6:72)
as evident by Example 6.24. If we express the data in binary form, (6.72) becomes
cn dn cn1
(6:73)
Thus, the data cn can be obtained from the original data dn by differential encoding. A
quadrature implementation of MSK is illustrated in Figure 6.21.
Note that the data dn is clocked in at the bit rate of 1/Tb bits per second. On the other hand,
the data c2n and c2n1 are clocked at the rate of 1/2Tb bits per second.
A variation of MSK is also employed, namely, precoded MSK, where the differential
encoder from Figure 6.21 is omitted. This no longer guarantees that data bit dn = 1 will
be transmitted at fc 1=4Tb and data bit dn = 0 at fc 1=4Tb , hence noncoherent
249
t
cos
2Tb
dn {0,1}
cn {0,1}
dn
cos 2 fct
Antipodal
signal
generator
cn1
1 bit
delay
1 bit
delay
Antipodal
signal
generator
t
sin
2Tb
Figure 6.21
MSK
sin 2 fct
2Tb
cos 2 fct
Antipodal
signal
generator
dn {0,1}
1 bit
delay
Antipodal
signal
generator
t
sin
2Tb
Figure 6.22
Pre-coded
MSK
sin 2 fct
250
Modulation
Table 6.1 Occupied bandwidth normalized to 1/Tb for a given percentage of power
BTb
90%
99%
99.9%
0.2
0.25
0.5
1
0.52
0.57
0.69
0.78
0.79
0.86
1.04
1.20
0.99
1.09
1.33
2.76
GMSK
MSK
dn {0,1}
Figure 6.23
Antipodal
signal
generator
FM
modulator
f = 1
4Tb
Gaussian
filter
GMSK
2 2
(6:74)
(6:75)
a
B
2
B
2
Note that jH Bj2 exp0:592 0:5, the half-power value, as expected. The impulse
response of the Gaussian lter is
p
2 =a2 t2
h t
e
(6:76)
a
The signal g(t) at the output of the Gaussian lter is frequency-modulated to produce the
GMSK signal, which can be expressed as follows:
"
#
t X
n
st A cos 2fc t 2af
2di 1 g iTb d
(6:77)
1 i1
251
252
Modulation
n 1; 2; . . . ; M
(6:78)
where p(t) is the pulse shape of a symbol and cos 2fc t is the carrier. An is the signal
amplitude given by
An 2n M 1A
(6:79)
where A is the smallest signal amplitude. It is seen that 2A is the distance between adjacent
signal amplitudes. The symbol rate is taken to be 1/Ts, where Ts is the symbol time.
Signal space
From the analysis of Section 2.4, we observe that the basis function for the signal set sn(t) in
(6.78) is given by
A
x1 t p pt cos 2fc t
E
where E is the smallest symbol energy dened as
1
Apt cos 2fc t 2 dt
E
(6:80)
(6:81)
1
p
Thus, it is seen that the signal energy En of signal sn t 2n M 1 E x1 t can be
expressed as follows:
1
En
s2n t dt 2n M 12 E
(6:82)
1
The MASK signal space is shown in Figure 6.24 for the cases M = 4 and M = 8.
Each symbol d0 d1 dk1 is assigned a signal level via Gray code. Adjacent symbols
differ by only one bit, thus a symbol
error only results in one bit error. It is common to
p
normalize the signal level with E so that the mapping can be represented as in Tables 6.2
and 6.3.
253
Input bits d0 d1
Signal level = 2n M 1
00
01
11
10
3
1
+1
+3
1
2
3
4
Input bits d0 d1 d2
Signal level = 2n M 1
000
001
011
010
110
111
101
100
7
5
3
1
+1
+3
+5
+7
1
2
3
4
5
6
7
8
00
01
3 E
11
0
10
x1
3 E
(a) M = 4
000
7 E
001
011
5 E
010
3 E
110
111
101
100
3 E
5 E
7 E
x1
(b) M = 8
Figure 6.24
1
X
(6:83)
i1
where Di 2 f1; 2; . . . ; M g represents the ith symbol in the symbol stream. The complex
envelope or equivalent lowpass signal of s(t) is given by
m t
1
X
i1
2Di M 1 pt iTs
(6:84)
254
Modulation
k
k
2
P
f
Sm f j P f j
Ts
Ts k1 Ts
Ts
(6:85)
where P( f ) is the Fourier transform of the pulse shape p(t) and and 2 are the mean value
and variance of the signal levels f2Di M 1g, respectively. Since the symbol levels are
equally spaced and symmetrical around zero, the mean value = 0. The variance 2 can be
calculated as follows:
n
o
n
o
2 E 2Di M 12 E 4D2i 4Di M 1 M 12
o
4E D2i 4M 1EfDi g M 12
)
X
M
X
4 M 2 4
2
n M 1
n M 1
M n1
M
n1
o
2
M 12M 12M 12 M 12
3
1
M2 1
3
(6:86)
For example, we have 2 5 for 4ASK and 2 21 for 8ASK. Substituting (6.86) with
= 0 into (6.85) yields
Sm f
M2 1
jP f j2
3Ts
(6:87)
The MASK power spectral density is obtained by substituting (6.87) into (6.5). If a squared
pulse shape p(t) is employed, then the MASK power spectral density would look identical to
that of PSK with an amplitude scaling factor of M 2 1, and with Tb replaced by Ts
as illustrated in Figure 6.6 (recall that PSK is simply a DSB-AM signal with an antipodal signaling message). For the raised-cosine pulse shape, the bandwidth is
B 1 =Ts , where is the raised-cosine roll-off factor. Note that the symbol rate
1=Ts 1=kTb , where 2k M . Thus, in terms of the bit rate, 1=Tb , the bandwidth in this
case is B 1 =kTb . The spectral efciency, which is the bit rate-to-bandwidth ratio, is
r 1=BTb k=1 bits/s/Hz, which is between k/2 and k bits/s/Hz. Therefore, the
larger the number of bits per symbol k log2 M , the smaller the bandwidth required, and
the higher the spectral efciency that can be achieved. We observe that an increase in power
by a factor of M 2 1 accompanied a spectral efciency gain of k log2 M 5M 2 1.
Hence, it is obvious that MASK must use more transmitter power to achieve the same
p
performance as PSK, given the fact that the minimum Euclidean distance for MASK is 2 E,
which is the same as that of PSK.
255
{0,1}
k1
dk1
Symbol
mapping
Pulse
shaping
MASK
d0
cos 2 fct
Figure 6.25
MASK modulator.
Modulator
The MASK modulator is illustrated in Figure 6.25. The binary data in bits are shifted
with rate 1=Tb into a serial-to-parallel converter (demultiplexer) whose k outputs form a
stream of k-bit symbols with rate 1=Ts 1=kTb . Each symbol d0 d1 dk1 is then mapped
into its appropriate signal level. The pulse shape generator provides a pulse p(t) with the
appropriate signal level, which then modulates a carrier cos 2fc t to produce the MASK
signal.
n 1; 2; . . . ; M
(6:88)
where p(t) is the pulse shape of a symbol, cos 2fc t is the carrier, and A is the constant
amplitude. The symbol rate is taken to be 1/Ts, where Ts is the symbol time, which is k times
the bit interval Tb. The symbol rate is thus equal to 1/k times the bit rate 1/Tb. The carrier
frequency is assumed to be an integer multiple of the symbol rate.
256
Modulation
Signal space
The MPSK signals in (6.88) can be expressed in the I-Q representation as
2n 1
2n 1
sn t cos
Apt cos 2fc t sin
Apt sin 2fc t
M
M
(6:89)
Simply observing (6.89), we conclude that there are two basis functions for the signal set
sn t , n 1; 2; . . . ; M , namely,
A
x1 t p pt cos 2fc t
Es
(6:90)
A
x2 t p pt sin 2fc t
Es
(6:91)
1
1
(6:92)
1
In the cases where p(t) is a unit amplitude pulse of duration Ts, the basis functions x1 t and
x2 t form an orthonormal set per (2.87).
1
A2 1
x1 tx2 t dt
ptpt sin 4fc t dt
(6:93)
2Es 1
1
Applying the Parseval formula in Table 2.2 to the right-hand side of (6.93), we get
1
1
A2 1
P f P f 2fc P f 2fc df
j4Es 1
A2 1
P f Pf 2fc Pf 2fc df
j4Es 1
x1 tx2 t dt
(6:94)
If P( f ) is bandlimited to a spectrum W such that W < fc, then P( f ) does not overlap with
Pf 2fc or Pf 2fc . Therefore, P f Pf 2fc 0 and P f Pf 2fc 0.
Thus, the right-hand side of (6.94) is zero and x1(t) is orthogonal to x2(t). Note that this
condition is sufcient only.
A stronger condition can be established by letting P f Pf , that is, P( f ) is an even
function, a condition that is normally met by practical pulse shapes. In this case,
257
Input bits d0 d1
1
2
3
4
I
p
p2=2
p2=2
p2=2
2=2
11
01
00
10
x2
01
Q
p
p2=2
p2=2
p2=2
2=2
x2
010
11
110
011
111
x1
Es
x1
001
10
00
Es
101
000
(a)
Figure 6.26
100
(b)
Signal space: (a) QPSK; (b) 8PSK. The I- and Q-channels are represented by x1 and x2, respectively.
1
1
P f Pf 2fc df
1
1
P f Pf 2fc df
(6:95)
Using the basis functions x1(t) and x2(t) in (6.90) and (6.91), we can express the MPSK
signal sn(t) in (6.89) as follows:
s n t
p
p
2n 1
2n 1
x1 t Es sin
x2 t
Es cos
M
M
p
p
2n 1
2n 1
Es cos 2n1
M
sn p
x1 Es sin
x2
Es cos
2n1
M
M
Es sin M
(6:96)
(6:97)
258
Modulation
Input bits d0 d1 d2
111
110
010
011
001
000
100
101
cos =8
sin =8
sin =8
cos =8
cos =8
sin =8
sin =8
cos =8
sin =8
cos =8
cos =8
sin =8
sin =8
cos =8
cos =8
sin =8
1
2
3
4
5
6
7
8
1
X
2Di 1
Apt iTs cos 2fc t
M
i1
1
X
2Di 1
Apt iTs sin 2fc t
sin
M
i1
cos
(6:98)
where Di 2 f1; 2; . . . ; M g represents the ith symbol in the symbol stream. Let sL t be the
complex envelope of s(t), then
st A Re sL t ej2fc t
(6:99)
where
sL t
1
X
i1
cos
1
X
2Di 1
2Di 1
pt iTs j
pt iTs
sin
M
M
i1
(6:100)
The power spectral density S( f ) of the MPSK signal s(t) can be evaluated as
S f
A2
SL f fc SL f fc
4
(6:101)
where SL f is the power spectral density of the complex envelope sL t , which is the sum of
the power spectral density SL;I f of the in-phase (real) component and the power spectral
density SL;Q f of the quadrature (imaginary) component of sL t . Both SL;I f and SL;Q f
can be evaluated according to the expression (6.85), which is displayed here for clarity:
2 X
1 2
2I
I
2
P k f k
(6:102)
SL;I f jP f j
Ts
Ts
Ts k1 Ts
259
2 X
1 2
2Q
Q
k
k
2
SL;Q f
P
f
j P f j
Ts
Ts k1 Ts
Ts
(6:103)
where P( f ) is the Fourier transform of the pulse shape p(t), I and Q are the mean values of
the M/2 signal levels cos2Di 1=M and sin2Di 1=M , respectively, and 2I and 2Q are
the corresponding variances. Since the M/2 signal levels are symmetrical around zero,
I Q 0. With each of the M/2 signal levels assumes a probability of 2/M, the variances
2I and 2Q are
2I
2X
2n 1 1
cos2
M n1
M
2
(6:104)
2Q
2 X 2 2n 1 1
sin
M n1
M
2
(6:105)
M =2
M =2
jP f j2
2Ts
(6:106)
j P f j 2
Ts
(6:107)
Consequently, the MPSK power spectral density is obtained by substituting (6.107) into
(6.101). Figure 6.27 shows the power spectral density of an MPSK signal with unit
S( f )
f
1 fc
fc
Ts
1
fc+
Ts
fc
1
fc
Ts
1
fc +
Ts
(a)
S( f )
fc
1+
2Ts
fc
fc+
1+
2Ts
fc
1+
2Ts
fc
(b)
Figure 6.27
Power spectral density of MPSK: (a) squared pulse shape; (b) raised-cosine pulse shape.
fc+
1+
2Ts
260
Modulation
amplitude squared pulse and with raised-cosine pulse. The latter has P( f ) given in (6.17)
with Ts replacing Tb, and the former has jP f j jTs sin fTs =fTs j (see Table 2.2). It is
clear from (6.87) and (6.107) that MPSK and MASK have the same spectral efciency,
namely, r k=1 bits/s/Hz for a raised-cosine pulse shape, which is between k/2 and
k bits/s/Hz.
From the power efciency point of view we observe that the minimum Euclidean distance
p
(or minimum signal separation) for MPSK is 2 Es sin =M per (2.91), which gets smaller
with increasing M. This in turn requires more power for a larger M. Obviously, MPSK is a
modulation that trades power efciency for bandwidth efciency. Of particular interest is
QPSK, thepmost
used digital modulation. The minimum Euclidean distance for
widely
p
QPSK is 2 Es =2 2 Eb , since Es 2Eb , where Eb is the bit energy. Thus, QPSK has the
same minimum Euclidean distance as PSK, suggesting that QPSK is as power efcient as
PSK. This is reinforced by the fact that a QPSK signal is a superposition of two orthogonal
PSK signals as viewed by (6.96). Thus, if QPSK can be detected as two orthogonal PSK
signals, the performance would be identical to PSK. On the other hand, the spectral
efciency of QPSK is twice that of PSK. This is certainly a rare combination.
Modulator
MPSK can be implemented by the I-Q modulator shown in Figure 6.28. The binary data in bits
are shifted with rate 1/Tb into a serial-to-parallel converter (demultiplexer). The k outputs form
a symbol with the symbol rate 1=Ts 1=kTb . Each symbol d0 d1 dk1 is mapped into
the I- and Q-levels that determine the amplitudes of the I- and Q-pulse shape p(t). The
I-pulse modulates the in-phase carrier cos 2fc t, while the Q-pulse modulates the quadrature
carrier sin 2fc t. The two signals are added to form the MPSK signal.
{0,1}
k1
Pulse
shaping
dk1
Symbol
mapping
d0
cos 2 fct
Pulse
shaping
sin 2 fct
Figure 6.28
MPSK
261
{0,1}
Pulse
shaping
Symbol
mapping
cos 2 fct
1 bit
delay
OQPSK
Pulse
shaping
sin 2 fct
Figure 6.29
OQPSK modulator.
QPSK spectrum to regrow even if the signal is pre-ltered to suppress the sidelobes. This
can cause interference to adjacent channels. Since post-ltering at the output of an RF power
amplier is not practical due to a high lter insertion loss (the lter bandwidth is normally
very small compared to the carrier frequency), OQPSK might be employed to reduce
sidelobe regrowth. The staggering of I and Q components implies that only one bit in a
symbol can be succeeded by the next bit. If Gray code is used in symbol mapping, the phase
changes can only occur with 0 and 90. Thus, the phase change of 180 is eliminated and
the sidelobe regrowth can be reduced. The OQPSK modulator is shown in Figure 6.29. Note
the similarity between OQPSK modulator and pre-coded MSK modulator in Figure 6.22.
Pre-coded MSK can be viewed as OQPSK with sinusoidal weighted pulse shapes.
di
1
0
1
1
0
0
0
1
1
1
Q
+1
+1
+1
+1
1
1
1
1
+1
+1
1
1
+1
+1
1
1
+1
+1
+1
+1
QOQPSK
QPSK phase
OQPSK phase
7/4
1
1
+1
+1
1
1
+1
+1
+1
+1
/4
5/4
3/4
/4
7/4
7/4
/4
3/4
5/4
5/4
3/4
/4
/4
262
Modulation
I
t
0
Tb
Ts
(a)
Q
t
0
(b)
QOQPSK
t
0
(c)
Figure 6.30
Note that QPSK can change phase every symbol time, while OQPSK can only change phase every bit
time (from the result of a change in either the I or Q bit, but not both). Despite switching phase every bit
interval, OQPSK is still a superposition of two orthogonal PSK signals. Therefore, its power spectral
density is still the same as that of QPSK.
263
i i1
00
10
11
01
0
/2
3/2
1
2
3
4
{0,1}
dI,i
dQ,i
Pulse
shaping
cI,i
Differential
encoder
Symbol
mapping
cQ,i
cos 2 fct
DQPSK
Pulse
shaping
sin 2 fct
Figure 6.31
DQPSK modulator.
st Apt cos2fc t i
i i1 Di 1
2
;
M
Di 2 f1; 2; . . . ; M g
(6:108)
(6:109)
where p(t) is the pulse shape, Di represents the ith symbol in the data stream, and i and i1
are the phases of the ith symbol and the previous (i 1)th symbol, respectively. Note that in
the case of DPSK (M = 2) we have Di 2 f1; 2g, and Di 1 di 2 f0; 1g is simply the ith
data bit, per (6.25). Table 6.6 illustrates the differential encoding in (6.109) for DQPSK,
which, in practice, can also be implemented via the following logical expressions:
cI;i dI;i dQ;i cI;i1 dQ;i dI;i dQ;i dI;i cQ;i1
(6:110)
cQ;i dI ;i dQ;i dI;i cQ;i1 dI;i dQ;i cI;i1 dQ;i
(6:111)
where the subscripts (I, i) and (Q, i) represent the ith bit in the in-phase and quadrature bit
streams, respectively. Figure 6.31 shows a DQPSK modulator which consists of a differential encoder and a QPSK modulator.
264
Modulation
Di
dI,i
dQ,i
0
1
2
3
4
1
4
2
1
3
0
0
1
0
1
0
1
0
0
1
cI,i
cQ,i
i i1 (mod 2)
0
0
0
0
1
0
1
0
0
1
5/4
3/4
5/4
5/4
/4
3/2
/2
0
(6:112)
cQ;i dI ;i dQ;i dI;i cQ;i1 dI;i dQ;i dI ;i cI;i1
(6:113)
Di
dI,i
dQ,i
0
1
2
3
4
5
6
7
4
3
2
3
2
4
3
1
1
1
0
1
0
1
1
0
1
0
1
0
1
1
0
0
cI,i
cQ,i
i i1 (mod 2)
1
1
1
1
1
0
0
0
1
0
1
0
1
0
1
1
/4
7/4
/4
7/4
/4
5/4
3/4
3/4
3/2
/2
3/2
/2
3/2
0
265
i i1
00
01
10
11
0
/2
3/2
1
2
3
4
Es cos i x1 t
Es sin i x2 t
(6:114)
Ii x1 t Qi x2 t
where Es is the symbol energy, x1 t and x2 t are the two basis functions in (6.90) and
(6.91), and Ii and Qi are the ith symbols I-level and Q-level, respectively, which are given by
p
(6:115)
Ii Es cos i
Qi
p
Es sin i
(6:116)
Substituting (6.109) into (6.115) and (6.116), we obtain the direct symbol mapping rule
where Ii1
(6:117)
(6:118)
p
p
Es cos i1 and Qi1 Es sin i1 ; i i i1 Di 12=M :
Modulator
Figure 6.32 presents two implementations of DMPSK, one via differential encoding then
symbol mapping, and one via direct symbol mapping.
DMPSK belongs to the family of MPSK; therefore, it has the same power spectral density
as MPSK.
266
Modulation
{0,1}
k1
dk1, j
Ii
ck 1, j
Differential
encoder
d0,i
Pulse
shaping
Symbol
mapping
cos 2 fct
c0,i
Qi
sin 2 fct
{0,1}
Ii
dk1, i
Pulse
shaping
cos 2 fct
d0,i
DMPSK
Pulse
shaping
(a)
k1
Qi
DMPSK
Pulse
shaping
sin 2 fct
(b)
Figure 6.32
(a) DMPSK modulator via separate differential encoding and symbol mapping for M = 4. (b) DMPSK modulator
via direct symbol mapping with i = (Di1) 2/M.
i i1 2Di 1 ;
4
Di 2 f1; 2; 3; 4g
(6:119)
(6:120)
6.13 /4-DQPSK
267
i = i i1
00
01
11
10
/4
3/4
5/4
7/4
1
2
3
4
X2
(Q)
X2
X1
(Q)
(I)
X1
(a)
(I)
(b)
X2
(Q)
X1
(I)
(c)
Figure 6.33
(a) Signal space for even symbol; (b) signal space for odd symbol; (c) signal space for two consecutive symbols.
where pt is the pulse shape, Di represents the ith symbol in the data stream, and i and i1
are the phases of the ith symbol and the previous (i 1)th symbol, respectively. The
differential encoding rule for /4-DQPSK is shown in Table 6.8.
Signal space
The study of DQPSK indicates that a phase change in multiples of /2 radians or 90 occurs
between two consecutive symbols. Thus, the signal space for DQPSK is the same as that of
QPSK (Figure 6.26(a)). On the other hand, a phase change in odd-multiples of /4 radians or
45 occurs in consecutive symbols in /4-DQPSK. This forces consecutive symbols to
alternate between two signal spaces as shown in Figure 6.33(a) for the ith symbol in the
268
Modulation
Ii
{0,1}
dI, i
dQ, i
Pulse
shaping
cos 2 fct
/4-DQPSK
Pulse
shaping
sin 2 fct
Figure 6.34
/4-DQPSK modulator.
symbol stream where the time index i is even (even symbols) and Figure 6.33(b) for odd
symbols. The superposition of the even and odd symbol signal spaces in Figure 6.33(c)
provides a single signal space for any two consecutive symbols.
The
levels for both I- and Q-channels can now take on ve values, namely,
psignal
0,p 2=2, 1 (for QPSK and DQPSK, there are only two I- and Q-values, namely,
2=2).
(6:122)
6.13 /4-DQPSK
269
p p pp
1 1
2
2
2
2
1
2 2
2
2
2
2
pp p p
3
3
1 1
2
2
2
2
Q1 I0 sin Q0 cos
0
4
4
2 2
2
2
2
2
3
3
2. I1 I0 cos Q0 sin
4
4
p
p p
2
2
2
0
2
2
2
p
p
p
2
2
2
7
7
Q2 I1 sin Q1 cos 1
0
2
2
2
4
4
pp pp
1 1
2
2
2
2
1
4. I3 I2 cos Q2 sin
4
4
2 2
2
2
2
2
pp pp
1 1
2
2
2
2
Q3 I2 sin Q2 cos
0
4
4
2 2
2
2
2
2
p
p
p
5
5
2
2
2
0
5. I4 I3 cos Q3 sin 1
4
4
2
2
2
p
p
p
5
5
2
2
2
Q4 I3 sin Q3 cos 1
0
4
4
2
2
2
7
7
3. I2 I1 cos Q1 sin 1
4
4
Di
dI,i
dQ,i
i = i i1 =(2Di 1)
/4 (mod 2)
0
1
2
3
4
3
2
4
1
3
1
0
1
0
1
1
1
0
0
1
5/4
0
7/4
0
5/4
5/4
3/4
7/4
/4
5/4
Ii
p
2=2
1p
2=2
1 p
2=2
Qi
p
2=2
0 p
2=2
0 p
2=2
270
Modulation
n 1; 2; . . . ; M
(6:123)
where p(t) is the pulse shape of a symbol, cos 2fc t is the carrier, and An and n represent the
amplitude and phase of the nth symbol, respectively. The symbol time is denoted as Ts,
hence the symbol rate is 1/Ts, which is 1/k times the bit rate 1/Tb, where Tb is the bit time
Tb Ts =k .
Signal space
The MQAM signal can be expressed in the I-Q representation as
st cos n An pt cos 2fc t sin n An pt sin 2fc t
n cos n Apt cos 2fc t n sin n Apt sin 2fc t
(6:124)
where n An =A is the normalized signal amplitude, and A is the smallest signal amplitude.
The two orthonormal basis functions x1 t and x2 t are given as
A
x1 t p pt cos 2fc t
E
(6:125)
A
x2 t p pt sin 2fc t
E
(6:126)
1
(6:127)
271
6.14 MQAM
The orthogonality of x1 t and x2 t was given in (6.93)(6.95). Thus, it is seen that the
signal energy En of a particular signal sn t n Apt cos2fc t n is
1
En
s2n t dt 2n E
(6:128)
1
(6:129)
p
p
n cos n
E
p
E n cos n x1 E n sin n x2
sn
E n sin n
(6:130)
where x1 and x2 are the two-dimensional basis vectors given in (2.60). Equation (6.130)
completely describes the signal space of MQAM. Figure 6.35 shows some popular signal
spaces for MQAM. Both rectangular 16QAM and 64QAM are specied in the WLAN IEEE
802.11a,g standards [7,8] to be used as subcarrier modulation in orthogonal frequency
division multiplexing (OFDM) (to be studied in Section 6.18).
Each symbol is assigned an I-level and a Q-level via Gray code, that is, adjacent symbols
differ by only one bit. For
MQAM signal space,
the signal levels are
p
p aprectangular
commonly
normalized
to
E
=
2
so
that
the
I-level
is
I
cos
n and the Q-level
n
n
p
is Qn 2n sin n for the symbol represented by signal sn.
Tables 6.9 and 6.10 show the symbol mapping for rectangular 16QAM and 64QAM. For
16QAM, the rst two bits d0 d1 of the symbol d0 d1 d2 d3 determine the I-value, and the last
two bits d2 d3 determine the Q-value. For 64QAM, the rst three bits d0 d1 d2 of the symbol
d0 d1 d2 d3 d4 d5 determine the I-value and the last three bits d3 d4 d5 determine the Q-value.
Figure 6.36 illustrates the symbol mapping for 16QAM in Table 6.9. This procedure can
be generalized to any rectangular MQAM with M = 22m.
1
X
i1
1
X
i1
(6:131)
where Di 2 f1; 2; . . . ; M g represents the ith symbol in the symbol stream. The I-Q
representation in (6.131) can be transformed into the complex envelope representation as
A
st p Re sL t ej2fc t
2
where the complex envelope sL t is given by
(6:132)
272
Modulation
M=8
(a)
Input bits d2 d3
3
1
+1
+3
00
01
11
10
3
1
+1
+3
M = 16
M = 16
(b)
(c)
M = 256
M = 128
M = 64
M = 32
M = 16
M=4
(d)
Figure 6.35
Various signal spaces for MQAM. The rectangular one is the most popular.
273
6.14 MQAM
Input bits d3 d4 d5
7
5
3
1
+1
+3
+5
+7
000
001
011
010
110
111
101
100
7
5
3
1
+1
+3
+5
+7
Q
0010
0110
1110
1010
1111
1011
+1
+3
1101
1001
1100
1000
+3
0011
3
0001
0111
+1
0101
1
0000
0100
3
Figure 6.36
1 p
1 p
X
X
2Di cos Di pt iTs j
2Di sin Di pt iTs
i1
(6:133)
i1
The power spectral density of the MQAM signal s(t) can be evaluated by (6.101). Following
the procedure in Section 6.11, as indicated by (6.102) and (6.103), we need to evaluate
I , Q , 2I , and 2Q . For symmetric signal spaces, we have I Q 0. The variances 2I
and 2Q depend on particular
signal spaces. For a rectangular
MQAM with M = 22m signal
p
p
points, there are
I-levels and M normalized Q-levels. These are
normalized
pM
1; 3; . . . ; M 1 . Thus,
n o
n
o
2I E ID2 i E 22Di cos2 Di
p
2
2
M 1
p 12 32
(6:134)
M
1
M 1
3
Similarly,
274
Modulation
1
s 2Q M 1
3
(6:135)
M 1jP f j2
3Ts
(6:136)
2M 1jP f j2
3Ts
(6:137)
The MQAM power spectral density can be obtained by substituting SL f into (6.101) and
by noting that A is the smallest signal amplitude. By comparing (6.137) to (6.107) and
(6.87), we conclude that MQAM, MPSK, and MASK have the same spectral efciency.
Their power spectral densities have the same shape but with different amplitudes. MQAM
can be generated by the I-Q generator shown in Figure 6.28 for MPSK. The only difference
is that the symbol mapping for MQAM generates non-constant symbol amplitudes.
275
Q
II
I
1011
1001
0010
0011
0000
0001
+3
1010
1000
1101
1100
+1
+1
+3
0100
0110
0101
0111
1
1111
1110
3
III
Figure 6.37
IV
Message symbol
Sector
Encoded symbol
0000
1101
1001
1100
0011
I
III
IV
II
II
0000
1101
0101
1000
1011
1
3
1
1
3
1
1
3
1
3
276
Modulation
will be presented in this section, and M-ary frequency shift keying, which will be addressed
in Section 6.16.
CSK employs a set of M = 2k orthogonal sinusoidal Walsh functions to represent a set of M
distinct k-bit symbols
sn t Awn t cos 2fc t;
n 1; 2; . . . ; M
(6:138)
where wn t is the Walsh function in (2.52), A is the amplitude, and cos 2fc t is the carrier.
The Walsh chip rate is 1/Tc, where Tc is the Walsh chip time. Since there are M Walsh chips
per symbol, Tc Ts =M , where Ts is the symbol time. Furthermore, Ts kTb , where Tb is
the bit time; therefore, the Walsh chip rate can be related to the bit rate as
1=Tc 1=Tb M =k.
1
X
(6:139)
i1
where Di 2 f1; 2; . . . ; M g represent the ith symbol in the symbol stream and pt is a
squared pulse of unit amplitude and duration Ts. If we substitute (2.52) into (6.138), we
obtain a stream of PSK waveforms clocking at the chip rate 1=Tc . Therefore, the power
spectral density of CSK is identical to that of PSK or DSB-AM given in (6.7), with Tc
replacing Tb. Note that 1=Tc 1=Tb M =k. Thus, CSK bandwidth is M/k times PSK
bandwidth for a given bit rate.
Modulator
Generating CSK signals is quite straightforward from (6.139). The incoming bits are
demultiplexed to form symbols, which are mapped to corresponding Walsh functions.
The Walsh functions bi-phase modulate a carrier to form a CSK signal. The CSK modulator
is shown in Figure 6.38.
{0,1}
k1
Walsh
mapping
CSK
0
cos 2 fct
Figure 6.38
CSK modulator.
277
n 1; 2; . . . ; M
(6:140)
where for a given positive integer 1, symbol rate 1=Ts , and fc =Ts ,
fn fc 2n M 1
1
2Ts
(6:141)
The minimum frequency spacing is 1=Ts to preserve the orthogonality for the entire n
sinusoids, as proved in (2.45). Note that the phase does not need to be continuous at the
symbol transitions. For this case, the bandwidth of MFSK is approximately M 1=Ts.
The bandwidth can be made narrower if the phase is forced to be continuous at the symbol
transitions. This implies that a continuous phase MFSK (CP-MFSK) signal must look like
an FM signal with digital modulated data, and can take the following form for an Nth data
symbol:
t X
N
1
st A cos 2fc t 2
2Di M 1p iTs d
4Ts 1 i1
(6:142)
where Di 2 f1; 2; . . . ; M g is the ith symbol in the symbol stream, and p(t) is the squared
pulse of unit amplitude and duration Ts. Following the derivation in (6.57), we can write s(t)
from (6.142) in a more compact form for the Nth data symbol:
278
Modulation
st A cos 2fc t
2DN M 1t NTs f N
2Ts
(6:143)
where f N is dened as
fN
1
NX
2Di M 1
2 i1
(6:144)
Let h 2Ts f be dened as the digital modulation index, where f is the peak frequency
deviation of s(t). Since f 1=4Ts per (6.142), it is seen that h = 0.5. Using h we can write
s(t) in the form
st A cos2fc t f t; DN
(6:145)
(6:146)
2Ts
q t
p d
(6:147)
CP-MFSK is a special case of continuous phase modulation (CPM), which will be studied in
Section 6.17.
By observing (6.142), we see that the message signal m(t) for the CP-MFSK signal s(t) is
simply an MPAM with amplitudes in the set f1; 3; . . . ; M 1g as seen from (6.84):
m t
N
X
(6:148)
i1
(6:150)
279
h=1
h = 0.5
0.3
0.3
0.2
0.2
0.1
0.1
fTs
0
0.5
1.5
fTs
0.5
(a)
Figure 6.39
1.5
(b)
{0,1}
Switch
MFSK
1
f1
fM
Figure 6.40
Bij f
cos 2fTs vij u cos vij
1 u2 2u cos 2fTs
vij h1 j M 1
u
sin M h
M sin h
(6:151)
(6:152)
(6:153)
By using h = 0.5, we can obtain the power spectral density for CP-MFSK. For h = 1, there
are impulses occurring at M frequencies. Figure 6.39 shows the power spectral densities for
M = 4 for the cases h = 0.5 and h = 1 as a function of the normalized frequency fTs and
normalized amplitude for fTs 40 only. The origin corresponds to the carrier frequency fc. For
h = 0.5, the bandwidth is approximately M =2Ts 2=Ts, while for h = 1, it is about
M =Ts 4=Ts .
Modulator
The MFSK signal can be implemented with independent carriers as in Figure 6.40, or using
an FM modulator to generate a continuous phase signal as in Figure 6.41.
280
Modulation
Waveform
1
0 t LTs
otherwise
1
2t
1
cos
LTs ; 0 t LTs
pt L
0;
otherwise
h
i
h
i
t T2s Q p2B
t T2s ; 0 BTs 1
pt Q p2B
ln 2
ln 2
1 1 x2 =2
p
Q x x 2 e
dx
pt
L-REC
L-RC
GMSK
L;
0;
(
{0,1}
FM
modulator
MPAM
generator
Figure 6.41
f =
1
4Ts
CP-MFSK
Modulator for CP-MFSK with a minimum frequency spacing equal to half the symbol rate or twice the peak frequency
deviation f.
q t
1
2Ts
p d
(6:154)
The symbol pulse shape p(t) can take on various shapes, such as a rectangular pulse of
duration L symbols, which is called L-REC, a raised cosine pulse shape of duration L
symbols, which is called L-RC, and a Gaussian minimum shift keying pulse, which is called
GMSK. Table 6.11 shows the equations that represent these common pulse shapes, which
are also plotted in Figure 6.42. The CPM signal is commonly referred to as full response
CPM when pt 0 for t 4Ts and partial response CPM when pt 6 0 for t 4Ts. For
GMSK pulse shapes, the parameter B in its equation represents the 3 dB bandwidth of the
Gaussian pulse dened in (6.75).
281
p(t)
p(t)
1
0.5
0
Ts
2Ts
(b)
(a)
p(t)
p(t)
2
Ts
2Ts
Ts
(d)
(c)
p(t)
BTs = 1
BTs = 0.2
0.55
t
3Ts
Ts
Ts
3Ts
(e)
Figure 6.42
(a) 1-REC; (b) 2-REC; (c) 1-RC; (d) 2-RC; (e) GMSK with BTs = 0.2 and BTs = 1.
282
Modulation
{0,1}
MPAM
generator
Figure 6.43
FM
modulator
Symbol
pulse
shaping
p(t)
f =
h
2Ts
CPM
CPM modulator.
{0,1}
Modulator 1
f1
OFDM
Modulator N
fN
Figure 6.44
Modulator
The CPM signal can be implemented with an FM modulator and a pulse shaping network as
illustrated in Figure 6.43. Note that the smaller the digital modulation index h, the smaller
the peak frequency deviation f, which results in a decrease in bandwidth.
283
6.18 OFDM
orthogonal with the minimum frequency spacing of 1/Ts, which is the subcarrier symbol
rate. For example, if PSK is the modulation technique for the subcarriers, then,
1=Ts 1=NT . On the other hand, if QPSK is used, then 1=Ts 1=2NT . For MQAM, we
would have 1=Ts 1=NT log2 M . The OFDM signal in Figure 6.44 is simply the sum of N
modulated subcarriers. When the number of subcarriers is large, the implementation in
Figure 6.44 is not practical because of a large number of modulators. In the following
discussion we investigate the implementation of OFDM at baseband, which is used in
practical applications.
zk t Z k Ik jQk ;
(6:156)
where Ik and Qk, the in-phase and quadrature components of zk t , respectively, are the I- and
Q-values of the kth subcarrier symbol.
For scaling purposes, we normalize the OFDM signal to N, the number of subcarriers, so
that
st
N 1
1X
sk t ;
N k0
0 t 5T s
(6:157)
1
n
o
1 NX
Re Z k ej2 fc kf t
N k0
1
1 NX
Re Z k ej2kft ej2fc t
N k0
(
!
)
N 1
1X
j2kft
j2fc t
Z k e
Re
e
;
N k0
(6:158)
0 t 5Ts
N 1
1X
Z k ej2kft ;
N k0
0 t 5Ts
(6:159)
284
Modulation
Consider a sampling rate equal to N times the subcarrier symbol rate, that is, the sampling
time is simply Ts =N. Applying the sampling process to (6.159), we obtain the nth sample at
time t nTs =N as follows:
sL
nTs
N
N 1
1X
Zk ej2kfnTs =N ; n 0; 1; . . . ; N 1
N k0
(6:160)
N 1
1X
Zk ej2kn=N ;
N k0
n 0; 1; . . . ; N 1
(6:161)
IDFTfZkg
Equation (6.161) allows us to generate the sequence sL n from the frequency samples
Z k Ik jQk ; k 0; 1; . . . ; N 1, which consists of the I- and Q-values of N subcarrier symbols. This relation is referred to as the N-point inverse discrete Fourier transform
(IDFT) of Z k [12]. In turn, the inverse operation that allows the frequency samples fZkg
to be obtained from the sequence sL n is called the N-point discrete Fourier transform
(DFT) and is given by
Z k
N1
X
sL n ej2kn=N ;
k 0; 1; . . . ; N 1
n0
(6:162)
DFTfsL ng
N 1
X
sL n ej2kn=N
n0
N 1 X
N 1
1X
Zm ej2nmk=N
N n0 m0
"
#
N1
N1
X
1X
j2nmk=N
Zm
e
N m0
n0
(6:163)
Zk
The right-hand side of the above equation reduces to Z(k) by virtue of the following identity:
N 1
X
n0
j2nmk=N
N;
0;
m k rN ; r is an integer
otherwise
(6:164)
285
6.18 OFDM
cos 2 fct
{0, 1}
Z(N1)
N1
Subcarrier
symbol
mapping
{Z(k)}
sL(N1)
Guard
interval
addition
IFFT
Z(0)
I
Symbol
pulse
shaping
(D/A
converter)
sL(0)
OFDM
Q
sin 2 fct
Figure 6.45
N 1
X
sL npt nTs =N
(6:165)
n0
where p(t) is the interpolation pulse shape. The pulse shape p(t) can have the form sin x=x,
which is used for reconstruction of a sampled signal (see Section 2.10). Note the similarity
of (6.165) and the NyquistShannon interpolation formula (2.247).
286
Modulation
^sL t
N 1
X
sL npt nTs =N
n0
N 1 X
N 1
1X
Zk ej2kn=N pt nTs =N
N n0 k0
N 1
N 1
X
1X
Zk
ej2kn=N pt nTs =N
N k0
n0
(6:166a)
N 1
1 X
p
Zkgk t
N k0
We can choose p(t) so that the N pulses fgk tg form a set of orthonormal pulses and they
also satisfy the Nyquist criterion for zero ISI (Chapter 9). We have
N 1
1 X
ej2kn=N pt nTs =N ; k 0; 1; . . . ; N 1
gk t p
N n0
r
N sinNt=Ts
pt
Ts Nt=Ts
(6:166b)
(6:166c)
The above result shows that an OFDM symbol is indeed a normalized sum of N subcarrier
symbols dened by the set of orthonormal pulse shapes fgk tg.
In applications where intersymbol interference (ISI) is present (see Chapter 9), a guard time
TGI is added to the start of a symbol. The guard time will be deleted at the receiver before the
FFT is performed to recover fZk g. This method can eliminate ISI if the maximum delay spread
of the ISI can be reasonably predicted. The drawback is the loss in throughput because the
symbol time is now Ts TGI . The guard time is created by prepending the IFFT sequence a
circular extension or cyclic prex of itself. For example, the cyclic prex for
fsL ng; n 0; 1; . . . ; N 1 is the last m samples sL N m, sL N m 1; . . . ;
sL N 1. These are added to the beginning of the sequence sL 0, sL 1; . . . ; sL N 1.
The new sequence now consists of N + m samples. Naturally, the rst m samples of the
received version of fsL ng are discarded at the receiver before the FFT process. Due to the
periodic nature of IFFT, the junction at the start of the original symbol will always be
continuous. Note that discontinuities can exist at the junctions between adjacent symbols.
Figure 6.46 shows how a guard time is created using cyclic prex.
In wireless applications, the transmitted signal can arrive at the receiver via many
propagation paths due to reection from large objects (such as the ground, hills, or buildings), and diffraction which allows the signal to slide via rooftops or bend around objects,
Cyclic
prefix
Figure 6.46
OFDM symbol
Cyclic
Prefix
OFDM symbol
287
6.18 OFDM
and scattering from smaller objects (such as trees or lamp posts). Thus, the receiver can
receive multiple attenuated and delayed copies of the transmitted signal. Often, a few
dominant copies are present and the others are too weak to be signicant. These dominant
copies not only interfere with themselves, but the delayed ones also interfere with the next
symbol. The former effect is called multipath fading resulting in a scaling and rotation of the
symbol, which leads to performance degradation (higher symbol error rate). The latter effect
is called ISI and can be eliminated by the guard time as discussed above.
Cyclic prefix
In this section we present a mathematical description of cyclic prex and its role in removing
ISI resulting from transmitting the OFDM signal through a multipath fading channel (see
Chapter 10).
(6:167)
The discrete-time Fourier transform (DTFT) of y(n) is the product of DTFT {h(n)} and
DTFT {x(n)}
Y ej2f Hej2f X ej2f
(6:168)
where
Hej2f DTFTfhng
hn ej2fn
X ej2f DTFTfxng
xn ej2fn
(6:169)
In OFDM the sequences x(n) and h(n) are of nite length N. Therefore, taking Fourier
transform of y(n) does not give the multiplication in the frequency domain unless we use
cyclic prex to convert linear convolution to circular convolution. Assume that the length of
the sequences x(n) and h(n) is N (if their lengths are not the same, zeros can be used to pad
the sequences to the same length). The length Nc needed so that linear convolution and
circular convolution are the same from 0 to Nc 1 is 2N 1. The circular convolution of
two sequences h(n) and x(n) is given by
yn hn xn
N 1
X
hkxn kN
(6:170)
k0
288
Modulation
circle with a xed reference. A counterclockwise rotation gives a new reference and a
circularly shifted sequence. This is also equivalent to a linear shift of the periodic extension
of x(n). Consider the case of k = 2, we have the following circularly shifted sequence:
xn kN fxN 2; xN 1; x0; x1; . . . ; xN 3g.
The N-point DFT of y(n) is given by
Y k HkX k; k 0; 1; . . . ; N 1
(6:171a)
where H(k) is the N-point DFT of h(n) and X(k) is the N-point DFT of x(n).
Hk DFTfhng
N 1
X
hn ej2kn=N ;
k 0; 1; . . . ; N 1
n0
X k DFTfxng
N 1
X
(6:171b)
xn ej2kn=N ;
k 0; 1; . . . ; N 1
n0
Note that the DFT corresponds to samples of the DTFT taken every 1/N. Thus, we have
Hk Hej2f f k=N Hej2k=N
X k X ej2f f k=N X ej2k=N
(6:172)
N 1
1X
Y k ej2kn=N
N k0
N 1
1X
HkX k ej2kn=N
N k0
hl
xm
e
N l0
m0
k0
N 1
X
hlxn lN
l0
hn xn; n 0; 1; . . . ; N 1
(6:173)
(6:174)
289
6.18 OFDM
Now consider a transmitted OFDM symbol that is the sequence sL n of length N. This
sequence serves as the input to a channel characterized by a nite impulse response (FIR)
h(n) of length 1 < L < N. Let us form an expanded input sequence ~sL n of length N + L 1
by copying the last L 1 samples of sL n, L 1 n N 1, that is, the cyclic prex of
sL n, and append them to the front of the sequence sL n. Thus, the expanded sequence is
~sL n sL nN ; L 1 n N 1. Consequently, we have ~sL n k sL n kN ,
L 1 n k N 1. Also, we expand the sequence h(n) to length N + L 1 with
zero padding. The output of the FIR channel is the linear convolution of h(n) and ~sL n as
given by
yn hn ~sL n
NX
L1
hk~sL n k
k0
NL1
X
hksL n kN
k0
L1
X
hksL n kN
(6:175)
k0
hn sL n; n L 1; L 2; . . . ; 1; 0; 1; . . . ; N 1
The rst L 1 samples of y(n), n L 1; L 2; . . . ; 1, are redundant because they are
associated with the cyclic prex of sL n. Furthermore, these samples also contain the ISI that
results from passing the last L 1 samples of the previous OFDM symbol through the channel.
These samples are discarded by the receiver prior to the DFT operation. If we take DFT of both
sides of the above expression we obtain the frequency-domain samples as follows:
Y k HkZk; k 0; 1; . . . ; N 1
(6:176)
Conversely, the time-domain sample y(n) can be expressed in terms of Y(k) via IDFT as
y n
1
1 NX
HkZk ej2kn=N ;
N k0
n 0; 1; . . . ; N 1
(6:177)
The channel frequency response H(k) can be measured by sending a known symbol Z(k) and
calculating the ratio of the received symbol Y(k) to Z(k), that is, H(k) = Y(k)/ Z(k). The length
of cyclic prex is a compromise between how much ISI one wishes to eliminate and how
much reduction in system throughput one can accept.
The envelope of an OFDM signal is not constant. Occasionally, a large signal peak can
occur when many subcarrier symbols are added in phase. This problem can result in the
clipping of the signal amplitude in the D/A converter, causing errors in the transmitting of
the symbols. Furthermore, it can saturate the transmitter power amplier, causing intermodulation distortion. The latter problem can be solved by operating the amplier in the
linear region with enough power back-off. The former problem can be remedied by using
error correction codes and interleaving as shown in Figure 6.47. Often a single metric called
{0,1}
Figure 6.47
Encoder
Interleaver
OFDM
modulator
OFDM
290
Modulation
peak-to-average power ratio (PAR) is used to describe the amount of back-off required for
the power amplier. Since the peak occurs infrequently, PAR does not seem to be very
useful. For example, let the PAR of an OFDM signal be 8 dB. If a peak power exceeds a level
of 4 dB above the average power level for only 1% of the time (that is, the probability of the
signal being clipped is 0.01 if the amplier is backed-off by 4 dB), then we might as well
operate the amplier with 4 dB back-off instead of 8 dB and design a code that can correct
the error resulting from clipping.
The problem of signal clipping does not exist if OFDM is implemented at RF, as in
Figure 6.44. The random phase shifts between subcarriers prevent the coherent addition of
subcarriers. This suggests that pseudo-random phase shifts can be purposely introduced to
the N subcarriers for baseband implementation. The information on the set of pseudorandom phase shifts for particular symbols can be transmitted on one of the N subcarriers
for the receiver to remove during demodulation.
OFDM signal is the sum of N orthogonal subcarriers; therefore, its power spectral density
is the sum of the power spectral densities of N subcarriers. If the subcarrier modulation is
MPSK or MQAM, the spectral density of a subcarrier simply has a sin x=x2 shape, and
hence, the OFDM signal has a spectrum that is fairly at across the top. The bandwidth of an
OFDM signal is approximately equal to N 1=Ts , that is, about N + 1 times the subcarrier
symbol rate 1=Ts .
291
Ungerboeck TCM
In 1982 Ungerboeck [33] introduced trellis coded modulation (TCM), which is exible and
effective for expanded MPSK and MQAM with convolutional codes of arbitrary rate
r k=k 1. Conventional MPSK and MQAM employ Gray symbol mapping. With
soft decision decoding, the Euclidean free distance of the code, which is the minimum
Euclidean distance between any two code vectors, should be as large as possible for a given
code rate and constraint length (the Euclidean free distance is not the free distance dfree ,
which is the smallest Hamming distance between any two distinct code sequences). Note
that the Euclidean free distance has to be larger than the minimum Euclidean distance of the
uncoded signal space of M 2k to have any coding gain. Gray symbol mapping may fail to
accomplish this goal with some codes and usually leads to low coding gain. To overcome
this problem Ungerboeck proposed symbol mapping by set partitioning to obtain good
coding gain for expanded MPSK and MQAM using simple convolutional codes. The
essence of mapping by set portioning is to partition the expanded signal space into subsets
whose minimum Euclidean distances increase as the subsets become smaller.
The next question is how to map code symbols (a code symbol has k + 1 bits) to signal
vectors in the expanded signal space. The symbol mapping depends on the code structure.
For example, one mother code of rate is used to generate two output coded bits from one
input information bit while the remaining input information bit serves as the third output
coded bit. This results in a rate 2=3 code. This code has a parallel transition between states
because of the uncoded bit. Similarly, one can generate a higher rate code using more
uncoded bits. For example, rate 3=4 code is generated by using remaining input information
bits as the third and fourth output coded bits. This code has two parallel transitions between
states. Another example to obtain rate 3=4 code is to use a rate 2=3 mother code to generate
three output coded bits from two input information bits while the remaining input information bit serves as the fourth output coded bit. This code has a parallel transition between
states. The following rules are used for symbol mapping:
1. The output coded bits are used to select the subsets. This maximizes the Euclidean
distance between transitions leaving or entering the same state.
2. The uncoded bits are used to select the signal points in the subsets. This maximizes the
Euclidean distance between parallel transitions.
These rules are ad hoc methods for symbol mapping but they are effective in generating
good coding gain for simple convolutional codes. The following examples illustrate the
concept of symbol mapping by set partitioning.
292
Modulation
00
01
11
10
Figure 6.48
Symbol mapping by set partitioning for QPSK with rate convolutional code. The bold face bit is the second coded bit
and the other bit is the rst coded bit.
01
00
10
11
Rate
convolutional
code
Figure 6.49
Implementation of TCM-QPSK.
quadrant (second quadrant) when it is 0 and the signal vector in the third quadrant (fourth
quadrant) when it is 1. Note that the symbol mapping is clearly not Gray symbol mapping.
We observe that two coded bits are transmitted in one QPSK symbol interval Ts 2Tb ,
which is equivalent to transmitting one uncoded bit in one PSK bit interval Tb . This explains
why TCM-QPSK has the same spectral efciency as uncoded PSK. Figure 6.49 shows the
implementation of TCM-QPSK.
293
000
001
010
011
110
100
Figure 6.50
101
111
Symbol mapping by set partitioning for 8PSK with rate 2=3 convolutional code. The rst bit is the uncoded bit, the bold
face and underlined bit is the second coded bit and the bold face bit is the rst coded bit.
294
Modulation
001
000
Rate
convolutional
code
111
110
Q
100
Figure 6.51
101
Implementation of TCM-8PSK
0000 1000
Figure 6.52
011
010
1100 0100
0010 1010
1110 0110
1001 0001
0101 1101
1011 0011
0111 1111
Symbol mapping by set partitioning for 16QAM with rate 3=4 convolutional code. The rst bit is the uncoded bit, the
underlined bit is the third coded bit, the bold face and underlined bit is the second coded bit, and the bold face bit is
the rst coded bit.
when it is 1. The second partition has four subsets, each with four signal vectors and the
p
p
minimum Euclidean distance is 4 Es as compared to 2 2Es of the rst partition. The
second coded bit selects the subset on the left when it is 0 and the subset on the right when it
is 1. The third partition has eight subsets, each with two signal vectors and the minimum
p
p
Euclidean distance is 4 2Es as compared to 4 Es of the rst partition. The third coded bit
selects the subset on the left when it is 0 and the subset on the right when it is 1. The uncoded
bit selects the signal vector in each quadrant. Note that the symbol mapping is clearly not
Gray symbol mapping. We observe that four coded bits are transmitted in one 16QAM
symbol interval Ts 4Tb , which is equivalent to transmitting three uncoded bits in one
295
Rate 2/3
convolutional
code
Figure 6.53
Implementation of TCM-16QAM.
Y
X
Figure 6.54
8QAM symbol interval Ts 3Tb . This explains why TCM-16QAM has the same spectral
efciency as uncoded 8QAM. Figure 6.53 shows the implementation of TCM-16QAM.
Pragmatic TCM
Ungerboeck TCM requires the use of a convolutional code with precise code rate k/(k+1) for
the expanded signal space. For example, rate 2=3 must be used for 16QAM. Pragmatic TCM
(adopted by the IEEE 802.16) employs various code rates such as and 2=3 for 16QAM, 2=3 and
3=4 for 64QAM, and 3=4, 5=6, and 7=8 for 256QAM. Furthermore, one mother code of rate is
used for all modulation techniques with rates 2=3 and 3=4 codes derived from rate code via
puncturing. Rates 5=6 and 7=8 are derived from rate 3=4 punctured code via the use of uncoded bits
(two uncoded bits for rate 5=6 and four uncoded bits for rate 7=8). The use of a mother code of rate
for all required modulation techniques simplies the encoder and the decoder. Figure 6.54
shows the industry standard rate convolutional encoder with constraint length K = 7.
296
Modulation
u2
c3
u1
c1
16QAM
u0
Figure 6.55
Rate 1/2
convolutional
code
c2
c0
c1c0
00
01
10
11
11 10 01 00 c3c2
Pragmatic rate 3=4 TCM-16QAM. The I-value of a signal vector is generated by c3 c2 and its Q-value is generated by c1 c0
via set partitioning for 4ASK.
bit of every three X-bits, namely X1 X2 X3, and by puncturing the third bit of every three Y-bits,
namely Y1 Y2 Y3. Thus, the sequence of transmitted coded bits is X1 Y1 Y2 X3 ; X4 Y4 Y5 X6 ; . . ..
Since it takes three information bits to generate four transmitted coded bits, the rate is 3=4.
297
16QAM
u0
Rate 1/2
convolutional
code
c3
c1
or
c2
c0
Q
Figure 6.56
c1c0
01
00
10
11
11 10 00 01 c3c2
Pragmatic rate TCM-16QAM. The I-value of a signal vector is generated by c3 c2 and its Q-value is generated by c1 c0
via Gray mapping.
u4
c5
u3
c2
64QAM
u2
u1
u0
Figure 6.57
Rate 3/4
convolutional
code
c4
c3
c1
c0
298
Modulation
c2 c1 c0
011
010
000
001
101
100
110
111
111 110 100 101 011 010 000 001
Figure 6.58
64QAM
u1
u0
Figure 6.59
c5 c4 c3
c5
Rate 1/2
convolutional
code
c2
or
c4
c1
c
c
3
0
6.20 Summary
In this chapter we have provided a detailed study of major digital modulation techniques. We divided them into
binary modulation and M-ary modulation. The binary modulation includes PSK, DPSK, ASK, FSK, MSK, and GMSK.
PSK is a popular modulation that is employed widely in practice including signal in the wireless 802.11a and
802.11g standards. DPSK is employed to resolve the phase ambiguity and in applications that require
noncoherent demodulation because phase tracking is not possible. DPSK is used in the wireless IEEE 802.11
standard. ASK is employed explicitly in ber optic communication systems. GMSK is employed in the most
299
Problems
popular cellular standard, namely, GSM. The M-ary modulation techniques covered MASK, MPSK including
OQPSK, DMPSK including =4-DQPSK, MQAM, DMQAM, CSK, MFSK, CPM, and OFDM. We noted that 8ASK is
employed in the HDTV standard in the United States. QPSK is perhaps the most popular modulation for satellite
communication. QPSK and 16QAM, 64QAM, and 256QAM are used in the wireless 802.11a,g and 802.16
standards. =4-DQPSK is used in the cellular IS-136 standard. CSK is employed in the reverse channel of the
IS-95 cellular standard, and OFDM is the orthogonal multiplexing scheme for all the current wireless LAN and
MAN standards cited above. The encoding of an I-Q signal with a convolutional code results in trellis coded
modulation. Both Ungerboeck TCM and the pragmatic TCM used in 802.16 family are studied.
Problems
1.
2.
3.
4.
5.
6.
7.
The ideal bandwidth of a PSK signal is 1=Tb if a sin x=x pulse shape is employed.
What is the noiseless spectral efciency of PSK? What is the noiseless spectral
efciency of PSK if a squared pulse shape is employed, assuming a null-to-null
bandwidth? At Eb =N0 8 dB compare the result of sin x=x pulse shape to the
Shannon channel capacity using Figure 4.8 and Figure 6.7.
Consider a PSK signal with raised-cosine pulse shape with a roll-off factor 0:5.
What is the noiseless spectral efciency of PSK? For the same spectral efciency what
is the bit energy-to-noise density ratio Eb =N0 to achieve a reliable transmission
dictated by the Shannon channel capacity?
Generate the DPSK signal for the following data bit sequence 0100110.
Consider the use of two Walsh functions derived from the two rows of the Hadamard
matrix
1 1
H2
1 1
We have w1 t pt pt Tb and w2 t pt pt Tb where p(t) is a
squared pulse of unit amplitude and duration Tb . These two Walsh functions are
orthogonal by inspection over (0, 2Tb ). Let us use these two functions for binary
communication. Show that this modulation technique is actually DPSK.
What is the noiseless spectral efciency of FSK if a squared pulse shape is employed,
assuming a null-to-null bandwidth? At Eb =N0 8 dB compare the above result to the
Shannon channel capacity. For the same spectral efciency as FSK, what is the
required Eb =N0 dictated by the Shannon channel capacity for reliable transmission?
What is the spectral efciency of MSK, assuming a null-to-null bandwidth? At
Eb =N0 8 dB compare the above result to the Shannon channel capacity. For the
same spectral efciency as MSK, what is the required Eb =N0 dictated by the Shannon
channel capacity for reliable transmission?
What is the spectral efciency of GMSK with BTb 0:25, assuming a 90%-power
bandwidth? At Eb =N0 8 dB compare the above results to the Shannon channel
capacity. For the same spectral efciency as GMSK, what is the required Eb =N0
dictated by the Shannon channel capacity for reliable transmission?
300
Modulation
8.
9.
10.
11.
12.
13.
14.
15.
16.
17.
18.
19.
20.
21.
22.
Generate the in-phase and quadrature bit sequences for MSK given the input bit
sequence 110011001.
What are the noiseless spectral efciencies of 8PSK and 16QAM, assuming raisedcosine pulse shapes with roll-off factors 0? At Eb =N0 8 dB compare the above
results to the Shannon channel capacity using Figure 4.8 and Figure 6.7. For the same
spectral efciency, what is the required Eb =N0 dictated by the Shannon channel
capacity for reliable transmission?
Derive the power spectral density expression (6.85), assuming uncorrelated data symbols.
Find the corresponding I-Q-values and phases of OQPSK for the data sequence
0010110110.
Use Table 6.6 to provide the differential encoding for DQPSK for the following data
sequences:
(a) {3, 4, 2, 1, 3}, assuming the initial phase is /4 radians.
(b) {2, 3, 4, 1, 2}, assuming the initial phase is 7/4 radians.
Use direct symbol mapping to verify the result in Example 6.28.
Provide the direct symbol mapping for DQPSK for the data sequence {1, 2, 4, 1, 3},
assuming the initial phase is 5/4 radians.
Provide the direct symbol mapping for =4-DQPSK for the data sequence {3, 2, 4, 1,
3}, assuming the initial phase is /2 radians.
Provide the direct symbol mapping for =4-DQPSK for the following data sequences:
(a) {3, 2, 4, 1, 3}, assuming the initial phase is /4 radians.
(b) {4, 2, 4, 1, 3}, assuming the initial phase is 7/4 radians.
Design three signal spaces for 16QAM with Gray encoding.
Perform differential encoding for D16QAM using Figure 6.37 and the message
symbols {0 1 1 1, 1 0 0 0, 0 0 0 1, 1 1 0 0, 0 0 10}.
Perform the differential encoding for D16QAM using Figure 6.37 and the message
symbols {0 1 1 0, 1 0 0 1, 1 0 0 0, 1 1 0 0, 0 1 0 1}.
Find the minimum Euclidean distances and average symbol energies of the signal
spaces in Figure 6.35(a)(c). For M = 16, compare thepresults
with
16QAM.
prectangular
(a) The radii of the inner and outer circles are E and 2 E , respectively, for
Figure 6.35(a).
p p
p
p
(b) The radii of the circles are E , 3 E=2, 2 E , and
5
for
p E=2p
Figure 6.35(b).
(c) The radii of the inner and outer circles are E and 2 E , respectively, for
Figure 6.35(c).
The Walsh functions satisfy the closure property, that is, they form a group under
multiplication. This means for any wi t, wj t 2 WN , wi twj t wk t 2 WN ,
where WN is the set of N Walsh functions.
(a) Demonstrate this property using W4 as an example.
(b) Convert the Walsh indices i, j, k to their corresponding Walsh binary indices Qi ,
Qj , Qk . Construct a set W4 such that wi twj t wk t 2 W4 implies
Qk Qi Qj and vice versa.
Show that the Walsh sequences obtained from the rows of the Hadamard matrix via
the mapping 1 ! 0; 1 ! 1 satisfy the closure property, that is, they form a group
under modulo-2 addition.
301
Problems
23.
24.
25.
26.
27.
28.
29.
30.
si s, i = 1, 2, . . . ,
these vectors be s 1=M M
i
i1 i
M. Calculate the signal energy of the new set. Find the crosscorrelation of any pair of
signals in the new set. When M = 2 show that this simplex signaling reduces to PSK.
Consider four-dimensional signals which can be represented by a set of vectors. From
one vector s1 1; 1; 1; 1 the other vectors can be formed by replacing the components of s1 by their negatives.
(a) How many bits does a symbol vector carry?
(b) What are the symbol and bit energies?
(c) What is the minimum Euclidean distance?
(d) What is the minimum correlation between signal pairs?
302
Modulation
31.
32.
33.
34.
35.
36.
37.
38.
39.
Further reading
For a thorough review of amplitude modulation and frequency modulation we refer the
reader to the books by Couch [2], Haykin [3], Ziemer and Tranter [4], Proakis and Salehi [5],
and Carlson et al. [6]. For further exploration of digital modulation the reader should consult
[1323]. For CPM in particular, the following books go into greater detail than our
coverage: [10,11,13,15,21].
Bibliography
1. G. Keiser, Optical Fiber Communications, 4th edition, New York: McGraw-Hill, 2007.
2. L. W. Couch, II, Digital and Analog Communication Systems, 7th edition, Upper
Saddle River, NJ: Prentice Hall, 2007.
303
Bibliography
3. S. Haykin, Communication Systems, 3rd edition, Chichester: John Wiley & Sons,
1994.
4. R. E. Ziemer and W. H. Tranter, Principles of Communications, 4th edition, Chichester:
John Wiley & Sons, 1995.
5. J. G. Proakis and M. Salehi, Fundamentals of Communication Systems, Upper Saddle
River, NJ: Pearson Prentice Hall, 2005.
6. A. Bruce Carlson, P. B. Crilly, and J. C. Rutledge, Communication Systems, 4th edition,
New York: McGraw-Hill, 2002.
7. IEEE Standards 802.11a, 1999.
8. IEEE Standards 802.11g, 2003.
9. W. J. Weber, Differential encoding for multiple amplitude and phase shift keying
systems, IEEE Trans. Communications, Vol. 26, No. 3, pp. 385391, 1978.
10. F. Xiong, Digital Modulation Techniques, Boston, MA: Artech House, 2000.
11. J. G. Proakis and M. Salehi, Digital Communications, 5th edition, New York: McGrawHill, 2008.
12. J. G. Proakis and D. G. Manolakis, Digital Signal Processing, 4th edition, Upper Saddle
River, NJ: Prentice Hall, 2007.
13. J. B. Anderson, T. Aulin, and C.-E. Sundberg, Digital Phase Modulation, New York:
Plenum Press, 1986.
14. J. R. Barry, E. A. Lee, and D. G. Messerschmitt, Digital Communication, 3rd edition,
Amsterdam: Kluwer Academic Publishers, 2004.
15. S. Benedetto, E. Biglieri, and V. Castellani, Digital Transmission Theory, Englewood
Cliffs, NJ: Prentice-Hall, 1987.
16. R. E. Blahut, Digital Transmission of Information, Addison-Wesley, 1990.
17. R. G. Gallager, Principles of Digital Communication, Cambridge: Cambridge
University Press, 2008.
18. R. D. Gitlin, J. F. Hayes, and S. B. Weinstein. Data Communications Principles, New
York: Plenum Press, 1992.
19. U. Madhow, Fundamentals of Digital Communication, Cambridge: Cambridge
University Press, 2008.
20. M. B. Pursley, Introduction to Digital Communications, Upper Saddle River, NJ:
Pearson Prentice-Hall, 2005.
21. M. K. Simon, S. M. Hinedi, and W. C. Lindsey, Digital Communication Techniques.
Signal Design and Detection, Englewood Cliffs, NJ: Prentice-Hall, 1995.
22. B. Sklar, Digital Communications. Fundamentals and Applications, 2nd edition, Upper
Saddle River, NJ: Prentice-Hall, 2001.
23. R. E. Ziemer and R. L. Peterson, Introduction to Digital Communication, 2nd edition,
Upper Saddle River, NJ: Prentice-Hall, 2001.
24. G. C. Clark, Jr. and J. B. Cain, Error-Correction Coding for Digital Communications,
New York: Plenum Press, 1981.
25. A. Lapidoth, A Foundation in Digital Communication, Cambridge: Cambridge
University Press, 2009.
26. S. Lin and D. J. Costello, Jr., Error Control Coding, 2nd edition, Upper Saddle River,
NJ: Pearson Prentice-Hall, 2004.
304
Modulation
27. K. Murota and K. Hirade, GMSK modulation for digital mobile radio telephony,
IEEE Trans. Commun., Vol. COM-29, No.7, pp. 10441050, 1981.
28. T. S. Rappaport, Wireless Communications, 2nd edition, Upper Saddle River, NJ:
Prentice Hall, 2002.
29. C. E. Shannon, A mathematical theory of communication, Bell Syst. Tech. J., Vol. 27,
pp. 379423, 1948.
30. C. E. Shannon, A mathematical theory of communication, Bell Syst. Tech. J., Vol. 27,
pp. 626656, 1948.
31. S. G. Wilson, Digital Modulation and Coding, Englewood Cliffs, NJ: Prentice-Hall,
1996.
32. J. M. Wozencraft and I. M. Jacobs, Principles of Communication Engineering, New
York: Wiley, 1965; Waveland Press, 1990.
33. G. Ungerboeck, Channel coding with multilevel/phase signals, IEEE Trans.
Information Theory, Vol. IT-28, No.1, pp. 5567, 1982.
Demodulation
Introduction
The purpose of digital demodulation is to recover the information (bits, symbols) carried
by the digitally modulated signals. This process can be achieved via coherent or
noncoherent demodulation. The former requires a local reference carrier to be
matched exactly in frequency and phase to the received signal. The latter requires only a
match in frequency. Both types of demodulation involve two steps. In the rst step, a
signal processor is employed to convert the received signal that represents a bit
(binary modulation) or a symbol (M-ary modulation) into a decision sample at maximum
signal-to-noise ratio for the case of coherent demodulation. For noncoherent demodulation, the signal processor converts the received signal into a nonnegative decision
sample independent of its unknown initial phase, which is cleverly discarded. Since the
phase information is not employed, a loss in the signal-to-noise ratio for the decision
sample results. The coherent signal processors are the matched lter and the correlator,
both are equivalent at the time the decision sample is taken. The noncoherent signal
processors are the matched lter-envelope detector (also known as noncoherent
matched lter) and the quadrature correlator-square law detector (also known as noncoherent correlator), both are also equivalent at the time the decision sample is obtained.
Binary demodulation employs one or two signal processors depending on the type of
modulation. On the other hand, M-ary modulation requires a bank of L signal processors,
1 L M.
In the second step, an optimum detector (maximum likelihood detector) is employed to
determine the bit or the symbol represented by the decision sample. The optimum detector
recovers the information with a minimum bit error probability for a given received signal-tonoise ratio. Figure 7.1 illustrates the digital demodulation concept. For binary demodulation,
the optimum detector is the threshold detector or the maximum detector depending on the
type of modulation. For M-ary demodulation, the optimum detector is the minimum
Euclidean distance detector or the M-ary maximum detector, again, depending on the type
of modulation.
Demodulation will be studied for the digital modulation techniques discussed in
Chapter 6, which includes the family of I-Q modulation such as MASK, MPSK,
and MQAM, the family of orthogonal signals such as CSK and MFSK, and the
family of continuous phase modulation (CPM). We also provide a study of OFDM
demodulation.
305
306
Demodulation
Signal + noise
Figure 7.1
Signal
processor
Optimum
detector
Decision
vector
Bit or symbol
Generic digital demodulator. The decision vector may contain up to M decision samples.
t=T
s(t) + n(t)
Figure 7.2
Matched
filter
h(t)
s0(t) + n0(t)
X = s0(T) + n0(T)
s20 T
n20 T
(7:1)
The matched lter concept is illustrated in Figure 7.2. Note that SNR0 is simply the ratio of
the square of the mean to the variance of the decision variable X s0 T n0 T .
(7:2)
(7:3)
1
1
307
Using (7.3), we can evaluate the mean-square value of the noise sample n0 T :
1 1
2
n0 T E
nt n hT t hT dt d
1 1
1 1
Ef nt n g hT t hT dt d
1 1
1 1
N0
t hT t hT dt d
2 1 1
N0 1 2
h T t dt
2 1
We have used the fact that noise is wide-sense stationary
E fnt n g N0 =2t . Substituting (7.2) and (7.4) into (7.1), we have
1
2
s hT d
2
1
1
SNR0
N0
h2 T t dt
2
N0
1
(7:4)
so
that
(7:5)
st hT t dt
1
1
1
h2 T t dt
The right-hand side of (7.5) can be maximized with the help of the CauchySchwarz
inequality, which states that
1
2 1
1
2
a
t
dt
at
dt
(7:6)
j
j
jbtj2 dt
1
1
1
where at and bt are complex and nite energy functions. The equality holds if and only if
within a scaling constant
bt a t
(7:7)
1
1
s2 t dt
(7:9)
308
Demodulation
(7:10)
ht s T t sT t
(7:11)
or, equivalently,
when st is real. This shows that ht is simply a time-reversed and T-translated version of
st , hence the name matched lter. For simplicity we set 1 in (7.11) and observe that at
the detection time t T , the matched lter collects the signal energy E, that is, the signal
must have passed completely through the lter. It is then sufcient to assume that st is timelimited to T seconds to ensure that SNR0 2E=N0 at t T , which is the maximum signalto-noise ratio.
s( t)
s(t)
s0(t)
A2T = E
t
0
Figure 7.3
2T
309
(7:12)
A
A
pt pt cos 4fc t
2
2
The high-frequency term 12Apt cos 4fc t is ltered out by the matched lter that matches
the component Apt . Alternatively, one can implement the matched lter that matches
the signal s(t) directly, as shown in Figure 7.5. Note that the output signal s0 t is a
modulated triangular pulse with a peak at the time t T , where the maximum SNR0
is achieved. The drawback of the implementation in Figure 7.5 is the requirement for
more accurate sampling timing because of the sharp roll-off of the signal around the time
t T.
s0(t)
A2T/2 = E
s(t)
v(t)
h(t) = Ap(T t)
s0(t)
t
0
cos2 fct
Figure 7.4
2T
s0(t)
Envelope of s0(t)
s(t)
Figure 7.5
h(t) = s(Tt)
t
T
310
Demodulation
Applying the Parseval formula to (7.4) we obtain the output noise power at t T as
follows:
N0 1
n20 T
(7:14)
jH f j2 df
2 1
Substituting (7.13) and (7.14) into (7.1), we obtain
1
2
H f S f ej2fT df
2
1
1
SNR0
N0
jH f j2 df
(7:15)
1
2
N0
2
N0
2E
N0
1
1
jH f j df
jS f j2 df
1
1
1
jH f j2 df
1
jS f j2 df
2
N0
1
1
(7:16)
s2 t dt
useof the CauchySchwarz inequality and the Parseval formula, and the fact that
via thej2fT
S f e
jS f j. The maximum SNR0 is obtained when
H f S f ej2fT
(7:17)
(7:18)
311
s0 t
1
1
s sT t d
1
1
s sT t d
(7:19)
Rs T t Rs t T
where Rs , by denition (2.191), is the autocorrelation of the input signal st with a time
lag . Note that the autocorrelation is an even function by (2.194). At time t T , we get
1
s2 t dt
s0 T Rs 0
(7:20)
1
E
Furthermore, the output noise in (7.3) is given by
1
n sT t d
n0 t
(7:21)
1
and hence, at t T ,
n0 T
1
1
nt st dt
(7:22)
Decision variable
In order to evaluate the matched lter performance in signal detection, we need to know that
statistics of the decision variable X s0 T n0 T at the output of the matched lter in
Figure 7.2. First, we note that the matched lter is a linear lter; therefore, the output noise
n0 t is also Gaussian. Thus, X is a Gaussian random variable with the following statistics:
X s0 T E
(7:23)
2 n20 T 2 EN0 =2
(7:24)
2
2
1
fX x p exX =2
2
(7:25)
where X , 2 , and fX x are the mean, variance, and probability density function of X,
respectively. Again is a scaling factor which can always be set to unity if so desired.
The probability distribution function FX x PrX x is given by
x
fX ydy
FX x PrX x
1
x
2
2
1
1 1
2
p
eyX =2 dy p
ey =2 dy
(7:26)
2 1
2 X x=
X x
Q
312
Demodulation
1 1 y2 =2
e
dy
Qa p
2 a
(7:27)
and
Qa 1 Qa
(7:28)
PrX 4x 1 PrX x
xX
Q
(7:29)
Note that
Summary
The matched lter is a coherent signal processor because the signal waveform must be
known exactly. Its impulse response is the time-reversed and T-translated waveform of the
signal. It achieves the maximum output signal-to-noise ratio of 2E=N0 at the sampling time
t T . The output of the matched lter, expressed by X s0 T n0 T , completely
represents the received signal rt st nt . In other words, X represents sufcient
statistics that summarize rt for the purpose of detecting the input signal st . Sufcient
statistics allow viewing an innite dimensional signal xt as a scalar X well enough for the
detection of st . It can be shown via (7.20) that et rt Xx1 t st nt
T
x 1 t nt n0 T x1 t ; wherepx1t is a unit energy function dened
s0 T x1 t n0 p
as x1 t st = E and is selected to be 1= E , and is uncorrelated to X, that is,
Efet X g 0. Thus, et contains no information relevant to the detection of st . Note that
st s0 T x1 t , thus s0 T is the projection of st into the basis function x1 t . Similarly,
n0 T is the projection of input noise nt onto x1 t . Also note that the noise
et nt n0 T x1 t is Gaussian, therefore, Efet X g 0 also implies that et and
X are independent. Therefore, et is irrelevant in the detection of st . From the statistical
point of view, the Gaussian random variable X s0 T n0 T requires only two parameters to fully characterize it, namely, its mean and variance. Since the mean is s0 T and
the variance is the variance of n0 T , sufcient statistics are contained in X.
313
t=T
s(t) + n(t)
X = s0(T ) + n0(T )
s(t)
Figure 7.6
The correlator.
s0(t)
t=T
s(t)
v(t)
A2T/2=
s0(t)
s(t)
Figure 7.7
nt st dt
(7:31)
The above two equations allow us to alternatively implement the matched lter by using a
voltage multiplier and an integrator as illustrated in Figure 7.6. This equivalent version of
the matched lter is called the correlator. Note that the equivalence is valid only at the
sampling time t T . This also implies that the output of the correlator represents sufcient
statistics for the detection of rt st nt .
(7:32)
Note that p2 t pt and the high-frequency term 12A2 pt cos 4fc t is ltered out by the
integrator, which acts as a lowpass lter. The output signal is s0 t 12A2 t pt , which
yields at t T the sample s0 T A2 T =2 E. Thus, we obtain the same result as that of
the matched lter approach in Example 7.2.
314
Demodulation
t=T
s(t,) + n(t)
Figure 7.8
Matched
filter
h(t) = s(T t)
s0(t) + n0(t)
Envelope
detector
x(t)
X = x (T)
applications, the phase of a modulated carrier cannot be tracked accurately. Therefore, the
matched lter or the correlator will fail to work in these cases. Let us examine the case in
Example 7.3. Suppose the signal st arrives at the receiver with an unknown phase , that is,
st Apt cos2fc t . In this case, the output signal vt is vt 1=2A2 pt cos
1=2A2 pt cos4f t . Thus, the output of the integrator is s t 1=2A2 t pt cos ,
c
0
which yields s0 T 1=2A2 T cos . It is obvious that s0 T 0 when 90 , which
implies a zero output signal! To avoid the situation where coherent demodulation cannot be
achieved with a matched lter or a correlator, we can employ a combination of matched
lterenvelope detector (also referred to as noncoherent matched lter) or quadrature
correlatorsquare-law detector (or noncoherent correlator). In this section we study the
former while the latter will be examined in Section 7.4.
For a digital signal whose information is contained in its complex envelope (lowpass
equivalent signal), an envelope detector can be employed at the output of the matched lter
to extract the information bit. Obviously, we can just employ the envelope detector to extract
the information bit without the use of a matched lter in some modulation techniques, such
as standard AM (OOK or ASK, for example). On the other hand, the matched lter
envelope detector will give us better performance, as we will prove later. Figure 7.8
illustrates the matched lterenvelope detector.
We assume that the matched lter is matched to the transmitted signal st , but when st
arrives at the receiver, the channel introduces an unknown phase . Let us describe st by its
envelope-phase representation as follows:
st Rt cos 2fc t
(7:33)
(7:34)
where is the unknown phase. The matched lter impulse response ht is simply a timereversed and T-translated version of st as given by (7.11) as follows:
ht sT t
RT t cos2fc T t
(7:35)
The channel attenuation can be included in the parameter in (7.35) if desired. The output
signal s0 t at the matched lter output is
315
s0 t
1
1
1
s; ht d
1
1
s; sT t d
R cos2fc RT t cos2fc T t d
1
1
R RT t cos2fc cos 2fc d cos2fc T t
1
1
R RT t cos2fc sin 2fc d sin2fc T t
1
(7:36)
Further simplication yields
s0 t s0;I t cos2fc T t s0;Q t sin2fc T t
(7:37)
where
s0;I t
R RT t cos d
1
1
(7:38)
1
R RT t cos4fc d
1
s0;Q t
R RT t sin d
2 1
1
R RT t sin4fc d
2 1
(7:39)
are the in-phase and quadrature components of s0 t, respectively. Now consider the output
noise n0 t . From (7.21), we get
1
1
n0 t
n ht d
n sT t d
1
1
1
n RT t cos2fc T t d
1
1
(7:40)
n RT t cos 2fc d cos2fc T t
1
1
n RT t sin 2fc d sin2fc T t
1
n0;Q t
1
1
1
n RT t cos 2fc d
(7:41)
n RT t sin 2fc d
(7:42)
316
Demodulation
Combining (7.37) and (7.40), we obtain the output s0 t n0 t of the matched lter as
follows:
s0 t n0 t s0;I t n0;I t cos2fc T t
(7:43)
s0;Q t n0;Q t sin2fc T t
Consequently, the output of the envelope detector is simply
q
2
2
x t
s0;I t n0;I t s0;Q t n0;Q t
The decision variable X at the sampling time t = T is given by X xT as
q
2
2
X
s0;I T n0;I T s0;Q T n0;Q T
where
1
1 2
s0;I T cos
R2 t dt
R t cos4fc t dt
2
2 1
1
s0;Q T sin
2
1
1
R2 t dt
n0;I T
n0;Q T
1
1
1
1
1
R2 t sin4fc t dt
(7:44)
(7:45)
(7:46)
(7:47)
nt Rt cos 2fc t dt
(7:48)
nt Rt sin 2fc t dt
(7:49)
1 1 2
1 1 2
R t dt
R t cos 4fc t dt
2 1
2 1
(7:50)
The second integral on the right is very small compared with the rst integral on the right
because of the oscillation due to the carrier at 2fc . For example, let Rt Apt , where pt is
the unit amplitude pulse of T seconds duration. Then, the rst integral on the right is A2 T =2,
and the second integral is A2 T =2sin 4fc T =4fc T , which is much smaller than A2 T =2 for
fc T 1, a case we always encounter in practice. Thus, for all practical purposes s0;I T and
s0;Q T in (7.46) and (7.47) can be reasonably approximated by
s0;I T E cos
(7:51)
s0;Q T E sin
(7:52)
317
1
2 N 0 1 2
n20;I T
R t cos2 2fc t dt
2 1
2 N 0 1 2
2 N 0 1 2
(7:54)
R t dt
R t cos 4fc t dt
4 1
4 1
2 EN0
2
Similarly,
n20;Q T
2 EN0
2
(7:55)
We remark that the matched lter is a linear time-invariant systems; therefore, its output
noise components are both Gaussian random variables with zero mean and variance
2 EN0 =2.
(7:56)
The power of the signal component is 2 E 2 while that of noise is 2 EN0 . Thus, SNR0 is
given by
SNR0
2 E 2
E
2 EN0 N0
(7:57)
Comparing (7.57) to (7.8), we conclude that the matched lterenvelope detector produces
half the output signal-to-noise ratio of the coherent matched lter, or 3 dB less.
318
Demodulation
Decision variable
In order to predict the performance of a digital communication receiver employing the
matched lterenvelope detector, we need the statistics of the decision variable X in (7.45).
First we observe that s0;I T n0;I T and s0;Q T n0;Q T are both Gaussian random
variables with means s0;I T E cos and s0;Q T E sin , and common variance
2 2 EN0 =2. The decision variable X, being the square root of the sum of two squared
Gaussian random variables, with non-zero mean, is therefore a Rice random variable. Its
probability density function is given by
ax
2
2
2
x
fX x 2 ex a =2 I0 2 ; x 0
a2 s20;I T s20;Q T 2 E2
(7:58)
2
EN
0
2
2
Note that a2 =2 2 SNR0 E=N0 . For the special case when the signal st is not present,
then X is simply the square root of the sum of two squared zero mean Gaussian random
variables with a common variance 2 2 EN0 =2. Thus, X is a Rayleigh random variable
with a probability density function
f X x
x x2 =2 2
e
;
2
x0
(7:59)
Summary
The matched lterenvelope detector is a noncoherent signal processor that can be
employed to detect the presence of a signal with an unknown phase. It achieves a maximum
output signal-to-noise ratio of E=N0 at the sampling time t T . This is 3 dB less than that
produced by a coherent matched lter. The decision variable X at the output of the matched
lterenvelope detector completely represents the envelope of rt st; nt . In other
words, X represents sufcient statistics that summarize jrt j for the purpose of detecting st
with an unknown phase . From the statistics point of view, the Rice random variable X
requires only two parameters a and 2 to characterize it. Both parameters are contained in the
in-phase and quadrature components of the matched lter output.
319
s0;I T cos
2
s0;Q T sin
2
R t dt
2
R2 t dt
n0;I T
R2 t cos4fc t dt
(7:60)
R2 t sin4fc t dt
(7:61)
nt Rt cos 2fc t dt
(7:62)
nt Rt sin 2fc t dt
(7:63)
n0;Q T
T
0
1 y=22
e
;
2 2
y0
(7:65)
The output of the quadrature correlatorsquare law detector represents sufcient statistics that
summarize jrt j jst; nt j for the purpose of detecting st with an unknown phase .
t=T
( )2
s(t, ) + n(t)
+
+
t=T
( )2
Figure 7.9
The quadrature correlatorsquare law detector for an input signal s(t) with unknown phase .
320
Demodulation
(7:66)
where m1 E X jH1 is the mean value of X given hypothesis H1 . Similarly, the probability
density function of X given hypothesis H2 is
2
1
2
fX xjH2 p exm2 =2
2
(7:67)
Optimum threshold
Without loss of generality, let m1 4VT and m2 VT . Therefore, given a value x of the
decision variable X, hypothesis H1 is chosen when x4VT and hypothesis H2 is chosen when
x5VT . Obviously, an error occurs when x5VT given H1 . Similarly, an error occurs when
x4VT given H2 . Thus, by using (7.26), we have
m1 VT
(7:68)
PrX VT jH1 Q
(7:69)
Applying the total probability theorem and using the fact that H1 and H2 are equally likely, that
is, PrH1 PrH2 0:5, we obtain the error probability for threshold detection as follows:
Pe PrH1 PrX VT jH1 PrH2 PrX 4VT jH2
1
m1 VT
1
VT m2
Q
Q
2
2
(7:70)
321
Threshold
detector
m + m2
VT = 1
2
Figure 7.10
H2
<
> VT
H1
(7:71)
(7:72)
m1 m2
2
(7:73)
(7:74)
PrH1 jx 9 PrH2 jx
(7:75)
H1
fX xjHi PrHi
;
fX x
i 1; 2
(7:76)
2
X
i1
fX xjHi PrHi
(7:77)
322
Demodulation
(7:78)
Substituting (7.66) and (7.67) into (7.78) yields the following expression:
2
H2
H1
PrH2
PrH1
(7:79)
The left-hand side is a monotonically increasing (or decreasing) function; therefore, the
inequality is preserved if we take the natural logarithm of both sides of (7.79).
Also, since PrH1 PrH2 0:5, lnPrH2 =PrH1 0 and hence
H2
x m2 2 x m1 2 9 0
(7:80)
H1
x9
H1
m1 m2
VT
2
(7:81)
Therefore, the maximum likelihood criterion establishes the same optimum threshold
VT m1 m2 =2 that is required by the minimum error probability criterion. The threshold detector is thus a maximum likelihood detector.
x 1 9 x2
(7:82)
H1
Figure 7.11 illustrates the maximum detector. We can rewrite the decision rule in (7.82)
using x x1 x2 as
X1
Maximum
detector
X2
Figure 7.11
H2
< x2
x1 >
H1
323
X1
+
Threshold
detector
VT = 0
H2
<
x>
0
H1
X2
Figure 7.12
x x1 x2 9 0
(7:83)
H1
The decision rule in (7.83) allows an equivalent implementation using a threshold detector
as shown in Figure 7.12. This implies that the maximum likelihood detector can be used.
(7:84)
2 EfX2 jH2 g
(7:85)
(7:86)
EfX2 jH1 g 0
(7:87)
Then X X1 X2 is obviously a Gaussian random variable with variance 2 2 and conditional means
m1 Ef X jH1 g EfX1 jH1 g EfX2 jH1 g 1
(7:88)
(7:89)
Therefore, applying the error probability expression for threshold detection of X in (7.74),
we obtain the error probability for the maximum detector or its equivalent implementation
using the threshold detector with Gaussian input decision variables as follows:
1 2
p
(7:90)
Pe Q
2 2
324
Demodulation
(7:91)
1
0
x1
1
x1
(7:92)
x2 x2 =22
e 2 dx2 dx1
2
x21 =2 2
0
Let x 2x1 be a new integral variable, and then Pe in (7.93) can be written as
1
x x2 a2 =42 ax
2
2
Pe ea =4
dx
e
I0
2
2 2
0 4
(7:94)
Where 2 2 2 . The integral on the right-hand side of (7.94) is the integral of the Rice
probability density function with parameter a2 and 2 over its entire range; hence, it is equal
to 1. Thus,
1 2 2
Pe ea =4
2
(7:95)
Substituting a2 and 2 from (7.58) into (7.95) yields the error probability for the maximum
detector with input Rice and Rayleigh decision variables as follows:
1
Pe eE=2N0
2
(7:96)
325
(7:97)
(7:98)
(7:99)
Substituting (7.98) and (7.99) into (7.97), and making use of (7.58) and (7.59) for Rice and
Rayleigh random variables, we arrive at the following result:
1
H2
I0 ax
2
ax
1
(7:100)
9
I0 22 H1
which implies
I0
ax H2 ax
1
2
9 I0 2
2 H1
(7:101)
x1 9 x 2
(7:102)
H1
The above results should hold for the case when X1 and X2 are chi-squared random
variables with probability density function given in (7.64) and (7.65). This is expected because
squaring the two positive decision variables for comparison should not alter the outcome of the
maximum detector. Thus, for non-central chi-squared and central chi-squared decision variables, the bit error probability of the maximum detector is also given by (7.96), that is,
1
Pe eE=2N0
2
(7:103)
326
Demodulation
s(t) + n(t)
Signal
processor
Threshold
detector
H2
<
> VT
H1
VT
Figure 7.13
X1
s(t) + n(t)
Maximum
detector
Signal
processor
2
Figure 7.14
X2
H2
x1 < x2
>
H1
Coherent PSK
As indicated by (6.13), PSK employs two signals s1 t st : H1 and s2 t st : H2 ,
where st Apt cos 2fc t. Therefore, the one-path coherent demodulator shown in
Figure 7.15 can be used to recover the information bits. As usual, we denote Tb as the bit
time. Also, without loss of generality we assume that the pulse shape pt that represents a
bit is a squared pulse of unit amplitude and duration Tb . The coherent carrier obtained from
the carrier recovery circuit is denoted as 2 cos 2fc t if the matched lter is employed to
be consistent with Figure 7.4. On the other hand, the recovered carrier is denoted as
2pt cos 2fc t when the integrator is used. Obviously, the integration time is Tb . The bit
timing recovery circuit provides the clock signal with frequency 1=Tb to sample the output
voltage of the matched lter or the integrator. The detection threshold is set to zero volts.
Table 7.1 presents all the parameters related to Figure 7.15, where Eb is the bit energy.
Using these parameters in (7.74), we obtain the bit error probability Pb as follows:
m m
1
2
Pb Q
2
r
(7:104)
2Eb
Q
N0
where Eb =N0 is the bit energy-to-noise density ratio, commonly referred to as the signal-tonoise ratio per bit (SNR per bit). Figure 7.16 plots the bit error probability Pb versus Eb =N0
(dB) for coherent PSK.
327
Table 7.1 Parameters for coherent PSK using (7.23) and (7.24)
Hypothesis
H1
H2
Signal
m1 = E(X|H1)
s1(t) = s(t)
s2(t) = s(t)
Eb
Matched
filter
h(t) = Ap(T t)
or
integrator
s(t) + n(t)
Figure 7.15
m2 = E(X|H2)
Noise variance, 2
Eb
2 EbN0/2
2 EbN0/2
t = Tb
X
Threshold
detector
VT = 0
H2
<
> 0
H1
Bit timing
recovery
Pb = Q(2Eb / N0)
101
102
103
Pb
104
105
106
107
108
109
Figure 7.16
6
Eb/N0(dB)
10
12
328
Demodulation
Coherent DPSK
DPSK can be coherently demodulated and differentially decoded as shown in Figure 7.17.
The demodulated bit ci is modulo-2 added to the previously demodulated bit ci1 to provide
the data bit di , that is,
di ci ci1
(7:105)
Since ci ci1 ci ci1 , inverted data caused by phase ambiguity in the recovered
carrier (i.e., the output of the carrier recovery circuit is cos 2fc t instead of cos 2fc t) is
re-inverted and the phase ambiguity is resolved. Note that by substituting ci di ci1 of
(6.26) into (7.105) one gets back that data bit di , since di di ci1 ci1 .
DPSK
i +
ci
di
0
1
2
3
4
5
6
2
3
3
4
4
0
0
1
0
0
1
1
0
0
1
1
0
1
0
PSK
demodulator
ci
di {0, 1}
+
ci1
1 bit delay
Figure 7.17
DPSK demodulator.
329
ci
0
1
2
3
4
5
6
2
2
4
3
1
0
0
1
1
1
0
di
1
1
0
1
0
0
1
gerrors
gerrors
t = Tb
PD or
APD
Figure 7.18
Transimpedance
amplifier
Tb
Threshold
detector
{0, 1}
2Eb
2Eb
1Q
Pb 2Q
N0
N0
(7:106)
Apparently, at high bit energy-to-noise density ratio Eb =N0 , the bit error probability of
coherent DPSK is approximately twice that of coherent PSK, as illustrated by Table 7.3.
>
1
<
21 re
it
(7:107)
2
>
re
>
> i2 t RA
pt is t : H2
:
2
1 re
where hypotheses H1 and H2 represent bits 1 and 0, respectively, and is t is the shot noise
current. The transimpedance amplier converts the photocurrent into a voltage signal xt at
330
Demodulation
its output. The amplier transimpedance is Z0 and its internal noise current is ia t . We can
write the output voltage signal xt as
xt Z0 it ia t
Substituting (7.107) into (7.108) yields
8
Z0 RA2
>
>
x
pt Z0 in t : H1
>
1
<
21 re
x t
2
>
re
>
> x2 t Z0 RA
p t Z 0 i n t : H 2
:
2
1 re
(7:108)
(7:109)
where
i n t i s t i a t
(7:110)
is the total
2 shotnoise plus amplier noise current2 and in t has a spectral density designated
as N0 A =Hz . Note that the unit is ampere /hertz and not watt/hertz. The integrator
integrates xt for one bit time Tb and the sampler produces a decision variable X at the
input of the threshold detector
8
Z0 RA2 Tb
>
>
N : H1
<
21 re
X
(7:111)
2
re
>
Z
RA
T
>
0
b
:
N : H2
1 re
2
The conditional mean value of X and the noise sample N are given as
m1 E f X jH1 g
Z0 RA2 Tb
21 re
Z0 RA2 Tb
re
m2 E f X jH2 g
2
1 re
Tb
N
Z0 in t dt
(7:112)
(7:113)
(7:114)
The noise voltage Z0 in t can be modeled as AWGN with zero mean and power spectral
density Z02 N0 (W/Hz). Thus, the variance 2 of the noise sample N in (7.114) is
2 Z02 N0 Tb
(7:115)
Substituting (7.112) and (7.113) into (7.73), we obtain the optimum threshold for the
threshold detector as follows:
m1 m2 Z0 RA2 Tb
VT
(7:116)
2
4
Note that the average received power P of the lightwave ASK signal is given by (6.28) as
1
A2
1
A2 re
P
2 21 re
2 21 re
(7:117)
A2
331
Combining (7.116) and (7.117), we get the optimum threshold in terms of average power as
VT Z0 RPTb
(7:118)
The minimum bit error probability is given by (7.74) with the help of (7.112), (7.113),
(7.115), and (7.117) as follows:
r
1 re Tb
RP
(7:119)
Pb Q
1 re N0
1
9:11 Gb=s
Tb
Coherent FSK
Coherent FSK can be conceptually demodulated by a two-path demodulator shown in
Figure 7.19.
t = Tb
Tb
0
s(t) + n(t)
A cos 2
X1
1
fc
t
2T
b
1
A cos 2 fc +
t
2T
b
X2
t = Tb
Tb
0
Figure 7.19
Maximum
detector
H2
< x2
x1 >
H1
332
Demodulation
Without loss of generality we assume that the carrier phase n in (6.42) is zero radian, or
equivalently, 1 2 0 radian for (6.36). Using the result for the maximum detector with
Gaussian decision variables in Section 7.6, we obtain via (7.84) and (7.85)
1 EfX1 jH1 g Eb
(7:120)
2 EfX2 jH2 g Eb
(7:121)
(7:122)
(7:123)
The minimum bit error probability of coherent FSK can be obtained via (7.90) as follows:
r
Eb
Pb Q
(7:124)
N0
By comparing (7.124) with (7.104), we observe that coherent FSK requires 3 dB more in
Eb =N0 (a factor of 2) to achieve the same bit error probability as coherent PSK. This is the
reason why coherent FSK is not employed in practice.
(7:125)
333
A cos 2fct
t = (2n + 1) Tb
(2n + 1)Tb
(2n 1)T
X1
H2
<
>0
H1
Threshold
detector
{0, 1}
t
cos
2Tb
VT = 0
t
sin
2Tb
t = (2n + 2) Tb
(2n + 2)Tb
2nT
X2
H2
<
> 0
H1
Threshold
detector
A sin 2fct
Figure 7.20
Coherent precoded MSK demodulator. The integration time is 2Tb reecting the fact that an I-bit or a Q-bit spans an
interval of 2Tb seconds, as seen in Figures 6.17 and 6.20.
Coherent
precoded
MSK
demodulator
cn
1 bit delay
Figure 7.21
dn
{0, 1}
cn 1
2Tb
t
t
C
cos
cos 2fc t sin
sin 2fc t dt
2Tb
2Tb
0
1 2Tb
t
sin
sin 4fc t dt
4 0
Tb
1 2Tb
1
1
cos 2 2fc
t cos 2 2fc
t dt
8 0
2Tb
2Tb
(7:126)
0
where fc is assumed to be an integer multiple of 1=Tb . If fc is not an integer multiple of 1=Tb
but fc 1=Tb , then the I-channel and Q-channel are quasi-orthogonal, which is approximately orthogonal for all practical purposes. Due to orthogonality, the I-noise and Q-noise
are both uncorrelated, and since they are Gaussian, they are also statistically independent.
334
Demodulation
2Tb
0
t
A sin 2fc t sin
nt dt
2Tb
(7:128)
Tb 2Tb
t
2 2
EfN1 N2 g A
Efnt n g cos
cos 2fc t sin
sin 2fc dt d
2Tb
2Tb
Tb 0
Tb 2Tb
N0
t
2 2
A
t cos
cos 2fc t sin
sin 2fc dt d
2Tb
2Tb
2
Tb 0
2Tb
1 2 2
t
t
cos
cos 2fc t sin
sin 2fc t dt
A N0
2
2Tb
2Tb
0
1
2 A2 N0 C 0
2
(7:129)
by virtue of (7.126). Thus, N1 and N2 are uncorrelated and, since they are Gaussian random
variables, they are independent. It is obvious from (7.127) and (7.128) that both N1 and N2
have zero mean since nt has zero mean. Let us calculate the variances of N1 and N2 . From
(7.127), we get
Tb Tb
2
t
2
2 2
E N1 A
Efnt n g cos
cos
cos 2fc t cos 2fc dt d
2Tb
2Tb
Tb Tb
T b Tb
N0
t
2 2
t cos
A
cos
cos 2fc t cos 2fc dt d
2Tb
2Tb
Tb Tb 2
Tb
1 2 2
t
2
A N0
cos
cos2 2fc t dt
2
2T
b
Tb
1
(7:130)
2 A2 N0 Tb
4
Similarly, it can be shown that the variance of N2 is the same as that of N1 , that is,
1
2 E N22 2 A2 N0 Tb
4
(7:131)
335
To evaluate the bit error probability of either coherent MSK or coherent pre-coded MSK, we
need to know the statistic of the decision variables X1 and X2 at the input of the threshold
detectors in Figure 7.20. Obviously, the variance of X1 or X2 is just the variance 2 of noise
sample N1 or N2 as indicated in (7.130) or (7.131). It remains to nd the mean value of X1 and the
mean value of X2 . The waveform of I-bits can be written with the help of Figure 6.20 as follows:
t
(7:132)
cos 2fc t; 2n 1Tb t 52n 1Tb
sI t A2n1 cos
2Tb
t
sQ t A2n sin
(7:133)
sin 2fc t; 2nTb t 52n 2Tb
2Tb
where 2n1 and 2n take on values of 1 given hypothesis H1 , or 1 given hypothesis H2 . The
mean value of X1 given H1 is
2n1Tb
t
dt
sI t A cos 2fc t cos
m1 EfX1 jH1 g
2Tb
2n1Tb
2n1Tb
t
2
2
(7:134)
A cos
cos2 2fc t dt
2T
b
2n1Tb
1
A2 Tb
2
The mean value of X1 given H2 is simply
1
m2 EfX1 jH2 g A2 Tb
2
(7:135)
1
m1 EfX2 jH1 g A2 Tb
2
(7:136)
1
m2 EfX2 jH2 g A2 Tb
2
(7:137)
Similarly,
Substituting (7.130), (7.134), and (7.135) into (7.74) yields the I-bit error probability
m m
1
2
Pb I Q
2
s!
A 2 Tb
Q
(7:138)
N0
r
2Eb
Q
N0
where Eb A2 Tb =2 is the energy of the MSK bit. Note that both the I-bit and Q-bit have 2Tb
duration, hence both have twice the energy of the MSK bit. Similarly, the Q-bit error
probability can be obtained by substituting (7.131), (7.136), and (7.137) into (7.74)
r
2Eb
Pb Q Q
(7:139)
N0
336
Demodulation
Because the I-noise and Q-noise are statistically independent, and if we assume the I-bits
and Q-bits are equally likely, then the bit error probability of coherent precoded MSK is
1
1
Pb Pb I Pb Q
2 r 2
2Eb
: coherent precoded MSK
Q
N0
(7:140)
Due to differential decoding, the bit error probability of coherent MSK suffers a penalty of
approximately a factor of two and is given by
r
2Eb
Pb 2Q
: coherent MSK
(7:141)
N0
Note that by observing Figures 6.21 and 7.20, we can remove the inverter dn preceding
the differential encoder in the MSK modulator, and the one that follows the differential
decoder in the coherent MSK demodulator. Thus, the removal of dn does not change the
operation since inverting the data at the modulator and re-inverting it at the demodulator
amounts to no change. If we map the binary data 0 into 1 and 1 into 1 via dn ! 1 2dn
instead of dn ! 2dn 1, as in (6.57), we would not indeed have to add the inverter dn to
both modulator and demodulator for coherent MSK.
Coherent GMSK discussed in Section 6.8 employs Gaussian pulse shape g t pt ht ,
where pt is the pulse of unit amplitude and duration Tb , and ht is the impulse response of
the Gaussian lter given in (6.76). Thus, GMSK can be coherently demodulated as suggested
in [3,4] by the demodulator in Figure 7.22 employing a Costas loop.
The bit error probability for coherent GMSK [3,4] is given by
r
2Eb
Pb Q
(7:142)
N0
LPF
/2
Loop
filter
IF local
oscillator
/2
LPF
Clock
recovery
Figure 7.22
337
where 0:68 for BTb 0:25 as described in Section 6.8. This yields a degradation of
1.6 dB relative to coherent precoded MSK due to intersymbol interference.
(7:143)
where Eb is the bit energy and N0 =2 is the power spectral density of the input noise.
Since MSK is a special type of FSK, both demodulators in Figures 7.23 and 7.24 can be
employed to noncoherently demodulate MSK. For MSK, the frequencies f1 and f2 are given
by f1 fc 1=4Tb and f2 fc 1=4Tb per (6.60). The bit error probability in (7.143) also
applies to noncoherent MSK.
Both FSK and MSK are digital FM signals, meaning that the modulating information
signal is digital. Therefore, both FSK and MSK can be demodulated by a slope detector used
in FM demodulation [5,6]. Figure 7.25 shows a generic demodulator using slope detector.
The IF lter bandwidth is chosen to avoid the distortion of the modulated signal. The bit
error probability is given in [7] as follows:
Pb eEb =N0
(7:144)
Comparing (7.144) with (7.143), we observe that the demodulation with a slope detector
is superior to noncoherent demodulation with a matched lterenvelope detector or a
t = Tb
Matched
filter, f1
Envelope
detector
X1
Maximum
detector
t = Tb
Matched
filter, f2
Figure 7.23
X2
Envelope
detector
x1
H2
< x2
>
H1
338
Demodulation
t = Tb
Tb
( )2
A cos 2f1t
A sin 2f1t
t = Tb
Tb
( )2
X1
Maximum
detector
t = Tb
Tb
0
A cos 2f2t
A sin 2f2t
t = Tb
Tb
0
Figure 7.24
X2
( )2
x1
H2
<
> x2
H1
( )2
IF
filter
Slope
detector
m(t)
Tb
Threshold
detector
H2
x < 0
>
H1
cos 2f0t
Figure 7.25
Continuous-phase FSK or MSK demodulator with slope detector. The signal m(t) is the antipodal information signal.
quadrature correlationsquare law detector by 3 dB in bit energy-to-noise density ratio
(signal-to-noise ratio per bit). This improvement results from the fact that a slope detector
is affected by quadrature noise only. Thus, the in-phase noise is not a factor, and the noise as
seen by the threshold detector is 3 dB less in power.
For the case of
continuous-phase
FSK, coherent demodulation yields a bit error proba
p
bility of Pb Q Eb =N0 , which is larger than the bit error probability of a demodulator
with slope detector, Pb expEb =N0 . For example, at Pb 104 the required Eb =N0 for
coherent demodulation is 11.4 dB. On the other hand, the required Eb =N0 for demodulation
with slope detector is 9.2 dB. This happens because the slope detector extracts the baseband
signal mt , as seen in Figure 7.25, which is an antipodal signal. Then, mt is optimally
detected via a threshold detector. Note that an antipodal signal is the complex envelope of a
PSK signal; therefore, threshold detection of an antipodal signal yields the same bit error
probability as that of PSK. Since the slope detector output noise is no longer Gaussian, this
339
leads
to a degradation
in the bit error probability and compared to that of PSK, which is
p
Q 2Eb =N
.
Thus,
the
bit error
probability
of the demodulator with slope detector falls
0
p
p
between Q Eb =N0 and Q 2Eb =N0 .
Noncoherent DPSK
Recall from Section 6.4 that when the ith data bit di 0 is transmitted, its differentially
encoded bit ci remains the same as the previously encoded bit ci1 . In other words, the
waveform for two consecutive encoded bits ci1 ci can be written as
s1 t si1 t si t ;
i 1Tb t 5i 1Tb
(7:145)
where si1 t and si t are the transmitted PSK waveforms for the encoded bits ci1 and ci ,
respectively. Assume that the carrier phase varies slowly over any two consecutive bits,
then we can express si1 t and si t as follows:
si1 t Apt i 1Tb cos2fc t
(7:146)
(7:147)
as seen in Example 6.20. On the other hand, when the ith data bit di 1 is transmitted, its
differentially encoded bit ci is the complement of the previously encoded bit ci1 , that is,
ci ci1 . Thus, the waveform for two consecutive encoded bits ci1 ci can be written as
s2 t si1 t si t ;
i 1Tb t 5i 1Tb
(7:148)
Clearly, s1 t and s2 t are orthogonal over the two-bit interval 2Tb , that is
i1Tb
i1Tb
s1 t s2 t dt
i1Tb
i1Tb
s2i1 t dt
i1Tb
i1Tb
s2i t dt 0
(7:149)
since both si1 t and si t have the same bit energy Eb A2 Tb =2.
The above observations lead us to the following conclusion. The data bit di 0 is
represented by the waveform s1 t , and di 1 is represented by s2 t . Thus, DPSK is
truly an orthogonal modulation over any interval of 2Tb . This enables DPSK to be demodulated noncoherently as shown in Figure 7.26. Note that the energy of s1 t or s2 t is 2Eb
since they span an interval of 2Tb . Thus, according to (7.56) and (7.57), the signal-to-noise
ratio at the output of the matched lter is
SNR0
2 2Eb 2
2Eb
2 2Eb N0
N0
(7:150)
which is 3 dB higher than an FSK signal. Consequently, the bit error probability of noncoherent DPSK is given by
1
Pb eEb =N0
(7:151)
2
as compared to (7.143) for FSK.
340
Demodulation
t = i Tb, i = 2, 3,
Matched
filter for
s1(t)
Envelope
detector
X1
t = i Tb,
i = 2, 3,
Matched
filter for
s2(t)
Figure 7.26
X2
H2
x1 < x2
>
H1
Envelope
detector
+
iTb
A sin 2 fct
( )2
(i1)Tb
X2c
1 bit
delay
A cos 2 fct
iTb
t = iTb
(i1)Tb
X1c
X2s
( )2
X2
Maximum
detector
X1
( )2
H2
x1 < x2
>
H1
t = iTb
1 bit
delay
+
+
Figure 7.27
Maximum
detector
X1s
( )2
341
si(t)
X1c
X1s
X2c
X2s
X1
X2
di
0
1
2
3
4
5
6
Ap(t)cos(2fct + )
Ap(t Tb)cos(2fct + )
Ap(t 2Tb)cos(2fct + )
Ap(t 3Tb)cos(2fct + )
Ap(t 4Tb)cos(2fct + )
Ap(t 5Tb)cos(2fct + )
Ap(t 6Tb)cos(2fct + )
/2 A2Tbcos
A2Tcos
0
0
A2Tbcos
0
A2Tbcos
/2 A2Tbsin
A2Tbsin
0
0
A2Tbsin
0
A2Tbsin
/2 A2Tbcos
0
A2Tbcos
A2Tbcos
0
A2Tbcos
0
/2 A2Tbsin
0
A2Tbsin
A2Tbsin
0
A2Tbsin
0
2/4 A4T2
2 A4 Tb2
0
0
2 A4 Tb2
0
2 A4 Tb2
2 =4A4 Tb2
0
2 A4 Tb2
2 A4 Tb2
0
2 A4 Tb2
0
0
1
1
0
1
0
In the following discussion we will analyze the performance of the noncoherent demodulator in Figure 7.27 by considering the AWGN that accompanies the received signal. Let us
assume that hypothesis H1 represents the data bit 0, that is, we receive the signal
s1 t si1 t si t , i 1Tb t 5i 1Tb in (7.145). Then, given the bit energy
Eb A2 Tb =2, we have
0
00
X1c A2 Tb cos N1c
A2 Tb cos N1c
2
2
(7:152)
A2 Tb cos N1c 2Eb cos N1c
0
00
A2 Tb sin N1s
X1s
A2 Tb sin N1s
2
2
2
A Tb sin N1s
2Eb sin N1s
2
(7:153)
where
0
N1c
0
N1s
iTb
i1Tb
iTb
i1Tb
00
N1c
00
N1s
i1Tb
nt A cos 2fc t dt
(7:154)
nt A sin 2fc t dt
(7:155)
iTb
i1Tb
iTb
and
0
00
N1c N1c
N1c
;
0
00
N1s N1s
N1s
(7:156)
Given the input noise nt with power spectral density N0 =2 W=Hz and, hence, autocor0
00
0
relation Efnt n g N0 =2t , we can calculate the variance of N1c
, N1c
, N1s
, and
00
2 2
N1s to be A Tb N0 =4. This, in turn, yields the variance of N1c and N1s in (7.156) to be
0
00
0
00
2 A2 Tb N0 =2 2 Eb N0 . Note that N1c
, N1c
, N1s
, and N1s
are statistically independent
because they are uncorrelated zero mean Gaussian random variables. Similarly, we have
0
00
N1c
X2c N1c
(7:157)
0
00
N1s
X2s N1s
(7:158)
342
Demodulation
Thus, both X2c and X2s are independent Gaussian random variables with zero mean and
variance 2 A2 Tb N0 =2 2 Eb N0 .
The decision variables X1 and X2 are therefore given by
2
X1s2
X1 X1c
(7:159)
2
X2s2
X2 X2c
(7:160)
We observe that X1 is the sum of two squared Gaussian random variables with means
2Eb cos and 2Eb sin , and variance 2 Eb N0 . Thus, X1 is a non-central chi-squared
random variable with two degrees of freedom whose probability density function is given
in (7.64) with X1 replacing Y, and a2 42 Eb2 and 2 2 Eb N0 . On the other hand, the
decision variable X2 is the sum of two squared Gaussian random variables with zero mean
and variance 2 Eb N0 . Thus, X2 is a central chi-squared random variable whose probability
density function is given in (7.65) with X2 replacing Y, and 2 2 Eb N0 . The conditional bit
error probability given H1 can be derived in a similar fashion to (7.91)(7.96). It is the same
as the conditional bit error probability given hypothesis H2 that corresponds to data bit 1 as
dictated by symmetry. The bit error probability of noncoherent DPSK using the demodulator
in Figure 7.27 is given by (7.103), which is the same as (7.96) with E 2Eb using the fact
that a2 =4 2 in (7.95) is equal to Eb =N0 . Thus
1
Pb eEb =N0
2
(7:161)
In addition to the noncoherent demodulator in Figure 7.27, DPSK can also be noncoherently
demodulated by another demodulator, such as the one shown in Figure 7.28, called a delayand-multiply demodulator.
t = iTb
Xi
iTb
(i1)Tb
A cos 2 fct
1 bit delay
Xi 1
t = iTb
iTb
(i1)Tb
A sin 2 fct
Figure 7.28
Yi
1 bit delay
Yi 1
Threshold
detector
H2
<
> 0
H1
343
s(t)
Xi
Xi1
Yi
Y i1
di
0
1
2
3
4
5
6
Ap(t)cos(2fct+)
Ap(t Tb)cos(2fct + )
Ap(t 2Tb)cos(2fct + )
Ap(t 3Tb)cos(2fct + )
Ap(t 4Tb)cos(2fct + )
Ap(t 5Tb)cos(2fct + )
Ap(t 6Tb)cos(2fct + )
/2 A2Tbcos
/2 A2Tbcos
/2 A2Tbcos
/2 A2Tbcos
/2 A2Tbcos
/2 A2Tbcos
/2 A2Tbcos
0
/2 A2Tbcos
/2 A2Tbcos
/2 A2Tbcos
/2 A2Tbcos
/2 A2Tbcos
/2 A2Tbcos
/2 A2Tbsin
/2 A2Tbsin
/2 A2Tbsin
/2 A2Tbsin
/2 A2Tbsin
/2 A2Tbsin
/2 A2Tbsin
0
/2 A2Tbsin
/2 A2Tbsin
/2 A2Tbsin
/2 A2Tbsin
/2 A2Tbsin
/2 A2Tbsin
0
2 =4 A4 Tb2
2 =4 A4 Tb2
2 =4 A4 Tb2
2 =4 A4 Tb2
2 =4 A4 Tb2
2 =4 A4 Tb2
0
1
1
0
1
0
Xi A2 Tb cos Nc;i
2
(7:162)
(7:163)
Yi
A2 Tb sin Ns;i
2
(7:164)
(7:165)
where
Nc;i
i1Tb
Nc;i1
iTb
Ns;i
i1Tb
iTb
Ns;i1
iTb
i1Tb
iTb
i1Tb
nt A cos 2fc t dt
(7:166)
nt A sin 2fc t dt
(7:167)
Note that Nc;i , Nc;i1 , Ns;i , Ns;i1 are independent Gaussian random variables with
zero mean and variance 2 A2 Tb N0 =4 2 Eb N0 =2, where Eb A2 Tb =2 is the bit energy.
The decision variable X at the input of the threshold detector in Figure 7.28 is
X Xi1 Xi Yi1 Yi
(7:168)
344
Demodulation
Nc;i1 Nc;i
Nc;i1 Nc;i 2
A2 Tb cos
2
2
2
2
2
2
2
Ns;i1 Ns;i
Ns;i1 Ns;i 2
A Tb sin
2
2
2
2
2
(7:169)
Nc;i1 Nc;i
2
2
(7:170)
N2
Ns;i1 Ns;i
2
2
(7:171)
N3
Nc;i1 Nc;i
2
2
(7:172)
N4
Ns;i1 Ns;i
2
2
(7:173)
Thus, we can express the decision variable X in terms of the new noise variables N1 , N2 , N3 ,
and N4 as follows:
2
2
X A2 Tb cos N1
A2 Tb sin N2 N32 N42
(7:174)
2
2
Furthermore, let us dene the non-central chi-squared and central chi-squared random
variables X1 and X2 as follows:
2
2
X1 A2 Tb cos N1
A2 Tb sin N2
(7:175)
2
2
X2 N32 N42
(7:176)
(7:177)
(7:178)
345
e
T
RP
Q 1r
=N
b
0
1re
p
Q Ep
=N0
b
2Q
2E =N
pb 0
Q p2E
b =N0
Q 2Eb =N0
1 Eb =2N0
2e
1 Eb =2N0
2e
eEb =N0
1 Eb =N0
2e
Bandwidth
2=Tb
2=Tb
2=Tb
3=Tb
3=2Tb
3=2Tb
3=2Tb
3=Tb
3=2Tb
3=2Tb
2=Tb
(7:179)
346
Demodulation
Pb comparison
100
102
Pb
104
106
108
1 e Eb /N0
2
Q(2Eb / N0)
1010
Figure 7.29
6
Eb /N0(dB)
10
12
(7:180)
where N is the zero mean Gaussian noise vector with independent components whose
variance is 2 . Thus, the probability density function of X given Hi is
fX xjHi
L
Y
k1
1
1
p L e22
2
L
Y
2
1
2
p exk sik =2
k1 2
PL
2
fXk xk jHi
k1
(7:181)
xk sik
We wish to design a detector that selects a hypothesis Hi based on the minimum error
probability criterion. This is equivalent to selecting Hi corresponding to the maximum of the
set of a-posteriori probabilities PrHi jx; i 1; 2; . . . ; M . This is called the maximum
a-posteriori probability criterion (MAP), as rst discussed in Section 7.5. When all symbols
are equally likely, then the a-priori probabilities PrHi , i = 1, 2, . . . , M, are equal to 1/M and
the MAP criterion becomes the maximum likelihood (ML) criterion. Using the Bayes rule,
we can express the a-posteriori probabilities PrHi jx as follows:
fX xjHi PrHi
;
i 1; 2; . . . ; M
(7:182)
fX x
P
We note that PrHi 1=M , and fX x M
i1 fX xjHi PrHi is completely independent
of Hi , that is, fX x is independent of which symbol is transmitted. Therefore, nding the
PrHi jx
347
ln fX xjHi L ln
(7:183)
p
where the term L ln 2 is common to all ln fX xjHi . Hence, the maximum of the set
ln fX xjHi corresponds to the minimum of the set of Euclidean distances
v
u L
uX
(7:184)
xk sik 2 ; i 1; 2; . . . ; M
d x; si kx si k t
k1
The above analysis concludes that an optimum detector based on the maximum likelihood
criterion selects the signal vector si that is closest in Euclidean distance to the sampled value
x of the decision vector X. Hence, the term minimum Euclidean distance detector is used.
x s1 2 9 x s2 2
(7:185)
H2
which is identical to (7.80). Hence, the threshold detector is equivalent to the minimum
Euclidean distance detector for M = 2.
Figure 7.30 illustrates the minimum Euclidean distance detector. This type of detector is
employed for coherent M-ary demodulation of signals, such as PSK, QPSK, MPSK, and
X1
X2
XL
Figure 7.30
.
.
.
Select signal si
closest in
Euclidean
distance to x
si
348
Demodulation
PrX 2 Ri jHi
fX xjHi dx
(7:186)
Ri
The symbol detection probability can be obtained via the total probability theorem as
follows:
Pd
M
X
i1
M
1X
M i1
(7:187)
fX xjHi dx
Ri
M
1X
M i1
fX xjHi dx
(7:188)
Ri
It is not feasible to evaluate (7.188) for a large signal set such as MQAM. Fortunately,
we can provide a tight upper bound for the symbol error probability by using the famous
union bound. Let Ai ; i 1; 2; . . . ; M , be a set of M events, then the union bound states that
!
M
M
X
[
Ai
PrAi
(7:189)
Pr
i1
i1
Returning to the minimum Euclidean distance detector and assume that we have hypothesis
H1 , that is, s1 is transmitted. An error occurs if one of si ; i 2; 3; . . . ; M , is closer to the
sampled vector X than s1 . Dene the event Ai as the event that si is closer to X than s1 given
that s1 was transmitted. The symbol error probability given H1 is
!
M
M
[
X
Ai
PrAi
(7:190)
PejH1 Pr
i2
i2
Hence, we only need to evaluate PrAi , which is the pairwise error probability in selecting
between two signals si and s1 , that is,
349
PrAi Prs1 ! si Pr kX si k2 5kX s1 k2 H 1
(7:191)
Let s1 EfXjH1 g and si EfXjHi g be two sampled L-dimensional vectors. Let N Nj
be the L-dimensional noise vector whose components are independent Gaussian random
variables with zero mean and variance 2 . Assume hypothesis H1 , that is, s1 was the
transmitted vector, then X s1 N. We are interested in evaluating (7.191). Substituting
X into (7.191), we get
PrAi Prs1 ! si Prks1 si N k2 5ks1 s1 N k2 H1
(7:192)
Prks1 si N k2 5kN k2 H1
Since
ks1 si N k2 ks1 si k2 2s1 si t N kN k2
(7:193)
Then
PrAi Prs1 ! si Prks1 si k2 2s1 si t N kN k2 5kN k2 jH 1
(7:194)
Dene
Z
si s1 t N
ksi s 1 k
(7:195)
Note that si s1 t =ksi s1 k is a unit length vector, therefore, the random variable Z in
(7.195) is a Gaussian random variable with zero mean and variance 2 . Substituting (7.195)
into (7.194) we get
1
PrAi Prs1 ! si Pr Z 4 ksi s1 kH1
2
(7:196)
k si s 1 k
Q
2
via (7.29). Dene di;1 as the Euclidean distance between two vectors si and s1 , that is,
di;1 ksi s1 k
(7:197)
di;1
PrAi Prs1 ! si Q
2
(7:198)
Then
Hence, the pairwise error probability involving two vectors s1 and si in a Gaussian channel
depends solely on their Euclidean distance and the variance of noise.
If we substitute PrAi in (7.198) into the union bound (7.190), we obtain the upper bound
on the symbol error probability given H1 as
350
Demodulation
PejH1
M
X
di;1
Q
2
i2
(7:199)
The Q function decreases rapidly as its argument increases, thus, the left-hand side of
(7.199) is dominated by the terms on the right-hand side with the smallest argument. Let us
dene dmin;1 as the smallest Euclidean distance of the set di;1 ; i 2; 3; . . . ; M,
dmin;1 min di;1
i
(7:200)
and N1 as the number of signals that are dmin;1 distance from s1 . Then, we can approximate
(7.199) as follows:
dmin;1
(7:201)
PejH1 N1 Q
2
Assuming all hypotheses are equally likely, then the average symbol error probability is
Pe
M
X
PejHk PrHk
k1
M
1X
dmin; k
Nk Q
M k1
2
(7:202)
Using the same argument as above, we dene dmin as the minimum Euclidean distance
between any signal pair and Nn as the average number of symbols at the minimum Euclidean
distance dmin , that is, the average number of nearest neighbors
dmin min dmin; k
k
Nn
M
1X
Nk
M k1
Then, the nal approximation for the symbol error probability is given by
dmin
Pe Nn Q
2
(7:203)
(7:204)
(7:205)
351
X1
X2
.
.
.
Select the
largest
input
si
XM
Figure 7.31
(7:208)
352
Demodulation
Xj Nj ; j 2; 3; . . . ; M
(7:209)
Note that Xj are independent Gaussian random variables with the following probability
density functions:
2
1
2
fX1 x1 p ex1 jjs1 jj =2
2
2
2
1
fXj xj p exj =2 ;
2
j 2; 3; . . . ; M
(7:210)
(7:211)
It is seen that a correct detection occurs when every xj is smaller than x1 , where x1 is a
sampled value of X1 . Such an event occurs with a conditional correct detection probability
Pcjx1 PrfX2 5x1 \ X3 5x1 \ \ XM 5x1 jH1 g
PrX2 5x1 jH1 M 1
(7:212)
We have used the fact that Xj ; j 2; 3; . . . ; M , are independent and identically distributed
random variables. Via (7.26) we get
x1
x
1
2
2
1
PrX2 5x1 jH1 p
(7:213)
ey =2 dy 1 Q
2 1
Substituting (7.213) into (7.212) yields
h
x iM 1
1
Pcjx1 1 Q
(7:214)
By taking the expected value of Pcjx1 with respect to x1 , using the probability density
function in (7.210), we obtain the correct detection probability given hypothesis H1 :
1
h
x iM 1
2
1
2
1
p ex1 jjs1 jj =2 1 Q
dx1
Pc
1 2
(7:215)
1 1 y2 =2
ks1 k M 1
p
e
1Q y
dy
2 1
Since all hypotheses are equiprobable, this is also the correct detection probability. The
symbol error probability Pe is simply 1 Pc, and by using (7.206), we get
1 1 y2 =2
dmin M 1
e
1 Q y p
dy
(7:216)
Pe 1 p
2 1
2
353
c (x, s1)
X1
X2
.
.
.
Calculate
M metrics
c (x, si)
i = 1, 2, , M
c (x, s2)
.
.
.
XL
Figure 7.32
c (x, sM)
Select the
largest
input
si
(7:217)
Since kxk2 is common to all ln fX xjHi , the maximum of the set ln fX xjHi corresponds to
the maximum of the set
cx; si 2xt si ksi k2 ;
i 1; 2; . . . ; M
(7:218)
The parameter cx; si is often called the correlation metric. Thus, a minimum Euclidean
distance detector actually computes the set of M metrics and selects the signal that corresponds to the largest metric. Figure 7.32 illustrates the alternative implementation of a
minimum Euclidean distance detector.
All other M 1 decision variables Xk, k 2; 3; . . . ; M , are Rayleigh variables with the
probability density function given by (7.59):
fXk xk
xk x2 =22
e k ;
2
xk 0
(7:220)
A correct detection occurs when every Xk is smaller than x1 , where x1 is a sample value of
X1 . Such event occurs with a conditional correct detection probability
Pcjx1 PrfX2 5x1 \ X3 5x1 \ \ XM 5x1 jH1 ; x1 g
PrX2 5x1 jH1 ; x1 M 1
(7:221)
354
Demodulation
by virtue of independence between all Xk and the fact that they are identically distributed. By
using (7.220) in (7.221) we obtain
Pcjx1
x1
M 1
fX2 x2 dx2
0
x1
x2 x2 =22
e 2
dx2
2
0
h
i
2
2 M 1
1 ex1 =2
M 1
(7:222)
1 ex1 =2
e
0
1
2
2
0
1 M
1
X
M 1 kx2 =22 x1 x2 a2 =2 2 ax1
e 1
1k
e 1
I0 2 dx1
2
k
0 k0
(7:223)
yec y I0 y dy
2 2
1 1=4c2
e
2c2
(7:224)
M
1
X
k
1
k0
M 1
k
k
1 k1
e
k1
2
a
2 2
(7:225)
M
1
X
k1
1
k1
M 1
k
k
1 k1
e
k1
2
a
2 2
(7:226)
We would also like to nd a union bound for Pe. Suppose s1 was transmitted, an error
occurs when X1 is smaller than all Xk, k 2; 3; . . . ; M . Thus
Pejx1 PrfX2 4x1 [ X3 4x1 [ [ XM 4x1 jH1 ; x1 g
M
X
Pr Xj 4x1 H1 ; x1
(7:227)
j2
The conditional probability Pr Xj 4x1 H1 ; x1 has been evaluated in (7.95). Furthermore,
since all signals are equiprobable, the symbol error probability is simply the same for all
signals si(t). Thus, via (7.95) we get the union bound
355
Signal
processor 1
s(t) + n(t)
.
.
.
Signal
processor L
Figure 7.33
Symbolto-bit
converter
{0, 1}
Select
the
largest
input
Symbolto-bit
converter
{0, 1}
Signal
processor 1
s(t) + n(t)
.
.
.
Signal
processor M
Figure 7.34
Minimum
Euclidean
distance
detector
M-path demodulator.
1
2
2
Pe M 1ea =4
2
(7:228)
which is the rst term of the summation on the right-hand side of (7.226).
356
Demodulation
t = Ts
Ts
X1
x1(t)
t = Ts
Ts
XL
xL(t)
Figure 7.35
Signal processor for L-path demodulator. The received signal st represents one of the M signals si t,
i 1; 2; . . . ; M, and nt is noise.
si t
L
X
sik xk t ;
i 1; 2; . . . ; M
(7:229)
k1
(7:230)
1
(7:231)
We are interested in the statistics of N, and hence, the statistics of X given si . As always,
we assume that nt is wide-sense stationary AWGN with zero mean and power spectral
density N0 =2 W=Hz. The noise autocorrelation function is Efnt n g
N0 =2t . The noise random variables Nk are given by
Ts
Nk
nt xk t dt; k 1; 2; . . . ; L
(7:232)
0
357
Ts Ts
0
Efnt n gxk t x dt d
(7:233)
0
Ts
N0
xk t x t dt
2 0
N0
k
2
where k 0, k 6 , and k 1, k . It is quite obvious that the noise variables Nk are
mutually uncorrelated. Since they are Gaussian, they are also statistically independent. Their
variance is given by (7.233) by setting k , that is,
N0
2 E Nk2
2
(7:234)
Thus, the decision variables Xk sik Nk , k 1; 2; . . . ; L, are also statistically independent Gaussian random variables with conditional mean sik and variance 2 N0 =2.
Figure 7.36 shows the complete block diagram of an L-path demodulator employing the
minimum Euclidean distance detector. Figure 7.37 shows an alternative L-path demodulator
employing the metric computer and maximum detector in Figure 7.32. Note that the
received signal X t si t nt is completely represented by the vector X si N.
In other words, the L components Xk sik Nk , k 1; 2; . . . ; L, of the vector X are the
only data based on X t useful for detecting the symbol represented by the signal si t . They
represent sufcient statistics that summarize X t for the purposes of detection of si t . It can
P
P
P
be shown that et X t Xk xk t si t nt sik xk t Nk xk t nt
k
k
k
P
Nk xk t is uncorrelated to Xj , that is, E et Xj 0, j 1; 2; . . . ; N . Furthermore, et
k
Ts
0
s(t) + n(t)
Minimum
Euclidean
distance
detector
x1(t)
t = Ts
Ts
XL
xL(t)
Figure 7.36
si
Symbolto-bit
converter
{0, 1}
358
Demodulation
t = Ts
Ts
0
s(t) + n(t)
X1
c(x. s1)
c(x. s2)
x1(t)
Metric
computer
Maximum
detector
si
t = Ts
XL
Ts
Symbolto-bit
converter
{0, 1}
c(x. sM)
xL(t)
Figure 7.37
MF
x1(Ts t)
Minimum
Euclidean
distance
detector
s(t) + n(t)
t = Ts
MF
xL(Ts t)
Figure 7.38
si
Symbolto-bit
converter
{0, 1}
XL
Matched lter implementation of the L-path demodulator with a minimum Euclidean distance detector.
signal. Sufcient statistics are important in digital communications because they allow
the receiver to be practically built. They render a continuous-time signal X t into a nitedimensional vector X that is good enough for the detection of the transmitted signal.
The correlators in Figures 7.36 and 7.37 can also be replaced by the matched lters as
indicated by Figures 7.38 and 7.39. The symbol error probability of the L-path demodulator
is given by (7.205). Note that a symbol has log2 M bits. Therefore, when M symbols are
encoded with a Gray code, adjacent symbols differ by only one bit. Since adjacent symbols
are closest in Euclidean distance, a symbol error due to noise results in the selection of one of
the adjacent symbols for the true symbol. Thus a symbol error contains only one bit error.
Gray code mapping thus results in the following approximate bit error probability:
Pe
Nn
dmin
Pb
Q
(7:235)
log2 M log2 M
2
359
t = Ts
MF
x1(Ts t)
X1
c(x. s1)
c(x. s2)
s(t) + n(t)
Metric
computer
Maximum
detector
t = Ts
MF
xL(Ts t)
Figure 7.39
XL
si
Symbolto-bit
converter
{0, 1}
c(x. sM)
Matched lter implementation of the L-path demodulator with the metric computer and maximum detector.
t = Ts
X1
Ts
0
s(t) + n(t)
x1(t)
Minimum
detector
si
t = Ts
Symbolto-bit
converter
{0, 1}
XM
Ts
0
xM(t)
Figure 7.40
si t
M
X
sik xk t sii xi t
k1
p
Es xi t ;
(7:236)
i 1; 2; . . . ; M
where Es is the signal (symbol) energy. Since all si(t) are mutually orthogonal, it is obvious
that all xi(t) are mutually orthonormal. Equation (7.236) implies the following coherent
demodulators shown in Figures 7.40 and 7.41.
p
Note that in the absence of noise, the output of the ith path sampler is Es and the
remaining outputs are all zero. Thus, the maximum detector just selects the signal vector si.
The symbol error probability of the M-path demodulator is given by the union bound (7.207)
or by the exact expression in (7.216). To calculate the bit error probability, we note that when
there is a symbol error, all M 1 wrong symbols are equiprobable with probability
Pe =M 1. Since each symbol has k log2 M bits, the probability distribution for i bit
k
errors in a wrong symbol is
Pe =M 1. Thus, the average number of bit errors per
i
wrong symbol is
360
Demodulation
t = Ts
X1
MF
x1(Ts t)
s(t) + n(t)
Minimum
detector
si
Symbolto-bit
converter
{0, 1}
t = Ts
XM
MF
xM(Ts t)
Figure 7.41
Pe
2k1
k
Pe
M 1
M 1
(7:237)
There are k bits in a wrong symbol, thus the average bit error probability is simply the righthand side of (7.237) divided by k:
Pb
M =2
Pe
M 1
(7:238)
Substituting the union bound in (7.207) into (7.238) yields the union bound for Pb of the
coherent M-path demodulator:
Pb
M
dmin
Q
2
2
The exact bit error probability is obtained via (7.216) and (7.238) as follows:
"
#
M=2
1 1 y2 =2
dmin M 1
1 p
Pb
e
1 Q y p
dy
M 1
2 1
2
(7:239)
(7:240)
361
t = Ts
Ts
( )2
x1(t)
t = Ts
Ts
( )2
X 21
Maximum
detector
x1(t, /2)
t = Ts
Ts
X 2M
si
Symbolto-bit
converter
( )2
xM(t)
t = Ts
Ts
( )2
xM(t, /2)
Figure 7.42
Noncoherent M-path demodulator employing the quadrature correlatorsquare law detector. The phase is the
unknown carrier phase.
t = Ts
x1(Ts t)
X1
Envelope
detector
s(t,) + n(t)
Maximum
detector
t = Ts
xM(Ts t)
Figure 7.43
Envelope
detector
si
Symbolto-bit
converter
{0, 1}
XM
Noncoherent M-path demodulator employing the matched lterenvelope detector. The phase is the unknown
carrier phase.
For example, if x1 t
We also note that
p
p
2=Ts cos 2f1 t, then we have x1 t; =2 2=Ts sin 2f1 t.
Ts
xi t; xi t dt cos
(7:242)
xi t; xi t; =2 dt sin
(7:243)
and
Ts
0
362
Demodulation
where xi t; is xi t with a phase shift of radians. From (7.236), it is seen that the output
of the quadrature
correlatorsquare
p
2 p law detector2given that si t was the transmitted signal is
2
Xi
Es cos NI;i Es sin NQ;i . The outputs of the remaining M 1
2
2
NQ;j
. The noise random variquadrature correlatorsquare law detectors are Xj2 NI;j
ables NI;i and NQ;i are all independent Gaussian random variables with zero mean and
variance 2 N0 =2. From Section 7.4, we conclude that Xi2 is a non-central chi-squared
random variable and that Xj2 is a central chi-squared random variable. On the other hand, the
output of the ith matched lterenvelope detector Xi is a Rice random variable and Xj is a
Rayleigh random variable as in Section 7.3. Both demodulators in Figures 7.42 and 7.43
yield identical symbol error probability as described in (7.226), where the parameters a2 and
2 are given by a2 Es and 2 N0 =2, that is,
Pe
M
1
X
1k1
k1
M 1
k
k
1 k1
e
k1
Es
N0
(7:244)
(7:245)
As a comparison, the union bound for bit error probability can be evaluated via (7.228) and
(7.238):
M
Pb e
4
Eb
2N0
log2 M
(7:246)
r
2
cos 2fc t
Ts
(7:247)
with the aid of E A2 Ts =2, where E is the smallest signal (symbol) energy and Ts is the
signal interval. Figure 7.44 shows the coherent demodulator
pfor
MASK.
The minimum Euclidean distance for MASK is dmin 2 E ; hence, according to (7.235),
the approximate bit error probability is
363
t = Ts
Minimum
Euclidean
distance
detector
Ts
s(t) + n(t)
Symbolto-bit
converter
{0, 1}
2
cos 2fc t
Ts
Figure 7.44
Nn
Q
Pb
log2 M
r
2E
N0
(7:248)
To calculate the average number of nearest neighbors Nn , we observe from Figure 6.24 that
the two outer symbols have one nearest neighbor. On the other hand, the inner symbols have
two nearest neighbors. Thus,
1
1 1 2 M 2
M
2 M 1
Nn
(7:249)
It is more relevant to express the bit error probability in terms of the average bit energy Eb
than the smallest symbol energy E. Since Eb Es =log2 M , where Es is the average symbol
energy, we are interested in calculating Es . From (6.82), we have
M
M
1X
1X
En
2n M 12 E
M n1
M n1
"
#
M
M
X
X
E
2
2
4
n 4 M 1
n M M 1
M
n1
n1
E 2
2
2
M M 12M 1 2M M 1 M M 1
M 3
1
M2 1 E
3
Es
(7:250)
and hence
Eb
M2 1
E
3 log2 M
(7:251)
Substituting (7.249) and (7.251) into (7.248), we obtain the approximate bit error
probability
2 M 1
Q
Pb
M log2 M
s
!
6 log2 M Eb
M 2 1 N0
(7:252)
364
Demodulation
(7:253)
where N is the AWGN sample with zero mean and variance 2 N0 =2. We then have
PejH1 PrfX 2 R2 jH1 [ X 2 R3 jH1 [ X 2 R4 jH1 g
PrX 2 R2 jH1 PrX 2 R3 jH1 PrX 2 R4 jH1
(7:254)
using the fact that all regions Ri , i 1; 2; 3; 4, do not overlap as indicated in Figure 7.45.
We can establish the conditional bit error probability PbjH1 from PejH1 by noting the
following facts: when X 2 R2 , one bit error occurs since symbols 00 and 01 differ by
only one bit; when X 2 R3 , two bit errors occur since symbols 00 and 11 differ by two bits;
when X 2 R4 , only one bit error occurs since symbols 00 and 10 differ by one bit. Therefore,
we obtain PbjH1 as follows:
1
1
PbjH1 PrX 2 R2 jH1 PrX 2 R3 jH1 PrX 2 R4 jH1
2
2
(7:255)
Now let us evaluate the three terms on the right-hand side of (7.255):
3
PrX 2 R2 jH1 Prdmin 5s1 N 0 Pr dmin 5 dmin N 0
2
1
3
dmin
3dmin
5
Pr dmin N dmin Q
Q
2
2
2
2
(7:256)
Similarly,
3
PrX 2 R3 jH1 Pr05s1 N dmin Pr 05 dmin N d min
2
3
5
3dmin
5dmin
Q
Pr dmin 5N dmin Q
2
2
2
2
(7:257)
Finally,
3
PrX 2 R4 jH1 Prs1 N 4d min Pr dmin N 4dmin
2
5
5dmin
Pr N 4 dmin Q
2
2
(7:258)
365
Substituting (7.256)(7.258) into (7.255) yields the conditional bit error probability assuming hypothesis H1 :
PbjH1
1
dmin
1
3dmin
1
5dmin
Q
Q
Q
2
2
2
2
2
2
(7:259)
(7:260)
Following the argument that was used to evaluate PbjH1 , we obtain the conditional bit error
probability PbjH2 as follows:
1
1
PbjH2 PrX 2 R1 jH2 PrX 2 R3 jH2 PrX 2 R4 jH2
2
2
(7:261)
The three terms on the right-hand side of (7.261) can be evaluated in the same fashion:
1
PrX 2 R1 jH2 Prs2 N dmin Pr dmin N dmin
2
1
dmin
Pr N 5 dmin Q
2
2
(7:262)
Similarly,
1
PrX 2 R3 jH2 Pr05s2 N dmin Pr 05 dmin N dmin
2
1
3
dmin
3dmin
Q
Pr dmin 5N dmin Q
2
2
2
2
(7:263)
Also,
1
PrX 2 R4 jH2 Prs2 N 4dmin Pr dmin N 4dmin
2
3
3dmin
Pr N 4 dmin Q
2
2
(7:264)
By substituting (7.262)(7.264) into (7.261), we obtain the conditional bit error probability
given hypothesis H2 :
PbjH2 Q
dmin
1
3dmin
Q
2
2
2
(7:265)
The average bit error probability Pb can be obtained via the total probability theorem, and
the fact that PejH1 PejH4 and PejH2 PejH3 :
366
Demodulation
s1
00
s2
01
3
dmin
2
Figure 7.45
dmin
1 dmin
2
s3
11
0
1
d
2 min
s4
10
dmin
3
d
2 min
p
4ASK signal space with dmin 2 E . There are four detection regions: 15R1 dmin ; dmin 5R2 0;
05R3 dmin ; and dmin 5R4 1.
Pb
4
X
PbjHi PrHi
i1
3
dmin
1
3dmin
1
5dmin
Q
Q
Q
2
2
2
4
2
4
(7:266)
p
p
Using the fact that dmin 2 E 2 Eb 3 log2 M =M 2 1 via (7.251) and 2 N0 =2,
we get the exact bit error probability of 4ASK as
3
Pb Q
4
s
s
s
!
!
!
4 Eb
1
4 Eb
1
4 Eb
Q 3
Q 5
5 N0
2
5 N0
4
5 N0
(7:267)
For a large bit energy-to-noise density ratio, that is, Eb =N0 1, the second and third terms
on the right-hand side of (7.267) are negligible compared to the rst term, and Pb can be
approximated by
3
Pb Q
4
s
!
4 Eb
5 N0
(7:268)
(7:269)
367
t = Ts
I-channel
Ts
2
cos 2fc t
Ts
t = Ts
Q-channel
Figure 7.46
Minimum
Euclidean
distance
detector
Symbolto-bit
converter
{0, 1}
Ts
2
sin2fc t
Ts
(7:270)
log2 M sin
Pb
;
M 42
(7:271)
Q
log2 M
M
N0
p
Note that for PSK, that is, M = 2, we have Nn 1, hence Pb Q 2Eb =N0 as derived in
(7.104).
368
Demodulation
t = Ts
Ts
Threshold
detector
{0, 1}
2
cos 2fc t
Ts
t = Ts
Ts
Figure 7.47
Threshold
detector
2
sin2fc t
Ts
01
00
Figure 7.48
s2
s1
s3
s4
11
Es
10
(7:273)
Here, X is the received vector, that is, X s1 N, where N is the AWGN vector with
statistically independent elements N1 and N2 and common variance 2 N0 =2 via (7.233).
The explanation for (7.273) is as follows: if X 2 Ri ; i 2; 3; 4, then X is closer to
si ; i 2; 3; 4, than s1 , and hence an error would occur. Since R2 , R3 , R4 do not overlap,
we can express PejH1 as
PejH1 PrX 2 R2 jH1 PrX 2 R3 jH1 PrX 2 R4 jH1
(7:274)
369
The conditional bit error probability can be established from the conditional symbol error
probability PejH1 by the following reasoning: as X 2 R2 , one bit error occurs since symbols
11 and 01 differ by only one bit; as X 2 R3 , two bit errors occur since the symbol 11 and 00
differ by two bits; as X 2 R4 , one bit error occurs since the symbol 11 and 10 differ by only
one bit. Thus, with X 2 R2 and X 2 R4 , a symbol of two bits was transmitted, but only one
bit error occurs. On the other hand, if X 2 R3 , a symbol of two bits was transmitted, and
both bits are in error. Therefore, the conditional bit error probability PbjH1 can be related to
PejH1 as
1
1
PbjH1 PrX 2 R2 jH1 PrX 2 R3 jH1 PrX 2 R4 jH1
2
2
(7:275)
s11 N1
s12 N2
2d
min
N1
6
7
4 2
5
dmin
N2
2
(7:276)
p
where s11 s12 dmin =2 Es =2 and Es ksi k2 s2i1 s2i2 ; i 1; 2; 3; 4, is the signal
(symbol) energy. Using (7.276), we can calculate the three terms on the right-hand side of
(7.275). Specically, from Figure 7.47 we have
PrX 2 R2 jH1 Prfs11 N1 0 \ s12 N2 40g
Prs11 N1 0 Prs12 N2 40
PrN1 dmin =2 PrN2 4 dmin =2
dmin
dmin
Q
1Q
2
2
(7:277)
dmin
dmin
PrX 2 R4 jH1 Q
1Q
2
2
(7:278)
(7:279)
Substituting (7.277)(7.279) into (7.275), we obtain the conditional bit error probability
370
Demodulation
dmin
dmin
dmin
1Q
Q2
2
2
2
dmin
Q
2
r
r
Es
2Eb
Q
Q
N0
N0
PbjH1 Q
(7:280)
where Eb Es =2 is the bit energy and 2 N0 =2. By symmetry and the fact that all
symbols are equiprobable, we conclude that the bit error probability of QPSK is indeed
the right-hand side of (7.272):
r
4
4
X
X
1
2Eb
PbjH1 PrHi
Pb
Q
N0
4
i1
i1
(7:281)
r
2Eb
Q
N0
si1
si2
pE
s cos i
Es sin i
(7:282)
where i 2i 1=M is the encoding phase of a symbol. Note that i tan1 si2 =si1 , and
ksi k2 Es for i 1; 2; . . . ; M. From (7.218), the detection of a symbol is to select the largest
correlation metric cx; si 2xt si where ksi k2 is ignored because it is the same for all signals si ,
and x is the sampled value of X si N. We have
p
p
Es cos i
2 Es x1 cos i x2 sin i
cx; si 2x1 x2 p
Es sin i
p
2 Es R cosi
where
R
q
x21 x22
x2
tan
x1
1
(7:283)
(7:284)
(7:285)
Note that in the absence of noise, we have tan1 si2 =si1 i . Furthermore, R is
independent of any transmitted signal si . Therefore, in the alternative implementation of
371
Ts
t = Ts
X1
s(t) +n(t)
2
cos 2 fct
Ts
tan1
Figure 7.49
Ts
0
x2
x1
Calculate
|i |
i = 1,2, , M
Select the
smallest
input
si
Symbolto-bit
converter
{0, 1}
t = Ts
X2
2
sin 2 fct
Ts
j
j
i
cosi 4 cos j , and consequently, c x; si 4c x; sj for all j 6 i. Thus, for
coherent MPSK, the two-path demodulator in Figure 7.46 can be alternatively implemented
in Figure 7.49, which yields the same approximate bit error probability in (7.271).
372
Demodulation
t = Ts
Ts
Threshold
detector
2
Ts
1 bit
delay
{0, 1}
cos 2 fct
Tb + Ts
Tb +Ts
Threshold
detector
Tb
2
Ts
sin 2 fct
(a)
Ts
t = Ts
1 bit
delay
2
Ts
Minimum
Euclidean
distance
detector
cos 2 fct
2
Ts
Tb+Ts
Symbolto-bit
converter
{0, 1}
t = Ts
Tb
sin 2 fct
(b)
Figure 7.50
(a) Coherent OQPSK demodulator employing threshold detectors; (b) coherent OQPSK demodulator employing the
minimum Euclidean distance detector.
recovery process. Second, differential encoding allows DMPSK to be demodulated noncoherently in applications where coherent demodulation is not feasible. In this section, we
concentrate on the coherent demodulation of DMPSK with a differential decoding process
to resolve phase ambiguity. The most important member of the DMPSK family is DQPSK,
which we study rst.
373
i + /2
cI;i
cQ;i
dI;i
dQ;i
0
1
2
3
4
7/4
5/4
7/4
7/4
3/4
1
0
1
1
0
0
0
0
0
1
1
0
1
0
1
0
1
0
0
1
t = Ts
Ts
Threshold
detector
cI,i
dI,i
{0, 1}
2
cos 2 fct
Ts
Differential
decoder
t = Ts
Ts
Figure 7.51
Threshold
detector
cQ,i
dQ,i
2
sin 2 fct
Ts
(7:288)
To illustrate the phase ambiguity resolution, we use the data in Example 6.28 and rotate the
signal space by =2 radians, that is, the phase i of the received DQPSK signal is advanced
by =2 radians, to become i =2. Table 7.7 illustrates the differential decoding process to
recover the correct transmitted data despite a phase error of =2. The same result holds for a
phase rotation of , 3=2 or 2 radians.
374
Demodulation
1 bit
delay
{0, 1}
Pulse
shaping
Symbol
mapping
cos 2 fct
DQPSK
Pulse
shaping
1 bit
delay
Figure 7.52
sin 2 fct
t = Ts
Ts
Threshold
detector
1 bit
delay
2
cos 2 fct
Ts
{0, 1}
t = Ts
Ts
Threshold
detector
1 bit
delay
2
sin 2 fct
Ts
Figure 7.53
The above two examples demonstrate the simplicity in implementing a coherent DQPSK
demodulator. For higher-order DMPSK, differential decoding must be done via (6.109),
which provides a recursive relation between the phases i1 and i of consecutive DMPSK
symbols Di1 and Di , that is,
i i1 i
(7:289)
where
i Di 1
2
;
M
Di 2 f1; 2; . . . ; M g
(7:290)
If i can be recovered properly, then Di would be detected, and hence the k-bit symbol
represented by Di, where k log2 M . Figures 7.54 and 7.55 show two equivalent coherent
DMPSK demodulators.
These coherent demodulators are simply coherent MPSK demodulators followed by
differential decoders.
375
Ts
t = Ts
Minimum
Euclidean
distance
detector
2
cos 2 fct
Ts
Ts
t = Ts
si
1 symbol
delay
Calculate
i Di
si1
Symbolto-bit
converter
Di
{0, 1}
2
sin 2 fct
Ts
Figure 7.54
Coherent DMPSK demodulator. The phases of si1 and si are i1 and i , respectively, as indicated by the signal
space (see (7.282)), and i i i1 .
2
cos 2fct
Ts
Ts
t = Ts
X1
Ts
t = Ts
x2
tan1 x
1
|k|
k = 1,2, , M
X2
Select the
smallest
input
Figure 7.55
+
1 symbol
delay
{0, 1}
sin 2fct
Ts
mod 2
Symbolto-bit
converter
Calculate
Di
i1
Pb
(7:291)
Q
log2 M sin
log2 M
M
N0
376
Demodulation
(7:292)
Then the phase difference i remains the same as in (7.290), that is,
i i i1 i i1
2
Di 1 ; Di 2 f1; 2; . . . ; M g
M
(7:293)
Thus, the unknown carrier phase is eliminated and the data symbol can be recovered by
deciding which one of the M equiangular decision regions i belongs. This suggests the
noncoherent demodulator illustrated in Figure 7.56. The detection regions are chosen in
Figure 7.57 for the case of M = 4. These equiangular detection regions are dened by
M phase angles f j j 12=M ; j 1; 2; . . . ; M . For example, if for a given
k 2 f1; 2; . . . ; M g, we have ji f k j min ji f j j; j 1; 2; . . . ; M; then we
j
choose Di k as the output symbol. In general, we have M detection regions
Rj ; j 1; 2; . . . ; M , where f j =M Rj 5f j =M .
To evaluate the symbol error probability of the noncoherent DMPSK in Figure 7.56, let us
assume that an unknown carrier phase is present as in (7.292). The received DMPSK
signal for the ith symbol is
st Apt cos2fc t i nt
(7:294)
2
cos 2fct
Ts
Ts
t = Ts
X1
Ts
t = Ts
X2
x2
tan1 x
1
mod 2
i
+
i
1 symbol
delay
i1
Calculate
|i j|
Di Symbolto-bit
j = 1,2, , M
converter
and choose
the smallest
2
sin 2fct
Ts
Figure 7.56
{0, 1}
377
R2
+
R3
R1
I
+
R4
Figure 7.57
Four equiangular detection regions Ri, i = 1, 2, 3, 4, for M = 4; /4 R1 < /4, /4 < R2 3/4, 3/4 < R3
< 5/4, 5/4 R4 < 7/4; j = ( j1) 2/M, j = 1, 2, . . . , M.
X1;i
N1;i
E
s cosi
p
Xi
(7:295)
X2;i
N2;i
Es sini
The noise samples N1;i and N2;i are independent Gaussian random variables with zero mean
and variance 2 N0 =2. The vector X i can be written in phase notation as
X i X1;i j X2;i kX i keji
(7:296)
Note that, without noise, the phase i of X i is simply i . Similarly, the decision vector
X i1 for the previous i 1th symbol is
N1;i
X1;i1
E
s cos i1
p
X i1
(7:297)
X2;i1
N2;i
Es sini1
or, in phasor notation,
X i1 X1;i1 j X2;i1 kX i1 ke ji1
(7:298)
In the absence of noise, i1 i1 . Dene the phase difference i i1 , which is
equal to i without noise. Since is our decision variable, we are interested in its
probability distribution function F , which has been derived in [1]:
F
sin
4
=2
(7:299)
From Figure 7.56, it is seen that a symbol is correctly detected if and only if f j falls into
the angular detection region Rj given the differential phase i f j . Since the probability
distribution function F is independent of f j , we can arbitrarily set f j 0 (which
corresponds to j 1 in f j j 12=M ; j 1; 2; . . . ; M ) and hence the correct symbol
detection probability is
378
Demodulation
=M
n
o
Pc Pr 5
f d F
F
M
M
M
M
=M
(7:300)
=M
Ps 1
f d
f d
f d
=M
=M
h
i
2 F F
M
(7:301)
1
sin
2
M
=2
(7:302)
For the case of M = 2, the above expression gives the bit error probability of noncoherent
DPSK in (7.179). A simple approximation can be found in [1] for Ps in (7.302), as
follows:
r
2Es
(7:303)
Ps 2Q
sin p
N0
2M
For Gray code bit-to-symbol mapping, the bit error probability for noncoherent DMPSK is
given by
r
2
2Eb
Pb
Q
log2 M sin p
(7:304)
N0
log2 M
2M
Compared with coherent
in (7.271), noncoherent DMPSK requires approximately
p MPSK
2
2
Eb=N0 to achieve the
sin =M sin = 2 M more
p error
inp
same
2 rate. For large M, this is
approximately sin2 =M sin2 = 2 M =M 2 = = 2 M 2 or 3 dB.
and
cos 0 ) H2
(7:305)
379
R2
R1
I
Figure 7.58
Two equiangular detection regions R1 (right-half plane) and R2 (left-half plane), with 1 = 0 and 2 = for M = 2.
Let us expand cos cosi i1 into the following expression:
cos cosi i1 cos i cos i1 sin i sin i1
(7:306)
Using (7.296) and (7.298), we can scale cos by a positive constant kX i k kX i1 k without
altering the decision rule in (7.306):
kX i k kX i1 k cos kX i1 k cos i1 kX i k cos i kX i1 k sin i1 kX i k sin i
X1;i1 X1;i X2;i1 X2;i
(7:307)
where X1;i ; X2;i are given in (7.296) and X1;i1 ; X2;i1 are given in (7.298). Equation (7.307)
is identical to (7.168); therefore, we conclude that the two demodulators in Figure 7.28 and
7.56 are equivalent.
380
Demodulation
R2
R1
R3
R4
Figure 7.59
Detection regions for /4-DQPSK: 0 R1 /2, /2 R2 < , R3 3/2, 3/2 R4 < 2; j = (2j 1) /4,
j = 1, 2, 3, 4.
and
and
and
and
sin 40 ) H1
sin 40 ) H2
sin 50 ) H3
sin 50 ) H4
(7:309)
We note that cos leads to the expression X1 X1;i1 X1;i X2;i1 X2;i as in (7.307).
Similarly, sin sini i1 sin i cos i1 cos i sin i1 leads to the expression
X2 X1;i1 X2;i X2;i1 X1;i . The detection rule in (7.309) can be expressed in terms of the
decision variables X1 and X2 as
X1 40
X1 50
X1 50
X1 40
and
and
and
and
X2 40 ) H1
X2 40 ) H2
X2 50 ) H3
X2 50 ) H4
(7:310)
381
2
cos 2fct
Ts
Ts
Ts
t = Ts
X1,i
1 symbol
delay
X1
+
Threshold
detector
{0, 1}
X1,i1
t = Ts
sin 2fct
Ts
Figure 7.60
+
X2,i
1 symbol
delay
X2,i1
X2
Threshold
detector
Ts
t = Ts
Minimum
Euclidean
distance
detector
2
cos 2fct
Ts
Ts
Symbolto-bit
converter
{0, 1}
t = Ts
Figure 7.61
2
cos 2fct
Ts
3;
M
3
M
1;
M 3 7
M
1;
M
3
E6
7
6
7
6
..
..
7
26
.
.
5
4
p
p
p
p
p
p
M 1; M 1
M 3; M 1 . . .
M 1; M 1
(7:311)
382
Demodulation
M
1X
1
En M 1E
M n1
3
(7:312)
M
4
M
p
2
M 2 inner points. The corner point has two nearest neighbors, the edge point has three
nearest neighbors, and the inner points have four nearest neighbors. Therefore, the average
number of nearest neighbors is
p
p
2
1
424 M 2 3
M 2 4
Nn
M
(7:313)
4
4 p
M
For example, M = 16 yields Nn = 3 and M = 64 yields Nn = 3.5. For very large M, Nn
approaches 4.
The minimum Euclidean distance for rectangular MQAM is
p
(7:314)
dmin 2E
Substituting Es in (7.312) for E into (7.314), we get
dmin
r r
6Es
6Eb log2 M
M 1
M 1
(7:315)
Assuming that Gray code is employed for symbol-bit mapping, we can obtain the bit error
probability for coherent MQAM using (7.235) and (7.313):
s
p
!
4 1 1= M
3 log2 M Eb
Q
Pb
M 1 N0
log2 M
(7:316)
A glance at (7.271) and (7.316) shows that the two expressions are identical for M = 4. For
M > 4, MQAM has a signal-to-noise ratio advantage equal to
3=M 1
2 sin2 =M
(7:317)
383
g (dB)
8
16
32
64
128
256
1.65
4.20
7.02
9.95
12.92
15.92
s
p
!
8 1 1= M
3 log2 M Eb
Q
Pb
M 1 N0
log2 M
(7:318)
This is twice the bit error probability of coherent MQAM, with a factor of 2 accounting for
the error propagation of the differential encoding.
i 1; 2; . . . ; M
(7:319)
where wi t ; i 1; 2; . . . ; M , are the Walsh functions in (2.52). On the other hand, the
basis functions for MFSK are
xi t
r
2
cos 2fi t;
Ts
i 1; 2; . . . ; M
fi fc 2i M 1
1
2Ts
(7:320)
(7:321)
Figures 7.62 and 7.63 show the coherent demodulators for CSK and MFSK, respectively.
As usual, the basis functions must be obtained via carrier recovery.
The bit error probability of coherent CSK or MFSK can be evaluated via the exact
expression in (7.240) or the union bound in (7.239). Specically, the minimum Euclidean
p
distance is given in (7.206) as dmin 2Es . Therefore, by substituting dmin into (7.239) and
(7.240), and noting that 2 N0 =2, we obtain the union bound
384
Demodulation
Ts
0
s(t) + n(t)
t = Ts
X1
2
w (t)cos 2 fct
Ts 1
Ts
t = Ts
Maximum
detector
si
Symbolto-bit
converter
{0, 1}
Maximum
detector
si
Symbolto-bit
converter
{0, 1}
XM
2
w (t)cos 2 fct
Ts m
Figure 7.62
Ts
0
s(t) + n(t)
t = Ts
X1
2
cos 2 f1t
Ts
Ts
t = Ts
XM
2
cos 2 fMt
Ts
Figure 7.63
M =2
1 1 y2 =2
2Eb
1 p
e
1Q y
log2 M
dy
Pb
M 1
N0
2 1
(7:322)
(7:323)
For Pb 5102, the union bound is very close to the exact expression.
385
r
2
xi t
wi t cos 2fc t;
Ts
r
2
xi t;
wi t sin 2fc t;
2
Ts
i 1; 2; . . . ; M
(7:324)
i 1; 2; . . . ; M
(7:325)
For MFSK the basis functions and the corresponding quadrature functions are given by
r
2
cos 2fi t; i 1; 2; . . . ; M
(7:326)
xi t
Ts
r
2
xi t;
sin 2fi t;
2
Ts
i 1; 2; . . . ; M
fi fc 2i M 1
(7:327)
1
2Ts
(7:328)
Figures 7.64 and 7.65 show the noncoherent CSK and MFSK demodulators, respectively.
The exact bit error probability and its union bound are given in (7.245) and (7.246),
respectively, and are cited here again for convenience:
t = Ts
Ts
2
w (t)cos 2 fct
Ts 1
t = Ts
Ts
( )2
( )2
X12
s(t,) + n(t)
2
w (t)sin 2 fct
Ts 1
t = Ts
Ts
2
w (t)cos 2 fct
Ts M
t = Ts
Ts
2
w (t)sin 2 fct
Ts M
Figure 7.64
Maximum
detector
2
XM
( )2
( )2
si
Symbolto-bit
converter
{0, 1}
386
Demodulation
t = Ts
Ts
0
2
cos2 f1t
Ts
( )2
t = Ts
Ts
( )2
X12
s(t,) + n(t)
2
sin2 f1t
Ts
Ts
Maximum
detector
t = Ts
2
cos2 fMt
Ts
Ts
2
XM
si
Symbolto-bit
converter
{0, 1}
)2
t = Ts
( )2
2
sin2 fMt
Ts
Figure 7.65
1
k
X
M =2 M
1 k1
NEb0 log2 M
k1 M 1
e
Pb
1
k
M 1 k1
k1
(7:329)
M
Pb e
4
Eb
2N0
log2 M
(7:330)
For all practical purposes, the union bound approximates the exact expression well. For
Pb 5102 coherent demodulation enjoys an advantage of less than 1 dB in signal-to-noise
ratio over noncoherent demodulation.
(7:331)
387
where f t; DN represents the phase of the carrier up to the end of the Nth symbol, that is,
NTs t 5N 1Ts :
N
X
f t; DN 2h
NTs t 5N 1Ts
2Di M 1 qt iTs ;
(7:332)
i1
The digital modulation index h is assumed to be xed for all symbols in our analysis and
Di 2 f1; 2; . . . ; M g is the ith symbol in the symbol stream. The pulse qt is dened in
(6.147) as the integral of the symbol pulse shape pt, which we assume to have duration L
symbols. Thus, symbols with the indexes i N L 1, N L 2; . . . ; N inuence the
phase ft; DN via its pulse shape, while symbols with indexes i 1; . . . ; N L
produce the phase accumulation in the past. Thus, we can represent f t; DN as follows:
f t; DN h
N
L
X
2Di M 1 2h
i1
N t; DN ;
N
X
2Di M 1 qt iTs
iN L1
(7:333)
NTs t N 1Ts
The above observations allow us to conclude that the phase function of CPM is a nitestate machine and can be represented by a state diagram or a state trellis. The state trellis is
preferable because it allows us to view the state transitions in time. From (7.333) we can
represent a state at time t NTs as
N ; L 1
(7:334)
SN
fN ; DN 1 ; DN 2 ; . . . ; DN L1 g; L41
and a state at time t N 1Ts can be updated from SN as follows:
N 1 ; L 1
S N 1
fN 1 ; DN ; DN 1 ; . . . ; DN L2 g; L41
where
(7:335)
388
Demodulation
1
4
,1
(0,1)
(0, 2)
1
2
2
,1
Figure 7.66
2
,2
4
, 2
N 1 N h2DN L1 M 1
(7:336)
4 3
3
3
3
(7:337)
and DN 2. The state diagram in Figure 7.66 shows the transition from state N ; DN 1 to
state N1 ; DN . The trellis in Figure 7.67 shows the transition of state N ; DN 1 to state
N 6 ; DN 5 over six symbol periods. For example, the path (0,1), (4/3,1), (2/3,2), (4/3,
2) through the trellis represents the detected symbol stream 1,1,2,2. On the other hand, the
path (2/3,1), (0,2), (2/3,1), (0,1) represents the detected symbol stream 1,2,1,1.
The above example shows that the trellis can be used to detect a sequence of CPM
symbols. The algorithm that performs sequence detection via a trellis is known as the Viterbi
algorithm. To implement the Viterbi algorithm, one needs to evaluate the path metric at each
389
(N, DN1)
(N+1, DN)
(N+2, DN+1)
(N+6, DN+5)
(0, 1)
(0, 2)
2
3 ,1
2
,2
3
4
,1
3
4
,2
3
Figure 7.67
Trellis for CP-FSK with h = 2/3 and L = 2 for six symbol intervals. The starting state can be any one of the six states.
The bold lines are the trellis for starting state (0,1), which reaches steady-state after three symbol periods.
node. For coherent demodulation, the most logical choice of metric is the correlation metric
similar to (7.218) for symbol-by-symbol detection. Let us dene the correlation metric for a
received CPM signal plus noise xt given the semi-innite symbol sequence DN :
CN
N 1Ts
1
N 1Ts
1
xt cos2fc t f t; DN dt
xt cos 2fc t cos f t; DN dt
N1Ts
1
N 1Ts
NTs
N 1Ts
NTs
xt cos2fc t f t; DN dt
xt cos 2fc t cos f t; DN dt
N 1Ts
NTs
CN 1 N
(7:339)
We term CN the path metric and N the branch metric. At each node in the trellis, a surviving
path with the largest metric is retained. There are bM L1 or 2bM L1 surviving paths, one for
each node. The path metrics are then updated by bM L or 2bM L branch metrics and new
surviving paths are again retained for the next node. Thus, each surviving path has its metric
updated by M new branch metrics. The coherent CPM demodulator is illustrated in
Figure 7.68 based on (7.333) and (7.339).
390
Demodulation
x(t)
cos 2fct
Phase
generator
(t, DN)
Metric
computer
Viterbi
algorithm
Di
Symbolto-bit
converter
{0, 1}
sin 2fct
Figure 7.68
Obviously, the initial signal phase must be known for coherent demodulation. In the
following discussion we study the performance of the Viterbi algorithm which implements
sequence detection. Therefore, we rst establish the sequence error probability. If the
observed sequence has a length of N symbols assuming the initial time is t = 0, then there
are M N different sequences. The Viterbi algorithm chooses the path of N symbols that is
closest in Euclidean distance to the received sequence. Hence, the sequence error probability
is upper bounded by the union bound
X dij
Q
Pse
(7:340)
2
i6j
where 2 N0 =2 is the noise variance, and dij is the Euclidean distance between two signals
si t and
sj t that are unmerged from nTs to n N Ts , whose phase functions f t; DN ;i
and f t; DN;j contain two N-symbol sequences that must be different in their rst symbol.
Setting the index n to zero, we have
NTs
2
si t sj t dt
dij2
0
(7:341)
NTs
NTs
NTs
2
2
si t dt
sj t dt 2
si t sj t dt
0
The rst two terms on the right-hand side of (7.341) are the energy of the sequence of N
symbols. Therefore, each term is equal to NEs , where Es is the symbol energy. Thus,
NTs
2
A2 cos 2fc t f t; DN ;i cos 2fc t f t; DN;j dt
dij 2NEs 2
0
(7:342)
2Es NTs
2NEs
cos f t; DN ;i f t; DN ;j dt
Ts 0
We have assumed that fc is an integer multiple of 1=Ts so that the term containing the second
harmonic is zero. Let us denote
(7:343)
f t; DN;i ; DN;j f t; DN;i f t; DN ;j
391
(7:344)
k1
By using (7.343) and (7.344) in (7.342), we obtain the squared Euclidean distance as
follows:
2Es NTs
1 cos f t; DN;i ; DN;j dt
dij2
Ts 0
"
#)
(7:345)
(
N
X
2Es NTs
1 cos 4h
Dk;i Dk;j qt kTs
dt
Ts 0
k1
The sequence error probability in (7.340) is dominated by the term corresponding to dmin , the
minimum of all possible dij s. Let Ad be the average number of paths of distance d from a given
path (this concept is similar to the average number of nearest neighbors of a signal point), we
can approximate the sequence error probability with the rst few terms as follows:
Pse
dmin
m
X
Ad Q
ddmin
d
2
(7:346)
(7:347)
2
of CP-MFSK of any digital modulation index h a=b, where a and b are
The value of dmin
relative prime positive integers, is given for 1REC pulse shape (full response L = 1) as
follows [7]:
8
2Es ; integer h
>
>
< minf4Es 1 sinch; b M ; 2; 4; 6; . . . ; 2M 1
2
dmin
(7:348)
>
>
: 4Es 1 sinch; h50:3017; b5M
2Es ; h40:3017; b5M
p
This implies that dmin 2 Eb , which is the same minimum Euclidean distance as PSK. This
explains why coherent precoded MSK has the same bit error probability as coherent PSK.
392
Demodulation
When h is not rational, the minimum Euclidean distance varies with both h and N. For CPFSK with 1REC pulse shape, it was shown that when N = 3, h reaches its maximum at 0.715
2
and dmin
4:87Eb [8]. Compared with coherent precoded MSK, a gain of 0.85 dB in signalto-noise ratio per bit is achieved. The value of h can be approximated by h = 5/7 and the
Viterbi algorithm can be implemented using the phase trellis. Another observation from
(7.348) shows that dmin increases with increasing M, since a larger M results in a larger
symbol energy Es. This also holds true for h that is not rational.
5
2
; 2; 4; 6
dmin min 4Es 1 sinc
7
sin 5=7
min 4Es 1
(7:350)
5=7
minf4:87Es ; 3:81Es ; 3:77Es g
3:77Es 7:54Eb
Comparing (7.350) with (7.349) shows that CP-4FSK with h = 5/7 achieves a 2.75 dB gain
over precoded MSK.
For h = 17/19, we obtain
17
2
; 2; 4; 6
dmin
min 4Es 1 sinc
19
sin 17=19
min 4Es 1
(7:351)
17=19
minf4:44Es ; 4:35Es ; 4:22Es g
4:22Es 8:44Eb
Thus, CP-4FSK with h = 17/19 achieves a 3.24 dB gain over precoded MSK.
This example shows that by increasing M one can achieve a better power efciency,
but only at the expense of bandwidth, since a higher digital modulation index is
required.
393
0
2
1
2
3
2
Figure 7.69
i=0
0
1
i=2
i=3
i=4
i=5
i=6
2
1
1
2
1
2
2
3
2
Figure 7.70
Trellis for precoded MSK over six bit intervals. The starting state is state 0.
Note that it takes a minimum of two bit intervals for a path to leave a state and re-enter the
2
for precoded MSK, the observed sequence
same state. Therefore, in the calculation of dmin
has length N = 2 symbols. Let us assume that the algorithm starts at state 0 at time t = 0 and
that the initial phase is zero. Consider the received bit sequence {1, 1}, which corresponds to
the sequence of symbols fD0 2; D1 2g. We wish to use the Viterbi algorithm to recover
this sequence of symbols. Assuming no noise, the received signal for each symbol interval is
given in (7.331). The state SN 2 f0; =2; ; 3=2g at each time index N will be denoted as
0N ; =2N ; N ; 3=2N .
1. There are two paths leaving the initial sate 00 to enter state =21 (when D0 = 2) and
state 3=21 (when D0 = 1). The phases f t; D0 0 t; D0 are
f t; 1 0 2 2 1qt qt and f t; 2 0 4 2 1qt
qt . The metrics C0 at each state =21 and 3=21 are given by
394
Demodulation
C0 =21
Ts
(7:352)
ATs
2
and
C0 3=21
Ts
A Ts
A Ts
cosf t; 2 f t; 1 dt
cos2qt dt
2 0
2 0
A Ts
2t
cos
dt 0
2 0
2Ts
0
(7:353)
2. There are two paths leaving =21 to enter state 02 (when D1 1) and state 2 (when D1 2).
There are also two paths leaving 3=21 to enter state 02 (when D1 2) and state 2 (when
D1 1). We calculate the two metrics for each state and retain only the survivor path with the
largest metric. The phase f t; D0 ; D1 1 t; D0 ; D1 0 2 2D0 2 1
2D1 2 1qt Ts is given by
t Ts
t
f t; 2; 1 4 2 1 2 2 1
2Ts
2
2Ts
(7:354)
t Ts
t
f t; 2; 2 4 2 1 4 2 1
2
2Ts
2Ts
(7:355)
t Ts
t
f t; 1; 2 2 2 1 4 2 1
2Ts
2
2Ts
(7:356)
t Ts
t
f t; 1; 1 2 2 1 2 2 1
2
2Ts
2Ts
(7:357)
There are four new metrics to be computed and they are listed below with the rst two for
state 02 and the last two for state 2 :
C1 02 C0 =21
ATs A
2
2
2Ts
2Ts
Ts
^ 1 02 C0 3=21
C
A
2
2Ts
Ts
Ts
t
ATs
:
cos dt
Ts
2
2Ts
(7:358)
survivor
Ts
dt
ATs
5 C 1 02 :
2
(7:359)
eliminated
395
C1 2 C0 =21
ATs A
2
2
2Ts
2Ts
2Ts
Ts
(7:360)
dt ATs :
survivor
Ts
^ 1 2 C0 3=21
C
A
Ts
2Ts
Ts
t
cos dt 05C1 2 :
Ts
(7:361)
eliminated
We
now have two surviving paths: f00 ; =21 ; 2 g with metric C1 2 ATs and
00 ; =21 ; 02 with metric C1 02 ATs =2. Since C1 2 4C1 02 , we choose path
f00 ; =21 ; 2 g as the detected path. The data symbols associated with this path is
fD0 2; D1 2g and the data bits can be calculated via 2Di M 1 with M = 2 as
{1, 1}, which is the correct data sequence.
The above example shows that the Viterbi algorithm can be used for sequence detection.
Of course, in the case of precoded MSK, the demodulator in Figure 7.20 works well and is
optimum for symbol-to-symbol detection. For partial response CPM (L > 1) with M > 2, the
Viterbi algorithm is advantageous because the algorithm can be implemented easily as part
of the digital baseband processor of the receiver. Furthermore, when the channel has
memory, as in the case of intersymbol interference, symbol-by-symbol detection is no
longer optimal and the Viterbi algorithm offers the best detection. In (7.346), we give the
expression for sequence error probability. As an end user, we are more interested in the
symbol error probability, which is given by
Ps
dmin
m
X
ddmin
d
Bd Q
2
(7:362)
where 3 m 5 would sufce and Bd is the total number of symbol errors on all paths of
distance d from a given path. The value of Bd can be obtained for each CPM with rational h
by searching the trellis.
In summary, the Viterbi algorithm is a viable trellis detection scheme for coherent CPMFSK with rational h. The Viterbi algorithm is indeed a maximum likelihood sequence
detection scheme N ! 1 that offers the best detection (as opposed to a symbol-bysymbol detection that has been the theme of the previous sections).
396
Demodulation
algorithm is one scheme that applies well to CP-MFSK with rational h. In this section, we
study a method that does not rely on the trellis; therefore, it applies to any CPM in general.
The method calculates the metrics of an observable sequence of N symbols, and then uses
these metrics to make a decision on the rst symbol only. Thus, the detection made on the kth
symbol is based on the metrics of the observable sequence starting with the kth symbol and
ending with the (k + N 1)th symbol. Since the decision is made on one symbol
at a time,
we could use the probability density function of the received signal xt s t Hi ; Dn;N 1 +
noise, given hypothesis Hi of M possible hypotheses (for M symbols) and N 1 observable
symbols Dn;N 1 that follow the kth symbol. There are n M N 1 equally likely combinations of observable sequences Dn;N1 of N 1 symbols. Noise is Gaussian with variance
2 N0 =2. Using signals instead of vectors in (7.181), we get
NTs
2
1
1=2 2
xtstjHi ; Dn;N 1 dt
0
fX xjHi ; Dn;N 1 p e
(7:363)
2
Using
(7.217) and noting that x2 t dt is common to all Dn;N 1 and all signals
s t Hi ; Dn;N 1 have the same energy NEs, which is N times the symbol energy Es, we
can write the right-hand side of (7.363) as follows:
NTs
1= 2
xt stjHi ; Dn;N 1 dt
fX xjHi ; Dn;N 1 p e 0
(7:364)
2
where the constant is independent of Dn;N 1 and is given by
h
i
NTs
1=2 2
x2 t dtNTs Es
0
e
(7:365)
To remove the condition on Dn;N 1 , we need to average (7.364) over the probability density
function of Dn;N 1 , which is discrete with a value of 1=M N 1 for each possible symbol Di in
the set f1; 2; . . . ; M g:
1
NY
fD Dn;N1
fDi Di
i1
N
1
Y
M N 1
i1
(7:366)
Di 1 Di 2 Di M
N 1
M
X
1
M N1
n1
p e1=
2
NTs
0
xt stjHi ; Dn;N 1 dt
(7:367)
N1
M
X
1= 2
NTs
0
xt stjHi ; Dn;N1 dt
i 1; 2; . . . ; M
(7:368)
n1
The detector chooses a hypothesis that corresponds to the largest metric and outputs the rst
symbol of N observable symbols (rst symbol plus Dn;N 1 following symbols) that is
397
NTs
e(
s(t 1,D1,N1)
NTs
0
x(t)
e(
)
c(x H1)
Maximum
detector
Di {1,2,L, M}
c(x HM)
NTs
e(
s(t M,D1,N1)
NTs
e(e
s(t M, DM N1,N1)
Figure 7.71
N 1
M
X
n1
1= 2
NTs
0
i 1; 2; . . . ; M
(7:369)
398
Demodulation
where xt is again the received CPM signal st plus noise over 0 t NTs and
s tjHi ; Dn;N1 ; A cos 2fc t f t; Dn;N 1 Hi
0 t NTs
(7:370)
(7:372)
s t; Hi ; Dn;N 1 A sin 2fc t f t; Dn;N 1 Hi
(7:373)
2
Using the above result, we obtain the correlation between xt and s tjHi ; Dn;N 1 ; as
follows:
NTs
xt s tjHi ; Dn;N 1 ; dt cos
NTs
0
xt s tjHi ; Dn;N 1 dt
NTs
Hi ; Dn;N 1 dt
2
0
In;i cos Qn;i sin Zn;i cos n;i
sin
xt s t;
(7:374)
where
In;i
NTs
xt s tjHi ; Dn;N 1 dt
(7:375)
xt s t; Hi ; Dn;N 1 dt
2
(7:376)
Qn;i
NTs
0
Zn;i
q
2 Q2 ;
In;i
n;i
n;i tan
1
Qn;i
In;i
(7:377)
N1
M
X
2
e1= Zn;i cosn;i ;
i 1; 2; . . . ; M
(7:378)
n1
The unconditional metrics can be obtained by averaging c xjHi ; using the probability
density function f 1=2; 0 2:
399
c xjHi
c xjHi ; f d
0
N 1
M
X
1 1= 2 Zn;i cosn;i
d
e
2
n1 0
N 1
M
X
Zn;i
I0
;
i 1; 2; . . . ; M
2
n1
(7:379)
N1Ts
NTs
xt cos2fc t f t; DN dt
NTs
N 1Ts
sin
xt sin2fc t f t; DN dt
NTs
(7:381)
NTs
QN
N 1Ts
xt sin2fc t f t; DN dt
(7:382)
NTs
q
ZN IN2 Q2N ;
N tan
1
QN
IN
(7:383)
It is obvious that the information about the Nth symbols is contained in the envelope ZN of
the correlation function. Therefore, the generic demodulator in Figure 7.68 also applies in
this case but with a noncoherent path metric cN dened for a path of N symbol as follows:
cN
N
X
Zi
(7:384)
i0
where Zi is the noncoherent branch metric for the ith symbol dened in (7.383). The Viterbi
algorithm can be carried out as described in Example 7.29.
400
Demodulation
NTs
( )2
s(t 1, D1,N1)
NTs
I0( )
( )2
s t , 1, D1,N1
2
NTs
c (x H1)
( )2
NTs
I0( )
( )2
x(t)
Di {1, 2, , M}
s t , 1, DM N1,N1
2
NTs
Maximum
detector
( )2
NTs
I0( )
( )2
s t , M, D1,N1
2
c (x H1)
NTs
( )2
s(t M, DM N1,N1)
NTs
I0( )
( )2
s t , M, DM N1,N1
2
Figure 7.72
401
Table 7.9 Performance of M-ary modulation techniques (the symbol duration is Ts Tb log2 M)
Modulation
Coherent MASK
q
2Eb
N0 log2 M sin M
2
log2 M
Coherent OQPSK
2Eb
N0
4
log2 M
q
2Eb
N0 log2 M sin M
2
log2 M
q
2Eb
p
N0 log2 M sin 2M
Noncoherent /4-DQPSK
q
Coherent DMQAM
q
1:11 NEb0
r
p
411= M
3 log2 M Eb
Q
log2 M
N0
M 1
r
p
811= M
3 log2 M Eb
Q
log M
N0
M 1
q
Eb
5 M2 Q
N log2 M
Coherent CPM
Ps
Coherent MQAM
dmin
P5
Bd Q
ddmin
Noncoherent CPM
About 1 dB less
d
2
: Viterbi
Bandwidth
2
Ts
2
Ts
2
Ts
2
Ts
2
Ts
2
Ts
2
Ts
2
Ts
2M
Ts
for CSK;
M
Ts
for MFSK
2M
Ts
for CSK;
M
Ts
for MFSK
[8]
[8]
402
Demodulation
cos 2 fct
Filter p(t)
and
A/D
converter
Remove
guard
interval
FFT
Subcarrier
symbol
detection
sin 2 fct
Figure 7.73
Preamble:
12 symbols
Figure 7.74
Signal:
One symbol
Data:
Variable number of symbols
403
Table 7.10 Data rate and subcarrier modulation of IEEE 802.11a [9]
Data rate
Coding
Coded bits per
(Mbits/s) Modulation rate (R) subcarrier (NBPSC)
6
9
12
18
24
36
48
54
BPSK
BPSK
QPSK
QPSK
16-QAM
16-QAM
64-QAM
64-QAM
1/2
3/4
1/2
3/4
1/2
3/4
2/3
3/4
48
48
96
96
192
192
288
288
24
36
48
72
96
144
192
216
1
1
2
2
4
4
6
6
Value
48
4
52 (NSD + NSP)
0.3125 MHz (= 20 MHz/64)
3.2 s (1/F)
16 s (TSHORT + TLONG)
4.0 s (TGI + TFFT)
0.8 s (TFFT/4)
1.6 s (TFFT/2)
4 s (TGI + TFFT)
8 s (10 TFFT/4)
8 s (TGI2 + 2 TFFT)
802.11a are shown in Table 7.11. The IEEE 802.11a standard species a 64-point IFFT (and
hence 64-point FFT) as illustrated in Figure 7.75. The coefcients 1 to 26 are mapped into
the same numbered IFFT inputs, while the coefcients 26 to 1 are copied into IFFT
inputs 38 to 63. The rest of the inputs, 27 to 37, and the 0 (dc) input, are set to zero. After
performing an IFFT, the output is cyclically extended to the desired length. The reverse
process is carried out for an FFT.
Timing synchronization
In a packet-switched WLAN, timing synchronization must be acquired during the preamble
in order to retrieve the data packet. The length and the content of the IEEE 802.11a preamble
have been carefully designed to facilitate timing synchronization while keeping the frame
overhead to a minimum. The preamble in Figure 7.76 consists of ten short OFDM symbols
404
Demodulation
Null
#1
0
1
#2
#26
26
27
Null
Null
IFFT
26
27
Null
37
#26
38
Figure 7.75
0
1
37
38
#2
62
#1
63
62
63
10 0.8 = 8.0 s
t1
t2
t3
t4
t5
t6
Figure 7.76
t7
t8
t9 t10
GI2
T1
T2
Coarse
Channel and
frequency fine frequency
offset
offset estimation
estimation
timing
synchronize
Signal
Rate
length
Data 1
Service + data
Data 2
Data
and two long OFDM symbols. The parameters t1 to t10 denote short training OFDM
symbols, and T1 and T2 denote long training OFDM symbols. The total training length is
16 s including two guard intervals (GI2) between the short symbol t10 and the long symbol
T1. The dashed boundaries in Figure 7.76 denote repetitions due to the periodicity of the
IFFT. A short OFDM symbol consists of 12 subcarriers, which are modulated by the
elements of the sequence S, given by
r
13
S26;26
f0; 0; 1 j; 0; 0; 0; 1 j; 0; 0; 0; 1 j; 0; 0; 0;
6
1 j; 0; 0; 0; 1 j; 0; 0; 0; 1 j; 0; 0; 0; 0; 0; 0; 0; 1 j; 0; 0; 0;
1 j; 0; 0; 0; 1 j; 0; 0; 0; 1 j; 0; 0; 0; 1 j; 0; 0; 0; 1 j; 0; 0g
(7:385)
405
Symbol 1
CP2
Symbol 2
Early FFT
window timing
CP3
ISI
Late FFT
window timing
Figure 7.77
Symbol 3
406
Demodulation
mn
j cn j 2
Pn2
(7:387)
where the function cn is the correlation and Pn is the received power of the second symbol
and both are given by
cn
D1
X
rnk rnkD
(7:388)
k0
Pn
D1
X
jrnkD j2
(7:389)
k0
The symbol timing estimate in an AWGN channel can be achieved using only the short
training sequence that consists of a few identical OFDM symbols, with 16 samples in each
symbol. Here, the delay D takes the values
D 16;
1; 2; . . . ; 5
(7:390)
In Figure 7.78 an 80-sample delay D was used. The simulation was run by setting the
frequency offset F between the transmitted and received signals to 100 kHz. The correct
symbol timing occurs at the peak with n = 401 and the frequency offset is estimated to be
^
F 99:88 kHz at n = 401.
0.9
0.8
0.7
0.6
m1
0.5
0.4
0.3
0.2
0.1
0
Figure 7.78
100
200
300
Sample index, n
400
500
600
407
rk rkD
(7:391)
k0
between two consecutive symbols, where rk is the received complex baseband signal. If
there is a carrier frequency offset F between the transmitted and received signals, we can
write the correlation c as follows:
c
D1
X
sk ej2 F k s skD ej2 F kD s
(7:392)
k0
where rk sk ej2 F ks , and 1= s denotes the sampling rate, that is, 1= s N =Ts , where
1=Ts is the subcarrier symbol rate, which is also the subcarrier separation, and N is the
number of subcarriers. Simplifying c we obtain
c
D1
X
k0
D1
X
j2 F D s
(7:393)
jsk j
k0
c
2D s
(7:394)
where c is the angle of c, that is, c tan1 Imc=Rec. Note that c is dened
unambiguously only in ; , hence the absolute value of F must obey
jF j5
2D s 2D s
(7:395)
Using IEEE 802.11a as an example for the short training OFDM symbol, the delay is D = 16,
and for the long training OFDM symbol, the delay is D = 64. Furthermore, we use
1= s N =Ts 20 MHz as seen in Table 7.10. Thus,
jFshort j5
20 106
625 kHz
2 16
(7:396)
408
Demodulation
and
6
Flong 5 20 10 156:25 kHz
2 64
(7:397)
Thus the carrier frequency offset estimate would be more reliable with the short training
sequence. For the IEEE 802.11a, the carrier frequency is about 5.3 GHz. The maximum
clock error is 20 106 . Therefore, the maximum error between the transmitter and receiver
clocks is 40 106 , and this results in a maximum frequency offset
Fmax 5:3 109 40 106 212 kHz. This frequency offset can only be estimated
accurately with the short training sequence. If (7.388) is employed instead of (7.391) for
frequency offset estimation, then (7.394) becomes
^
c^n
2D s
(7:398)
where c^n is the angle of c^n , the correlation value corresponding to the symbol timing
estimate ^
n.
The carrier frequency offset can be estimated quite accurately. Figure 7.79 displays the
probability distribution function (PDF) of the frequency offset estimate in an AWGN
0.25
0.2
0.15
0.1
0.05
0
85
Figure 7.79
90
95
100
Frequency [kHz]
105
110
115
409
channel. The simulation was run with F = 100 kHz, SNR = 10 dB, and D = 80. The PDF is
bell-shaped and centered at 100 kHz. The maximum frequency offset error can be taken to
be 4 kHz.
2
1.5
1
Imaginary
0.5
0
0.5
1
1.5
2
2
Figure 7.80
1.5
0.5
0
Real
0.5
Rotation of IEEE 802.11a QPSK signal points due to 3 kHz frequency error [11].
1.5
410
Demodulation
(7:399)
Np
X
^ k P;k
R;k H
k1
j2f
Np
X
(7:400)
^ P;k 2
Hk H
k
k1
411
dM
X
(7:402)
Ad Pd
ddfree
From the event error probability we can obtain the union bound on the coded bit error
probability by weighting the pair-wise error probability P(d) with the information weight Bd
(number of information bit errors) of all paths of Hamming weight d. Since there are k
information bits per branch for a rate k/n code, the union bound is given by
Pb 5
dM
1 X
Bd Pd
k dd
(7:403)
free
Tables 7.127.14 display the weight structures of the best codes for rates 1/3, 1/2, and 2/3,
respectively. The larger the constraint length the more powerful the code is.
Hard decoding
To evaluate the union bound for the bit error probability we need to calculate the pair-wise
error probability P(d). If the channel is a BSC then hard decoding is implemented. The
output of the detector is the channel output 1s or 0s. These demodulated coded bits serve as
Table 7.12 Generator (octal) and information weights of the best rate 1/3 codes [12]
K
Generators
dfree
Bdfree
Bdfree 1
Bdfree 2
Bdfree 3
Bdfree 4
3
4
5
6
7
8
7,7,5
17,15,13
37,33,25
75,53,47
171,165,133
367,331,225
8
10
12
13
15
16
3
6
12
1
7
1
0
0
0
8
8
0
15
6
12
26
22
24
0
0
0
20
44
0
58
58
56
19
22
113
Table 7.13 Generator (octal) and information weights of the best rate 1/2 codes [12]
K
Generators
dfree
Bdfree
Bdfree 1
Bdfree 2
Bdfree 3
Bdfree 4
3
4
5
6
7
8
9
7,5
17,15
35,23
75,53
171,133
371,247
753,561
5
6
7
8
10
10
12
1
2
4
2
36
2
33
4
7
12
36
0
22
0
12
18
20
32
211
60
281
32
49
72
62
0
148
0
80
130
225
332
1404
340
2179
412
Demodulation
Table 7.14 Generator (octal) and information weights of the best rate 2/3 codes [12]
K
Generators
dfree
Bdfree
Bdfree 1
Bdfree 2
Bdfree 3
Bdfree 4
3
4
5
6
7
8
9
7,5,7
15,13,15
31,33,31
73,41,73
163,135,163
337,251,337
661,473,661
3
4
5
6
6
8
8
1
8
25
75
1
395
97
10
34
112
0
81
0
0
54
180
357
1571
402
6695
2863
226
738
1858
0
1487
0
0
853
2 989
8 406
31 474
6 793
235 288
56 633
the decoder input bits. Let p be the channel transition probability; the probability that the
path of Hamming weight d is chosen over the all-zero path is given by
8
d
P
d i
>
>
p 1 pdi ;
d odd
>
<
i
id1=2
Pd
(7:404)
d
P
>
1 d
d i
d=2
di
>
d=2
>
p
p
1
p
1
p
;
d
even
:
i
2 d=2
id=21
413
where s0 represents the code vector c0 with Hamming weight d, and 2 N0 =2 is the noise
variance. The quantity ks0 sk is simply the Euclidean distance between s and s0 .
102
103
Q(2(Eb/N0))
104
Pb
K = 7 soft decode
K = 7 hard decode
K = 8 soft decode
K = 8 hard decode
K = 9 soft decode
K = 9 hard decode
105
106
107
Eb /N0(dB)
Figure 7.81
Bit error probability of PSK with convolutional codes of rate , K = 7, 8, 9. (Courtesy of Peter Harley.)
414
Demodulation
fR rjc Prc
fR r
(7:407)
Since convolutional codes are linear, all code vectors of the same length are equally likely;
P
therefore, Pr(c) is identical for all possible values of c. Furthermore, fR r fR rjc Prc
is independent of whichever code vector was transmitted. The decoder that maximizes the
a-posteriori probability Prcjr, that is, maximizes the probability of selecting the transmitted code vector correctly given the received vector r, is the ML decoder. Consequently,
the ML decoder also maximizes the likelihood fR rjc of the transmitted code vector c. Let
us establish the likelihood ratio
2m
Q
fR rjc i1
2m
r
fR rjc0 Q
i1
As usual we assume that the all-zero-code vector c was transmitted. The code vector c0 is
represented by the path through the trellis that differs with the all-zero path in d bits. Thus,
fR rjc and fR rjc0 differ in only d products and therefore by rearranging these different
products to be the rst d products we have
415
d
Q
Lr
i1
d
Q
i1
d
Q
i1
d
Q
i1
Lr i1
p
d
Q
a r
I0 22;i
(7:410)
i1
2
i1
i1
(7:411)
d
X
r1;i 5Y2
i1
d
X
r2;i
(7:412)
i1
The two random variables Y1 and Y2 are 2 -distributed. The pair-wise error probability
Pd PrY1 5Y2 is given by the following expression [8,12]:
d 1
d 1
X
X
1
2i i
k d 1!
Pd e=2
2
k
i!2kd1
i!d
i
1!
i0
ki
(7:413)
where
drEb
N0
(7:414)
416
Demodulation
102
103
104
Pb
1 Eb/2N0
e
2
K = 7 soft decode
K = 7 hard decode
K = 8 soft decode
K = 8 hard decode
K = 9 soft decode
K = 9 hard decode
105
106
107
10
11
12
13
14
Eb /N0(dB)
Figure 7.82
Bit error probability of noncoherent FSK with convolutional codes of rate , K = 7, 8, 9. (Courtesy of Peter Harley.)
Matched
filter
Quantizer
Soft
decoder
cos 2fct
Matched
filter
{0,1}
Quantizer
sin 2fct
Figure 7.83
417
for the I-Q coordinates. The pair-wise error probability between two vector s and s0 is
Prs ! s0 and is given by
0
k s sk
Prs ! s0 Q
(7:415)
2
where 2 N0 =2 is the noise variance. The quantity ks0 sk is simply the Euclidean
distance between s and s0 . Let the Euclidean free distance of the code be the minimum
Euclidean distance between different code words, that is,
d free min dsi ; sj
all i6j
(7:416)
Then we have
d free
Prs ! s Q
2
0
(7:417)
On the other hand, the pair-wise error probability Prsi ! sj between two symbols of the
uncoded signal space is given by
dmin
Prsi ! sj Q
(7:418)
2
where dmin is the minimum Euclidean distance of the uncoded signal space.
The TCM asymptotic coding gain relative to the uncoded signal space with the same
spectral efciency and average power is conventionally dened as follows:
G
d 2free
2
dmin
(7:419)
418
Demodulation
G (dB)-8PSK
3
3.6
4.1
4.6
5
5.2
5.7
G (dB)-16PSK
3.54
4
4.4
5.1
5.3
5.3
5.5
G (dB)-16QAM
3
4
4.8
4.8
5.4
6
6
G (dB)-64QAM
2.8
3.8
4.6
4.6
5.2
5.8
5.8
419
100
Uncoded 8QAM
Ungerboeck rate 3/4
101
BER
102
103
104
105
106
10
12
Eb/N0 (dB)
14
16
18
20
100
Uncoded 8QAM
Pragmatic rate 3/4
101
BER
102
103
104
105
106
Figure 7.84
10
12
Eb/N0 (dB)
14
16
18
20
Bit error probability: (a) rate 3=4 Ungerboeck TCM-16QAM, (b) rate 3=4 pragmatic TCM-16QAM. (Courtesy of Lin
Kiat Peh.)
420
Demodulation
mapped into an I-Q vector via symbol mapping by set partitioning for Ungerboeck TCM or
via pragmatic symbol mapping for pragmatic TCM. The squared Euclidean distance
between the code symbol of a branch and the corresponding received I-Q vector (at the
outputs of the samplers) is calculated and added to the previous path squared Euclidean
distance for each node. At each node in the trellis, the Viterbi algorithm chooses a branch
that belongs to a path with the smallest path squared Euclidean distance. This retained
path is called the survivor path. All other paths at each node with higher squared Euclidean
distances are not retained. In the case of a tie, a survivor path might be chosen randomly.
At the end of the decoding window, the survivor path with the smallest path squared
Euclidean distance is selected and the associated code vector is chosen as the transmitted
code vector.
malized to rEs =2 according to Figure 6.49. The path that associates with the code vector
(00,00,00,00,00) has a normalized minimum squared Euclidean distance of 6.5 and hence is
selected as the decoding path as shown in Figure 7.85. The two symbol errors that are caused
by two underlined voltage errors are corrected.
1, 1
0.5, 1
0.5, 1
00
00
1, 1
1, 1
State
00
4
00
4.3
11
11
8.3
6.5
11
00
6.5
11
00
10.5
11
11
8.5
11
10.5
00
01
01
12.5
10
8.5
10
+
10
14.3
01
Figure 7.85
10
14.5
10
10
10
01
10.3
10
00
11
00
01
01
6.5
+ 16.5
12.5
01
01
16.5
Unquantized soft Viterbi algorithm for Ungerboeck TCM-QPSK with rate , K = 3 convolutional code.
11
421
0:5; 0:1; 0:8; 0:6; 0:3 normalized to rEs =2. The code symbols labeled on the branches are
mapped to I-Q vectors according to Figure 6.55 as follows: 011 ! 0:92; 0:38,
000 ! 0:38; 0:92, 001 ! 0:38; 0:92, 010 ! 0:92; 0:38, 111 ! 0:92; 0:38,
100 ! 0:38; 0:92, 101 ! 0:38: 0:92, 110 ! 0:92; 0:38. The decoded code
symbol sequence is (000,010,011,001,011) as indicated by the heavy dashed line.
Figure 7.86
0.3, 0.5;
0.2, 0.7;
0.3, 0.5;
0.1, 0.8;
0.6, 0.3
State
0.1
0.7
0.9
100
000
1.5
00
110
010
1.4
0.5
1.7
1.2 110
010
1.6
+
1.6
1.1
0.9
10
111
011
1.1
101
001
Figure 7.87
01
1.7
111
011
2.5
100
000
101
001
1.2
1.3 11
Unquantized soft Viterbi algorithm for Ungerboeck TCM-8PSK with rate 2=3, K = 3 convolutional code.
422
Demodulation
7.26 Summary
In this chapter we presented the principles of coherent demodulation and noncoherent demodulation. A digital
demodulator basically consists of a signal processor and an optimum detector. The signal processor processes the
received signal plus noise into a decision vector. The optimum detector uses the decision vector to recover the
transmitted information with a minimum bit error probability. Four fundamental signal processors were studied.
They are the coherent matched lter and correlator, which are equivalent at the sampling time, and the matched
lterenvelope detector (noncoherent matched lter) and the quadrature correlatorsquare law detector
(noncoherent correlator), which are also equivalent at the sampling time. Two optimum binary detectors were
studied, namely, the threshold detector and the maximum detector (comparator). These optimum detectors are
governed by the maximum likelihood criterion. The threshold detector is used with coherent binary demodulation, and the maximum detector can be employed for both coherent and noncoherent binary demodulations.
We then proceeded to analyze the performance of binary demodulation such as coherent PSK, coherent DPSK,
direct detection ASK for lightwave signals, coherent MSK and precoded MSK and GMSK, noncoherent FSK and
MSK, and noncoherent DPSK.
Next, we studied the detector for M-ary signals, namely, the minimum Euclidean distance detector for coherent
demodulation and the M-ary maximum detector for both coherent and noncoherent demodulation. Both
detectors are governed by the maximum likelihood criterion. Afterward, we provided a detailed design of the
coherent L-path demodulator based on the signal space approach. This optimum demodulator employs either L
matched lters or L correlators for signal processing and either the minimum Euclidean distance detector or the
metric computer and maximum detector for detection. The design of a coherent M-path demodulator was
presented next. This type of demodulator employs either M matched lters or M correlators for signal processing
and a maximum detector for detection. We then provided the design of a noncoherent M-path demodulator,
which employs either M matched lterenvelope detectors or M quadrature correlatorsquare law detectors for
signal processing, and a maximum detector for detection. Using the above designs we analyzed the performance
of coherent MASK, coherent MPSK, coherent DMPSK, noncoherent DMPSK, noncoherent =4-DQPSK, coherent
MQAM and DMQAM, coherent CSK and MFSK, and noncoherent CSK and MFSK. We also considered two methods
of detection for coherent CPM, namely, sequence detection via the Viterbi algorithm and symbol-by-symbol
detection. Noncoherent CPM with symbol-by-symbol detection was also discussed. We concluded our study of
M-ary demodulation techniques with a summary of their performance in an AWGN channel.
The study of OFDM demodulation was conducted in fair detail. We used the IEEE 802.11a standard as an
example for our discussion, which included symbol timing estimation; frequency offset estimation; and carrier
phase synchronization. The chapter ends with the performance analysis of TCM based on the asymptotic coding
gain and simulation results.
423
Problems
100
101
Q(X)
102
103
104
105
106
107
Figure 7.88
0.5
1.5
2.5
X
3.5
4.5
Problems
1.
2.
3.
4.
5.
6.
7.
8.
9.
Determine the output of the matched lter of the pulse p(t) when the input waveform is
st Apt pt Tb where p(t) is the squared pulse of unit amplitude and
duration Tb .
Given two complex vectors u and v, show that ju vj kukkvk with equality if and only
if u and v are dependent, that is, u v for some arbitrary constant (the Cauchy
Schwarz inequality). The notation u stands for the conjugate transpose of the vector u.
2Bt
Determine the output of the matched lter of the waveform st 2B sin2Bt
.
Consider the complex voltage signal st Apt jpt, where p(t) is the squared
pulse of unit amplitude and duration Tb . Find the output voltage of the complexvalued matched lter at the sampling time t Tb .
In Figure 7.3 assume that the sampling time has an error ", that is, T + ". Find the
output signal-to-noise ratio of the matched lter.
Determine the output signal-to-noise ratio of the matched lter of the pulse p(t) at the
sampling time T " (0 " T ), where p(t) is the squared pulse of unit amplitude
and duration T , given the input signal st Apt pt T .
Find a pulse shape pt with Fourier transform P f whose matched
lter works for
p
both itself and its counterpart qt with Fourier transform Q f P f .
Using the time-domain approach design a matched lter for signal s(t) plus additive
non-white noise n(t) with real autocorrelation Rn whose Fourier transform Sn f is
a rational function. This can be achieved by rst designing a noise whitening lter
then the matched lter.
Using the frequency-domain approach, design a matched lter for the signal s(t) with
additive non-white noise n(t) with rational power spectral density function Sn f .
424
Demodulation
10.
11.
12.
13.
14.
15.
16.
Consider a bank of L matched lters that receive the same signal s(t) but with different
attenuation ai , i = 1, 2, . . . , L; that is, the ith matched lter input is ai st. The matched
lter outputs are summed and sampled at time t = T where T is the duration of the
signal s(t). The noise signals n1 t; n2 t; :::; nL t are independent and have identical
power spectral density N0 =2. Find the output signal-to-noise ratio of the sum signal
assuming the signal energy is E.
Consider a bank of L matched lters that receive the same signal s(t) but with different
attenuation ai , i = 1, 2, . . . , L; that is, the ith matched lter input is ai st. The matched
lter outputs are summed and sampled at time t = T where T is the duration of the
signal s(t). The noise signals n1 t; n2 t; :::; nL t are independent and have identical
power spectral density N0 =2. Assume that the knowledge of the attenuation ai is
available, the sample of the ith matched lter is then weighted by ai. Find the output
signal-to-noise ratio of the sum signal assuming the signal energy is E.
Consider a PSK demodulator using a local reference carrier that is offset in phase with
respect to the received signal by . Determine the bit error probability.
Consider a PSK demodulator with a sample timing error ", that is, the sampling time
for the ith bit is t = iTb + ". Determine the bit error probability.
Consider a PSK demodulator using a local reference carrier that is offset in phase with
respect to the received signal by , and with a sample timing error ", that is, the
sampling time for the ith bit is t = iTb + ". Determine the bit error probability.
Consider the carrier recovery circuit in Figure 7.89. Show that the carrier of a PSK
signal s(t) can be reproduced at the output but it has a phase ambiguity of radians.
This means that the recovered carrier has a phase (modulo ). This might result in
inverted data at the detector output. Suggest a method to correct this problem without
using differential encoding.
A PSK signal with bit rate 1/Tb = 19.6 kbps is demodulated to the baseband signal,
which is a random sequence of antipodal signals. Use the circuit in Figure 7.90 to
PSK
Figure 7.89
Frequency
divider by
2
BPF
@2fc
Squarer
BPF
Delay
Tb/2
1, x > 0
sgn( x) =
1, x 0
Figure 7.90
cos 2fc t
Sgn
425
Problems
Antipodal
sequence
Figure 7.91
Squarer
Sgn
BPF
r2(t)
r1(t)
DPSK
Integrator
Sampler
Threshold
detector
1 Bit
delay
Figure 7.92
17.
18.
19.
20.
21.
22.
recover the data clock signal from the antipodal signal sequence via simulation. It
should be a periodic waveform with period Tb. The bandpass lter (BPF) should have
a center frequency equal to 1/Tb and a bandwidth B 1/Tb.
Consider a squaring timing recovery circuit for PSK signal shown in Figure 7.91. The
PSK signal is rst demodulated to baseband to produce an antipodal signal sequence.
The data clock is recovered by using a squarer. Let the antipodal pulse shape be
denoted as p(t) and its Fourier transform be P( f ). Using the Poisson sum formula
P
P
given by k pt kTb Tb1 l Pl=Tb exp j2lt=Tb , show that a period signal
with period Tb can be produced at the output. The sgn-function is given in Problem 16.
Consider the suboptimum demodulator of a DPSK signal shown in Figure 7.92.
(a) Find the signal r1 t and r2 t when noise is absent.
(b) Determine the detection threshold and decode the data bits for the following
received differentially encoded bit sequence 0110001.
(c) Assume the case of small noise, derive the approximate bit error probability.
A direct detection optical receiver operates at an input power level of 30 dBm. The
noise spectral density is 1016 mA2/Hz. The photo detectors responsitivity is 0.5 and
re . It is desired to achieve a bit error probability of at least 108 , determine the
maximum bit rate.
Prove that if s1 t and s2 t are orthogonal baseband signals with identical absolute
bandwidth B, then s1 t cos 2fc t and s2 t cos 2fc t are orthogonal signals as well,
provided that fc 4B. Give an example for which the statement fails when fc 5B.
Consider an FSK signal employing a raised-cosine pulse shape with a roll-off factor .
The bit rate is Rb 1=Tb . Determine the minimum frequency spacing for strict
orthogonality.
Consider a noncoherent orthogonal FSK
demodulator
employing noncoherent corre
p
lators. The received signal is st 2=Tb cos2f1 f t . The frequency
error is f 1=8Tb and the phase is =4. Ignoring the high-frequency terms, nd
the output sample of the noncoherent correlator of the signal branch. Find the power
loss relative to the case of no frequency error. Assume that fi , i = 1, 2 are integer
multiples of the bit rate and noise is negligible.
426
Demodulation
23.
24.
25.
26.
27.
28.
29.
30.
n
o
M =2
M =2
M =2
Consider the set of biorthogonal signals fsi tgi1 ; fsi tgi1 , where fsi tgi1
is the set of M/2 orthogonal signals.
(a) Design the demodulator using M/2 correlators, M/2 sign functions, and a maximum detector.
(b) Derive the union bound for the symbol error probability.
The null-to-null bandwidth of PSK is 2=Tb;PSK and that of FSK is 3=Tb;FSK when the
squared pulse shape is used. Consider two communication systems employing PSK
and FSK with coherent detection. Both systems have identical bandwidth which is
taken to be the null-to-null bandwidth of their signals. This implies that FSK would
have to transmit at a lower rate than PSK.
(a) If the bit rate of PSK is Rb;PSK what would be the bit rate of FSK.
(b) Assume that both systems receive signals with the same power P. What would be
the bit error probability of each system in terms of P and Tb;PSK ?
(c) Both systems are required to have identical bit error probability, how much more
(or less) power does FSK need?
The null-to-null bandwidth of PSK is 2=Tb;PSK and that of MSK is 1:5=Tb;MSK when
the squared pulse shape is used. Consider two communication systems employing
PSK and precoded MSK with coherent detection. Both systems have identical
bandwidth which is taken to be the null-to-null bandwidth of their signals.
(a) If the bit rate of PSK is Rb;PSK what would be the bit rate of MSK.
(b) Assume that both systems receive signals with the same power P. What would be
the bit error probability of each system in terms of P and Tb;PSK ?
(c) Both systems are now required to have identical bit error probability, how much
more (or less) power does MSK need?
Find the I-data and Q-data of MSK for the input bit sequence 110011001.
Consider a PSK receiver with the following parameters: system signal-to-noise ratio
SNR = 7 dB, bit rate = 400 kbps, noise bandwidth = 500 kHz, system noise temperature = 500 K. What is the received power? Calculate the bit error probability.
Consider a PSK system where the a-priori probabilities of the transmitted bits are
Pr(1) = q and Pr(0) = 1 q. Derive the bit error probability.
Perform the demodulation of DPSK given the phase sequence 2; 2; ; 2; 2;
3; 3.
Consider a PSK signal with the squared pulse shape of unit amplitude and duration Tb ,
and carrier frequency fc 1=4Tb. This PSK signal is passed through a bandpass lter
transfer function
H f sin f fc Tb =4= f fc Tb =4
sin f fc Tb =4= f fc Tb =4
31.
Show that the output of the lter is indeed an MSK signal (serial MSK).
The two-path or IQ demodulator makes a decision on the two-dimensional vector
X si N. The symbol vector si si1 si1 t can be mapped to the complex
symbol si si1 jsi2 . The norm ksi k is identical to the magnitude jsi j. Design an
equivalent complex demodulator using the complex reference carrier ej2fc t.
427
Problems
MPSK
Figure 7.93
( )M
BPF
@ Mfc
Frequency
divider by
M
cos 2 fct
33.
34.
35.
36.
37.
38.
39.
40.
41.
is
p p7.94,
p pE
t
example, we have si 3 E= 2 3 E = 2 p
for
symbol 1000. The minimum
Euclidean distance of this signal space is dmin 2E .
Consider two communication systems employing 16PSK and 16QAM with coherent
detection. Assume that both systems have the same received power P. Both systems
are required to have approximately the same bit error probability of 105 , what would
be the bit rate of 16PSK relative to that of 16QAM?
Consider a 16QAM receiver with the following parameters: system signal-to-noise
ratio SNR = 17 dB, bit rate = 24 Mbps, noise bandwidth = 30 MHz, system noise
temperature = 500 K. What is the received power? Calculate the bit error probability.
Does the carrier recovery circuit in Figure 7.93 for MPSK with M = 4 apply to any
rectangular QAM?
428
Demodulation
Q
0010
0110
1110
1010
1111
1011
+3
0011
3
0001
0111
1
+1
0101
+1
+3
1101
1001
1100
1000
1
0000
0100
3
Figure 7.94
42.
43.
Consider coherent 64CSK and 64QAM. If the bit rate of 64CSK is Rb , what would be
the bit rate of 64QAM, assuming that both systems receive the same power and are
required to have identical bit error probability of 106?
A channel produces a random received signal-to-noise ratio . It is desired to keep the
received signal-to-noise ratio constant at the value SNR. Let thetransmit power P
1
vary with subject to the average power constraint P, that is, 0 Pf d P,
where f is the density of . The power adaptation strategy is given as
SNR=. Find a closed-form expression for SNR. If is uniformly distributed
P=P
between 5 and 50, nd the value of SNR. Based solely on the minimum Euclidean
distances, if 64FSK is used at minimum , what modulation technique should be used
for maximum with the least modication to the transmitter and receiver?
What are the noiseless spectral efciencies of CSK assuming raised-cosine pulse
shapes with roll-off factors 0 and 0:5? At these spectral efciencies, what are the
required Eb =N0 dictated by the Shannon channel capacity for reliable transmission?
Compare the result with DPSK and draw the conclusions.
Consider OFDM with 16 QPSK-subcarriers and a single carrier 256QAM.
(a) For the same bit rate and assuming squared pulse shape, compare their
bandwidths.
(b) For the same received power, and assuming that the bit error probability of OFDM
is twice that of 256QAM, compare the bit rates.
Consider OFDM-1 with 64 QPSK-subcarriers and OFDM-2 with 16 64QAMsubcarriers. Both have the same received power
(a) Determine the subcarrier bit rate of OFDM-1 in terms of the subcarrier bit rate of
OFDM-2 for the same subcarrier bit error probability of 104.
44.
45.
46.
47.
M !1
429
Problems
48.
49.
50.
51.
52.
53.
54.
55.
56.
57.
(b) Determine the bit rate of OFDM-1in term of the bit rate of OFDM-2.
(c) Determine the bandwidth of OFDM-1in term of the bandwidth of OFDM-2.
Consider a CROSS-8QAM, which can be obtained from the rectangular 16QAM by
retaining four inner signal points and four corner signal points and deleting eight edge
signal points.
(a) Find the number of nearest neighbors for the inner signal point and the corner signal
point and then the average number of nearest neighbors for the signal space.
(b) Derive the bit error probability using the union bound.
(c) For the same bit error probability of 105 nd the required received power of
CROSS-8QAM relative to that of 8PSK.
Derive the coding gains for convolutional encoded PSK with hard decoding and soft
decoding at low bit error probability. Compare the coding gains. (Hint: use the rst
2
term of the union bound and the following Chernoff bound Qx ex =2 . These
coding gains are referred to as asymptotic coding gains.)
Derive the bit error probability for a block code with hard decoding and soft decoding,
assuming the block code minimum Hamming distance is dmin and the number of code
words of weight d is Ad . Derive the coding gains for hard decoding and soft decoding
at low bit error probability; assume that the modulation is PSK. Compare the coding
gains. (Hint: use the rst term of the union bound of the bit error probability and the
2
following Chernoff bound Qx ex =2 . These coding gains are referred to as
asymptotic coding gains.)
Consider a PSK signal encoded by a rate convolutional code with constraint length
K = 4. Assuming that soft decision decoding is employed, what is the required signalto-noise ratio per bit at Pb 106 ? What is the coding gain at this bit error
probability?
Consider a TCM-8PAM obtained via set partitioning. Assume that the asymptotic
coding gain for this TCM is 4.6 dB; calculate the Euclidean free distance of the
encoder.
Consider the TCM-8PSK shown in Figure 6.53. Assume that a convolutional code
2
with rate is employed and that the Euclidean free distance is dfree
d02 2d12 , where
q
p
p
d0 2 2Es and d1 2Es . Calculate the asymptotic coding gain for TCM8PSK relative to uncoded QPSK.
Consider the TCM-16QAM shown in Figure 6.55. Assume that the asymptotic coding
gain for the TCM-16QAM is 6 dB relative to uncoded 8PSK. Calculate the Euclidean
free distance of the TCM.
Design a TCM-64QAM with separate I and Q encoding and decoding.
Perform the Viterbi algorithm for Ungerboeck TCM-QPSK using the trellis in
Figure 4.13 for the received
r 0:7; 0:5; 0:8; 0:7; 0:8; 1; 0:9;
pvector
0:6; 1; 0:5 normalized to rEs =2, where r is the code rate.
Perform the Viterbi algorithm for Ungerboeck TCM-8PSK using the trellis in
Figure 7.87 for the received vector r 0:6; 0:2; 0:3; 0:7; 0:9; 0:5; 0:3; 0:8;
p
0:4; 1 normalized to rEs , where r is the code rate.
430
Demodulation
Further reading
Simon et al. [1] is recommended for further study of additional modulation and demodulation techniques. For readers who are interested in optical communications we recommend
[2]. References [911, 1317] provide a detailed description of OFDM.
Bibliography
1. M. K. Simon, S. M. Hinedi, and W. C. Lindsey, Digital Communication Techniques.
Signal Design and Detection, Englewood Cliffs, NJ: Prentice-Hall, 1995.
2. G. Keiser, Optical Fiber Communications, 4th edition, New York: McGraw-Hill, 2007.
3. K. Murota and K. Hirade, GMSK modulation for digital mobile radio telephony,
IEEE Trans. Commun., Vol. 29, No.7, pp. 10441050, 1981.
4. T. S. Rappaport, Wireless Communications, 2nd edition, Upper Saddle River, NJ:
Prentice-Hall, 2002.
5. R. E. Ziemer and R. L. Peterson, Introduction to Digital Communication, 2nd edition,
Upper Saddle River, NJ: Prentice-Hall, 2001.
6. S. Haykin, Communication Systems, 3rd edition, Chichester: John Wiley & Sons, 1994.
7. F. Xiong, Digital Modulation Techniques, London: Artech House, 2000.
8. J. G. Proakis and M. Salehi, Digital Communications, 5th edition, New York: McGrawHill, 2008.
9. IEEE Standards 802.11a, 1999.
10. IEEE Standards 802.11g, 2003.
11. M. Segkos, Advanced techniques to improve the performance of OFDM Wireless
LAN, unpublished Masters thesis, Naval Postgraduate School, Monterey, CA, June
2004.
12. G. C. Clark, Jr. and J. B. Cain, Error-Correction Coding for Digital Communications,
New York: Plenum Press, 1981.
13. T. Pollet, M. van Bladel and M. Moeneclaey, BER sensitivity of OFDM systems to
carrier frequency offset and Wiener phase noise, IEEE Trans. Commun., Vol. 43, No.
2, Part 3, pp. 191193, 1995.
14. P. H. Moose, A technique for orthogonal frequency division multiplexing frequency
offset correction, IEEE Trans. Commun., Vol. 42, No. 10, pp. 29082914, 1994.
15. J.-J. Van de Beek, M. Sandell, and P. O. Brjesson, ML estimation of time and
frequency offset in OFDM systems, IEEE Trans. Signal Processing, Vol. 45, No. 7,
pp. 18001805, 1997.
16. M. Speth, D. Daecke, and H. Meyr, Minimum overhead burst synchronization for
OFDM based broadband transmission, IEEE Global Telecommunications Conference,
Vol. 5, pp. 27772782, 1998.
431
Bibliography
17. J.-J. van de Beek, O. Edfors, M. Sandell, S. K. Wilson, and O. Borjesson, On channel
estimation in OFDM systems, IEEE Proc. VTC-1995, Vol. 2, pp. 815819, 1995.
18. J. B. Anderson, T. Aulin, and C. E. Sundberg, Digital Phase Modulation, London:
Plenum Press, 1986.
19. J. R. Barry, E. A. Lee, and D. G. Messerschmitt, Digital Communication, 3rd edition,
Boston, MA: Kluwer Academic Publishers, 2004.
20. S. Benedetto, E. Biglieri, and V. Castellani, Digital Transmission Theory, Englewood
Cliffs, NJ: Prentice-Hall, 1987.
21. R. E. Blahut, Digital Transmission of Information, London: Addison-Wesley, 1990.
22. R. G. Gallager, Principles of Digital Communication, Cambridge: Cambridge
University Press, 2008.
23. R. D. Gitlin, J. F. Hayes, and S. B. Weinstein, Data Communications Principles,
London: Plenum Press, 1992.
24. A. Lapidoth, A Foundation in Digital Communication, Cambridge: Cambridge
University Press, 2009.
25. S. Lin and D. J. Costello, Jr., Error Control Coding, 2nd edition. Pearson Education,
2004.
26. U. Madhow, Fundamentals of Digital Communication, Cambridge: Cambridge
University Press, 2008.
27. M. B. Pursley, Introduction to Digital Communications, Upper Saddle River, NJ:
Pearson Prentice-Hall, 2005.
28. J. I. Marcum, Statistical theory of target detection by pulsed radar, IEEE Trans. Inf.
Theory, Vol. 6, No. 2, pp. 5967, 1960.
29. B. Sklar, Digital Communications. Fundamentals and Applications, 2nd edition, Upper
Saddle River, NJ: Prentice-Hall, 2001.
30. A. J. Viterbi and J. K. Omura, Principles of Digital Communication and Coding, New
York: McGraw-Hill, 1979.
31. S. G. Wilson, Digital Modulation and Coding, Englewood Cliffs, NJ: Prentice-Hall,
1996.
32. J. M. Wozencraft and I. M. Jacobs, Principles of Communication Engineering, New
York: Wiley, 1965; Waveland Press, 1990.
Spread spectrum
Introduction
A spread spectrum modulated signal has a bandwidth much larger than the bandwidth of its
narrowband counterpart for transmitting the message signal. For example, a spread spectrum PSK signal employed in the forward link of the 2G cellular standard IS-95 carrying a
coded message signal of symbol rate 19.2 ksps has a bandwidth of 1.25 MHz. The narrowband PSK signal would only require a bandwidth of 19.53 kHz. This is an increase of
bandwidth by a factor of 64. What is the reason behind the use of spread spectrum
modulation? Historically, spread spectrum modulation originated from military applications. The purpose is to protect the received signals integrity by reducing the effectiveness
of a jamming signal. In order to jam a spread spectrum signal, the jamming signal must
distribute its xed transmitted power over a larger spread bandwidth. This would lower the
magnitude of its power spectral density and correspondingly its jamming power in the
smaller message bandwidth. There are basically two types of spread spectrum modulated
signals: direct sequence (DS) and frequency hop (FH).
433
{0,1},Tb
Repeat
N times
{0,1},Tc
Mapping
01
1 1
Pulse
shaping
DS-PSK
{0,1},Tc
cos 2fc t
PN sequence
generator
Figure 8.1
2Tc
0
Figure 8.2
Tc
3Tc
4Tc
The main feature of direct sequence spread spectrum is the PN sequence. There are many
types of PN sequences that can be used in practice, but the most important one is the
maximal sequences (also called maximal-length sequences) or for short, m-sequences. They
can be generated by linear feedback shift-registers (LFSR). When the shift register has n
stages the m-sequence has a period of N 2n 1 chips. In practice a zero can be appended
at the end of the m-sequence to produce an augmented PN sequence of period 2n chips. An
m-sequence can be generated by a monic binary irreducible primitive polynomial h(x):
h x 1 h1 x h2 x2 hn1 xn1 xn ; hi 2 f0; 1g
(8:1)
434
Spread spectrum
{0,1},Tb
Mapping
01
1 1
Repeat
N times
Pulse
shaping
DS-PSK
cos 2fc t
Mapping
01
1 1
PN sequence
generator
Figure 8.3
h1
Figure 8.4
h2
hn1
h3
ci1
ci2
ci3
x2
x3
...
cin1
cin
x n1
xn
(8:2)
ones and 2
1 zeros. The discrete periodic autocorrelation of
An m-sequence has 2
an m-sequence is a two-valued function and is given by the number of agreements minus
disagreements for the overall length of the sequence and a shifted replica of itself. These two
values are Rm i 2n 1 for i 0 and Rm i 1 for i 6 0. The m-sequence has the
smallest side lobe magnitude of any binary sequence with the same period. Applying the
mapping of 0 ! 1 and 1 ! 1 to an m-sequence we obtain the same discrete periodic
autocorrelation, that is,
n1
n1
Rm i
n
2X
1
ck cki
k0
2n 1; i 0
1;
i 6 0
(8:3)
Figure 8.5 shows the autocorrelation function of a periodic m-sequence when a unit
amplitude squared pulse of duration Tc is employed as the pulse shape of a chip.
435
Rm()
NTc
Tc
Tc
Figure 8.5
NTc
Figure 8.6
x2
x3
m-sequence is 1 1 1 0 1 0 0. It has been established that the number of agreements minus the
number of disagreements for this m-sequence and any of the circular shifted replica of itself
is 1.
Phase shift keying is not the only carrier modulation for direct sequence spread
spectrum. The reverse link of IS-95 employs code shift keying (CSK) as a carrier modulation
for direct sequence spread spectrum. In direct sequencecode shift keying (DS-CSK),
k log2 M bits form a symbol. Each of the M symbols is assigned a distinct
Walsh sequence from the set of M Walsh sequences that can be obtained from the rows of
the Hadamard matrix in (2.50). The elements of the Walsh sequences are 1 or 1. To employ
modulo-2 addition instead of multiplication, the mapping of 1 ! 0 and 1 ! 1 can be
applied to the M Walsh sequences to obtain the equivalent M Walsh sequences
whose elements are in {0,1}. Let Rb be the bit rate, then the symbol rate Rs is Rb/k.
Consequently, the Walsh chip rate Rw is MRs. The PN sequence chip rate Rc is selected to
be an integer multiple of the Walsh chip rate Rw , that is, Rc = NRw , where N is an integer
greater than or equal to 1. Thus, expressing the symbol rate in terms of the PN chip rate we
have Rc = NMRs. Hence, the spreading factor of DS-CSK is NM. The number of PN chips
per Walsh chip is N. The PN sequence period can be an integer multiple of the Walsh period
(the symbol period). To generate the DS-CSK signal, the Walsh chips of 0s or 1s are rst
repeated N times. Then, they are modulo-2 added to the PN chips of 0s or 1s. The mapping of
0 ! 1 and 1 ! 1 is then applied to the resulting spread Walsh symbol. A pulse shape
assigned to the chip then modulates a carrier. Figure 8.7 illustrates the generation of a DSCSK signal.
436
Spread spectrum
{0,1}
k1
Walsh
sequence {0,1}
generator
and repeat
Walsh chip
N times
Mapping
01
1 1
Pulse
shaping
DS-CSK
cos 2fct
PN sequence
generator
Figure 8.7
Orthogonal covering
Orthogonal covering or code division multiplexing (CDM) is a technique that employs
orthogonal functions to multiplex different user bit streams onto one carrier without incurring interference between user data. Orthogonal Walsh functions are used in practice for this
technique, such as in the forward link of IS-95. The concept is similar to the use of two
orthogonal carriers cos 2fc t and sin 2fc t in I-Q modulation techniques. These two
orthogonal carriers coexist in the same channel without mutual interference. Orthogonal
covering is also employed by DS systems such as the 3G cellular systems, namely,
WCDMA or CDMA 2000. Each user bit modulates an assigned Walsh sequence (symbol)
of 0s and 1s and of chip rate Rw MRb , where Rb is the bit rate. This is done by repeating the
data bit M times and then modulo-2 adding the repeated bits to the Walsh chips. A PN
sequence of 0s and 1s of chip rate Rc = N Rw, where N is an integer greater than or equal to 1,
is added (modulo-2 ) to the modulated Walsh chips, where each Walsh chip is repeated N
times. The PN sequence is common to all users. The mapping of 0 ! 1 and 1 ! 1 is
then applied to the resulting spread Walsh symbol. The multiplexing is then performed by
algebraically adding the symbols of all users together. The set of M Walsh functions can
accommodate up to M users. The sum of M user symbols serves as the baseband signal of a
transmitted PSK signal with carrier cos 2fc t. Figure 8.8 illustrates the basic concept of
Walsh orthogonal covering.
To provide an effective method for carrier phase synchronization at the receiver, two
different PN sequences can be employed for spreading in conjunction with I-Q or quadrature modulation. The I-channel employs a common spread PNI sequence for all users and
the Q-channel employs a common spread PNQ sequence for all users as indicated in
Figure 8.9. Note that the same sum of M user symbols is used in both I- and Q-channels.
This technique is referred to as quadrature spreading.
437
User channel 1
{0,1}
Repeat
M times
Mapping
01
1 1
Repeat
N times
{0,1}
{0,1}
Walsh
sequence 1
PN sequence
Pulse
shape
cos 2fct
User channel M
Figure 8.8
PNI sequence
User channel 1
{0,1}
cos 2fct
Mapping
I
01
1 1
{0,1}
Repeat
M times
Repeat
N times
Mapping
Q
01
1 1
{0,1}
Walsh
sequence 1
{0,1}
Pulse
shape
Pulse
shape
PNQ sequence
sin 2fct
I
User channel M
Figure 8.9
0
60
6
H4 4
0
0
0
1
0
1
0
0
1
1
3
0
17
7
15
0
(8:4)
(8:5)
These four Walsh sequences represent four orthogonal channels. Let the data sequences of
these four channels be given by (each data bit is repeated four times)
d 1 0000; d 2 1111; d 3 0000; d 4 1111
(8:6)
438
Spread spectrum
The resulting symbols in each channel are represented by the following sequences via
modulo-2 addition:
s1 d 1 w1 0000; s2 d 2 w2 1010;
s3 d 3 w3 0011; s4 d 4 w4 1001
(8:7)
(8:8)
Note that scaling does not alter the orthogonality of the Walsh sequences (sequence and
function are used interchangeably since a sequence becomes a function when a positive
squared pulse shape of duration Tc is assigned to a +1 chip or binary 0 and a negative squared
pulse shape of duration Tc is assigned to a 1 chip or binary 1), therefore, let us assume that
we scale the symbol s1 by a factor of two, then, the orthogonal covering symbol is given by
s 2s1 s2 s3 s4 1 5 1 1
(8:9)
Figure 8.10 shows the waveform s(t) of the orthogonal covering symbol s using the squared
pulse shape.
Now let us consider a simple process to recover the data bit in each of the four channels.
Let w1 t; w2 t; w3 t; and w4 t represent the Walsh functions whose corresponding
sequences are w1 ; w2 ; w3 ; and w4 , respectively. Dene the normalized inner product of
the two sequences s and wi with elements in f1; 1g as follows (the mapping
0 ! 1 and 1 ! 1 must be applied to the binary Walsh sequences before calculating
the inner products):
1 4Tc
t
=4
stwi tdt; i 1; 2; 3; 4
(8:10)
s;
w
sw
h i i=4
i
4Tc 0
We now have the following results:
hs; w1 i =4 2; hs; w2 i =4 1; hs; w3 i =4 1; hs; w4 i =4 1
(8:11)
Suppose we employ a threshold detector with the threshold set to zero, and for any inner
product with a positive value we detect bit 0, and negative value bit 1. Then, for channel 1
we detect bit 0, for channel 2 bit 1, for channel 3 bit 0, and for channel 4 bit 1.
s(t)
t
0
Figure 8.10
Tc
2Tc
3Tc
4T
439
The above orthogonal covering technique applies to user channels with the same common
rate, that is, a multiple of any user channel rate. Repetition of data bits must be used to bring
the user channel rates to a common rate before orthogonal covering. Since the spread
bandwidth is the same for all users as in WCDMA or CDMA 2000, multi-rate transmission
from the base station requires different spread factors for different user channels. This
necessitates the use of variable-length orthogonal sequences that can be constructed via a
tree with Walsh sequences as its branches (rows of Hadamard matrix with binary elements):
1. Start at the root of the tree with binary 0.
2. Move to the next branch to create sequences with twice the length. The upper branch
contains the cascade of two parent node sequences. The lower branch contains the
cascade of the parent node sequence and its complement.
3. Variable-length orthogonal sequences cannot come from the same branch of the tree.
Figure 8.11 shows the tree for variable-length orthogonal sequences. All sequences that
have the same length are in fact Walsh sequences but they do not appear in the order of the
rows of the Hadamard matrix. For example, if we assign Walsh sequence (0101) for channel
1 (spread factor of 4), then Walsh sequences (01010101) and (01011010) cannot be used for
any channel that requires a spread factor of 8. The remaining six Walsh sequences of length 8
are legitimate candidates for orthogonal covering.
Example 8.4 Orthogonal covering with 4-ary and 8-ary Walsh functions
Consider two user channels, one with bit rate Rb 1=Tb , where Tb is the bit time (denoted as
channel 1) and the other with bit rate 2Rb (denoted as channel 2). Note, one bit of channel 1
covers the same duration as two bits of channel 2. Channel 1 employs a Walsh sequence
w1 for spreading. This Walsh sequence has eight chips per bit; therefore, the chip rate is
8Rb. Channel 2 employs a Walsh sequence w2 for spreading. This Walsh sequence has four
chips per bit; therefore, the chip rate is also 8Rb. Hence, the two channels meet the requirement of common chip rate and spread bandwidth. Let the data sequences of these two
channels be given by (each data bit is repeated eight times for Channel 1 and four times
for Channel 2)
0000
00000000
00001111
00
0011
00110011
00111100
0
0101
01010101
01011010
01
0110
Figure 8.11
01100110
01101001
440
Spread spectrum
(8:12)
Furthermore, let the Walsh sequences be selected from Figure 8.11 as follows:
w1 00001111; w2 0101
(8:13)
The resulting symbols in each channel are represented by the following sequences via
modulo-2 addition:
s1 d 1 w1 11110000; s2;0 d 2;0 w2 0101
and s2;1 d 2;1 w2 1010
(8:14)
(8:15)
Let us assume that we scale the symbol s1 by a factor of two, then, the orthogonal covering
symbol is given by
s 2s1 s2;0 ; s2;1 s11 ; s12 1 3 1 3; 1 3 1 3
(8:16)
We now calculate the following normalized inner products using the Walsh sequences
w1 1 1 1 1 1 1 1 1 and w2 1 11 1:
hs; w1 i=8 2; hs11 ; w2 i=4 1 and hs12 ; w2 i=4 1
(8:17)
Suppose we again employ a threshold detector with the threshold set to zero, and for any
inner product with a positive value we detect bit 0, and negative value bit 1. Then, for
channel 1 we detect bit 1, and for channel 2 bits 0 and 1.
(8:18)
(8:19)
441
(8:20)
442
Spread spectrum
Walsh
function 0
PN chip
1.2288 Mcps
Pilot
channel
(all 0s)
Walsh
function 23
Sync
channel
bits
1.2 kbps
Modulation
symbol
Code
symbol
Convolutional
encoder
r = 1/2 K = 9
Symbol
repetition
2.4 ksps
Block
interleaver
4.8 ksps
PN chip
1.2288 Mcps
Modulation
symbol
A
4.8 ksps
Walsh
function p
Paging
channel
bits
9.6 kbps
4.8 kbps
Convolutional
encoder
r = 1/2 K = 9 19.2 ksps
9.6 ksps
Symbol
repetition
19.2 ksps
8.6 kbps
4.0 kbps
2.0 kbps
0.8 kbps
Block
interleaver 19.2 ksps
A
19.2 ksps
Forward
traffic channel
information
bits for user
m (172, 80,
40, or 16
bits/frame)
PN chip
1.2288 Mcps
Modulation
symbol
Modulation
symbol
Code
symbol
Add frame
quality
indicators
for 9600 and
4800 bps rates
9.2 kbps
4.4 kbps
2.0 kbps
0.8 kbps
Long code
generator 1.2288 Mcps
Add 8 bit
encoder
9.6 kbps
tail
4.8 kbps
2.4 kbps
1.2 kbps
Decimator
Code
symbol
Convolutional
encoder
19.2 ksps
r = 1/2 K = 9
9.6 ksps
4.8 ksps
2.4 ksps
Modulation
symbol
Symbol
repetition
19.2 ksps
Walsh
function n
Power
control bits
PN chip
1.2288 Mcps
800 bps
MUX
Modulation
symbol
Block
interleaver 19.2 ksps
19.2 ksps
Long code
mask for
user m
Long code
generator
1.2288 Mcps
Decimator
Decimator
cos 2fc t
I(t)
Baseband
filter
A
Q
Figure 8.12
Baseband
filter
Q(t)
sin 2fc t
s(t)
443
The long code is generated with a 42-stage LFSR. It has a period of 242 1 chips. The
scrambled symbols are then spread by a designated Walsh sequence (the second to eighth rows
of the 64 64 Hadamard matrix). Figure 8.12 shows the generation of the paging channel.
The trafc channel carries voice or data to the mobile user. Up to 55 trafc channels are
available in each forward link. Some of the paging channels can be used for trafc channels.
Figure 8.12 shows the generation of the trafc channel. Voice data are selected on a frameby-frame (20 ms) basis. At 9.6 kbps (4.8 kbps) the frame contains 172 (80) information bits
followed by 12 (8) frame quality indicator bits (cyclic redundancy check (CRC) error
detection bits) and 8 encoder tailing bits used to reset the state of the convolutional encoder
after each frame. At 2.4 kbps (1.2 kbps) the frame contains 40 (16) information bits followed
by 8 encoder tailing bits. The information bits are convolutionally encoded and repeated to
produce a common symbol rate at 19.2 kbps. Block interleaving is implemented for protection against burst errors. The interleaved code symbols are then scrambled by a long code
PN sequence clocked at 19.2 kcps as in the paging channel to provide data privacy. The
decimated long code PN sequence is further decimated to yield a clock signal at 800 Hz by
sampling once every 24 chips. This clock is used by the power control bits at a rate of
800 bps to puncture the scrambled symbols. The power control bits are sent to the mobile
station to control the mobile transmit power. This control process equalizes the near-far
effect. The resulting symbol stream is spread by a designated Walsh sequence (the ninth to
32nd rows, and the 34th to 64th rows of the 64 64 Hadamard matrix). The above forward
trafc channel is for Rate Set 1 (RS1), which is specied in the 2G standard IS-95A. A later
version IS-95A species a Rate Set 2 (RS2) of 14.4, 7.2, 3.6, and 1.8 kbps. At 14.4 kbps
(7.2 kbps) the frame contains 267 (125) information bits followed by 12 (10) frame quality
indicator bits and 8 encoder tailing bits. At 3.6 kbps (1.8 kbps) the frame contains 55 (21)
information bits followed by 8 (6) frame quality indicator bits and 8 encoder tailing bits.
The convolutional code for RS2 is of rate . It is obtained from the convolutional code of
rate in RS1 by puncturing (deleting) two output code symbols for every six output code
symbols (which correspond to every three input code symbols). The 2.5G standard IS-95B
can assign up to eight Walsh functions to a dedicated user for a maximum rate of 115.2 kbps
(8 14.4 kbps), although in practice the rate is about half the maximum rate for constraints
imposed on the air interface.
444
Spread spectrum
To avoid the near-far effect, mobile signals on the same reverse link must arrive at the base
station with approximately the same signal power. Thus, CDMA requires near real-time
closed-loop power control for all mobile stations in the same cell. The base station measures
the mobile signal strength and sends an instruction on the forward link for the mobile station
to adjust its transmitted power up or down by an incremental unit amount. Assume perfect
power control, that is, all mobile signals in the same cell and on the same link arrive at their
base station with exactly the same power P. Furthermore, assume that the reverse link has
K mobile signals and each mobile signal has an identical spread factor equal to N. This
means that the decision sample for each mobile signal at the base station has an interference
variance (power) equal to P(K1)/N. Also, the output signal-to-interference ratio is equal to
N/(K1) (this is a conservative estimate since it does not take into account the modulation
type and whether the demodulation is coherent or noncoherent. A more precise analysis can
be found in a later section of this chapter). The output signal-to-interference ratio is the
factor that limits the number of voice channels in a CDMA cellular system like IS-95. Note
that the forward link of IS-95 has no multi-user interference because of the use of orthogonal
covering. Thus, the IS-95 forward link can carry up to 62 voice channels if no paging
channel is used. On the other hand, the IS-95 reverse link is constrained by multi-user
interference and carries a smaller number of voice channels unless other efcient techniques
are incorporated, such as voice activation, diversity at the base station, and sectoring. In
fact, the capacity of IS-95 is forward link-limited rather than reverse link-limited once these
efcient techniques are incorporated.
445
446
Spread spectrum
Access
channel
information
4.4 kbps
bits
(88 bits/frame)
Add 8 bit
encoder
tail
4.8 kbps
Code
Convolutional symbol
encoder
r = 1/3 K = 9 14.4 ksps
Code
symbol
Symbol
repetition
Code
symbol
64-ary
orthogonal
modulator
Block
interleaver 28.8 ksps
28.8 ksps
Modulation
symbol
(Walsh chip)
cos 2fct
I
I(t)
Baseband
filter
1/2 PN chip
delay = 406.9 ns
PN chip
1.2288 Mcps
Q(t)
Baseband
filter
s(t)
Long code
sin 2fct
Q-channel sequence
1.2288 Mcps
Long code
mask
Reverse
traffic
channel
information
bits (172,
80, 40, or 16
bits/frame)
8.6 kbps
4.0 kbps
2.0 kbps
0.8 kbps
Add frame
quality
indicators
for 9600 and
4800 bps rates
Add 8 bit
encoder
tail
9.2 kbps
4.4 kbps
2.0 kbps
0.8 kbps
Code
symbol
Convolutional
encoder
r = 1/3 K = 9 28.8 ksps
14.4 ksps
7.2 ksps
3.6 ksps
9.6 kbps
4.8 kbps
2.4 kbps
1.2 kbps
Code
symbol
Symbol
repetition
Code
symbol
Block
interleaver
28.8 ksps
28.8 ksps
I-channel sequence
1.2288 Mcps
Frame data rate
Modulation
symbol
64-ary
(Walsh)
orthogonal
4.8 ksps
modulator
(307.2 kcps)
cos 2fct
I
Data burst
randomizer
PN chip
1.2288 Mcps
1/2 PN chip
delay = 406.9 ns
Q
D
Long code
Q-channel sequence
1.2288 Mcps
I(t)
Baseband
filter
s(t)
Q(t)
Baseband
filter
sin 2fct
Long code
mask
Figure 8.13
447
DS signal
Narrowband
demodulator
PN sequence
Figure 8.14
The demodulation process of a direct sequence spread spectrum signal can be illustrated
by the general block diagram shown in Figure 8.14. The received DS signal is rst despread
by multiplying it by a synchronized replica of the PN sequence. This operation simultaneously spreads the jamming signal power over a bandwidth much larger than the data
bandwidth. The despread signal is now a narrowband signal and can be demodulated by a
conventional demodulator, depending on the modulation format. Note that the despreading
process provides no advantage with respect to white noise due to the fact that the spectrum
of white noise is theoretically innite and can no longer be spread. Thus the performance of
a DS signal in a white Gaussian noise channel is identical to the performance of its
narrowband counterpart. Only when a jamming signal is present the benet of spreading
obtained.
ATb di
J N
2
(8:22)
where the random variables J and N represent the jamming and noise samples, respectively,
and are expressed as follows:
1
J Aj cos
2
Tb
ct dt and N
Tb
ntct cos 2fc tdt
(8:23)
Let L be the number of 1s minus the number of 0s in the PN sequence c(t), and hence L can
be positive or negative. Using the fact that the chips in the PN sequence have 1 amplitude,
we have
J
1
Aj Tc L cos
2
(8:24)
448
Spread spectrum
t = Tb
Tb
Threshold
detector
c(t)cos 2fct
Figure 8.15
DS-PSK demodulator.
Assuming that the random phase is uniformly distributed, we calculate the jamming
variance as
2j
1 2 2 2
A T L
8 j c
(8:25)
Using the fact that c2 t 1, the noise variance can be calculated as follows:
2
N 0 Tb
4
(8:26)
Consequently, the output signal-to-noise ratio SNR0 and the output signal-to-jamming ratio
SJR0 are given by
SNR0
ATb di =22
A2 Tb
2Eb
N0
N0
2 2 !
2
ATb di =22
N
A
N
2
SJR0
2
SJRi
2
2
L
L
j
Aj
(8:27a)
(8:27b)
where SJRi A2 =A2j is the input signal-to-jamming ratio. Note that the output signal-tonoise ratio is identical to that of the narrowband PSK as expected since spreading the signal
has no effect on noise. On the other hand the output signal-to-jamming ratio is enhanced by a
processing gain equal to 2(N/L)2 relative to the input signal-to-jamming ratio. For example,
if c(t) represents a maximal sequence, then L = 1, and the processing gain is 2N2. Thus, tone
jamming is not effective against a DS signal with a large spread factor N. Note the difference
between spread factor and processing gain. The latter is modulation, demodulation, and
jamming dependent while the former is not.
J0 Tb
4
(8:28)
449
SJR0
ATb di =22
A2 Tb 2Eb
J0
J0
2j
(8:29)
A2 =2
Pj
(8:30)
where Pj is dened as the power of the jamming signal in the equivalent noise bandwidth of
the DS-PSK signal, which is equal to the chip rate 1/Tc:
Pj
J0
Tc
(8:31)
Thus we have
SJRi
A2 =2
A2 Tc
Eb
Pj
2J0
NJ0
(8:32)
(8:33)
The above result shows that a processing gain equal to 2N is achieved against broadband
jamming. The output signal-to-jamming-and-noise ratio is given by SJNR0
1 1
SJR1
0 SNR0 , where SNR0 2Eb =N0 . The jamming sample at the integrator output
is not Gaussian, but if N is large, the sample can be approximated as a Gaussian sample
(based on the central limit theorem, any broadband signal that is passed through a narrowband lter whose bandwidth is much smaller than that of the broadband signal produces an
output signal that is approximately Gaussian). Thus, the bit error probability of a DS-PSK
signal in the presence of a broadband jamming signal can be approximated as
p
Pb Q SJNR0
450
Spread spectrum
jamming the output signal-to-pulse jamming ratio is decreased by a factor equal to . Using
the Gaussian approximation and based on the total probability theorem, the bit error
probability for the DS-PSK signal in pulse jamming is given by
q
p
1 1
1
Q
SNR0
Pb Q
SJR1
SNR
(8:34)
0;p
0
If we consider the case when SNR0 SJR0;p , then the bit error probability can be approximated as follows:
p
p
p
Pb Q SJR0;p Q
2Eb =J0
2N SJRi Q
(8:35)
For a xed average jamming power, the jammer can select to maximize the bit error
probability. The worst-case bit error probability and the optimum can be found numerically as
Pb;max
0:083
0:083
;
N SJRi Eb =J0
0:71
0:71
N SJRi Eb =J0
(8:36)
For illustration purpose, consider the case of SJRi = 1/20 and N = 128, then Pb,max 0.013
and 0:11. For comparison purposes consider the case of broadband jamming in the
previous example. Assume that the effect of noise is negligible compared to that of
the jamming signal. Then,
the bit error
probability
pfor broadband
p
p jamming is given by
p
Pb Q SJNR0 Q 2N SJRi Q 2Eb =J0 Q 12:8 1:7 104 . This
shows that pulsed jamming is more effective than broadband jamming in general. The
disadvantage is the requirement of a high peak power jammer. To combat pulsed jamming,
error-correction coding may be employed.
(8:37)
where di,k 2 f1; 1g is the ith bit in the kth channel. The I- and Q-channel PN functions
are cI(t) and cQ(t), respectively. The arbitrary phasep represents the phase difference
between the received signal and the local carrier. The 2 factor means that half the power
goes into the I-channel and the other half goes into the Q-channel. We assume that noise n(t)
has the power spectral density N0/2 (W/Hz). The following operations are performed by the
demodulator to extract the I-decision variable:
451
(8:38)
M
M
A X
A X
p
di;k wk twl t cos p
di;k cI tcQ twk twl t sin
2 2 k1
2 2 k1
(8:39)
M
M
A X
A X
di;k cI tcQ twk twl t sin p
di;k wk twl t cos
p
2 2 k1
2 2 k1
The high-frequency terms are ignored because they will be integrated to zero, since as usual
we assume that the carrier frequency is an integer multiple of the symbol rate. Add (8.38)
and (8.39) and include noise. We get the signal and noise of the I-channel:
sI t stcI twl t cos 2fc t stcQ twl t sin 2fc t
M
A X
di;k wk twl t cos
p
2 k1
(8:40)
(8:41)
Similarly, the following operations are performed by the demodulator to extract the Qdecision variable:
stcI twl t sin 2fc t
M
A X
p
di;k c2I twk twl t cos2fc t sin 2fc t
2 k1
M
A X
p
di;k cI tcQ twk twl t sin2fc t sin 2fc t
2 k1
M
M
A X
A X
di;k wk twl t sin p
di;k cI tcQ twk twl t cos
p
2 2 k1
2 2 k1
(8:42)
452
Spread spectrum
(8:43)
M
M
A X
A X
p
di;k cI tcQ twk twl t cos p
di;k wk twl t sin
2 2 k1
2 2 k1
Add (8.42) and (8.43) and include noise. We get the signal and noise of the Q-channel:
sQ t stcI twl t sin 2fc t stcQ twl t cos 2fc t
M
A X
p
di;k wk twl t sin
2 k1
(8:44)
(8:45)
To obtain the I- and Q-decision variables XI;l and XQ;l of the lth channel we integrate
sI t nI t and sQ t nQ t with respect to t, where iTb 5t i 1Tb . We have
ATb
XI;l p di;l cos NI;l
2
(8:46)
ATb
XQ;l p di;l sin NQ;l
2
(8:47)
This phase error can be converted to a dc voltage which can be used by a VCO to correct the
phase error between the received signal and the local carrier. In both scenarios, the I-decision
variable of the lth channel, namely XI ;l, is fed to a threshold detector to make a decision on
which bit had been transmitted. Figure 8.16 illustrates the coherent demodulation of the I-Q
orthogonal covering signal.
453
cos 2fct
cI(t)
w1(t)
Ts
XQ,1
tan1
XI,1
cI(t) w1(t)
cos 2fct
Ts
cQ(t) w1(t)
180
90
cos 2fct
cI(t) w1(t)
VCO
Ts
Threshold
detector
di,l
Figure 8.16
(8:51)
454
Spread spectrum
Ts
...
cI(t) w1(t)
di,l
Ts
Figure 8.17
(8:52)
455
spreading codes are cI t and cQ t, and the carrier frequency is fc , which is assumed to be an
integer multiple of the symbol rate 1=Ts .
A
A
st p cI twk t cos2fc t p cQ twk t sin2fc t ; 0 t Ts (8:53)
2
2
The noncoherent demodulator has M branches corresponding to M distinct transmitted
symbols. Each branch has two quadrature correlators that perform the following operations:
A
stcI twl t cos 2fc t p c2I twk twl t cos2fc t cos 2fc t
2
A
p cI tcQ twk twl t sin2fc t cos 2fc t
2
A
A
p wk twl t cos p cI tcQ twk twl t sin
2 2
2 2
(8:54)
A
stcQ twl t sin 2fc t p cI tcQ twk twl t cos2fc t sin 2fc t
2
A
p c2Q twk twl t sin2fc t sin 2fc t
2
A
A
p cI tcQ twk twl t sin p wk twl t cos
2 2
2 2
(8:55)
The high-frequency terms are ignored because they will be integrated to zero. Add
(8.54) and (8.55) and include noise. We get the signal and noise of the I-correlator of the
lth branch:
A
sI t stcI twl t cos 2fc t stcQ twl t sin 2fc t p wk twl t cos
2
(8:56)
where the noise part is given by the following expression:
nI t ntcI twl t cos 2fc t ntcQ twl t sin 2fc t
(8:57)
Similarly, the following operations are performed by the demodulator to extract the signal
of the Q-correlator of the lth branch:
A
stcI twl t sin 2fc t p c2I twk twl t cos2fc t sin 2fc t
2
A
p cI tcQ twk twl t sin2fc t sin 2fc t
2
A
A
p wk twl t sin p cI tcQ twk twl t cos
2 2
2 2
(8:58)
456
Spread spectrum
A
stcQ twl t cos 2fc t p cI tcQ twk twl t cos2fc t cos 2fc t
2
A 2
p cQ twk twl t sin2fc t cos 2fc t
2
A
A
p cI tcQ twk twl t cos p wk twl t sin
2 2
2 2
(8:59)
Add (8.58) and (8.59) and include noise. We get the signal and noise of the Q-correlator of
the lth branch:
A
sQ t stcI twl t sin 2fc t stcQ twl t cos 2fc t p wk twl t sin
2
(8:60)
where the noise part is given by the following expression:
nQ t ntcI twl t sin 2fc t ntcQ twl t cos 2fc t
(8:61)
To obtain the I- and Q-decision variables XI;k and XQ;k of the kth branch (signal
branch) we integrate sI t nI t and sQ t nQ t with respect to t, where 0 t Ts .
We have
ATs
XI;k p cos NI;k
2
(8:62)
ATs
XQ;k p sin NQ;k
2
(8:63)
N 0 Ts
2
(8:64)
(8:65)
XQ;l NQ;l
(8:66)
The I- and Q-outputs of each branch are squared and added to provide M decision variables
for the maximum detector:
2
2
XQ;l
; l 1; 2; . . . ; M
X 2 XI;l
(8:67)
The bit error probability for an AWGN channel is given in Chapter 7 and for a fading
channel in Chapter 10. Figure 8.18 shows the noncoherent demodulator for a quadrature
DS-CSK signal.
457
Ts
()2
l=1
...
Maximum
detector
Symbol/bit
converter
l=M
cI(t) w1(t)
Ts
()2
Figure 8.18
K
X
k2
K
X
k2
iTb 5t i 1Tb
(8:68)
where k is the kth path delay and k is uniformly distributed in (0, 2) radians. The N-chip
PN function ck(t) has period NTc = Tb. The demodulator is the familiar one shown in
Figure 8.14. The decision variable at the threshold detector input can be calculated as
X
ATb di;1
I N
2
(8:69)
458
Spread spectrum
where the noise variable N is a zero mean Gaussian random variable with the following
variance:
N 0 Tb
4
(8:70)
A
2
Tb X
K
0
di;k cos k c1 t ck t k dt
k2
K
AX
di;k cos k
2 k2
Tb
c1 t ck t k dt
(8:71)
K
AX
di;k Ik cos k
2 k2
where
Ik
Tb
c1 t ck t k dt
(8:72)
If the number of chips per period N of the PN sequences is large, the PN sequences can be
approximated as independent zero mean random binary sequences. The autocorrelation of
the random f1; 1g-sequence is given by
R 1
j j
; j j Tc and zero elsewhere
Tc
(8:73)
We note that binary data has zero mean and unit variance, and that the variable
cos k has zero mean and variance . Therefore, the variance of the MUI variable I is
given by
VarI
K
K
A2 X
A2 X
Vardi;k Varcos k VarIk
VarIk
4 k2
8 k2
(8:74)
We now proceed to evaluate the variance of the variable Ik in (8.72). Using the fact that all
user PN sequences are independent we obtain
Tb T b
VarIk
R2 t dt d
(8:75)
0
1 Tb 2
VarIk
R u du
dv 2
R2 uTb u du
(8:76)
2 Tb
u
0
Substituting (8.73) into (8.76) we have the variance given by
2
Tb =N
Nu 2
T
T2
VarIk 2
1
Tb u du 2 b b 2
Tb
3N 12N
0
(8:77)
459
For large N, which is the case in practice, the variance can be approximated as
VarIk
2Tb2
3N
(8:78)
Substituting (8.78) into (8.74) we obtain the MUI variance necessary for further analysis:
VarI
K 1A2 Tb2
12N
(8:79)
Returning to (8.69) we evaluate the output signal-to-interference ratio SIR0 and the output
signal-to-noise ratio SNR0 , as follows:
SNR0
2
N0
N0
and SIR0
VarI
K1
(8:80)
SIR1
0
1
SNR1
0
K 1 N0
2Eb
3N
1
(8:81a)
Analysis that does not assume a random spreading sequence can be found in [1]. It is
common in practice to approximate the despread input signal-to-interference ratio SIRi and
output SIR0 by assuming the MUI power is distributed uniformly over the spread bandwidth. Thus, the MUI power spectral density is J0 K 1Eb =N and we have
SIRi N =K 1
(8:81b)
SIR0 2N =K 1
(8:81c)
This conservative approach compensates for the non-randomness of the PN sequence and it
is up to the engineer to select which SIR0 to use. We thus have an inequality for SINR0, as
follows:
K 1 N0
2Eb
2N
1
SINR0
K 1 N0 1
2Eb
3N
(8:81d)
(8:82)
3N
2N
2Eb
2Eb
460
Spread spectrum
Bandpass
filter
Modulator
Hopping
local
carrier
Frequency
synthesizer
controlled
by PN sequence
Figure 8.19
Frequency
hop signal
461
FH signal
Bandpass
filter
Demodulator
{0, 1}
Synchronized
hopping local
carrier
Frequency
synthesizer
controlled
by PN sequence
Figure 8.20
Partial-band jamming
Partial-band jamming of a FH signal is the counterpart of pulsed jamming of a DS signal. It
allows the jammer to concentrate all power into a smaller bandwidth than the spread
bandwidth in order to inict greater damage.
(8:83)
Pb eEb =2J0 eWP=2Pj Rb eN SJRi =2
2
2
2
where N W =Rb is the spread factor, SJRi P=Pj is the input signal-to-jamming ratio,
and N SJRi Eb =J0 . The maximum bit error probability can be calculated as follows:
Pb;max
e1
0:368
0:368
;
2
2
N SJRi Eb =J0
(8:84)
Note the similarity between pulse jamming of DS-PSK signal in (8.36) and partial-band
jamming of FH-FSK in (8.84). Both jamming strategies convert the bit error probability as
an exponentially decayed function of N SJRi Eb =J0 into an inversely proportional
function of N SNRi . To illustrate this, consider the case of SJRi = 1/20 and N = 128, then
0:3 and Pb;max 0:058. On the other hand, for a broadband jamming strategy the bit
error probability is Pb 0:5eN SJRi =2 0:02. This shows that partial-band jamming is in
general more effective than broadband jamming. Also a partial-band jammer does not need
the high peak power transmitter required by a pulsed jammer.
462
Spread spectrum
(8:85)
e
e
4
4
(8:86)
Pb
Pb
where N W =Rb is the spread factor, SJRi P=Pj is the input signal-to-jamming ratio and
N SJRi Eb =J0 . The maximum bit error probability can be calculated as follows:
M e1 =2
0:184M
0:184M = log2 M
N SJRi log2 M
Eb =J0
Pb;max
(8:87)
Consider the case of M = 16, SJRi = 1/20, and N = 128, then 0:08 and we have
Pb;max 0:115. On the other hand, for broadband jamming strategy the bit error probability
is Pb 4e4N SJRi =2 105 .
Let us consider the use of adaptive MFSK to improve the anti-jamming capability by
starting with M = 2 and increasing M when partial-jamming is present. Assume that
the jammer achieves optimum jamming for M = 2 with 2=N SJRi . Substituting
into (8.86) for M > 2 we have Pb M e log2 M =2N SJRi M e log2 M =2Eb =J0 . For
SJRi = 1/20, and N = 128, then we have Pb 0:023 (M = 16) and Pb 0:012 (M = 64), as
compared to Pb;max 0:058 for M = 2.
Multi-tone jamming
With some knowledge of the FH signal structure, the jammer may further increase its
effectiveness by randomly placing tones in the bandwidth of a MFSK signal.
463
error is made if the jamming tone is within the FH bandwidth but not in the same noncoherent matched lters bandwidth of the signal tone (because it enhances the detection of
the signal by adding to the signals energy). When Pj =q P, a symbol error occurs with a
probability 0:5 under the conditions described for Pj =q4P. The optimum number of
jamming tones is approximately given by
Pj
(8:88)
q
P
where bc denotes the largest integer less than or equal to Pj =P. Note that 1 q NH , where
the number of hop bins is NH . This value of q is optimum since a larger q results in Pj =q5P
and no error results; a smaller q results in fewer hop bins being jammed but no larger error
probability for any one bin. The probability that any hop bin is jammed is given by
q
Prhop bin jammed
(8:89)
NH
The probability that the hop is not jammed is
Prhop bin not jammed 1
q
NH
(8:90)
Let us assume that when a hop bin is jammed only one jamming tone is present in the
bandwidth of the MFSK signal. The symbol error probability is one minus the probability
that the jamming tone and the signal are in the same noncoherent matched lters bandwidth.
This probability is 1 1=M and the symbol error probability Pe is
8
M 1
q4 N H
>
>
;
>
>
>
< M
M 1 q
q
M 1 q
Pe q
1 q NH
(8:91)
1
;
Pe 0
>
>
M
N
N
M
N
>
H
H
H
>
>
:
q5 1
Pe 0;
where Pe 0 represents the probability of symbol error when jamming is not present. The
approximation in the second equation of (8.91) is valid when the effect of noise is negligible
compared to that of the jamming signal. The bit error probability is given by
Pb
M =2
Pe q
M 1
(8:92)
For numerical illustration consider the case of SJRi P=Pj 1=20 and N W =Rb = 129.
Consider the case when the hop bins are contiguous and the hop bin bandwidth is equal to
the MFSK bandwidth, which is M 1Rs . Hence, the number of hop bins is given as
NH W =M 1Rb = log2 M N log2 M =M
consider M = 2; this yields
1. First
NH 43. The number of jamming tones is q Pj =P b1=SJRi c 20. The resulting
bit error probability is Pb q=2NH 0:23. There is no advantage for M > 2 in multi-tone
jamming. It is not necessary to jam a hop bin with more than one jamming tone since one
jamming tone can do as much damage as many.
464
Spread spectrum
Follower jamming
Another jamming strategy that can potentially nullify the effectiveness of frequency hopping is follower jamming or repeater jamming. This jamming strategy requires the jammer
to be able to receive the hopped signal destined to the receiver either via the antenna main
beam or via its side lobe. The jammer also needs a broadband receiver that can intercept the
hopped signal. If the hop energy exceeds a preset threshold, then the jammer transmits a
jamming signal at the same frequency as the hopped signal. In order to jam a hop, the
jamming signal has to arrive at the receiver within a fraction of the dwell time of the hop
(the duration of the hopped signal). The dwell time is smaller than or equal to the hop time.
If the jamming energy in the fraction of the dwell time exceeds the hop energy, an error could
occur. This procedure requires the sum of the jammers processing time, the propagation
delay between the transmitter and the jammer (intercept link), and the propagation delay
between the jammer and the receiver (jamming link) to be smaller or equal to the fraction of
the hop dwell time plus the propagation delay between the transmitter and the receiver. Thus
the smaller the hop time and hence the dwell time, the closer to the receiver the jammer has
to be in order to be effective. This increases the vulnerability of the jammer. In order to
neutralize the effect of follower jamming, fast frequency hopping (FFH) is necessary. A FFH
system employs multiple hops per bit to reduce the hop time. The demodulation can be done
via hard-decision or soft-decision demodulation. A hard-decision demodulator detects L
hops and outputs a sequence of L bits which is fed to a majority voter. If the number of 1s
exceeds the number of 0s, the detected bit is bit 1; otherwise it is bit 0. If L is even, and there
are the same number of 1s and 0s, choose either bit 1 or bit 0 with probability 0.5. A softdecision binary noncoherent demodulator combines (sums) L hopped decision samples at
the outputs of the I- and Q-channel noncoherent signal processors. The combined I-channel
and the combined Q-channel decision variables are compared by a maximum detector to
determine bit 1 or bit 0. Soft-decision demodulation may perform better than hard-decision
demodulation. Both types of demodulation suffer performance loss for L > 1 compared to
the case of L = 1 in an AWGN channel without any jamming. The combining loss [24] is a
penalty to pay in exchange for an additional ability to neutralize a potent jammer.
465
BPF
Envelope
detector
BPF
Envelope
detector
Select
largest
.
.
.
BPF ND
Figure 8.21
Envelope
detector
(8:93)
Note that Pe (not jammed) is the symbol error probability of MFSK with noise spectral
density N0 =2, and Pe jammed is the symbol error probability of MFSK with noise spectral
density N0 =2 replaced by the jamming and noise power spectral density J0 N0 =2. The
bit error probability is Pb M =2Pe =M 1. Let us assume that the jamming signal is
strong enough to cause the maximum symbol error probability equal to 1 1=M for a
jammed hop. Furthermore, assume that the probability of correctly identifying the frequency
bin is 0.5. Then, the worst-case symbol error probability is M 1=2M , assuming the
effect of noise is negligible compared to jamming. The corresponding worst-case bit error
probability is Pb 1=4:
Let us consider the case of M = 2, and again assume that the effect of noise is negligible
compared to jamming. Using the ratio Eb =J0 W =ND Rb SJRi , the bit error probability of
slow FH-FSK is Pb Pc;h Pb jammed 0:5Pc;h expW =2ND Rb SJRi . For the case
W =ND Rb 2, Pc;h 0:5, and Pb 0:058, the required input signal-to-jamming ratio is
given by SJRi P=Pj 1:47. For the same bit error probability, the partial-band jamming
strategy in Example 8.8 requires SJRi 1=20. Thus the follower noise jamming strategy
may save the jammer 14.7 dB in power.
466
Spread spectrum
Eh =J0 W =ND LRb SJRi , where Eh Eb =L, the hop error probability of fast FH-FSK is
Ph Pc;h Ph jammed 0:5Pc;h expW =2ND LRb SJRi . The fraction of a hop being
jammed is now smaller than that of the case considered in Example 8.11. For numerical
illustration, let W =ND Rb 6, Pc;h 0:5, and SJRi P=Pj 1:47, then we have
Ph 0:058. By using (4.37) with L replacing n, Ph replacing p, and t = 1, we obtain the
bit error probability Pb 0:01 for hard decoding.
Now let us consider the case of soft decoding where L hops are combined for a decision.
Using (7.413) with L replacing d, r = 1/L we obtain the bit error probability Pb 0:004, a
slightly better performance than hard decoding.
8.4 Summary
The main theme of this chapter is the discussion of direct sequence spread spectrum, which is the cornerstone of
commercial and military systems. We explored important techniques for direct sequence CDMA, such as xed
length and variable length orthogonal coverings. We also used IS-95 as an example to show the implementation
of direct sequence spread spectrum in a major commercial system. We also presented various jamming scenarios
applicable to both direct sequence and frequency hop spread spectrum systems. Pulsed (partial time) jamming is
an effective jamming strategy against direct sequence spread spectrum systems, but this strategy also requires a
high peak power transmitter. Partial band jamming is an effective jamming strategy against frequency hop
spread spectrum systems and does not require a high peak power transmitter. Fast frequency hop systems can be
deployed to combat the potent follower (repeater) jammers.
Problems
1.
2.
3.
4.
5.
Consider a DS-PSK signal using the chip pulse shape pt. Express the waveform si t
for a data bit di 2 f0; 1g or di 2 f1; 1g. Express the waveform for an innite number
of bits.
Write the mathematical expression for the autocorrelation of a periodic msequence with squared chip pulse shape. Derive the corresponding power spectral
density.
List the seven states and the outputs of the m-sequence generator hx
1 x x3 .
Draw the generators for the polynomials hx 1 x2 x5 and hx 1 x3 x5 .
The Gold sequences are generated by modulo-2 adding a pair of preferred
m-sequences from two distinct generators of degree n. This results in 2n 1 Gold
sequences (which include the pair of preferred m-sequences) whose crosscorrelation
has three values given by
467
Problems
Rl;n
6.
7.
8.
9.
10.
11.
8
< tn
1
:
tn 2
tn
2n1=2 1; odd n
2n2=2 1; even n; n 6 0 mod 4
Generate the set of Gold sequences from the following preferred pair of m-sequence
generators, hx x3 x 1 and hx x3 x2 1.
Consider three variable orthogonal covering channels 1, 2, and 3 that employ
the Walsh sequences w1 0110; 0110; 0110; 0110, w2 0011; 1100, and
w3 0000. The data bits for the three channels are d1 1; d2;0 1; d2;1 1;
d3;0 0; d3;1 0; d3;2 1; d3;3 1.
(a) Find the transmitted sequences s1 ; s2;0 ; s2;1 ; s3;0 ; s3;1 ; s3;2 ; s3;3 using the mapping
0 ! 1 and 1 ! 1.
(b) Scale the sequence s1 by a factor of two. Find the orthogonal covering sequence s.
(c) Recover the data bits via inner products and threshold detection.
Consider three variable orthogonal covering channels 1, 2, and 3 that employ the
Walsh sequences w1 0000; 1111, w2 0110; 1001, and w3 0101. The data
bits for the three channels are d1 0; d2 1; d3;0 1; d3;1 1.
(a) Find the transmitted sequences s1 ; s2 ; s3;0 ; s3;1 using the mapping
0 ! 1 and 1 ! 1.
(b) Scale the sequence s1 by a factor of two. Find the orthogonal covering sequence s.
(c) Recover the data bits via inner products and threshold detection.
Consider a DS-PSK signal with a received signal power of 1 mW. The tone jamming
signal has 500 mW of power. Assuming the PN spread sequence is an m-sequence and
the output signal-to-jamming ratio is required to be at least 10 dB, nd the necessary
spread factor.
Design a DS-QPSK modulator and demodulator. Carry out the demodulation analysis
to recover the I- and Q-channel bits.
Consider a DS-PSK signal with tone jamming.
(a) The PN sequence employed by the DS-PSK signal is an m-sequence. Find the
output signal-to-jamming ratio given the input signal-to-jamming ratio of20 dB.
(b) Given the output signal-to-noise ratio of 20 dB, nd the output signal-to-noise and
interference ratio.
Consider a DS-PSK signal with tone jamming:
st Adi ct cos 2fc t ; iTb 5t i 1Tb ; and
jt Aj cos2fc f t
where f is the offset frequency from the carrier frequency fc. The spread factor is
N Tb =Tc . Assume coherent demodulation.
(a) Write the decision variable X S J N by identifying the signal component
S, the jamming component J, and the noise component N. Do not try to solve the
integrals.
(b) Consider a situation where the single-tone jamming signal is replaced by a multiM
P
tone jamming signal jt
Aj;k cos2fc t k . Calculate the jamming
k1
component and its variance assuming all M jamming tones have independent k .
468
Spread spectrum
12.
13.
14.
15.
16.
17.
18.
Consider a DS-PSK signal with a spread factor of 128 under pulsed jamming. The
input signal-to-jamming ratio is 14 dB.
(a) What is the worst-case output signal-to-pulse jamming ratio and the corresponding bit error probability?
(b) What would be the power of a broadband jamming signal relative to the peak
power of a pulsed jamming signal to achieve the same bit error probability at the
jammed receiver?
Consider a DS-PSK signal that encounters a pulsed jamming signal with duty cycle
0:1. The average input signal power is 1 mW and the average jamming power is
10 mW. The received bit energy-to-noise density ratio is Eb =N0 7 dB, and the
spread factor is N = 128. Calculate the bit error probability. What is the bit error
probability if the jammer optimizes ? Whatpis
the corresponding pulse duty cycle?
2
Using the tight upper bound Qx5ex =2 = 2 x, nd the optimum pulse duty cycle
and the worst-case bit error probability for pulsed jamming of a DS-PSK signal,
assuming noise is negligible compared to the jamming signal.
The DS-PSK signal in Example 8.7 is encoded with a convolutional code of rate
and constraint length K. The Viterbi algorithm is carried out for each data frame as in
IS-95, and the signal is jammed by a pulsed jammer. Jamming occurs for a complete
data frame, that is, when a data frame is jammed, all bits in that frame are jammed and
when the data frame is not jammed no bits are jammed.
(a) Consider the case K = 5 (Table 7.13). Evaluate the bit error probability of the
coded signal with soft decoding.
(b) Consider the case K = 9 (Table 7.13). Evaluate the bit error probability of the
coded signal with soft decoding. Compare the result with that of (a).
(c) Repeat (a) for hard decoding. Compare the result with that of (a).
(d) Evaluate the bit error probability with Golay code. Compare the result with that
of (a).
Consider a DS-PSK-CDMA system that allows a total of K multiple access users.
(a) The receiver of a user encounters a broadband Gaussian jamming signal with a
power spectral density function J0 =2. The bit energy-to-jamming density ratio is
Eb =J0 12 dB. The bit energy-to-noise density ratio is Eb =N0 14 dB. Find the
minimum spread factor N that can accommodate at least K = 25 users at a bit error
probability of 103 .
(b) Now assume pulsed jamming with duty cycle 0:1, and evaluate the bit error
probability.
Generalize (8.82) for K users with different powers Pi .
Consider a DS-PSK-CDMA system that allows a total of K multiple access users.
(a) The bit energy-to-noise density ratio is Eb =N0 14 dB. Find the minimum
spread factor N that can accommodate at least K = 25 users at a bit error
probability of 103 .
(b) Now consider a convolutional code of rate , constraint length K = 9 (Table 7.13)
and soft decoding. Using the parameters in (a) to determine the coded bit
error probability. Compare the result to that of the uncoded signal with spread
factor 2N.
469
Problems
19.
20.
21.
22.
23.
24.
25.
26.
27.
Provide the pulse shape that can uniformly distribute the MUI power over the spread
bandwidth.
Consider a DS-QPSK signal under Gaussian broadband jamming with the I-channel
jamming variance many times as large as the Q-channel jamming variance.
Assume that the transmitter has the channel side information and is aware of the
jamming strategy. What should the transmitter do to maximize the receiver
performance?
Consider a coherent DS-CSK signal under pulsed jamming with duty cycle . Derive
the bit error probability under the worse-case situation, assuming that the channel
noise is negligible compared to the jamming signal.
Consider a noncoherent DS-CSK signal under pulsed jamming with duty cycle .
Derive the bit error probability under the worse-case situation, assuming that the
channel noise is negligible compared to the jamming signal.
Consider a coherent DS-CSK signal with M = 16 and a symbol rate spread factor
N = 128 under pulsed jamming given the input signal-to-jamming ratio of 8 dB.
Assuming that the channel noise is negligible compared to the jamming signal, what is
the bit error probability? What would be the power of a broadband jamming signal
relative to the peak power of a pulsed jamming signal in order to achieve the same bit
error probability at the jammed receiver?
Consider a noncoherent DS-CSK signal with M = 16 and a symbol rate spread factor
N = 128 under pulsed jamming given the input signal-to-jamming ratio of 8 dB.
Assuming that the channel noise is negligible compared to the jamming signal, what is
the bit error probability? What would be the power of a broadband jamming signal
relative to the peak power of a pulsed jamming signal to achieve the same bit error
probability at the jammed receiver?
Consider a coherent DS-CSK signal with M = 16 that encounters a pulsed jamming
signal with duty cycle 0:1. The average input signal power is 4 mW and the
average jamming power is 10 mW. The received bit energy-to-noise density ratio is
Eb =N0 7 dB, and the symbol rate spread factor is N = 128.
(a) Calculate the bit error probability.
(b) To achieve the same bit error probability as in (a) what would be the required
symbol rate spread factor if the jammer optimizes ?
The coherent DS-CSK signal is used with a convolutional code of rate and
constraint length K = 5 (Table 7.13). The Viterbi algorithm is carried out for each
data frame as in IS-95. Jamming occurs for a complete data frame, that is, when a data
frame is jammed, all bits in that frame are jammed and when the data frame is not
jammed no bits are jammed. Assuming a symbol rate spread factor of 128 and the
average input signal-to-jamming ratio of 8 dB, what is the coded bit error probability
with hard decoding when the jamming duty cycle is 0.1? Assume that the channel
noise is negligible compared to the jamming signal.
Consider a noncoherent DS-CSK-CDMA system that allows a total of K multiple
access users.
(a) The receiver of a user encounters a broadband Gaussian jamming signal with a
power spectral density function J0 =2 uniform over the spread bandwidth. The bit
470
Spread spectrum
28.
29.
30.
31.
32.
33.
34.
35.
energy-to-jamming density ratio is Eb =J0 12 dB. The bit energy-to-noise density ratio is Eb =N0 14 dB. Find the minimum symbol rate spread factor N that
can accommodate at least K = 25 users at a bit error probability of 103 , given
M = 64.
(b) Now assume pulsed jamming with duty cycle = 0.1, using the parameters in
(a) to determine bit error probability.
Consider a noncoherent DS-CSK-CDMA system with M = 64 that allows a total of K
multiple access users.
(a) The bit energy-to-noise density ratio is Eb =N0 14 dB. Find the minimum
symbol rate spread factor N that can accommodate at least K = 25 users at a bit
error probability of 103 .
(b) If the symbol rate spread factor is doubled, how many multiple access users can
the system accommodate for the same bit error probability?
What would be the power of a broadband jamming signal relative to the power of a
partial-band jamming signal to achieve the same bit error probability at a slow
noncoherent FH-FSK receiver? Give the numerical answer for a bit error probability
of 102 .
Consider a slow noncoherent FH-DPSK signal with a spread factor of 128 under
partial-band jamming given the input signal-to-jamming ratio of 14 dB. What are
the optimum and the corresponding bit error probability? What would be the power
of a broadband jamming signal relative to the power of a pulsed jamming signal to
achieve the same bit error probability at the jammed receiver? Assume that channel
the noise is negligible compared to the jamming signal.
Consider a slow noncoherent FH-FSK signal that encounters a partial-band jamming
signal with 0:1. The average input signal power is 1 mW and the average jamming
power is 10 mW. The received bit energy-to-noise density ratio is Eb =N0 10 dB,
and the spread factor is N = 128.
(a) Calculate the bit error probability.
(b) To achieve the same bit error probability what would be the required spread factor
if the jammer optimizes ?
Consider a slow noncoherent FH-MFSK signal that encounters a partial-band jamming signal with 0:1. The average input signal power is 1 mW and the average
jamming power is 10 mW. The received bit energy-to-noise density ratio is
Eb =N0 10 dB, and the spread factor is N = 128.
(a) Calculate the bit error probabilities for M = 16, 64.
(b) To achieve the same bit error probabilities what would be the required spread
factor if the jammer optimizes ?
Compare the two cases of slow noncoherent FH-FSK in Examples 8.8 and 8.10 from
the point of view of jamming power to achieve the same bit error probability at the
jammed receiver.
Derive the bit error probability for slow noncoherent FH-DPSK under multi-tone
jamming. Use the numerical values in Example 8.10 to evaluate the bit error probability.
Derive the bit error probability for slow noncoherent FH-MSK under multi-tone
jamming.
471
Bibliography
36.
37.
38.
39.
40.
Further reading
For advanced study on direct sequence spread spectrum we recommend Lee and Miller
[5] and Viterbi [6]. For an introduction to CDMA 2000 and WCDMA references [7,8] may
be consulted. Detailed discussion of frequency hop spread spectrum systems can be found in
[912]. Sarwate and Pursley [13] provides a comprehensive discussion of PN sequences for
CDMA applications.
Bibliography
1. M. B. Pursley, Introduction to Digital Communications, Upper Saddle River, NJ:
Pearson Prentice Hall, 2005.
2. J. S. Lee, R. H. French, and L. E. Miller, Probability of error analysis of a BFSK
frequency-hopping system with diversity under partial-band interference Part I:
Performance of square-law linear combining soft decision receiver, IEEE Trans.
Commun., Vol. 32, No. 6, pp. 645653, 1984.
3. J. S. Lee, L. E. Miller, and Y. K. Kim, Probability of error analysis of a BFSK frequencyhopping system with diversity under partial-band interference Part II: Performance of
square-law nonlinear combining soft decision receiver, IEEE Trans. Commun., Vol. 32,
No. 6, pp. 12431250, 1984.
472
Spread spectrum
Introduction
Intersymbol interference (ISI) is a phenomenon in which the energy of a symbol spills over
into succeeding symbols causing interference. Tight ltering at the transmitter and receiver
and/or channel distortion can cause the waveform that represents a symbol to spread out into
succeeding symbol periods. In a terrestrial wireless channel, a signal can travel from the
transmitter to the receiver via multiple delayed paths (echoes). If the delays between paths
are large compared to a symbol period, the energy carried by these echoes can cause
signicant ISI to the succeeding symbols. In this chapter, we study the Nyquist criterion
for zero ISI and provide the design of an optimum demodulator that can achieve both
maximum output signal-to-noise ratio and zero ISI. We then study linear equalizers, such as
the zero-forcing linear equalizer (ZF-LE), and the mean-square error linear equalizer (MSELE), and nonlinear equalizers, such as the zero-forcing decision-feedback equalizer (ZFDFE) and the mean-square error decision-feedback equalizer (MSE-DFE). Both nonlinear
equalizers perform better than their linear counterparts. We then study the maximum likelihood sequence detection (MLSD), which is the optimum nonlinear equalization technique.
Finally, the fractionally spaced equalizer is introduced to solve the timing error problem in
the previous equalizers.
474
p(t)
p(t)
Figure 9.1
2T
t
T
3T
Matched filter
h(t) = p(T t)
3T
0
Figure 9.3
p0(t)
RC-ltering of a squared pulse. The output pulse is p0 t 1 et=RC for 05t 5T and
p0 t 1 eT=RC etT=RC for t 4T.
Figure 9.2
P0(t)
2T
P2
i0
P2
2T
3T
pt iT .
Matched filter
h(t) = p(T t)
i0 p0 t
t
0
2T
3T
4T
iT .
channel can be modeled as an RC-lter, then p0 t would appear at the input of the matched
lter in Figure 7.15. Obviously, the pulse pt is the complex envelope of a PSK bit and the
pulse p0 t is the distorted complex envelope of a PSK bit after going through an RCltering channel.
Let us now consider the output of a matched lter designed for the squared pulse and pt
and p0 t at its input, respectively, as illustrated in Figures 9.2 and 9.3 for three consecutive
bits 1, 0, 1.
We observe that the squared pulse shape presents no post-ltering ISI at the sampling time
t = T, 2T, 3T. This happens because the output of the matched lter is a series of three
triangular pulses whose positive or negative peak occurs right at the zero amplitude of other
pulses. On the other hand, post-ltering ISI is clearly present for the series of RC-ltering
pulses at the sampling time t = T, 2T, 3T. At the sampling time t = 3T, ISI due to p0 t is
constructive but ISI due to p0 t T is more destructive, so the net ISI effect is destructive.
The same ISI effect occurs at t = 2T. The ISI effect on a particular bit can be either
constructive or destructive, but the net effect is always destructive for a long series of bits
or symbols as far as the bit or symbol error probability is concerned.
We note that the pre-ltering ISI is determined by the channel ltering. On the other hand,
the post-ltering ISI is determined by both channel ltering and matched ltering. It is the
latter that determines the performance degradation of a digitally modulated signal. Without
475
confusion, we simply use the word ISI to describe both pre-ltering ISI and post-ltering ISI
unless otherwise stated. Note that pre-ltering ISI is irreversible since a matched lter
designed for the RC-ltering pulse would not eliminate post-ltering ISI. In the following
discussion, we seek to remedy ISI that arises from channel ltering by presenting the
Nyquist criterion for zero ISI. The criterion provides the necessary and sufcient condition
for a pulse shape to have zero ISI at the time a decision sample is taken. It makes no mention
whether the sample is taken at maximum signal-to-noise ratio (SNR) or not. It is clear that a
decision sample from an optimum demodulator designed for a bandlimited channel should
meet both requirements, namely, zero ISI and maximum SNR. This is a subject that will be
studied later in this section.
(9:1)
where k is an integer, if and only if the Fourier transform P f of pt satises the following
condition:
1
1 X
m
P f
1
T m1
T
(9:2)
1
1
P f e j2ft df
(9:3)
1
1
P f e j2fkT df
m1 2m1=2T
1 1=2T
X
m1
1 2m1=2T
X
1=2T
m j2fkT
P f
df
e
T
1=2T
Pf e j2fkT df
(9:4)
m j2fkT
P f
df
e
T
1=2T m1
1
X
1=2T
e j2fkT df
1=2T
sin k
1;
k0
0;
k 6 0
(9:5)
476
To prove that (9.1) implies (9.2), let us express the periodic function in (9.2), whose period is
1/T, as a Fourier series:
1
1
X
X
m
P f
ck e j2kfT
(9:6)
T
m1
k1
where the coefcient ck is given by
ck T
1=2T
m j2fkT
P f
df
e
T
1=2T m1
1
X
(9:7)
(9:8)
Substituting (9.8) into (9.6) yields (9.2) and thus concludes the proof.
Note that when P f is bandlimited to B, then the sampling rate 1/T must be less than or
equal to 2B. When 1=T 42B, there is no pulse shape that can produce zero ISI. Also note
that the left-hand side of (9.2) is simply the Fourier transform of the sampled signal
P
P
pt t mT pt t mT (by virtue of symmetry).
m
sint=T
t=T
(9:9)
477
1
0.8
p(t)
0.6
0.4
0.2
P( f )
0
0.2
0.4
4
Figure 9.4
fT
3
0
t/T
1
2
1
2
The ideal pulse for zero ISI and minimum bandwidth spectrum.
p t
sint=T cost=T
t=T 1 42 t 2 =T 2
(9:11)
The parameter is commonly referred to as the roll-off factor and takes a value in the range
05 1. Note that when = 0, the raised-cosine pulse becomes the sin x/x pulse. The
Fourier transform of the raised-cosine pulse is given as
8
0 j f j 1
>
T;
2T
>
>
>
>
>
<T
1
1
T
1
2T j f j 2T
(9:12)
1 cos
;
P f
jfj
>
2
2T
>
>
>
>
>
j f j4 1
: 0;
2T
Figure 9.5 shows both pt and P f for various values of . The energy spectrum of pt
looks like the frequency response of a lowpass Butterworth lter with roll-off dictated by .
Therefore, it is easy to generate pt in practice.
When the raised-cosine pulse shape is employed for data bits or symbols with rate 1/T, the
positive or negative peak of each bit or symbol coincides with the zero amplitude of other
bits or symbols. Thus, there is no intersymbol interference.
478
1
0.5
0.8
0.8
P( f )
p(t)
0.6
0.4
0.6
0.4
0.2
1
0.2
0
0.5
0.2
4
Figure 9.5
0
t/T
0
1
0
fT
0.2
0.4
0.6
0.8
Impulse
generator
1
Transmitter
filter
Bandpass
channel
cos2fc t
Receiver
filter
Threshold
detector
{0, 1}
cos2fc t
Figure 9.6
PSK modulator and demodulator for a bandlimited bandpass channel with a bandwidth of 2B.
{0, 1}
Impulse
generator
1
Transmitter
filter
Equivalent
lowpass
channel
HT ( f )
H (f)
s(t)
+ n(t)
Receiver
filter
s0(t)
t=T
+ n0(t)
Threshold
detector
{0, 1}
HR ( f )
f B
0
Figure 9.7.
Equivalent lowpass model for a bandlimited PSK signal. The channel is bandlimited to B.
a case study but our analysis also applies to other modulation techniques. Figure 9.6 shows a
generic PSK modulator and demodulator for a linear bandlimited bandpass channel.
It is more convenient to analyze the model in Figure 9.6 using its equivalent lowpass
model. This model deals only with the complex envelope of the PSK signal and the
corresponding equivalent lowpass channel of the bandlimited bandpass channel.
Figure 9.7 illustrates the equivalent lowpass model. Let HT f , HR f , and H f denote
the transfer functions of the transmitter lter, receiver lter, and linear equivalent lowpass
channel, respectively. Furthermore, let st and nt denote the complex envelope of the
received PSK signal and noise, respectively. For PSK, st is just an antipodal baseband
signal. Furthermore, let N f be the power spectral density of the equivalent lowpass noise
nt , and for AWGN we take N f N0 W=Hz by convention.
479
Obviously, the bandpass noise that accompanies the PSK signal at the input of the
receiver in Figure 9.6 has a power spectral density N f fc =2 N f fc =2, which is
N0 =2 for AWGN. The output of the receiver lter is s0 t n0 t , where s0 t is the output
signal and n0 t is the output noise. At the sampling time t = T, we obtain the decision sample
s0 T n0 T . We want to design HT f and HR f to achieve maximum SNR0 at the
sampling time and also have zero ISI. We begin by noting that the Fourier transform S0 f
of s0 t that represents a bit is
S0 f S f HR f HT f H f HR f
(9:13)
1
1
S0 f e j2ft df
1
1
HT f H f HR f e j2ft df
(9:14)
(9:15)
1
It is obvious that the output signal power is s20 T , and the output noise power is given by
1
n20 T
N f jHR f j2 df
(9:16)
1
The output signal-to-noise ratio at the output of the receiver lter at the sampling time
t = T is
SNR0
js0 T j2
n20 T
1
2
j2fT
df
1 HT f H f HR f e
1
2
1 N f jHR f j df
1 HT f H f e j2fT p
2
p
N f HR f df
1
N f
1
2
1 N f jHR f j df
(9:17)
Applying the CauchySchwarz inequality to the right-hand side of (9.17), we obtain the
upper bound for SNR0 as follows:
1
SNR0
1
1
jHT f j2 jH f j2
df 1 N f jHR f j2 df
N f
1
2
1 N f jHR f j df
jHT f j2 jH f j2
df
N f
1
(9:18)
480
(9:19)
HT f H f ej2fT
N f
(9:20)
In practice, lters are designed to have linear phase and, furthermore, we assume that the
channel is linear; therefore, we can express HT f , HR f , and H f as follows:
HT f jHT f jej2ftT
(9:21)
HR f jHR f jej2ftR
(9:22)
(
H f
jH f jej2ft0 ;
0;
jfj B
j f j 4B
(9:23)
where tT, tR, and t0 are the time delays of the physically realizable transmitter and receiver
lters and channel delay, respectively.
Substituting (9.21)(9.23) into (9.20) yields, with jj expfj2f tT tR t0 T g,
HR f jj
jHT f j jH f j
N f
(9:24)
jfj B
(9:25)
or, equivalently,
jHT f j jHR f j jH f j P f ;
jfj B
(9:26)
481
jHT f j2
1 N f P f
;
jj jH f j2
jHR f j2 jj
jfj B
P f
;
N f
(9:27)
jfj B
(9:28)
p
P f ;
jfj B
(9:30)
Desired channel
{0, 1}
Impulse
generator
1
P( f )
1
H( f )
Channel
H( f )
t=T
P( f )
Threshold
detector
{0, 1}
1
Noise
0
Figure 9.8
An ideal channel with zero ISI is created by equalization in the transmitter, which results in an optimum demodulator
that achieves maximum output SNR0 at the sampling time.
482
jHT f j2 jH f j2
df
N0
1
1 1
1 1 2
2
SNR0;max
s t dt
jS f j df
N0 1
N0 1
(9:32)
(9:33)
2Eb
N0
where Eb is the bit energy. Note that st is the complex envelope of the PSK signal,
therefore the energy of st is 2Eb . The bit error probability is simply
r
p
2Eb
Pb Q SNR0;max Q
(9:34)
N0
which is the same as the bit error probability for PSK with squared pulse shape. This
result comes as no surprise, since the equalizer 1=jH f j in the transmitter totally compensates for the channel distortion and eliminates pre-ltering ISI before the noise enters the
receiver.
483
In a noisy bandlimited linear channel, the locations of the channel equalizer are not
equivalent. When an equalizer is placed in the receiver, it can remove post-ltering ISI,
but only at the expense of the output signal-to-noise ratio. This happens because the
equalizer destroys the optimality of the matched lter, since the cascade of the matched
lter and equalizer, whatever the order, is no longer a matched lter. Therefore, the receiver
equalization cannot achieve optimum demodulation because it is an ad hoc method which
tries to achieve a performance as close to that of an optimum demodulator as it can.
Nevertheless, this type of equalization is more than adequate and widely used in practical
applications.
Receiver equalization of the channel distortion is based on the knowledge of the channel
contained in the received signal a training sequence, for example. Any signal processor
must process the received signal without loss of information and no signal processor can do
it better than a matched lter matched to the received signal pulse. The decision sample at
the matched lter output has the maximum signal-to-noise ratio, but it contains post-ltering
ISI. Also, the noise samples are dependent since the noise power spectral density at the
output of the matched lter is no longer white. To eliminate ISI, a linear equalizer is
employed to force the post-ltering ISI to zero at the sampling time. Unfortunately, the
equalizer also enhances noise and, therefore, destroys the optimality of the matched lter.
Figure 9.9 illustrates the concept of linear equalization at the receiver using a lowpass
equivalent model. It is obvious that the output of the equalizer must meet the Nyquist
criterion in (9.1) or equivalently (9.2) to have zero ISI. This requirement implies that the
frequency response of the equalizer must be the inverse of its input signal spectrum.
Let us consider a transmitted pulse gt with Fourier transform G f passing through a
channel with impulse response ht and corresponding transfer function H f . The received
signal is st gt ht with the Fourier transform S f G f H f . A matched lter
with impulse response s t and corresponding frequency response S f is employed.
Note that the matched lter is non-causal for simplicity. A delay can always be introduced to
make the matched lter causal without altering the result. The signal s0 t at the matched
lter output is sampled with rate 1=T 2B, where B is the bandwidth of the bandlimited
P
P
channel. The sampled signal xt s0 t k t kT s0 t k t kT is passed
{0, 1}
Impulse
generator
Channel
h(t), H( f )
g(t), G( f )
k d(k)(t kT)
s(t), S( f )
AWGN
n(t)
d(k) {1, 1}
t = kT
Matched
filter
s*(t), S*(f)
Figure 9.9
s0(t)
x (t)
S0( f )
X (f)
Equalizer
L( f )
Threshold
detector
Continuous-time lowpass equivalent channel model with linear equalizer at the receiver.
{0, 1}
484
through an equalizer to eliminate ISI. Let S0 f be the Fourier transform of the matched
lter output signal s0 t , then
S0 f S f S f jS f j2
(9:35)
Xf
S0 f
S0 f
T k1
T
T k1
T
(9:36)
2
2
1
1
X
1 X
k
1
k
S f
S f
T
T
T
T
k1
k1
Note that X f is periodic with frequency 1=T. Furthermore, the sampling rate must be in
the range B 1=T 2B for Nyquist criterion of zero ISI to hold. X f is called the folded
spectrum of the received signal st .
In order for the sampled signal xt to have zero ISI at the sampling time t = 0, its spectrum
X f must be equal to unity as dictated by the Nyquist criterion (9.2). This can only happen
if and only if s0 t is a Nyquist pulse. Since this is not possible, the only way to satisfy (9.2) is
to force the sampled signal xt through an equalizer whose transfer function L f is equal to
the inverse of X f , that is,
L f
1
Xf
(9:37)
Now let us assume that the power spectral density of the equivalent lowpass AWGN nt at
the matched lter input is N0 . The matched lter output noise n0 t has the power spectral
P
density N0 jS f j2 N0 jS f j2. Let ns t n0 t k t kT be the sampled noise with
the following autocorrelation:
XX
t kT t nT dt
Rns ; n k En0 n0 t
Rn0
Rn0
XX
k
XX
t kT t nT dt
(9:38)
n kT
We are interested in the autocorrelation at the sampling time index k = 0, that is,
P
Rns Rn0 nT , whose Fourier transform is the power spectral density of
n
S f
F f
T k1
T k1
T
T
N0 X f
Let F0 f be the power spectral density of the equalizer output noise, then we have
(9:39)
485
F0 f F f jL f j2
N0
Xf
1
T
N0
1
P
S f k 2
T
(9:40)
k1
It is apparent that if the spectrum X f of the sampled signal xt in (9.36) contains nulls
or small values, then the output noise spectrum could become innite or very large.
Consequently, the variance of output noise tends to innity or very large. Thus, the zeroforcing linear equalizer tries to obtain zero ISI at the expense of the output signal-to-noise
ratio. Let us evaluate the noise power at the sampling time index k = 0. Denote R0 the
autocorrelation of the equalizer output noise, which is the inverse Fourier transform of
F0 f . Note that the frequency response L f 1=X f of the equalizer is periodic with
period 1=T , therefore it can be expressed as a Fourier series. We have the following
relationship:
1
X
F0 f N0 L f N0
k T
1=2T
R0 N0
1=2T
1
X
k e j2kfT ;
k1
L f e
j2kfT
df T
1=2T
1 j2kfT
df
e
X
f
1=2T
(9:41)
k kT
k1
Using (9.40), we obtain the noise power at the sampling time index k = 0 as follows:
R0 0 N0 0 N0 T
1=2T
1=2T
L f df N0 T
1=2T
1
df
X
f
1=2T
(9:42)
P
Let yt pt k t kT be the sampled signal at the equalizer output and Y f its
Fourier transform. Since Y f L f X f 1, the pulse pt obviously satises the
Nyquist criterion (9.2), and hence at the sampling time index k = 0 we have p0 1 per
(9.1). Thus, the output signal power is p2 0 1. Alternatively, we can view yt as the
inverse Fourier transform of Y f , and since Y f 1 we have yt t . Hence, at the
sampling index k = 0, y0 is simply the weight of t , which is unity, and consequently
the output signal power is y2 0 p2 0 1. Therefore, the output signal-to-noise ratio is
given by
SNR0; ZF -LE
1
1
1
2 N0 0 N0 mA X 1
(9:43)
where mA X 1 0 denotes the arithmetic mean of 1=X f per (9.41). Note that SNR0
cannot exceed the maximum output signal-to-noise ratio SNR0; max of the optimum lters
given in (9.33), which is 2Eb =N0 , that is, 1=N0 mA X 1 2Eb =N0 .
486
sampled version xt satises (9.1). The Fourier transform X f of xt also satises (9.2),
that is, X f 1. Therefore, no equalizer is needed in this case. Furthermore,
x 0 1
1
1 2
2
2
via (9.1) and the noise power is simply N0 1 s t dt N0 1 jS f j df N0
since the energy 2Eb of the Nyquist pulse is unity. Hence, SNR0 x2 0=N0 1=N0 .
(9:44)
which also implies X f jF f j2 jF f j2 . This suggests that the zero-forcing equalizer whose frequency response L f 1=X f can be decomposed into a cascade of two
lters, one with response 1=F f , and the other response 1=F f . This concept is shown in
Figure 9.10. Let us evaluate the noise spectrum at the output of the lter 1=F f . The
power spectral density of noise at the input of this lter is given in (9.39). Therefore, the
output noise power spectral density is F f = jF f j2 N0 X f =X f N0 , and this
implies that the output noise is white.
The error probability for ZF-LE can be evaluated via the Q-function. For example, for
PSK modulation
or QPSK modulation, the bit error probability is simply
p
Pb Q SNR0; ZF -LE , where SNR0; ZF -LE is given in (9.43).
Up to this point, we have studied the concept of zero-forcing linear equalization (ZF-LE)
at the receiver with the criterion of minimizing the effect of noise at the ZF-LE output
subject to the constraint of zero ISI at the sampling time. As far as linear equalization is
concerned, a demodulator that consists of a matched lter followed by a symbol-rate
sampler and a ZF-LE provides the optimum performance in terms of minimizing the error
probability. In practice, the use of a continuous-time matched lter, which is not easily
adapted, poses some implementation problems if the channel response is not known with
t = kT
s(t) + n(t)
Matched
filter
s*(t), S*(f )
s0(t)
x(t)
S0( f )
X( f )
Noisewhitening
filter
1
F *( f )
Figure 9.10
Causal
1
F( f )
Threshold
detector
{0, 1}
Zero-forcing
linear equalizer (ZF-LE)
The whitened matched lter, which consists of a matched lter, a symbol-rate sampler, and a noise-whitening lter.
487
(9:45)
1
1
1 X
k
1 X
k
V0 f
V0 f
T k1
T
T k1
T
1
1
1 X
k
k
1 X
k
k
HFE f
HFE f
S f
S f
T k1
T
T
T k1
T
T
(9:46)
To satisfy the Nyquist criterion (9.2) for zero ISI, the ZF-LE must have a frequency response
equal to 1=Q f . This yields an output signal q0 t with a corresponding Fourier transform
Q0 f Q f =Q f 1. Thus, at the sampling time t = 0, the output sample is unity via
(9.1) and the sample power is also unity.
To evaluate the effect of noise, we again assume white noise nt with power spectral
density N0. The front-end lter output noise has a power spectral density equal to
N0 jHFE f j2 . Therefore, the power spectral density of the sampled noise is given by
{0, 1}
Impulse
generator
g(t), G(f)
Channel
h(t), H( f )
s(t)
S( f )
AWGN
n(t)
t = kT
Front-end
filter
HFE( f )
Figure 9.11
V0(t)
q(t)
V0( f )
Q( f )
Sub-optimum
ZF-LE
1
Q( f )
Threshold
detector
{0, 1}
488
2
2
1
1
X
N0 X
k
N
k
0
HFE f
C f
HFE f
T k1
T k1
T
T
(9:47)
Consequently, the ZF-LE output noise possesses the following power spectral density:
C0 f C f
1
P
HFE f k 2
T
N0
k1
jQ f j2
1
P
2
1
k
k
S f T HFE f T
T
(9:48)
k1
(9:49)
N0
k1
C0 f
1
T
1
1
P
S f k 2 P HFE f k 2
T
T
k1
1
T
k1
(9:50)
N0
1
P
S f k 2
T
k1
Note that equality in (9.49) occurs if and only if bk ak where is an arbitrary constant.
By comparing C0 f to the output noise power spectral density F0 f of the ZF-LE in
(9.40), we conclude that C0 f F0 f . Consequently, the output SNR0 for the suboptimum
ZF-LE is less than or equal to that of the ZF-LE given in (9.43). Note that C0 f F0 f if
and only if HFE f S f , that is, the front-end lter is a matched lter!
So far, we have studied ZF-LE design via the analysis of the continuous-time sampled
signal and its Fourier transform. Alternatively, we can also achieve the same goal by
considering the discrete-time sequence of samples and its corresponding discrete-time
Fourier transform. Let us consider the sampled signal xt at the input of the equalizer in
P
Figure 9.10. We note that xt k s0 kT t kT , therefore the samples
are
0
xk s0 kT . The sequence xk and its discrete-time Fourier transform X e j2f are
related by the following pair of equations:
xk
X e
1=2
0
0
X e j2f e j2kf df 0
1=2
j2f 0
1
X
k1
(9:51)
xke
j2kf 0
489
1
X
xkej2kfT
(9:52)
k1
0
Comparing (9.51) and (9.52), we see that X e j2f and X f are related by
0
f
j2f 0
X e
X
; j f 0 j 1=2
T
0
Consequently, we obtain X e j2f in terms of S f in (9.36) as follows:
2
2
1 0
1 0
1 X
X
0
S f k 1
S f k
X e j2f
T k1
T
T k1
T
(9:53)
(9:54)
To deal with lter transfer functions, we employ the z-transform of the sequence xk, which
is dened as
X z
1
X
xkzk
(9:55)
k1
0
The z-transform reduces to the discrete-time Fourier transform when z e j2f , that is, on
the unit circle in the complex z-plane. The sequence xk can also be determined from X z
via the inverse z-transform using contour integration as follows:
1
X zzk1 dz
(9:56)
xk
2j C
0
Since the discrete-time Fourier transform X e j2f is nonnegative real, its corresponding
z-transform X z has a spectral factorization as follows:
X z F zF
1
z
(9:57)
490
t = kT
s(t) + n(t)
Matched
filter
s*(t), S*( f )
s0(t)
x(k)
S0( f )
X(z)
r(k) + w(k)
y(k) + (k)
Causal
ZF-LE
1
F(z)
Threshold
detector
Noisewhitening
filter
1
F*(1/z*)
Figure 9.12
ZF-LE
Composite
channel
X(z) =
F(z)F*(1/z*)
Noisewhitening
filter
1
F*(1/z*)
x(k)
Non-white
noise n0(k)
Figure 9.13
{0, 1}
Causall
ZF-LE
1
F(z)
y(k) + (k)
Threshold
detector
dk
ZF-LE
Discrete-time channel
for the
ZF-LE sampled at the symbol rate. The equalizer input noise wk has the
j2f
model
0 2
j2f 0
spectrum e
=F e
, where f is given in (9.39).
y(k) + (k)
d(k)
Synthetic
channel
F(z)
r(k)
Causal
ZF-LE
1
C(z) =
F(z)
Threshold
detector
d(k)
AWGN
w(k)
Figure 9.14
Simplied discrete-time channel model for the ZF-LE sampled at symbol rate.
channel model is illustrated by Figure 9.13. The sequence xk is the sampled data at the
output of the matched lter in Figure 9.12, and the noise sequence n0 k is the non-white
noise sample of the matched lter output noise n0 t . The output of the noise-whitening lter
1=F 1=z is the signal sequence rk plus the white noise sequence wk whose spectrum
is N0 . The simplied discrete-time channel model for the optimum ZF-LE is shown in
Figure 9.14.
Let us denote the transfer function 1/F(z) of the equalizer in Figures 9.13 and 9.14 as
C z
1
X
1
ckzk
F z k0
(9:58)
491
where c(k) represents the lter coefcients (often called tap weights). Given the input signal
sequence rk, the output signal sequence y(k) is given by
yk rk ck
1
X
rick i
1
X
i0
cirk i
(9:59)
i0
The output noise sequence k at the equalization output has the following power spectral
density and variance:
N0
N0
0
F0 e j2f
(9:60)
2
X e j2f 0
jF e j2f 0 j
2
1=2
1=2
1=2
0
F0 e j2f df 0 N0
1=2
df 0
N0 mA X 1
0
j2f
1=2 X e
df 0
mA X
j2f 0
1=2 X e
0
where mA X 1 is the arithmetic mean of X 1 e j2f .
1
(9:61)
The output s0 t of the matched lter in Figure 9.12 is simply the autocorrelation of the pulse
1
s s t d
s0 t s0 t
(9:63)
1
ajtj
2Ee
Thus, the sequence xk s0 kT is simply given by
xk 2EeaT jk j 2Ebjk j
(9:64)
where b eaT . Using (9.55) the z-transform of the sequence xk can be evaluated as
follows:
X z 2E
b2 1
z
b
z bz b1
(9:65)
2
z b b1 z1
X z 2E b 1
z
2E b 1
(9:66)
492
Consequently,
1
X e
j2f 0
1
b j2f 0
0
2
e
b b1 ej2f
2E b 1
(9:67)
Integrating F0 f 0 in (9.61) using (9.67) for j f 0 j 1=2 yields the output noise variance
2
1=2
N0
N 0 1 b2
0
0 df
j2f
2E 1 b2
1=2 X e
(9:68)
1
2E 1 b2
2 N0 1 b 2
(9:69)
Note that X z possesses poles at z b and z 1=b. Therefore, as b eaT ! 1, that is,
poles on the unit circle, SNR0 ! 0, and the equalizer fails. Compared to the matched lter,
the ZF-LE suffers a reduction in SNR0 by a factor 1 b2 =1 b2 .
In general, the linear equalizer is an innite impulse response (IIR) lter. It can be
approximated by a transversal lter, which is a classic name for a nite impulse response
(FIR) lter with 2N + 1 coefcients or tap weights c(i). Specically, we approximate the
output of y(k) of a linear equalizer with input r(k) via (9.59) as follows:
yk
N
X
cirk i
(9:70)
iN
1;
k0
0;
k 1; 2; . . . ; N
(9:71)
r0
r1
..
.
6
6
6
6
6
6 rN
6
6
..
4
.
r2N
r1
r0
..
.
rN 1
..
.
r2N 1
3 2 3
32
cN
r2N
0
7
6
r2N 1 7
07
76 cN 1 7 6
7
7 6
76 ..
..
6 .. 7
7
6
7
.
.
.
6
76 7
76
7 617
6 c0
rN 7
7 6 .7
76
7 4 .7
6
7
..
.
5
54 ..
.5
.
0
cN
r0
(9:72)
493
rk
z1
c(N)
z1
z1
c(N + 1)
c(N 1)
c(N)
y(k)
Adjustment of coefficients
Figure 9.15
or in compact form as
Rc y
(9:73)
(9:74)
Figure 9.15 illustrates the transversal lter with 2N + 1 coefcients. If the input is causal, the
number of coefcients reduces to N + 1.
Summary
The ZF-LE equalizes the folded spectrum of the received signal via a lter with an inverse
frequency response. It forces the ISI to be zero at the sampling time at the expense of the
output signal-to-noise ratio. When the folded spectrum has nulls, the equalizer cannot
equalize it since the output signal-to-noise ratio can go to zero. The output of the matched
lter contains both anti-causal ISI (which affects preceding pulses) and causal ISI (which
affects succeeding pulses) since it is symmetric at the sampling time. Nevertheless, the
output of the whitened matched lter is causal because the whitened matched lter transfer
function is F z, a minimum-phase function. In other words, the noise-whitening lter
1=F 1=z totally eliminates anti-causal ISI (Figure 9.16). For the suboptimum ZF-LE,
there is no spectral factorization for Q f in (9.46) since it is not positive-real. The
discrete-time channel model in Figure 9.14 still holds with F z replaced by Q z. In this
case, Q z is not minimum-phase and the input to the ZF-LE 1=Q z is not guaranteed to be
causal.
494
r(k)
x(k)
k
3
Anti-causal ISI
(k < 0)
Figure 9.16
Causal ISI
(k > 0)
The output xk of the sampled matched lter contains both anti-causal ISI and causal ISI. The output rk of the whitened
matched lter contains only causal ISI, which is eliminated by the equalizer 1/F(z).
g(t), G( f )
Channel
h(t), H( f )
Matched
filter
s*(t), S *( f )
AWGN
n(t)
Equalizer
c(k), C(z)
Figure 9.17
y(t) + (t)
Threshold
detector
{0, 1}
Discrete-time channel model for the MSE-LE sampled at the symbol rate.
(9:75)
Denote k the equalizer output noise. We wish to minimize the mean-square error (MSE)
between yk k and the transmitted data dk. Let the error ek be dened as
495
9.4 MSE-LE
ek yk dk k
(9:76)
n
o
n
o
2e E jekj2 E jyk dk kj2
(9:77)
(9:78)
Since MSE is simply the variance of ek, we can evaluate the power spectral density Se f 0
of ek and integrate it to get MSE. Assume that data and AWGN are uncorrelated, we can
evaluate Se f 0 directly from (9.78) as follows (for the sake of simplicity, we drop the
argument f 0 in the lter response and power spectral densities):
Se Sd jXC 1j2 S
(9:79)
where S is the power spectral density of the equalizer output noise. Note that the power
spectral density of the equalizer input noise is given in (9.39) as N0 X , where N0 is the power
spectral density of the input AWGN. Thus,
S N0 X jC j2
(9:80)
Substituting (9.80) into (9.79) yields the power spectral density of the equalizer output
error as
Se Sd jXC 1j2 N0 X jC j2
(9:81)
Note that Se is positive-real, therefore to minimize the MSE, which is the integral of Se , we
only need to minimize Se . We observe that when C 1=X , Se reduces to F0 in (9.40) and 2e
in (9.77) reduces to 2 in (9.42), which is the case of the ZF-LE for which yk dk. Now,
we wish to have some residual ISI at the equalizer output but minimize it together with noise
and use the fact that the positive-real folded spectrum X can be factored into the following
product:
X FF j F j2
(9:82)
Returning to (9.81) and completing the square term on the right-hand side, we obtain by
using (9.82)
Se Sd X 2 jC j2 1 XC X C N0 X jC j2
Sd X N0 X jC j2 Sd XC X C 1
(9:83)
Si j F j2 jC j2 Sd j F j2 C Sd j F j2 C Sd
where
Si Sd X N0 Sd j F j2 N0
Continuing, we get
(9:84)
496
Se Si j F j2 jC j2 Sd j F j2 C Sd j F j2 C Sd2 Si1 j F j2 Sd Sd2 Si1 j F j2
2
Si FC Sd Si1 F Sd 1 Sd Si1 j F j2
Note that 1 Sd Si1 j F j2 N0 Si1 via (9.84), therefore
2
Se Si FC Sd Si1 F Sd N0 Si1
(9:85)
(9:86)
We wish to minimize the rst term on the right-hand side of (9.86) that contains the spectra
of both ISI and noise at the equalizer input. This yields FC Sd Si1 F 0, and hence the
following frequency response for the equalizer is obtained:
C Sd Si1
Sd
Sd
2
S
X
N0
Sd j F j N0
d
(9:87)
N0 Sd
2
Sd j F j N0
N0 Sd
Sd X N 0
(9:88)
For white data we can set Sd f 1 to obtain the following discrete-time transfer function
for the equalizer:
C z
1
F zF 1=z N0
(9:89)
0
For white data the minimum MSE 2e in (9.88) is simply the integral of Se e j2f from 1/2
to 1/2:
2e
mA Se
1=2
1=2
0
Se e j2f df 0
(9:90)
Again, we use the notation mA Se to denote the arithmetic mean of Se . Comparing 2e with
2 of the ZF-LE in (9.61), we observe that Se N0 =X N0 5F0 N0 =X due to the extra
term N0 in the denominator of Se . Therefore, we always have 2e 2 . Furthermore, 2e does
not approach innity when X has spectral nulls. Thus, the MSE-LE avoids the innite noise
enlargement.
Figure 9.18 shows the discrete-time channel model for the MSE-LE, which incorporates a
noise-whitening lter. Figure 9.19 shows an equivalent but simplied discrete-time channel
model with a discrete-time pre-equalizing lter F 1=z . This matched lter does not exist
for the ZF-LE in Figure 9.14 since the ZF-LE already inverts the channel.
497
9.4 MSE-LE
d(k) {1, 1}
Composite
channel
X(z) =
F(z)F *(1/z*)
Noisewhitening
filter
1
F*(1/z*)
x(k)
d(k)
Threshold
detector
F*(1/z*)
F(z)F *(1/z*) + N0
MSE-LE
Non-white
noise n0(k)
Figure 9.18
Equalizer
Discrete-time channel model for MSE-LE with a noise-whitening lter sampled at the symbol rate.
Synthetic
channel
F(z)
r(k)
u(k)
AWGN
w(k)
Figure 9.19
Discretetime
matched
filter
F (1/z)
MSE-LE
1
F(z)F(1/z) + N0
Equalizer, L(z) =
Threshold
detector
d ( k )
F (1/z)
F(z)F(1/z) + N0
Simplied discrete-time channel model showing a pre-equalizing discrete-time matched lter sampled at the
symbol rate.
ek n
yk k
o dk vk dk, and we wish to minimize the mean-square error
2e E jekj2 . Let F z be expressed as
F z
1
X
f kzk
(9:91)
k1
1
X
f mdk m wk
(9:92)
m1
1
X
kzk
(9:93)
k1
n
o
The orthogonality principle of mean-square estimation states that 2e E jekj2
is minimum when the error e(k) is orthogonal to the input u k j; 15j51. That is,
Efeku k jg 0
Note that the equalizer output v(k) is equal to uk k, thus
(9:94)
498
vk
1
X
iuk i
(9:95)
i1
(9:96)
i1
or, equivalently,
1
X
(9:97)
i1
1
1
X
X
f mf lEfdk m id k l jg
m1 1
Efwk iw k jg
1
X
f mf m i j N0 ij
(9:98)
m1
Also, since white noise w(k) and data d(k) are uncorrelated, we have
1
X
Efdku k jg
f mEfdkd k m jg
m1
(9:99)
f j
Substituting (9.98) and (9.99) into (9.97), we get
"
#
1
1
X
X
i
f mf m i j N0 ij f j
i1
(9:100)
m1
Taking the z-transform on both sides of (9.100) yields the following equation:
1
1
L z F zF N0 F
z
z
Thus,
L z
F 1=z
F zF 1=z N0
(9:101)
(9:102)
Therefore, L(z) consists of a discrete-time matched lter F 1=z , and the MSE-LE is given
by
C z
1
F zF 1=z
N0
(9:103)
499
9.4 MSE-LE
g(t), G( f )
Channel
h(t), H( f )
Front-end
filter
HFE( f )
AWGN
n(t)
Sub-optimum
MSE-LE
C(z)
Figure 9.20
Threshold
device
{0, 1}
Discrete-time channel model for a suboptimum MSE-LE sampled at the symbol rate.
(9:104)
where S^ is the power spectral density of the equalizer output noise ^k and is given in terms
of the input noise power spectral density C as
2
^
S^ CC
(9:105)
where the general expression of C is given in (9.47). Combining (9.104) and (9.105), we
obtain
2
^ 12 CC
^
S^e Sd QC
(9:106)
Let Sq denote the power spectral density of signal plus noise at the equalizer input, then
Sq Sd jQj2 C
By using (9.107) in (9.106), we can write S^e as follows:
2
^
Sd Sq1 Q Sd CSq1
S^e Sq C
(9:107)
(9:108)
500
d(k) Channel
Q(z)
Discretetime
matched
filter
Q(1/z)
y (k) + (k)
Equalizer
1
Q(z)Q(1/z) + (z)
Threshold d(k)
device
Noise
(z)
Sub-optimum MSE-LE
C (z)
Figure 9.21
Discrete-time channel model for the suboptimum MSE-LE sampled at the symbol rate and white data with Sd(z) = 1.
^
C
Sd Q
Sd jQj2 C
(9:109)
Sd zQ 1=z
Sd zQ zQ 1=z C z
(9:110)
^ z via its
Note that the role of the discrete-time matched lter reoccurs in the numerator of C
transfer function Q 1=z , which we did not want in the rst place! Figure 9.21 shows the
discrete-time channel model for the suboptimum MSE-LE with white data input.
2
jQe j2f 0 j Ce j2f 0
0
2
1
P
N0
HFE fT Tk
k1
(9:111)
1
1
P
f 0 k
f 0 k 2
P
1
HFE f 0 k 2
H
S
FE T
T
T
T N0
T
T
T
k1
k1
N0
N0
j2f 0
P
S
e
e
1 0
j2f 0 N
0
1
S f k 2 N0 X e
T
T
T
k1
Consequently, the MSE ^2e of the suboptimum MSE-LE is always greater than or equal to the
MSE 2e of the minimum MSE-LE. Obviously, ^2e 2e when HFE f 0 S f 0 , that is, the
front-end lter is a matched lter.
To complete the comparison, we only need to examine the equalizer output ^yk for a
symbol d(k). We wish to show that j^ykj2 jykj2 , that is, the suboptimum equalizer
produces an output signal power that is less than or equal to that of the
equalizer.
optimum
0
Without loss of generality, we can assume that data is white, that is, Sd e j2f 1. We only
501
9.5 ZF-DFE
need to compare the end-to-end transfer functions Y^ and Y of the two equalizers. This is
possible since j^ykj2 and jykj2 are just the arithmetic means of the frequency responses
jY^ j2 and jY j2 , respectively, and both Y^ and Y are nonnegative-real. We have
Y^
jQj2
2
jQj C
1 C=jQj2
1
P
2
P
1
1 N0 k jHFE j2
k SHFE
T
1 N0 =
1
P
1
T
jS j
(9:112)
X
Y
X N0
Integrating both sides of (9.112) over a full Nyquist period of unity yields j^ykj2 jykj2 ;
the equality holds if and only if HFE S . In conclusion, any MSE-LE that does not employ
a front-end matched lter has a smaller output signal-to-noise-and-ISI ratio than that of the
minimum MSE-LE.
To assess the performance of the MSE-LE, we wish to nd the equalizer output signal
power so that we can establish a gure-of-merit such as the signal-to-MSE ratio. The
output signal power jykj2 of a symbol d(k) is the arithmetic mean of the end-to-end
frequency response jY j2 , where Y X =X N0 as seen in Figures 9.18 and 9.19. We
can express Y in terms of Se N0 =X N0 as Y 1 Se . Therefore, integrating over a
h
i
full Nyquist period of unity we have jykj2 mA 1 Se 2 . The output signal-to-MSE
h
i
ratio SNR0;MSE -LE jykj2 2e mA 1 Se 2 mA Se establishes a gure-of-merit for
assessing the performance of the MSE-LE. Note that 1 Se 2 1 Se , therefore we get
SNR0;MSE-LE mA 1 Se =mA Se 1 mA Se =mA Se . Since ISI is not Gaussian, the
Q-function cannot be employed to evaluate the error probability. Nevertheless, SNR0;MSE-LE
is a useful quantity for comparing the performance of MSE-LE to that of MSE-DFE. We also
note that it is not straightforward to compare the performances of MSE-LE and ZF-LE via
SNR0;MSE-LE versus SNR0;ZF -LE in (9.43) unless we evaluate both mA Se and mA X 1 .
Although we have 2e mA Se 2 N0 mA X 1 , we cannot deduce from this fact alone
that the MSE-LE performs better than the ZF-LE.
502
1
X
f kzk
(9:113)
k0
where f k denotes the channel coefcients (tap weights). We can express the output
sequence rk of the synthetic channel as
rk dk f k
1
X
f idk i
i0
f 0dk
1
X
(9:114)
f idk i
i1
where the rst term is the desired symbol and the second term contains causal ISI produced
by earlier (preceding) symbols dk i. The ZF-DFE computes the second term based upon
^ i of dk i, which is assumed to be correct. The estimated causal ISI
the estimated dk
is then subtracted from the received symbol rk to yield
yk rk
1
X
^ i
f idk
i0
f 0dk
1
X
h
i
^ i
f i dk i dk
(9:115)
i1
f 0dk
^ i dk i, that is, assuming the previous symbols were detected correct
assuming dk
at the threshold detector output. The equalization is illustrated in Figure 9.22. Note that the
^ of the threshold detector contains no noise; therefore, noise does not
output decision dk
503
9.5 ZF-DFE
d(k)
Synthetic
channel
r(k)
F(z) = f(k)zk
y(k) + w(k)
Threshold
detector
d(k)
k=0
AWGN
w(k)
Feedback
filter
i)
= f(i)d(k
y(k)
k=1
i=1
Figure 9.22
Simplied discrete-time channel model for ZF-DFE sampled at the symbol rate.
circulate back to the threshold detector input. The AWGN wk whose power spectral
density is N0 (see Figure 9.14) is thus the threshold detector input noise.
(9:116)
^ z f 1 0F z k1
F
M
Q
1 k z1
;
jk j 1; jj51
(9:117)
1 k z1
k1
X e j2f
0
L
Q
1 k ej2f 0 2
f 2 0
k1
M
Q
(9:118)
0
j1 k ej2f j
k1
504
1=2
1=2
ln X e j2f df 0 ln f 2 0
L 1=2
X
k1
1=2
M 1=2
X
k1
1=2
0 2
ln 1 k ej2f df 0
0 2
ln 1 k ej2f df 0
(9:119)
Example 9.13
Geometric
mean and arithmetic mean of nonnegative real functions X e j2f
and X 1 e j2f
From
we can express the geometric mean mG X of the nonnegative real function
(9.120)
0
X e j2f via the Jensen inequality, as follows:
ln mG X
1=2
1=2
1=2
0
j2f 0
0
ln X e
X e j2f df 0 ln mA X
df ln
(9:121)
1=2
This implies
mG X mA X
(9:122)
j2f 0
, is also the dc component in the Fourier series
where mA X , the average value of X e
1
expansion of X f . If we replace X by X in (9.122), we get
1=2
1
df 0 mA X 1
(9:123)
j2f 0
X
e
1=2
0
where mA X 1 , the average value of 1=X e j2f , is also the dc component in the Fourier
series expansion of 1=X f as seen in (9.41). But we also have, via (9.118)(9.120),
mG X 1 f 2 0 m1
(9:124)
G X
mG X
1
(9:125)
505
9.5 ZF-DFE
In order to compare the ZF-DFE to its closest cousin, the ZF-LE, we only need to evaluate
the output signal-to-noise ratio for the received symbol yk wk at the input of the
threshold detector. We already know that the noise sample wk has the power spectral
density N0. Thus, its variance 2 is simply the integral of N0 over the full Nyquist period of
unity with respect to the discrete-time Fourier transform frequency f 0,
2 N0
(9:126)
(9:127)
y2 k f 2 0
2
N0
(9:128)
Comparing the SNR0 of the ZF-DFE to that of the ZF-LE in (9.43), with the help of (9.125),
we get
SNR0; ZF -DFE SNR0; ZF -LE
(9:129)
Hence, in the absence of error propagation, the ZF-DFE always performs better than or equal
to the ZF-LE. For a channel with substantial ISI and a low signal-to-noise ratio, the
performance of ZF-DFE could be worse than that of ZF-LE. The error probability of the
ZF-DFE can be evaluated
via the Q-function. For PSK or QPSK modulation, it is given by
p
Pb Q SNR0; ZF -DFE at high signal-to-noise ratios.
1
1
1 bz1 1 bz
^ zF
^ 1=z
f 2 0F
(9:130)
f 2 0 2E
1 b2
N0
N0
(9:131)
Note that SNR0 ! 0 as b ! 1, that is, the channel pole approaches the unit circle.
Comparing (9.131) to (9.69) it can be seen that the ZF-DFE has an advantage of 1 b2 in
SNR0 over the ZF-LE. Also note that the matched lter output signal-to-noise ratio is 2E=N0
in the case of zero ISI.
506
We have shown that in the absence of error propagation, which normally happens at high
signal-to-noise ratios, the ZF-DFE is clearly superior to that of the ZF-LE. We explain this
improvement by looking at Figures 9.14 and 9.22. The noise at the output of the ZF-LE is no
longer white, therefore, they are correlated and depend on the channel transfer function. The
ZF-DFE in effect adds a lter F z at the output of the ZF-LE to whiten the output noise, that
is, to make the noise sample uncorrelated and hence independent since it is Gaussian. The
added lter reproduces the data sample rk, which contains causal ISI and hence can be
removed by the feedback lter F z f 0 without enhancing noise. This added lter is
called a linear predictor. Figure 9.23 shows the ZF-DFE obtained from the ZF-LE by adding
a linear predictor. Note that the synthetic channel F z is the discrete-time representation of
the whitened matched lter and no lter can minimize noise better than it. This explains why
SNR0;ZF -DFE SNR0;ZF -LE .
In summary, in the absence of error propagation, the performance of the ZF-DFE is
superior to that of the ZF-LE. There is no noise enhancement due to spectral nulls. At high
output signal-to-noise ratios (more than 10 dB), the effect of error propagation is small,
especially if the data is interleaved to make it look random. Error propagation tends to end
after a few correct detections as the ZF-DFE clears itself of errors.The
error probability
for
p
PSK, including error propagation, can be expressed as Pb Np Q SNR0; ZF -DFE , where
Np is a positive constant that accounts for the increase in the error rate due to error
propagation.
d(k) Synthetic
channel
F(z)
r(k)
Causal
ZF-LE
1
F(z)
Linear
predictor
F(z)
r(k) + w(k)
+
AWGN
w(k)
Figure 9.23
The ZF-DFE as the ZF-LE with a linear predictor and a feedback lter.
Threshold
detector
Feedback
filter
F(z) f(0)
d ( k )
507
9.6 MSE-DFE
d(k)
Synthetic
channel
F(z)
r(k)
Equalizer
L(z) =
F *(1/z*)
F(z)F *(1/z*) + N0
Linear
predictor
P(z)
AWGN
w(k)
Figure 9.24
Threshold
detector
d(k)
Feedback
filter
P(z) 1
Simplied discrete-time channel model for the MSE-DFE sampled at the symbol rate.
The error spectrum of the MSE-DFE can be derived by assuming that the previous
symbols were correctly detected at the threshold detector output. We have
ek ^yk dk ^k
dk f k lk pk dk pk dk dk ^k
dk pk f k lk k ^k
S^e Sd jXC 1j2 j Pj2 N0 X jC j2 j Pj2
Sd jXC 1j2 N0 X jC j2 j Pj2
Se j P j
(9:132a)
(9:132b)
(9:134)
Thus, byusing a linear predictor and a feedback lter, we can whiten the noise plus ISI error.
0
Since S^e e j2f is a constant, the MSE of the MSE-DFE is simply
^e2
1=2
1=2
0
S^e e j2f df 0 mG Se
(9:135)
It is seen that ^e2 2e , where 2e is the MSE of the MSE-LE given in (9.90). This occurs
because the geometric mean mG Se is less than or equal to the arithmetic mean mA Se . The
combined equalizerpredictor transfer function is given by
508
dk
Composite
channel
X(z) = F(z)F (1/z)
xk
mG(Se)
N0P(1/z)
Non-white
noise n0, k
Figure 9.25
Threshold
detector
dk
Feedback
filter
P(z) 1
Discrete-time channel model for the MSE-DFE sampled at the symbol rate.
F 1=z P z
F zF 1=z N0
1
F 1=z P zSe z
N0
mG Se F 1=z
N0
P 1=z
L zP z
(9:136)
Incorporating the above result into Figure 9.24 yields the discrete-time channel model for
the MSE-DFE in Figure 9.25. The lter 1=P 1=z is similar to the noise-whitening lter
1=F 1=z in Figure 9.18. We can call it the noise-ISI-whitening lter. This lter also
attempts to eliminate the anti-causal ISI in the sequence xk, which arises via ltering by
F 1=z in (9.136). The feedback lter P z 1 attempts to cancel out the remaining
causal ISI. Since we are dealing with MSE, some residual ISI remains at the threshold
detector input. This is a compromise to reduce the effect of noise.
To assess the performance of the MSE-DFE, we wish to establish a gure-of-merit such as
the output signal-to-MSE ratio. This implies that we need to evaluate the output signal
power j^ykj2 for a symbol dk at the threshold detector input. Note that j^ykj2 is just
the arithmetic mean of the end-to-end frequency response jY^ j2 . Also we have
Y^ XP=X N0 P 1 1 N0 P=X N0 1 Se P, where Se N0 =X N0
is the error spectrum of the MSE-LE, assuming no error at the threshold detector output.
Using (9.133), we get Y^ 1 mG Se =P . Integrating jY^ j2 over a full Nyquist period of
unity yields the power of the signal component j^ykj2 1 mG Se 2 . We have used the
fact that 1=P is a maximum-phase function with 1=P 0 1; therefore, its average value
is unity. Thus, the output signal-to-MSE ratio is
2
SNR0; MSE-DFE j^ykj2 ^e2 1 mG Se 2 mG Se 1 ^e2 ^e2
(9:137)
h
i
For comparison with MSE-LE, where SNR0;MSE-LE mA 1 Se 2 mA Se , we have
h
i
SNR0; MSE-DFE SNR0; MSE-LE when mA 1 Se 2 1 mG Se 2 . For a channel with
substantial ISI and a low signal-to-noise ratio, the performance of MSE-DFE could be worse
than that of MSE-LE.
509
p
We use the same pulse st 4Ea eat ut as used in Example 9.8, where
1
X z 2Eb2 1 b1 zz b1 z b1 . For simplicity, we assume an input signalto-noise ratio E=N0 7 dB and b = 0.5. Thus,
Se z
N0
z 0:5z 2
2
z 17:5z 1
X z N0
z 0:5z 2
1 0:5z1 1 0:5z
0:115
z 0:057z 17:44
1 0:057z1 1 0:057z
(9:138)
K 1
X
dkst kT nt
(9:139)
k0
where dk is the kth symbol, st is the received symbol waveform, and nt is the AWGN.
510
For M-ary modulation, there are M K possible transmitted sequences of length K symbols.
Thus, we can readily apply the maximum likelihood principle for sequence detection
developed in Chapter 4. Recall that maximum likelihood sequence detection states that
minimizing the error probability is equivalent to selecting a sequence si t in the sequence
set that is closest in Euclidean distance to the received sequence r(t). We can write the
squared Euclidean distance d 2 r; si as follows:
1
d 2 r; si
(9:140)
jrt si t j2 dt; i 1; 2; . . . ; M K
1
1
1
1
"
2
jrt j dt 2Re
K 1
X
1
d k
k0
K 1 X
K 1
X
dkd m
k0 m0
1
1
rt s t kT dt
(9:141)
st kT s t mT dt
Since the rst term involving the integral of jrt j2 is common to all d 2 r; si , the minimum
Euclidean distance corresponds to the maximum of the following correlation metric:
cDK 2Re
K 1
X
d kvk
K 1 X
K 1
X
dkd mxm k
(9:142)
k0 m0
k0
(9:143)
1
xm k
1
1
st kT s t mT dt x k m
(9:144)
In summary, the MLSD chooses a symbol that corresponds to the smallest Euclidean
distance d r; si ; i 1; 2; . . . ; M K , or one that has the largest correlation metric cDK .
Such MLSD employs a front-end matched lter that matches the symbol waveform st .
The output of the matched lter is sampled at the symbol rate to produce a sequence of
output samples vk . These are decision variables that are used by the sequence detector
to detect the sequence DK . The sequence vk provides sufcient statistics for demodulation of DK . The sequence xk is the discrete-time autocorrelation of the known symbol
waveform st .
Obviously one does not want to build a sequence detector that follows the sampled
matched lter to evaluate M K correlation metrics when K is large. Fortunately, one can use
the Viterbi detector, which implements an efcient dynamic programming algorithm to
511
Impulse
generator
d(t)
g(t)
Channel
h(t)
si(t)
Matched
filter
s *(t)
n(t)
K1
K1
k=0
k= 0
Viterbi sequence
detector with
correlation metric
c(DK)
Figure 9.26
D
K
The MLSD with a front-end matched lter and a Viterbi sequence detector.
estimate the sequence of transmitted symbols. We have used the Viterbi algorithm in the
decoding of convolutional codes in Chapter 4, and the detection of CPM signals in Chapter 7
based upon the minimum squared Euclidean distance. For MLSD, the correlation metric in
(9.142) is used instead. This correlation metric can be computed recursively if we assume
ISI affects only L past symbols and L future symbols, that is, xk 0 for jk j4L. The
recursive formula is given by [1,2]:
cDk cDk1 2Red kvk
2Re d k
k1
X
dmxk m
!
jdkj2 x0
(9:145)
mkL
The Viterbi algorithm employs cDk to compute paths through a trellis with M states.
At each state only the path that has the largest correlation metric is retained, called
the survivor path. All paths that enter this same state (commonly referred to as the
node) with smaller correlation metrics are rejected. Thus, the complexity grows linearly
with the sequence length K and not exponentially with M K . Figure 9.26 shows the
MLSD with the Viterbi sequence detector. For M-ary modulation, such as MPSK or
MQAM, the data dk is complex with Redk representing the I-channel data and
Im(d(k)) representing the Q-data channel. Figure 9.27 shows the equivalent discrete-time
model for the MLSD.
512
Composite
channel
X(z)
DK
v(k)
x(k)
Viterbi sequence
detector with
correlation metric
c(DK)
D K
Non-white
noise n0(k)
Figure 9.27
DK
Impulse
generator
d(t)
g(t)
Channel
h(t)
si(t)
n(t)
t = kT
Matched
filter
s*(t)
vk
Noisewhitening
filter
1
F *(1/z)*
u(k)
Viterbi sequence
detector with
Euclidean
distance metric
d(uK, rK)
D K
Figure 9.28
lter output is symmetric about the sampling time. Hence, it contains both anti-causal ISI
and causal ISI. Furthermore, the output noise samples are non-white and hence correlated
but still Gaussian. On the other hand, the whitened matched lter output contains only causal
ISI and the noise samples are white, hence, uncorrelated. Since they are Gaussian, they are
also independent. We want to show that replacing the sampled matched lter in Figure 9.26
with a whitened matched lter does not change the optimality of the MLSD. Therefore, the
demodulator shown in Figure 9.28 is entirely equivalent to that in Figure 9.26.
Let X z denote the z-transform of the sequence xk dened in (9.144), that is, the
sequence of signal components at the sampled matched lter output. Performing a spectral
factorization of X z yields
1
(9:146)
X z F zF
z
where F z is a minimum-phase function and F 1=z is a maximum-phase function as
previously shown in (9.57). The noise-whitening lter in Figure 9.28 has the transfer
function 1=F 1=z . The MLSD in Figure 9.28 can be represented by the simplied
discrete-time model in Figure 9.29.
513
r(k)
DK
F(z)
Viterbi sequence
detector with
Euclidean
distance metric
d(uK, rK)
u(k)
D K
AWGN
w(k)
Figure 9.29
Simplied discrete-time model for MLSD with a whitened matched lter. The sequence rK represents the ltered data
sequence and uK = rK + wK, where wK is the AWGN sequence.
K
1
Y
fUK ukjDK
k0
K
1
Y
2
1
2
p ejukrkj =2
2
k0
(9:147)
PK1
2
2
2
1
p K e1=2 k0 jukrkj =2
2
where 2 N0 is the variance of the AWGN wk. Minimizing the ln-likelihood function
ln fUK uK jDK results in minimizing the Euclidean distance
v
uK1
uX
d uK ; D K t
(9:148)
juk rkj2
k0
over the set of M K data sequences. The Euclidean distance metric is entirely equivalent to
the correlation metric in (9.142) since both detectors apply the same maximum likelihood
principle to the sequence of data DK .
K 1
X
juk rkj2
k0
(9:149)
K 1
X
k0
jukj2 2Re
K 1
X
k0
ukr k
K 1
X
k0
jrkj2
514
The rst term on the right-hand side of (9.149) is common to all d 2 uK ; DK ; therefore,
minimizing d 2 uK ; DK corresponds to maximizing the following correlation metric:
^cDK 2Re
K 1
X
ukr k
K 1
X
jrkj2
(9:150)
dmf k m
(9:151)
k0
k0
K 1
X
m0
L
X
f kzk
(9:152)
k0
K1 X
K1
X
d mukf k m
m0 k0
2Re
K 1
X
m0
2Re
K 1
X
m0
d m
K1 X
K1 X
K1
X
dmd nf k mf k n
m0 n0 k0
K1
X
ukf k m
dmd n
m0 n0
k0
d mvm
K1 X
K 1
X
K 1 X
K 1
X
K1
X
f k mf k n
k0
dmd nxn m
m0 n0
c D K
(9:153)
The right-hand side of (9.153) is precisely (9.142) as expected. Note that the transfer
function of the noise-whitening lter is 1=F 1=z . Therefore, uk, the output of this lter
with input vk, is given in terms of the z-transform by U z V z=F 1=z , where U(z)
and V(z) are the z-transforms of the sequences uk and vk, respectively. Hence,
V z F 1=z U z is equivalent to vk, the output of F 1=z with input uk, that
is, vk uk f k where F 1=z is the z-transform of the time-reversal sequence
f k.
515
u(1) = 0.8
d(0)
r(1) = 1.2
u(2) = 0.2
d(1)
r(2) = 1.2
u(3) = 0.4
d(2)
r(3) = 1.2
u(4) = 1
d(3)
r(4) = 1.2
d(4)
+1
0.8
0.8
0.8
0.8
0.8
0.8
0.8
0.8
1
1.2
Figure 9.30
1.2
1.2
d(0)
+1
0.16 d(1)
2.12 d(2)
d(3)
0.72 d(4)
0.68
2.56
Figure 9.31
1.2
0.52
3.92
3.08
can be one of M L 2 states. The states at time k are labeled with dk 1 and the
corresponding channel outputs are rk f 0dk f 1dk 1 dk 0:2dk 1.
The channel outputs are labeled on the branches of the trellis. We assume the following
received samples (channel outputs plus noise): u1; u2; u3; u4; . . . 0:8; 0:2;
0:4; 1; . . ..
Since the channel is a nite-state machine, we have a starting state and an ending state. We
can achieve this condition by arbitrarily starting the sequence DK d0; d1; . . . ;
dK with d0 1 or d0 1 and end the sequence with dK 1 or
dK 1. We then calculate the branch metric between time k 1 and k as given by
d 2 uk; rk juk rkj2 . This branch metric is added to the path metrics at time
P
2
k 1, that is, K1
m0 jum rmj . Afterwards, the path at time k that has the smallest path
metric, called the survivor path, is retained and all other paths are discarded. Thus, at each
node, only one path survives and its metric is stored for the next computation. For our
example, we assume the starting state is d0 1. In Figure 9.31 we show the metric of the
survivor path at each node. At time k 4, there are two survivor paths with metric 0.72 and
3.92. As these two paths merge at time k 3, we can readily detect the information
sequence up to this time, which is 1; 1; 1; 1. If there is no merger, then the process
must continue until the end of the sequence. If the sequence is too long, it may be truncated
at the predetermined length K. If K is much larger than the channel memory L, there is little
loss in performance.
516
1
2 j2ft0
0
CT f 0 jS f j e
;
T
jfj
1
2T 0
(9:155)
where only the k = 0 term is counted since S f 6 0 only in j f j 1=2T 0 . Thus, CT 0 f can
equalize any phase error ej f that is experienced by S f . The output of the FSE is then
sampled again at the symbol rate 1=T to yield a spectrum Y f of period T as follows:
t = kT '
s(t) + n(t)
Figure 9.32
Matched
filter
s*(t), S*( f )
S0( f )
t = kT
X'( f )
FSE
CT '( f )
Y'( f )
Y( f )
517
9.9 Summary
1
1 X
k
0
Y f
Y f
T k1
T
1
1 X
k
k 2 j2 f k=T t0
0
S f
0
CT f
e
TT k1
T
T
(9:156)
Note that Y f is the aliased sum of equalized Y 0 f k=T , while X 0 f is the sum of
aliased component jS f k=T 0 j2 . The FSE, therefore, can be designed to assume the role
of ZF-LE, MSE-LE, and the linear predictor of the DFE.
9.9 Summary
In this chapter we have provided a detailed study of intersymbol interference and fundamental equalization
techniques. We introduced the concept of zero ISI via the Nyquist criterion. The raised-cosine pulse is the Nyquist
pulse that is widely used in practice. The optimum demodulator was designed based on the criteria of zero ISI and
maximum output signal-to-noise ratio. In other words, the pulse shape at the threshold detector input is a
Nyquist pulse that achieves zero ISI at the sampling time. Furthermore, the sample presented to the threshold
detector has a maximum signal-to-noise ratio. The optimum demodulator employs two lters in the transmitter.
One lter has a frequency response equal to the inverse of the channel frequency response. In effect, the
combination of this lter and the channel yield an ideal channel with no distortion. This process is called
pre-equalization, where the channel distortion effect is pre-corrected at the transmitter. The receiver lter is
designed to match the second lter in the transmitter to provide an output Nyquist pulse and to emulate the
matched lter.
Pre-equalization in the transmitter is not practical because the transmitter often does not have the means
to estimate the channel. Therefore, equalization is commonly carried out in the receiver where the channel
response can be estimated. We presented two types of linear equalizers, namely, the zero-forcing linear equalizer
(ZF-LE) and the mean-square error linear equalizer (MSE-LE). The ZF-LE forces the sample at the threshold
detector input to have zero ISI. It does so at the expense of the output signal-to-noise ratio, which can go to
zero if the channel response has spectral nulls. The MSE-LE compromises between ISI and the output signal-tonoise ratio.
The performance of linear equalizers can be improved by using feedback decision. We presented two
nonlinear equalizers, namely, the zero-forcing decision-feedback equalizer (ZF-DFE) and the mean-square error
decision-feedback equalizer (MSE-DFE). These nonlinear equalizers suffer error propagation but nevertheless may
perform better than their linear counterparts. The best nonlinear equalizer is the maximum likelihood sequence
detector (MLSD), which can be implemented by the Viterbi algorithm. Finally, we studied the fractionally spaced
equalizer (FSE), which can correct the timing error problem in previously studied equalizers. The FSE works on
the aliased sum of equalized components instead of the sum of aliased components, thereby solving the timing
problem before it can be rendered uncorrectable by aliasing.
518
Problems
1.
2.
Show that the raised-cosine pulse shape and its Fourier transform satisfy the Nyquist
criterion of zero ISI.
P
P
Show that 1=T m P f m=T is the Fourier transform of pt t mT
P
m
pt t mT .
m
3.
4.
5.
6.
7.
8.
9.
10.
11.
12.
Show that when the sampling rate 1/T is greater than twice the absolute bandwidth B
of the Fourier transform P( f ) of the pulse shape p(t), then the Nyquist criterion of
zero ISI cannot be satised.
Consider an antipodal signal with squared pulse shape p(t) of unit amplitude and
duration T. Sketch the waveform at the output of the matched lter when the input bit
sequence is 10011. Specify the amplitudes at the sampling times. Is there any
intersymbol interference?
Find the impulse response of the lter having the frequency response equal to the
square root of P f T ; 1=2T f 1=2T and zero otherwise.
Consider an antipodal signal with squared pulse shape p(t) of unit amplitude and
duration T. Derive the output waveform for one bit, say 1, at the output of an RC-lter
(series resistance R and shunt capacitor C).
(a) Sketch the output waveform when the input bit sequence is 10011and specify the
amplitudes at the sampling times for the case RC = T. Is there any intersymbol
interference?
(b) Derive the output SNR for the RC-lter for a single bit at the sampling time T,
assuming the noise spectrum is N0 =2. The antipodal signal amplitudes are A.
(c) Derive the one-shot output SNR for each bit in the sequence 10011 for the case
above that includes intersymbol interference.
(d) Derive the one-shot bit error probability for each bit in the sequence 10011 for the
case (c) that includes intersymbol interference, assuming that noise is Gaussian.
Find the impulse response of the square root raised-cosine lter.
Verify (9.63).
Verify (9.65).
Consider a bandlimited QPSK signal whose bandwidth is B.
(a) Draw the equivalent lowpass model for the transmitter, channel, and receiver.
(b) Draw the block diagram of an optimum system that can achieve the maximum
output SNR0 at the sampling time.
Consider the following samples of the output pulse of a matched lter:
. . . ; 0; x1 a; x0 1 a2 ; x1 a; 0; . . .. Evaluate the composite channel
transfer function, the synthetic channel transfer function, and the ZF-LE transfer function.
Consider a channel with impulse response ht at bt t0 . The input to this
channel is the pulse pt.
(a) Evaluate the channel output st.
(b) Consider a lter that is matched to st with a designated impulse response
sT 0 t, where is a scaling constant and T 0 is a time delay. Evaluate the
output s0 t of the matched lter at the time t kT 0 when T 0 t0 .
519
Problems
13.
14.
15.
16.
17.
18.
19.
20.
21.
(c) Assuming that pt is orthogonal to its t0 -translates, evaluate the output sequence
xk s0 kt0 . Evaluate the z-transform of xk.
(d) Find the synthetic channel transfer function of the sequence xk 1.
Consider the following sequence of pulses at the input of the matched lter:
P
vt n dn st nT , where dn 2 f1; 1g represents the data bit at the time
index n. Furthermore, p
let
the equivalent lowpass pulse of the modulated pulse
st cos 2fc t be st 4Ea eat ut, a > 0 (see Example 9.8).
(a) Calculate the signal-to-noise ratio at the output of the sampler at the time index
k = 1, assuming d0 1; d1 1.
(b) Calculate the signal-to-noise ratio at the output of the sampler at the time index
k = 2, assuming d0 1; d1 1; d2 1.
(c) Calculate the signal-to-noise ratio at the output of the sampler at the time index
k = 3, assuming d0 1; d1 1; d2 1; d3 1.
(d) Compare the results in (a)(c) to that of the ZF-LE.
Consider the following sequence of pulses at the input of the matched lter:
P
vt n dn st nT , where dn 2 f1; 1g represents the data bit at the time
index n. Furthermore, p
let
the equivalent lowpass pulse of the modulated pulse
st cos 2fc t be st 2Ea eajtj , a > 0 (see Example 9.8).
(a) Calculate the signal-to-noise ratio at the output of the sampler at the time index
k = 1, assuming d0 1; d1 1.
(b) Calculate the signal-to-noise ratio at the output of the sampler at the time index
k = 2, assuming d0 1; d1 1; d2 1.
(c) Calculate the signal-to-noise ratio at the output of the sampler at the time index
k = 3, assuming d0 1; d1 1; d2 1; d3 1.
Using Example 9.8, determine the sampling rate so that SNR0;ZF -LE 1:5E=N0
Consider the composite channel X z 1 0:5z1 0:5z. Calculate the meansquare error of MSE-DFE via a spectral factorization.
1
Consider the channel specied p
bythe
Fz
FIR lter p
f 0 f 1z , where the
channel tap weights are f 0 2E; f 1 0:4 2E. The input to this channel is a
sequence of two bits fdk; dk 1g, where di 2 f1; 1g. Calculate the power
jrkj2 of the output bit rk for all four bit patterns. Find the one-shot bit error
probability for each of the four bit patterns, assuming the modulation is PSK.
Consider the composite channel X z a2 bz bz1 .
0
(a) Derive the condition on a and b so that X e j2f is a positive-real function, that is,
0
X z can be expressed as X z FzF 1=z . Find Fz and X e j2f . Does
0
X e j2f have spectral nulls?
(b) Calculate SNR0;ZF -LE .
Consider the synthetic channel Fz f 0 f 1z1 .
(a) Calculate the mean-square errors of the MSE-LE and MSE-DFE.
(b) Calculate SNR0;MSE-LE and SNR0;MSE-DFE .
Verify (9.121).
Consider a synthetic channel with transfer function Fz f 0 f 1z1 .
(a) Derive the bit error probability for the received PSK signal.
(b) Derive the bit error probability for the received PSK signal with ZF-DFE.
520
22. Carry out the Viterbi algorithm using Figure 9.30 with the following data:
u1; u2; u3; u4; . . . 1; 0; 0:2; 1:1; . . ., Fz 1 0:1z1 .
Further reading
For further studies on ISI the reader is referred to the excellent text by Proakis [2], which also
discusses controlled ISI. The effect of ISI can be found in the excellent book by Pursley [3].
More information on equalization techniques can be found in [2] or in the excellent books by
Barry et al. [4] and Gitlin et al. [5]. The above books also cover adaptive equalization, in
which the equalizer taps can be adjusted to accommodate a time-varying channel. More
advanced topics on adaptive lters can also be found in Haykin [6]. A good discussion of
sequence demodulation can be found in Blahut [1].
Bibliography
1. R. E. Blahut, Digital Transmission of Information, New York: Addison-Wesley, 1990.
2. J. G. Proakis, Digital Communications, 5th edition, New York: McGraw-Hill, 2008.
3. M. B. Pursley, Introduction to Digital Communications, Upper Saddle River, NJ:
Pearson Prentice Hall, 2005.
4. J. R. Barry, E. A. Lee, and D. G. Messerschmitt, Digital Communication, 3rd edition,
London: Kluwer Academic Publishers, 2004.
5. R. D. Gitlin, J. F. Hayes, and S. B. Weinstein, Data Communications Principles,
London: Plenum Press, 1992.
6. S. Haykin, Adaptive Filter Theory, London: Prentice Hall, 1996.
7. S. A. Alteka and N. C. Beaulieu, Upper bounds on the error probability of decision
feedback equalization, IEEE Trans. Inform. Theory, Vol. 39, pp. 145156, 1993.
8. C. A. Belore and J. H. Park, Decision feedback equalization, Proc. IEEE, Vol. 67,
No. 8, pp. 11431156, 1979.
9. G. D. Forney, Jr., Maximum likelihood sequence estimation of digital sequences in
the presence of intersymbol interference, IEEE Trans. Inform. Theory, Vol. 18,
pp. 363378, 1972.
10. G. D. Forney, Jr., The Viterbi algorithm, Proc. IEEE, Vol. 61, No. 3, pp. 268278, 1973.
11. R. W. Lucky, Automatic equalization for digital communications, Bell Syst. Tech. J.,
Vol. 44, pp. 547588, 1965.
12. R. W. Lucky, Techniques for adaptive equalization of digital communication, Bell
Syst. Tech. J., Vol. 45, pp. 255286, 1966.
13. R. W. Lucky, J. Salz, and E. J. Weldon, Jr., Principles of Data Communication, New
York: McGraw-Hill, 1968.
14. H. Nyquist, Certain topics in telegraph transmission theory, AIEE Trans., Vol. 47,
pp. 617644, 1928.
521
Bibliography
15. A. Papoulis, Probability, Random Variables, and Stochastic Processes, 3rd edition,
New York: McGraw-Hill, 1968.
16. S. Qureshi, Adaptive equalization, Proc. IEEE, Vol. 53, No. 9, pp. 13491387, 1985.
17. G. Ungerboeck, Fractional tap-spacing equalizer and consequences for clock recovery
in data modems, IEEE Trans. Commun., Vol. 24, No. 8, pp. 856864, 1976.
18. S. Verdu, Maximum likelihood sequence detection of intersymbol interference channels: a new upper bound on error probability, IEEE Trans. Inform. Theory, vol. 33,
pp. 6268, 1987.
10
Fading channels
Introduction
Besides the intersymbol interference (ISI) that occurs via channel ltering, a digitally
modulated signal can also have ISI when it is transmitted over a multipath fading channel.
This type of channel is encountered in all forms of mobile wireless communication. In a
multipath fading channel, the transmitted signal arrives at the receiver via multiple paths.
These paths generally arise via signal reection from the ground, hills, buildings, and any
other large structures. They also arise from signal diffraction via bending around the
corners of buildings or sliding across rooftops. They also can arise via signal scattering
from small objects such as vehicles, lamp posts, trees, etc. Each signal path results in a
randomly delayed, attenuated, and phase-shifted copy of the transmitted signal. These
multipath copies combine at the receiver to give rise to a received signal whose envelope
may be described by a Rayleigh fading process (no line-of-sight path), or a Rice fading
process (one line-of-sight path), or a Nakagami fading process. Also, because the arrival
times of the multipath copies are random, especially in a mobile environment, the multipath copies might overlap the next bit or symbol and hence cause intersymbol interference. This type of ISI cannot be eliminated by pulse shaping dictated by the Nyquist
criterion for zero ISI, but can be alleviated by equalization (as discussed in Chapter 9). The
above effects are collectively called fading. A fading channel that exhibits ISI is called a
frequency-selective fading channel. A channel that exhibits negligible ISI is called a at
fading channel.
Figure 10.1 illustrates a frequency-selective fading channel in which a signal travels
several propagation paths creating multiple received copies of it at the receiver.
Figures 10.2 and 10.3 illustrate the ISI process due to multipath fading effects for three
bits 1, 0, 1 with a squared pulse shape at the input and output of the matched lter,
respectively, in the case of two propagation paths. ISI occurs at the sampling time t = 2T,
3T. Note that the amplitudes of these bits are not the same to reect the channel fading
effect. In this case, the complex envelope fading varies from bit to bit but remains
essentially the same during a bit time. This phenomenon is commonly called slow fading
to reect the fact that the channel varies slowly within one or several bit times. When the
channel varies substantially during a bit time, the channel is said to be a fast fading
channel. Both slow and fast fading phenomena result from the movement of the transmitter and/or receiver, the so-called Doppler effect. In a bandlimited multipath fading
channel, both ltering-induced ISI and multipath-induced ISI occur simultaneously and
can cause serious performance degradation. Also, the time variation can induce carrier
522
523
Figure 10.1
Introduction
Multipath fading channel with multiple copies of the signal arriving at the receiver.
Figure 10.2
3T
Pre-ltering ISI for a squared pulse shape. The solid line is for the rst arrival and the dashed line is for the delayed
second arrival.
2T
0
Figure 10.3
3T
4T
Post-ltering ISI at the output of the matched lter for two propagation paths.
phase error, which results in a loss of the signal-to-noise ratio at the coherent receiver. The
combined time variation and multipath effect give rise to a time-varying and space-varying
channel. This channel can be represented by an impulse response h; t, where the time t
represents the variations of the channel and the time represents the time arrivals of the
multipath copies. When the channel is time-invariant, we have h; t h. Wireless
channels such as cellular channels, WLAN, WMAN, and underwater channels are common examples of Rayleigh fading channels, which have two non-conicting attributes of
the four attributes described above (slow fading, fast fading, at fading, and frequencyselective fading).
524
Fading channels
Time-varying effect
We examine rst the cause of time variation of the channel. It arises mainly from the
Doppler effect due to the motion of the transmitter and/or receiver and the time variation
of the structures of the medium. Let us consider a received free space E-eld (V/m) at
distance d greater than a reference distance d0 from the transmitter. We have
Ed; t
E0 d0
d
cos 2fc t
d
c
(10:1)
where E0 is the free space E-eld at d0 . If we now assume that the receiver moves away from
the transmitter with radial velocity v, the distance d increases in time by vt. The received
E-eld is given by
E0 d0
d vt
Ed; t
cos 2fc t
c c
d vt
(10:2)
E0 d0
v
d
cos 2fc 1 t
d vt
c
c
Thus the carrier frequency fc becomes fc 1 v=c. The received E-eld encounters a
Doppler shift of f fc v=c. If the receiver moves toward the transmitter, the Doppler
shift is f fc v=c. The Doppler shift induces two effects: the time-varying phase shift
t 2ft and the time-varying amplitude E0 d0 =d vt. Let us consider the rst example
where the carrier frequency is fc 1 GHz and the receiver radial velocity is v = 100 km/hour
(moving away). Then, the Doppler shift is 92:6 Hz. Consider a channel bit rate of
14.4 kbps, then the bit interval is 69.4 s. Let us consider a bit which starts at time t = 0
and ends at time t = 69.4 s. The phase shift at the end of the bit is 0:04 radians or 2:3. If
the criterion for channel estimation is no more than 45 (=4 radians) phase error then the
delay requirement for channel estimation cannot exceed 1=8f seconds or 20 bit times. Also,
during 20 bit times the receiver moves a distance 0.039 m. This induces a small variation in
the received E-eld amplitude.
Now let us consider the second example of a transmitter and a receiver separated by a
distance d. If we assume that the receiver moves straight toward the transmitter with radial
velocity v, the distance d1 decreases in time by vt. The Doppler shift of the LOS E-eld is
f fc v=c. In addition, suppose the receiver also receives a reected E-eld from a perfect
reector directly behind it. The distance traveled by this reected E-eld is d2 . Since
the receiver is moving straight away from the reector, the distance increases in time by vt.
525
The Doppler shift of the reected E-eld is fc v=c. The reected signal is radians outof-phase with its incident E-eld. Therefore, the total received E-eld at the receiver is
given as
E0 d0
d1 vt
E0 d0
d2 vt
cos 2fc t
cos 2fc t
Ed; t
d1 vt
c
d2 vt
c
c
c
E0 d0
v
d1
E0 d0
v
d2
cos 2fc 1 t
cos 2fc 1 t
c
c
d1 vt
c
d2 vt
c
(10:3)
For simplicity let us consider the case where the receiver is close to the reector so that the
amplitudes of both LOS and reected E-elds are approximately the same. Then the sum of
the two E-elds can be approximated by the following expression:
Ed; t
2E0 d0
sin2fc vt=c d2 d1 =2c sin2fc t d2 d1 =2c
d2 vt
(10:4)
The received E-eld now has a time-varying amplitude which varies according to the
Doppler shift. The larger the Doppler shift, the faster the amplitude varies. This results in
deep fades at intervals of 1/4f (or a quarter-wavelength) from the peak amplitudes. For the
numerical values used in the rst example, a deep fade occurs every 39 bits from the peak
amplitudes. The difference between the Doppler shifts of the LOS and reected E-elds is
referred to as the Doppler spread, which we dene below.
(10:5)
where the maximum is taken over all positive Doppler shifts fi and negative Doppler shifts fj
of the propagation paths.
For the above example the Doppler spread is 2fc v=c. In general, Doppler shift induces
both time variation in signal amplitude and carrier phase error. These two phenomena cause
a loss in the signal-to-noise ratio and degrade performance.
Denition 10.2 The coherence time of a fading channel is related to the Doppler spread as
follows:
Tc
1
4fD
(10:6)
According to this denition, the coherence time is an empirical relation that can be
adjusted to t the requirement. Obviously, the coherence time is innite when there is no
Doppler shift. This implies that the channel is time-invariant and no phase error occurs.
There is still loss in signal amplitude due to the sum of the various signal rays at the receiver,
but this loss can be compensated for with power control.
526
Fading channels
Space-varying effect
The fading wireless channel is also space-varying, that is, the multipath effect varies
according to different locations of transmitter and receiver even when the transmitter and
receiver are stationary and the structures of the medium do not change. Thus, a multipath
intensity prole (MIP) which shows power versus delay or magnitude versus delay of each
multipath copy is space-dependent, that is, it is a function of the transmitter and receiver
coordinates. An example of a realization of MIP is shown in Figure 10.4. As previously
discussed, the multipath effect might cause ISI in the received symbol stream. To characterize this effect we introduce two parameters called multipath delay spread and coherence
bandwidth.
Denition 10.3 The multipath delay spread Td of a fading channel is empirically dened
as the maximum of the difference in propagation times between the rst path with power
P1 and the last signicant path of power Pl , such that P1 =Pl is greater than but closest to a
given threshold . For example, the threshold can be taken to be SNR P1 =N , the signalto-noise ratio of the rst path. The maximum is carried out over all multipath intensity
proles:
(10:7)
This denition implies that the ISI effect of multipath copies should be kept small when
one estimates the multipath delay spread for waveform design purposes. If the receiver locks
onto the rst path, then the spill-over symbol energy from delayed paths might cause ISI in
the next symbol. Thus, one should include as many signicant paths as possible so that the
ISI effect is not underestimated. Equation (10.7) also implies that the transmission time of a
symbol should far exceed the multipath delay spread to have a small ISI effect. Instead of
designing for the worst case, real-time MIP can also be obtained from channel estimation to
Magnitude
Time
Figure 10.4
527
vary the transmission rate to take advantage of the channel condition. This implies a higher
symbol rate for small Td and a lower symbol rate for a large Td . For the MIP realization in
Figure 10.4, the multipath delay spread can be taken to be the time spead between the rst
path and the fth path, assuming P1 =Pl ; l 2; 3; 4, and P1 =P5 4.
Denition 10.4 The coherence bandwidth Bc of a fading channel is dened as the inverse
of the multipath delay spread
Bc
1
Td
(10:8)
This is the bandwidth necessary to accommodate the symbol time no smaller than the
multipath delay spread and with sin x=x pulse shape.
Frequency (MHz)
900
900
2400
215 s
0.32 s
20300 ns
528
Fading channels
529
(
xt Re
(
Re
XX
i
aik tsL t i t e
j2fc t i tf ik t
XX
i
!
aik te
j2fc i tf ik t
sL t i t e
(10:10)
j2fc t
The complex envelope xL t of the received signal x(t) is given by the following expression:
xL t
XX
i
(10:11)
hi tsL t i t
where the term hi t is the ith complex path attenuation, and is often referred to as the ith
channel tap
X
hi t
aik tej2fc i tf ik t
(10:12)
k
By the central limit theorem the channel tap hi t can be modeled as a complex Gaussian
random variable whose in-phase and quadrature components have the same variance. Thus
the magnitude jhi tj of the ith channel tap is a Rayleigh or Rice process (when a line-ofsight component is present). Alternatively, one can model the channel tap by a Nakagami
process (see Chapter 3). From (10.11) we obtain the complex envelope of the impulse
response (equivalent lowpass impulse response, baseband impulse response) of an uncorrelated fading channel as follows:
X
h; t
hi t i t
(10:13)
i
hi tej2f i t
(10:14)
For the special case of the time-invariant channel the complex envelope and the frequency
response are obtained from (10.13) and (10.14) as
h
hi i ;
Hf
hi ej2f i
(10:15)
4
X
i1
hi li T
(10:16)
530
Fading channels
Tap hi
Channel 1
Channel 2
1
2
3
4
0.8
0.4
0.33
0.3
0
0.2
0.3
0.5
0
0.5
1.2
2
Hf
4
X
hi ej2li f T
(10:17)
i1
The magnitude and phase versus the frequency-time product fT of the frequency responses
are shown in Figure 10.5. The rst channel is a at fading channel when the signal
bandwidth-multipath delay spread product is smaller than fT = BcTs = 0.5, where Bc is the
channel coherence bandwidth and Ts is the symbol time. In this range, the magnitude
spectrum is fairly at and the phase is linear. For the same range of fT, the second channel
has frequency-selective fading, since the magnitude spectrum shows considerable variation
and the phase is not linear.
Substituting h; t
Rh ; 0 ; t1 ; t2 Eh; t1 h 0 ; t2
(10:18)
XX
i
(10:19)
Ehi t1 hk t2 i t1 0 k t2
This means that the channel autocorrelation is non-zero only when all the delays are the
same. Thus, the channel autocorrelation reduces to the following expression:
"
Rh ; t1 ; t2
X
i
#
Ehi t1 hi t2 i t1
(10:20)
531
0.8
1.8
0.6
1.6
0.4
1.4
0.2
( f )in radians
H( f )
1.2
1
0.8
0.4
0.6
0.6
0.4
0.8
1
0.2
0
0.5
1.5
(a)
2.5
fT
3.5
4.5
0.5
1.5
(b)
2.5
fT
3.5
4.5
4.5
1.8
1.5
1.6
1
( f )in radians
1.4
H( f )
0
0.2
1.2
1
0.8
0.6
0.5
0
0.5
1
1.5
0.4
0.2
2.5
0
0.5
(c)
Figure 10.5
1.5
2.5
fT
3.5
4.5
0.5
1.5
(d)
2.5
fT
3.5
Frequency responses of (a),(b) the at fading channel 1 and (c),(d) the frequency-selective fading channel 2.
If we assume that the fading process fhi tg is a WSS random process and also ergodic we
can take the ensemble average of Rh ; t1 ; t2 to be its time average, which depends only on
t0 t2 t1 , and hence eliminate the time origin t1 . From now on we assume that the channel
is WSS and ergodic and undergoes uncorrelated fading. Thus we have the multipath
autocorrelation prole
X
Rhi t0 i
(10:21)
Rh ; t0
i
(10:22)
532
Fading channels
Sh ; f 0
Shi f 0 i
(10:23)
where the ith path Doppler power spectrum Shi f 0 is the Fourier transform of the path
autocorrelation Rhi t0 . This means that all multipath copies of the transmitted signal are
affected by the Doppler shift. The Doppler spread of the fading channel is approximately the
maximum of the path Doppler spreads of signicant paths (the path Doppler spread is twice
the absolute bandwidth of the path Doppler spectrum.). The coherence time of the fading
channel is dened as in (10.6).
(10:24)
k
Assuming that the scatterers are evenly spaced around the mobile, we can express the arrival
angle as k k and 2=L.
"
#
X
P
0
e j2fm cos kt
Rh ; t0
L k
"
#
(10:26)
P X j2fm cos kt0
e
2 k
As the number of scatterers around the mobile approaches innity, the summation becomes
an integral and we have
P
0
Rh ; t0
e j2fm cos t d
2
(10:27)
0
PJ0 2fm t
533
Integral approach
The multipath autocorrelation prole for random horizontal arrival angles is given as
follows:
"
#
n
o
X
0
Ejhk j2 E e j2fm cos k t
Rh ; t0
"
#
P X n j2fm cos k t0 o
E e
L k
h n
oi
0
PE e j2fm cos k t
(10:28)
P
0
j2fm cos t0
e
d
Rh ; t
2
(10:29)
PJ0 2fm t0
The multipath Doppler prole is the Fourier transform of Rh ; t0 in the t 0 variable, and is
given by
8
P
>
0
>
< q ; jf j fm
2
0
2
0
Sh ; f fm f
(10:30)
>
>
:
0;
otherwise
The ClarkeDoppler power spectrum is the value of Sh ; f 0 at 0, and is given by the
following expression:
8
P
>
0
>
< q ; jf j fm
2
0
2
0
fm f
Sh f
(10:31)
>
>
:
0;
otherwise
This power spectrum has singularities at jf j fm , which make it spike to innity. This
happens because the scattered waves are assumed to be on the horizontal plane only and
uniformly distributed around the receiver. In practice, Doppler power spectra do resemble
this theoretical one but do not have singularities at fm . The Doppler spread fD of the Clarke
fading channel is twice the maximum Doppler shift, that is,
fD fm fm 2fm
(10:32)
1
1
4fD 8fm
(10:33)
534
Fading channels
(10:34)
where dk is the distance traveled by the kth wave, k0 2fc =c is the free space wave
number, and f k is the wave phase. The random variables E0;k dk , k ; k ; f k are assumed
to be independent. The total received E-eld at the mobile is the sum of L incident
waves:
X
X
Ed; t
Ek dk ; t
E0;k dk e jk0 x sin k cos k y sin k sin k z cos k f k (10:35)
k
where d d1 ; d2 ; . . . ; dL and the coordinates x; y; z are given in terms of the speed v and
the initial mobile position x0 ; y0 ; z0 as
kth wave
k
y
k
Mobile direction
Figure 10.6
535
x x0 v cos t
y y0 v sin t
(10:36)
z z0
We consider the case of small-scale variations in E-eld amplitudes and approximate the
mean-square value of the E-elds as constant and independent of distance, that is,
EjE0;k dk j2 P=L, where P is the total received power. The maximum Doppler shift is
fm fc v=c. The total received E-eld is
X
Ed; t
Ek dk ; t
k
(10:37)
Dene
hk
where
k0 x0 sin k cos k y0 sin k sin k z0 cos k f k .
2
2
j
E0;k dk e , then Ejhk j EjE0;k dk j P=L. The corresponding impulse response
of the channel is
"
#
X
j2fm sin k cos k t
hk e
h; t
(10:38)
k
Ejhk j E e
Rh ; t
k
PX
0
(10:39)
(10:40)
0 0
The functions f and f are the probability density functions of the polar angle and
azimuth angle , respectively. Since the mobile position is arbitrary on the horizontal plane,
we can assume a uniformly distributed azimuth angle and hence we have f 1=2.
Therefore, the autocorrelation can be written as
J0 2fm t0 sin f d
(10:41)
Rh t0 P
0
Taking the Fourier transform of the autocorrelation Rh t0 in the variable t0 we obtain the
generalized Doppler power spectrum
536
Fading channels
Sh f
(
fm
P
p
f d ; jf 0 j fm
2
2
sin sin
0;
otherwise
; sin
f0
fm
(10:42)
For the special case of incident waves restricted to the horizontal plane, we have
f sin =2 and the generalized Doppler power spectrum reduces to the
ClarkeDoppler power spectrum. The generalized at fading channel is also a slow fading
channel so long as the channel estimation time is smaller than the coherence time, which is
one-eighth of the maximum Doppler shift.
sin
;
2 sin
=2 =2
(10:43)
Thus, the arrival angle of the incident waves is restricted to the angular region
=2 =2 . Substituting f into (10.42) we obtain the Aulin
Doppler power spectrum, which does not have singularities at f 0 fm :
8
>
sin
1 p
P
>
; 0 jf 0 j fm cos
>
< fm sin sin
1f 0 =f m 2
(10:44)
Sh f 0
P
;
fm cos jf 0 j fm
>
>
> 2fm sin
:
0;
otherwise
537
t = Ts
X1
MF1
2
Ts
s(t) + n(t)
Minimum
Euclidean
distance
detector
cos 2 fct
t = Ts
si
Symbolto-bit
converter
{0, 1}
X2
MF2
2
Ts
Figure 10.7
sin 2 fct
t = Ts
sL(t) Complex-
s(t) + n(t)
valued
MF
Mapper
si si
X = X1 + jX2
Minimum
Euclidean
distance
detector
Symbol-tobit
converter
{0, 1}
Figure 10.8
(10:45)
(10:46)
where N N1 jN2 is the complex noise. As the signal passes through the channel, it is
multiplied by the channel taps. Let us consider the case where the channel is a at fading
538
Fading channels
channel. The received signal now has a complex envelope given by hsL t, where the
complex channel tap h jhje j is assumed to be available via perfect channel estimation.
To achieve
the maximum signal-to-noise ratio (SNR)
we require the use of the local complex
p
p
carrier 2=Ts h =jhjej2fc t , which is equal to 2=Ts ej ej2fc t , such as the pilot tone in
IS-95 or the pilot subcarriers in OFDM-based signals (IEEE 802.11a,g and 802.16,e).
Consequently, the output of the complex-valued matched lter is the complex variable
h
hsi N
j hj
h
jhjsi N
j hj
(10:47)
jhjsi1 jsi2
h
N1 jN2
jhj
Both I-noise N1 h =jhjN1 and Q-noise N2 h =jhjN2 have the same variance 2 .
The above complex decision variable can be written in vector form as follows:
X jhjsi N
(10:48)
where N N1 N2 t . Conditional on jhj, the decision variable vector for a slow-at fading
channel is identical to that of an AWGN channel, since the zero mean I-noise and Q-noise
samples are statistically independent and have the same variance 2 . For a slow-at fading
channel, the signal vector si is weighted by the channel tap magnitude jhj. The complex
decision variable or its equivalent decision variable vector is a sufcient statistic. Given a
realization x of X we wish to develop a metric for the maximum likelihood detection. The
conditional probability density function of X given jhj; Hi is
fX xj jhj; Hi
2
Y
k1
fXk xk j jhj; Hi
2
Y
2
1
2
p exk jhjsik =2
k1 2
P2
2
1
1
p 2 e 22 k1 xk jhjsik
2
2
1
1
p 2 e 22 kxjhjsi k
2
(10:49)
(10:51)
This is the metric that the minimum Euclidean distance detector, with the available channel
tap h from the pilot tone or pilot symbol, makes the decision on the transmitted signal. That
is, it selects the signal vector jhjsi that is closest in Euclidean distance to the sampled value
539
x of the decision vector X. The conditional pair-wise error probability for a at fading
channel between two vectors jhjsi and jhjsj is
jhjsi sj
Pr jhjsi ! jhjsj Q
(10:52)
2
The conditional bit error probability for the Gray-coded signal set fsi gM
i1 is given by the
following approximation:
Nn
jhjdmin
Q
(10:53)
Pb jhj
log2 M
2
where dmin min jjsi sj jj is the minimum Euclidean distance of the signal set, 2 N0 =2,
i;j
and Nn is the average number of symbols at the minimum Euclidean distance dmin or the
average number of nearest neighbors. Here N0 =2 is the power spectral density of noise. In
(10.53) the equality holds for both PSK and QPSK. Knowing the statistics of the fading
channel (Rayleigh, Rice, Nakagami) we can evaluate the bit error probability as follows:
Nn
jhjdmin
Q
Pb EPb jhj E
log2 M
2
q
(10:54)
Nn
Q
2jhj2 SNR
E
log2 M
where the symbol signal-to-noise ratio SNR and the positive constant which depends on
the modulation are given by
Es
1 dmin 2
; SNR
(10:55)
SNR
N0
2 2
The quantity jhj2 SNR is referred to as the instantaneous SNR, and does not exist in an
AWGN channel.
Rayleigh
For convenience we normalize the channel tap such that E jhj2 1. The channel tap jhj is
Rayleigh-distributed and jhj2 is exponentially distributed with density functions
fjhj x 2xex ; fjhj2 y ey
2
(10:56)
The corresponding bit error probability is given by the following closed-form expression
(see Appendix 10B):
r!
Nn
SNR
1
Pb
(10:57)
2 log2 M
1 SNR
p
PSK: the minimum Euclidean distance is dmin 2 Eb . With 2 N0 =2, we obtain
SNR Eb =N0 . Furthermore, Nn 1, M = 2, therefore the bit error probability for
PSK is
540
Fading channels
1
Pb
1
2
s!
Eb =N0
PSK
1 Eb =N0
(10:58)
p
QPSK: the minimum Euclidean distance is dmin 2Es . With 2 N0 =2, we obtain
SNR Es =2N0 Eb =N0 . Furthermore, Nn 2, M = 4, therefore the bit error probability for QPSK is identical to that of PSK as expected
s!
1
Eb =N0
QPSK
(10:59)
Pb
1
2
1 Eb =N0
p
MPSK: the minimum Euclidean distance is dmin 2 Es sin=M . Using 2 N0 =2,
we obtain SNR Es =N0 sin2 =M log2 M Eb =N0 sin2 =M . Furthermore,
Nn 2, therefore the bit error probability for MPSK is given by
s!
1
log2 M Eb =N0 sin2 =M
Pb
1
MPSK
(10:60)
log2 M
1 log2 MEb =N0 sin2 =M
p
MQAM: the minimum Euclidean distance is dmin 6Eb log2 M =M 1. Using
we
2 N0 =2, p
obtain SNR 3=2log2 M =M 1Eb =N0 . Furthermore,
Nn 4 4= M , therefore the bit error probability for MQAM is given by
s!
p
2 2= M
3=2log2 M =M 1Eb =N0
1
MQAM (10:61)
Pb
1 3=2log2 M =M 1Eb =N0
log2 M
Rice
When there is a line-of-sight component among the non-direct (diffuse) components the
channel tap jhj is Rice-distributed:
x x2 A2 =22s
Ax
I0 2
fjhj x 2 e
s
s
(10:62)
The function I0 x is a modied Bessel function of the rst kind of zero order. When there is
no signal, that is, A = 0 and I0 0 1, the Rice density function reduces to the Rayleigh
density
function. We normalize the channel tap jhj such that its mean-square value is
E jhj2 A2 2 2s 1. The ratio A2 =2 2s is called the direct signal-to-diffuse signal
ratio. We have A2 = 1 and 2 2s 1= 1. Therefore, the Rice density function
of jhj and the non-central 2 -density function of jhj2 can be rewritten in term of the
parameter as follows:
p
I0 2x 1
p
fjhj2 y 1e1y I0 2 1y
fjhj x 2 1xe1x
(10:63)
541
There is no closed-form expression for Pb with a Rice-distributed channel tap, and numerical integration has to be carried out to evaluate (10.54).
Nakagami-m
The Nakagami-m density function represents the most general fading statistics.
The channel
tap is Nakagami-distributed with the following density functions for E jhj2 1:
fjhj x
(10:64)
where m is the Gamma function. For m = 1, the Nakagami-m density function fjhj x
reduces to the Rayleigh density function. The Rice density function can be approximated by
the Nakagami-m density function by selecting m 12 =2 1. The bit error
probability has the following closed-form expression (Appendix 10B):
s
"
k #
m1
2k
Nn
SNR=m X
1
; m : integer
1
Pb
1 SNR=m k0 k
41 SNR=m
2 log2 M
"
#
p
m 12
SNR=m
Nn
1
1
1
p
;
Pb
2 F1 1; m 2; m 1;
1 SNR=m
log2 M 2 1 SNR=mm1=2 m 1
m : noninteger
(10:65a)
where 2 F1 is the Gauss hypergeometric function given by the following innite series:
2 F1 a; b; c; z
1
c X
a nb n zn
n!
ab n0
c n
(10:65b)
The equals sign holds for both PSK and QPSK. The bit error probabilities of PSK, QPSK,
MPSK, and MQAM can be found by substituting the appropriate expressions for SNR into
the above equation.
SNR
= 4SNR. On the other hand, the bit error probability
approximated as Q 2SNR e
of a slow-at fading channel decreases inversely with SNR. This results in very poor
performance, particularly for Rayleigh fading. If we consider the case of PSK at SNR =
10.5 dB, the bit error probability in an AWGN channel is about 106 . The same SNR yields a
bit error probability of 0.02 in a slow-at Rayleigh fading channel. What is the reason for
such a poor performance? Let us look at the instantaneous SNR of a fading channel, that is,
jhj2 SNR. Since the channel tap jhj is Rayleigh-distributed, it can take on any value from zero
to innity. Thus the probability that the instantaneous SNR can fall below a given value, no
matter how small, is nite. Hence a deep fade can always occur with nite probability and
542
Fading channels
the result is a very high bit error probability on average. On the other hand, the AWGN
channel has no instantaneous SNR, that is, no fades occur.
=SNR
ey dy 1 e=SNR
(10:67)
For numerical illustration, let the threshold be 1 and SNR = 10. This yields a SNR-outage
probability pout 0:1 and a bit error probability Pb 0:08 for both PSK and QPSK. Either
measure shows a very poor performance even for large SNR. There is almost a 10% chance
that the instantaneous SNR is less than 1 (0 dB), and even the average SNR is 10 (10 dB)!
Many communication devices cannot operate with a SNR below 0 dB, therefore it is possible
that the link is cut off entirely during the outage. Hence, if one takes into account the
dynamic range of the receiver, the bit error probability may be much larger than that
predicted by the above analysis.
(10:68)
(10:69)
543
We are interested in a linear estimate ^h a X of h for some complex constant a given the
known pilot symbol sp . The criterion is to minimize Ejh ^hj2 , the mean-square error. The
orthogonality principle states that E^h hY 0, and this yields the constant a as follows:
!
E jhj2 sp
SNRp
sp
a 2
(10:70)
2
E jhj2 s 2 2 SNRp 1 sp
p
2
where SNRp sp =2 2 is the signal-to-noise ratio of the pilot symbol. Thus we have
!
E jhj2 sp
sp
SNRp
^
X
h a X 2
(10:71)
X
SNRp 1 sp 2
E jhj2 s 2 2
p
(10:73)
(10:74)
544
Fading channels
j^
^ h
^e , and " .
^ Both I-noise N1 ^h =^hN1 and Q-noise
where h jhje j , h
h = ^
N2 ^
hN2 have the same variance 2 . The above complex decision variable can be
written in vector form as follows:
X jhjTsi N
(10:75)
where N N1 N2 t , and the transformation matrix T is the phase error rotation matrix
T
(10:76)
The minimum Euclidean distance detector has the channel tap ^h available to it, but not h
or the
error ". Therefore, the detector uses the faulty set of metrics
phase
d x; ^
hsi x ^
hsi ; i 1; 2; . . . ; M , forthe detection of vector si instead of the
correct set of metrics d x; jhjTsi x jhjTsi ; i 1; 2; . . . ; M , for maximum likelihood detection. This results in a higher bit error probability.
p
Eb as
H2
5
^hx
40
(10:78)
H1
This is threshold detection with the threshold set at 0 volts independently of ^h. The
conditional bit error probability of PSK with threshold detection is
q
Pb jhj; " Q
2jhj2 SNR cos "
(10:79)
where SNR Eb =N0 . The bit error probability of PSK is the expected value of Pb jhj; "
with respect to both jhj and ". In practice, the phase of the estimated channel tap is used to
drive a voltage-controlled oscillator (VCO) to correct the phase of the local carrier. The
phase error " between the received signal and the local carrier has the following Tikhonov
density function:
545
(
f" "
0;
otherwise
(10:80)
where 2" is the variance of the phase error ". The bit error probability of PSK in a Rayleigh
fading channel is
1
Pb
2
r!
SNR cos2 "
1
f" " d"
1 SNR cos2 "
(10:81)
q
q
1
2
2
2jhj SNRcos " sin " Q
2jhj SNRcos " sin "
2
(10:82)
1
s!
SNRcos " sin "2
(10:83)
Example 10.10 Special case of I-Q signals with phase error in an AWGN channel
Let us consider the special case of an I-Q signal in a non-fading AWGN channel. The
coherent demodulation is carried out by a local reference carrier that is phase-synchronized
to the received signal via a carrier recovery circuit. Let " be the phase error between the
transmitted symbols and the local carrier symbol. The decision variable vector is obtained as
follows:
X Tsi N
(10:84)
546
Fading channels
where T is the phase error rotation unitary matrix given by (10.76). The bit error probability
is a special case of that of a fading channel and is given by
Nn
dmin "
Pb "
Q
(10:85)
log2 M
2
where the erroneous minimum Euclidean distance dmin " (the detector has no knowledge of
the phase error) is given by
Tsi sj 2 kTsi si k2
dmin " min di;j " min
(10:86)
s i s j
i;j
i;j
p
For PSK, we get dmin " 2 Eb cos " and 2 N0 =2; therefore, the bit error probability of
PSK is given by
dmin "
Pb " Q
2
(10:87)
p
Q 2 SNR cos "
where
SNR
Eb
N0
(10:88)
For " 22:5 the loss of SNR is 0.7 dB, which shows that PSK is rather resistant to phase
error.
0:924 0:383
T
0:383 0:924
(10:91)
547
Furthermore,
kTs1 s1 k2 0:305Es =2;
(10:92)
Therefore,
d1;2 1:08
p
p
Es =2; d1;3 d1;4 2:6 Es =2
(10:93)
(10:94)
p
When there is no phase error, that is, " 0 we have dmin 0 2 Es =2, which is the
minimum Euclidean distance of QPSK. Thus the phase error " 22:5 reduces the
minimum Euclidean distance by a factor of 1.85 relative to the case where there is no
phase error. This results in a loss of SNR by a factor of 1:852 or 5.4 dB. This large loss of
SNR is mostly due to the crosstalk between the I- and Q-channels. Thus, it is necessary to
keep the phase error as small as feasible, particularly in MQAM, which is very sensitive to
the phase error.
548
Fading channels
he j2ftsL(t)
+n(t)
s(t) + noise
t = Ts
Complexvalued
MF
2/Tsej(2fct+)
Figure 10.9
Demux
Mapper
si si
and
detector
Symbolto-bit
converter
{0, 1}
Pilot
p
Ts 0
channel tap is hp hp e jp. The
Doppler
channel tap of the pilot symbol is
hp hp p hp e jJp ; where hp hp p and Jp p p . The received lth data
symbol si;l after the pilot symbol (which corresponds to hypothesis Hi) is hl si;l hl l si;l .
l1T
The complex Doppler factor of the lth data symbol si;l is l jl jej l T1s lTs s e j2ft dt,
and the corresponding channel tap is hl jhl je jl . The Doppler channel tap of the lth data
symbol si;l is hl hl l jhl je jJl ; where jhl j jhl jjl j and Jl l l . The pilot symbol
plus noise Yp and the lth data symbol plus noise Yl are respectively given as
Yp h p s p N
Yl hl si;l Nl
(10:95)
1 l1Ts j2ft
sin fTs j2fTs l1=2
l
(10:97)
e
dt
e
Ts lTs
fTs
549
The Doppler loss jl j and the Doppler phase error l are respectively given as
sin fTs
; 2fTs l 1=2
(10:98)
j l j
fTs l
Using (10.98) with l = 0, the corresponding Doppler loss p and Doppler phase error p for
the pilot symbol are
sin fTs
p
(10:99)
fT ; p fTs
s
The differential Doppler phase error between the lth data symbol and the pilot symbol is
l p 2fTs l, and increases with time. Thus the Doppler phase error is compounded
and needs to be corrected.
To illustrate this, consider a symbol rate of 14.4 ksps, and a Doppler shift of 180 Hz. The
Doppler loss is 0.999, which amounts to virtually no loss for the amplitude of the signal. The
differential Doppler phase errors for the fth and tenth symbols after the most recent channel
update are 0.39 and 0.78 radians or 22:5 and 45, respectively.
550
Fading channels
phase error and the l 1th symbols estimated Doppler phase error. The bit error probability
of decision-feedback demodulation is now dependent on "l instead of "l . The former is much
smaller than the latter hence yielding a smaller error probability. The phase jitters of
"^l1 and "l determine the accuracy of the tracking algorithm. The differential Doppler
phase error should remain fairly constant over a few symbols. If the previous symbol si;l1
was detected incorrectly there may be a spike in the post-estimated ^
"l relative to the previous
^
^
"l1 . If this happens "^l2 should be used in recalculating
Xl in (10.100) instead
of "^l1 . Also
in the rst equation of (10.100) one could use ^hl1 =^hl1 instead of ^hp =^hp , but the pilot
symbol is commonly transmitted with higher energy, hence its estimated phase is more
"l can be calculated and used to
accurate. Note that once si;l is detected, the post-estimated ^
^
^
"l needed for the next sample, Xl1 Xl1 ej^"l .
evaluate the "^l "l1 ^
If the differential Doppler phase error per symbol is greater than the tolerable limit, further
correction may be implemented by using the following sample:
X^l Xl ej2^"l1 ^"l2 ^hp =^hp Yl ej2^"l1 ^"l2
(10:101)
jhl je j"l si;l N l ej2^"l1 ^"l jhl je j"l ^"l1 ^"l1 ^"l si;l N^ l
jhl je j"l ^"l1 si;l N^ l
where ^
"l1 "^l1 "^l2 is the post-estimated differential Doppler phase error of the
l 1th symbol. This is the double-differential decision-feedback algorithm (DD-DF).
The decision-feedback algorithm can be carried out in practice using a voltage proportional
^ p 2^
^ p "^l1 D-DF or J
"l1 "^l2 DD-DF as input to a
to the phase J
voltage-controlled oscillator (VCO) to correct the phase error between the received signal
and the local carrier at the VCO output.
The complex decision variables in (10.96), (10.100), and (10.101) can be written in twodimentional vector form as follows:
X l jhl jTl si;l Nl
The transformation matrix Tl is the phase error rotation matrix given by
cos e l sin e l
Tl
sin e l
cos e l
(10:102)
(10:103)
551
2
2
jhl jTl si;l ^hl1 sj;l jhl jTl si;l ^hl si;l
^
d i;j hl1 ; jhl j; e l
^hl1 si sj
(10:107)
Therefore, we get
!
^hl1 ; jhl j; e l
d
i;j
Pr si;l ! sj;l jHi ; ^
hl1 ; jhl j; e l Q
2
(10:108)
PrHi Q
log2 M i1 j2
2
hl1 ; jhl j; e l
P b ^
(10:109)
Knowing the statistics of the fading channel (Rayleigh, Ricean, Nakagami) and the Doppler
error we can evaluate the bit error probability as follows:
Pb E Pb ^hl1 ; jhl j; e l
(10:110)
For a slow fading channel where the channel taps remain constant over
symbol
several
times, and in addition where the signal-to-noise ratio is high, we can set ^hl1 jhl j in the
bit error probability expression above and obtain a simpler expression.
552
Fading channels
10.5 OFDM
In OFDM systems such as the IEEE 802.11a,g, 802.16, and 802.16e the pilot symbols are
transmitted simultaneously with data symbols in the pilot subcarriers. For high mobility
applications such as 802.16e the channel update is instantaneous and no Doppler compounded effect occurs. The Doppler effect of a data symbol in pilot-aided demodulation
nevertheless does exist because the pilot subcarriers are at different frequencies to the data
subcarriers. In addition, the Doppler shift may destroy the subcarrier orthogonality and
hence creates intercarrier interference (ICI). Furthermore, although the subcarriers may be
designed to experience slow fading, the entire OFDM channel is a frequency-selective
fading channel. Thus, the phase shifts of subcarriers can be different and pilot symbol-aided
demodulation still results in phase errors in the decision sample because of the phase rotation
of the signal space of each subcarrier.
553
10.5 OFDM
st Re ^sL te j2fc t
("
)
#
N
1
X
j2fc t
Re
sL lpt lTs =N e
(10:111)
l0
Now assume that the signal encounters a Doppler effect with Doppler frequency f . The
carrier frequency is now fc f and therefore the Doppler shifted OFDM signal is given by
st Re ^sL te j2ft e j2fc t
#
("
)
N
1
X
(10:112)
j2ft
j2fc t
Re
sL le
pt lTs =N e
l0
N
1
X
sL le j2ft pt lTs =N
(10:113)
l0
The A/D converter at the receiver samples this complex envelope with sample time nTs =N to
produce the set of N samples for the DFT. We have
^sL n
N
1
X
sL le j2nfTs =N pn lTs =N
l0
N 1
X
sL le j2nfTs =N n l
(10:114)
l0
sL ne j2nfTs =N ; n 0; 1; . . . ; N 1
If the Doppler shift f is known via frequency offset estimation, the phase rotation due to the
Doppler effect can be corrected via the multiplication of ^sL n with ej2nfTs =N to yield the
distortionless sample sL n. Substituting the original time-domain samples sL n generated
by the transmitter as given in (6.161) we get
^sL n
1
1 NX
Zk e j2nkfTs =N ; n 0; 1; . . . ; N 1
N k0
(10:115)
N 1
X
^sL nej2kn=N ; k 0; 1; . . . ; N 1
(10:116)
n0
We can substitute ^sL n into (10.116) to see the Doppler effect on a subcarrier symbol. We
have
!
N
1
N1 h
i
X
X
1
j2nfT
=N
j2mn=N
s
^
Zk
Zme
ej2kn=N ; k 0; 1; . . . ; N 1 (10:117)
e
N
n0
m0
Exchanging the order of summation in (10.117) we get
554
Fading channels
1 X
N 1
1 NX
^
Zme j2nmkfTs =N
Zk
N m0 n0
1
N 1
X
1 NX
Zm
e j2nmkfTs =N
N m0
n0
!
N 1
N 1
N
1
X
X
1
1X
j2nfTs =N
Zk
e
Zm
e j2nmkfTs =N
N n0
N m0
n0
m6k
N 1
1 e
1
1X
e j2mkfTs 1
Zk
Zm
N e j2fTs =N 1
N m0 e j2mkfTs =N 1
j2fTs
m6k
e
Zk
N sinfTs =N
N 1
1X
sinm k fTs jmkfTs N 1=N
Zm; k 0; 1; . . . ; N 1
e
N m0 sinm k fTs =N
m6k
(10:118)
The right-hand side of the above equation is obtained via the following identity:
N
1
X
e jN f 1 sinN f=2 jN 1=2
e jnf jf
(10:119)
e
e 1
sinf=2
n0
The rst term on the right-hand side of the above equation is the distorted subcarrier symbol
and the second term is the intercarrier interference (ICI).
Now let us include the channel distortion effect. Denote the complex envelope of the
channel impulse response as
XL1
ht
hit i
(10:120)
i0
where 0 0. To be consistent with the notation in Section 6.18 we use the notation h(i) to
represent a channel tap for OFDM. The complex envelope of the received OFDM signal is
given by
1
yt ht ^sL t
h^sL t d
1 X
L1
1 i0
L1
X
i0
L1
X
i0
1
hi i ^sL t d
hi
1
(10:121)
i ^sL t d
hi^sL t i
555
10.5 OFDM
where ^sL t is given in (10.113). We assume the A/D converter contains the matched lter
p(t), which is a sin x=x pulse shape, therefore the output of the matched lter is just p(t). The
A/D converter samples the OFDM signal every Ts =N seconds to produce the set of N
samples after discarding the cyclic prex
yn
L1
X
hi^sL
i0
L1
X
hi
i0
nTs
i
N
N 1
X
l0
j2nfTs =N
L1
X
hie
j2f i
i0
e j2nfTs =N
L1
X
N 1
X
(10:122)
sL lpn lTs =N i
l0
~
hi~sL n; i; n 0; 1; . . . ; N 1
i0
where
~hi hiej2f i
~sL n; i
N 1
X
(10:123)
sL lpn lTs =N i
(10:124)
l0
Note that (10.121) reduces to (10.114) when there is no channel distortion, that is, when
ht t. If we take the DFT of y(n) we obtain the frequency domain samples which are
distorted and contain ICI. We can eliminate ICI with Doppler correction before the DFT, that
P ~
is, multiply y(n) with ej2nfTs =N to yield ~yn L1
sL n; i and then take the DFT.
i0 hi~
The resulting frequency-domain samples still contain distortion because of the phase
rotation of the channel taps h(i) and the ISI in the time-domain samples sL l via the sum
factors in (10.124). This type of distortion cannot be corrected and results in degradation.
Let us consider the case when the channel can be modeled as a FIR lter as discussed in
Section 6.18. In this case we have L channel taps with delay i iTs =N. Again assuming
that the Doppler shift has been corrected as described above, we obtain the following timedomain samples:
L1
N 1
X
X
~
~yn ynej2nfTs =N
sL lpn lTs =N i
hi
i0
L1
X
i0
L1
X
~
hi
N 1
X
l0
sL lpn l iTs =N
l0
L1
X
i0
~hi
N 1
X
l0
sL ln l i
(10:125)
~
hisL n iN
i0
~
hn sL n; n 0; 1; . . . ; N 1
Taking the DFT of ~yn we obtain the distorted frequency-domain samples given as
556
Fading channels
~
~
Zk
HkZk;
k 0; 1; . . . N 1
(10:126)
~
where Hk
is the DFT of ~
hn, that is,
~
Hk
L1
X
~
hnej2kn=N
n0
L1
X
hnej2nkfTs =N
(10:127)
n0
~
In the case of no Doppler shift, that is, f 0 we have Hk
Hk, where
PL1
j2kn=N
Hk n0 hne
is the DFT of h(n), and we obtain the frequency-domain
~
samples Zk
HkZk; k 0; 1; . . . N 1, which is the result derived in
Section 6.18.
Returning to the general case in (10.122) and assuming that the Doppler shift has been
corrected as described above we obtain the following time-domain samples:
~yn ynej2nfTs =N
L1
X
~
hi
i0
~
h0sL n
L1
X
~
hi
i1
N 1
X
sL lpn lTs =N i
l0
N 1
X
(10:128)
sL lpn lTs =N i ; n 0; 1; . . . ; N 1
l0
Taking the DFT of ~yn we obtain the distorted frequency-domain samples as follows:
(
)
L1
N
1
X
X
~
~
~
hi
sL lpn lTs =N i ; k
Zk
h0Zk DFT
(10:129)
i1
l0
0; 1; . . . ; N 1
Even in the case of no Doppler shift we note that the second term on the right-hand side of
(10.129) still exists (with ~
hi replaced by h(i)). This means that the cyclic prex, which
removes the ISI between consecutive OFDM symbols, cannot remove the intersample
interference (ISI) within an OFDM symbol as in the case of L channel taps with delay
i iTs =N . To remove the intersample interference when the channel tap delay i is not an
integer multiple of the sampling interval Ts =N , equalization must be employed after the
removal of the cyclic prex.
Low mobility
Wireless IEEE Standards such as 802.11a,g and 802.16 were designed for low mobility and
hence their applications experience virtually no Doppler effects. In this scenario we add
~
noise and have Zk
HkZk Nk ; k 0; 1; . . . ; N 1 for the case of L channel taps
with delay i iTs =N. Note that this expression is identical to its counterpart (10.95) in
single channel demodulation. Therefore, the bit error probability of the mth subcarrier is
given by (10.110) with appropriate replacement of variables:
^ l m; el m
^ l1 m; H
Pb;m E Pb;m H
(10:130)
^ l m, and el m are the estimated DFTs of the subchannel taps of the
^ l1 m, H
where H
l 1th symbol of the mth subcarrier, the subchannel taps of the lth symbol of the mth
557
10.5 OFDM
subcarrier, and the differential phase error (10.100) or double-differential phase error
(10.101) between these two DFTs, respectively. The bit error probability can be averaged
over N subcarriers to yield
Pb
1
1 NX
Pb;m
N m1
(10:131)
In evaluating the bit error probability we need to know, for example, the distribution of
the differential phase errors el m of the subcarriers. One possibility is to assume the
differential phase error to be uniformly distributed in a given range e; e. At a
high signal-to-noise ratio the estimation of subchannel taps can be quite accurate.
0
Hence, the differential
phase
error can be insignicant, el m 0 , and
the
estimated
^
^ l m by
^
subchannel taps Hl1 m can be closely approximated as Hl1 m H
virtue of no Doppler shift. In this case, the bit error probability of low-mobility OFDM
approaches (10.54).
High mobility
~
~
Incorporating noise and we have Zk
HkZk
Nk ; k 0; 1; . . . ; N 1 for the case
of L channel taps with delay i iTs =N. As seen in the above discussion, Doppler-induced
ICI and Doppler loss can have a pronounced effect on OFDM. Once the DFT is carried out to
generate the subcarrier I-Q samples for detection, ICI is added directly to them like AWGN
and cannot be undone. This phenomenon is similar to aliasing in undersampling. Therefore,
ICI can only be cancelled out prior to DFT, namely, in the time domain samples ~sL n. This
requires knowledge of the Doppler shift. The remaining question is how to estimate the
Doppler error. A training sequence can be transmitted in the packet header of each transmitted packet to allow for the estimation of the Doppler shift. Only one chosen subcarrier is
used during this estimation time, that is, the OFDM signal is actually a single-carrier signal.
Data symbols in all other subcarriers are set to zero to avoid ICI. The training sequence
enhances the estimation of the subcarrier Doppler phase error via the decision-feedback
algorithm since all training symbols are known a priori. If a training symbol is detected
incorrectly, it can be corrected via the knowledge of its position in the sequence relative to
the previously correctly detected symbol. Since only a single subcarrier is used, the channel
phase shift is eliminated by the detection process in the previous section, assuming that it is
constant for each symbol time. In practice, the training sequence is fairly short and it is likely
that the channel phase shift remains constant during the entire sequence. The estimated
differential Doppler phase error "^ for each training symbol can be averaged out for the
entire training sequence. Alternatively, the average differential Doppler phase error "^ can be
used to adjust the phase of the local in-phase and quadrature carriers to correct the Doppler
effect before the DFT operation is carried out.
In the general case where the channel tap delay i is not an integer multiple of the
sampling interval Ts =N , equalization must be employed to remove the intra-subcarrier
interference from the sequence ~yn immediately after Doppler correction and cyclic prex
removal.
558
Fading channels
SNR
DMPSK
sin2 p
N0
2M
Eb
=4-DQPSK
SNR 0:555
N0
(10:132)
(10:133)
Rayleigh
For convenience we normalize the channel tap such that E jhj2 1. The channel tap jhj is
Rayleigh-distributed with density function given in (10.56). The corresponding bit error
probabilities are given by the following closed-form expressions:
*
DMPSK, p=4-DQPSK:
1
Pb
log2 M
r!
SNR
1
1 SNR
(10:134)
CSK, MFSK:
Pb
1
X
M =2 M
1
M 1
1k1
k
M 1 k1
1 k 1 SNR
(10:135)
Rice
There is no closed-form expression for Pb of DMPSK and =4-DQPSK with a Ricedistributed channel tap and numerical integration has to be carried out to evaluate
(10.132) and (10.133). The closed-form expression for Pb of CSK and MFSK is given as
559
1
X
M =2 M
1
nSNR
n1 M 1
Pb
1
exp
n
M 1 n1
1 n 1 SNR
1 n 1 SNR
(10:136)
Nakagami-m
When m is an integer, the bit error probabilities have the following closed-form expressions:
DMPSK, p=4-DQPSK:
s
"
k #
m1
2
SNR=m X
1
2k
1
Pb
; m : integer (10:137)
log2 M
1 SNR=m k0 k
41 SNR=m
*
CSK, MFSK:
Pb
1
X
M=2 M
k 1m1
M 1
m
1k1
k
M 1 k1
1 k 1 SNR
m
(10:138)
i1Ts
jhk jA cos2fk f t k
p
2=Ts cos 2fl t dt Nl;I
iTs
jhk jA i1Ts
p
cos2k l=Ts f t k dt Nl;I
(10:139)
2Ts iTs
r
A2 Ts sin2k l f Ts i 1 k sin2k l f Ts i k
jhk j
2k l f Ts
2
Nl;I
p
Similarly, we demodulate this signal by multiplying it by 2=Ts sin 2fl t and integrating
the resulting signal from t iTs to t i 1Ts in order to obtain the lth tones Q-decision
sample:
560
Fading channels
Xl;Q
i1Ts
jhk jA cos2 fk f t k
p
2=Ts sin 2fl t dt Nl;Q
iTs
jhk jA i1Ts
p
sin2k l=Ts f t k dt Nl;Q
(10:140)
2Ts iTs
r
A2 Ts cos2k l f Ts i 1 k cos2k l f Ts i k
j hk j
2k l f Ts
2
Nl;Q
Both Nl;I and Nl;Q are independent Gaussian random variables with zero mean and variance
2 N0 =2. The high-frequency term is insignicant compared to the rst term and is
dropped from the above expression. The decision variable of the lth tone is given by
q
2
(10:141)
Xl Xl;I2 Xl;Q
Thus, given jhk j, Xl is Rice distributed and its density function is given as
m x
xl
2
2
2
l l
fXl xl 2 exl ml =2 I0
(10:142)
where the parameter m2l is the sum of the squares of the means of Xl;I and Xl;Q and is
expressed as
2
2
m2l X l;I X l;Q
(10:143)
sin2 f Ts
jhk j2 Es
2
k l f Ts
where Es A2 Ts =2 is the symbol energy. Note that m2k jhk j2 Es sin f Ts =f Ts 2 . By
using the results in Section 7.9 we can calculate the conditional symbol error probability
2
3
of MFSK. We get
1 Y
M xk
6
7
fXl xl dxl 5fXk xk dxk
Pe jhk j; f 1
4
0
l1
l6k
2
3
1 Y
1
M
6
7
1
1
fXl xl dxl 5fXk xk dxk
4
0
(10:144)
xk
l1
l6k
The above integration can be expressed in term of the Marcum Q-function, which is dened
as follows:
b
1
2
2
2
2
xex a =2 I0 ax dx 1 xex a =2 I0 ax dx
Qa; b
b
ea b =2
2
1 n
X
a
n0
1 ea
b2 =2
In ab
1 n
X
b
n1
In ab; b4a40:
(10:145)
561
Therefore, we have
Pe jhk j; f 1
1 Y
M h
m x i
l
k
;
1Q
fXk xk dx
0 l1
(10:146)
l6k
The inequality in the above expression results from considering the worst-case scenario that
is encountered by the middle tone at k M =2. The bit error probability for MFSK is
obtained via the following expression:
!
M
M =2
1X
Pb f
EPe jhk j; f
(10:147)
M 1 M k1
Since the difference in Doppler shifts for adjacent tones is negligible, the Doppler shift f can
be calculated from (10.143) in the case of small noise by establishing the ratio r2 m2k =m2l ,
called the signal-to-intertone interference ratio. For the worse-case scenario we consider
two adjacent tones and let l k 1. For a given value of r2 we obtain two Doppler shift
values, one negative and one positive, as follows:
f
1
1
m2
; f
; r2 2 k
r 1Ts
r 1Ts
mk1
(10:148)
The sign ambiguity of the Doppler shifts can be resolved if the movement of both transmitter
and receiver is known (moving toward or moving away from each other). The global
positioning system (GPS) can supplement the receiver with the sign information. The
Doppler shift can be employed by a tracking phase-locked loop (PLL) to demodulate the
signal and cancel out the Doppler effect. In practice, the noisy ratio r2 is obtained as
2
. The frequency jitter due to noise determines the accuracy of the estimate
r2 Xk2 =Xk1
of the Doppler shift.
The sign ambiguity of the Doppler shifts can also be resolved via the use of
dual demodulators with a PLL tracking Doppler shift f and another PLL tracking
f using the following decision-feedback selection algorithm (DFS). Suppose the ith
symbol in the symbol stream represented by tone fk was correctly detected. The Doppler
shifts f and f are then estimated as described in (10.148). The dual demodulators
demodulate the next i 1th symbols (two of them) represented by tone fn . The values
of Xn2 f and Xn2 f are compared and the i 1th symbol corresponding to the larger
value is selected. Suppose the correct Doppler shift is f f and assume that the signal-tonoise ratio is high, then the decision variables for the dual demodulators are
Xn2 f m2n f jhn j2 Es sin2 f f Ts = f f Ts 2 jhn j2 Es
and
Xn2 f m2n f jhn j2 Es sin2 f f Ts = f f Ts 2 5jhn j2 Es :
The decision is to select the symbol corresponding to Xn2 f :
The above analysis for MFSK applies equally well to all orthogonal signals, such as CSK
employed in the reverse link of IS-95, MSK, and DPSK, although (10.148) differs for
different modulation techniques.
562
Fading channels
m2k1
m2n f
m2k
m2n f
Speed
(km/h)
Doppler
shift f (Hz)
jhk j Es
jhk j Es
f (Hz)
f (Hz)
jhn j Es
jhn j2 Es
500
500
463
926
0.023
0.046
0.883
0.597
6.18
3.59
334
523
463
926
0.69
0.25
1
1
i1Ts
p
2=Ts wk t cos 2fc t dt Nk;I
iTs
Z
jhk jA i1Ts
p
cos2ft k dt Nk;I
2Ts iTs
r
A2 Ts sin2fTs i 1 k sin2fTs i k
j hk j
Nk;I
2fTs
2
(10:149)
563
i1Ts
p
2=Ts wk t sin 2fc t dt Nl;Q
iTs
jhk jA i1Ts
p
sin2ft k dt Nl;Q
2Ts iTs
r
A2 Ts cos2fTs i 1 k cos2fTs i k
j hk j
Nl;Q
2
2fTs
(10:150)
Both Nl;I and Nl;Q are independent Gaussian random variables with zero mean and variance
2 N0 =2. The high-frequency term is insignicant compared to the rst term and is dropped
from the above expression. The decision variable of the kth Walsh function is given by
q
2 X2
Xk Xk;I
(10:151)
k;Q
Thus, Xk is Rice distributed and its density function is given as
m x
xk
2
2
2
k k
fXk xk 2 exk mk =2 I0
(10:152)
where the parameter m2k is the sum of the squares of the means of Xk;I and Xk;Q and is
expressed as
2
2
m2k X k;I X k;Q
(10:153)
sin2 fTs
j hk j 2 E s
2
fTs
p
where A2 Ts=2 is the symbol energy.
Let us now establish the decision variable of the lth Walsh function wl t. In the case of
zero Doppler shift, this decision variable represents the envelope of Gaussian noise and is
Rayleigh distributed. We have
Xl;I
i1Ts
iTs
jhk jA
p
2Ts
jhk jA
p
2Ts
p
2=Ts wl t cos 2fc t dt Nl;I
(10:154)
wj t cos2ft k dt Nl;I
iTs
Note that the product of two Walsh functions in the same set is another Walsh function in the
same set (closure property). Without loss of generality we choose the Walsh function that
represents the sequence of M/2 zeros followed by M/2 one or, in term of 1 amplitudes, the
sequence of M/2 plus one amplitudes followed by M/2 minus one amplitudes, namely,
564
Fading channels
wl 11; . . . 1 ; 1 1; . . . 1
|{z} |{z}
M =2
M =2
This is the (M/2 + 1)th row of the Hadamard matrix. Thus we get
jhk jA i1Ts
Xl;I p
wj t cos2ft k dt Nl;I
2Ts iTs
"
#
i1Ts
jhk jA i1=2Ts
p
cos2ft k dt
cos2ft k dt Nl;I
(10:155)
2Ts iTs
i1=2Ts
3
r2 sin2fTs i1=2k sin2fTs ik
A2 T s 4
2fTs
5 Nl;I
jh k j
2fTs i1=2k
2
sin2fTs i1k sin
2fTs
r
2
A Ts 2 sin2fTs i 1=2 k sin2fTs i k sin2fTs i 1 k
jh k j
2fTs
2
Nl;I
p
Similarly we demodulate this signal by multiplying it with
2=Ts wl t sin 2fc t and
integrating the resulting signal from t iTs to t i 1Ts in order to obtain the lth
Walshs Q-decision sample:
i1Ts
p
Xl;Q
jhk jAwk t cos2 fc f t k 2=Ts wl t sin 2fc t dt Nl;Q
iTs
jhk jA i1Ts
p
wj t sin2ft k dt Nl;I
(10:156)
2Ts iT
r s
Consequently, we obtain
2
2
m2l X l;I X l;Q
jhk j2 Es
(10:157)
The bit error probability of CSK is identical to that of MFSK discussed in the previous section.
The Doppler shift f can be calculated in the case of small noise by establishing the ratio
2
r m2k =m2l , which is called the signal-to-interchannel interference ratio. For a given value
of r we obtain two Doppler shift values, one negative and one positive, as follows:
!
!
2
1
2
1
1
1
p ; f
p ;
cos
cos
f
Ts
Ts
1 1=r2
1 1=r2
(10:158)
m2
r2 k2 ; wl 11; . . . 1 ; 1 1; . . . 1
|{z} |{z}
ml
M =2
M =2
565
m2l
m2k
m2n f
m2n f
Speed
(km/h)
Doppler
shift f (Hz)
jhk j Es
jhk j Es
f (Hz)
f (Hz)
jhn j Es
jhn j2 Es
500
500
463
926
0.086
0.287
0.883
0.597
3.2
1.44
463
926
463
926
0.596
0.073
1
1
The sign ambiguity of the Doppler shifts can be resolved if the movement of both transmitter
and receiver is known (moving toward or moving away from each other). The global positioning system (GPS) can supplement the receiver with the sign information. The Doppler
shift can be employed by a tracking phase-locked loop (PLL) to demodulate the signal and
cancel out the Doppler effect. In practice, the noisy ratio r2 is obtained as r2 Xk2 =Xl2 . The
frequency jitter due to noise determines the accuracy of the estimate of the Doppler shift.
The sign ambiguity of the Doppler shifts can also be resolved via the use of dual
demodulators with a PLL tracking Doppler shift f and another PLL tracking f using the
following decision-feedback selection algorithm. Suppose the ith symbol in the symbol
stream represented by Walsh function wk t was correctly detected. The Doppler shifts f
and f are then estimated as described in (10.158). The dual demodulators demodulate the
next (i + 1)th symbols (two of them) represented by Walsh function wn t. The values of
Xn2 f and Xn2 f are compared and the (i + 1)th symbol corresponding to the larger value
is selected. Suppose the correct Doppler shift is f f and assume that the signal-to-noise
ratio is high, then the decision variables for the dual demodulators are
Xn2 f m2n f jhn j2 Es sin2 f f Ts = f f Ts 2 jhn j2 Es
and
Xn2 f m2n f jhn j2 Es sin2 f f Ts = f f Ts 2 5jhn j2 Es :
The decision is to select the symbol corresponding to Xn2 f :
566
Fading channels
a symbol time while Walsh functions may change phases every chip time. This effect causes a
larger leakage of signal energy onto other Walsh channels when Doppler shift is present and
leads to larger signal-to-interchannel interferences.
M h
X
i
dk cI t d^k cQ t wk t
(10:159)
M h
i
X
d^k cI t dk cQ t wk t
(10:160)
k1
sL;Q t
k1
M
X
d k ctwk t
(10:161)
k1
567
sL tc t
(10:162a)
The high-frequency term is ltered out, and the following operation results:
sL tc t sL tcI t jcQ t sL;I t jsL;Q tcI t jcQ t
sL;I tcI t sL;Q tcQ t jsL;Q tcI t sL;I tcQ t
The I-value is given by
sL;I tcI t sL;Q tcQ t
M h
X
(10:162b)
i
dk c2I t d^k cI tcQ t wk t
k1
M h
i
X
d^k cI tcQ t dk c2Q t wk t
k1
M
X
dk wk t
k1
M
X
(10:163)
dk wk t
k1
M
X
dk wk t
k1
M h
X
i
d^k c2I t dk cI tcQ t wk t
k1
M h
i
X
dk cI tcQ t d^k c2Q t wk t
k1
M
X
d^k wk t
k1
M
X
M
X
(10:164)
d^k wk t
k1
d^k wk t
k1
The I- and Q-bits contain no I-Q crosstalk interference with complex spreading and
despreading. Figure 10.10 shows the modulator for the complex spreading orthogonal
covering signal. Figure 10.11 shows the mathematically equivalent complex modulator
where the lines are used for complex operations.
Doppler analysis
The received quadrature orthogonal covering signal can be represented in an arbitrary
symbol interval lTs 5t l 1Ts as follows:
568
Channel 1
Fading channels
Mapping
01
1 1
Repeat
N times
Repeat
M times
cI
Pulse
shape
w1
cos 2 fct
Repeat
N times
Mapping
01
1 1
Repeat
N times
Mapping
01
1 1
cQ
sin 2 fct
Pulse
shape
w1
Mapping
01
1 1
Repeat
N times
Repeat
M times
Channel M
Figure 10.10
Repeat
N times
Repeat
M times
Channel 1
w1
w1
e j2fct
Channel M
Figure 10.11
rt jhjA
M h
X
i
dk cI t d^k cQ t wk t cos 2 fc f t
k1
jhjA
M h
X
i
d^k cI t dk cQ t wk t sin 2 fc f t nt ;
(10:165)
k1
lTs 5t l 1Ts
where dk and d^k are the lth in-phase and quadrature bits in the kth channel normalized to the
smallest symbol energy E A2 Ts =2, respectively. The I- and Q-channel PN functions are
cI(t) and cQ(t), respectively. The phase represents the channel phase shift (the phase of the
569
channel tap h jhje j ) and the arbitrary phase of the carrier. We assume the channel tap is
time-invariant in a symbol time, that is, the channel fading rate is much smaller than the
symbol rate. The phase represents the phase error between st and the local carrier
that translates it to baseband. We assume that noise n(t) has the power spectral density
N0 =2 (W/Hz). In terms of the complex envelope representation we have
(10:166)
rt Re rL t e j2fc t nt
where the complex envelope rL t is given by the following expression:
rL t Ahe j2ft
M
X
d k ctwk t
(10:167)
k1
(10:168)
With the aid of the pilot symbol, the channel tap h is available
pand is used to demodulate the signal
rt to complex baseband via a complex local carrier 2=Ts h =jhjej2fc t. Next, the complex
despreading process is carried out by multiplying the complex baseband signal by c t=2,
where jctj2 2. Then, the dechannelization process is carried out to separate individual
channels via the multiplication of the despreading baseband signal with wm t, where
w2m t 1. The signal processing part is completed via the integration of the complex dechannelized signal over a symbol time to obtain the predetection complex sample Xm . We have
l1Ts r
M
X
1
jhje j2ft
A
d k wk twm t dt Nm
Xm
2Ts
lTs
k1
r
l1T
M
s
X
1
dk
jhj
A
wk twm te j2ft dt Nm
(10:169)
2Ts
lTs
k1
jhj
M
X
k;m si;k Nm
k1
2
where si;k (hypothesis Hi ) with energy Ei;k si;k and complex Doppler factor k;m are
given by
r
p
2
A2 Ts
si;k
d k E d k ; Ei;k si;k Ejd k j2 ; E smallest symbol energy
2
(10:170)
1 l1Ts
k;m
wk twm te j2ft dt
Ts lTs
Note that, with zero Doppler shift, the term k;m is 1 for k m and 0 for k 6 m, since the
product of two different complex Walsh functions is another complex Walsh function in
the same set (closure property) and the integration over its period is always zero except
for the rst Walsh function (Walsh sequences have the same number of zeros and ones
except for the rst Walsh sequence). The Gaussian noise variable Nm has zero mean and
variance N0 =2.
570
Fading channels
cos 2 fct
cI(t) w1(t)
sin 2 fct
cQ(t) w1(t)
Ts
Pilot-aided
phase recovery
Minimum
Euclidean
distance
detector
Channel m
Ts
0
Figure 10.12
Pilot-aided demodulation of complex spreading orthogonal covering signal with Doppler tracking of local in-phase
and quadrature carriers.
M
X
hk;m si;k ; Nm Im Nm
(10:171)
k1
k6m
where hk;m jhjk;m , and the term Im is the interchannel interference (ICI) caused by
the Doppler
shift
that destroys the orthogonality of the orthogonal covering signal.
Note that m;m 1 and, therefore, both pilot and data symbols experience losses in
their magnitudes which translate into losses in the signal power. We have seen this
phenomenon before in OFDM and CSK with noncoherent demodulation. The interchannel interference in an orthogonal covering signal is an analogy of the intercarrier
interference (ICI) in OFDM. Since the pilot tone is transmitted on the same carrier as
the data signals, there is no differential Doppler phase error between the received
pilot and data symbols. The bit error probability of a Walsh channel is evaluated in a
similar manner to OFDM, with appropriate replacement of variables. Figures 10.12
and 10.13 show the demodulators for the complex spreading orthogonal covering
signal with Doppler tracking.
571
Pilot-aided
phase
recovery
ej 2 fct
Minimum
Euclidean
distance
detector
c(t) w1(t)
Channel m
Doppler
tracking
Figure 10.13
E si;k k;m
(10:172)
k1
k6m
2
The parameter k;m has been calculated in (10.157) using the Walsh function that
represents the M =2 1th row of the Hadamard matrix as follows:
k;m
(10:173)
The remaining question is how to estimate the Doppler frequency. A training pilot sequence
can be transmitted in the packet header of each transmitted packet to allow for the estimation
of the Doppler shift. Only the pilot Walsh channel is used during this estimation time, that is,
the orthogonal covering signal is actually a single-carrier signal. Data symbols in all other
Walsh channels are set to zero to avoid ICI. The training sequence enhances the estimation
of the Doppler shift via the decision-feedback algorithm since all training pilot symbols are
known a priori. If a training pilot symbol is detected incorrectly, it can be corrected via the
knowledge of its position in the sequence relative to the previously correctly detected pilot
symbol. Since only a single Walsh channel is used the channel phase shift is eliminated by
572
Fading channels
the detection process in (10.100), assuming that it is constant for each symbol time. In
practice the training pilot sequence is short and the channel phase shift remains constant with
high probability during the entire sequence. The differential Doppler phase error for each
training symbol can be estimated and averaged out for the entire training sequence. The
Doppler shift f can be corrected by using the average differential Doppler phase error "^ of the
training sequence to drive a VCO to correct the phase of the local in-phase and quadrature
carriers. This is equivalent to multiplying the right-hand side of (10.167) by ej2ft to obtain
the Doppler-corrected received envelope:
rL t Ah
M
X
d k ctwk t
(10:174)
k1
This is also the complex baseband signal for the low-mobility case without interchannel
interference. Thus, the complex spreading orthogonal covering signal has the same performance as the single channel discussed previously. Mathematically, the decision variable
Xp in (10.169) for a training pilot sequence with the all-zero Walsh function is given as
follows:
l1Ts r
1
Xp
jhje j2ft d p dt Np
A
2Ts
lTs
"r #
(10:175)
A2 Ts
j2f l1=2Ts
jhjjje
d p Np
2
hsp Np
where jj is given in (10.98). The Doppler phase error h=jhj can be estimated from the
knowledge of the pilot symbol sp and used to correct the Doppler effect.
(10:176)
The I- and Q-channel PN functions are cI(t) and cQ(t), respectively. The phase k represents
the phase of the channel tap hk . We assume that the channel tap is time-invariant in a symbol
time, that is, the channel fading rate is much smaller than the symbol rate. We assume that
573
noise n(t) has the power spectral density N0/2 (W/Hz). In terms of complex envelope
representation we have
rt Re rL t e j2fc t nt
(10:177)
where the complex envelope rL t is given by the following expression:
Ajhk j
rL t p e jk e j2ft ct
2
(10:178)
(10:179)
The
signal rt is rst translated to complex baseband by a complex local carrier
p
2=Ts ej2fc t , which is offset by the unknown phase k with respect to rt. Next, the
complex despreading process is carried out by multiplying the complex baseband signal
with c t, where jctj2 2. The signal processing part is completed via the integration of
the complex despreading signal over a symbol time to obtain the M predetection complex
samples Xm . We have
i1Ts r
1
A
Xm
jhk je jk e j2ft ck tcm t dt Nm
2Ts
iTs
r i1T
s
1
jk
jhk je A
ck tcm te j2ft dt Nm
(10:180)
2Ts iTs
M
X
p
Es jhk je jk
k;m Nm
k1
where the symbol energy Es and complex Doppler factor k;m are given by
A 2 Ts
2
1 i1Ts
Es
k;m
(10:181)
Let us consider the case of zero Doppler shift, where the term k;m is 2 for k m. For k 6 m,
k;m is not zero as expected because the I-Q crosstalk is present due to a half-chip delay in the
Q-channel. This I-Q crosstalk is much smaller than 2 and can be ignored. Without the halfchip delay there would be no I-Q crosstalk and k;m would be zero for k 6 m. The complex
Gaussian noise variable Nm has independent I-noise and Q-noise, each with zero mean and
variance N0 . The M decision variables at the input of the maximum detector are
jXm j2 ; m 1; 2; . . . ; M. The bit error probability of noncoherent offset quadrature
DS-CSK is identical to that of noncoherent MFSK and CSK, which were presented in
Section 10.7. The Doppler tracking method is also identical to that of noncoherent CSK.
Figure 10.14 shows the block diagram of the noncoherent demodulator for offset quadrature
DS-CSK.
574
Fading channels
sin 2fc t
Figure 10.14
Ts
( )2
l=1
w1(tTc /2)
cQ(tTc /2)
cI (t) w1(t)
cos 2fct
Ts
Maximum
detector
Symbol/bit
converter
l=M
( )2
w1(tTc/2)
cQ(tTc/2)
575
Interleaved
symbols
Symbols
Interleaver
1
1
2
2
Figure 10.15
5
5
6
6
Modulator
7
7
8
8
9
9
10
10
Figure 10.16
3
3
4
4
Modulated
signal
Signal
processor
Deinterleaver
Diversity
combiner
Detector
VT
X
1 R
i
R V T i1
(10:182)
where R V is the level crossing rate, which is dened as the average rate at which the signal
envelope crosses the threshold V in the positive direction, as indicated in Figure 10.17.
576
Fading channels
Figure 10.17
r_ f V ; r_ d_r
RV
T
0
(10:184)
Rice has shown that the random variables r and r_ are independent and their joint density
function for Rayleigh fading is given by
f r; r_
r r2 =2 2s
e
2s
1
2
2 2 2
e_r =4 s fm
4 3 2s fm2
(10:185)
The rst term on the right-hand side of (10.185) is the familiar Rayleigh density function of
the signal envelope, where Er2 2 2s . The second term contains the parameter fm, which
denotes the maximum Doppler shift experienced by the signal. Therefore, the level crossing
rate is obtained via direct integration as
p
2
(10:186)
RV 2 fm e
577
where
2
V2
2 2s
V
Vrms
2
The parameter 2 is given in term of the rms signal voltage level Vrms
(10:187)
p
Er2 .
VT
X
1R
i
T i1
1
1 V r r2 =22s
Prr V
e
dr
RV
R V 0 2s
e 1
p
2 fm
2
(10:188)
(10:189)
We note that when there is no Doppler shift, that is, fm 0, then the level crossing rate is
zero (no level crossings). Consequently, the average fade duration is also zero (not innite as
in (10.189), which is valid only for RV 6 0 ). This implies that the channel is time-invariant
(the coherence time is innite) and interleaving would not be effective. Thus, time diversity
may not be effective in time intervals where the channel experiences no fading.
578
Fading channels
(10:190)
The complex Gaussian random variable Nl has a variance 2 2 . We can express L in vector
form as follows:
(10:191)
Y hsi N
where Y Y1 Y2 . . . YL t ; h h1 h2 . . . hL t ; and N N1 N2 . . . NL t .
Now assume perfect channel estimation, then the sufcient statistic for coherent demodulation is h =khkY. We use the notation h for the conjugate and transpose of vector h. We
have the following MRC decision variable:
h
X
hsi N
khk
(10:192)
h
N
khksi
khk
The complex Gaussian noise h =khkN has variance 2 2 . Note that the above operation is
L
L
L
P
P
P
equivalent to khkX
hl Yl
jhl j2 si hl Nl as mentioned above (co-phasing and
l1
l1
sl1
L
P
jhl j2 still preserves the sufcient statistic.
weighting). The normalization with khk
l1
The instantaneous MRC output signal-to-noise ratio SNR0;i , given the symbol si , is
P
P
SNR0;i khk2 jsi j2 =2 2 l jhl j2 jsi j2 =2 2 l SNRl;i , where SNRl;i jhl j2 jsi j2 =2 2
is the instantaneous signal-to-noise ratio of the pre-combining lth copy of symbol si . Thus,
MRC achieves the maximum output signal-to-noise ratio, which is the sum of L input signalto-noise ratios (a fact that can also be veried by CauchySchwarz inequality). The name
maximal ratio combining is the truncated version of maximal signal-to-noise ratio combining.
Assume perfect channel estimation, and use the mapping si ! si for a complex variable
to a two-dimensional vector, then the conditional pair-wise error probability between two
vectors khksi and khksj is
579
khksi sj
Pr khksi ! khksj Q
2
(10:193)
The conditional bit error probability for the Gray-coded signal set fsi gM
i1 is given by the
following approximation:
Nn
khkdmin
Pb khk
Q
(10:194)
log2 M
2
where dmin min jjsi sj jj is the minimum Euclidean distance of the signal set, 2 N0 =2,
i;j
and Nn is average number of symbols at the minimum Euclidean distance dmin or the average
number of nearest neighbors. Here N0 =2 is the power spectral density of noise. Knowing the
statistics of the fading channel (Rayleigh, Rice, Nakagami) we can evaluate the bit error
probability as follows:
Nn
khkdmin
Q
Pb EPb khk E
log2 M
2
q
(10:195)
Nn
E
Q
2khk2 SNR
log2 M
where the diversity symbol signal-to-noise ratio SNR and the positive constant , which
depends on the modulation, are given by
Es
Es
1 dmin 2
SNR
; SNR
(10:196)
2 2
N0 LN0
The diversity symbol energy Es is given by Es Es =L.
Rayleigh
When the iid subchannel tap magnitudes jhi j are Rayleigh distributed with normalized
L
P
mean-square value E jhi j2 1, the combining channel tap magnitude khk2
jhl j2
l1
yL1 ey
L 1!
(10:197)
The corresponding bit error probability is given by the following two equivalent closedform expressions from Appendix 10B:
"
k #
L1
X
Nn
1 2
2k
Pb
1
k
2 log2 M
4
k0
L1
Nn
1 LX
1 l
L1l
Pb
l
2
2
log2 M
l0
(10:198a)
(10:198b)
580
Fading channels
where
r
SNR
(10:199)
1 SNR
p
PSK: the minimum Euclidean distance is dmin 2 Eb , where the diversity bit energy is
Eb . With the noise variance 2 N0 =2, we obtain SNR Eb =N0 Eb =LN0 .
Furthermore, Nn 1, M = 2, therefore the bit error probability for PSK is
"
k #
L1
X
1
1 2
2k
Pb
1
PSK
(10:200a)
k
4
2
k0
L1
1 LX
1 l
L1l
PSK
(10:200b)
Pb
l
2
2
l0
p
QPSK: the minimum Euclidean distance is dmin 2Es , where the diversity symbol energy
is Es . With 2 N0 =2, we obtain SNR Es =2N0 Es =2LN0 . Furthermore, Nn 2,
M = 4, therefore the bit error probability for QPSK is identical to that of PSK as expected
"
k #
L1
X
1
1 2
2k
1
Pb
QPSK
(10:201a)
k
2
4
k0
1 l
Pb
QPSK
2
l0
p
MPSK: the minimum Euclidean distance is dmin 2 Es sin=M ,
diversity symbol energy is Es . Using the noise variance 2 N0 =2,
Furthermore,
SNR Es =N0 sin2 =M Es =LN0 sin2 =M.
therefore the bit error probability for MPSK is given by
"
k #
L1
X
1
1 2
2k
1
Pb
MPSK; M 42
k
4
log2 M
k0
1
2
L X
L1
L1l
l
(10:201b)
where the
we obtain
Nn 2,
(10:202a)
L1
2
1 LX
1 l
L1l
MPSK; M 42
(10:202b)
l
log2 M
2
2
l0
p
MQAM: the minimum Euclidean distance is dmin 6Es =M 1. Using 2 N0 =2,
we obtain p
SNR
3Es =N0 =2M 1 3Es =LN0 =2M 1. Furthermore,
p
L1
4 4= M 1 L X
1 l
L 1 l
Pb
MQAM
l
log2 M
2
2
l0
(10:203b)
581
Rice
When the independent subchannel tap magnitudes jhi j are Rice distributed with normalized
L
P
mean-square value E jhi j2 1, the combining channel tap magnitude khk2
jhl j2 has
l1
(10:204)
where was rst identied in (10.63). The bit error probability can be evaluated via direct
integration.
Nakagami-m
When the independent subchannel tap magnitudes jhi j are Nakagami-m distributed with
normalized mean-square value E jhi j2 1, the combining channel tap squared magnitude
k h k2
L
P
l1
fjjhjj2 y
(10:205)
(10:206)
Nn
SNR
Nn
1
Pb
1
1 1
1 SNR
2 SNR
2 log2 M
2 log2 M
(10:207)
Nn
1
log2 M 4 SNR
For the diversity signal we make the following approximations for the parameters involving
in (10.198):
582
Fading channels
1
1
1
1; and
2
2
4SNR
(10:208)
Furthermore,
L1
X
L1l
l
l0
2L 1
L
(10:209)
Note that the time diversity signal has the transmission symbol rate equal to L times that of
the signal without diversity. Thus the diversity symbol energy is only 1/L times the nondiversity symbol energy or equivalently, SNR SNR=L. Thus, the bit error probability of
an I-Q signal with diversity at high SNR is given by
L
2L 1
Nn
1
Pb
log2 M
4 SNR
L
(10:210)
L
Nn
1
L 2L 1
L
4 SNR
log2 M
L
Thus, with diversity, the bit error probability decreases inversely with the Lth power of the
input signal-to-noise ratio. On the other hand, without diversity, the bit error probability
decreases inversely with the input signal-to-noise ratio.
For illustration consider the case when SNR = 100 (20 dB), L = 3, Nn 2; M 4
(QPSK). We have Pb 2:5 103 for no diversity as compared to Pb 3:9 106 for a
three-fold diversity with MRC. In other words, to achieve the same bit error probability the
signal without diversity must have an SNR of about 48 dB instead of 20 dB. The diversity
gain is approximately 28 dB.
SNR
(10:211)
For a slow-at Rayleigh fading channel we can use the 2 -density function with 2L degrees
of freedom. Thus, the outage probability is given by
=SNR
yL1 ey
dy
L 1!
0
L1
X
1
k
1 e=SNR
k! SNR
k0
pout
(10:212)
583
For numerical illustration, we compare the performance of an L-fold time diversity I-Q
signal with SNR to the performance of its non-diversity counterpart with SNR, assuming that
both signals have the same transmitted power and input bit rate. Thus, we have
SNR SNR=L. Let the threshold be 1 and SNR = 11.2 (10.5 dB), and L = 3. This
yields an outage probability pout 2:5 103 as compared to 0.09 for the case where there
is no diversity (see Example 10.5).
(10:213)
Now assume
perfect
channel estimation, then the sufcient statistic for coherent demodulation is hl =jhl j Yl . We have the following pre-combining decision variables:
hl
hl si Nl
j hl j
(10:214)
h
jhl jsi l Nl ; l 1; 2; . . . ; L
j hl j
Both the I- and Q-noise components of hl =jhl j Nl have the same variance 2 . The
selection combiner chooses the decision variable with the largest subchannel tap squared
magnitude:
(10:215)
jhj2 max jh1 j2 ; jh2 j2 ; . . . ; jhL j2
Xl
Nn
jhjdmin
Q
Pb EPb jhj E
log2 M
2
q
(10:217)
Nn
2
E
Q
2jhj SNR
log2 M
584
Fading channels
(10:218)
Fjhl j2 y 1 ey
(10:219)
1
L elz
l
l1
(10:220)
Finally, the bit error probability in a Rayleigh fading channel can be evaluated using the
above density function. We get
r!
L
Nn X
SNR
l1 L
1
1
Pb
(10:221)
l
2 log2 M l1
l SNR
For illustration we compare the performance of an L-fold time diversity I-Q signal with SNR to
the performance of its non-diversity counterpart with SNR, assuming that both signals have the
same transmitted power and input bit rate. Thus, we have SNR SNR=L. Consider the case
when SNR = 100 (20 dB), L = 3, Nn 2; M 4 (QPSK), hence SNR SNR=L 100=3.
We have Pb 2:5 103 for no diversity as compared to Pb 2:2 105 for a three-fold
diversity with SC. In other words, to achieve the same bit error probability the signal without
diversity must have an SNR of about 41 dB instead of 20 dB. The diversity gain is approximately 21 dB as compared to 28 dB for MRC in Example 10.19.
(10:222)
Taking the expectation of Pb jhj with respect to jhj2 by using the density function of jhj2 in
the above example we obtain the bit error probability as follows:
585
Pb
L
X
l1
1
l1
L1
X
l
1
L
L1
1l
L
l 2l SNR
l
2l 1 SNR
l0
(10:223)
For illustration consider the case of FSK with time diversity L = 3. The nondiversity signalto-noise ratio is SNR = 100 (20 dB). Thus, the diversity signal-to-noise ratio is
SNR SNR=L 100=3. We have Pb 1=2 SNR 102 for no diversity as compared to Pb 4:6 104 for a three-fold diversity with SC. In other words, to achieve the
same bit error probability the signal without diversity must have an SNR of about 33 dB
instead of 20 dB. The diversity gain is approximately 13 dB.
Coherent EGC
Let us consider the pre-combining samples of L copies of an arbitrary transmitted symbol si
of a coherent I-Q signal at the output of the matched lter. Each copy has a corresponding iid
complex subchannel tap hl . We have
Yl hl si Nl ; l 1; 2; . . . ; L
(10:224)
Now assume
perfect
channel estimation, then the sufcient statistic for coherent demodulation is hl =jhl j Yl . We have the following pre-combining decision variables:
hl
hl si Nl
j hl j
(10:225)
hl
jhl jsi
Nl ; l 1; 2; . . . ; L
j hl j
Both the I- and Q-noise components of hl =jhl j Nl have the same variance 2 . The EGC
decision variable is given by
Xl
L
X
Xl jhjsi N
(10:226)
l1
where jhj jh1 j . . . jhL j and N is a complex Gaussian variable with variance 2L 2 . The
bit error probability is given in (10.194) if the density function of jhj is known. Since jhj is
586
Fading channels
the sum of L iid random variables, its density function is the L-fold convolution of L
identical density functions. There is no closed-form expression for Rayleigh, Rice, or
Nakagami-m cases. We note that coherent EGC does not equal the performance of MRC
because it weights all subchannels equally. Furthermore, there is no reduction in complexity
with respect to MRC because the channel estimation is carried out in the samemanneras
MRC. In fact, the sufcient statistic for MRC is Xl hl Yl , as compared to Xl hl =jhl j Yl
for coherent EGC.
Noncoherent EGC
In this section we consider the application of noncoherent EGC to orthogonal signals such as
MFSK and CSK in a Rayleigh fading channel. From (7.236), the output of the rst
quadrature correlatorsquare
law detector
1;l t was the lthcopy of the transmitted
p
2 given
ps
2
2
signal is X1;l
jhl j Es cos l NI;1;l jhl j Es sin l NQ;1;l . The diversity symbol
energy Es is 1/L times the symbol energy Es, that is, Es Es =L. The outputs of the remaining
2
M 1 quadrature correlatorsquare law detectors are Xj;l2 NI2;j;l NQ;j;l
; j 2; 3; . . . ; M .
The noise random variables NI;i;l and NQ;i;l are all independent Gaussian random variables
with zero mean and variance 2 N0 =2. Note that both jhl j cos l and jhl j sin l are zero2
and Xj;l2 are central chimean Gaussian random variables, therefore we conclude that X1;l
squared random variables given the subchannel tap magnitudes jhl j; l 1; 2; . . . ; L. The
independent outputs of all L noncoherent EGC combiners are
U1
L
X
2
X1;l
; Uj
l1
L
X
Xj;l2
(10:227)
l1
euj =2j ; uj 0
uL1
j
2
2 2j L L 1!
(10:228)
where the parameter 2j is the mean-square value of each component of Xj;l2 and is given as
21 Es =2 2 Es =2 N0 =2; 2j 2 N0 =2; j 2; 3; . . . ; M
(10:229)
For convenience
we normalize the mean-square value of the subchannel tap magnitude to
unity: E jhl j2 1. The conditional symbol error probability given u1 is one minus the
probability that all M 1 variables Uj are less than u1 . Furthermore, the probability that a
variable Uj is less than u1 is
PrUj 5u1
u1
fUj uj duj
1e
u1 =2 2j
L1
X
1
l0
u1
l! 2 2j
!l
(10:230)
587
du1 A
1 e
M 1
l! 2 2j
2 21 L L 1!
0
l0
0
1
#
1 "
L1 l M 1
X
M=2 @
u1
1
u1
L1 u1 =1SNR
1e
du1 A
1
u1 e
l!
M 1
1 SNRL L 1!
0
l0
(10:231)
where SNR Es =N0 Es =LN0 .
L1
X
1 SNRl
L1l
l
2 SNRLl
l0
(10:232)
For illustration consider the case of FSK and time diversity L = 3. The nondiversity signalto-noise ratio is SNR = 100 (20 dB). Thus, the diversity signal-to-noise ratio is
SNR SNR=L 100=3. We have Pb 1=2 SNR 102 for no diversity as compared to Pb 2:2 104 for a three-fold diversity with EGC. In other words, to achieve the
same bit error probability the signal without diversity must have an SNR of about 36.6 dB
instead of 20 dB. The diversity gain is approximately 16.6 dB as compared to 13 dB for
noncoherent SC.
588
Fading channels
Symbols
Figure 10.18
FFH
modulator
Jamming
& fading
channel
Dehopper
& demod
Diversity
combiner
To detector
k 0; 1; . . . ; N 1
(10:233)
For diversity to work the subchannel taps Hk must be uncorrelated. This puts constraints on
the number of subcarriers, the separation between diversity groups, and the length of the
cyclic prex that is employed to remove subcarrier ISI. Consider a frequency selective
589
cos 2fc t
{0, 1}
Z(N 1)
N1
Subcarrier
symbol
mapping
{Z(k)},
repetition,
diversity
interleaving
sL(N 1)
Cyclic
prefix
IFFT
Z(0)
I
+
D/A
converter
OFDM
sL(0)
sin 2fc t
Figure 10.19
cos 2fct
A/D
converter
Remove
cyclic
prefix
FFT
Subcarrier
symbol
demapping,
diversity,
deinterleaving,
MRC
sin 2fct
Figure 10.20
590
Fading channels
subchannel can be lled with N LbN =Lc unused subcarriers, which can be discarded
before combining takes place at the demodulator. After diversity deinterleaving the precombining variables of L diversity subcarriers for the case of low-mobility OFDM are
^
Zk
Hl;k Zk Nl;k ;
l 1; 2; . . . ; L; k 0; 1; . . . ; bN =Lc
(10:234)
Applying MRC to the above L pre-combining variables for each diversity subcarrier
symbol, the bit error probability of the L-fold diversity system is given by (10.194). The
performance of an L-fold diversity OFDM signal with diversity SNR as compared to the
performance of its non-diversity counterpart with non-diversity SNR is the same as that for
time diversity discussed previously, assuming that both signals have the same transmitted
power and input bit rate.
Rake receiver
Multipath-delay diversity that occurs naturally in a frequency-selective fading channel can
be implemented via a special receiver called a Rake receiver. This type of receiver is
particularly suited to a direct sequence spread spectrum signal with a wide bandwidth
covering many coherence bandwidths. Such a wideband channel is modeled as an L-tap
channel with the following impulse response:
L1
X
h; t
hi t i t
(10:235)
l0
where 0 t5 1 t5. . .5 L1 t. We assume that the channel varies slowly so that the
channel taps remain constant over many symbols (the channel coherence time is several
symbols long). Furthermore, the multipath delay spread is assumed to be less than a symbol
time (the channel is underspread) so that ISI affects only the next symbol. These assumptions are valid for most practical wireless systems. Thus, the L-tap channel model is timeinvariant and can be expressed as follows:
h
L1
X
hl l
(10:236)
l0
With the time-varying component t eliminated we can replace the delay-time with t to work
with the received signal conventionally expressed in time t. Hence, we have
ht
L1
X
hl t l
(10:237)
l0
591
sL t d k ct kTs
(10:239)
The complex spreading PN sequence ct with unit amplitude squared pulse shape and chip
time Tc and period NTc (selected to be an integer multiple of Ts ) is expressed in terms of both
in-phase and quadrature PN sequences cI t and cQ t as
ct cI t jcQ t; 05t NTc
(10:240)
For the paths to be resolvable it is necessary for the chip time to be smaller than the smallest
delay between paths. The convolution of sL t and ht results in the received complex
envelope rL t, which contains L copies of each transmitted symbol and its associated ISI
from the preceding and succeeding symbols. We have
rL t A
L1
X
(10:241)
l0
where noise n(t) has the power spectral density N0 =2 (W/Hz). The term Ik1 t represents
L 1 ISI signals from the preceding k 1th symbol and the term Ik1 t represents L 1
ISI signals from the succeeding symbol k 1th, respectively. We have
Ik1 t A
L1
X
(10:242)
hl d k1 ej2fc l ct k 1Ts l
(10:243)
l1
Ik1 t A
L1
X
l1
Any arbitrary mth path of the kth symbol experiences L m ISI signals from the preceding
k 1th symbol and m 1 ISI signals from the k 1th symbol. Assume that the
They
channel taps hl and path delays l are available via perfect channel estimation.
are then used to demodulate the received signal rt Re rL t e j2fc t to L complex
baseband
(corresponding to L paths) via L complex local carriers
p j2fsignals
c t l
2=Ts e
; l 1; 2; . . . ; L. Next, the complex despreading process is carried out
for the interested symbol by multiplying the complex baseband signals with
c t kTs l =2; l 1; 2; . . . ; L, where jctj2 2. This is followed by integrating the
demodulated baseband signals for one symbol time to obtain L pre-combining decision
variables. This demodulation process creates L Rake ngers. The output of the lth Rake
nger for the kth symbol is given as
k1Ts l r
1
Yl
hl d k dt Il Il Il Nl
A
2T
(10:244)
s
kTs l
hl sk Il Il Il Nl
where sk is the unnormalized kth symbol with energy Ek jsk j2 given by
r
p
A2 Ts
d k E d k ; E k j sk j 2 E jd k j2 ;
sk
2
E A2 Ts =2 smallest symbol energy
(10:245)
592
Fading channels
The complex Gaussian noise variable Nl has zero mean and variance 2 2 N0 , where
2 N0 =2 is the variance of the in-phase noise or quadrature noise. The self-interference
Il is referred to as intrapath interference (IPI), and Il and Il are ISI. Given the unnormalized k 1th and k 1th symbols, the expressions for these three parameters are
r
L1
1 k1Ts l X
1
A
Il
hm d k ej2fc m l ct kTs m c t kTs l dt
2 kTs l
2T
s
m0
m6l
k1Ts l
L1
1 X
j2fc m l
hm s k e
ct kTs m c t kTs l dt
2Ts m0
kTs l
m6l
(10:246)
Il
1
2
r
kTs l m X
Ll
1
A
hm d k1 ej2fc m l ct k 1Ts m
2T
s
kTs l
m2
m4l
c t kTs l dt
X
1 Ll1
hm sk1 ej2fc m l
2Ts m1
m4l
kTs l m
kTs l
ct k 1Ts m c t kTs l dt
(10:247)
Il
r
k1Ts l m X
l2
1
A
hm d k1 ej2fc m l ct k 1Ts m c t kTs l dt
2T
s
k1Ts l
m0
m5l
l2
X
1
hm sk1 ej2fc m l
2Ts m0
m5l
k1Ts l m
k1Ts l
ct k 1Ts m c t kTs l dt
(10:248)
In the above integrals the product of two complex PN sequences is dened over their
overlapping interval only. The independent IPI and ISI variables can be approximated as
zero mean complex Gaussian variables with variances 2 2I , 2 2I , and 2 2I , respectively,
where 2I , 2I , and 2I are the variances of the corresponding in-phase interference or
quadrature interference. Combining the three interferences with Gaussian noise and denoting it as Nl Il Il Il Nl , with the total variance as the sum of variances
2 2 2 2I 2 2I 2 2I 2 2 , we obtain L Rake nger outputs for the kth symbol as
follows:
Yl hl sk Nl ; l 1; 2; . . . ; L
(10:249)
Dening the vector h as h h1 h2 . . . hL t and applying maximal ratio combining to the
above L Rake nger outputs we obtain the bit error probability for the Rake receiver of an
I-Q signal as
q
Nn
Pb E
Q
2khk2 SINR
(10:250)
log2 M
593
where the path signal-to-interference-and-noise ratio SINR and the positive constant ,
are given by
SINR
1 dmin 2
2 2
(10:251)
where dmin min jjsi sj jj is the minimum Euclidean distance of the I-Q signal set (as
i;j
usual we use the mapping si ! si to convert complex-valued symbols to twodimentional vector symbols). Note that the path SINR of the Rake receiver is similar
to the diversity SNR discussed previously for controlled time diversity and frequency
diversity such as OFDM. In a controlled diversity method with L copies of a transmitted symbol, the energy going into each diversity symbol is 1/L times the nondiversity symbol energy. On the other hand, the DS-SS transmitter transmits only one
copy of the transmitted symbol (no diversity), and the channel provides diversity for
the receiver in the form of L resolvable paths. Given a transmitted power P, the lth
path has an average power Pl 5P and, hence, the symbol in a path has only a fraction
of the energy of the transmitted symbol. If we assume that all paths have a normalized
unit mean-square value, that is, Ejhl j2 1; l 1; 2; . . . ; L, then each path symbol
has 1/L times the energy of the transmitted symbol. Thus, without the Rake receiver,
one would have a power loss of 10 log L (dB). The bit error probability of the Rake
receiver is the same as that of MRC, with SINR replacing SNR in the appropriate bit
error probability expressions. To evaluate the bit error probability of the Rake receiver
it is necessary to nd the variances of IPI and ISI variables. These variances can be
found using the results developed for CDMA in Chapter 8, assuming that the PN
sequence can be modeled as a random binary sequence (when the period of the PN
sequence, in chips, is much larger than the number of chips per symbol). In practice,
the multipath delay spread is much smaller than the symbol time of a direct sequence
spread spectrum signal in the cellular and PCS bands. Hence, IPI is much larger than
ISI and consequently we have 2I 2I 2I . For this case, we can approximate the
total variance of interference and noise as 2 2I 2 L 1Es =N N0 =2, where
Es Ejsk j2 is the average symbol energy and N is the spread factor. Note that in
time and OFDM diversities with diversity SNR, the performance comparision with a
non-diversity receiver with SNR is carried out with SNR SNR=L. In the case of the
Rake receiver, the performance comparision with a non-Rake receiver is carried out
with both receivers having the same SINR, since the non-Rake receiver just locks
onto only one path. Thus, when channel noise dominates IPI, the performance of the
Rake receiver with SINR approaches the performance of the maximal ratio combiner
as compared to a non-Rake receiver with the same SINR. The bit error probability of
a non-Rake receiver is given by (10.195) with the magnitude of the path channel tap
jhj replacing khk, and this is identical to (10.54) with SINR replacing SNR. Thus, an
additional power gain of 10 log L (dB) besides diversity gain is achieved for a Rake
receiver when compared to a non-Rake receiver. This is not the case for controlled
diversity. Figure 10.21 shows the conceptual complex-valued Rake receiver.
594
Fading channels
Integrator
h1 j2f (t ) c(t )
e
c
1
1
h1
MRC
Detector
Symbol-bit
converter
Integrator
hL j2fc(t ) c(t )
L
e
L
hL
Figure 10.21
(10:252)
The channel taps can be estimated via MMSE estimation (Example 10.6) or direct estimation (Example 10.7).
595
(10:253)
where V 0 ; is unity for isotropic antenna elements and k0 2fc =c 2= is the free
space wave number. To determine the degree of correlatedness between the two waves we
examine the spatial correlation coefcient , which is dened as the cosine of the angle
between the two phasors. We have
Transmitter
Figure 10.22
Receiver
596
Fading channels
Incident wave
Figure 10.23
E V 1 V 2
r
2
2
E jV 1 j E jV 2 j
(10:254)
2
0 0
V 1 ; V 2 ; f ; dd
q
2
2
2
2
0 0 jV 1 ; j f ; dd 0 0 jV 2 ; j f ; dd
2
(10:255)
The function f ; is the joint probability density function of the polar angle and
azimuth angle , respectively. In our analysis we assume that and are independent and
hence f ; f f . Note that 0 jj2 1. A larger value of jj2 indicates a
stronger correlation between the two incident plane waves. On the other hand, when jj2 0
the two incident plane waves are uncorrelated.
2
1
ejk0 d cos cos sin d ejk0 d sin cos f d
0 2 0
(10:256)
597
J0 k0 d sin f d
(10:257)
ejk0 d cos f d
(10:258)
(10:259)
On the other hand, 1 for a vertical array since the waves traveling in the horizontal
plane to arrive at the two vertically separated antennas are identical.
We can also use the Aulin model to assume that the arrival angle of the incident waves is
restricted to the angular region =2 =2 . In this model the density
function of the polar angle is given in (10.43) and the spatial correlation coefcient for a
vertical array is
sink0 d sin
k0 d sin
(10:260)
598
Fading channels
degrees at most. To ensure that all mobile stations at the edge of the cell (with the weakest
signals) can benet from diversity, the inter-element spacing must be large enough for
diversity paths to be uncorrelated. Let us consider a vertical array Aulin model with the
requirement jj2 0:012 or 19 dB. We get k0 d sin 5:6 radians. This implies that
d 5:6=k0 sin . For numerical illustration let 50 , then d = 2m (about 12.8 ).
Thus, the inter-element spacing at the base station antenna array is much larger than interelement spacing at the mobile station antenna array.
Performance
The following examples illustrate the performance of communication systems employing
receive antenna diversity alone or with other controlled diversity techniques.
599
signal with MRC to the performance of its counterpart without diversity assuming both
signals have the same transmitted power and input bit rate. We have L1 L2 3 and L = 9.
Assume that the required bit error probability for the diversity system is Pb 4:2 106 .
Using (10.210) with Nn 2; M 4 for QPSK and L = 9, the L1 -fold diversity signal-tonoise ratio is SNR 3 or 4.8 dB. A non-diversity system with SNR L1 SNR 9 or
9.5 dB would have a bit error probability of 2:6 102 . A three-fold controlled diversity
would require SNR 33:4 or 15.2 dB, and a non-diversity system must have
SNR 48 dB to have Pb 4:2 106 . The L-fold diversity gain is approximately
48 9:5 38:5 dB as compared to 28 dB for the L1 -fold diversity system (Example 10.19).
600
Fading channels
Space-time coding
For Ls symbol times and Lt -fold transmit antenna diversity we obtain a space-time block
code represented by a complex Ls Lt matrix G, which consists of Lt parallel code words
(the columns of the matrix) whose elements are the transmitted symbols. The repetition code
is a special case of the space-time block code with Lt 1, and the code word contains Ls
copies of one transmitted symbol, thus providing a diversity order of Ls . This indicates that
the rate of the repetition code is 1=Ls , which is quite inefcient. A space-time block code
allows the transmission of m symbol copies in each code word of Ls symbols, thereby
increasing the code rate to m/Ls . The diversity order of the space-time code is Lt , the number
Transmitter
Figure 10.24
Receiver
601
of transmit antennas. To separate the symbol copies for combining it is necessary that
the columns of the code matrix G be (complex) orthogonal since the transmitter sends
all Lt code words concurrently over Ls symbol times. Also, the channel is assumed to
remain constant for m symbol times. If we assume perfect channel estimation, then, for
a given transmitted space-time block code matrix G, the sufcient statistics for maximal
ratio combining is dictated by the ability to obtain a matrix H from the vector Gh as
follows:
Gh ! Hs
H H k h k2 I m
(10:261)
where h h1 h2 hLt t is the channel tap vector representing Lt paths; s is the symbol
vector whose m components are the transmitted symbols on their complex conjugates; H is
referred to as the channel tap matrix, an Ls m matrix whose elements are the channel taps
and its complex conjugates and whose structure depends on G; and Im is the m m identity
matrix. The columns of H are orthogonal, a necessary condition for the detection of s to be
decomposed into m separate complex variables. This implies that the columns of G must
also be orthogonal such that G G ksk2 ILt . The matrix H is known by the receiver via
channel estimation and the known structure of G. We use the notation A for the conjugate
and transpose of a matrix A. Note that the vector Gh is the sum of Lt channel tap weighted
code words. Let N N 1 N 2 N Ls t be the Gaussian noise vector whose elements are
independent zero mean complex Gaussian variables with identical variance 2 2 . The premapped vector at the input of the combiner is
Z Gh N
(10:262)
The vector Z is mapped into vector Y by changing some of its elements into their complex
conjugates as dictated by the structure of G such that
Y Hs N
(10:263)
The noise vector N N1 N2 NLs t has independent zero mean complex Gaussian components with identical variance 2 2 . The maximum ratio combiner performs the operation
H Y=khk to obtain the decision vector X:
1
1
H Hs
H N
khk
khk
1
H N
khks
khk
(10:264)
The
complex jointly Gaussian noise vector H =khkN has a diagonal covariance matrix
2
2
E H NN H=khk H ENN H=khk 2 2 H H=khk2 2 2 Im . This implies that
the components of this noise vector are white and have identical variance 2 2 , and hence
are independent. Thus, the decision vector X consists of m independent complex decision
variables Xk , k = 1, 2, . . . , m, that can be detected by m parallel minimum Euclidean distance
detectors.
602
Fading channels
Alamouti code
The Alamouti space-time block code is recommended in the IEEE 802.16 standard for
WMAN. The code is specied by the matrix G whose code words s1 s2 t and s2 s1 t
are the columns of G:
Antenna 1 Antenna 2
"
"
s1 s2 ! Symbol time 1
G
s2 s1 ! Symbol time 2
(10:265)
Note that the columns of G are orthogonal and so are the rows, that is,
G G GG ksk2 I2 , where s s1 s2 t . The Alamouti code allows the transmission of
two symbols over two antennas in two consecutive symbol times. Thus, the code rate is
unity since m=Ls 2=2 1. Hence, the Alamouti code requires no bandwidth expansion
and is the perfect space-time block code. The sum of the channel-tap weighted code words
plus noise is
s1
s2
N 1
Z h1
2
s2
s1
N 2
(10:266)
h 1 s 1 h2 s 2
N 1
Gh
h2 s1 h1 s2
N 2
The vector Z Gh N can be mapped into Y Hs N by changing the second element
of Z Z1 Z2 t , which is Z2 h2 s1 h1 s2 N 2 , into its complex conjugate
Z2 h2 s1 h1 s2 N 2 . Thus we have
603
Y Hs N
where N N1 N2 t N
matrix G is given by
Z1
h1 s 1 h 2 s 2
N1
Z2
h2 s1 h1 s2
N2
h1 h 2
s1
N1
h2 h1 s2
N2
(10:267)
h1
h2
h2
h1
(10:268)
Assume that the channel taps h1 and h2 are obtained from perfect channel estimation. The
receiver uses the matrix H to perform maximal ratio combining via the operation
H Y=khk H Hs N=khk khks H N=khk, where h h1 h2 t . Let us examine
the noise vector H N=khk, we have
1
1 h1 N1 h2 N2
(10:269)
H N
khk
khk h2 N1 h1 N2
2
The jointly Gaussian components
have zero mean
and identical
variance 2 . Furthermore,
0. Thus, the noise comthey are uncorrelated since E h1 N1 h2 N2 h2 N1 h1 N2
ponents are independent.
6
6
6
6
6
6
6
G6
6
6
6
6
6
4
Antenna 4
.........
"
s1
s2
s3
s2
s3
s1
s4
s4
s1
s4
s1
s3
s2
s2
s3
s2
s3
s1
s4
s4
s1
s4
s3
s2
! Symbol
7
s3 7 ! Symbol
7
s2 7 ! Symbol
7
s1 7
7 ! Symbol
7
s4 7 ! Symbol
7
s3 7
7 ! Symbol
7
s2 5 ! Symbol
s4
s1
time 1
time 2
time 3
time 4
(10:270)
time 5
time 6
time 7
! Symbol time 8
There are four symbols transmitting over eight symbol times (rate ) via four antennas
(four-fold diversity). The mapping of vector Z Gh N into Y Hs N is accomplished via the matrix H given by
604
Fading channels
2
6
6
6
6
1 6
H p 6
26
6
6
6
4
h1
h2
h3
h4
h1
h2
h3
h4
h2
h1
h4
h3
h2
h1
h4
h3
h3
h4
h1
h2
h3
h4
h1
h2
h4
h3
h2
h1
h4
h3
h2
h1
3
7
7
7
7
7
7
7
7
7
7
5
(10:271)
Sufficient statistics
The sufcient statistics in (10.261) do not apply to some space-time block codes. In this
section we present other sufcient statistics, and that they apply to space-time block codes
of smaller rate, such as rate . If we assume perfect channel estimation then, for a
given orthogonal space-time block code matrix G, where G G ksk2 ILt , maximal ratio
combining is achieved for any received
Z Gh N, if there exist m vectors hi ,
vector
2
i = 1,2, . . . , m, with identical norm hi khk2 such that
1
h Y i khksi Ni ; i 1; 2; . . . ; m
khk i
hi hi
khk
(10:272)
605
Note that the columns of G are orthogonal, that is, G G ksk2 I4 , where s s1 s2 s3 t .
Four code words (four columns of G) contain three symbols each, and are transmitted over
four symbol times (rate ) and via four antennas (four-fold diversity). The statistics in
(10.261) are not sufcient for this code. Instead we use the sufcient statistics (10.272). We
have
(10:274)
h1 h1 h2 h3 h4 ; h2 h2 h1 h4 h3 ; h3 h3 h4 h1 h2
2
Z Gh N;
3 2
3 2
h1 s 1 h 2 s 2 h 3 s 3
Z1
N
6 Z2 7 6 h2 s1 h1 s2 h4 s3 7 6 N
7 6
7 6
Z6
4 Z3 5 4 h 3 s h 4 s 2 h 1 s 5 4 N
1
3
h4 s1 h3 s2 h2 s3
Z4
N
3
1
7
7
5
3
2
(10:275)
t
t
t
Y 1 Z1 Z2 Z3 Z4 ; Y 2 Z1 Z2 Z3 Z4 ; Y 3 Z1 Z2 Z3 Z4
t
t
N^ 1 N 1 N 2 N 3 N 4 ; N^ 2 N 1 N 2 N 3 N 4 ; N^ 3 N 1 N 2 N
3
t
4
(10:276)
606
Fading channels
coding with a complex Ls Lt code matrix G, which provides Lt -fold diversity via Lt
transmit antennas for m complex symbols transmitting over Ls symbol times (code rate is
m/Ls ). On the receive side we implement diversity via Lr receive antennas. We wish to
show that with proper maximal ratio combining we can achieve the full diversity of order
L L t Lr .
Let us consider space-time codes that possess the sufcient statistics (10.261). For each
lth receive antenna, there corresponds an orthogonal channel tap matrix Hl obtained from
the mapping Ghl ! Hl s, where hl hl;1 hl;2 hl;Lt t is the lth channel tap vector
representing Lt paths from Lt transmit antennas to the lth receive antenna. The receivers
compute the following pre-combining vectors:
X l Hl Hl s N l khl k2 s Hl N l ; l 1; 2; . . . ; Lr
(10:278)
(10:279)
The normalized sum of the pre-combining vectors, X, exhibits the maximal ratio combining
for all L Lt Lr paths and is given by
X
Lr
Lr
Lr
1 X
1 X
1 X
Xl
H N l
k h l k2 s
khk l1
khk l1
khk l1 l
Lr
1 X
khks
H N l
khk l1 l
(10:280)
The decision vector X can be detected by m parallel minimum Euclidean distance detectors.
The noise vector has the covariance 2 2 Im .
Let us now consider space-time codes that possess the sufcient statistics (10.272). For
each lth receive antenna, there correspond m channel tap vectors hi;l , i = 1, 2, . . . , m. The
receiver computes Xi;l hi;l Y i;l khl k2 si khl kNi;l ; i 1; 2; . . . ; m; l 1; 2; . . . ; Lr ,
where hl hl;1 hl;2 hl;Lt t is the lth channel tap vector representing Lt paths from Lt
transmit antennas to the lth receive antenna. The receivers compute the following combining variables based on the channel tap vector h h1 h2 hLr t :
Xi
Lr
Lr
Lr
1 X
1 X
1 X
Xi;l
khl k 2 s i
khl kNi;l
khk l1
khk l1
khk l1
Lr
1 X
khksi
khl kNi;l ; i 1; 2; . . . ; m
khk l1
(10:281)
607
coding system suffers 10 log Lt (dB) power penalty. On the other hand, there is no power
penalty for receive antenna diversity; however, physical constraints often prevent the
implementation of many antennas on a receiver such as a cell phone, personal digital
assistant, or notebook computer, for example. Thus, transmitreceive antenna diversity
offers a compromise solution by reducing the number of both transmit and receive antennas
to satisfy both physical contraints and power requirements.
s1
s2
N 11
Z 1 h11
h12
s2
s1
N 12
(10:282)
h11 s1 h12 s2
N 11
Gh1 N1
s1
s2
N 21
Z 2 h21
22
s2
s1
N 22
(10:283)
h21 s1 h22 s2
N 21
Gh2 N2
Zl1
Y l Hl s N l
(10:284)
; l 1; 2
Zl2
Specically, we obtain
Y1
h11 s1 h12 s2
N
N
11
h12 s1 h11 s2
12
h11 h12
N
s1
h12 h11 s2
N
11
12
(10:285)
H1 s N 1
where
H1
h11
h12
h12
; H1 H1 kh1 k2 I2
h11
(10:286)
608
Fading channels
Also,
h21 s1 h22 s2
N 21
h22 s1 h21 s2
N 22
h21 h22
N 21
s1
h22 h21 s2
N 22
Y2
(10:287)
H2 s N 2
where
h
H2 21
h22
h22
; H2 H2 kh2 k2 I2
h21
(10:288)
(10:289)
The MRC decision variable is the normalized sum of X 1 and X 2 . For the channel tap vector
h h11 h12 h21 h22 t we have
X
1
k h 1 k2 k h 2 k 2
1
X 1 X 2
H N 1 H2 N 2
s
k hk
khk 1
khk
(10:290)
Since kh1 k2 kh2 k2 jh11 j2 jh12 j2 jh21 j2 jh22 j2 khk2 , we obtain the following MRC
decision vector:
X khks N
where the noise vector is N H1 N 1 H2 N 2 =khk. The covariance of noise
EN 2 N 2 H2. Since EN l N l 2 2 I2 , l
by ENN H1 EN1 N 1 H1 H2
2
2
2
obtain ENN 2 kh1 k kh2 k I2 =khk2 2 2 I2 .
(10:291)
N is given
= 1, 2, we
609
instantaneous channel capacity can also be theoretically zero during a deep fade. Hence, a
positive transmission rate is always greater than C during a deep fade, and no coding
systems exist that allow the code words to be transmitted during an outage with an
arbitrarily small frequency of errors. Thus, the average channel capacity does not exist
in most practical fading channels. However, we can still dene channel capacity in the
outage sense, which we will explore for the various channels studied in the previous
sections.
Slow fading
In a slow fading single-tap (at fading) channel, the channel estimation time is smaller than
the channel coherence time. With perfect channel estimation where the Doppler effect is
nullied completely, the instantaneous signal-to-noise ratio is simply jhj2 SNR, where h is the
channel tap. Thus, the instantaneous channel capacity can be dened as
(10:292)
C log2 1 jhj2 SNR bits=symbol
When the instantaneous channel capacity falls below the transmission rate R due to fading
the error probability cannot be made arbitrarily small and an outage occurs. We dene the
capacity-outage probability as follows:
h
i
PrC 5R Pr log2 1 jhj2 SNR 5R
(10:293)
For a specied po , there exists an outage capacity Co which is the maximum transmission
rate for which the capacity-outage probability is less than po . Setting PrC 5R po and
solving for R Co we have
h
i
po Pr log2 1 jhj2 SNR 5Co
2Co 1
Pr jhj2 5
(10:294)
SNR
Co
2 1
Fjhj2
SNR
1
2
where Fjhj2 y is thehdistribution
i function of jhj . Let Fjhj2 y be the inverse function of
1
Fjhj2 y, that is, Fjhj2 Fjhj2 po po , we can evaluate Co as follows:
h
i
1
(10:295)
Co log2 1 Fjhj2 po SNR bits=symbol
1
610
Fading channels
1
Fjhj2 po ln
1
1 po
(10:297)
(10:298)
For an outage probability po 0:01, the outage signal-to-noise ratio is 0:01 SNR. This is
almost 100 times smaller than that of an AWGN channel. Using the identity
log2 x ln x= ln 2 and SNR = 10, the outage capacity is approximately 4% of the AWGN
channel capacity, which is C log2 1 SNR.
2 Co 1
2
Pr khk 5
SNR
Co
2 1
Fkhk2
SNR
(10:301)
For a slow-at Rayleigh fading channel we can use the 2 -density function with 2L degrees
of freedom. Thus, the distribution function for L-fold diversity is given by
Fjjhjj2 y 1 ey
L1 k
X
y
k0
1
k!
(10:302)
There is no closed-form solution for Fjjhjj2 po , and hence Co , so the result has to be obtained
numerically.
611
SNR
5R
PrC 5R Pr log2 1 khk2
(10:304)
Lt
Given an outage probability po, the outage capacity is given as
SNR
1
bits=symbol
Co log2 1 Fjjhjj2 po
Lt
(10:305)
1
where po Fjjhjj2 Lt 2Co 1=SNR and Fjjhjj2 po can be calculated from (10.302).
612
Fading channels
SNR
1
Co log2 1 Fjjhjj2 po
bits=symbol
(10:306)
Lt
1
where po Fjjhjj2 Lt 2Co 1=SNR and Fjjhjj2 po can be calculated from (10.302) for
L L t Lr .
Example 10.36 Outage capacity of a slow fadingtransmit and receive antenna diversity
Rayleigh channel
Consider an outage probability po 0:01 and Alamouti code (m = 2, Ls = 2, Lt 2 )
working with two-fold receive diversity for a total of four-fold diversity, for SNR = 10, the
outage capacity is approximately 68% of the AWGN channel capacity, which is
C log2 1 SNR. As a comparison, the outage capacity of a non-diversity channel is
4% of the AWGN channel capacity, the outage capacity of a two-fold transmit antenna
diversity channel is 24% of the AWGN channel capacity, and the outage capacity of a twofold receive antenna diversity channel is 38% of the AWGN channel capacity.
Slow fadingOFDM
OFDM is designed to work in an underspread frequency-selective channel. With N subcarriers, OFDM provides N parallel slow fading subchannels with no ISI (at fading) in the
ideal case. The use of a cyclic prex to reduce ISI could only lower the channel capacity but
is necessary in practice. For a xed transmitted power P, each subchannel power is P=N .
This implies that the instantaneous signal-to-noise ratio of the lth subchannel at the receiver
is kHl k2 SNR=N . The instantaneous channel capacity is
C
N
X
l1
2 SNR
log2 1 jHl j
bits=symbol
N
(10:307)
613
If the OFDM target rate is R NRs , where Rs is the subcarrier symbol rate, then an outage
occurs if C < R, and the outage probability is
"
#
N
X
2 SNR
5R
log2 1 jHl j
PrC 5R Pr
(10:308)
N
l1
The above expression shows that as long as the OFDM capacity is above R, there exists a
coding system such that the code words can be transmitted over the channel with an
arbitrarily small frequency of errors. This happens irrespective of the fades experienced
by the subchannels, which could make some subchannel rates drop below R/N. This implies
that coding should be carried out across all subcarriers (this also means across many
coherence bandwidths) instead of over individual subcarriers. This ensures that if errors
occur in some subcarriers during a deep fade, they could still be corrected if the remaining
subcarriers have high signal-to-noise ratios. On the other hand, if coding is carried out for
each individual subcarrier, then a deep fade may wipe out a subcarrier entirely with no
chance of correcting the errors even if the remaining subcarriers have no error. Therefore,
coding for each individual subcarrier provides poor diversity, and coding across all subcarriers provides full diversity. The advantage of coding for each individual subcarrier is that
the decoding can be carried out at a lower subcarrier rate and parallel processing of all
subcarriers shortens the code latency. Furthermore, there are codes such as trellis coded
modulation that can provide excellent performance when implemented with MQAM subcarriers. On the other hand, codes such as low-density parity-check codes are excellent for
coding across all subcarriers.
P1 ...PN
N
X
log2 1 jHk j2 SNRk bits=OFDM symbol
(10:309)
k1
P
where SNRk Pk = 2 Pk =N0 , Nk1 Pk P,
and Hk is the DFT of the L-tap channel
p
P
j2kl=N
. Hence,
h h0 h1 hL1 0 0t multiplied by N , that is, Hk L1
l0 hl e
the power constraint optimization problem is the water lling strategy, which yields the
following power allocation:
!
1
N0
Pk max 0;
(10:310)
jHk j2
where the Lagrange multiplier is selected such that
channel model as follows:
Pn
k1
614
Fading channels
ht
L1
X
hl t l
(10:311)
l0
The corresponding frequency response is the Fourier transform of the channel impulse
response ht. We have
H f
L1
X
hl ej2f l
(10:312)
l0
For simplicity let the path delay be l lT , where T is the delay between two adjacent paths.
We have
H f
L1
X
hl ej2flT
(10:313)
l0
P
j2kl=N
Thus, the subchannel tap Hk L1
is the value of Hf sampled at
l0 hl e
frequency f k=NT, k 0; 1; . . . ; N 1, namely, Hk Hk=NT . Note that Hk is
also the discrete-time Fourier
transform
of the nite sequence hl , sampled at
j2kl=N
f k=N , namely, Hk H e
. Let us consider the case when the number of
subcarriers N is large as it tends to innity. This implies that the optimal power
allocation converges to the water lling power allocation over the channel frequency
response as follows:
!
1
N0
P f max 0;
(10:314)
jH f j2
where the Lagrange multiplier is selected such that
1=T
P f df P
(10:315)
(10:316)
The water lling power allocation achieves the maximum reliable rate of communication over independent subchannels. This is simpler and more effective than coding
across all subcarriers. It can be carried out by independent subcarrier coding with
adaptive rate and by using adaptive modulation to meet the power requirement. The
water lling power allocation requires the transmitter to know the state of the channel,
namely, the transmitter has the channel side information (CSI). If the channel state is
known only at the receiver, that is, only the channel side information at the receiver
(CSIR) is available, than coding across all subcarriers is the appropriate strategy.
615
Fast fading
Fast fading occurs when the channel estimation time is larger than the channel coherence
time. In previous material we have shown that Doppler tracking and correction can be done
down to the symbol time. This means that the Doppler phase error can be effective reduced
to no more than the phase error that accumulates over one symbol time. If the Doppler shift
is too large, as is the case with high-speed platforms such as high-speed trains, aircraft, and
low-Earth-orbit satellites, the Doppler phase error over a symbol time may incur a large loss
in the symbol signal-to-noise ratio of MQAM. This results in fast fading where the channel
coherence time is less than a symbol time. In such a fast fading channel, the average fade
duration is short and the symbols fade independently. For a quadrature (complex) channel
this means that symbol errors can occur randomly and there exists a coding system that
ensures the maximum reliable rate, which is the ensemble average rate given by
h
i
C E log2 1 jhj2 SNR
bits=symbol
(10:317)
If we assume Rayleigh fading, at low SNR the capacity can be approximated as
h
i
h
i E jhj2 SNR
C E log2 1 jhj2 SNR
ln 2
(10:318)
SNR
ln 2
2
where E jhj 1. The right-hand side of the above equation is also the capacity of an
AWGN channel at low SNR. At high SNR we have
h
i
h
i
(10:319)
C E log2 jhj2 SNR log2 SNR E log2 jhj2
At high SNR
h the capacity
i of an AWGN channel is log2 SNR. For a fast Rayleigh fading
channel E log2 jhj2 0:83 and C log2 SNR 0:83 log2 0:56 SNR. Thus, a
fast Rayleigh fading channel needs an additional 2.5 dB in SNR to achieve the same
capacity as that of an AWGN channel. For SNR = 10, the capacity of a fast Rayleigh
fading channel is 72% of the capacity of an AWGN channel. This is much better than a
slow Rayleigh fading channel where the outage capacity is about 30% the capacity of an
AWGN channel for an outage probability of 0.1. From the capacity point of view, slow
fading is more detrimental than fast fading. From the bit error probability point of view, a
more complex coding system is needed for a fast fading channel to achieve the same bit
error probability as compared with an AWGN channel. This is necessary because in a fast
fading channel one must deal with both envelope fading (such as Rayleigh fading) and
Doppler phase error. On the other hand, a slow fading channel only requires the coding
system to deal with envelope fading since the effect of Doppler shift is virtually nullied
by Doppler tracking and correction. Furthermore, since fast fading commonly occurs at
low bit rates and with high-speed platforms, most wireless applications for high data rates
and low mobility are not affected.
616
Fading channels
10.18 Summary
A fading channel is characterized by a time-varying effect and a space-varying effect. The former results from the
Doppler effect due to the movement of the transmitter and receiver. The latter results from the signal envelope
variation due to multipath echoes; these multipath components also vary according to the location of the
transmitter and receiver. The time-varying effect can be empirically described by the channel coherence time or
its counterpart the Doppler spread. If the channel coherence time is greater than the channel estimation time we
have a slow fading channel. If the channel coherence time is smaller than the channel estimation time we have a
fast fading channel. The space-varying effect can be empirically described by the multipath delay spread or its
counterpart the channel coherence bandwidth. If the multipath delay spread is smaller than the symbol time we
have a at fading channel. If the multipath delay spread is greater than the symbol time we have a frequencyselective fading channel. The Doppler spectrum (Clarke or Aulin) allows one to calculate the Doppler spread or
channel coherence time.
The performance of ideal coherent demodulation in a Rayleigh or Nakagami-m fading channel was evaluated
using known closed-form expressions. Bit error probability for IQ signals with known phase errors were also
derived. To correct the phase error due to Doppler shift we proposed the pilot symbol-aided decision-feedback
demodulation. This demodulation method employs differential decision-feedback (D-DF) and double-differential
decision-feedback (DD-DF) algorithms. The latter can correct the Doppler phase error down to the symbol time.
The performance of OFDM in the presence of Doppler shift was analyzed to show the effect of intercarrier
interference, which can be removed prior to DFT. The cyclic prex can remove ISI between OFDM symbols but the
intrasubcarrier interference can only be alleviated via equalization.
The performance of ideal noncoherent demodulation in a Rayleigh or Nakagami-m fading channel was
evaluated using known closed-form expressions. Doppler tracking of both MFSK and CSK were studied. Pilot-tone
aided demodulation of complex spreading orthogonal covering signals used in IS-95, CDMA 2000, and WCDMA
was investigated as well as noncoherent demodulation of offset quadrature DS-CSK signals.
Diversity is employed in practice to combat fading based on the fact that multiple receptions of the same
symbol are affected differently in a fading channel. Some receptions might survive and others might be wiped
out. The survivors may be of help in detecting the symbol. There are primarily four types of diversity, namely,
time diversity, frequency diversity, path diversity, and antenna diversity. Time diversity effectiveness requires
symbols to be interleaved over a time frame larger than the average fade duration. Frequency diversity is
commonly employed in fast frequency hopping and possibly in OFDM. Path diversity is exploited by the Rake
receiver operating in a resolvable multipath fading environment. Antenna diversity is the most powerful diversity
technique that can be implemented either at the receiver, at the transmitter, or at both transmitter and receiver.
Antenna diversity is also referred to as MIMO. Diversity signal is detected via maximal ratio combining, selection
combining, and equal gain combining. Transmit diversity in many practical systems is implemented via an
orthogonal space-time code such as the well-known Alamouti block code. The maximum rate is unity.
The capacity of a slow fading channel is best dened as the outage capacity since the ensemble average
capacity does not exist. On the other hand, the ensemble average capacity of a fast fading channel does exist.
Slow fading is more detrimental to the channel capacity than fast fading. However, fast fading requires a more
complex coding scheme to compensate for both envelope fading and Doppler error.
617
2
X
(10A:1)
k1
(10A:2)
(10A:3)
1
The sufcient condition for orthogonality of the basis functions is demonstrated in (6.94) of
Example 6.26. Using complex envelope representation we can express the waveform si t as
follows:
1
si t Re sL te j2fc t sL tej2fc t sL tej2fc t
2
(10A:4)
(10A:5)
To demodulate the
IQ symbol si si1 jsi2 we multiply the received signal si t by a local
p
complex carrier 2=Ts ej2fc t to obtain
r
2 j2fc t
1
1
e
si t p sL t p sL tej4fc t
(10A:6)
Ts
2Ts
2Ts
The high-frequency term is rejected by the complex-valued matched lter
ht pt Ts , where is a normalization constant. The output of the complex-valued
matched lter is si si1 jsi2 . The mapper maps si si1 jsi2 to the symbol vector
si si1 si2 t by taking the real part and imaginary part of si .
618
Fading channels
Pb EPb jhj E
Nn
Q
log2 M
q
2jhj2 SNR
(10B:1)
Rayleigh
The density function of jhj is fjhj x 2xex and the following expression can be employed
to evaluate Pb [7]:
1
1
a
2 2
Qaxeb x xdx 2 p 1 ; Rea2 42Reb2
(10B:2)
4b
a2 2b2
0
p
Setting b2 1, a 2SNR, we obtain the bit error probability
2
r!
SNR
1
1 SNR
Nn
Pb
2 log2 M
(10B:3)
Nakagami-m
m
The density function of jhj2 is fjhj2 y m
ym1 emy and the following expression can be
employed to evaluate Pb [8]:
m
1
0
p
c=2a
1
b b 12
p b1 ay
Q cyy e dy p
2 1 c=2ab1=2 ab b 1
1
1
2 F1 1; b ; b 1; 1 c=2a
2
(10B:4)
;
m
1;
2
1
2
1 SNR=m
log2 M 2 1 SNR=mm1=2 m 1
(10B:5)
When b is an integer the following expression can be used to obtain a simpler closed-form
bit error probability
s
"
k #
1
b1
2k
b
c=2a X
1
p b1 ay
Q cyy e dy
1
; b : integer
2ab
1 c=2a k0 k
41 c=2a
0
(10B:6)
where b b 1! Setting a m, b = m, and c=2a SNR=m, we obtain
619
s
"
k #
m1
Nn
SNR=m X
1
2k
1
Pb
; m : integer
2 log2 M
1 SNR=m k0 k
41 SNR=m
(10B:7)
Alternatively we can use the following closed-form expression to evaluate the bit error
probability [9]:
1
p
Q cyyb1 eay dy
b
b
a
!b 2
!k 3
p
p
b1
b1k
1 2a=c 1 4X
1 2a=c 1 5
p
p
;
k
2 1 2a=c
2 1 2a=c
k0
(10B:8)
b : integer
Setting a m, b = m, and c=2a SNR=m, we obtain
Nn
Pb
log2 M
!m 2
!k 3
p
p
m1
m1k
1 m=SNR 1 4X
1 m=SNR 1 5
p
p
;
k
2 1 m=SNR
2 1 m=SNR
k0
m : integer
(10B:9)
Nn
Pb E
Q
2khk2 SNR
(10B:10)
log2 M
Setting a = 1, b = L, and c=2a SNR, and noting that L L 1!, we obtain
"
r L1
k #
Nn
SNR X 2k
1
(10B:11)
1
Pb
1 SNR k0 k
41 SNR
2 log2 M
Alternatively, using (10B.8) we have
Pb
Nn
log2 M
!L 2
!k 3
p
p
L1
X
L
1
k
1 1=SNR 1 4
1 1=SNR 1 5
p
p
;
k
2 1 1=SNR
2
1 1=SNR
k0
L : integer
(10B:12)
620
Fading channels
or, equivalently,
" L1
#
Nn
1 L X
1 k
L1k
; L : integer
Pb
k
2
2
log2 M
k0
(10B:13)
1 SNR
(10B:14)
Problems
1.
2.
3.
A mobile approaches a base station at 100 km/hour. The mobile transmits a packet of
200 bits at a rate of 100 kbps. The carrier frequency is 1.9 GHz. The multipath delay
spread is Td 0:25 s. The signal-to-ISI ratio is approximated by SIR 0:9Tb =Td.
The signal-to-noise ratio is SNR 100 cos2 , where the carrier phase error due to
Doppler shift is . Calculate the signal-to-noise-and-ISI of the last bit of the packet.
Let us consider the scenario when a mobile traveling at speed of 100 km/hour receives
a 1950 MHz signal from a base station. The data rate is 14.4 kbps.
(a) Find the Doppler spread.
(b) Find the channel coherence time.
(c) If the channel estimation time is of the channel coherence time, what is the
channel throughput, assuming each channel estimation update is achieved via a
transmitted pilot symbol that replaces a data symbol?
Using (10.17) plot the magnitude and phase spectra for the following channel:
4.
Tap, hi
Delay, li
1
2
3
4
0.75
0.35
0.2
0.01
0
0.9
1.7
3
Using (10.17) plot the magnitude and phase spectra for the following channel:
5.
Tap, hi
Delay, li
1
2
3
0.75
0.55
0.40
0.8
0.85
0.9
Plot the ClarkeDoppler power spectrum in (10.31) and the AulinDoppler power
spectrum in (10.44).
621
Problems
6.
7.
8.
9.
10.
11.
12.
13.
14.
15.
16.
17.
18.
19.
20.
21.
Using polar coordinate transformation, evaluate the bit error probability of a coherent
I-Q signal in a Rayleigh fading channel.
Determine the SNR-outage probability of a slow-at Nakagami-m fading channel. Let
m = 3, the threshold is 1, and SNR = 10.
Determine the SNR-outage probability of a slow-at Ricean fading channel. Let
4:5, the threshold is 1, and SNR = 10.
Plot the bit error probability of PSK in an AWGN channel with phase error
" 0 ; 10 ; 22:5 ; 30 ; 45 ; 60 .
Plot the bit error probability of QPSK in an AWGN channel with phase error
" 0 ; 10 ; 22:5 ; 30 ; 45 .
Plot the bit error probability of 16QAM in an AWGN channel with phase error
" 0 ; 10 ; 22:5 ; 30 ; 45 .
Plot the bit error probability of 64QAM in an AWGN channel with phase error
" 0 ; 10 ; 22:5 ; 30 ; 45 .
Verify (10.82).
Plot the bit error probability of 16QAM in a Rayleigh fading channel with phase error
" 5 ; 10 ; 22:5 .
Plot the bit error probability of 64QAM in a Rayleigh fading channel with phase error
" 5 ; 10 ; 22:5 .
Show that the bit error probabilities of MPSK and MQAM in a Rayleigh fading
channel are inversely proportional to the signal-to-noise ratio.
Evaluate the bit error probability of PSK with convolutional code K = 7, rate ,
and soft decision in a Rayleigh fading channel at 10 dB signal-to-noise ratio
per bit.
Consider the reception of a DS-PSK signal in a Rayleigh fading channel with pulsed
jamming.
(a) Derive the bit error probability in term Eb =N0 , assuming AWGN is neglible
compared to the jamming signal.
(b) Assume a pulse duty cycle equal to 0:1 and Eb =N0 20 dB. Calculate the bit
error probability.
A packet of length L is sent over a Rayleigh fading channel via PSK. Assuming that bit
errors occur randomly and that the required packet error probability is 102 , calculate
the corresponding signal-to-noise ratio per bit for L = 200 bits.
Consider a mobile traveling toward a base station at a speed of 100 km/hour that
receives a 5400 MHz PSK signal from a base station. It is required that the throughput
be at least 80% and that the loss in SNR relative to the ideal case is no more than 2 dB.
Calculate the minimum bit rate assuming Rayleigh fading.
Consider a fading channel with Doppler spread fD 200 Hz. A PSK signal operating
with a bit rate of 128 kbps employs pilot bits for channel estimation.
(a) Specify the condition for slow fading. Estimate the maximum channel estimation
time in bits.
(b) Calculate the phase error for the above maximum estimation time.
(c) Revise the estimation time to get a reasonable phase error of no more than =8
radians.
622
Fading channels
22.
Consider a mobile moving toward a base station at radial velocity v = 110 km/hour.
The mobile transmits a PSK signal at 28.8 kbps to the base station. The data frame is
300 bits including 20 overhead bits. The channel estimation is achieved by inserting
pilot bits periodically. The carrier frequency is 1.9 GHz.
(a) Calculate the channel estimation time in bits so that the Doppler phase error
cannot exceed 30 .
(b) Calculate the number of pilot bits per frame.
(c) Calculate the frame efciency.
23. Consider a perfect reector model in Equation (10.4). Assume that the receiver
radial velocity is 80 km/hour, and the received signal has a carrier frequency of
1.9 GHz.
(a) Calculate the deep fade frequency.
(b) Calculate the deep fade period.
(c) Suggest a method to eliminate the deep fades.
(d) Suggest a method to improve the reception via signal combining.
24. Consider a QPSK signal with phase error " 22:5 in a Rayleigh fading channel. The
bit error probability is given by
s!
s!
1
SNRcos " sin "2
1
SNRcos " sin "2
1
1
Pb
4
4
1 SNRcos " sin "2
1 SNRcos " sin "2
25.
26.
27.
28.
29.
30.
623
Problems
31.
32.
33.
34.
35.
36.
37.
38.
39.
40.
41.
42.
43.
probability that the instantaneous despread input signal-to-jamming ratio falls below a
threshold value . N is the spread factor.
Consider a DS-PSK signal under pulse jamming, assuming that both signal and
jammer undergo independent Rayleigh fading and the AWGN effect is negligible
relative to the jamming effect. Derive the N SJRi -outage probability, that is, the
probability that the instantaneous despread input signal-to-pulsed jamming ratio falls
below a threshold value . N is the spread factor and is the pulsed jamming duty
cycle.
For a Rayleigh fading channel derive the bit error probability of noncoherent FHMFSK under partial-band jamming, assuming that the AWGN effect is negligible
relative to the jamming effect.
Derive the bit error probability of noncoherent FH-MFSK under partial-band jamming, assuming that both signal and jammer undergo independent Rayleigh fading
and the AWGN effect is negligible relative to the jamming effect.
For a Rayleigh fading channel derive the bit error probability of noncoherent slow FHFSK under follower jamming, assuming that the AWGN effect is negligible relative to
the jamming effect.
It is required that the rst symbol after the most recent channel update suffers no more
than 10 of differential Doppler phase error when the Doppler shift is 180 Hz.
Estimate the maximum bit rate for 64QAM. Calculate the Doppler loss at this bit
rate for a Rayleigh channel.
Consider a mobile traveling toward a base station at a speed of 100 km/hour that
receives a 5400 MHz QPSK signal from a base station. It is required that the
throughput be at least 80% and that the loss in SNR relative to the ideal case is no
more than 3 dB at a bit error probability of 5 103 . Calculate the minimum bit rate
assuming Rayleigh fading.
Consider a mobile traveling toward a base station at a speed of 100 km/hour that
receives a 5400 MHz 16QAM signal from a base station. It is required that the
throughput be at least 50% and that the loss in SNR relative to the ideal case is no
more than 6 dB.
(a) Calculate the pilot symbol rate.
(b) Calculate the minimum bit rate assuming Rayleigh fading.
Derive (10.136).
Derive (10.138).
Rework Table 10.3 for 128FSK.
Rework Table 10.4 for 128CSK.
Establish the signal-to-interchannel interference ratio for pilot tone-aided demodulation of an orthogonal covering signal in the presence of Doppler shift f =180 Hz for the
following cases:
(a) QPSK with bit rate of 28.8 kbps.
(b) 64QAM with bit rate of 86.4 kbps.
Approximate interchannel interference as AWGN and assume a bit error probability
of 103 for orthogonal covering QPSK with normalized Doppler shift fTs 0:01.
Calculate the required Eb =N0 .
624
Fading channels
44.
45.
46.
47.
48.
49.
50.
51.
52.
53.
Nn
Nn
1
Q
2khk2 SNR
Pb E
log2 M
log2 M 1 SNRL
Repeat Problem 51 for both Nakagami-m and Ricean fading channels.
(a) Derive the result via direct integration.
(b) Derive the result using the Laplace transform (or moment generating function).
The function Qx can also be expressed by the following integral [10]:
Qx
=2
ex
=2 sin2
d; x40
Use the above integral to express the bit error probability of the I-Q signal as follows:
Pb E
54.
55.
56.
57.
Nn
Q
log2 M
q
2khk2 SNR
Nn 1
log2 M
=2
SNR
F
d
sin2
0
625
Problems
58.
59.
60.
61.
62.
63.
64.
65.
same transmitted power and input bit rate. Consider the case of 64QAM with
SNR 1000 (30 dB) and Rayleigh fading. Calculate the diversity gain with selection combining.
Derive the bit error probability of an L-fold diversity noncoherent DPSK signal
with EGC in a Nakagami-m fading channel. (Hint: Use (11.1-13)(11.1-14)
from [11].)
Consider a Rayleigh channel and an L-fold diversity combining technique where the
two largest of L decision samples of a coherent I-Q signal are combined (second-order
SC). The density function of the largest random variable of L random variables is
derived in (10.216).
(a) Derive the density function of the kth largest random variable of L random
variables.
(b) Derive the joint density funtion of the two largest random variables of L random
variables.
(c) Derive the density function of the sum of the two largest random variables of L
random variables and the bit error probability for receive antenna diversity.
Derive the bit error probability for an L-fold diversity convolutionally coded PSK
signal with MRC in a slow Rayleigh fading channel.
Derive the bit error probability for an L-fold diversity TCM-MQAM signal with MRC
in a slow Rayleigh fading channel.
Consider a three-tap channel with L = 3, SINR = 200, and 16QAM.
(a) Calculate the bit error probability for a non-Rake receiver.
(b) Calculate the bit error probability for a Rake receiver.
(c) For a spread factor of 256, calculate SINR 1=2dmin =2s2 and specify the
signal-to-noise ratio per bit Eb =N0 to achieve SINR 200.
Verify that Z Gh N maps to Y Hs N via (10.270) and (10.271).
Show that (10.272) is satised using (10.274)(10.276) as an example.
For a real signal space, the space-time block code of unity rate does not have the limit
of two transmit antennas as in the Alamouti code. In fact, a space-time block code with
unity rate for any real signal space can have any number of transmit antennas.
Consider the following real code matrix:
G1 Gt2
G
G2 Gt1
(a) Find the necessary and sufcient condition for G to be orthogonal.
(b) Use the following two matrices to construct G and perform the MRC operation
for a four-fold transmit diversity, assuming that the signal is PSK:
s1 s 2
s3 s4
G1
; G2
s2 s1
s4 s3
66.
626
Fading channels
s1
6 0
G6
4 s2
s3
67.
3
s2
s3 7
7
s1 5
0
0
s1
s3
s2
3
s3
07
7
s1 5
s2
68.
69.
Investigate the possibility of using the following real matrix as orthogonal space-time
block code for a real signal space:
2
s1
6 s2
G6
4 s3
s4
70.
71.
72.
s2
s1
s4
s3
s3
s4
s1
s2
3
s4
s3 7
7
s2 5
s1
Find the outage capacity of a slow fading Rayleigh channel for po 0:01 for the
following cases:
(a) no diversity;
(b) receive antenna diversity with L = 2;
(c) transmit antenna diversity with Alamouti code;
(d) transmit antenna diversity with a rate space-time code and four antennas;
(e) transmit and receive antenna diversity with Alamouti code and two receive antennas.
Find the outage capacity of a fading channel characterized by the channel tap with the
following density function: fjhj x 2x=x2 12 ; x40.
Consider a MIMO system characterized by the Lr Lt channel tap matrix H hij ,
where hij represents the channel tap from the jth transmit antenna ( j = 1, 2, . . . , Lt ) to
the ith receive antenna (i = 1, 2, . . . , Lr ). Given the transmit signal vector s, the
received signal vector y is given by
y Hs N
where N is the zero mean complex Gaussian noise vector whose iid components have
identical variance (circular symmetric). Assuming that H H1 exists, that is, H is of
full column rank, design a zero-forcing (ZF) detector to recover the transmit vector s.
Is the output noise white? Find its covariance matrix.
627
Bibliography
73.
74.
Further reading
For a comprehensive study of propagation models we recommend [1,2]. Excellent treatment
of capacity of a fading channel for various wireless systems such as CDMA, OFDM, and
MIMO can be found in [3]. For the readers who wish to explore MIMO in depth, references
[46] should be consulted.
Bibliography
1. W. C. Jakes, Microwave Mobile Communications, IEEE Press, 1974.
2. R. Janaswamy, Radiowave Propagation and Smart Antennaa for Wireless Communications,
London: Kluwer, 2001.
3. D. Tse and P. Viswanath, Fundamentals of Wireless Communication, Cambridge:
Cambridge University Press, 2005.
4. E. Biglieri and G. Taricco, Transmission and Reception with Multiple Antennas:
Theoretical Foundations, Now Publishers, 2004.
5. E. G. Larsson and P. Stoica, Space-Time Block Coding for Wireless Communications,
Cambridge: Cambridge University Press, 2003.
628
Fading channels
Index
629
630
Index
codes (cont.)
rate, 132
repetition, 132
ReddSolomon, 150
self-punctuating, 124
ShannonFano, 125
soft decoding, 291, 410
uniquely decodable, 124
variable-length, 124
code division multiplexing, 436
coding gain, 146, 152, 160
asymptotic, 417
coherence bandwidth, 527
coherence time, 525, 532, 533, 536, 547
combining loss, 415, 464
comparator, 322
complex envelope, 4, 69, 70, 90, 232, 237, 241, 253,
258, 274, 283, 478, 528, 534, 537, 554, 569
complex spreading, 7, 566, 591
complex despreading, 7, 567, 573, 591
complex exponential, 37
constraint length, 160
convolution, 35, 61, 287, 306
circular, 287
co-phasing, 578, 578, 585
correlation metric, 353, 389, 510, 513, 596
correlator, 5, 356
noncoherent, 5, 318, 360
Costas loop, 336
coverage area, 115
CPM, 5, 280, 386, 397,
trellis, 388
CP-MFSK, 277
CSK, 5, 275, 383, 385, 435, 444, 558, 562, 572
cyclic prex, 286, 287, 402, 445
cyclic redundant check, 443
cyclostationary process, 4, 85, also see random process
D/A conversion, 2, 285, 289, 402
DC value, 38
decision-feedback selection algorithm, 561, 565
decoding, 163
hard, 163, 410, 411
soft, 165, 167, 410, 413
decorrelation, 92
delay spread, 92, 97
demodulator, 355, 358, 481
complex-valued, 536, 617
detector, 5
M-ary maximum, 5, 350
maximum, 5, 322, 456
maximum likelihood, 305, 397
minimum Euclidean distance, 5, 345
optimum, 305
phase, 370
slope, 337
threshold, 5, 320, 322, 347, 452, 457
differential encoding, 263, 274
631
Index
632
Index
633
Index
PSTN, 191
probability, 78, 133, 134, 150, 151
a posteriori, 79, 129, 154, 164, 168, 321, 414
a priori, 79, 129
capacity-outage, 609
density function, 80, 109, 110, 111, 112, 113, 135,
311, 319, 538
distribution function, 80, 311
minimum error, 321
outage, 114, 115, 117, 542, 582, 613
pair-wise error, 348, 410, 412, 415, 417, 539, 551,
578
sequence error, 391
symbol or bit error, 348, 395, 539, 557, 561, 579,
587, 592, 617
total, 79, 81
transition, 128, 131
processing gain, 432, 448, 449
PSK, 5, 14, 30, 88, 94, 135, 150, 231, 326, 432, 482,
486, 539, 544, 580
puncturing, 161
Q-function, see Gaussian integral function
quadrature spreading, 436
quantization, 165
QPSK, 15, 33, 89, 92, 94, 367, 486, 540, 545, 580, 598
radiometer, 464
raised cosine pulse shape, 232, 238, 254, 260, 476
Rake receiver, 590, 599
random process, 82
baseband, 88
cyclostationary, 85
Gaussian, 4, 100
independent, 83
lowpass, 88
stationary, 85
uncorrelated, 83
wise-sense cyclostationary, 85
wise-sense stationary, 85, 90, 307, 356, 531
random variable, 80
central chi-squared, 112, 113, 319, 325, 342, 344,
579, 586, 610, 619
continuous, 80
discrete, 80
equally likely, 84, 86
expected value (mean value), 81, 85
exponential, 542
Gaussian, 6, 100, 137, 311, 320
independent, 100, 103, 104
mean-square value, 81
moment, 81
non-central chi-squared, 319, 325, 342, 344, 540, 581
standard deviation, 81
uncorrelated, 100, 103
uniformly distributed, 83
variance, 81
634
Index
time-average, 88
timing synchronization, 403
total probability theorem, 79, 320, 321, 365, 450
transfer function, 46, 54, 58, 63, 106, 310, 502
canonical factor, 503
discrete-time, 496
lowpass, 46, 478
minimum phase, 489, 503, 507, 512
transformation, 100
transimpedance amplier, 329
transmission line, 185
transversal lter, 492
triangle inequality, 134
trunking efciency, 209
Ungerboeck, 291
union, 78
bound, 348, 351, 354, 360, 362, 383, 385,
390, 411
unit impulse function, 39
unit step function, 39
upconverter, 76, 177
vector, 27
Euclidean distance, 29
orthonormal basis, 28
squared norm, 28
Viterbi algorithm, 4, 6, 163, 165, 166, 168, 390, 392,
399, 410, 417, 510, 514
voice activation, 444
Walsh functions, 3, 6, 25, 276, 435, 436, 440, 445,
452, 563
variable length, 439
water-lling strategy, 141, 614
WCDMA, 3, 5, 7, 211, 436, 439, 457, 536, 547
weighting, 578, 585
white data, 97, 496
white noise, 179, 601
whitening lter, 107, 508, 514
wise-sense cyclostationary random process, see random
process
wise-sense stationary random process, see random
process
Wiener, 4, 92
Z-transform, 489