Block Transceivers OFDM and Beyond
Block Transceivers OFDM and Beyond
SeriesISSN:
Series ISSN:1932-1244
ISSN: 1932-1244
1932-1244
M
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Block
BlockTransceivers
Transceivers
Series
SeriesEditor:
Series Editor:William
Editor: WilliamTranter,
William Tranter,Virginia
Tranter, VirginiaTech
Virginia Tech
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Block
BlockTransceivers:OFDM
Block Transceivers:OFDMand
Transceivers:OFDM andBeyond
and Beyond
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Paulo
PauloS.R.
Paulo
Wallace
S.R.Diniz,
S.R.
WallaceA.
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Markus
MarkusV.S.
Markus V.S.Lima,
V.S. Lima,Universidade
Lima, UniversidadeFederal
Universidade Federaldo
Federal doRio
do Riodede
Rio deJaneiro
Janeiro
Janeiro
The
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BLOCK
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book presentsOFDM-inspired
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With
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address whatlies
what liesbeyond
lies beyondthe
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solutions inbroadband
broadbandtransmissions.
broadband transmissions.
transmissions.
About
AboutSYNTHESIs
About SYNTHESIs
SYNTHESIs
This
Thisvolume
This volumeisisisaaaprinted
volume printedversion
printed versionofof
version ofaaawork
workthat
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458295
William
WilliamTranter,
William Tranter,Series
Tranter, SeriesEditor
Series Editor
Editor
Block Transceivers
OFDM and Beyond
Synthesis Lectures on
Communications
Editor
William Tranter, Virginia Tech
All rights reserved. No part of this publication may be reproduced, stored in a retrieval system, or transmitted in
any form or by any means—electronic, mechanical, photocopy, recording, or any other except for brief quotations in
printed reviews, without the prior permission of the publisher.
DOI 10.2200/S00424ED1V01Y201206COM007
Lecture #7
Series Editor: William Tranter, Virginia Tech
Series ISSN
Synthesis Lectures on Communications
Print 1932-1244 Electronic 1932-1708
Block Transceivers
OFDM and Beyond
M
&C Morgan & cLaypool publishers
ABSTRACT
The demand for data traffic over mobile communication networks has substantially increased during
the last decade. As a result, these mobile broadband devices spend the available spectrum fiercely,
requiring the search for new technologies. In transmissions where the channel presents a frequency-
selective behavior, multicarrier modulation (MCM) schemes have proven to be more efficient, in
terms of spectral usage, than conventional modulations and spread spectrum techniques.
The orthogonal frequency-division multiplexing (OFDM) is the most popular MCM method,
since it not only increases spectral efficiency but also yields simple transceivers. All OFDM-based
systems, including the single-carrier with frequency-division equalization (SC-FD), transmit re-
dundancy in order to cope with the problem of interference among symbols. This book presents
OFDM-inspired systems that are able to, at most, halve the amount of redundancy used by OFDM
systems while keeping the computational complexity comparable. Such systems, herein called mem-
oryless linear time-invariant (LTI) transceivers with reduced redundancy, require low-complexity
arithmetical operations and fast algorithms. In addition, whenever the block transmitter and re-
ceiver have memory and/or are linear time-varying (LTV), it is possible to reduce the redundancy in
the transmission even further, as also discussed in this book. For the transceivers with memory it is
possible to eliminate the redundancy at the cost of making the channel equalization more difficult.
Moreover, when time-varying block transceivers are also employed, then the amount of redundancy
can be as low as a single symbol per block, regardless of the size of the channel memory.
With the techniques presented in the book it is possible to address what lies beyond the use
of OFDM-related solutions in broadband transmissions.
KEYWORDS
block transceivers, multicarrier modulation (MCM), orthogonal frequency-division
multiplexing (OFDM), reduced-redundancy transceivers, broadband digital commu-
nications
vii
Contents
Preface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . xi
Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . xv
3 OFDM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
3.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
3.2 Analog OFDM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
3.2.1 From TDM to FDM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
3.2.2 Orthogonality Among Subcarriers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
3.2.3 Orthogonality at Receiver: The Role of Guard Interval . . . . . . . . . . . . . . . . 62
3.2.4 Spectral Efficiency, PAPR, CFO, and I/Q Imbalance . . . . . . . . . . . . . . . . . 69
3.2.5 Implementation Sketch . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 71
3.3 Discrete-Time OFDM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73
3.3.1 Discretization of The OFDM Symbol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73
3.3.2 Discretization at Receiver: The CP-OFDM . . . . . . . . . . . . . . . . . . . . . . . . . 76
3.3.3 Discrete-Time Multipath Channel . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
3.3.4 Block-Based Model . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 82
3.4 Other OFDM-Based Systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
3.4.1 SC-FD . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
3.4.2 ZP-Based Schemes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
3.4.3 Coded OFDM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
3.4.4 DMT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
3.5 Concluding Remarks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
Bibliography . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 175
Preface
The widespread use of mobile devices with high processing capabilities, like smartphones and tablets,
as well as the increasing number of users and growing demand for higher data rates are some of the
main reasons why data traffic over mobile communication networks has increased so much during
the last decade. As a result, these mobile broadband devices spend the available spectrum fiercely,
requiring the search for new technologies.
From the physical-layer viewpoint, the first step to address this problem is to choose a mod-
ulation scheme which is more adequate for the type of channel through which the signal (wave)
propagates. Indeed, the spectral efficiency of communications systems can significantly increase
by properly choosing the modulation scheme. For example, in broadband transmissions in which
the channel presents a frequency-selective behavior, multicarrier modulation (MCM) schemes have
proven to be more efficient, in terms of spectral usage, than conventional modulations and spread
spectrum techniques.The multicarrier transmission illuminates the physical channel utilizing several
non-overlapping narrowband subchannels, where each subchannel appears to be flat, thus turning
the equalization process simpler.
Among the existing MCM schemes, the orthogonal frequency-division multiplexing
(OFDM) is the most notorious, since it not only increases spectral efficiency but also yields simple
transceivers. OFDM is capable of eliminating the intersymbol interference (ISI) with very simple
transmitter and receiver, by performing low-complexity computations such as insertion and removal
of a prefix, and by using fast algorithms such as the fast Fourier transform. It is worth noting that
ISI is one of the most harmful effects in broadband transmissions. Simple transceivers are attractive
since they lead to lower latency and require less power consumption. Therefore, it is no surprise
that OFDM has been adopted by many wired and wireless broadband communication technologies.
For instance, the long-term evolution (LTE) is a wireless communication standard whose down-
link connection is based on OFDM, whereas its uplink connection is based on the single-carrier
with frequency-division equalization (SC-FD), which is similar to OFDM in many aspects and is
composed of the same building blocks. The enhancements introduced by LTE physical layer are so
significant that LTE achieves much higher data rates, as compared to 3rd generation (3G) systems,
and is already being considered a 4th generation (4G) system.
At this moment, given the desirable features of OFDM and SC-FD schemes and their
widespread use in both wired and wireless communication standards, one could ask the following
questions: Is this the best we can do in terms of spectral efficiency? When LTE spectrum gets
overloaded, what comes next? Can we further improve these schemes?
This book tries to provide some directions to address these questions. Both OFDM and SC-
FD systems transmit redundancy (the prefix) in order to cope with the problem of ISI. Indeed, a
xii PREFACE
portion of each data block is reserved for the prefix, whose size must be larger than the channel
memory for OFDM and SC-FD systems. Thus, spectral efficiency may be increased if less redun-
dancy could be used. This book presents OFDM-inspired systems that are able to, at most, halve
the amount of redundancy used by OFDM systems while keeping the computational complexity
comparable. Such systems, herein called memoryless linear time-invariant (LTI) transceivers with
reduced redundancy, require low-complexity arithmetical operations and fast algorithms. In addition,
whenever the block transmitter and receiver have memory and/or are linear time-varying (LTV), it
is possible to reduce the redundancy in the transmission even further, as discussed in the last chapter
of this book. For the transceivers with memory it is possible to eliminate the redundancy at the
cost of making the channel equalization a more difficult task. Moreover, when time-varying block
transceivers are also employed, then the amount of redundancy can be as low as a single symbol per
block, regardless of the size of the channel memory. An example is the code-division multiple access
(CDMA) system with long spreading codes which is always able to achieve ISI elimination as long
as the system is not at full capacity, i.e., at least one spreading/de-spreading code is unused.
The approach followed in this book is to present both OFDM and SC-FD systems as particular
cases of the so-called transmultiplexer (TMUX). In fact, these two systems belong to the category
of memoryless TMUXes. Special attention is given to OFDM and its desirable properties, since
OFDM is being employed in many standards. Then, the TMUX is used as the main framework to
derive both LTI and LTV transceivers with reduced redundancy.
With the techniques presented in the book it is possible to address what lies beyond the use of
OFDM-related solutions in broadband transmissions. In summary, this book presents solutions to
reduce the redundancy in transmission aiming at increasing data throughput. However, it is worth
mentioning that reducing redundancy might increase mean square error (MSE) and bit-error rate
(BER), as well as turn the design of the block transceiver more challenging. The optimum solutions
are environment-dependent and its proper sensing leads to much more efficient spectral usage.
PREREQUISITES
We attempted to make this book as self contained as possible. Although basic knowledge of wireless
communications, digital transmission, and multirate signal processing is highly desirable, it is not
necessary since the first two chapters revisit the main concepts which are used throughout the book.
Thus, the main prerequisites to follow this book are: digital communications, basic concepts of
stochastic processes—involving expected values, means, and variances of random variables—and
linear algebra—involving operations with vectors and matrices, ranks, determinants, null and range
spaces. More advanced concepts, such as structured matrices and displacement theory, are explained.
Acknowledgments
The authors are grateful to Joel Claypool for kindly pushing us to finish this project. They are also
thankful to Professors M.L.R. de Campos, E.A.B. da Silva, L.W.P. Biscainho, and S.L. Netto of
UFRJ, and Professor T.N. Ferreira of UFF for their incentive and always being available to answer our
questions. They would like to thank Professors R. Sampaio Neto of PUC-RJ and V.H. Nascimento
of USP who influenced some parts of this book. Wallace thanks his colleagues at the Federal Center
for Technological Education Celso Suckow da Fonseca (CEFET/RJ–UnED-NI), in particular at
the Department of Control and Automation Industrial Engineering.
We also would like to thank our families for their patience and support during this challenging
process of writing a book. Paulo would like to thank his parents, his wife Mariza, and his daughters
Paula and Luiza for illuminating his life. Wallace thanks his fiancee Claudia and his parents, Renê
and Perpétua. Markus thanks his parents, Luiz Álvaro and Aracy, and his girlfriend Bruna.
List of Abbreviations
2G 2nd Generation
3G 3rd Generation
3GPP 3rd Generation Partnership Project
4G 4th Generation
ADSL Asymmetric Digital Subscriber Line
BER Bit Error Rate
CDMA Code-Division Multiple Access
CFO Carrier-Frequency Offset
CP-OFDM Cyclic-Prefix Orthogonal Frequency-Division Multiplexing
CP-SC-FD Cyclic-Prefix Single-Carrier with Frequency-Division equalization
CSI Channel-State Information
DAB Digital Audio Broadcasting
DFT Discrete Fourier Transform
DHT Discrete Hartley Transform
DMT Discrete MultiTone
DS-CDMA Direct Sequence CDMA
DSP Digital Signal Processing
ETSI European Telecommunications Standards Institute
FDD Frequency-Division Duplex
FDM Frequency-Division Multiplexing
FDMA Frequency-Division Multiple Access
FFT Fast Fourier Transform
FH-CDMA Frequency-Hopping CDMA
FIR Finite Impulse Response
GSM Global System for Mobile communications
IBI InterBlock Interference
ICI InterCarrier Interference
IDFT Inverse Discrete Fourier Transform
IEEE Institute of Electrical and Electronics Engineers
IIR Infinite Impulse Response
ISI InterSymbol Interference
LAN Local Area Network
LTE Long Term Evolution
LTI Linear Time-Invariant
xviii LIST OF ABBREVIATIONS
MA Multiple Access
MAN Metropolitan Area Network
Mbps Megabits per second
MC-MRBT MultiCarrier Minimum-Redundancy Block Transceiver
MC-RRBT MultiCarrier Reduced-Redundancy Block Transceiver
MIMO Multiple-Input Multiple-Output
MISO Multiple-Input Single-Output
MMSE Minimum Mean Square Error
MSC Mobile Switching Center
MSE Mean Square Error
MUI MultiUser Interference
OFDM Orthogonal Frequency-Division Multiplexing
OFDMA Orthogonal Frequency-Division Multiple Access
OLA Overlap-And-Add
PAM Pulse-Amplitude Modulation
PAN Personal Area Network
PAPR Peak-to-Average Power Ratio
PSD Power Spectrum Density
PSK Phase-Shift Keying
QAM Quadrature Amplitude Modulation
QPSK Quadrature PSK
SC Single Carrier
SC-FD Single-Carrier with Frequency-Domain equalization
SC-FDMA Single-Carrier Frequency-Division Multiple Access
SC-MRBT Single-Carrier Minimum-Redundancy Block Transceiver
SC-RRBT Single-Carrier Reduced-Redundancy Block Transceiver
SGSN Serving GPRS Support Node
SIMO Single-Input Multiple-Output
SISO Single-Input Single-Output
SNR Signal-to-Noise Ratio
SVD Singular Value Decomposition
VDSL Very high-speed Digital Subscriber Line
TDD Time-Division Duplex
TDM Time-Division Multiplexing
TDMA Time-Division Multiple Access
TMUX Transmultiplexer
UMTS Universal Mobile Telecommunications System
LIST OF ABBREVIATIONS xix
WAN Wide Area Network
Wi-Fi Wireless Fidelity
WiMAX Worldwide interoperability for Microwave ACCess
WLAN Wireless Local Area Network
WPAN Wireless Personal Area Network
WSS Wide-Sense Stationary
xDSL high-speed Digital Subscriber Line
ZF Zero-Forcing
ZP Zero-Padding
ZP-OFDM Zero-Padding OFDM
ZP-OFDM-OLA ZP-OFDM OverLap-and-Add
ZP-SC-FD Zero-Padding SC-FD
ZP-SC-FD-OLA ZP-SC-FD OverLap-and-Add
ZP-ZJ Zero-Padding Zero-Jamming
xxi
List of Notations
Scalars Lowercase letters, e.g., x
Vectors Lowercase boldface letters, e.g., x
Matrices Uppercase boldface letters, e.g., X
Definition
N Set of natural numbers, which is defined as N {1, 2, 3, . . .}
Z Set of integer numbers
R Set of real numbers
C Set of complex numbers
t Real-valued variable representing continuous time
n Integer number representing discrete time
j Imaginary unit, j2 = −1
ω Angular frequency
δ(t) Dirac impulse
δ[n] Kronecker delta
WM Unitary DFT matrix of size M × M
IM Identity matrix of size M × M
em Canonical vector, e.g., e0 = [1 0 . . . 0]T
(·)T Transpose of matrix (·)
(·)H Hermitian (conjugate) transpose of matrix (·)
0M×N M × N matrix with all entries equal to 0
E[(·)] Expected value of (·)
Z {(·)} Z -transform applied to (·)
Z −1 {(·)} Inverse Z -transform applied to (·)
tr{(·)} Trace of matrix (·)
rank{(·)} Rank of matrix (·)
diag{(·)} Diagonal matrix whose entries in its diagonal are (·)
ker{(·)} Kernel (Null space) of matrix (·)
R{(·)} Range (Column space) of matrix (·)
F {(·)} Fourier transform of (·)
[(·)]↓N Decimation operator by N
[(·)]↑N Interpolation operator by N
[(·)]ml Entry of matrix (·) in the mth row and lth column
DFT{(·)} Discrete Fourier transform of sequence (·)
1
CHAPTER 1
Channel
Receiver
version of the transmitted waveform since the latter suffers attenuation and other wave propagation
effects, and interferences caused by environmental noise and by other signals being transmitted
through the same channel.
At the receiver end, all the strategies utilized at the transmitter to improve the channel usage
are undone. The received waveform is converted to bits by the digital demodulator, the channel decoder
is responsible for correcting some bits that were erroneously detected, and then the source decoder
undoes the compression process, which is not a perfect reversion process in cases of using lossy
compression schemes (as explained before), generating an output signal as close as possible to the
input signal.
The next section addresses the OFDM, a transmission scheme that enables efficient trans-
mission of symbols (waveforms) through the channel.
• Asymmetric DSL (ADSL) – the asymmetric term means that upstream (i.e., from costumer to
network) and downstream (i.e., from network to costumer) data rates are different. Nowadays,
evolved versions of the ADSL, such as the ADSL2+M, can achieve upstream rates up to
3.5 Mbps (megabits per second) and downstream rates up to 24 Mbps.
• Very high-speed DSL (VDSL) – the first VDSL systems, known as VDSL1, provided data
rates higher than the ADSL. The drawback of VDSL1 is that its data rate decreases too fast
as the distance from the subscriber premises to the network increases, limiting its usage to
short local loops. Nowadays, VDSL2 systems can achieve upstream rates up to 10 Mbps and
downstream rates up to 50 Mbps, and perform quite similar to the ADSL2+ when transmitting
over long distances, unlike VDSL1.
Both ADSL and VDSL use discrete multitone (DMT), which is essentially a sophistication of
OFDM.
Figure 1.2 depicts these main classes according to their coverage area, and also shows examples
of wireless standards. Note that some technologies may appear in more than one class of wireless
networks, since they may receive upgrades that extend their original coverage area. For instance,
3 Some of these classes also exist for wired systems. Indeed, PAN and LAN can be used for both wired and wireless systems. In
cases when one wants to refer just to the wireless part of these systems, the terms wireless PAN (WPAN) and wireless LAN
(WLAN) can be used.
1.3. ORTHOGONAL FREQUENCY-DIVISION MULTIPLEXING 5
WAN
IEEE 802.20 ETSI GSM (2G) UMTS (3G)
(Mobile-Fi) 3GPP
MAN LTE-Advanced (4G)
the WiMAX (worldwide interoperability for microwave access) is usually considered an example of
MAN, but it can also be seen as a WAN system.
Many wireless network standards use OFDM in the air interface. Two examples are the LTE
(long-term evolution) and WiMAX standards. LTE was designed to fully replace the 3rd generation
(3G) networks for mobile communications. The WiMAX, although originally conceived to provide
wireless broadband services to homes, has been upgraded to be employed by mobile phones as access
method in recent years, competing with LTE.
(A) WiMAX
The main target of IEEE 802.16 standard, known as WiMAX, is to deliver wireless high-speed
Internet access over longer distances than the ones supported by the IEEE 802.11 standard,
commonly known as Wi-Fi (wireless fidelity). Indeed, WiMAX provides a MAN with wireless
broadband service in an area of 50 km (about 30 miles) of radius. The WiMAX is guided by an
association called WiMAX Forum and its data rates can reach up to 40 Mbps for low mobility
access and up to 15 Mbps for mobile access. The capabilities and coverage area of WiMAX
systems, especially due to the amendment e of IEEE 802.16 standard (IEEE 802.16e, also
called Mobile WiMAX), increased in such a way that the IEEE 802.20 standard was put to
hibernation.
WiMAX employs an adaptive modulation scheme as illustrated in Figure 1.3 in which the
digital modulator is adjusted according to the signal-to-noise ratio (SNR). Through low-SNR
channels, usually consisting of channels where the user is far from the base station, a sparser
modulation scheme, the quadrature phase-shift keying (QPSK), is used. On the other hand,
users near the base station are likely to have high-SNR channels and, therefore, transmissions
with higher-order modulation schemes, such as the 16- or 64-quadrature amplitude modulation
6 1. THE BIG PICTURE
High SNR
Medium SNR
Low SNR
64-QAM
16-QAM
QPSK
(QAM), are allowed. In addition, the multiple access scheme used in WiMAX is the orthogonal
frequency division multiple access (OFDMA).
Other goals of WiMAX are:
(B) LTE
LTE is a standard for wireless data communication that is capable of overcoming some limi-
tations of GSM/UMTS (global system for mobile communications/universal mobile telecom-
munications system) standards. It was developed by the 3rd Generation Partnership Project
(3GPP) and its first version was established in December 2008, when 3GPP Release 8 was
frozen. The main motivations for LTE are the user demands for higher data rates and quality
of service, the necessity of optimizing the packet-switched system, and the demands for cost
and complexity reduction.
1.3. ORTHOGONAL FREQUENCY-DIVISION MULTIPLEXING 7
Δf
Similar to the WiMAX, LTE uses OFDMA in the downlink connection. However, in the
uplink connection, LTE opted for using the single-carrier frequency-division multiple access
(SC-FDMA). In addition, LTE has very low latency and can operate on different bandwidths:
1.4, 3, 5, 10, 15, and 20 MHz.
3GPP Releases are continuously provided such as the most recent version called Release 11. It is
worth mentioning that from Release 10 and beyond, the LTE is usually called LTE-Advanced,
since it became IMT-Advanced compliant. Some of the key features of IMT-Advanced com-
pliant standards are:
OFDM subcarriers should occupy most of the available bandwidth so that the amount of symbols
transmitted is maximized.
In the next section we introduce the concept of cellular division, which was crucial to the
success (widespread use) of mobile communications.
These MA methods are briefly described in the following subsections. Note, however, that
they are not mutually exclusive. Indeed, the most successful 2G system, the GSM, is a hybrid of
TDMA and FDMA. In GSM, each 200 kHz of channel bandwidth is shared among 8 users using
TDMA.
1.5.1 TDMA
In TDMA, the separability among users occurs in the time domain, i.e., each user receives a time
slot (or a set of them) for his transmission. Figure 1.6 illustrates two users sharing the same medium
using a TDMA scheme. The duration of each slot is T seconds, and each user receives two of them.
During the slots that User 1 is transmitting, User 2 is in silence, and vice versa. The signal that arrives
at the receiver, assuming the channel does not introduce any distortion to the transmitted signals, is
the superposition of the signals sent by the two users. Therefore, in TDMA the transmitters must
be tightly synchronized in order to avoid interference among them. It is clear that the receiver must
10 1. THE BIG PICTURE
be synchronized with the transmitters as well, since it must know how to properly chop the received
signal in order to isolate the signals sent by the different users.
T T T T
T T T T
User 1
t
User 2
t Received Signal
(Without Distortion)
1.5.2 FDMA
Unlike TDMA, in FDMA users may transmit all the time. The separability among them occurs in
the frequency domain, i.e., for each user it is assigned a different frequency band. Figure 1.7 depicts
two users employing an FDMA scheme to share the channel resources. Both users transmit over
the channel using a bandwidth of f Hz, but the central frequency of User 1 is f1 , whereas it is f2
for User 2. Clearly, f1 and f2 must be distant enough from each other in order to avoid interference
between the two users. At the receiver side, the signals belonging to the two users can be separated
through bandpass filters that dramatically attenuate all frequencies, except the ones that fall within
the desired band.
As the number and types of services available to users increase, the fixed channel assignment
inherent to FDMA and TDMA becomes less efficient. As a result, more spectrum would be required
with such fixed assignment. Even if some kind of flexible channel assignment is incorporated in the
current FDMA and TDMA schemes, there is always a fixed upper bound in the number of users
that can be served.
1.5.3 CDMA
The CDMA, sometimes called spread-spectrum, appears in two distinct forms: the direct sequence
CDMA (DS-CDMA) and the frequency-hopping CDMA (FH-CDMA).
In DS-CDMA, the spectrum of the baseband signal is spread to occupy a wider bandwidth,
as depicted in Figure 1.8. In this figure, the power spectrum density (PSD) is represented as a
function of the frequency. Before transmitting the waveform conveying information, this waveform
1.5. MULTIPLE ACCESS METHODS 11
Δf Δf
Δf Δf
User 1
f1 f2 f
f1 f2 f
User 2
f1 f2 f Received Signal
(Without Distortion)
is modified in order to spread it over a wider frequency band. At the receiver, the reverse process
is performed and the spread-spectrum waveform is converted back to the original waveform from
which the information can be extracted.
PSD PSD
Original
Signal Spreading
Spread
Original
Signal
Despreading
Frequency Frequency
One of the main benefits from using spread-spectrum is reducing narrowband interference.
For instance, consider Figure 1.9 in which the spread original signal suffers interference due to
two narrowband interferers. In this case, the de-spreading process performed at the receiver de-
spreads the original signal and also spreads the interferers, as depicted in Figure 1.10. Consequently,
a significant portion of the signal power corresponding to the interferers is spread over frequency
bands different from the ones used by the original signal, thus reducing the inband interference.
Another main advantage of CDMA is that it allows users to share all the available bandwidth
simultaneously. Hence, since the signal transmitted by each user is wideband, CDMA must resolve
such type of wideband interference in order to guarantee the separability among users. Indeed,
CDMA introduces a new domain, called code domain, in which multiple users can be fully separated.
12 1. THE BIG PICTURE
PSD Narrowband
Interferers
Spread
Original
Signal
Frequency
Original
PSD Signal
Spread
Narrowband
Interferers
Frequency
Figure 1.11 illustrates a DS-CDMA passband-transmission scheme. In this case, the mth
user, m ∈ N, intends to transmit a sequence of symbols bm (nTb ), in which Tb denotes the symbol
duration and the sequence of symbols is indexed by n ∈ Z. Before transmitting, the symbols bm (nTb )
are multiplied by a code sequence cm (kTc ), where k ∈ Z represents the sequence index and Tc
corresponds to the duration of each element of the code sequence. In addition, cm (kTc ) is a very simple
sequence whose elements are 1 or −1. The main difference between the sequences bm (nTb ) and
cm (kTc ) is that the sampling rate of the latter is much higher, i.e., Tc Tb . Hence, the signal resulting
from the multiplication of bm (nTb ) by cm (kTc ) is a spread-spectrum signal whose bandwidth is
increased by a factor equal to Tb /Tc , also known as spreading factor or processing gain. Finally, this
resulting signal modulates the carrier cos (ωc t), where ωc is the carrier frequency, resulting in a
passband signal ready to be transmitted. The key point that guarantees the separability among users
is that these code sequences are unique, i.e., a different code sequence is assigned to each user, and
they are also orthogonal to each other.4
4 In fact, due to some transmission issues, such as synchronization, orthogonal code sequences are sometimes exchanged by (or
used together with) pseudonoise sequences.
1.5. MULTIPLE ACCESS METHODS 13
bm (nTb )
Note that while TDMA and FDMA have a maximum number of simultaneous users, CDMA
does not have such a constraint. Nevertheless, in CDMA, if the number of users increases the service
smoothly degrades.
In cellular systems, CDMA allows soft handover between neighboring cells. Close to the
frontier, where the signal is weaker, the user communicates with two base stations simultaneously so
that the diversity helps compensating for signal degradation at cell edges. In TDMA and FDMA
neighboring cells must use different frequencies because they control interference based on spatial
attenuation of the signals (frequency reuse). In CDMA, all cells use the same frequency range, elim-
inating the necessity for frequency-use planning, whereas TDMA and FDMA may use adaptive
frequency reallocation. In addition, CDMA requires strict power control and base station synchro-
nization, and allows intercell interference to be suppressed at the receiver.
Another type of spread-spectrum technique is the FH-CDMA, in which each user employs
a different frequency band within a given time frame. During transmission, the user frequency
band hops to different bands according to a prescribed hopping pattern (code). The receiver hops
synchronously with the transmitter with the knowledge of the code. The hopping can be slow, where
hopping occurs at the symbol rate, or fast where more than one hop occurs during symbol duration.
The latter case is more difficult to implement.
As a rule, wideband MA schemes can operate in the frequency range of existing narrowband
services and allow flexibility in the number of users and services provided to each user.They also allow
improved interference rejection, originated from multiuser, multipath, and narrowband interferences.
On the other hand, wideband MA systems require more advanced technology for implementation.
Figure 1.12 summarizes how the radio resources are assigned to one user for TDMA, FDMA,
and CDMA schemes. As we have already discussed, the separability among the multiple users occurs
in one of the following domains: time, frequency, or code domains. Therefore, in TDMA the user
receives a time interval for his transmission, which may use the whole channel bandwidth. In FDMA,
a frequency band is dedicated for the user’s transmission. In CDMA, a single user may transmit all
the time and using the entire channel bandwidth, but with a unique code.
1.5.4 OFDMA
In some standards, sets of OFDM subchannels can be assigned to distinct users leading to an MA
scheme known as orthogonal frequency-division multiple access (OFDMA). In the simplest case,
14 1. THE BIG PICTURE
Frequency Frequency
Slot
Slot
Time Time
Code Code
(a) TDMA. (b) FDMA.
Frequency
Slot
Time
Code
(c) CDMA.
Figure 1.12: Channel sharing for: (a) TDMA; (b) FDMA; (c) CDMA.
multiple access can be implemented in a TDMA format, where at a given time slot a specific user
is allowed to employ all subchannels for his transmission, as illustrated in Figure 1.13. As depicted
in Figure 1.14, it is also possible to assign distinct frequency bands for different users provided the
users know in which subchannels (bands) they can transmit at a given time slot.
1.5. MULTIPLE ACCESS METHODS 15
Frequency
·
User 1
··
User 2
User 3
···
Time
Frequency
·
User 1
··
User 2
User 3
···
Time
The most efficient way to assign the subchannels to multiple users is through random assign-
ment, which guarantees that all users enjoy approximately the same quality of service. Figure 1.15
illustrates an OFDMA scheme with random assignment of subchannels. As depicted in this fig-
ure, the subchannels used by each user may change at each time slot and, therefore, this kind of
OFDMA scheme avoids users to get stuck in low-quality subchannels. Note, however, that the
OFDMA schemes that are usually employed in standards are the ones depicted in Figures 1.13
and 1.14. This is justified by some issues and limitations concerning the OFDMA with random
assignment. For example, some multiple-antenna transmission schemes, especially those transmis-
sions with diversity, cannot be employed in OFDMA with random assignment, since they usually
require the transmission over adjacent subchannels.
16 1. THE BIG PICTURE
Frequency
·
User 1
··
User 2
User 3
···
Time
In this section we have seen how different users can transmit over the same medium. But
what if the users want to be capable of both transmitting and receiving data? In this case, duplex
methods must be used.
1.6.1 TDD
As illustrated in Figure 1.16, TDD schemes assign different time intervals to uplink and downlink
connections. Both of these connections can use the whole channel bandwidth during their trans-
missions. It is common practice to separate the uplink and the downlink connections in TDD by
a time interval known as guard time, which avoids interference between these two connections that
might be caused by propagation effects such as multipath, which are addressed in the next section.
5That is, bidirectional communications systems also have uplink and downlink connections like cellular systems. The difference
can be on the elements that are at the ends of these connections, which can be computers and Internet service providers instead
of mobile stations and base stations.
1.7. WIRELESS CHANNELS: FADING AND MODELING 17
1.6.2 FDD
Figure 1.17 illustrates the FDD method in which different frequency bands are assigned to the
uplink and downlink connections.
1.7.1 FADING
The main effects on transmitted signals inherent to wireless channels can be summarized in one
word: fading. Fading is a phenomenon concerning the time-variation of the channel strengths. If
these variations are due to transmissions over long distances (ranging from hundreds to thousand
meters), then they are known as large-scale fading, whereas the term small-scale fading is used for
channel variations due to relative movements of transmitter/receiver over short distances (of the
order of the carrier wavelength).
Frequency
ϕ2
Uplink Downlink
ϕ1
t1 t2 t3 t4 Time
Guard Time
Frequency
ϕ4
Uplink
ϕ3
ϕ2
Downlink
ϕ1
t1 t2 Time
Some examples of wave-propagation effects that fall into the category of large-scale fading
are the path loss and the shadowing effects. Path loss, also known as path attenuation, is the power
reduction of an electromagnetic signal as it propagates through the medium, i.e., longer distances
between transmitter and receiver leads to lower power of the received signal, assuming that the power
of the transmitted signal does not vary. The path loss Ploss is defined as
Pt
Ploss , (1.1)
Pr
where Pt and Pr stand for the power of the electromagnetic signal at the transmitter and receiver,
respectively. The path loss, measured in decibels, has a linear dependence on log10 (d), where d is
the distance that the signal travels. The other effect mentioned above, namely shadowing, is a fading
caused by the obstruction of the line-of-sight between transmitter and receiver. The shadowing
generates signal variations that are usually modeled by log-normal distribution.
There are two main types of small-scale fading: the frequency-selective and the flat fading. If
different frequency components of the electromagnetic signal are affected differently by the channel,
such effect corresponds to frequency-selective fading. Otherwise, in cases the variation induced on the
signal by the channel does not depend on/vary with the frequency of the electromagnetic wave, then
such effect falls into the category of flat fading.
The multipath fading is the most common type of small-scale fading that is present in mobile
communications. It originates from the transmitted-signal reflections in local buildings, hills, or
structures around a few hundred wavelengths from the mobile. Therefore, the multipath fading
can be understood as variations on the signal caused by interferences from attenuated and delayed
versions of the same signal (the reflections). Figure 1.18 illustrates an example in which two reflected
1.7. WIRELESS CHANNELS: FADING AND MODELING 19
signals are interfering with the signal received from the line-of-sight. This figure depicts the three
waveforms independently, but it is clear that the received signal y(t) is a summation of the three
signals. Note also that in the time interval [0, Ts ], in which Ts stands for the duration of one symbol
generated by the digital modulator, the interference among the three signals will be constructive,
whereas some destructive interferences occur within other intervals, such as [Ts , 2Ts ].
Received Signals
... ...
0 Ts 2Ts time
Figure 1.19 depicts an example of power loss in wireless environment. For a given transmitted
power Pt , such figure illustrates how the received power Pr varies as a function of the transmitter-
receiver distance d. These power variations occur due to the path loss, shadowing, and multipath
fading phenomena.
1.7.2 MODELING
In rough environments, such as urban areas, frequency-selective fading occurs due to the reflections
of the transmitted signal that arrive with distinct delays at the receiver (multipath). In these cases,
the propagation medium (channel) is usually modeled as a linear system with memory, which can
be characterized by its impulse responses. Indeed, since wireless channels are time-varying, their
impulse responses h(t, τ ) can change along the time t. Thus, the signal arriving at the receiver at a
time instant t ∈ R can be written as
t
y(t) = s(τ )h(t, τ )dτ, (1.2)
0
where h(t, τ ) corresponds to the channel response to an impulse applied at the instant t and s(τ )
represents the transmitted signal (concatenation of waveforms representing symbols). In addition,
20 1. THE BIG PICTURE
Pr
(dB) Mulipath - Fading
Pt
Path-loss + Shadowing
Path-loss
d (log)
it is common practice to consider that the memory of h(t, τ ) is finite since, due to path loss, not all
existing reflections will have enough power to be sensed at the receiver.
If one is interested only in the discrete-time representation of the signals, the signal arriving
at the receiver at instant k ∈ Z can be written as
k
y(k) = s(i)h(k, i), (1.3)
i=0
where h(k, i) is the channel response to an impulse applied at instant k, and s(i) represents the
transmitted signal as a sequence of symbols.
Note that, in practical communications systems, the sequence of symbols s(i) are usually
divided into blocks, and then each block is transmitted through the channel, as will be shown in the
next section.
y(n − 1)
s(n) s(n − 1)
y(n)
interference (ISI) as well as interblock interference (IBI) at the receiver end. In this case, the currently
received block y(n) and the previously received block y(n − 1), corresponding to the transmitted
blocks u(n) and u(n − 1), respectively, will have an overlap as depicted in Figure 1.20. Then, the
SISO receiver is responsible for eliminating both ISI and IBI and detecting the transmitted symbols,
yielding a data block ŝ(n) that must be as close as possible to the originally transmitted block s(n).
In the case of wideband transmission, the ISI can be severe enough to make the SISO receiver
very complex to implement. Roughly speaking, each transmitted symbol would spread over the time
slot(s) of the neighboring symbol(s), turning their correct detection more challenging.
A naive, but widely used, solution to avoid interference among symbols of different blocks
(i.e., IBI) is to allow a guard period between each block transmission. In Figure 1.20, this solution
corresponds to separate each block u(n) by an amount of time that is sufficiently large to guarantee
that overlaps do not occur between the received blocks y(n), n ∈ Z. The drawback of such a solution
is the reduction of the data-transmission rate. This reduction can be significant in cases the blocks
u(n) are not much larger than the guard period.
A more general block-transmission framework is illustrated in Figure 1.21. This MIMO
model for block transmission encompasses many block-transmission schemes, including the SISO
model, multicarrier schemes, and multiple antenna configurations. In the MIMO model for block
transmission, a given data block s is modified by a MIMO Transmitter, yielding the block u whose
length is greater than the length of s. This larger length is due to many different reasons, such as the
replication of previously transmitted blocks, in case the transmitter has memory, or just by the use
of a guard period. The MIMO channel model can be described through a MIMO transfer function
possibly with memory, generating ISI and IBI. It is up to the MIMO receiver to process the received
signal block y in order to generate a reliable estimate of the transmitted signal block ŝ.
22 1. THE BIG PICTURE
s MIMO
u MIMO
y MIMO
ŝ
The MIMO model might represent a wide range of signal-processing tasks. In this book the
main type of MIMO processing is the multicarrier transmission, which consists of transmitting each
symbol in a block through a narrowband subcarrier. The benefits from using this technique are the
following:
• Each subcarrier illuminates a narrow range of channel frequencies, so that the equivalent
subchannel appears to be flat. This turns the equalization for each subcarrier much simpler.
• Since it consists of a block transmission, the time support for transmission of a symbol modu-
lated by a subcarrier is roughly multiplied by the number of symbols in the block. That means
there is much more time to decode the information conveyed by each subcarrier, reducing or
even avoiding ISI within a block.
• If a guard period is inserted to avoid IBI, the time overhead is relatively low as long as each
block carries several symbols. The guard time is a function of the length of the channel impulse
response (time-delay spread).
As we have seen, if ISI and IBI are not tackled, they can deteriorate the performance of
communications systems. We have also seen that IBI can be avoided just by using a simple guard
period between the blocks. In the following section, we will introduce systems that tackle the ISI
problem in a simple and efficient manner.
jω
H e
Zoom
Δω
Δω
ω
Figure 1.22: Multicarrier system dividing the channel bandwidth into non-overlapping flat subchannels.
The basic idea of multicarrier systems, whose most popular implementation is the OFDM, is
the transport of information through a wideband channel by energizing it with several narrowband
subcarriers simultaneously.The success of this technique relies on the partition of the physical channel
into non-overlapping narrowband subchannels through a transmultiplexer, as will be explained in
the next chapter. If the subchannels are narrow enough, the associated channel response in each
subchannel-frequency range appears to be flat, thus avoiding the use of sophisticated equalizers.
Figure 1.22 illustrates the effect of splitting a wideband channel in flatter subchannels.
Figure 1.23 depicts a transmultiplexer implementing a 4-band multicarrier system, i.e., the
channel is divided into 4 subchannels. At the transmitter end, a set of symbols (represented by colors)
is prepared for transmission through distinct subcarriers, where each subcarrier is represented by a
finite-impulse response (FIR) filter whose transfer function is denoted by Fi (z), i ∈ {0, 1, 2, 3}. At
the receiver side there are related FIR filters Gi (z). The FIR filters can be thought as narrowband
filters with distinct central frequencies, so that the symbols sent at different subcarriers travel through
different subchannels, thus not interfering with each other. In addition, Figure 1.23 assumes that
perfect transmission (reconstruction) is possible, i.e., the symbol transmitted at each subcarrier is
perfectly recovered at the receiver.
24 1. THE BIG PICTURE
F0 (z) G0 (z)
Noise
F1 (z) G1 (z)
Channel
F2 (z) G2 (z)
F3 (z) G3 (z)
In practical systems, perfect reconstruction is usually not achievable due to degradations caused
by physical-channel and noise effects, as well as power limitations. In this case, the subchannel
division allows, whenever possible, the exploitation of the signal-to-noise ratio (SNR) in the distinct
subbands by managing their data load in each subchannel. Indeed, if the transmitter has knowledge
about the SNR at the channel output for each subcarrier, then some loading scheme could be applied,
as illustrated in Figure 1.24. As can be observed in this figure, at the subcarriers with higher SNR it is
possible to transmit symbols belonging to higher-order modulations, such as an 8-PSK modulation,
whereas low SNR ones use lower-order modulation schemes, such as binary-PSK (BPSK). For very
low SNR subcarriers, it can be even decided not to transmit any symbol at all.
In a general setup some redundancy is required at the transmission in order to keep the
equalization as simple as possible. This is an important issue that will be addressed in the following
chapters. In addition, several methods for jointly optimizing the transmitter and receiver of FIR
MIMO systems can be employed to combat near-end crosstalk and additive-noise sources.
F0 (z) G0 (z)
Noise
F1 (z) G1 (z)
Channel
F2 (z) G2 (z)
F3 (z) G3 (z)
ŝ = GHF s, (1.4)
where F represents the precoder matrix applied at the transmitter, G represents the postcoder matrix
applied at the receiver, and H is the MIMO channel matrix. In this simplified description, all matrices
are considered memoryless so that each input-signal vector is processed independently. In addition,
allowing the existence of additive noise v at the channel output and assuming that F modifies and
inserts a prefix on s, the block transmission can be modeled as
F s +
ŝ = GH Gv , (1.5)
u v̄
ŝ = GHFs
+ Gv
u v̄
Prefix
ŝ G H u G v
0
ŝ s v̄
0
s1 ŝ1
···
···
Tx Rx
···
···
sN ŝM
···
···
···
···
···
Figure 1.27: General setup of multiple block MIMO precoding with multiple antennas.
CHAPTER 2
Transmultiplexers
2.1 INTRODUCTION
The proposal of new techniques for channel and source coding, along with the development of
integrated circuits and the use of digital signal processing (DSP) for communications have allowed
the deployment of several communications systems to meet the demands for transmissions with
high data-rates. Typical DSP tools, such as digital filtering, are key to retrieving, at the receiver end,
reliable estimates of signals associated with one or several users who share the same physical channel.
There are various classes of digital filters. Those employed in communications systems can
be either fixed or adaptive, linear or nonlinear, with finite impulse response (FIR) or with infinite
impulse response (IIR), just to mention a few. Among such classes of systems, fixed, linear, and
FIR filters are rather common in practice because of their simpler implementation, good stability
properties, and lower costs as compared to other alternatives.
Nonetheless, modern communications systems require more sophisticated techniques, thus
calling for more features than fixed, linear, and FIR filters can offer. In this context, multirate signal
processing adds some degrees of freedom to the standard linear time-invariant (LTI) signal process-
ing through the inclusion of decimators and interpolators. These degrees of freedom are crucial to
develop some interesting representations of communications systems based on filter banks, especially
multicarrier transceivers. A filter bank is a set of filters (usually LTI FIR filters) sharing the same
input-output pair and internally employing decimators and interpolators.
Filter-bank representations are widely used in source coding and spectral analysis. In com-
munications, the transmultiplexer (TMUX) configuration can be employed to represent multicarrier
or single-carrier transceivers, and can be considered a system dual to the filter-bank configuration
in the sense that the signal processing which takes place at the input of a filter bank actually appears
at the output of a TMUX, and vice versa. Indeed, several practical systems can be modeled using
TMUXes.
Unlike filter banks that usually require sharp frequency-selective subfilters, practical multicar-
rier transceivers can be modeled as TMUXes which use short-length subfilters with poor frequency
selectivity. In the majority of practical cases, these transceivers are implemented as memoryless block-
based transceivers. The most commonly used memoryless block-based transceivers are the orthogonal
frequency-division multiplexing (OFDM) and the single-carrier with frequency-domain equaliza-
tion (SC-FD) systems.
The main feature of OFDM-based transceivers is the elimination of intersymbol interfer-
ence (ISI) with low computational complexity, i.e., using just a small amount of numerical opera-
30 2. TRANSMULTIPLEXERS
tions to undo the harmful effects induced by frequency-selective channels. A competing alternative
to OFDM is the SC-FD transceiver, which presents lower peak-to-average power ratio (PAPR)
and lower sensitivity to carrier-frequency offset (CFO), as explained in [63, 87]. In addition, for
frequency-selective channels, the bit-error rate (BER) of SC-FD can be lower than for its OFDM
counterpart, particularly for the cases in which the channel has high attenuation at some subchannels.
In this chapter some key multirate signal-processing tools are revised (Section 2.2) aiming at
their use in the modeling of communications systems (Section 2.3). These tools will be particularly
utilized to represent OFDM and SC-FD systems, as well as to introduce some initial results related
to what is beyond OFDM-based systems, namely: the memoryless LTI block-based transceivers
using reduced redundancy (Section 2.4).
where the integer number k denotes the time index at the new sampling rate. In the frequency
domain, the effect of interpolation can be described as (see, for example, [17])
in which
X(ejω ) F {x(n)}
= x(n)e−jωn (2.3)
n∈Z
is the discrete-time Fourier transform of the sequence x(n), with ω ∈ R denoting the frequency
variable.1
1 It is assumed that the discrete-time Fourier transform of the sequence x(n) exists, i.e., the series in expression (2.3) is convergent
for all real-valued scalar ω. For instance, an absolutely summable (i.e., an 1 -signal) x(n) is sufficient to guarantee the convergence
of the series.
2.2. MULTIRATE SIGNAL PROCESSING 31
The decimation by a factor N consists of discarding N − 1 samples from each non-overlapping
block containing N samples of the input signal. The resulting signal has a sampling rate N times
lower than the sampling rate of the original signal. Indeed, given the signal s(n), the decimated
signal sdec (k) is defined by
for all integer number k. In the frequency domain, it is possible to show that the decimated signal
is represented by (see, for example, [17])
1 j ω−2π n
where N {0, 1, · · · , N − 1}. Unlike the interpolation, the decimation is a periodically time-
varying operation.
|S(ejω )| |Sint(ejω )|
ω ω
−2π −π 0 π 2π −2π −π 0 π 2π
N
s(n) sint(k)
n k
−6 −4 −2 0 2 4 6 −6 −4 −2 0 2 4 6
It is worth mentioning that a more appropriate nomenclature for the interpolation and dec-
imation processes just described should be upsampling and downsampling, reserving the nouns in-
terpolation and decimation for the cases in which a filtering process is also present. However, it is
rather common in the literature and in practice to use interchangeably the nomenclatures upsam-
pling/interpolation and downsampling/decimation. We will follow this practice, but the reader will
be able to identify easily when a filtering process takes place or not.
Figures 2.1 and 2.2 depict the respective effects of interpolation and decimation by a factor
N = 2 in both time and frequency domains. These signals are only for illustration purposes and
they do not represent true time-frequency pairs. By examining Figures 2.1 and 2.2 it is possible to
32 2. TRANSMULTIPLEXERS
|S(ejω )| |Sdec(ejω )|
ω ω
−2π −π 0 π 2π −2π −π 0 π 2π
N
s(n) sdec(k)
n k
−6 −4 −2 0 2 4 6 −6 −4 −2 0 2 4 6
verify that, in order to avoid aliasing due to decimation and to eliminate the spectrum repetition
due to interpolation, a digital filtering operation is required before the decimation and after the
interpolation. The decimation filter narrows the spectrum of the input signal in order to avoid
that aliasing corrupts the spectrum of the resulting decimated signal. For a lowpass real signal,
for instance,
we have
to maintain the input signal information only at the low frequencies within
the range − N π π
, N , so that the spectrum at this range is not corrupted after decimation. The
interpolation filter smooths the interpolated signal sint (k), eliminating abrupt transitions between
non-zero and zero samples, which is the source of the spectrum repetitions (also known as spectral
images). The central frequencies of the spectrum repetitions are located at ± 2π N n, with n ∈ N .
Figure 2.3 illustrates how the decimation and interpolation operations are implemented in practice.
Determine Hint (z) = [H (z)]↑3 and Hdec (z) = [H (z)]↓3 , for all non-zero complex number z, in
which H (z) Z {x(n)} is the Z -transform of the sequence x(n). In addition, the notations [(·)]↑N
and [(·)]↓N denote the interpolation and decimation by N applied to (·), respectively.
2.2. MULTIRATE SIGNAL PROCESSING 33
Solution. We know that
H (z) = x(n)z−n
n∈Z
= 1 + 2z−1 + 4z−2 + 8z−3 + 16z−4 + 32z−5 + 64z−6 + 128z−7 , (2.7)
for all z = 0. The interpolation by a factor of 3 is equivalent to insert 2 zero-valued samples between
adjacent samples of x(n). Hence, we have
Hint (z) =1 + 0.z−1 + 0.z−2 + 2z−3 + 0.z−4 + 0.z−5 + 4z−6 + 0.z−6 + 0.z−8 + 8z−9
+ 0.z−10 + 0.z−11 + 16z−12 + 0.z−13 + 0.z−14 + 32z−15 + 0.z−16 + 0.z−17
+ 64z−18 + 0.z−19 + 0.z−20 + 128z−21
=1 + 2z−3 + 4z−6 + 8z−9 + 16z−12 + 32z−15 + 64z−18 + 128z−21
=H (z3 ), (2.8)
for all z = 0. The decimation by a factor of 3 will generate a discrete-time signal hdec (k) = h(3k) =
23k = 8k , if k ∈ {0, 1, 2}, or hdec (k) = 0, otherwise. Hence, we have
for all z = 0. 2
In multirate systems, there are very useful manners to manipulate the interpolation and dec-
imation building blocks. We are particularly interested in ways to commute the decimation and
interpolation operations with linear time-invariant filters. Some forms of commuting are based on
the so-called noble identities.
Figure 2.4 illustrates the building-block representations of the noble identities. In the inter-
polation process, instead of first upsampling the input signal and then filtering it, one can first filter
the input signal in a lower sampling rate and then upsample the resulting signal. This strategy allows
one to reduce the number of operations required by the overall interpolation process. As for the
decimation process, the decimator followed by a filter is equivalent to filter the input signal by the
34 2. TRANSMULTIPLEXERS
interpolated filter followed by the downsampling. These operations can be mathematically described
as
g1 (k) N N f1 (k)
gM −1 (k) N N fM −1 (k)
A widespread application of multirate systems is the filter-bank design. A filter bank consists
of a set of filters with the same input signal, or a set of filters whose outputs are added to form the
overall output signal, as depicted in Figure 2.5. The set of M ∈ N filters represented by the family
of impulse responses {gm (k)}m∈M , in which M {0, 1, · · · , M − 1}, is the so-called analysis filter
bank, whereas the set of filters represented by the family of impulse responses {fm (k)}m∈M is the
synthesis filter bank. It is possible to verify that the analysis filter bank divides the input signal in
subbands, generating narrowband signals which can be further decimated.The subband signals can be
employed for analyses and manipulations according to the particular application. For reconstruction,
the subband signals are interpolated and combined by the synthesis filter bank.
Transmultiplexers, also known as filter-bank transceivers, are considered systems dual to the
filter-bank configurations since the roles of analysis and synthesis filter banks are interchanged in
transmultiplexers. Indeed, the inputs of a transmultiplexer are first combined by the synthesis bank
and, after some further processing stages, the outputs are obtained as a result from the analysis bank,
as shown in Figure 2.6.
It is worth mentioning that this section is based on [17, 81], which contain a thorough
treatment of this subject.
synthesis filter bank
2.3. FILTER-BANK TRANSCEIVERS 35
v(k)
u(k) sm (i)fm (k − iN). (2.12)
(i,m)∈Z×M
y(k) h(j )u(k − j ) + v(k). (2.13)
j ∈Z
At the receiver end, the signal y(k) is processed in order to generate estimates of the transmitted
data according to
ŝm (n) gm (l)y(nN − l). (2.14)
l∈Z
By employing Equations (2.12), (2.13), and (2.14) it is possible to describe the relation between
the input signal sm (n) and its estimate ŝm (n), as given by
ŝm (n) = gm (l)h(j )sm (i)fm (nN − l − j − iN) + gm (l)v(nN − l). (2.15)
(i,j,l,m)∈Z3 ×M l∈Z
The description above is not the easiest one to analyze the system and draw conclusions. For
example, it is possible to employ a time-domain approach using matrix description, as described
in [70, 72]. Another approach is to apply polyphase decomposition in a Z -domain formulation as
described as follows.
and
where m ∈ M, and Fm (z) and Gm (z) are the Z -transforms of fm (k) and gm (k), respectively. The
transfer functions Fi,m (z) are the Type-I polyphase components of order N associated with Fm (z),
whereas the transfer functions Gm,i (z) are the Type-II polyphase components of order N associated
with Gm (z).
38 2. TRANSMULTIPLEXERS
Example 2.2 (Polyphase Decomposition) Let us consider the signal h(n) defined in Example 2.1.
Determine the Type-I polyphase decomposition of order 3 associated with the transfer function
H (z).
yielding
Observe that
Comparing
with the solution of Example 2.1, the reader should also notice that Hdec (z) = H0 (z) =
H0 (z3 ) ↓3 . This is a useful property that will be further exploited. 2
By using a matrix approach, we can rewrite Equations (2.16) and (2.17) as follows:
⎡ ⎤
F0,0 (zN ) ··· F0,M−1 (zN )
⎢ .. .. ⎥
F0 (z) · · · FM−1 (z) = 1 z−1 · · · z−(N −1) ⎣ .
..
. . ⎦
dT (z) FN −1,0 (z ) · · · FN −1,M−1 (z )
N N
F(zN )
= d (z)F(z ),
T N
(2.23)
⎡ ⎤ ⎡ ⎤⎡ ⎤
G0 (z) G0,0 (zN ) ··· G0,N−1 (zN ) 1
⎢ .. ⎥ ⎢ .. .. .. ⎥ ⎢ .. ⎥
⎣ . ⎦=⎣ . . . ⎦⎣ . ⎦
GM−1 (z) GM−1,0 (z ) · · · GM−1,N−1 (z )
N N z(N −1)
G(zN ) d(z−1 )
−1
= G(z )d(z
N
). (2.24)
2.3. FILTER-BANK TRANSCEIVERS 39
V (z)
z −1 z
S1 (z) Ŝ1 (z)
N N
F(z N ) z −1 z G(z N )
z −1 z
SM −1 (z) ŜM −1 (z)
N N
Now, by defining Sm (z) Z {sm (n)}, U (z) Z {u(k)}, X(z) Z {x(k)}, V (z) Z {v(k)},
Y (z) Z {y(k)}, and Ŝm (z) Z {ŝm (n)}, then one can write
⎡ ⎤
S0 (zN )
⎢ .. ⎥
U (z) = dT (z)F(zN ) ⎣ . ⎦, (2.25)
SM−1 (zN )
s(z)
X(z) = H (z)U (z), (2.26)
Y (z) = X(z) + V (z), (2.27)
⎡ ⎤
Ŝ0 (z)
⎢ .. ⎥ −1
⎦ = G(z )d(z )Y (z)
N
⎣ . . (2.28)
↓N
ŜM−1 (z)
ŝ(z)
Figure 2.7 illustrates the transceiver model utilizing the polyphase decompositions of the
transmitter and receiver subfilters. By employing the noble identities described in Section 2.2, it is
possible to transform the transceiver of Figure 2.7 into the equivalent transceiver of Figure 2.8.
The highlighted area of Figure 2.8 that includes delays, forward delays, decimators, interpola-
tors, and the SISO channel model can be represented by a pseudo-circulant matrix H(z) of dimension
40 2. TRANSMULTIPLEXERS
V (z)
z −1 z
S1 (z) U1 Y1 Ŝ1 (z)
N N
F(z) z −1 z G(z)
z −1 z
SM −1 (z) UN −1 YN −1 ŜM −1 (z)
N Pseudo-Circulant Channel N
H(z)
N × N, given by
⎡ ⎤
z−1 HN −1 (z) z−1 HN −2 (z) · · · z−1 H1 (z)
H0 (z)
⎢ H1 (z)
H0 (z) z−1 HN −1 (z) · · · z−1 H2 (z)⎥
⎢ ⎥
H(z) ⎢ .. .. .. .. .. ⎥, (2.29)
⎣ . . . . . ⎦
HN −1 (z) HN −2 (z) HN −3 (z) ··· H0 (z)
in which
H (z) Hi (zN )z−i and Hi (z) h(j N + i)z−j . (2.30)
i∈N j ∈Z
0≤j N +i≤L
Indeed, given the indexes m and l within the set N , the (m, l)th element of the matrix H(z),
denoted as [H(z)]ml , represents the transfer function from the lth input element of the highlighted
area shown in Figure 2.8 to the mth output element of this area. Hence, by assuming that v(k) = 0
for all integer number k, if Ul (z) is the lth input at the transmitter end of the highlighted area in
Figure 2.8 and Ym (z) is the mth output of this area at the receiver end, then
Ym (z) = z−l H (z)Ul (zN )zm
↓N
= zm−l H (z)Ul (zN )
↓N
= Ul (z) z H (z)
m−l
, (2.31)
↓N
2.3. FILTER-BANK TRANSCEIVERS 41
in which we have applied the noble identity described in Equation (2.11) and we also have considered
that the only non-zero input of the highlighted area in Figure 2.8 is Ul (z).
Therefore, based on Equation (2.31) and on the first type of polyphase representation of the
channel-transfer function, we can write
Ym (z)
[H(z)]ml =
Ul (z)
= zm−l H (z)
↓N
= zm−l Hi (zN )z−i
i∈N ↓N
= Hi (zN )zm−l−i
i∈N ↓N
= H0 (z )z
N m−l
+ H1 (zN )zm−l−1 + · · · + HN −1 (zN )zm−l−N +1 . (2.32)
↓N
We know that the decimation operation retains the first coefficient out of N coefficients within
a block, starting from the 0th element. In the Z -domain, this means that the decimation operation
keeps only the coefficients which multiply a power of zN . Thus, the j th coefficient of the decimated
signal corresponds to the (j N )th coefficient of the signal before the decimation. Another way of
interpreting this fact is that, given an index i0 ∈ N such that (m − l − i0 ) is a multiple of N, the
decimation operation which appears in expression (2.32) retains the i0 th term Hi0 (zN )zm−l−i0 and
decimates it, as illustrated in Example 2.2.
We also know that −(N − 1) ≤ m − l ≤ N − 1, since m and l are within the set N . Hence,
if m − l ≥ 0, then expression (2.32) yields
[H(z)]ml = H(m−l) (zN ) = H(m−l) (z). (2.33)
↓N
v(n)
From Figure 2.9, it is also possible to infer that the transfer matrix T(z) of the transceiver can
be expressed as
where we considered the particular case in which v(k) = 0 for all integer number k, inspired by the
zero-forcing (ZF) design. A transceiver is zero forcing whenever T(z) = αz−d IM , for some α ∈ C
and d ∈ N. Notice that, if there is no noise, a zero-forcing solution is able to retrieve a scaled and
delayed version of all transmitted signals.
An important observation about Figure 2.9 is that, in order to be able to recover a block with
M transmitted symbols, one must send through the channel at least M elements in a data block,
i.e., we must necessarily have N ≥ M (this fact explains the shapes of the boxes in Figure 2.9).
Nonetheless, if N = M (no redundancy is included), then the matrices F(z), H(z), and G(z) are
square matrices and, therefore, a zero-forcing solution would not be achieved using only FIR filters,
considering that the channel model is not a simple delay, as explained in [44, 45]. Hence, some
redundancy must be introduced in order to work with FIR transceivers.3
Now, let us assume that we choose N > L, i.e., the interpolation/decimation factor is greater
than or equal to the channel order L, a common situation in practice.4 Based on Equation (2.30),
we have that the only integer number j which satisfies the inequality constraint 0 ≤ j N + i ≤ L
is j = 0, which lead us to conclude that Hi (z) = h(i), for N
i ≤ L. On the other hand, if there
exists i > L within the set N , then Hi (z) = 0 since there is no term to be added in order to form
Hi (z). In other words, we can say that, for N > L, each element of the matrix H(z) will consist of
filters with a single (possibly null) coefficient. In this case, the pseudo-circulant channel matrix in
3 Employing IIR filters may bring about many drawbacks, such as instability issues. This is the reason why FIR transceivers are the
prevalent choice.
4 Usually, practical block transceivers use N = M + K, where K is an integer number larger than or equal to L.
2.3. FILTER-BANK TRANSCEIVERS 43
Equation (2.29) is represented by a first-order FIR matrix described as
⎡ ⎤ ⎡ ⎤
h(0) 0 0 ··· 0 0 · · · 0 h(L) · · · h(1)
⎢ h(1) h(0) 0 ··· 0 ⎥ ⎢ .. .. ⎥
⎢ ⎥ ⎢0 0 · · · 0 . . ⎥
⎢ .. .. .. .. .. ⎥ ⎢ ⎥
⎢ . . . . ⎥
. ⎥ ⎢ . . . . .. h(L)⎥
.
⎢ ⎢ .. .. .. .. ⎥
⎢ ⎥ −1 ⎢ ⎥
H(z) = ⎢h(L) h(L − 1) . . . · · · 0 ⎥ +z ⎢ 0 0 0 · · · 0 0 ⎥
⎢ ⎥ ⎢ ⎥
⎢ 0 ··· ··· 0 ⎥ ⎥ ⎢ ··· 0 ⎥
⎢ h(L) ⎢0 0 0 0 ⎥
⎢ . .. .. .. .. ⎥ ⎢. .. .. .. .. .. ⎥
⎣ .. . . . . ⎦ ⎣ .
. . . . . . ⎦
0 0 h(L) · · · h(0) 0 0 0 0 ··· 0
HISI HIBI
−1
= HISI + z HIBI . (2.38)
Notice that Equation (2.38) implies the following relation in the time domain:
where
The relationship described in Equation (2.39) makes clear the roles of the matrices HISI and
HIBI . Indeed, matrix HISI mixes the symbols transmitted in the current data block, i.e., such a matrix
introduces interferences among the current data-block symbols, while matrix HIBI mixes some of
the symbols transmitted in the past block. The channel output vector x(k) is the result of adding
both effects: ISI and IBI.
Another way to derive Equation (2.39) is by analyzing what happens in the time domain when
a signal u(k) passes through an FIR channel h(k) of order L. In this case, we know that the channel
output x(k) is the linear convolution between the signals u(k) and h(k), that is, x(k) = (h ∗ u)(k).
Hence, if we look at a block of size N > L containing the channel output signals (in other words,
if we examine the elements of the vector x(k)), then we can verify that the first L elements of this
block are affected by the last L elements of the previous block, due to the channel memory and the
way the linear convolution is computed.
It is worth pointing out that generalizations of standard multicarrier communications systems
may call for sophisticated transmultiplexer designs in which the transmitted signal is filtered by a
precoder with memory consisting of a MIMO FIR filter.The inherent memory at the transmitter can
be viewed as a kind of redundancy since a given signal block is transmitted more than once along with
neighboring blocks. Sophisticated transmitters may require more complex receivers, but they might
44 2. TRANSMULTIPLEXERS
allow a reduction in the amount of redundant signals necessary to attain zero-forcing solution, for
example. All of these facts indicate that communication engineers should master the TMUX-related
tools in order to pursue new advances in communications systems, especially regarding multicarrier
transceivers. The case of transceivers with memory will be addressed in Chapter 5. In this chapter,
we shall consider the widespread used memoryless systems.
The particular and very important case where the transceivers are memoryless, that is, F(z) =
F and G(z) = G, is addressed in this section. This case encompasses the memoryless block-based
transceivers, since these systems do not use data from previous or future blocks in the transmission
and reception processing of the current data block. That is, only the current block takes part in the
transceiver computations.The traditional OFDM and SC-FD transceivers are well-known examples
of memoryless block-based systems.
The non-overlapping behavior associated with memoryless transceivers is only possible if the
lengths of the FIR causal subfilters {fm (k)}m∈M and {gm (k)}m∈M are less than or equal to N .
Indeed, from Equations (2.16) and (2.17), we know that
[F(z)]im = fm (j N + i)z−j , (2.42)
j ∈Z
[G(z)]mi = gm (j N − i)z−j , (2.43)
j ∈Z
for all pairs of numbers (i, m) within the set N × M. Hence, the matrix F(z) will have memory
(i.e., will depend on z) if, and only if, there exists both a non-zero natural5 number j0 and a pair of
numbers (i0 , m0 ) ∈ N × M, such that fm0 (j0 N + i0 ) = 0, which occurs if, and only if, fm0 (k) is
a causal impulse response with length larger than N, since j0 N + i0 ≥ N. The same conclusion can
be drawn for the matrix G(z).
We shall briefly describe now the main memoryless LTI block transceivers which will be
considered throughout this book. Further details will be given in Chapters 3 and 4.
5The index j cannot be negative because we are only interested in causal subfilters.
0
2.4. MEMORYLESS BLOCK-BASED SYSTEMS 45
2.4.1 CP-OFDM
The OFDM transceiver employing cyclic prefix as redundancy (also known as cyclic-prefix OFDM,
or just CP-OFDM) is described by the following transmitter and receiver matrices, respectively:
0K×(M−K) IK
F H
WM , (2.44)
IM
ACP ∈CN×M
G EWM 0M×K IM , (2.45)
RCP ∈CM×N
where the integer number K denotes the amount of redundant elements, WM is the unitary M × M
discrete Fourier transform (DFT) matrix, that is,
W ml
[WM ]ml √M , (2.46)
M
In addition, IM is the M × M identity matrix, 0M×N is an M × N matrix whose entries are zero,
and E is an M × M diagonal equalizer matrix placed after the removal of the cyclic prefix and the
application of the DFT matrix.
As can be noted, the data block to be transmitted has length M, however, due to the prefix,
the transceiver actually transmits a block of length N = M + K, in which K must be larger than or
equal to the channel order L, i.e., one must necessarily have M ≥ K ≥ L so that the CP-OFDM
system works properly. The first K elements are repetitions of the last K elements of the inverse
discrete Fourier transform (IDFT) output in order to implement the cyclic prefix.
Matrix ACP adds and matrix RCP removes the related cyclic prefix. Note that, based on
Equation (2.38), the product RCP H(z)ACP Hc ∈ CM×M is given by
⎡ ⎤
h(0) 0 ··· 0 h(L) · · · h(1)
⎢ .. .. ⎥
⎢ h(1) h(0) ··· 0 0 . . ⎥
⎢ ⎥
⎢ .. .. . ⎥
⎢ . . . . h(L) ⎥
⎢ ⎥
⎢ .. .. ⎥
Hc = ⎢ h(L) h(L − 1) . . 0 ⎥ , (2.48)
⎢ ⎥
⎢ .. .. ⎥
⎢ 0 h(L) . . ⎥
⎢ ⎥
⎢ .. .. .. .. ⎥
⎣ . . . . 0 ⎦
0 ··· 0 h(L) ··· h(0)
46 2. TRANSMULTIPLEXERS
where we can observe that RCP removes the IBI (there is no dependency on z anymore), whereas
matrix ACP right-multiplies the resulting memoryless matrix RCP H(z) ∈ CM×N so that the overall
matrix product is a circulant matrix of dimension M × M. Indeed, one can observe that each row of
matrix Hc can be obtained by circular-shifting the related previous row.
After inclusion and removal of the cyclic prefix, the resulting circulant matrix can be diago-
nalized by its right-multiplication by the IDFT and left-multiplication by the DFT matrices, where
these matrices are placed at the transmitter and receiver sides, respectively. Indeed, we have
H
H
WM Hc WM = WM h0 h1 · · · hM−1 WM
H
= WM h0 WM h1 · · · WM hM−1 WM
= h0 h1 · · · H
hM−1 WM , (2.49)
where hm is the mth column of matrix Hc and hm WM hm is its DFT. Note that one can in-
terpret the elements of vector hm as a periodic discrete-time signal hm (k) whose period is M,
which respects the relation hm (k) = h0 (k − m), where h0 (0) = h(0), h0 (1) = h(1), · · · , h0 (L) =
h(L), h0 (L + 1) = 0, · · · , h0 (M − 1) = 0, h0 (M) = h(0), and so forth.Thus, by remembering the
circular-shifting property of the DFT stating that, given
DFT{h0 (k − m)} = WM
ml
H0 (l). (2.51)
yielding
hm = diag{WMml
}l∈M h0
m
= diag{WMl
}l∈M h0
m
= D h0 , (2.53)
2.4. MEMORYLESS BLOCK-BASED SYSTEMS 47
in which D l }
diag{WM denotes an M × M
l∈M
l ,
diagonal matrix whose the (l, l)th element is WM
for each l ∈ M. We can therefore rewrite Equation (2.49) as
H
WM Hc WM = h0 D
h0 D2
h0 · · · DM−1 H
h0 WM
lm lm ⎡ ⎤
−m.0
WM
⎢
⎢ WM
⎥
−m.1
⎥ √
= l0
H0 (l)WM l1
H0 (l)WM ···
l(M−1)
H0 (l)WM ⎢ ⎥/ M
⎢ ..⎥
⎣ .⎦
−m(M−1)
WM
H0 (l) li −im
= √ WM WM
M i∈M
H0 (l) i(l−m)
= √ WM . (2.54)
M i∈M
i(l−m)
If l = m, then WM = 1 for all i ∈ M, implying
H0 (l) √
H
WM Hc WM = √ × M = MH0 (l), (2.55)
ll M
(l−m)
while, if l = m, then WM = 1, implying
H0 (l)
M(l−m)
WM −1
H
WM Hc WM = √ (l−m)
lm M
WM − 1
H0 (l) 1−1
= √ (l−m)
M WM −1
= 0. (2.56)
WM Hc WM H
= WM RCP H(z)ACP WM H
√
= diag Mh0
√
= diag MWM h0
√ h
= diag MWM
0(M−L−1)×1
diag{λm }m∈M , (2.57)
s + Ev
,
ŝ E (2.58)
with v
WM RCP v and, for the sake of simplicity, the time dependency of the expressions was
omitted. As can be observed, the estimates of the transmitted symbols are uncoupled, that is, each
symbol can be estimated independently of any other symbol within the related block, avoiding
intersymbol interference. One can interpret this fact as if each symbol were transmitted through a
flat-fading subchannel.
From a signal processing perspective, the model described in Equation (2.58) has a simple
interpretation. Indeed, the addition and removal of the cyclic prefix turns the linear convolution
described in Equation (2.39) into a circular convolution. In this case, the CP-OFDM system loads
each subcarrier in the frequency domain with a constellation symbol and, after that, performs the
inverse discrete Fourier transformation, generating a vector in the time domain. The elements of
this vector can be thought as a periodic signal which is processed by the channel through a linear
convolution. After that, the signal is brought back to the original frequency domain. A basic fact
of digital signal processing is that the circular convolution of two signals can be implemented in
the frequency domain by performing the product of the DFTs of the related signals. Therefore,
the CP-OFDM system can be further simplified if we take this fact into account. All we have to
do is to perform the entire processing in the frequency domain. The symbols which are loaded at
each subcarrier can be directly mapped to the received signals at each subcarrier by performing the
product with the frequency response of the channel (DFT of the zero-padded impulse response, as
in Equation (2.57)).
The equalizer E for this transceiver can be defined in several ways, where the most popular are
the zero-forcing (ZF) and the minimum mean square error (MMSE) equalizers. In the ZF solution,
it is aimed to undo the distortions introduced by the channel. Indeed, when there is no noise, the
ZF solution is able to perfectly recover the transmitted vector. It is assumed that matrix can be
inverted, thus yielding
EZF −1 . (2.59)
As for the MMSE solution, there is no requirement that matrix be invertible since this
latter operation is not needed. In fact, the linear MMSE equalizer matrix is the solution to the
following optimization problem:
EMMSE arg min J (E) , (2.60)
∀E∈CM×M
2.4. MEMORYLESS BLOCK-BASED SYSTEMS 49
where J is a real-valued function of a complex-valued matrix argument defined as
J (E) E s − E( s + v
)22
H
= E s − E s − Ev
s − Ev
s − E
H
= tr E s − E s − Ev
s − E s − Ev
= tr σs2 IM + σs2 E H EH − σs2 E − σs2 H EH + σv2 EEH , (2.61)
where E[·] and tr{·} are the expected value and trace operators, respectively. The derivation above
assumes that the transmitted symbols and environment noise within a block are independent and
identically distributed (i.i.d.), originating from a wide-sense stationary (WSS) white random se-
quences with zero means and uncorrelated.These assumptions imply that E[sv
H ] = E[s]E[v
]H =
0M×M = E[v
]E[s]H = E[v
sH ] and that E[ssH ] = σs2 IM and E[v
v
H ] = σv2 IM , where the pos-
itive real numbers σs2 and σv2 are the variances of the related WSS random sequences.6
Now, by using the following derivatives of scalar functions of complex matrices [83]:
! "
∂tr ZAZH
= ZA, (2.62)
! ∗ H"
∂Z
∂tr AZ
= A, (2.63)
∂Z∗
then we have
∂ J (E)
= σs2 E
H − σs2 H + σv2 E. (2.64)
∂E∗
∂ J (EMMSE )
We know that the optimal solution EMMSE is such that ∂E∗ = 0M×M , which implies
that7
# $−1
σv2
EMMSE = H
H
+ 2 IM
⎧ σs ⎫
⎨ λ∗m ⎬
= diag . (2.65)
⎩ |λ |2 + σv2 ⎭
m σs2 m∈M
It is worth highlighting that the CP-OFDM transceiver is the most popular type of OFDM-
based techniques which are employed in practical applications.
2.4.2 ZP-OFDM
An alternative OFDM system inserts zeros as redundancy and is called zero-padding OFDM (ZP-
OFDM). There are many variants of ZP-OFDM. One possible choice is the ZP-OFDM-OLA
6 In this book, we shall not employ distinct notations for deterministic and random variables.
7 We encourage the reader to justify why this is actually the minimum solution of the objective function.
50 2. TRANSMULTIPLEXERS
(overlap-and-add) whose transmitter and receiver matrices are implemented as
IM
F WM H
, (2.66)
0K×M
AZP ∈CN×M
IK
G EWM IM , (2.67)
0(M−K)×K
RZP ∈CM×N
As can be observed, matrix AZP removes the interblock interference, whereas matrix RZP left-
multiplies the resulting memoryless Toeplitz matrix 8 H(z)AZP ∈ CN ×M so that the overall product
becomes a circulant matrix of dimension M × M. The reader should note that RZP H(z)AZP =
RCP H(z)ACP = Hc .
The ZP-OFDM-OLA transceiver discussed here is a simplified version of a more general
transceiver proposed in [55].9 In fact, the general transceiver allows the recovery of the transmitted
symbols using zero-forcing equalizers independently of the locations of the channel zeros, unlike
the ZP-OFDM-OLA or CP-OFDM that might have zero eigenvalues under certain channel con-
ditions. Unfortunately, from the computational point of view, this transceiver implementation is not
as simple as, for instance, the CP-OFDM, since the equivalent channel matrix is not circulant, turn-
ing impossible its diagonalization through fast transforms, such as fast Fourier transform (FFT).10
Furthermore, even for the design of a simple ZF equalizer, the general ZP-OFDM transceiver
8 See Subsection 4.3.2 for a formal definition of a Toeplitz matrix.
9There are other variants of ZP-OFDM, such as the ZP-OFDM-FAST.
10 Actually, it is possible to implement the general ZP-OFDM system using FFTs, but without diagonalizing the equivalent channel
matrix (see Chapter 4).
2.4. MEMORYLESS BLOCK-BASED SYSTEMS 51
would require the inverse of a Toeplitz matrix, being therefore more complex than the inversion of
a circulant matrix required by a ZP-OFDM-OLA system.
2.4.3 CP-SC-FD
The cyclic-prefix single-carrier with frequency-domain (CP-SC-FD) equalization transceiver em-
ploys cyclic prefix as redundancy and it is closely related to the CP-OFDM transceiver. The CP-
SC-FD system is described by the following transmitter and receiver matrices:
0K×(M−K) IK
F , (2.69)
IM
G WMH
EWM 0M×K IM . (2.70)
2.4.4 ZP-SC-FD
The zero-padding single-carrier with frequency-domain (ZP-SC-FD) equalization transceiver in-
serts redundant zeros to the block to be transmitted, as in the ZP-OFDM transceiver. The ZP-SC-
FD-OLA version may be modeled through the following transmitter and receiver matrices:
IM
F , (2.71)
0K×M
IK
G WM
H
EWM IM . (2.72)
0(M−K)×K
Observe that the requirement of having 2K − L ≥ 0 makes sense when we analyze the
above expression. Indeed, in order to recover the M transmitted symbols, the memoryless transfer
matrix T of dimension M × M must be full-rank. This means that min{M, M + 2K − L} ≥ M,
i.e., 2K − L ≥ 0 ⇔ K ≥ L2 .
For this transceiver there are some constraints to be imposed upon the channel impulse
response model so that a zero-forcing solution exists. These constraints are related to the con-
cept of congruous zeros.11 The congruous zeros of a transfer function H (z) are the distinct zeros
z0 , z1 , · · · , zμ−1 ∈ C, with μ ∈ N, which meet the following condition: ziN = zjN , with H (zi ) =
H (zj ) = 0, for all i, j ∈ {0, 1, · · · , μ − 1}. Note that μ is a function of N. As shown in [44], the
channel model must satisfy the constraint μ(N ) ≤ K, where μ(N ) denotes the cardinality (number
of elements) of the largest set of congruous zeros with respect to N .
Therefore, assuming the existence of minimum-redundancy solutions for a given channel, i.e.,
considering that μ(N) ≤ L/2 ∈ N, then the ZF solution is such that its associated receiver matrix
is given by
−1 −1
G (H F)−1 = F H , (2.78)
CHAPTER 3
OFDM
3.1 INTRODUCTION
As discussed in the previous chapters, the orthogonal frequency-division multiplexing (OFDM) is
a transmission technique that is currently used in a number of wired and wireless systems. This
chapter describes OFDM in more detail, starting from its original conception in the continuous-time
domain (herein called, analog OFDM ) and arriving at its current implementation in the discrete-time
domain. In fact, the discrete-time description of OFDM has already been addressed in Section 2.4
of Chapter 2. However, that description is solely based on the useful mathematical properties related
to circulant matrices, without necessarily calling for physical intuition of actual transmissions. The
focus of the present chapter, on the other hand, is to motivate the construction of the OFDM system
by analyzing its very insightful analog version and to derive the discrete-time implementation from
this physically meaningful continuous-time system. Indeed, it was only with the widespread use of
digital integrated circuit technology that the discrete-time OFDM transmission technique became
popular, especially due to the existence of fast Fourier transform (FFT) algorithms, which enable
efficient computations of the discrete Fourier transform (DFT) employed for modulation.
From a historical perspective, the origins of frequency-division multiplexing (FDM) date back
to the late nineteen century, according to the review article by S. B. Weinstein [88]. The analog
version of OFDM was first proposed by R. W. Chang in 1966 [10], who filed a patent that was
granted in 1970 [11]. A major breakthrough was the perception that the use of analog subcarrier
oscillators and their corresponding coherent demodulators could be avoided by replacing them by
DFT-based transceivers. In this context, S. B. Weinstein and P. M. Ebert [89] were the originators
of the DFT-based modulation and demodulation schemes. Another key result related to the digital
OFDM implementation was conceived by A. Peled and A. Ruiz [64] who advanced the use of cyclic
prefix as solution for maintaining orthogonality among subcarriers at the receiver side. Although the
analog and digital versions of OFDM systems are closely related, they are not always fully equivalent
as discussed in [43]. OFDM has become widely adopted in commercial applications, thus explaining
why there are so many works addressing its history [13, 88].
This chapter is organized as follows. Section 3.2 describes the origins of OFDM in its analog
version. Such topic is particularly interesting for understanding the choices of some important
parameters, such as the OFDM symbol duration, sampling period, and guard period. In addition,
Section 3.2 also introduces the importance of orthogonality in OFDM. Section 3.3 describes the
discrete-time implementation of OFDM systems. The idea of Section 3.3 is to connect what we
have seen in Section 2.4 of Chapter 2 with the theory of analog OFDM. Section 3.4 describes
56 3. OFDM
some variants of OFDM-based systems, including single-carrier with frequency-domain equalization
(SC-FD), zero-padding (ZP) schemes, coded OFDM (C-OFDM), and discrete multitone (DMT)
systems. Finally, some conclusions are drawn in Section 3.5.
where pm (t) is a continuous-time pulse signal. The choice of the pulse signal determines how the
elements of the discrete-time signal are distributed over the time-frequency plane.
For example, by choosing pm (t) in a TDM fashion, so that pm (t) p(t − mT ) whose time
support is the real interval [mT , (m + 1)T ), we generate the following continuous-time signal (see
3.2. ANALOG OFDM 57
Equation (3.1) as well):
sTDM (t) sm p(t − mT ). (3.3)
m∈M
The former equation implies that sTDM (t) is a concatenation of pulses p(t), each of them starting at
time t = mT with duration of T seconds, modulated by their corresponding symbol sm originating
from a digital modulator. From Fourier analysis we know that the Fourier transform (FT) of sTDM (t)
is
STDM (ω) = sm P (ω)e−jωmT , (3.4)
m∈M
where the FT of p(t) is represented by P (ω), whose bandwidth is
. Since in TDM schemes
each symbol is transmitted in a time slot with T seconds of duration, then the transmission of M
symbols lasts MT seconds. In frequency domain each of these symbols occupies the entire available
bandwidth
.
On the other hand, in FDM schemes we utilize a dual strategy for signal transmission. Indeed,
in FDM each symbol occupies a portion of the whole channel bandwidth
. The frequency response
of the transmitted signal is
SFDM (ω) sm P
(ω − m
), (3.5)
m∈M
m ∈ M {0, 1, . . . , M − 1} (3.7)
is the subcarrier index, and M is a positive integer number representing the number of subcarriers,
then the frequency separation between two consecutive subcarriers is
1
fm+1 − fm , (3.8)
for all m ∈ M \ {M − 1}. Note that, by assuming that a subcarrier is comprised of a single tone, let
us say fm , the time-domain representation of such subcarrier consists of a complex exponential at
that frequency,1 that is, ej2πfm t .
The transmission of a block with M symbols belonging to a given constellation C ⊂ C, in
which each symbol is denoted as sm (n) ∈ C , is performed by transmitting these symbols using
subcarriers with distinct central frequencies. In this context, n is an integer number that identifies the
block with M constellation symbols. Such association between symbols and subcarriers is exemplified
below as
s0 (n) → f0 0,
1
s1 (n) → f1 = ,
..
.
M −1
sM−1 (n) → fM−1 = . (3.9)
Hence, the mth symbol is associated with the subcarrier whose central frequency is fm = m
,
for each m within M. The nth data block to be transmitted, usually called OFDM symbol, is 2
a complex signal denoted by ûn (t), in which t is a real variable representing time. The OFDM
symbol ûn (t) is generated as the superposition of the subcarriers, each of them modulated by its
1 Indeed, a single-tone signal whose tone is f
m is a signal whose frequency-domain representation consists of an impulse centered
at frequency fm . From Fourier analysis, we know that the inverse Fourier transform of an impulse at frequency fm corresponds
to ej2πfm t .
2The reader should not confuse the terms symbol and OFDM symbol. While the former is a complex number generated at the
output of a digital modulator (see Section 1.2), the latter is associated with a collection of constellation symbols.
3.2. ANALOG OFDM 59
corresponding symbol sm (n), yielding
1
ûn (t) + sm (n)ej2πfm (t−nT̂ )
T̂ m∈M
1 2π
=+ sm (n)ej m(t−nT̂ ) , for nT̂ ≤ t < (n + 1)T̂ , (3.10)
T̂ m∈M
provided T̂ is a multiple of , that is, T̂ = κ, with κ being a positive integer. Indeed, the last
2π
equality follows easily by considering that e−j lt can be rewritten as
# $ # $
2π 2π 2π
e−j lt = cos − lt + j sin − lt , (3.12)
and remembering that both sine and cosine functions integrate to zero in intervals corresponding to
multiples of their fundamental period, which in this case is given by l . Therefore, we must choose
T̂ in Equation (3.11) in such a way that the cross-correlation is equal to zero for all l ∈ M \ {0}.
This implies that the choice of T̂ must be based on the slowest complex exponential,3 which occurs
when l = 1, which in turn shows that T̂ must be a multiple of 1 = .
Therefore, as pointed out in Equation (3.11), the orthogonality among subcarriers plays a key
role in the choice of T̂ and its relation with . Indeed, this orthogonality can be obtained by choosing
3The one that takes more time to complete a cycle. In addition, note that the fundamental period of the slowest complex exponential
is already a multiple of the periods of the other exponentials.
60 3. OFDM
an OFDM symbol duration T̂ = κ. Note that, as κ increases, the OFDM symbol duration also
increases, but the amount of transmitted data is exactly the same that would be transmitted if κ = 1,
that is, κ > 1 reduces the system throughput. That is why κ = 1 is the natural choice. In addition,
note that the orthogonality does not depend on the symbols that modulate the subcarriers.
This implies that the OFDM symbol ûn (t) can be redefined as
1 j 2π m(t−nT̂ )
ûn (t) + sm (n)e T̂ , for nT̂ ≤ t < (n + 1)T̂ ,
T̂ m∈M
1 j 2π mt
=+ sm (n)e T̂ e−j2πmn , for nT̂ ≤ t < (n + 1)T̂ ,
T̂ m∈M =1
j 2π mt
= sm (n) p̂(t − nT̂ )e T̂
m∈M
ϕ̂m (t−nT̂ )
= sm (n)ϕ̂m (t − nT̂ ), (3.13)
m∈M
where function ϕ̂m (t) represents the mth subcarrier and the pulse signal is
⎧
⎨ +1 , for 0 ≤ t < T̂ ,
p̂(t) (3.14)
⎩ T̂
0, otherwise.
Let us interpret Equation (3.13) pictorially. Each subcarrier ϕ̂m (t) is a complex exponential
multiplied by p̂(t), a rectangular window of duration T̂ . Figure 3.1(a) depicts a given pulse p̂(t),
whose Fourier transform is the well-known sinc function. The square of the sinc represents the
subcarrier spectrum, as depicted in Figure 3.1(b). Figure 3.1(c) illustrates many OFDM subcarriers,
placed at their correct positions,4 in order to show the distance between the central frequencies
of neighboring subcarriers and to emphasize that, at each subcarrier central frequency, all other
subcarriers have amplitude equal to zero, as illustrated by the dotted lines.
Figure 3.1(c) illustrates several subcarriers belonging to a single OFDM symbol as if they were
isolated (i.e., we have not added the curves associated with each subcarrier). However, as given in
Equation (3.13), an OFDM symbol is formed by the summation of all M subcarriers modulated by
their corresponding symbol. The result of such summation is represented in Figure 3.2. This figure
depicts a frequency-domain representation of an OFDM symbol comprising three subcarriers. Note
that the support of such a representation is the entire real axis.
At this point, it is worth mentioning that simple TDM- and FDM-based transmissions can
also yield orthogonal signals to the input of the communication channel, as illustrated in the dis-
cussions of Section 1.5 of Chapter 1 within the framework of multiple-access schemes. Indeed,
4 Remember that the Fourier transform of the product of two signals is equivalent to the convolution between the Fourier transforms
j 2π mt
of each individual signal. Thus, each subcarrier p̂(t)e T̂ can be represented in the frequency domain by the convolution of
a sinc function (Fourier transform of p̂(t)) with a Dirac impulse at frequency m (Fourier transform of the exponential), which
T̂
results in the sinc function centered at frequency m .
T̂
3.2. ANALOG OFDM 61
0 “ ” 12
Subcarrier p sin πf T̂
@ T̂ “ ” A
Spectrum πf T̂
Pulse Shape
1
p̂(t)
T̂
t
T̂ − T̂4 − T̂2 0 2
T̂
4
T̂
f
(c) Subcarriers.
Figure 3.1: Representation of OFDM subcarriers: (a) time-domain representation of p̂(t); (b) frequency-
domain representation of p̂(t); and (c) a set of non-interfering subcarriers.
Equations (3.3) and (3.5) are examples of (theoretically) orthogonal TDM and FDM. However, in
the case of FDM transmissions, one must necessarily let empty spectral regions for separating the
frequency content associated with each subcarrier, otherwise the filters employed in such separa-
tion would be for certain non-causal filters due to the required sharp transitions in the frequency
domain. The aforementioned analog OFDM avoids this waste of spectrum that generally occurs
in FDM-based systems by allowing spectrum superposition of the subcarriers. In standard TDM
transmissions, when the transmitted signal crosses a frequency-selective channel, the original time-
domain orthogonality is lost. A possible solution is adding guard intervals between the transmission
62 3. OFDM
Subcarrier A
Subcarrier B
Subcarrier C
A+B+C
··· ···
Figure 3.2: Example of frequency representation of an OFDM symbol comprised of three subcarriers.
of each constellation symbol, which also represents a waste of resources. Actually, TDM-based solu-
tions employ time-domain equalizers to decrease the interference among symbols due to the loss of
orthogonality. But, even in this case, if the interference level is too high, then the order of the time-
domain equalizer can turn its implementation impractical. Actual analog OFDM transmissions, on
the other hand, are able to circumvent the interferences introduced by frequency-selective channels
by using subcarriers that are orthogonal to each other at the receiver end, thus justifying the name
orthogonal FDM (OFDM). The key feature present in analog OFDM is the introduction of guard
intervals between each OFDM symbol, as explained in Subsection 3.2.3.
t
initial dashed version represents its time extension. This OFDM subcarrier extension is known as
cyclic prefix (CP). Assuming this carrier energizes a frequency-selective channel, whose time-delay
spread spans up to τmem ∈ R seconds, at the receiver end the first τmem seconds will be corrupted
by the previous OFDM symbol, thus generating the so-called interblock interference (IBI). After the
period τmem the IBI is over and if we keep the subcarriers illuminating the channel for T seconds,
then the subcarriers will be orthogonal to each other as long as the time period T − τmem is long
enough and as long as we are able to eliminate the interference among constellation symbols within
a given data block, i.e., the so-called intersymbol interference (ISI).
Figure 3.4 illustrates how OFDM symbols are concatenated taking into consideration a guard
interval5 of duration τ ∈ R seconds. The useful symbol time T̂ (see Figure 3.4) corresponds to the
original duration of the OFDM symbol whose subcarriers are orthogonal to each other only at
the transmitter side. Thus, the extended OFDM symbol has duration T T̂ + τ , where the guard
interval τ is longer than the longest multipath delay (τ > τmem ) and is used to avoid the harmful
interferences introduced by frequency-selective channels.
τ
T
guard useful
period symbol time
5 A guard interval or guard period is a more general concept which includes the cyclic prefix as a special case.
64 3. OFDM
1
Mathematically, if we define M subcarriers separated in frequency by T −τ Hz as
2π
ϕm (t) p(t)ej T −τ mt , (3.15)
for m ∈ M, where the subcarrier index m is within the set M and the pulse signal is6
⎧
⎨√ 1 , for − τ ≤ t < T − τ,
p(t) T −τ (3.16)
⎩
0, otherwise,
then the transmitted signal, which is the concatenation of extended OFDM symbols, can be written
as
u(t) un (t)
n∈Z
= sm (n)ϕm (t − nT ), (3.17)
n∈Z m∈M
=un (t)
where sm (n) is the mth symbol within the nth block representing an extended OFDM symbol.
An extended OFDM symbol obtained by the extension of each subcarrier-time support is
equivalent to a time-domain signal with a cyclic prefix of length τ , i.e., in which the first τ seconds of
the data block coincide with the last τ seconds. Indeed, consider the nth extended OFDM symbol,
whose time support is [nT − τ, nT + T − τ ). Let t be an arbitrary real number within the first
τ seconds of that interval, i.e., t ∈ [nT − τ, nT ). In addition, let t
be a real number defined as
t
t + (T − τ ), which denotes a time instant within the last τ seconds of the referred block. Thus,
we have
2π
un (t) = sm (n)p(t − nT )ej T −τ m(t−nT )
m∈M
1 2π
= sm (n) √ ej T −τ m[t −(T −τ )−nT ]
m∈M
T −τ
1 2π
= sm (n) √ ej T −τ m(t −nT ) · e−j2πm
T −τ
m∈M =1
=p(t
−nT )
= un (t
). (3.18)
Cyclic Prefix
τ
T
guard useful
period symbol time
being used and, as a consequence, the received signal corresponding to this ûn (t) after crossing a
multipath fading channel would not yield orthogonal subcarriers. On the other hand, the signal u(t)
in Equation (3.17) is the concatenation of infinitely many extended OFDM symbols using cyclic
prefix in order to ensure orthogonality among subcarriers at the receiver side.
Indeed, the orthogonality among subcarriers is maintained since the interference between
OFDM symbols (i.e., the IBI) can be eliminated by discarding the first τ seconds out of T seconds
of each received data block. As for the remaining interference (ISI) due to constellation-symbol
superpositions within the resulting block of duration T − τ , we can eliminate it using the subcarrier
orthogonality. In order to verify mathematically these facts, let us analyze the received signal y(t)
assuming a noiseless baseband channel model whose time-delay spread is τmem , as follows:
y(t) H {u(t)}
, -
=H sm (n )ϕm (t − n T )
∈Z m∈M
n ! "
= sm (n
)H ϕm (t − n
T )
∈Z m∈M
n ! "
= sm (n
)H ϕm (t − n
T ) , (3.19)
m∈M n
∈Z
where H {·} represents the linear system that models the referred baseband channel. Now, by con-
sidering that the channel model
! remains" constant during the interval of an OFDM symbol, we can
compute the quantity H ϕm (t − n
T ) through a convolution integral of ϕm (t − n
T ) with the
66 3. OFDM
channel impulse response hn
(t) associated with the n
th symbol. Therefore, we have
∞
j T2π
2π
y(t) =
sm (n )e −τ m(t−n T ) hn
(τ
)p(t − n
T − τ
)e−j T −τ mτ dτ
m∈M n
∈Z −∞
τmem
2π
2π
= sm (n
)ej T −τ m(t−n T ) hn
(τ
)p(t − n
T − τ
)e−j T −τ mτ dτ
, (3.20)
m∈M n
∈Z 0
As we are studying both interblock and intersymbol interferences associated with the trans-
mission of blocks un (t), with n ∈ Z, of length T , it is convenient to separate the received signal into
blocks with the same length T . Thus, we can write
y(t) = y(nT + t
)
n∈Z
yn (t
)
= yn (t ), (3.22)
n∈Z
where the time instant t
is in the interval [−τ, T − τ ). This way, it follows from Equation (3.20)
that the nth received block can be expressed by
2π
yn (t
) = sm (n
)ej T −τ m[(n−n )T +t ] In,n
(t
), (3.23)
m∈M n
∈Z
with
τmem
2π
In,n
(t ) hn
(τ
)p (n − n
)T + t
− τ
e−j T −τ mτ dτ
. (3.24)
0
Now, if we consider only time instants t
within the interval [0, T − τ ), with τ > τmem , then
min{t
+ τ, τmem } = τmem , and, therefore, the second integral which appears at the right-hand side
of Equation (3.25) will be zero. In other words, when we discard the first τ > τmem seconds of
each received block yn (t
), then there is no interference between the nth and (n − 1)th transmitted
OFDM symbols, i.e., for any t
∈ [0, T − τ ), we have
τmem
δ[n
− n] 2π
In,n
(t ) = √ hn
(τ
)e−j T −τ mτ dτ
T −τ
0
2π m
Hn
T −τ
# $
δ[n
− n] 2π m
= √ H n
, (3.27)
T −τ T −τ
sm (n)Hn T2π−τ
m
complex numbers √ . Hence, if we compute the temporal cross-correlation over a useful
T −τ
symbol duration T̂ = T − τ for any of those two modulated subcarriers, we would reach a similar
result to Equation (3.11). This means that the orthogonality between subcarriers is achieved at the
receiver side. This is the main feature of analog OFDM, since the IBI is eliminated and, at the
same time, the ISI can be eliminated by using the resulting orthogonality among subcarriers at the
receiver.
1
Δf =
T −τ
t
T
„ «2
Subcarrier √ sin (πf T )
T
Spectrum (πf T )
Pulse Shape
1
p(t) √
T
t
T − T4 − T2 0 2
T
4
T
f
Figure 3.7: Representation of extended OFDM subcarriers: (a) time-domain representation of p(t);
and (b) frequency-domain representation of p(t).
will exist interferences from adjacent subcarriers at those central frequencies. This means that the
subcarriers are not orthogonal at the transmitter side. However, this is not an issue since the infor-
mation extraction occurs at the receiver. Thus, the orthogonality among subcarriers at the receiver
end allows a proper extraction of the symbols associated with different subcarriers within a received
extended OFDM symbol, even when the received signal has been severely distorted by the channel.
That is why we exchange the original orthogonality present in OFDM symbols by an orthogonality
of the received extended OFDM symbols with cyclic prefix (after discarding the first τ seconds).
Mb
BR = (3.29)
T
in bits per second (bps). If we add the spectra of all subcarriers and consider that side lobes below
20 dB from the main lobe are negligible, corresponding to the second side lobes on each side of the
70 3. OFDM
main lobe, then the total bandwidth is9
M −1 3
BW = +2 . (3.30)
T −τ T
The ratio
BR Mb
= (M−1)T
BW
T −τ +6
b
=
(3.31)
M−1
M
T
T −τ + 6
M
is the so-called spectral efficiency, which tends to b(1 − Tτ ) for large M. That means the OFDM is
an optimal modulation in terms of spectral efficiency as long as Tτ 1. However, when τ T (very
dispersive environment), then the spectral efficiency of OFDM transmissions is quite small. Other
multicarrier and single-carrier transmissions which address this drawback of spectral efficiency will
be described in Chapters 4 and 5.
Figure 3.8 depicts the instantaneous power of a given OFDM symbol un (t). A dotted line is
used to represent the average power of the OFDM symbol. In this figure, we can observe that there
exist some peaks in the power of un (t) that are much higher than the average power, i.e., they are
well above the dotted line. Indeed, it is well known that the peak-to-average power ratio (PAPR) of
OFDM transmissions is higher than the PAPR of single-carrier transmissions (see Subsection 3.4.1
for further details in the discrete-time domain). High PAPR is undesirable because it implies a wide
dynamic range of the signal to be transmitted, which in turn requires power amplifiers with linear
response over a wide range, increasing the cost of such devices. This is one of the main reasons why in
LTE the use of OFDMA in the uplink was avoided. Therefore, PAPR is an important impairment
related to OFDM transmissions.
In cases where the carrier frequency of the received signal does not match the carrier frequency
of the transmitted signal, we have the so-called carrier-frequency offset (CFO). Thus, CFO is the
offset (difference) between two numbers representing carrier frequencies, one at the transmitter and
the other at the receiver end. Ideally, CFO should be close to zero, but there are many practical cases
in which non-negligible CFO occurs. For example, when the transmitter and/or receiver are moving,
which usually happens in mobile communications, the Doppler effect acts as a source of CFO. Note
that, from our previous discussion about the importance of orthogonality in OFDM systems, it is
rather intuitive that CFO has the potential to severely degrade the quality of OFDM transmissions.
Most of the solutions to the CFO issue rely on blind estimation of the frequency offset, and are a bit
complex and/or applicable to very particular cases. Some of the low-complexity solutions to CFO
are presented in [46, 92].
9The bandwidth must be computed by considering that the central frequencies of the M subcarriers are separated by 1 Hz,
T −τ
and each subcarrier is a sinc whose second side lobes decay more than 20 dB at T3 Hz, as can be observed in Figure 3.7(b).
3.2. ANALOG OFDM 71
un(t)2
Moreover, digital transmissions usually employ two branches: an in-phase (I) and a quadrature-
phase (Q) branch. These branches are associated with the real and imaginary parts of the transmitted
signal, respectively. I/Q imbalance occurs when there is phase and/or amplitude mismatches between
I and Q branches. Such mismatches are usually due to the imperfections in the process of the
radio-frequency signal down-conversion to baseband signal and are, therefore, unavoidable in the
analog front-end [74]. In most cases, I/Q imbalance can only deteriorate the bit-error rate (BER)
performance of OFDM systems when they are employing high-order modulation schemes, such as
64-QAM (quadrature amplitude modulation). When I/Q imbalance is a major issue, one can use
digital signal processing techniques to compensate such mismatch. Indeed, there already exists several
techniques to compensate for the I/Q imbalance without increasing significantly the computational
burden. For more details, see [6, 78, 85] and references therein.
the output of the filter ψm (t) will be equivalent to the temporal cross-correlation between the
OFDM symbol and the mth receiver filter. Following the same steps performed in Equation (3.11)
and remembering that the fundamental period of the slowest subcarrier is = T − τ , it is easy to
verify that ψm (t) will remove all subcarriers except ϕm (t) due to orthogonality. In fact, the pair of
functions ϕm (t) and ψm (t) are biorthogonal [17]. In addition, note that the basis function at the
receiver ψm (t) has a time support shorter than the basis function at the transmitter ϕm (t).
Analog
s1 (n)δ(t − nT ) ϕ1 (t) ψ1 (t)
Channel
···
···
sM −1 (n)δ(t − nT ) ϕM −1 (t) ψM −1 (t)
Each
T
Even though analog OFDM can be derived from a very insightful view of the digital-to-
analog conversion implemented in an orthogonal FDM-based fashion, the resulting implementation
sketch depicted in Figure 3.9 also summarizes its main drawback: in general, practical solutions
entail the use of a large number of orthogonal subcarriers, thus hindering the applicability of this
structure in practice. Indeed, if M is large, then we would have to implement a large amount of
different oscillators/modulators, which may not be practical. This is one of the main reasons why
this analog version of OFDM was not employed in commercial applications after its proposal by
R. W. Chang [10, 11]. However, many of its properties and interpretations are still useful, as S. B.
Weinstein and P. M. Ebert [89], as well as A. Peled and A. Ruiz [64] noticed when they realized
that OFDM could be efficiently implemented in the discrete-time domain. This implementation
will be addressed in the next section.
3.3. DISCRETE-TIME OFDM 73
3.3 DISCRETE-TIME OFDM
Digital signal processing (DSP) has emerged as a powerful and efficient tool in a growing number
of applications. Indeed, there are many situations where the use of DSP-based techniques has either
greatly simplified the implementation of practical systems or simply enhanced their performance.
The discrete-time implementation of OFDM systems is an example of how DSP can even enable
the practical usage of a given technique, which could be quite hard to implement otherwise.
In this section we will apply sampling to the OFDM symbols and show a discrete-time
implementation of OFDM systems. As will be shown, the discrete Fourier transform (DFT) plays
a central role in this process, enabling efficient implementations of OFDM by means of fast Fourier
transform (FFT) algorithms.
10This is true only for detection purposes. If one is interested in studying the spectral roll-off related to the output of OFDM
systems or the effect of CFO (just to mention a few examples), then one should work directly with the discrete-time OFDM
model after the digital-to-analog conversion [43].
74 3. OFDM
Indeed, when we sample an OFDM symbol,11 the spectral repetitions due to the sampling
process are spaced apart by fs Hz, where fs T1s denotes the sampling frequency and Ts ∈ R
denotes the sampling period. First of all, fs must be larger than M−1 , otherwise we would have
T̂
at least one central frequency being shared by two distinct subcarriers, constituting a harmful type
of aliasing in this case. But if we consider any integer multiple of 1 larger than M − 1, then we
T̂
would not have any kind of interference at the subcarrier central frequencies. For simplicity reasons,
it is better to use the smallest sampling frequency that does not cause interference at the subcarrier
central frequencies, i.e., fs = M . Thus, we have
T̂
T̂
Ts . (3.34)
M
Without loss of generality, since the OFDM symbols are non-overlapping in time, we can
analyze each block of symbols separately. The nth OFDM symbol is non-zero only for time instants
within the interval [nT̂ , (n + 1)T̂ ).This means that, for all t
∈ [0, T̂ ), we have from Equation (3.13)
that
ûn (t
+ nT̂ ) = sm (n)ϕ̂m (t
+ nT̂ − nT̂ )
m∈ M
= sm (n)ϕ̂m (t
). (3.35)
m∈M
Now, let ûn,k be the discrete-time signal stemming from sampling the continuous-time sig-
nal ûn (t
+ nT̂ ) described in the above equation at each time instant t
= kTs , where, due to the
definition of Ts in Equation (3.34) and as t
∈ [0, T̂ ), we must have k within the set M. Hence, the
resulting discrete-time representation of an OFDM symbol is
where, based on Equation (3.13), we have that the discrete-time version of the mth subcarrier in an
OFDM symbol is given as
1 j 2π m( kT̂ )
ϕ̂m,k = + e T̂ M
T̂
1 j 2π mk
=+ eM . (3.37)
T̂
11 Here, sampling means to multiply the continuous-time function by a train of Dirac impulses.
3.3. DISCRETE-TIME OFDM 75
Thus, we can rewrite the discrete OFDM symbol in a more convenient manner as follows:
1 2π
ûn,k = + sm (n)ej M mk
T̂ m∈M
√
MIDFT{sm (n)}k
.
M
= IDFT {sm (n)}k , (3.38)
T̂
for each k ∈ M, where the inverse DFT (IDFT) of the discrete-time signal sm (n), with m ∈ M,
is also a sequence with length M. Hence, the following relation also holds:
.
T̂ ! "
sm (n) = DFT ûn,k m , (3.39)
M
in which the DFT of the sequence ûn,k is also a sequence whose mth element is defined as
! " 1 2π
DFT ûn,k m √ ûn,k e−j M mk , (3.40)
M k∈M
T̂ ∗
sm (n) = ûn,k ϕ̂m,k . (3.41)
M
k∈M
In summary, the discrete-time version of OFDM symbols without guard intervals is easily
computed through an IDFT of the transmitted discrete-time signal, which can be efficiently im-
plemented by using an FFT algorithm. Thus, assuming the channel introduces no distortion on the
transmitted signal, the sequence of symbols sm (n) could be recovered at the receiver end by taking
the DFT of the sequence ûn,k and scaling the result, as shown in Equation (3.39). This is a rather
obvious conclusion since the DFT and IDFT are inverse operations.
Note that the discrete-time OFDM symbols ûn,k in expression (3.36) can be thought as the
superposition of M subcarriers ϕ̂m,k modulated by the symbols sm (n). It is worth mentioning that the
original orthogonality present in the analog OFDM symbol is preserved in the discrete-time case.
Indeed, we can see that the following relation between any two subcarriers with indexes i, j ∈ M
76 3. OFDM
is valid:
1 2π
∗
ϕ̂j,k ϕ̂i,k = ej M k(j −i)
k∈M T̂
⎧ k∈#M 2π $
⎨ 1 ej M M(j −i) −1 , if i = j,
= T̂ e
j 2π
M (j −i) −1
⎩ M
, otherwise.
, T̂
0, if i = j,
= M
, otherwise.
T̂
M
= δ[i − j ]. (3.42)
T̂
Equation (3.42) is nothing but the orthogonality between subcarriers which are synchronized
and have the same duration T̂ .This expression stems from the projection of a transmitted signal onto
a subcarrier for detection purposes at the receiver end, as exemplified in Equation (3.41). For analog
OFDM, we have shown that the subcarrier orthogonality at the transmitter side is not sufficient
to allow the detection of the transmitted symbols in practical situations, i.e., when the data faces a
frequency-selective channel. This occurs since frequency-selective fading channels extend the time
support of the transmitted signals generating both IBI and ISI, as explained in Subsection 1.7. In
Subsection 3.2.3, we ensure subcarrier orthogonality at the receiver by inserting a cyclic prefix before
transmission. In the following subsection, we shall generate a discrete-time version of the results
related to extended analog OFDM symbols using cyclic prefix.
where Ts = M T̂
is the sampling rate associated with the discretization process. In addition, let K be
a positive integer number such that K ≥ L. Thus, by choosing the length of the cyclic prefix as
τ KTs , (3.44)
then we have that τ > τmem and, therefore, expression (3.28) holds for any t
∈ [0, T − τ ).
Now, by remembering from Subsection 3.2.3 that the useful symbol time T̂ is given by T − τ ,
then the sampling rate can be written as
T −τ
Ts = , (3.45)
M
3.3. DISCRETE-TIME OFDM 77
which means that [0, T − τ ) = [0, MTs ).
Let yn,k be the discrete-time signal originating from sampling the continuous-time signal
yn (t
) described in Equation (3.28) at each time instant t
= kTs , with k ∈ M. As explained before,
this continuous-time signal is the nth received block after removing the first τ = KTs seconds.
Thus, the resulting discrete-time representation of the nth received extended OFDM symbol after
removing the first K elements12 is
yn,k yn (kTs )
# $
1 2π m 2π T −τ
=√ Hn sm (n)ej T −τ mk M
T − τ m∈M T −τ
# $
1 2π m 2π
=√ Hn sm (n)ej M mk
T − τ m∈M T −τ
# $ 2π
2π m ej M mk
= Hn sm (n) √
T −τ T −τ
m∈M
λm (n) =ϕ̂m,k
= λm (n)sm (n)ϕ̂m,k . (3.46)
m∈M
Equation (3.46) means that the received signal, after removing the first K elements associated
with the guard interval introduced at the transmitter side, is composed of M modulated subcarriers
ϕ̂m,k . Even though the kth element of such signal is affected by all transmitted symbols sm (n), with
m ∈ M, one can use the subcarrier orthogonality expressed in Equation (3.42) in order to recover a
scaled version of the transmitted symbols without ISI. Therefore, by projecting the received signal
onto the mth subcarrier, one gets
T −τ ∗
λm (n)sm (n) = yn,k ϕ̂m,k . (3.47)
M
k∈M
The former projection process can be implemented in a much more efficient way. Indeed, based
on Equation (3.46) and following a similar approach which was employed in Subsection 3.3.1, we
can rewrite the received discrete OFDM symbol in a more convenient manner as follows:
1
M
yn,k = IDFT {λm (n)sm (n)}k , (3.48)
T −τ
for each k ∈ M. Therefore, the following relation also holds:
1
T −τ ! "
λm (n)sm (n) = DFT yn,k m . (3.49)
M
12 If we are ignoring the first τ = KT seconds of the continuous-time received block, then we are ignoring the first K elements of
s
the related discrete-time signal with corresponding sampling rate of T1 Hz.
s
78 3. OFDM
Hence, in order to recover sm (n), all we need is to multiply the mth element of the DFT of the
received OFDM symbol by λm1(n) , assuming λm (n) = 0, for all m ∈ M. This is the so-called zero-
forcing (ZF) equalizer. In fact, there are many other ways to perform equalization in order to estimate
sm (n). The choice of the equalizer depends on the types of distortion faced by the transmitted signal.
For instance, in our previous discussion we have neglected the existence of additive noise. In the
presence of such type of noise, an equalizer that minimizes the mean square error (MSE) would be
more appropriate than an equalizer that eliminates only the ISI, such as the ZF equalizer. In this
case, the equalization would consist of multiplying the mth element of the DFT of the received
OFDM symbol by
λ∗m (n)
, (3.50)
σv2 (n)
|λm (n)|2 + σs2 (n)
where σs2 (n) and σv2 (n) represent the variance of symbols and noise, respectively.13 This type of
equalizer is known as minimum MSE (MMSE) equalizer, as previously discussed in Subsection 2.4.1.
As explained in Subsection 3.2.3, the subcarrier orthogonality at the receiver end is obtained
by including a cyclic prefix of length τ at the transmitter end, generating the extended OFDM
symbols. Thus, by following the same steps employed in Subsection 3.3.1, but now considering the
nth extended OFDM symbol un (t) of Equation (3.17), we have
un (t
+ nT ) = sm (n)ϕm (t
+ nT − nT )
m∈ M
= sm (n)ϕm (t
), (3.51)
m∈M
for all t
∈ [−τ, T − τ ) = [−KTs , MTs ). Hence, for each k ∈ {−K, · · · , −1, 0, · · · , M − 1}, we
can define the discrete-time representation of an extended OFDM symbol as (see Equation (3.15))
un,k un (kTs + nT )
1 2π
=√ sm (n)ej M mk , (3.52)
T − τ m∈M
13 It is assumed here that all constellation symbols within the nth OFDM symbol have the same variance σ 2 (n). In addition, it was
s
assumed that all noise components have the same variance σv2 (n) as well. Those assumptions are not necessary, but they simplify
the notation. See Subsection 3.4.4 for the case where we do not consider those equal-power assumptions.
3.3. DISCRETE-TIME OFDM 79
whereas, for k ∈ {−K, · · · , −1}, one has
2π 2π 2π
sm (n)ej M mk = sm (n)ej M mk ej MmM
m∈M m∈M =1
2π
= sm (n)ej M (M+k)m
√M
m∈
= MIDFT {sm (n)}(M+k) , (3.54)
M−1
u(t) un,k pT (t − (k + nN )Ts ) , (3.56)
n∈Z k=−K
where
N M +K (3.57)
80 3. OFDM
denotes the amount of transmitted elements per extended OFDM symbol, Ts is the sampling period
of the DAC, and pT (t) is the transmitting pulse that may be chosen based on several distinct criteria,
such as best spectral roll-off.
After applying u(t) to a continuous-time baseband channel represented by the linear operator
H {·}, we have the following resulting signal:
r(t) H {u(t)}
M−1
= un,k H {pT (t − (k + nN )Ts )} , (3.58)
n∈Z k=−K
where H {pT (t − (k + nN )Ts )} is the output of the linear channel corresponding to nth transmitted
block, which can be computed as
The function hn (t) represents the impulse response of the analog baseband channel associated with
the nth transmitted block.14 Hence, one has
M−1
r(t) = un,k (hn ∗ pT ) (t − (k + nN )Ts ) . (3.60)
n∈Z k=−K
At the receiver end, an analog-to-digital conversion is also implemented by first filtering the
received waveform and then sampling it at a given rate (e.g., the same sampling rate employed at
the transmitter). By considering that the receiving filter impulse response is pR (t), then the signal
y(t) resulting from this first filtering stage at the receiver front-end is given by
14 It is assumed that the channel model does not vary with k during the transmission of the nth block.
3.3. DISCRETE-TIME OFDM 81
We can now define the discrete-time version of y(t) as
yn
,k
y(k
Ts + n
NTs )
M−1
= un,k hn k
− k Ts + (n
− n)N Ts
k=−K
n∈Z
hdn ((k
−k)+(n
−n)N )
M−1
= un,k hdn (k
− k) + (n
− n)N , (3.63)
n∈Z k=−K
is the discrete-time equivalent channel model associated with the nth transmitted block.
Now, in order to determine the relation between λm (n) and hdn (l), consider that hn (t) is the
impulse response of a multipath fading channel with delay spread τmem ≤ LTs . This way, it can be
approximated as15
hn (t) = hdn (l)δ(t − lTs ), (3.65)
l∈L
15 We use the word “approximated,” since the delay associated with the lth tap is not necessarily equal to lT . However, if T is small
s s
enough, then this expression is a reasonable approximation for an equivalent multipath baseband channel whose corresponding
transmitting and receiving filters were properly designed.
82 3. OFDM
−τ
Taking this fact into account, assuming that Ts = TM , and considering Equations (3.27)
and (3.46), we have
# $
2π
λm (n) = Hn m
T −τ
τmem
2π
= hn (τ
)e−j T −τ mτ dτ
0
τmem
2π
= hdn (l) δ(τ
− lTs )e−j T −τ mτ dτ
l∈L
0
d −j T2π
= hn (l)e −τ mlTs
l∈L
2π
= hdn (l)e−j M ml + 0
l∈L
2π 2π
= hdn (l)e−j M ml + 0 × e−j M mk
l∈L k∈M\L
√ 2π
= MHnd ej M m
√
MHnd (m) , (3.67)
in which Hnd (m) denotes √ the mth bin of the M-length DFT of a discrete-time equivalent channel
hdn (l). The scaling factor M appeared since we are working with the normalized version of the
DFT. In general, we assume that the number of subcarriers M employed in CP-OFDM systems
is larger than or equal to the length L + 1 of the channel model, i.e., we assume that M > L. This
means that we may have to insert some zeros at the end of the channel model hdn (l) before computing
the M-length DFT Hnd (m). Such insertion is also known as zero-padding.16
It is worth mentioning that the notation here is “heavier” than the notation in Chapter 2
since in this chapter we are working with time-varying continuous and discrete variables. So, we
shall keep in this chapter the dependency on the index of the OFDM symbol denoted by n and we
shall use the superscript “d” for the discrete-time model of the channel hdn (l), instead of the simpler
notation h(l).
Using the results presented in Subsections 3.3.1, 3.3.2, and 3.3.3, we can now clearly connect
the discrete-time implementation of CP-OFDM described in Chapter 2, Subsection 2.4.1 and the
CP-OFDM derived through a discretization of the analog OFDM.
⎡ ⎤
1 yn,0
T −τ ⎢ .. ⎥
= WM ⎣ . ⎦
M
yn,(M−1)
1
T −τ
= WM 0M×K IM y(n)
M
RCP
1
T −τ
= WM RCP y(n). (3.72)
M
where the equalizer matrix E(n) is an M × M diagonal matrix containing the scaling factors corre-
sponding to each one of the M subcarriers. Once again, it is clear the similarities between the former
expression and Equation (2.45) of Chapter 2. In addition, as pointed out in Subsection 3.3.2, one
can choose
1
[E(n)]mm = , (3.74)
λm (n)
λ∗m (n)
[E(n)]mm = , (3.75)
σv2 (n)
|λm (n)|2 + σs2 (n)
for the MMSE solution (considering the presence of an additive noise signal at the receiver front-
end).
As mentioned before, the OFDM discussed here is the so-called cyclic prefix OFDM (CP-
OFDM), whose block diagram is depicted in Figure 3.10. For the CP-OFDM the equalizer consists
of a scalar correction performed at each DFT output to compensate for the channel distortion
λm (n) at the mth frequency bin. The digital channel appearing in Figure 3.10 can also be modeled
within the block-based framework by simply rewriting Equation (3.63) as follows (compare with
3.3. DISCRETE-TIME OFDM 85
Transmitter
s0(n)
s1(n) I
s2(n) D P Digital
···
···
F S Cyclic Channel
T Prefix
sM −2(n)
sM −1(n)
Equalizer
ŝ0(n)
ŝ1(n)
S D ŝ2(n)
F
···
···
Remove
Prefix
P T
ŝM −2(n)
ŝM −1(n)
Receiver
⎡ ⎤
hdn (0) 0 0 ··· 0
⎢ hdn (1) h d (0) 0 ··· 0 ⎥
⎢ n ⎥
⎢ .. .. .. .. .. ⎥
⎢ . . . . . ⎥
⎢ ⎥
⎢ d . ⎥
y(n) = ⎢h (L) hd (L − 1) .. ··· 0 ⎥ u(n)
⎢ n n ⎥
⎢ 0 d
hn (L) ··· ··· 0 ⎥
⎢ ⎥
⎢ . . .. .. .. ⎥
⎣ .. .. . . . ⎦
0 0 hdn (L) · · · hdn (0)
HISI (n)
86 3. OFDM
⎡ ⎤
0 ··· 0 hdn−1 (L) · · · hdn−1 (1)
⎢ .. .. ⎥
⎢0 0 ··· 0 . . ⎥
⎢ ⎥
⎢ .. .. .. .. .. ⎥
⎢. . . . . hn−1 (L)⎥
d
⎢ ⎥
+ ⎢0 0 0 ··· 0 0 ⎥ u(n − 1)
⎢ ⎥
⎢0 0 0 ··· 0 0 ⎥ ⎥
⎢
⎢. .. .. .. .. .. ⎥
⎣ .. . . . . . ⎦
0 0 0 0 ··· 0
HIBI (n)
= HISI (n)u(n) + HIBI (n)u(n − 1), (3.76)
where we use the fact that hdn (l) is a causal system with order L, which implies that hdn (l) = 0
whenever l ∈ Z \ L.
As pointed out in Subsection 2.4.1 of Chapter 2, the effect of matrices ACP and RCP in the
former block-based description is to eliminate the IBI while turning the resulting effective channel
matrix into a circulant matrix. Due to the use of IDFT and DFT at the transmitter and receiver
sides, respectively, the overall block-based system reduces to a set of M uncoupled subchannels whose
mth received signal is given as (see also Equation (2.58) of Chapter 2)
in which we include the effect of additive noise modeled by vm (n). The parameter λm (n) is the so-
called channel tap associated with the mth subchannel of the nth transmitted block. This uncoupled
subchannel model will be very useful in the description of DMT in Subsection 3.4.4.
Example 3.1 (Block-Based Model) Design a CP-OFDM communication system with four
subcarriers and a fixed noiseless channel with transfer function
The first operation at the receiver is the removal of the redundancy through the multiplication
by RCP = [04×1 I4 ]. This operation is equivalent to discarding the first element, yn,(−1) , of the
received data block, thus removing the effect of the IBI and yielding the following description:
⎡ ⎤ ⎡ ⎤⎡ ⎤
yn,0 1 0 0 0.1 un,0
⎢yn,1 ⎥ ⎢0.1 1 0 0⎥ ⎢ ⎥
⎢ ⎥ =⎢ ⎥ ⎢un,1 ⎥
⎣yn,2 ⎦ ⎣ 0 0.1 1 0 ⎦ ⎣un,2 ⎦
yn,3 0 0 0.1 1 un,3
⎡ ⎤ ⎡ ⎤
1 0 0 0.1 s0 (n)
⎢0.1 1 0 0⎥ ⎢ ⎥
=⎢ ⎥ W H ⎢s1 (n)⎥
⎣ 0 0.1 1 0 ⎦ 4 ⎣ s2 (n)⎦
0 0 0.1 1 s3 (n)
⎡ ⎤⎡ ⎤⎡ ⎤
1 0 0 0.1 1 1 1 1 s0 (n)
1 ⎢0.1 1 0 0⎥ ⎢ ⎥⎢
⎥ ⎢1 j −1 −j ⎥ ⎢s1 (n)⎥ .
⎥
= ⎢⎣ ⎦ ⎣ ⎦ ⎣ (3.80)
2 0 0.1 1 0 1 −1 1 −1 s2 (n)⎦
0 0 0.1 1 1 −j −1 j s3 (n)
88 3. OFDM
Now, by taking the DFT of the resulting signal, one has
⎡ ⎤ ⎡ ⎤⎡ ⎤⎡ ⎤⎡ ⎤
yn,0 1 1 1 1 1 0 0 0.1 1 1 1 1 s0 (n)
⎢yn,1 ⎥ 1 ⎢1 −j −1 j ⎥ ⎢0.1 1 0⎥ ⎢ ⎥⎢ ⎥
W4 ⎢ ⎥ ⎢ ⎥⎢ 0 ⎥ ⎢1 j −1 −j ⎥ ⎢s1 (n)⎥
⎣yn,2 ⎦ = 4 ⎣1 −1 1 −1⎦ ⎣ 0 0.1 1 0 ⎦ ⎣ 1 −1 1 −1 ⎦ ⎣ s2 (n)⎦
yn,3 1 j −1 −j 0 0 0.1 1 1 −j −1 j s3 (n)
⎡ ⎤⎡ ⎤
1.1 0 0 0 s0 (n)
⎢ 0 1 − 0.1j 0 0 ⎥ ⎢s1 (n)⎥
=⎢
⎣0
⎥⎢ ⎥, (3.81)
0 0.9 0 ⎦ ⎣s2 (n)⎦
0 0 0 1 + 0.1j s3 (n)
leading to an uncoupled model with four subchannels. 2
3.4.1 SC-FD
As previously discussed in Subsection 3.2.4, the OFDM symbol presents high PAPR imposing
a high linearity constraint in the transmitter power amplifier. In the case where the power of the
transmitted signal has high dynamic range, it is possible that the nonlinear behavior of the transmitter
amplifier introduces distortions to the transmitted OFDM symbol, leading to loss of orthogonality
among the subcarriers. The deterministic PAPR of a discrete-time OFDM symbol can be defined as
! "
max |un,k |2
k
PAPR OFDM (n) 1 2 , (3.82)
M |un,k |2
k∈M
3.4. OTHER OFDM-BASED SYSTEMS 89
! "
in which max |un,k denotes the peak power level associated with the nth OFDM symbol, whereas
|2
1 2
k
M |un,k | denotes the average power of this nth block. In the case of CP-OFDM, the insertion
2
k∈M
of the guard interval does not change the maximum achievable power (peak) of the elements that
compose an OFDM symbol. In addition, it is very unlikely that the guard interval could change
the average power as well. Therefore, we can consider only the elements after the application of the
IDFT, without considering the cyclic prefix which is inserted. Thus, based on Equation (3.55), one
has
1 2π
un,k = √ sm (n)ej M mk , (3.83)
T − τ m∈M
yielding (see also Equation (3.42))
1 1
|un,k |2 = un,k u∗n,k
M M
k∈M k∈M
1 1
∗ j 2π
)k
= sm (n)sm
(n)e M (m−m
M T −τ
k∈M m∈M m ∈M
1 1 2π
∗
= sm (n)sm
(n) ej M (m−m )k
T −τ M
m∈M m
∈M k∈M
1
= sm (n) sm
(n)δ[m − m
]
∗
T −τ
m∈M m
∈M
1
= |sm (n)|2
T −τ
m∈M
M 1
= |sm (n)| ,2
(3.84)
T −τ M
m∈M
and
3 3
3 3
1 3 2π 3
|un,k | = √ 3 sm (n)ej M mk 3
3
T − τ m∈M 3
33 3
1 2π 3
≤√ 3sm (n)ej M mk 3
T − τ m∈M
1
=√ |sm (n)|
T − τ m∈M
1
≤√ max {|sm (n)|}
T − τ m∈M m
M
=√ max {|sm (n)|} , (3.85)
T −τ m
90 3. OFDM
thus implying that
M2
max |un,k |2 ≤ max |sm (n)|2 . (3.86)
k T −τ m
The results of Equations (3.84) and (3.86) give us that
⎛ ! "⎞
max |sm (n)| 2
⎜ m ⎟
PAPR OFDM (n) ≤ M ⎝ 1 2 ⎠
M |sm (n)| 2
m∈M
PAPR SC (n)
= MPAPR SC (n), (3.87)
s0(n)
s1(n)
s2(n) P Digital
···
S Cyclic Channel
Prefix
sM −2(n)
sM −1(n)
Equalizer
ŝ0(n)
ŝ1(n)
D I
S D ŝ2(n)
F
···
···
···
Remove F
Prefix
P T T ŝM −2(n)
ŝM −1(n)
Receiver
properly designed equalizer. Thus, in general, ZP-based transceivers do not yield so simple (one-
tap) equalizers as CP-based systems do. These observations are valid for the general ZP versions
of the OFDM and SC-FD, herein called ZP-OFDM and ZP-SC-FD, respectively. In fact, there
are some tools related to structured matrix representations which can work around these difficulties,
yielding general ZP-transceivers which are still based on DFTs and one-tap equalizers (see Figure 4.3
of Chapter 4). However, there are particular versions of ZP-OFDM and ZP-SC-FD that have the
same simple equalizers as the ones used in CP-OFDM and CP-SC-FD systems. These particular
versions, known as ZP-OFDM-OLA and ZP-SC-FD-OLA, perform overlap-and-add (OLA)
operations at the receiver side and they are depicted in Figures 3.12 and 3.13. The mathematical
details regarding these particular versions of ZP-OFDM and ZP-SC-FD were already explained in
Subsections 2.4.2 and 2.4.3, respectively.
The following topics are research results concerning the differences between the general ZP-
OFDM and ZP-SC-FD and their cyclic-prefix counterparts [55, 87]:
s0(n)
s1(n) I
s2(n) D P Digital
···
···
F S Zero Channel
T Padding
sM −2(n)
sM −1(n)
Equalizer
ŝ0(n)
ŝ1(n)
Post S D ŝ2(n)
··· F
···
Processing
P T
ŝM −2(n)
ŝM −1(n)
Receiver
• ZP-OFDM has better performance in terms of BER or MSE than CP-OFDM, for a given
average-bit-energy-to-noise power ratio, Eb /N0 .
• The ZP-SC-FD has lower PAPR, presents robustness to CFO, and has also better uncoded
performance. However, the equalization is a bit more complex to implement.
• In the case some kind of channel coding is included (C-OFDM), the coded version is better
when code rate is low and the error correcting coding capability is enhanced. In the coded case
ZP-SC-FD is better than C-OFDM for high code rate.
• CP-OFDM with equal-gain power allocation has the same performance as zero-forcing equal-
ized CP-SC-FD transceiver.
3.4. OTHER OFDM-BASED SYSTEMS 93
Transmitter
s0(n)
s1(n)
s2(n) P Digital
···
S Zero Channel
Padding
sM −2(n)
sM −1(n)
Equalizer
ŝ0(n)
ŝ1(n)
I
Post S D
D ŝ2(n)
F
···
···
···
Processing
P F
T T ŝM −2 (n)
ŝM −1 (n)
Receiver
• CP-OFDM with AMBER (an approximately minimum BER) power allocation is better than
zero-forcing equalized CP-SC-FD.
Channel
OFDM Receiver
them in subsets with greater Euclidean distances, while maintaining the transmission energy. The
subsets and the points in the subsets are tied using a trellis diagram derived with the convolutional
code. If a Viterbi decoder [38] is used at the receiver some coding gain is expected [13].
Error bursts could occur if the inner decoder (decoder 2) chooses a wrong decoding path
which could then be corrected by the combination of the interleaver and the outer decoder (decoder
1).
In the final analysis the reason for coding is to provide a link among the symbols transmitted on
different subcarriers such that a symbol transmitted in a strongly faded subcarrier (i.e., λm (n) ≈ 0)
can be recovered at the receiver by estimating the symbols transmitted by other subcarriers. The
bottom line is that we are distributing and mixing the transmitted information to increase the
chance of proper detection at the receiver end.
3.4.4 DMT
A discrete multitone (DMT) transceiver is essentially an OFDM system comprised of three particular
features: (i) there is no passband conversion to a higher carrier frequency, which means that the actual
transmitted signals are baseband signals; (ii) since any actual transmission employs real-valued signals,
then the baseband transmitted signals must be real-valued. This means that the input constellation
∗
symbols sm (n) must have the conjugate symmetric property, i.e., sm (n) = s(M−m) (n), for all m ∈ M;
and (iii) there is some kind of channel-state information (CSI) at the transmitter side, so that the
transceiver can use some smart techniques in order to cope with possible channel impairments in
advance.
The third DMT property above is indeed its key feature since it enables transmissions with
higher data rates. Nevertheless, it is usually applicable in wired connections, in which the channel
state does not change too often. As described in Subsection 1.3.1 of Chapter 1, the DMT system is
currently employed in many digital subscriber line (xDSL) applications. The aim of this subsection
is to describe how DMT-based systems take into account the availability of information about the
channel at the transmitter side in order to enhance the overall transmission performance.
As described in Subsection 3.3.4, OFDM systems can be thought as M parallel uncou-
pled subchannels, whose mth received signal of the nth transmitted OFDM symbol is given by
3.4. OTHER OFDM-BASED SYSTEMS 95
λm (n)sm (n) + vm (n), in which we consider that sm (n) is a random sequence with zero-mean and
2 (n), whereas v (n) is a random sequence with zero-mean and variance σ 2 (n), where
variance σs,m m v,m
m ∈ M and n ∈ Z. In this case, we have the following signal-to-noise ratio (SNR) associated with
the mth subchannel:
|λm (n)|2 σs,m
2 (n)
SNR m (n) 2 (n)
. (3.88)
σv,m
If vm (n) is a Gaussian random sequence, then it follows from basic knowledge on digital
communications that, for each fixed pair of numbers (m, n) ∈ M × Z, one has the following error
probability of symbols [22, 26, 45, 66, 83]:
+
which means that SNR(n) is basically the geometric mean of the sum of the SNR gap with the mth
subchannel SNR, subtracting the SNR gap from the final result.
Thus, the parallel and independent subchannels together behave as a single white Gaussian
noise channel with SNR corresponding to SNR(n), which is essentially equal to the geometric mean
of the subcarrier SNRs increased by the SNR gap. This geometric mean is a good approximation
for moderate to high SNRs on all subcarriers. As the geometric mean is much more sensitive to
small values than the arithmetic mean,18 then the existence of poor subchannels (i.e., subchannels
with low SNRs) can significantly degrade the system performance in terms of error probability of
symbols. An alternative view is that, for a given error-probability performance, the presence of poor
subcarriers can significantly decrease the transmission data rates.
18 In fact, the geometric mean of positive real numbers is always smaller than or equal to the arithmetic mean of these numbers.
3.4. OTHER OFDM-BASED SYSTEMS 97
The former reasoning clarifies how the quality of the subchannels affects the overall system
performance, since the SNR of the mth subchannel is affected directly by the channel gain λm (n)
(see Equation (3.88)). Standard uncoded OFDM-based systems suffer critically from performance
degradations caused by deep fadings in some subchannels. In order to appreciate the advantages
of the DMT over standard OFDM-based systems, the SNR(n) must be compared with unbiased
SNR of equalized baseband systems. In practice, SNR(n) can be improved considerably when the
available energy is distributed non-uniformly over all, or a subset of parallel carriers, giving rise
to high-performance multicarrier systems. This energy allocation is implemented by the so-called
loading algorithms, whose most notorious example is the water-filling approach. Let us formulate
this approach from now on.
From information theory, it is well known that the ultimate transmission data rate is dictated
by the channel capacity [14]. Indeed, the channel capacity is the upper bound on the possible data
rates which any communication system is able to achieve, assuring that error probability of symbols
is zero. It is possible to show that the channel capacity (in bits/transmission, i.e., channel capacity
normalized by the channel bandwidth) of the mth Gaussian subchannel of a CP-OFDM system is
given by (in the case of complex-valued variables) [14]
implying that the average channel capacity, C(n), considering all M subchannels of the nth data
block, is
1
C(n) = log2 (1 + SNR m (n)) . (3.99)
M
m∈M
This expression is quite similar to the average amount of transmitted bits expressed in Equa-
tion (3.96).
Therefore, in order to increase the transmission data rates one could try to compensate for
the channel taps λm (n) leading to low the subchannel SNRs, and an intuitive solution to this
problem is maximizing the average channel capacity in expression (3.99) subject to a predefined
power constraint. It is worth mentioning at this point that, although a transmission scheme that
achieves the related capacity could be theoretically implemented by using codes with a long block
length [14], practical communications systems avoid high complexity codes, thus implying that the
capacity may not be achievable. In other words, for a given small average error probability of symbols
Pe (n), the channel code might require infinite complexity to achieve the capacity. In practice, at a
given fixed error probability of symbols, one can characterize the modulation/coding by a gap ∈ R+
(for example, = QAM in Equation (3.95)) that quantifies the effective loss in SNR with respect
to capacity. Extremely powerful codes (such as turbo code and concatenated code) can reduce the
gap to up 1 or 2 dB.
98 3. OFDM
Let us formulate the problem of maximizing the average channel capacity (or minimizing
−C(n) expressed using the natural logarithm) as the following optimization problem:
, -
1 |λm (n)|2 σs,m
2 (n)
min − ln 1 + 2 (n)
, (3.100)
σs,m (n)∈R M σv,m
m∈M
m∈M
subject to:
2
σs,m (n) = p0 (n), (3.101)
m∈M
in which the positive real number p0 (n) denotes the total transmission power of the nth block. In
2 (n) and λ (n) are fixed and known numbers, for
this optimization problem, it is assumed that σv,m m
all m ∈ M.
By applying the Lagrange-multiplier method (with Lagrange multiplier λ ∈ R), we have the
following cost-function:
1 |λm (n)|2 σs,m
2 (n)
J σs,0 (n), · · · , σs,(M−1) (n) − ln 1 + 2 (n)
M σv,m
m∈ M
+λ σs,m (n) − p0 (n) ,
2
(3.102)
m∈M
Note that, based on Equation (3.103), when σs,mo (n) = 0, one has λ = 0, which means that 1 is
λ
well defined.19
o (n), with m ∈ M, are
Now, we can consider, without loss of generality, that the numbers σs,m
ordered as follows:
o o o
σs,0 (n) ≥ σs,1 (n) · · · ≥ σs,(M−1) (n). (3.106)
19 We shall omit the proof that those optimal values correspond to solutions that minimize the original objective function.
3.4. OTHER OFDM-BASED SYSTEMS 99
In addition, let us assume that we will transmit using M
subchannels, where M
− 1 ∈ M is the
largest index of the smallest value σs,m (n), with m ∈ M. Note that M ≤ M and that
o
o o
σs,0 (n) ≥ · · · ≥ σs,(M
−1) (n) > 0. (3.107)
Thus, by defining M
{0, · · · , M
− 1}, the constraint in Equation (3.101) can be used to find
the Lagrange multiplier, as follows:
2 1
2 (n)
σv,m
o
σs,m
(n) = −
Mλ |λm
(n)|2
m ∈M m ∈M
M
1 σv,m
2 (n)
= −
M λ |λm
(n)|2
m
∈M
= p0 (n), (3.108)
yielding
1 1 σv,m
2 (n)
=
p0 (n) + . (3.109)
Mλ M |λm
(n)|2
m
∈M
σ 2 (n) o 2
Note that large values of |λv,m(n)|2 leads to low values of σs,m (n) , which means that less power
m
is allocated to poor subchannels. The solution in expression (3.111) is the so-called water-filling. It
essentially says that the power allocation among the subcarriers is implemented in such a way that
o 2 σ 2 (n)
σs,m (n) + |λv,m(n)|2 is a constant and that occasionally it will not be distributed any energy to some
m
of the last subcarriers due to the power constraint. The name water-filling is motivated by the way
it can be interpreted, namely, as communicating vessels which are filled with a homogeneous liquid
that achieves the same level in all containers, regardless of their volume and/or shape.
100 3. OFDM
Power
1
4λ
o 2
σs,1 (n)
o
2
σs,0 (n)
2 2
σv,2 (n) σv,3 (n)
|λ2 (n)|2 |λ3 (n)|2
2
σv,1 (n)
|λ1 (n)|2 2
σv,0 (n)
|λ0 (n)|2
1 2 0 3 Channel
Figure 3.15 illustrates pictorially a water-filling solution for the case of four subchannels (i.e.,
2
2
M = 4) where the total energy distributed is p0 (n) = σs,0 o (n) + σs,1o (n) . The subchannels
with indexes 3 and 4 did not receive any power (i.e., M
= 2) since the values of their corresponding
2 (n)
σv,0 σ 2 (n)
2 (n)
σv,2 2 (n)
σv,3 p0 (n)+ 2 + v,1 2
|λ0 (n)| |λ1 (n)|
factors and
|λ2 (n)|2
were both higher than 4λ
|λ3 (n)|2
1
= 2 . Example 3.2 is a toy
example which help us verify the gains introduced by the water-filling approach.
Example 3.2 (Water-Filling) Given a time-invariant channel model with transfer function
for all m ∈ {0, 1, · · · , 7} and n ∈ Z. Considering eight subchannels, compute the channel capacity
for each subchannel, Cm (n), and average channel capacity, C(n), before and after applying the water-
filling approach. Assume two possible values of total transmission power, namely p0 (n) = 0.04 and
p0 (n) = 0.40. In addition, determine the power distribution for each case.
Solution. In order to compute the initial SNR (i.e., the SNR before applying the water-filling
2 (n),
optimization approach), one has to determine three quantities for each subchannel, namely σv,m
3.4. OTHER OFDM-BASED SYSTEMS 101
2 (n), and |λ (n)|2 , as described in Equation (3.88). The noise variance is fixed at 0.08, while the
σs,m m
average power of the symbols is σs,m2 (n) = p (n)/8, i.e., σ 2 (n) = 0.04/8 = 0.005 or σ 2 (n) =
0 s,m s,m
0.40/8 = 0.05. As for |λm (n)| , one has (see Equation (3.67) or Equation (2.57) of Chapter 2)
2
3 d 332
3√
3 hn 3
|λm (n)|2 = 3 M WM 3 , (3.113)
3 05×1 m 3
0.12 0.12
0.1 0.1
0.08 0.08
Power
Power
0.06 0.06
0.04 0.04
0.02 0.02
0 0
6 2 0 3 7 4 1 5 6 2 0 3 7 4 1 5
Subchannel Index Subchannel Index
o 2
The power distribution, the initial and optimized SNRs (the optimized SNR uses σs,m (n)
given in expression (3.111)), the initial and optimized capacity (see Equation (3.98)) for each subcar-
rier, and the average capacity (see Equation (3.99)) are summarized in the following table, considering
both transmission powers:
102 3. OFDM
Subchannel index 6 2 0 3 7 4 1 5
|λm (n)|2 0.0 2.0 4.0 2.0 0.0 2.0 4.0 2.0
2 (n)/|λ (n)|2
σv,m ∞ 0.04 0.02 0.04 ∞ 0.04 0.02 0.04
m
p0 (n) 0.04
2 (n)
Initial σs,m 0.005 0.005 0.005 0.005 0.005 0.005 0.005 0.005
Initial SNRm (n) 0.00 0.12 0.25 0.12 0.00 0.12 0.25 0.12
Initial Cm (n) 0.00 0.17 0.32 0.17 0.00 0.17 0.32 0.17
Initial C(n) 0.1654
Optimized σs,m2 (n) 0.00 0.00 0.02 0.00 0.00 0.00 0.02 0.00
Optimized SNRm (n) 0.00 0.00 1.00 0.00 0.00 0.00 1.00 0.00
Optimized Cm (n) 0.00 0.00 1.00 0.00 0.00 0.00 1.00 0.00
Optimized C(n) 0.2500
p0 (n) 0.40
2 (n)
Initial σs,m 0.05 0.05 0.05 0.05 0.05 0.05 0.05 0.05
Initial SNRm (n) 0.00 1.25 2.50 1.25 0.00 1.25 2.50 1.25
Initial Cm (n) 0.00 1.17 1.81 1.17 0.00 1.17 1.81 1.17
Initial C(n) 1.0368
Optimized σs,m2 (n) 0.00 0.06 0.08 0.06 0.00 0.06 0.08 0.06
Optimized SNRm (n) 0.00 1.50 4.00 1.50 0.00 1.50 4.00 1.50
Optimized Cm (n) 0.00 1.32 2.32 1.32 0.00 1.32 2.32 1.32
Optimized C(n) 1.2414
CHAPTER 4
2The bottom line here is that many distinct and interesting ways of designing multicarrier systems are available. We will focus on a
particular type of solution (reduced-redundancy system) that allows us to present to the reader a set of tools related to structured
matrix representations, which can be eventually employed in many other contexts.
4.2. REDUCED-REDUNDANCY SYSTEMS: THE ZP-ZJ MODEL REVISITED 107
4.2 REDUCED-REDUNDANCY SYSTEMS: THE ZP-ZJ
MODEL REVISITED
As briefly described in Chapter 2, Subsection 2.4.5, Lin and Phoong proposed a family of memoryless
discrete multitone transceivers which employ a reduced amount of redundancy to eliminate both
IBI and ISI [44]. A useful particular type of reduced-redundancy system is the so-called zero-padded
zero-jammed (ZP-ZJ) system. Let us revisit now the ZP-ZJ model in a more detailed manner.
Consider a complex-valued finite impulse-response (FIR) channel model h(l), where the
channel-tap index l can assume any value within the set L {0, 1, · · · , L}. The integer number L
is the so-called channel order, which in turn is associated with the delay spread of the channel [80]: a
measure of how much a signal is spread over the time when passing through the channel.
In ZP-ZJ systems, the transmitter is responsible for linearly processing the input vector
s(n) ∈ C M×1 , where the integer number n is the time index, the natural number M denotes the
amount of symbols to be transmitted, and C ⊂ C is a given digital constellation. Such a processing
is implemented through the multiplication by the matrix
F
F ∈ CN ×M , (4.1)
0K×M
where N ∈ N is the total length of the transmitted message, whereas F ∈ CM×M is the actual
transmitter matrix, or just the transmitter matrix, which is in charge of preparing the vector for the
transmission through the FIR channel. The matrix 0K×M is responsible for adding the redundant
zero elements (zero-padding process). The amount of redundancy inserted is the natural number
K N − M. (4.2)
If one assumes that the channel order L is less than the total length of the transmitted message
N , then the received block y(n) ∈ CN ×1 will suffer from ISI and IBI effects according to the channel
model described in Chapter 2, Equation (2.39), which can be rewritten as
y(n) = HISI Fs(n) + HIBI Fs(n − 1) + v(n), (4.3)
where v(n) ∈ CN×1 accounts for an additive noise at the receiver front-end, as depicted, for example,
in Figure 2.9 of Chapter 2.
In order to generate an estimate ŝ(n) ∈ CM×1 of the transmitted data vector,3 the receiver
applies another linear transformation to the incoming signal vector through the matrix
G 0M×(L−K) G ∈ CM×N , (4.4)
where G ∈ CM×(N+K−L) is the actual receiver matrix, or just the receiver matrix. The matrix
0M×(L−K) is responsible for removing the remaining IBI (zero-jamming process) that was not com-
pletely eliminated by the transmitted redundancy.
3 Observe that the estimated elements do not have to pertain to the digital constellation C. Only after a hard-decision-detection
process the estimated elements are mapped into constellation symbols belonging to C.
108 4. MEMORYLESS LTI TRANSCEIVERS WITH REDUCED REDUNDANCY
Now, by using the definitions of HISI and HIBI given in Equation (2.38) of Chapter 2, the
previous ZP-ZJ model description yields the following estimate:
ŝ(n) Gy(n)
= GHISI Fs(n) + GHIBI F s(n − 1) + Gv(n)
=0M×M v(n)
= GHISI Fs(n) + v(n)
F
= 0M×(L−K) G HISI s(n) + v(n)
0K×M
=G H F s(n) + v(n)
T
= Ts(n) + v(n), (4.5)
Equation (4.5) shows that the estimated vector ŝ(n) does not depend on the transmitted vector
s(n − 1), which means that the overall transmission/reception process is performed on a block-by-
block basis. We can therefore simplify the notation by omitting the time-dependency of the related
variables. Thus, Equation (4.5) can be rewritten as
ŝ = Ts + v. (4.7)
Figure 4.1 illustrates the ZP-ZJ model omitting the time-dependency of the related variables.
Note that the choices for the dimensions of the all-zero matrices which implement the zero-padding
zero-jamming processes make sense. Indeed, by remembering the definition of matrix HIBI in
Equation (2.38), we can verify that only the first L elements of a received data block are affected by
the last L elements of the previous transmitted data block. This interblock interference occurs due to
the memory of the channel. Hence, the IBI effect can be eliminated by discarding the first L out of
N elements from all received data blocks. However, if one desires to transmit N = M + K elements,
4.2. REDUCED-REDUNDANCY SYSTEMS: THE ZP-ZJ MODEL REVISITED 109
v
⎡ ⎤ ⎡ ⎤
s0
T M ŝ0
⎢ s1 ⎥ M N HISI N N ⎢ ŝ1 ⎥
F 0(L−K)×M
s=⎢
⎣ ... ⎦
⎥ + T ⎢ . ⎥ = ŝ
⎣ .. ⎦
0K×M −1
G
sM −1 z HIBI ŝM −1
N =M +K
with K redundant zeros included at the end of each transmitted block, then instead of discarding the
first L elements from the received data block, one can discard only the first L − K elements, since the
immediately following L − (L − K) = K elements are “affected” by the zero-valued transmitted
elements appended at the end of the previous transmitted block (see Example 4.1 below).
In general, K can assume any non-negative integer value in order to eliminate the IBI effect.
In fact, if K ≥ L, then we must adjust the notation employed to define matrix G in Equation (4.4),
since L − K ≤ 0. In such a case, we could redefine G without zero jamming, i.e., G = G ∈ CM×N .
By doing so, we see that ZP-ZJ systems encompass general ZP-OFDM and ZP-SC-FD systems
which transmit K ≥ L redundant zeros.
Example 4.1 (Zero-Padding Zero-Jamming Process) Consider an FIR channel with transfer
function
Observe that, if we use the following matrices (see Equations (4.1) and (4.4))
I4
F= F, (4.10)
0
1×4
G = 04×2 G , (4.11)
110 4. MEMORYLESS LTI TRANSCEIVERS WITH REDUCED REDUNDANCY
in which F ∈ C4×4 and G ∈ C4×3 , then the IBI effect can be eliminated since
⎡ ⎤
0 0 h(3) h(2) h(1)
⎢0 0 0 h(3) h(2)⎥
⎢ ⎥ I4
GHIBI F = 04×2 G ⎢
⎢0 0 0 0 h(3)⎥⎥ 01×4 F
⎣0 0 0 0 0 ⎦
0 0 0 0 0
⎛ ⎞
⎜⎡ ⎤ ⎡ ⎤ ⎟
⎜ 0 0 h(3) h(2) h(1) ⎟
⎜ ⎟
⎜⎢ h(3)⎥ ⎢h(2)⎥ ⎟
⎜⎢0 0 0 ⎥ ⎢ ⎥ ⎟
= 04×2 G ⎜ ⎢
⎜⎢0 0 0 ⎥ I + ⎢ ⎥
0 ⎥ 4 ⎢h(3)⎥ 1×4 ⎟ 0 ⎟F
⎜⎣0 0 0 0 ⎦ ⎣ 0 ⎦ ⎟
⎜ ⎟
⎜ 0 ⎟
⎝ 0 0 0 0 ⎠
=05×4
⎡ ⎤
0 0 h(3) h(2)
⎢0 0 0 h(3)⎥
⎢ ⎥
= 04×2 G ⎢
⎢0 0 0 0 ⎥ ⎥F
⎣0 0 0 0 ⎦
0 0 0 0
⎛ ⎞
⎜ ⎟
⎜ 0 0 h(3) h(2) ⎟
=⎜ 0
⎜ 4×2 0 0 + G 0 ⎟
3×4 ⎟ F
⎝ 0 h(3) ⎠
=04×4
=04×4
= 04×4 . (4.12)
Moreover, it is also possible to verify that the structure of the equivalent channel matrix H
described in Equation (4.6) holds in this example. Indeed, we have
⎡ ⎤
h(0) 0 0 0 0
⎢h(1) h(0) 0 0 0 ⎥
⎢ ⎥ I4
GHISI F = 04×2 G ⎢
⎢h(2) h(1) h(0) 0 0 ⎥ ⎥ 01×4 F
⎣h(3) h(2) h(1) h(0) 0 ⎦
0 h(3) h(2) h(1) h(0)
4.2. REDUCED-REDUNDANCY SYSTEMS: THE ZP-ZJ MODEL REVISITED 111
⎛ ⎞
⎜⎡ ⎤ ⎡ ⎤ ⎟
⎜ h(0) 0 0 0 0 ⎟
⎜ ⎟
⎜⎢ 0 0 ⎥ ⎥ ⎢ ⎥ ⎟
⎜⎢h(1) h(0) ⎢ 0 ⎥ ⎟
= 04×2 G ⎜ ⎢
⎜⎢h(2) h(1) h(0) 0 ⎥ ⎢ ⎥ ⎟
⎥ I4 + ⎢ 0 ⎥ 01×4 ⎟ F
⎜⎣h(3) ⎦ ⎣ 0 ⎦ ⎟
⎜ h(2) h(1) h(0) ⎟
⎜ 0 ⎟
⎝ h(3) h(2) h(1) h(0) ⎠
=05×4
⎡ ⎤
h(0) 0 0 0
⎢h(1) h(0) 0 0 ⎥
⎢ ⎥
= 04×2 G ⎢
⎢h(2) h(1) h(0) 0 ⎥ ⎥F
⎣h(3) h(2) h(1) h(0)⎦
0 h(3) h(2) h(1)
⎛ ⎞
⎜ ⎡ ⎤⎟
⎜ h(2) h(1) h(0) 0 ⎟
⎜ h(0) 0 0 0 ⎟
=⎜
⎜ 0 4×2 +G ⎣h(3) h(2) h(1) h(0)⎦⎟ F
⎟
⎜ h(1) h(0) 0 0
⎝ 0 h(3) h(2) h(1) ⎟
=04×4 ⎠
=H∈C3×4
= G H F. (4.13)
As we have seen in Example 4.1, the redundancy is padded at the transmitter and jammed
at the receiver end in such a way that the IBI effect is completely eliminated. Nevertheless, the
amount of redundancy should be such that the ISI effect might also be eliminated, i.e., the amount
of redundant zeros should not be too small, otherwise it would prevent us from finding ZF solutions
(or, in other words, ISI elimination). Indeed, the definition of the M × M equivalent transfer matrix
T in Equation (4.5) makes clear that a zero-forcing solution is achievable if, and only if, the matrix
T = G H F is full-rank (i.e., rank{T} = M). Since the dimensions of matrices G, H, and F are
M × (M + 2K − L), (M + 2K − L) × M, and M × M, respectively, and since the rank of the
product of matrices is not larger than the rank of each individual matrix, then we must necessarily
have
GZF (H F)†
= [(H F)H (H F)]−1 (H F)H
H H H H
= [F H H F]−1 F H
−1 H
H
−1 H H
= F (H H)−1 F F H
−1 H H
=F (H H)−1 H
†
H
H †
=F H , (4.16)
−1
σ2
GMMSE (H F)H (H F) + v2 IM (H F)H
σs
# $−1
H H σv2 H H H
= F H HF + 2F F F H
σs
# $−1
−1 H σ2 H −1 H H
=F H H + v2 IM F F H
σs
# $−1
H H σv2 H
=F H H + 2 IM H , (4.17)
σs
where (·)† denotes the Moore-Penrose pseudo-inverse of (·). The optimal MMSE solution is de-
rived following the same procedure as in Subsection 2.4.1. In addition, the vectors s and v are
considered uncorrelated wide-sense stationary (WSS) random sequences with zero-mean,6 yielding
E[sv]H = E[s]E[v]H = 0M×M = E[v]E[s]H = E[vsH ]. We also assume that E[ssH ] = σs2 IM
and E[v v H ] = σv2 IM , in which σv2 , σs2 ∈ R+ denote the variances of the random sequences (they
do not depend on time because the random sequences are WSS). The given expression for GZF is
4 In Example 4.1, we see that, even though IBI is eliminated, it is not possible to achieve ISI elimination since matrix G ∈ C4×3
< =
and H ∈ C3×4 have ranks at most 3, instead of 4 = M. This happens since we adopted K = 1 < L 2 = 2, thus violating the
minimum required redundancy.
5The unitary property means F −1 = F H .
6 Remember that the time index was omitted for the sake of conciseness.
4.2. REDUCED-REDUNDANCY SYSTEMS: THE ZP-ZJ MODEL REVISITED 113
valid in the case H has full-rank. As a result, we are assuming that the FIR channel model leads to
a rank-M matrix H.
The ZF design does not require any estimate of the environment-noise variance and it is
mainly discussed for theoretical purposes. This lack of information about the noise variance leads
to poor performance when the SNR is low at the receiver end. On the other hand, the practical
MMSE design is computationally more demanding than the ZF solution, but leads to more effective
equalization.
The next step is to discuss how complex is performing both the equalization of a received
data block and the design of the related receiver. The equalization consists of processing the received
vector after the zero-jamming process through its multiplication by matrix G. This operation entails
O(M 2 ) complex-valued arithmetical operations for general unstructured matrices. The equalization
process requires the knowledge or the estimation of H and possibly its regularized pseudo-inverse.
This estimation is made during the receiver design. As a result, the computational complexity of the
receiver design is O(M 3 ) complex-valued arithmetical operations for general unstructured matrices.
The cyclic-prefix OFDM (CP-OFDM) and the CP-SC-FD transceivers have the advantage of
performing the equalization as well as the receiver design employing only O(M log2 M) complex-
valued operations due to their structural simplicity.
The use of discrete Fourier transform (DFT) and inverse DFT (IDFT) in order to decouple
the estimation of the symbols at the receiver end are paramount to the success of CP-OFDM-based
systems. Unfortunately, we cannot decouple so easily the estimation of the symbols in a ZP-ZJ
system with reduced redundancy. Indeed, such decoupling process would require the computation of
singular-value decompositions (SVDs), hindering its implementation in several practical problems.
Despite this potential drawback, we shall describe some tools that allow one to implement
low complexity ZP-ZJ systems with reduced redundancy. As a motivating example, let us consider
how a zero-forcing CP-SC-FD system is implemented. The insertion of the cyclic prefix turns the
linear convolution into a circular convolution between the transmitted data symbols and the channel-
impulse response. Using the vector notation for a noiseless channel, we can write y Hc s, where
Hc is a circulant matrix which contains the channel coefficients. From linear algebra, we know that
all circulant matrices may be diagonalized by using the same set of orthonormal eigenvectors. These
eigenvectors are the columns of the unitary DFT matrix, W. In addition, the eigenvalues of circulant
channel matrices are easily computed by means of the DFT of the first column of the circulant matrix
(see the discussions in Subsection 2.4.1). Thus, we have y = Hc s = W H Ws, which implies that
we can recover s by performing s = H−1 H −1
c y = W Wy, considering that
−1 is computable,
i.e., all eigenvalues of Hc are non-zero. Hence, the ZF-SC-FD system that employs cyclic prefix
decomposes the inverse of the equivalent channel matrix using DFT and diagonal matrices. In fact, this
decomposition is quite special since it is a diagonalization of the inverse of the equivalent channel
matrix.
Our aim is to follow a similar approach, that is, to look for an efficient decomposition for the
“inverse” of the equivalent channel matrix associated with ZP-ZJ systems with reduced redundancy.
114 4. MEMORYLESS LTI TRANSCEIVERS WITH REDUCED REDUNDANCY
In such systems, the equivalent channel matrix H is no longer circulant; rather, it is an (M + 2K −
L) × M Toeplitz matrix, as described in Equation (4.6). Nevertheless, we could take into account
the Toeplitz structure in order to decompose the generalized inverse of H, maybe using only DFT
and diagonal matrices. Such an approach employs the same basic ideas present in CP-OFDM-based
systems, except for two main features present only in OFDM-based systems: (i) the inverse of the
equivalent channel matrix has exactly the same structure as the equivalent channel matrix itself
(circulant structure); and (ii) the efficient decomposition of the inverse of the equivalent channel
matrix corresponds to its diagonalization.
Dν diag{νν }
⎡ ⎤
ν0
⎢ ν1 ⎥
⎢ ⎥
=⎢ .. ⎥
⎣ . ⎦
νM−1
= ν0 e0 ν1 e1 · · · νM−1 eM−1 , (4.21)
Zλ−1 = ZT1
λ
= ZH1 , (4.22)
λ∗
Hence, we have
⎡ ⎤
2 3 −1
CZ−1 = ⎣5 6 −4⎦ , (4.27)
8 9 −7
⎡ ⎤
−7 −8 −9
Z−1 C = ⎣ 1 2 3 ⎦. (4.28)
4 5 6
Therefore, right-multiplication by a λ-circulant matrix shifts all columns to the left, where the first
original column multiplied by λ is moved to the last column of the resulting matrix. On the other
hand, left-multiplying by a λ-circulant matrix shifts down all rows, where the last original row
multiplied by λ is moved to the first row of the resulting matrix.
Usually, structured matrices can be associated with some linear displacement operator. These
operators might reveal if a given structured matrix can be represented by a reduced number of
parameters. This representation is the key feature that allows the derivation of superfast algorithms 9
for inverting as well as performing matrix-to-vector multiplication involving the related structured
matrix.
The fast implementation of the reduced-redundancy transceivers rely on the displacement
rank of the matrices involved. The procedure entails the following steps.
2. Operation: The idea behind the displacement approach is that the compressed form of a
structured matrix contains all the information about the original matrix, but with a reduced
amount of elements. Therefore, instead of performing operations with the original matrix, it
is worth using the appropriate displacement-generator pair to perform such operations.
3. Decompression: Once the required operations are performed, the processed version of the orig-
inal matrix can be recovered through a displacement decompression operation, as long as the
operator matrices satisfy some mild constraints.
In order to illustrate the power of the displacement-rank tool, let us define some types of
structured matrices in a more formal manner in Subsection 4.3.2.
the (m2 , m1 )th element of T , denoted as [T T ]m2 m1 tm2 ,m1 ∈ C, satisfies the equality tm2 ,m1 =
t(m2 +1),(m1 +1) , as long as the pair of indexes (m2 + 1, m1 + 1) is also within the set M2 × M1 . In
4.3. STRUCTURED MATRIX REPRESENTATIONS 119
this case, matrix T can be rewritten in a more convenient form, as follows:
⎡ ⎤
t0 t−1 t−2 ··· t1−M1
⎢ t1 t0 t−1 ··· t2−M1 ⎥
⎢ ⎥
⎢ ··· ⎥
T =⎢ t2 t1 t0 t3−M1 ⎥
⎢ .. .. .. .. .. ⎥
⎣ . . . . . ⎦
tM2 −1 t tM2 −3 · · · tM2 −M1
M2 −2
t(m2 −m1 ) (m2 ,m1 )∈M2 ×M1
. (4.36)
In order to exemplify the compression capability of the displacement operators when dealing
with rectangular Toeplitz matrices, let us consider the application of the Sylvester displacement op-
erator ∇Zη ,Zξ , in which Zη ∈ CM2 ×M2 and Zξ ∈ CM1 ×M1 , on an M2 × M1 complex-valued Toeplitz
matrix T :
∇Zη ,Zξ (T
T ) = ZηT − T Zξ
⎡ ⎤
ηtM2 −1 ηtM2 −2 · · · ηtM2 −M1 +1 ηtM2 −M1
⎢ t0 t−1 ··· t2−M1 t1−M1 ⎥
⎢ ⎥
⎢ ··· ⎥
= ⎢ t1 t0 t3−M1 t2−M1 ⎥
⎢ . .. .. .. .. ⎥
⎣ .. . . . . ⎦
tM2 −2 tM2 −3 ··· tM2 −M1 tM2 −M1 −1
⎡ ⎤
t−1 t−2 ··· t1−M1 ξ t0
⎢ t0 t−1 ··· t2−M1 ξ t1 ⎥
⎢ ⎥
⎢ .. .. .. .. .. ⎥
−⎢ . . . . . ⎥
⎢ ⎥
⎣tM tM2 −4 · · · tM2 −M1 −1 ξ tM 2 ⎦
2 −3 −2
tM2 −2 tM2 −3 · · · tM2 −M1 ξ tM2 −1
⎡ ⎤
ηtM2 −1 − t−1 ηtM2 −2 − t−2 · · · ηtM2 −M1 +1 − t1−M1 ηtM2 −M1 − ξ t0
⎢ 0 0 ··· 0 t1−M1 − ξ t1 ⎥
⎢ ⎥
⎢ .. .. .. .. .. ⎥
=⎢ . . . . . ⎥
⎢ ⎥
⎣ 0 0 ··· 0 tM2 −M1 −2 − ξ tM2 −2
⎦
0 0 ··· 0 tM2 −M1 −1 − ξ tM2 −1
120 4. MEMORYLESS LTI TRANSCEIVERS WITH REDUCED REDUNDANCY
⎡ ⎤
1
⎢0 ⎥
⎢ ⎥
⎢ ⎥
= ⎢ ... ⎥ ηtM2 −1 − t−1 ηtM2 −2 − t−2 · · · ηtM2 −M1 +1 − t1−M1 ηtM2 −M1
⎢ ⎥
⎣0 ⎦
q̂1H
0
p̂1
⎡ ⎤
−ξ t0
⎢ t1−M1 − ξ t1 ⎥
⎢ ⎥
⎢ .. ⎥
+⎢ . ⎥ 0 0 ··· 0 1
⎢ ⎥
⎣tM − ξ tM 2 ⎦
2 −M1 −2 −2 q̂2
H
tM −M −1 − ξ tM2 −1
2 1
p̂2
q̂1H
= p̂1 p̂2
q̂2H
P̂
Q̂ H
= P̂Q̂ H
. (4.37)
Note that the resulting displacement matrix ∇Zη ,Zξ (T T ) can be represented by the
displacement-generator pair of matrices (P̂, Q̂ ) ∈ C M 2 ×2 × C ×2 . Thus, if one assumes that M2
M 1
and M1 are integer numbers much larger than 2, then the former example shows that rectangular
Toeplitz matrices can always be compressed, since matrix ∇Zη ,Zξ (T T ) has rank at most 2. In fact,
since the vectors p̂1 and q̂2 do not depend on T , then at most M2 + M1 coefficients (corresponding
to the elements of the vectors p̂2 and q̂1 ) are really required to represent the displacement of T . This
reduced amount of coefficients is rather close to the total number of possibly different entries of
matrix T , which is M2 + M1 − 1.
Now, let us define another important class of structured matrix: the Vandermonde matrix,
which is a matrix whose entries in each row are the terms of a geometric progression, as can be seen
in Equation (4.38) below. Indeed, given an M × 1 complex-valued vector ν [ ν0 ν1 · · · νM−1 ]T ,
an M × M matrix V ν is a Vandermonde matrix when, for each pair of indexes (m2 , m1 ) within the
set M2 M × M, the (m2 , m1 )th element of V ν , denoted as [V V ν ]m2 m1 νm2 ,m1 ∈ C, satisfies
4.3. STRUCTURED MATRIX REPRESENTATIONS 121
m1
the equality νm2 ,m1 = νm 2 . In this case, V ν presents the following form:
⎡ ⎤
ν00 ν01 · · · ν0M−2 ν0M−1
⎢ ν10 ν11 · · · ν1M−2 ν1M−1 ⎥
⎢ ⎥
⎢ .. ⎥
Vν = ⎢
⎢
..
.
..
.
..
.
..
.
⎥
. ⎥
⎢ 0 M−1 ⎥
⎣νM−2 νM−2 1 · · · νM−2
M−2
νM−2 ⎦
0
νM−1 νM−1 1 · · · νM−1
M−2 M−1
νM−1
[νm
m1
]
2 (m2 ,m1 )∈M
2. (4.38)
As in the Toeplitz case (see Equation (4.37)), a similar kind of compression can be applied
to Vandermonde matrices as well. Let us consider the application of the Sylvester displacement
operator ∇Dν ,Z0 , in which Dν = diag{νν } ∈ CM×M , on a given M × M Vandermonde matrix V ν .
In this case, we have
V ν ) = Dν V ν − V ν Z0
∇Dν ,Z0 (V
⎡ ⎤ ⎡ ⎤
ν0 · · · ν0M−1 ν0M ν0 · · · ν0M−1 0
⎢ ν1 · · · ν1M−1 ν1M ⎥ ⎢ ν1 · · · ν1M−1 0⎥
⎢ ⎥ ⎢ ⎥
=⎢ . .. .. ⎥−⎢ .. .. .. ⎥
⎣ .. . . ⎦ ⎣ . . .⎦
νM−1 · · · νM−1
M−1 M
νM−1 νM−1 · · · νM−1
M−1
0
⎡ ⎤
0 · · · 0 ν0M
⎢0 · · · 0 ν M ⎥
⎢ 1 ⎥
= ⎢. .. .. ⎥ , (4.39)
⎣ .. . . ⎦
0 · · · 0 νM−1
M
which consists of a rank-1 matrix with M degrees of freedom. Note that, even though the original
Vandermonde matrix V ν is comprised of M 2 entries, the M elements which compose the vector ν are
enough to completely define V ν . This compression example, therefore, shows that the displacement
approach is able to reveal analytically this reduced number of degrees of freedom.
In addition, there is a close relation between Vandermonde and DFT matrices. In order to
derive such a useful relation, let us first remember that the Mth roots of a given complex number
ξ consist of M distinct complex numbers ξm , with m ∈ M, such that ξmM = ξ = |ξ |ej ξ , where
j = −1 and ξ ∈ (−π, π ] ⊂ R represents the (principal) phase of the complex number ξ when
2 10
10 Remember that, if ξ is a phase of a given complex number ξ , then ξ + 2iπ is also a phase of ξ , for any integer number i. The
principal phase is the unique phase of ξ within the interval (−π, π ].
122 4. MEMORYLESS LTI TRANSCEIVERS WITH REDUCED REDUNDANCY
expressed in its polar form; that is,
1 ξ 2π
ξm |ξ | M ej M e−j M m
ξ0
⎛ ⎞m
⎜ 2π ⎟
= ξ0 ⎝e−j
⎠
M
WM
= ξ0 WM
m
. (4.40)
where WM is the M × M unitary DFT matrix previously defined in Equation (2.46). Such a
relation between Vandermonde and DFT matrices will be key to developing efficient factorizations
of structured matrices which appear in the ZP-ZJ model.
Another important structured matrix is the so-called Cauchy matrix, which may not be
T
familiar to some readers. Given two vectors, ν ν0 ν1 · · · νM1 −1 ∈ CM1 ×1 and λ
T
λ0 λ1 · · · λM2 −1 ∈ CM2 ×1 , an M1 × M2 complex-valued matrix C ν ,λλ is a Cauchy matrix
pairof indexes (m1 , m2 ) within the set M1 × M2 , the (m1 , m2 )th element of C ν ,λλ ,
when, for any
denoted as C ν ,λλ m m cm1 ,m2 ∈ C, satisfies the equality cm 1,m = 1 − νm1 λm2 , assuming that
1 2 1 2
4.3. STRUCTURED MATRIX REPRESENTATIONS 123
νm1 λm2 = 1. Thus, C ν ,λλ can be represented as follows:
⎡ ⎤
1 1
1−ν0 λ0 1−ν0 λ1 ··· 1
1−ν0 λ(M2 −1)
⎢ ⎥
⎢ 1 1
··· 1 ⎥
⎢ 1−ν1 λ0 1−ν1 λ1 1−ν1 λ(M2 −1) ⎥
C ν ,λλ =⎢ .. .. .. ⎥
⎢ ..
. ⎥
⎣ . . . ⎦
1
1−ν(M1 −1) λ0
1
··· 1
1−ν(M1 −1) λ1 1−ν(M1 −1) λ(M2 −1)
1
. (4.43)
1 − ν m 1 λm 2 (m1 ,m2 )∈M1 ×M2
A Cauchy matrix C ν ,λλ can be easily compressed by using the Stein displacement operator
Dν ,Dλ . Indeed, one has
which is also a rank-1 matrix. In fact, the resulting displacement matrix in Equation (4.44) enjoys
the most simple form that we have found up to now. This feature will be important for us in
g
Subsection 4.4.1. Moreover, a rectangular matrix C ν ,λλ is defined as a generalized Cauchy matrix, as
g
long as the Stein displacement matrix Dν ,Dλ (C C ν ,λλ ) has rank much smaller than the number of
g
rows and columns of C ν ,λλ .11
The last class of structured matrix of our interest is the Bezoutian matrix. As opposed to most
of the previous structured matrices, there is no simple closed-form expression for the entries of a
Bezoutian matrix. In fact, the easiest way to define a Bezoutian matrix is based on its displacement
rank, as in the case of generalized Cauchy matrices. Indeed, an M1 × M2 Bezoutian matrix B is such
that
B ) = PQ
∇Zν ,Zλ (B H
(4.45)
for some complex-valued scalars ν and λ, where (P, Q ) ∈ CM1 ×R × CM2 ×R , and with M2 R
and M1 R.
It is possible to show (see Subsection 4.3.3) that the inverse of a non-singular square Toeplitz
matrix is a particular type of Bezoutian matrix, also known as T-Bezoutian matrix.
11There are some other classes of Cauchy matrices which will not be of particular importance in this book. The interested reader
should refer to [27, 28] for detailed information on this topic.
124 4. MEMORYLESS LTI TRANSCEIVERS WITH REDUCED REDUNDANCY
So far, we have only illustrated the compression capabilities of the displacement operators.
Let us analyze now the operation stage of the displacement approach.
∇A,B (C) = AC − CB
= A C − A −1 CB
= AA −1 ,B (C), (4.46)
in which it is assumed that all matrices have compatible dimensions.12 Similarly, if B is invertible,
then
These relations between the Sylvester and Stein displacement operators show that, given two operator
matrices A and B, the displacement rank of a third matrix C does not depend on the particular
choice of the displacement operator (whether Sylvester or Stein), as long as at least one of the
operator matrices has full-rank and, in addition, some minor adjustments on the operator matrices
are performed, as for example, replacing A by A −1 or B by B−1 .
Let us now discuss two important results dealing with linear combinations of matrices and
matrix products, which basically show how these operations affect the displacement-generator
pairs of the original matrices. With respect to the generator of a linear combination, assume that
∇A,B (C) = PC QCH and that ∇A,B (D) = PD QDH , where matrices C and D have the same dimen-
12That is, the number of rows and columns are such that one can perform operations with the involved matrices, like addition or
product, without worrying about the dimensions of the related matrices.
4.3. STRUCTURED MATRIX REPRESENTATIONS 125
sions. Thus, for any complex-valued scalar α, one always has
= PQ H
. (4.48)
As for the generator of a product, consider that ∇A,B (C) = PC QCH and that ∇B,D (E) = PE QEH ,
where all matrices have compatible dimensions. Hence, we have
= PC QCH E + C PE QEH
Q HE
= PC CPE C
QEH
P
Q H
= PQ H
. (4.49)
Thus, the knowledge about the displacement-generator pairs of the individual matrices before the
algebraic operation (whether linear combination or product) helps one determine the displacement-
generator pair of the resulting matrix after the related operation.
Now, we will analyze the inversion operation. The importance of this result is the fact that the
compression of an inverse matrix can be easily generated from the compressed version of the original
matrix. Indeed, given an invertible square complex-valued matrix C and the operator matrices A
and B, one has
= B − C−1 AC C−1
= C−1 CB − AC C−1
= −C−1 ∇A,B (C)C−1 . (4.50)
In other words, if the displacement-generator pair of ∇A,B (C) is (PC , QC ), then the displacement-
generator pair of ∇B,A (C−1 ) is (−C−1 PC , C−H QC ), in which C−H denotes the Hermitian of the
inverse of C.
An immediate application of the inversion property is the verification of the following fact:
for a given non-singular square Toeplitz matrix T ∈ CM×M whose displacement generator pair
126 4. MEMORYLESS LTI TRANSCEIVERS WITH REDUCED REDUNDANCY
associated with the Sylvester displacement operator ∇Zη ,Zξ is (P̂, Q̂ ), one has ∇Zξ ,Zη (T T −1 ) =
(−T T −1 P̂)(Q̂ H T −1 ), whose rank is at most 2 due to Equation (4.37). Therefore, T −1 is a Bezoutian
matrix whenever M 2.
Actually, any regularized pseudo-inverse of an M2 × M1 complex-valued Toeplitz matrix T
is a Bezoutian matrix, as long as min{M2 , M1 } 4. Indeed, given a strictly positive real number ρ,
the regularized pseudo-inverse of T is defined as
−1
T †ρ T H T + ρIM1 T H. (4.51)
−1
Note that, by applying the matrix inversion lemma 13 to the expression T H T + ρIM1 that appears
in Equation (4.51), one has
−1 # $−1
1 1 H 1 1
T T + ρIM1
H
= IM1 − T T T + IM 2
H
T
ρ ρ ρ ρ
1
−1
= IM1 − T H
T T + ρIM2
H
T , (4.52)
ρ
−1
1
T †ρ = T −T
H H
T T + ρIM2
H
TT H
ρ
1 H
−1
= T IM2 − T T + ρIM2
H
TT H
ρ
1
−1
= T H T T H + ρIM2 T T H + ρIM2 − T T H
ρ
1
−1
= T H T T H + ρIM2 ρIM2
ρ
−1
= T H T T H + ρIM2 . (4.53)
−1
−1
13The following identity holds: A + BCDH = A −1 − A −1 B C−1 + DH A −1 B DH A −1 , assuming that all operations
are valid (see, for example, [83]).
4.3. STRUCTURED MATRIX REPRESENTATIONS 127
Therefore, the Sylvester displacement of the regularized pseudo-inverse described in Equa-
tions (4.51) and (4.53) is given by
−1
−1
∇Zξ ,Zη T †ρ = Zξ T H T T H + ρIM2 − T H T + ρIM1 T H Zη
−1
= T H T + ρIM1 T H T + ρIM1 Zξ T H
−1
− T H Zη T T H + ρIM2 T T H + ρIM2
−1
= T H T + ρIM1 T H T Z ξ T H − T H Zη T T H
−1
+ ρ Zξ T H − T H Zη T T H + ρIM2
−1
= T H T + ρIM1 −TT H ∇Zη ,Zξ (T
T )T H
−1
+ ρ∇Zξ ,Zη T H T T H + ρIM2 . (4.54)
Hence, if we define ∇Zη ,Zξ (T
T ) as P̂Q̂ H and ∇Zξ ,Zη T H as P̌Q̌ H , then we have
−1
−1
∇Zξ ,Zη T †ρ = −TT †ρ P̂Q̂ H T †ρ + T H T + ρIM1 P̌Q̌ H ρ T T H + ρIM2
−1
H ρ T T H + ρI
−1 Q̌ M
= T H T + ρIM1 P̌ −T
T †ρ P̂ 2
Q̂ H T †ρ
M ×4
P∈C 1
Q H ∈C4×M2
= PQ H
, (4.55)
Q̂ H T †
M ×4
P∈C 1
Q H ∈C4×M2
= PQ H
. (4.57)
The reader should notice the similarity between the above (regularized) pseudo-inverses and
the optimal ZF and MMSE linear solutions defined in Equations (4.16) and (4.17), apart from the
transmitter-matrix inverse which appears only in the definitions of the optimal receiver matrices.14
This means that the optimal receivers associated with ZP-ZJ transceivers are, essentially, Bezoutian
matrices multiplied by a predefined unitary matrix F. Hence, inspired by the fact that the standard
CP-OFDM and CP-SC-FD transceivers decompose the inverse of circulant channel matrices using
DFT, IDFT, and diagonal matrices, it seems to be a good idea to look for decompositions of general
Bezoutian matrices entailing the same type of matrices, namely DFT, IDFT, and diagonal.
Cauchy
Matrix
Factorized Efficient Factorized Efficient
Bezoutian Transformation Cauchy Decomposition
Matrix Matrix
This section describes how one can decompose a particular type of Cauchy matrix using
DFTs (Subsection 4.4.1) and, after that, how one can transform Bezoutian matrices into Cauchy
matrices in a simple and effective manner (Subsection 4.4.2). As a result, one arrives at an efficient
representation of Bezoutian matrices (Subsection 4.4.3).
T
ν ν0 ν1 · · · νM1 −1 ,
T
λ λ0 λ1 · · · λM2 −1 , (4.58)
m1 m2
where νm1 ν0 WM 1
, for all m1 within the set M1 , whereas λm2 λ0 WM 2
, for all m2 within the
set M2 (see Equation (4.40)). Let us also assume that M2 ≥ M1 . Thus, from expression (4.43), we
130 4. MEMORYLESS LTI TRANSCEIVERS WITH REDUCED REDUNDANCY
have
1
Cν,λ m =
1 m2 1 − ν m 1 λm 2
1
=
m1 m2
1 − ν 0 WM λ0 W M2
1
m1 m2 M 1
1 − (ν0 λ0 WM W M2 ) 1
= m1 m2 M 1 ×
1
1 − (ν0 λ0 WM WM ) 1 − W m1
ν λ W m2
1 2 M1 0 0 M 2
⎡ ⎤
m1 m2 M 1
1 1 − (W ν λ W
0 0 M2 )
= m1 m2 M 1
⎣ M1
⎦
1 − (ν0 λ0 WM 1
W M 2
) 1 − W m1
ν λ
0 0 W m2
M1 M2
1
M 1 −1
m
m1 m2
= m1 M1 m2 M 1 WM ν0 λ 0 W M
1 − (ν0 WM 1
) (λ0 WM 2
) m=0
1 2
M 1 −1
1 m1 m mm2
= WM (ν0 λ0 )m WM , (4.59)
1 − νλM
m2
1
m=0
1 2
1 − νλm
M1
2
= 0, ∀m2 ∈ M2 . (4.60)
mm2
Now, let us observe that, for each m within the set M1 , we have that WM is the mth entry
√ m1 m m
2
of the m2 th column of the matrix M2 WM2 , while WM1 (ν0 λ0 ) is the mth entry of the m1 th
√
row of the matrix M1 WM1 diag{(ν0 λ0 )m1 }m1 ∈M1 0M1 ×(M2 −M1 ) . Therefore, we can rewrite
Equation (4.59) as
√
M2 M1
Cν,λ m m = M1
WM1 diag{(ν0 λ0 )m1 }m1 ∈M1 0M1 ×(M2 −M1 ) WM2 m m , (4.61)
1 2
1 − νλm2 1 2
yielding
,√ -
M2 M1
Cν,λ = WM1 diag{(ν0 λ0 )m1 }m1 ∈M1 0M1 ×(M2 −M1 ) WM2 diag . (4.62)
1 − νλM
m2
1
m2 ∈M2
Equation (4.62) contains an efficient matrix factorization which is also useful to express
generalized Cauchy matrices. Indeed, as described in Subsection 4.3.2, we know that an M1 × M2
generalized Cauchy matrix Cν,λ g is such that
g
C ν,λ ) =
Dν ,Dλ (C pr qrH , (4.63)
r∈R
4.4. DFT-BASED REPRESENTATIONS OF BEZOUTIAN MATRICES 131
where R {0, 1, · · · , R − 1}, pr ∈ CM1 ×1 and qr ∈ CM2 ×1 are vectors, and (in fact, this is the
main condition) R M1 ≤ M2 . Observe that, by the definition of Stein displacement expressed in
Equation (4.19), we have
g g g
Dν ,Dλ (CC ν,λ ) = C ν,λ − Dν C ν,λ Dλ
m1 m 2
m1 m2
g g
= C ν,λ − Dν C ν,λ Dλ
m1 m2
m1 m2
g g
= C ν,λ − Dν m m C ν,λ Dλ m m
m1 m 2
1 1 m1 m2 2 2
g
= (1 − νm1 λm2 ) C ν,λ , (4.64)
m1 m2
Equation (4.67) is a quite efficient way of decomposing a given generalized Cauchy matrix,
since the decomposition only employs DFT and diagonal matrices. What remains is to find a way
of transforming a given rectangular Bezoutian matrix into such a type of Cauchy matrix.
132 4. MEMORYLESS LTI TRANSCEIVERS WITH REDUCED REDUNDANCY
4.4.2 TRANSFORMATIONS OF BEZOUTIAN MATRICES INTO CAUCHY
MATRICES
This subsection describes a computationally efficient way to transform rectangular Bezoutian ma-
trices into rectangular Cauchy matrices in order to benefit from the fact that Cauchy matrices enjoy
efficient decompositions. The transformation which we shall describe is based on the displacement
operators of Sylvester and Stein.15
From a displacement-approach point of view, we know that Bezoutian and Cauchy matrices
have an inherent feature in common: their resulting displacement matrices have small rank, as
compared to the dimensions of the original matrices. At a first look, it seems that the similarities
end here, since Bezoutian matrices employ Sylvester displacements, whereas Cauchy matrices use
Stein displacements. In addition, the operator matrices utilized to compress both classes of matrices
are also of different types. These facts might erroneously lead us to conclude that the transformation
from Bezoutian to Cauchy matrices is not possible.
Fortunately, this is not the case, i.e., it is possible to transform Bezoutian into Cauchy matrices.
Indeed, the difference between the Sylvester and Stein displacements is not a problem since we know
how to convert one into another, as explained in Equations (4.46) and (4.47). With regard to the
difference between the operator-matrix types, a Bezoutian matrix uses ν-circulant operator matrices,
such as Zν , whereas a Cauchy matrix uses diagonal operator matrices, let us say Dν . Even though
these operator matrices are quite different, there is a simple relationship between them which will
help us link Bezoutian to Cauchy matrices.
Consider the particular case in which vector ν is comprised of all distinct Mth roots of ν ∈ C,
i.e.,
T
ν ν0 ν1 · · · νM−1 , (4.68)
1 ν
where ν0 = |ν| M ej M and νm = ν0 WM m , for all m within the set M. Now, by remembering the
V ν Zν = Dν V ν (4.69)
15The results derived here are heavily based on the outstanding works developed by Heinig and Rost, for example, in [27]. These
authors, however, employed a polynomial formulation to develop most of their results.
4.4. DFT-BASED REPRESENTATIONS OF BEZOUTIAN MATRICES 133
and, for the useful case where ν = 0, one has
Zν = V ν−1 Dν V ν . (4.70)
We can now proceed to derive the expected transformation from Bezoutian to Cauchy ma-
trices. We know that, given two natural numbers M1 and M2 , with M1 ≤ M2 , and given two non-
zero complex numbers ξ and η, an M1 × M2 complex-valued Bezoutian matrix B is such that
∇Zξ ,Zη (B
B ) = PQ H , where the operator matrices have compatible dimensions and the resulting
displacement rank is much smaller than M1 . Our first task is to change from the Sylvester to the
Stein displacement and, after that, to convert the circulant operator matrices into diagonal ones.
Thus, by defining η 1/η and by using the relations in Equations (4.22), (4.47), and (4.70), we
have
PQ H
= ∇Zξ ,Zη (B
B)
= Zξ B − B Z η
= −Zξ ,ZT (B B )Zη
1/η
= − B − Zξ B ZηT Zη
= − B − V ξ−1 Dξ V ξ B V ηT Dη V η−T Zη
V ξ−1 V ξ B V ηT − Dξ V ξ B V ηT Dη V η−T Zη
= −V
V ξ−1 Dξ ,Dη (V
= −V V η−T Zη ,
V ξ B V ηT )V (4.71)
implying that
V ξ B V ηT ) = −V
Dξ ,Dη (V V ξ PQ H ZηH∗ V ηT
H
= −V V ξ P V η∗ Zη∗ Q
P
Q
H
H
=PQ
H
= p
0 p
1 · · · p
R−1 q0
q1
· · · qR−1 , (4.72)
V ξ B V ηT )} = rank{∇Zξ ,Zη (B
rank{Dξ ,Dη (V B )} ≤ R M1 . (4.73)
which yields
V ξ−1
B =V Dp
r WM1 diag{(ξ0 η0 )m1 }m1 ∈M1 0M1 ×(M2 −M1 )
r∈R ,√ -
M2 M1
×WM2 D∗q
diag V η−T . (4.75)
r
1 − ξ ηM
m2
1
m2 ∈M2
Equation (4.75) represents the last step of the displacement approach, namely the decom-
pression operation. It is worth mentioning that such expression is a rather efficient decomposition
of a general rectangular Bezoutian matrix, since it employs only DFT and diagonal matrices, and
considering that a given Vandermonde matrix can be described as a product of a DFT and a diagonal
matrix, as described in Equation (4.42).
Example 4.3 (Bezoutian Factorization) Let us analyze a numerical toy example which will help
us better understand the computations involved in Bezoutian factorizations. Consider the following
4 × 3 Toeplitz matrix
⎡ ⎤
2 1 0
⎢3 2 1⎥
T =⎢ ⎣0 3 2⎦ ,
⎥ (4.76)
0 0 3
−1 H
whose associated Moore-Penrose pseudo-inverse T † = T H T T is given by16
⎡ ⎤
0.1633 0.2245 −0.2041 0.0612
T † ≈ ⎣−0.0028 0.0019 0.3330 −0.2226⎦ . (4.77)
−0.0334 0.0223 −0.0037 0.3284
m1
m1 =2 −1
V ξ−1 = 3W3 diag{2 3 }m 1 =0
1 m
− 31 m1 =2
m2
m2 =3 −T
V η−T = 4W4 diag{1 4 }m 2 =0
1 H
= W4
2⎡ ⎤
1 1 1 1
1 ⎢1 j −1 −j ⎥
= ⎢ ⎥. (4.79)
4 ⎣1 −1 1 −1⎦
1 −j −1 j
In addition, we have
⎡ ⎤
1 0 0 0
⎢ 1 ⎥
diag{(ξ0 η0 )m1 }m1 ∈M1 0M1 ×(M2 −M1 ) = ⎣0 2 3 0 0⎦ , (4.80)
1
0 0 43 0
and
⎡ ⎤
,√ - −1 0 0 0
M2 M1 √ ⎢
⎢0
1
1−2j 0 0 ⎥
⎥
diag = 2 3⎢ 1 ⎥. (4.81)
1 − ξ ηM
m2
1 ⎣0 0 3 0 ⎦
m2 ∈M2 1
0 0 0 1+2j
The reader should note that we have just defined almost all matrices to be employed in the
factorization, but we have not used any information about matrix T yet. Indeed, all information of T is
contained in the entries of the diagonal matrices Dp
r and Dqr
, with r ∈ {0, 1, · · · , R − 1}. Thus, we
have to determine vectors p
r and qr
based on Equation (4.72). But, before that, we have to determine
matrices P and Q based on Equation (4.57). These latter matrices, on the other hand, depend on
the knowledge of the displacement matrices ∇Zη ,Zξ (T T ) = P̂Q̂ H and ∇Zξ ,Zη T H = P̌Q̌ H , which
136 4. MEMORYLESS LTI TRANSCEIVERS WITH REDUCED REDUNDANCY
can be straightforwardly acquired using Equation (4.37), as follows:
⎡ ⎤
1 −4
⎢0 −6⎥
P̂ = ⎢ ⎥
⎣0 1 ⎦ , (4.82)
0 2
⎡ ⎤
−1 0
Q̂ = ⎣ 0 0⎦ , (4.83)
3 1
and
⎡ ⎤
1 −2
P̌ = ⎣0 −1⎦ , (4.84)
0 0
⎡ ⎤
−3 0
⎢ 2 0⎥
Q̌ = ⎢ ⎥
⎣ 4 0⎦ . (4.85)
6 1
(4.88)
⎡ ⎤
−1.2022 0.0594 0.6957 0.3135
⎢−1.6531 − 1.8033j 0.0816 + 0.0891j 1.0816 − 0.4564j 0.3061 − 0.0297j⎥
Q
≈⎢
⎣
⎥.
⎦
3.3061 −0.1633 0.8367 0.3878
−1.6531 + 1.8033j 0.0816 − 0.0891j 1.0816 + 0.4564j 0.3061 + 0.0297j
(4.89)
4.5. REDUCED-REDUNDANCY SYSTEMS 137
By using matrices P
and Q we can define the diagonal matrices Dp
r and Dqr
, with r ∈ {0, 1, 2, 3}.
For example, we have
⎡ ⎤
0.0594 0 0 0
⎢ 0 0.0816 + 0.0891j 0 0 ⎥
Dq1
≈ ⎢⎣ 0
⎥.
⎦ (4.90)
0 −0.1633 0
0 0 0 0.0816 − 0.0891j
It is possible to verify that Equation (4.75) holds for this particular case using the numerical values
in Equations (4.78), (4.79), (4.80), (4.81), (4.88), and (4.89).
K F G ∈ CM×(M+2K−L) , (4.91)
with 2K ≥ L, then one can easily verify based on Equations (4.16) and (4.17) that
†
KZF = H , (4.92)
# $−1
H σv2 H
KMMSE = H H + 2 IM H (4.93)
σs
are (possibly regularized) pseudo-inverses of the tall channel Toeplitz matrix H, thus implying that
the transmitter-independent receiver matrix is also a particular type of rectangular Bezoutian matrix.
Hence, we can tailor the efficient decompositions of Bezoutian matrices previously described
in Subsection 4.4.3 to the particular cases of MMSE and ZF transmitter-independent receiver
matrices. As a result, a family of superfast multicarrier and single-carrier linear transceivers are
derived with their respective structures.
138 4. MEMORYLESS LTI TRANSCEIVERS WITH REDUCED REDUNDANCY
Firstly, we must carefully choose both parameters ξ and η which appear in the defini-
tion of the operator matrices employed in the compression stage of both the Toeplitz chan-
π
nel matrix and the receiver Bezoutian matrix. Let us assume that ξ = 1 and η = ej M . In this
case, by using Equation (4.37) one can readily compute the displacement-generator pair of ma-
trices (P̂, Q̂ ) ∈ C(M+2K−L)×2 × CM×2 and (P̌, Q̌
) ∈ CM×2 × C(M+2K−L)×2 Hassociated
with
H
matrices H and H , respectively, where ∇Zη ,Zξ H = P̂Q̂ and ∇Zξ ,Zη H
H = P̌Q̌ . By
H
using such displacement-generator pairs, one can precompute the displacement-generator pair
(P, Q ) ∈ CM×4 × C(M+2K−L)×4 associated with the transmitter-independent receiver matrix,
where ∇Zξ ,Zη (K) = PQ H . Indeed, for the MMSE solution, Equation (4.55) gives us
−1
H σv2
P = H H + 2 IM P̌ −KMMSE P̂ , (4.94)
σs
−H
2 H 2
Q = σv2 H H + σv2 I(M+2K−L) H
Q̌ KMMSE Q̂ , (4.95)
σs σs
Thus, by considering the previous compressed forms of the Bezoutian matrix K for either the
MMSE or the ZF solutions, one can employ the factorization in Equation (4.75), which depends
on the Vandermonde matrices Vξ and Vη . Since ξ = 1 and η = 1/η = e−j M , then
π
ξ
ξ0 = |ξ |1/M ej M
= 1 × e0
=1 (4.98)
and
η
η0 = |η|1/(M+2K−L) ej (M+2K−L)
−jπ
= 1 × e M(M+2K−L)
−jπ
= e M(M+2K−L) . (4.99)
These facts, along with Equation (4.42), imply that
√
Vξ = MWM , (4.100)
whereas
√ −jπ m
(M+2K−L−1)
Vη = M + 2K − L × W(M+2K−L) diag{e M(M+2K−L) }m=0 . (4.101)
4.5. REDUCED-REDUNDANCY SYSTEMS 139
We can therefore apply the decomposition presented in Equation (4.75) to obtain the following
result:
3
K = WMH
Dpr WM DM 0M×(2K−L) W(M+2K−L) D∗q W(M+2K−L) H
DH(M+2K−L) ,
r
r=0
(4.102)
in which the diagonal matrices DM and D(M+2K−L) do not depend on K and are, respectively,
defined as
M−1
−jπ m
DM diag e M(M+2K−L) , (4.103)
m=0
M+2K−L−1
−jπ m
D(M+2K−L) diag e M(M+2K−L) . (4.104)
m=0
In addition, based on Equations (4.72) and (4.75), the pair of matrices (P, Q ) ∈ CM×4 ×
C(M+2K−L)×4 are defined as
P p0 p1 p2 p3
= −V V P, (4.105)
ξ
Q q0 q1 q2 q3
M+2K−L−1
1
= diag V η∗ Zη∗ Q , (4.106)
1 − ξ ∗ (η∗m )M m=0
where P and Q are defined in Equations (4.94) and (4.95) for the MMSE solution, and in Equa-
tions (4.96) and (4.97) for the ZF solution, whereas ξ = 1 and η = e−j M .
π
Note that the choices of ξ and η were quite arbitrary. We have chosen ξ = 1, since, in the
case of multicarrier systems, we would like to cancel out the last IDFT matrix appearing in the
decomposition of matrix K described in Equation (4.102). Indeed, in the multicarrier systems, the
receiver matrix is G = WM K. If ξ = 1, one would not be able to cancel out the DFT matrix with
the last IDFT matrix presented in the decomposition of K, due to the presence of an additional
diagonal matrix. After fixing ξ = 1, we have chosen η in such a way that 1 − ξ ηM m = 1 − ηm = 0,
M
for all m within the set {0, 1, · · · , M + 2K − L − 1}, where η = 1/η. There are infinite possible
π
choices for η and we have arbitrarily chosen η = ej M . Note that, for this choice of η, one has
m = (η0 W(M+2K−L) )
ηM m M
−jπ −j2π mM
= e (M+2K−L) e (M+2K−L)
−jπ(2mM+1)
= e (M+2K−L)
= 1, (4.107)
where the definitions of vectors pr and qr depend on whether the ZF or the MMSE solution is
chosen. In any case, the resulting multicarrier structure is depicted in Figure 4.3.
Noise
Add Remove
Data Guard Phase
IDFT Channel Guard
Block Period IDFT
Period Shift
P/S S/P
One-Tap Phase
DFT One-Tap
Equalizer Shift DFT
Equalizer
Ignore
One-Tap Phase
DFT One-Tap
Equalizer Shift DFT
Data Equalizer
Block Ignore
Estimate
One-Tap Phase
DFT One-Tap
Equalizer Shift DFT
Equalizer
Ignore
One-Tap Phase
DFT One-Tap
Equalizer Shift DFT
Equalizer
Ignore
Figure 4.3 illustrates that, after removing the guard period, there are M + 2K − L phase shifts
to be performed in parallel, where the mth phase rotation is implemented through the multiplication
π
by ej M(M+2K−L) m . The first equalization step is performed after applying the IDFT to the data vector.
The resulting data vector is simultaneously processed by four different branches at the receiver. The
equalizers at this point consist of single-tap equalizers. The coefficients of the equalizers are the
entries of vectors q∗r , with r ∈ {0, 1, 2, 3}. After the DFT application, the last 2K − L elements
are discarded and new phase shifts are performed on the remaining M elements, where in this case
−j π m
the mth rotation consists of a multiplication by e M 2 . The final stage of the equalization consists
4.5. REDUCED-REDUNDANCY SYSTEMS 141
of applying DFTs in parallel to the output signal of the rotators, followed by another single-tap
equalization stage whose coefficients are the entries of vectors pr , with r ∈ {0, 1, 2, 3}.
−1
Also, a single-carrier system can be designed by setting F = IM and G = F K = K, yielding
3
∗
G = WM H
Dpr WM DM 0M×(2K−L) W(M+2K−L) Dq W(M+2K−L) H
DH
(M+2K−L) ,
r
r=0
(4.110)
in which, once again, the definitions of vectors pr and qr depend on whether the ZF or the MMSE
solution is chosen.
The superfast multicarrier and single-carrier transceivers of this chapter yield an additional
degree of freedom in the ZP-ZJ-based transmissions, since the amount of redundancy can vary from
the minimum value, L/2, to the most commonly used value, L.18 Nonetheless, one must deal with
two distinct DFT sizes, M and M + 2K − L. When M is a power of 2, then M + 2K − L is not
necessarily a power of two. Thus, a radix-2 FFT algorithm could only be applied to implement those
DFTs with size M. As for the DFTs with size M + 2K − L, one could implement the operations
using a radix-2 FFT of size 2M (which is assumed to be larger than M + 2K − L), along with zero-
padding of the related signals. Another possibility is to choose the amount of redundant elements
in such a way that M + 2K − L can be decomposed as a product of small prime numbers, leading
to fast implementations as well. This is an open research topic.
• ZP-OFDM-OLA: M log2 M − 2M + 4.
• ZP-OFDM-FAST: 5M
4 log2 M − 5M + 20.
• MC-RRBT: 15M
2 log2 M − 9M
2 + 20 + 5(2K − L).
In the MC-RRBT, it is possible to implement part of the receiver side using parallel processing
(see Figure 4.3). In this case, if we consider that the time that is required to perform a generic complex-
valued multiplication is T ∈ R+ s, then the MC-RRBT requires T (3M log2 M + 2(2K − L) + 8) s,
18 It is worth highlighting that Figure 4.3 is a way to implement general ZP-OFDM systems (K = L) which allows for superfast
equalization of received data blocks.
142 4. MEMORYLESS LTI TRANSCEIVERS WITH REDUCED REDUNDANCY
whereas the ZP-OFDM-OLA requires
T (M log2 M − 2M + 4) s, and the ZP-OFDM-FAST
requires T 4 log2 M − 5M + 20 s.
5M
It is worth noting that, in the minimum-redundancy case where K = L/2 (assuming a channel
with even order), the ZF solution can be further simplified. Indeed, as 2K − L = 0, then KZF =
−1
H , implying that Equation (4.97) can be rewritten as Q = [0M×2 KZF H Q̂ ]. As the product
PQ H defines completely the displacement matrix ∇Zξ ,Zη (KZF ), then we can also discard the first
two columns of P, since they will be multiplied by zero anyway. Thus, for ZF minimum-redundancy
systems, we can rewrite Equations (4.96) and (4.97) as follows:
Such a result could be alternatively obtained by using Equation (4.50). An important consequence
of this result is that all DFT matrices are M × M now and that the number of equalizer branches
at the receiver end changes from four to two branches.
We assumed that the pair of generator matrices (P, Q ) is known. In fact, these matrices
completely define the reduced-redundancy equalizers, since they are the only ones that contain
information about the channel. These matrices, however, must be previously computed in the so-
called receiver-design stage, which could be performed using up to O(M log22 M) operations, as
described, for example, in [51]. Besides, there are many applications in which the receiver-design
problem is not frequently solved. In wireline communications systems, the channel model is not
updated so often. In this case, the main computational burden refers to the equalization itself.
Reducing the amount of operations in the receiver-design stage is still an interesting and
rather important research direction in the context of superfast reduced-redundancy systems based
on the ZP-ZJ model.
4.5.2 EXAMPLES
Up to this point, we showed efficient structures for reduced-redundancy systems allowing the equal-
ization of received data blocks in a superfast manner. Nevertheless, we should note that ZP-ZJ
systems do not necessarily outperform OFDM-based transceivers. In fact, the authors in [54] have
proved mathematically that the MSE performance of reduced-redundancy transceivers degrades as
the number of transmitted redundant elements reduces. That is, transmissions using larger amounts
of redundancy leads to lower average MSE of symbols in ZP-ZJ systems.
If on one hand, we want to reduce the transmitted redundancy in order to save bandwidth;
on the other hand, we need to use as much redundancy as possible in order to have a good MSE
performance. The throughput is a good figure of merit to study the tradeoff between bandwidth
4.5. REDUCED-REDUNDANCY SYSTEMS 143
usage and error performance. Thus, let us analyze the throughput performance of superfast ZP-ZJ
systems through some simulation examples.
Example 4.4 (Minimum-Redundancy Transceivers) The figure of merit adopted here is the
throughput, defined as
M
Throughput = brc (1 − BLER)fs bps, (4.113)
M +K
in which b denotes the number of bits that is required to represent one constellation symbol, rc
denotes the code rate considering the protection of channel coding, K denotes the amount of
redundancy, fs denotes the sampling frequency, where symbol and channel models use the same
sampling frequency, and BLER stands for block-error rate, assuming that a data block is discarded
when at least one of its original bits is incorrectly decoded at the receiver end. In addition, the
definition of the SNR used throughout the simulations is the ratio between the mean energy of the
transmitted symbols at the input of the multipath channel and the power-spectral density of the
additive noise at the receiver front-end. Besides, we also consider that both synchronization and
channel estimation are perfectly performed at the receiver end.
In this example, we transmit 200 blocks, each one containing M = 32 BPSK data symbols
(without taking redundancy into account), and compute the throughput by using a Monte-Carlo
averaging process with 10,000 simulations. Consider these symbols are sampled at a frequency
fs = 1.0 GHz and that they are transmitted through a channel with a model operating at the same
frequency as the symbols and with a long impulse response of order L = 30.19 All the channel
taps
have the same variance, and the channel model is always normalized, that is, E h2 = 1. Both
2
the imaginary and real parts of the channel are independently drawn from a white and Gaussian
random sequence (random Rayleigh channel). For each simulation a new channel is generated.
Furthermore, since the ZP-ZJ transceivers use zeros as redundant elements, the adopted
OFDM and SC-FD systems in the simulations are the ZP-OFDM-OLA and ZP-SC-FD-
OLA [55], where ZP and OLA stand for zero-padding and overlap-and-add, respectively, (see
Subsections 2.4.2 and 2.4.4). Like the traditional cyclic-prefix-based systems, these ZP-based
transceivers also induce a circulant channel matrix. We chose these transceivers as benchmarks
since they are superfast transceivers that transmit L redundant zeros for each M data symbols. In
summary, from now on we shall consider that OFDM means ZP-OFDM-OLA and SC-FD means
ZP-SC-FD-OLA in all results here.
Figures 4.4 and 4.5 show the throughput curves for the OFDM, the SC-FD, the multi-
carrier minimum-redundancy block transceiver (MC-MRBT), and the single-carrier minimum-
redundancy block transceiver (SC-MRBT), using both ZF and MMSE designs. By observing these
figures it is possible to verify that the minimum-redundancy superfast transceivers outperform the
traditional OFDM and SC-FD transceivers in this particular setup, except for SNRs lower than
19 We consider that L = 30 corresponds to a long impulse response since L is very close to M in this case.
144 4. MEMORYLESS LTI TRANSCEIVERS WITH REDUCED REDUNDANCY
350
300
250
Throughput [Mbps]
200
150
100
ZF−OFDM
50 MMSE−OFDM
ZF−MC−MRBT
MMSE−MC−MRBT
0
0 5 10 15 20 25 30
SNR [dB]
Figure 4.4: Throughput [Mbps] as a function of SNR [dB] for random Rayleigh channels, considering
multicarrier transmissions.
12 dB in the ZF solutions. In this example, we use a convolutional code with constraint length 7,
rc = 1/2, and generators g0 = [133] (octal) and g1 = [165] (octal). This configuration is adapted
from the LTE (long-term evolution) specifications [94]. In addition, for the BLER computation,
we consider that a block (16 bits) is lost if, at least, one of its received bits is incorrect. We have
employed a MATLAB implementation of a hard-decision Viterbi decoder. Note that such favor-
able result stems from the choices for M and L, representing delay constrained applications in very
dispersive environment. These types of applications are suitable for the ZP-ZJ transceivers. In the
cases where M L, the traditional OFDM and SC-FD solutions are more adequate.
for which the throughput performance of the minimum-redundancy systems is poor. For this channel
(Channel A), we transmit 50,000 data blocks carrying M = 16 symbols of a 64-QAM constellation
(b = 6 bits per symbol). In fact, each data block stems from 48 data bits that, after channel coding,
yield 96 bits to be baseband modulated. The channel coding is the same as in Example 4.4 and we
assume that the sampling frequency is fs = 100 MHz.
Figure 4.6 depicts the obtained throughput results. We compare four different transceivers:
the ZP-OFDM-OLA and the three possible multicarrier reduced-redundancy block transceivers
(MC-RRBT).There are three possible MC-RRBT systems since the amount of redundant elements
respects the inequality L2 ≤ K ≤ L (i.e., K ∈ {2, 3, 4}). From Figure 4.6, one can observe that
the minimum-redundancy multicarrier system (MC-RRBT for K = 2) that employs an MMSE
equalizer is not able to produce a reliable estimate for the transmitted bits. However, if just one
350
300
250
Throughput [Mbps]
200
150
100
ZF−SC−FD
50 MMSE−SC−FD
ZF−SC−MRBT
MMSE−SC−MRBT
0
0 5 10 15 20 25 30
SNR [dB]
Figure 4.5: Throughput [Mbps] as a function of SNR [dB] for random Rayleigh channels, considering
single-carrier transmissions.
146 4. MEMORYLESS LTI TRANSCEIVERS WITH REDUCED REDUNDANCY
300
250
200
Throughput [Mbps]
MMSE−OFDM
MMSE−MC−RRBT (K = 2)
150
MMSE−MC−RRBT (K = 3)
MMSE−MC−RRBT (K = 4)
100
50
0
15 20 25 30 35
SNR [dB]
Figure 4.6: Throughput [Mbps] as a function of SNR [dB] for Channel A, considering multicarrier
transmissions.
additional redundant element is included in the transmission, the resulting MC-RRBT system (K =
3) is enough to outperform the MMSE-OFDM. One should bear in mind that such throughput
gains are attained without increasing substantially the computational complexity related to OFDM-
based systems. Moreover, the MC-RRBT system with K = 3 also outperforms the MC-RRBT
system with K = 4 in terms of throughput, especially for large SNR values, i.e., adding another
redundant element in the transmission (MC-RRBT for K = 4) does not contribute to improving
the throughput performance in this case.
Now, we will consider an FIR-channel model (Channel B) whose zeros are
0.999, −0.999, 0.7j, −0.7j, and −0.4j. This channel has zeros very close to the unit circle. We there-
fore expect that the performance of the traditional SC-FD system should be rather poor. Apart from
the channel model, all simulation parameters are the same of the previous experiment. Figure 4.7
depicts the throughput results. One can observe that the SC-RRBT systems always outperform the
traditional SC-FD system. Another important fact is that the throughput performance does not
necessarily improves as the number of transmitted redundant elements is increased. For example, for
low SNR values, it is better to use a reduced-redundancy system that transmits with a large number of
4.6. CONCLUDING REMARKS 147
200
180
160
140
Throughput [Mbps]
120
100
80
60
MMSE−SC−FD
MMSE−SC−RRBT (K = 3)
40 MMSE−SC−RRBT (K = 4)
MMSE−SC−RRBT (K = 5)
20
15 20 25 30 35
SNR [dB]
Figure 4.7: Throughput [Mbps] as a function of SNR [dB] for Channel B, considering single-carrier
transmissions.
redundant elements (K = 5), whereas for large SNR values, it is better to use a reduced-redundancy
system that transmits with a small number of redundant elements (K = 3). Once again, it is im-
portant to highlight that the superfast ZP-ZJ systems described in this chapter are just examples of
how to transmit with a small number of redundant elements while using superfast transforms and
single-tap equalizers.
CHAPTER 5
where Gq is an M × N complex-valued matrix, for each integer index q within the set Q
{0, · · · , Q − 1}. Once again, the entries of matrix Gq are not important in our discussions here, but
2 From now on, we will define the order of a matrix as the highest polynomial order among all of its entries. For instance, the order
of F(z) is T − 1.
152 5. FIR LTV TRANSCEIVERS WITH REDUCED REDUNDANCY
these entries are defined by G(z) which in turn is defined in Equation (2.43). It is worth mention-
ing that, if the transmitted redundant elements are not able to completely eliminate the interblock
interference (IBI), then matrix G(z) must have memory, i.e., it is mandatory to have Q > 1 in some
applications, as will be clearer soon.
Once again we will perform the same type of decomposition, but now for the FIR MIMO
channel matrix H(z) in Figure 2.9. Thus, we are interested in representing H(z) as
H(z) = Hb z−b , (5.3)
b∈B
where Hb is an N × N complex-valued Toeplitz matrix, for each integer index b within the set
B {0, 1, · · · , B}, where B stands for the order of the channel matrix H(z). The entries of Hb
can be determined by the entries of H(z), which in turn is a pseudo-circulant matrix defined by
Equations (2.29) and (2.30) of Chapter 2. Following these steps, it is possible to show that the
entries of Hb are related to the channel impulse response h(l) of order L by means of the following
equality:
Hb ml
h(bN + m − l), (5.4)
for any indexes m and l within the set N {0, 1, · · · , N − 1}. In addition, since we assumed that
the order of the causal channel is L, then h(bN + m − l) = 0 whenever bN + m − l is larger than
L or smaller than 0.
Example 5.1 (Determining Hb from H(z)) Suppose the order L of the FIR channel impulse
response h(l) is such that N < L < 2N, with N being the length of the transmitted block. Thus,
we are assuming that the channel memory L is longer than N. By using Equations (2.29) and (2.30)
of Chapter 2, we will decompose H(z) according to Equation (5.3), as follows:
⎡ ⎤
h(0) 0 0 ··· ··· ··· 0
⎢ h(1) h(0) 0 ··· ···⎥ ··· 0
⎢ ⎥
⎢ h(2) h(1) h(0) 0 ··· ⎥ ··· 0
⎢ ⎥
⎢ .. .. .. .. .. ⎥ .. ..
⎢ . .⎥
H(z) = ⎢ . . . ⎥ . .
⎢ .. .. .. .. .. ⎥ ..
⎢h(N −3) . . . . .⎥ .
⎢ ⎥
⎢ .. ⎥
⎣h(N −2) h(N −3) · · · · · · h(0)
. 0 ⎦
h(N −1) h(N −2) h(N −3) · · · · · · · · · h(0)
H0 ∈CN×N
5.2. TIME-VARYING REDUCED-REDUNDANCY SYSTEMS WITH MEMORY 153
⎡ ⎤
h(N) h(N −1) · · · h(4) h(3) h(2) h(1)
⎢ . . ⎥
⎢h(N +1) h(N ) .. . . h(4) h(3) h(2) ⎥
⎢ ⎥
⎢ .. .. .. .. .. ⎥
⎢ . h(N + 1) . . . . h(3) ⎥
⎢ ⎥
⎢ .. .. ⎥
+ z−1 ⎢ . .
..
.
..
.
..
.
..
. h(4) ⎥
⎢ ⎥
⎢ .. .. .. .. .. .. .. ⎥
⎢ . . . . . . . ⎥
⎢ ⎥
⎢ .. .. . . . . ⎥
⎣ . . . . . . . . . . h(N −1)⎦
··· ··· ··· ··· · · · h(N + 1) h(N )
H1 ∈CN×N
⎡ ⎤
0 ··· 0 h(L) · · · h(N + 1)
⎢ .. .. ⎥
⎢0 0 ··· 0 . . ⎥
⎢ ⎥
⎢ .. .. .. .. .. ⎥
⎢. . . . . h(L) ⎥
⎢ ⎥
+ z−2 ⎢0 0 0 ··· 0 0 ⎥
⎢ ⎥
⎢0 0 0 ··· 0 0 ⎥
⎢ ⎥
⎢. .. .. .. .. .. ⎥
⎣ .. . . . . . ⎦
0 0 0 0 ··· 0
H2 ∈CN×N
with B = 2. In addition, note that the Toeplitz matrices Hb are such that H0 is a lower triangular
matrix, whereas HB = H2 is an upper triangular matrix.
Equation (5.4) together with Example 5.1 suggest that B, the order of H(z), can be related
to the order of the channel impulse response L and the length of the transmitted block N . Now, let
us analyze such relation.
When L < N , Equation (5.3) coincides with Equation (2.38) of Chapter 2, with H0 = HISI
and H1 = HIBI , thus we have B = 1. When L = N, the decomposition in Equation (5.3) still
requires only two matrices, i.e., we still have B = 1, but in this case H1 = HIBI includes a non-
zero diagonal entry h(L) = h(N ). Physically, this means that the entire transmitted block suffers
interference from the previous transmitted block. As L becomes larger than N, other previously
transmitted blocks start interfering with the current transmitted block and, therefore, additional
matrices Hb appear in the decomposition of H(z). Example 5.1 illustrates the case when N < L <
2N, which leads to B = 2. In addition, when 2N < L ≤ 3N we have B = 3. Hence, we can define
154 5. FIR LTV TRANSCEIVERS WITH REDUCED REDUNDANCY
B as
/ 0
L
B . (5.6)
N
The definition of B in expression (5.6) implies that BN ≥ L. This observation will be particularly
useful in Section 5.3.
As previously observed, when the length of the transmitted data block N is smaller than the
channel order L, the resulting interblock interference (IBI) is caused by more than one block that have
been previously transmitted. This case (B > 1) is representative of delay-constrained applications
whose related channel-impulse responses are very long, thus hindering the transmission of data
blocks with length N ≥ L (B = 1). Another observation is that, since the channel order is L and
we are considering causal impulse responses, then h(L) must be necessarily different from zero. In
addition, we will assume that the channel model has a non-zero first term in its impulse response,
i.e., we assume h(0) = 0, which means that the channel transfer function has no zeros at the origin.
With those previous definitions, we can now derive a matrix description for the LTI transceiver
of Figure 2.9. To do that, let us first assume that the Z -transform of a generic time-domain vector
signal x(n) is denoted as
x(z) Z {x(n)} , (5.7)
where x(n) has size Nx ∈ N and can be expressed as
x(n) [ x(nNx − Nx + 1) x(nNx − Nx + 2) · · · x(nNx ) ]T . (5.8)
Thus, in the time domain, considering a noiseless channel, i.e., v(n) = 0N ×1 for all time
index n ∈ Z, we have the following expressions for the transmitted, received, and estimated vectors,
respectively:
u(n) Z −1 {F(z)s(z)}
! "
= Fτ Z −1 z−τ s(z)
∈T
τ
= Fτ s(n − τ ), (5.9)
τ ∈T
−1
y(n) Z {H(z)u(z)}
= Hb Z −1 z−b u(z)
b∈ B
= Hb u(n − b), (5.10)
b∈B ! "
−1
ŝ(n) Z G(z)y(z)
! "
= Gq Z −1 z−q y(z)
q∈Q
= Gq y(n − q). (5.11)
q∈Q
5.2. TIME-VARYING REDUCED-REDUNDANCY SYSTEMS WITH MEMORY 155
We can write the input-output relationship of FIR LTI transceivers in a much more compact
form by conveniently stacking the variables which appear in Equations (5.9), (5.10), and (5.11).
Indeed, note that Equation (5.11) can be rewritten as follows:
⎡ ⎤
y (n − (Q − 1))
⎢ .. ⎥
ŝ(n) = GQ−1 · · · G0 ⎣ . ⎦
G ∈CM×QN
y(n)
y(n)∈CQN ×1
= G y(n). (5.12)
Equation (5.12) depends on vectors y (n − Q + 1) , · · · , y(n), which can be determined using
Equation (5.10) as follows:
⎡ ⎤⎡ ⎤
HB · · · H0 0 ··· 0 u (n − (B + Q − 1))
⎢ 0 HB · · · H0 · · · 0 ⎥ ⎢ .. ⎥
⎢ ⎥⎢ . ⎥
y(n) = ⎢ . . . . . ⎥ ⎢ ⎥
⎣ .. .
. . . . . . ⎦⎣
. u(n − 1) ⎦
0 0 · · · HB · · · H0 u(n)
H ∈CQN ×(B+Q)N u(n)∈C(B+Q)N ×1
= H u(n). (5.13)
Equation (5.13), on its turn, depends on u (n − B − Q + 1) , · · · , u(n), which can be com-
puted using Equation (5.9) as follows:
⎡ ⎤⎡ ⎤
FT −1 ··· F0 0 ··· 0 s (n − (B + Q + T − 2))
⎢ 0 FT −1 · · · F0 ··· 0 ⎥ ⎢ .. ⎥
⎢ ⎥⎢ . ⎥
u(n) = ⎢ . . . . . ⎥ ⎢ ⎥
⎣ .. .. .. . . .. ⎦ ⎣ s(n − 1) ⎦
0 0 · · · FT −1 · · · F0 s(n)
F ∈C(B+Q)N ×[B+Q+(T −1)]M s(n)∈C[B+Q+(T −1)]M×1
= F s(n). (5.14)
Using Equations (5.12), (5.13), and (5.14), we have the following overall input-output de-
scription of the block-based LTI transceiver of Figure 2.9:
ŝ(n) = G HF s(n), (5.15)
which is the time-domain version of equation
ŝ(z) = G(z)H(z)F(z)s(z) (5.16)
in the Z -domain. This transceiver is time-invariant since the global transmitter and receiver matrices,
F and G , respectively, are constant matrices that do not depend on the time index n. The above
matrix description is rather useful for the time-varying case, since we cannot apply the Z -domain
description of the TMUX of Chapter 2 for this case.
156 5. FIR LTV TRANSCEIVERS WITH REDUCED REDUNDANCY
5.2.2 FIR MIMO MATRICES OF LTV TRANSCEIVERS
As previously mentioned, time-variance brings about extra degrees of freedom that play the role of
introducing additional diversity to the system. In order to generalize the previous LTI model to the
LTV case, we could consider that each transmitter matrix Fτ , with τ ∈ T , is actually a time-varying
matrix Fτ (n), which depends on the time index n. This implies that matrices F0 (n), · · · , FT −1 (n)
are employed to generate the channel input u(n), whereas matrices F0 (n − 1), · · · , FT −1 (n − 1)
are employed to generate the channel input u(n − 1), and so forth. Similarly, we could consider that
the receiver matrix Gq , with q ∈ Q, is in fact a matrix Gq (n), which varies with time as well.
In summary, by using this simple generalization we can rewrite Equations (5.9), (5.10),
and (5.11) as follows:
u(n) = Fτ (n)s(n − τ ), (5.17)
∈T
τ
y(n) = Hb u(n − b), (5.18)
b∈ B
ŝ(n) = Gq (n)y(n − q). (5.19)
q∈Q
and due to the time dependency of the transceiver matrices,3 there is no Z -domain counterpart for
such equation.
The reader should observe that, due to the definition of s(n) in Equation (5.14), we have
s(n) = 0M×(B+Q+T −2)M IM s(n), (5.23)
The specific constraints that the transceiver matrices F (n) and G (n) must satisfy in order to
achieve the ZF solution expressed in (5.25) will be described in the following section.
in which R {·} denotes the column or range space of the argument.5 This means that, for all such
vectors em , Equation (5.25) holds if there always exists a QN × 1 vector β m such that
em = F H (n)H H β m , (5.27)
Example 5.2 (Range Space of Matrix H H ) Let us start with the same pseudo-circulant chan-
nel matrix H(z) and its related matrices H0 , H1 , and H2 of Example 5.1, where N < L < 2N ,
thus implying that B = 2. In this example, we will study the range space of matrix H H . From
expression (5.13), we can write
⎡ ⎤H
H2 H1 H0 0 ··· 0
⎢0 H2 H1 H0 ··· 0 ⎥
⎢ ⎥
HH ⎢ . .. .. .. .. ⎥
⎣ .. . . . . ⎦
0 0 · · · H2 H1 H0
⎡ ⎤
HH2 0 ··· 0
⎢HH HH ··· 0 ⎥
⎢ 1 2 ⎥
⎢ H . .. ⎥
⎢H H1 H . . . ⎥
⎢ 0 ⎥
=⎢ .. ⎥, (5.29)
⎢ 0 H0 H . H2 ⎥
H
⎢ ⎥
⎢ . .. .. ⎥
⎣ .. . . H1 ⎦
H
0 0 · · · HH 0
5.3. CONDITIONS FOR ACHIEVING ZF SOLUTIONS 159
with HH
2 ∈ CN×N being a lower triangular Toeplitz matrix given by
⎡ ⎤
0 ··· 0 0 ··· 0
⎢ .. .. .. .. .. ⎥
⎢ . . . . . 0⎥
⎢ ⎥
⎢ .. .⎥
⎢ ··· . .. ⎥
⎢ ∗0 0 0 ⎥
HH2 ⎢ ⎢ h (L) 0 0 · ·· 0 0⎥ ⎥
. (5.30)
⎢ h∗ (L − 1) ∗
h (L) 0 ··· 0 0⎥ ⎥
⎢
⎢ . . . .. .. .. ⎥
⎣ .. .. .. . . .⎦
h∗ (N + 1) h∗ (N + 2) · · · h∗ (L) · · · 0
any vector formed by linear combination of the columns of H H will have zeros in its first BN − L
entries, as shown in Equation (5.28).
Taking Equation (5.28) into account and by defining α m H H β m , we have that the linear
system of Equation (5.27) can be rewritten as
em = F H (n)α
αm
0(BN −L)×1
= F (n)
H
αm
α
αm
= F H (n)α
αm , (5.31)
From linear algebra we know that the dimension of the null space of F H (n) is
K N −M (5.35)
α m = (n)α̂
α + γ̂γ m , (5.36)
α ∈ C[(Q+B)K−(T −1)M]×1 is a vector which parameterizes the general solution and introduces
where α̂
the additional degrees of freedom that allow α m to satisfy the requirement of being orthogonal to
all vectors within the null space of H (see Equation (5.31)). Note that
F H (n)α
α m = F H (n) α + F H (n)γ̂γ m
(n) α̂
=0 =em
= 0 × α̂
α + em
= em . (5.37)
Now, consider the restriction of having α m ∈ R H H , which is equivalent to find α m such
that z H α m = 0 for all vector z within the subspace ker H . But, instead of considering any vector
5.3. CONDITIONS FOR ACHIEVING ZF SOLUTIONS 161
z ∈ ker H , we can verify if α m is orthogonal to all vectors from a basis of ker H . Indeed,
if the dimension of the kernel ofmatrix H is D ∈ N, then there are D linear independent vectors
z0 , z1 , · · · , zD−1 which span ker H and which respect the identity H zd = 0QN ×1 , for all d within
the set D {0, 1, · · · , D − 1}. Since any vector z ∈ H can be written as
then
Hence,if zdH α m = 0 for all d within the set D, then z H α m = 0. Thus, our task is to find a basis
for ker H . It turns out that such a basis is associated with the zeros of the scalar channel transfer
function, as formulated below.
Given the FIR channel model, whose transfer function is
with h(0) = 0 = h(L), consider that the zeros of H (z) are all distinct, i.e., there are L distinct scalars
z0 , · · · , zL−1 ∈ C such that H (z0 ) = · · · = H (zL−1 ) = 0. Since h(0) = 0 = h(L) and due to the
Toeplitz structure of matrix H described in Equation (5.13), it turns out that this channel matrix is
full-row rank, i.e., rank of H is QN. As H is generated from H by discarding the first BN − L zero
columns, then the rank of H is QN as well. This means that the dimension of the null space of H is
D = QN + L − QN
= L. (5.41)
for each l ∈ {0, · · · , L − 1}, is a linearly independent set of vectors within the kernel of H , since
H zl = 0QN ×1 , (5.43)
for all l ∈ {0, · · · , L − 1}. Indeed, the ith row of H , with i ∈ {0, 1, · · · , QN − 1}, is given by
T
hi 0 · · · 0 h(L) · · · h(0) 0 · · · 0 ∈ C1×(QN +L) , (5.44)
162 5. FIR LTV TRANSCEIVERS WITH REDUCED REDUNDANCY
where there are i zero entries at the beginning of this vector and QN − i − 1 zero entries at its end.
Therefore, we have
T
hi zl = h(L)zli + h(L − 1)zli+1 + · · · + h(0)zlL+i
−(L−1)
= zlL+i h(L)zl−L + h(L − 1)zl + · · · + h(0)
= zlL+i H (zl )
=0
= 0, (5.45)
ZH α m = 0L×1 , (5.46)
Hence, we can substitute the general solution described in Equation (5.36) into the constraint
in Equation (5.46), thus giving
α + ZH γ̂γ m = 0L×1 ,
ZH (n)α̂ (5.48)
yielding
α = −ZH γ̂γ m .
ZH (n) α̂ (5.49)
H
H −1
α = − ZH (n)
α̂ ZH (n) ZH (n) ZH γ̂γ m , (5.50)
H
which is achievable only when matrix ZH (n) ZH (n) is non-singular. This fact has two
consequences:
The reader should observe that the equalizer-length bound in expression (5.52) depends on
some known variables, namely L, M, and T , and on the amount of redundancy K which is to be
defined. Given L,
! M, and T" , one might argue if it is always possible to satisfy inequality (5.52) and the
constraint rank ZH (n) = L (see discussions after expression (5.50)), regardless of the amount of
transmitted redundant elements K. The next subsection shows a particular example of time-varying
transceiver with memory that is able to achieve the lower bound on K in expression (5.53) and
discusses further design constraints to achieve such lower bound.
164 5. FIR LTV TRANSCEIVERS WITH REDUCED REDUNDANCY
5.3.4 ACHIEVING THE LOWER BOUND OF REDUNDANCY
Let us address this topic by first considering for the time being that the transmitter matrices are
memoryless (T = 1), since having more columns in matrix F (n) could only add more diversity
(more redundancy) to the system, so that the memoryless transmitter case is somehow the worst
case scenario for finding a lower bound on K which is actually required to achieve the ZF solution.
Indeed, it is reasonable to expect that the minimum achievable bound8 for K would also be sufficient
when we have memory at the transmitter end.
The remaining of this section will focus on proving that the lower bound on the number of
transmitted redundant elements in expression (5.53) can be reached and we encourage the reader
to keep this target in mind while examining the forthcoming lengthy mathematical computations.
Another important observation is that we are not interested in the performance of the described
transceiver, rather we focus only on the parameter bounds.
In the memoryless transmitter case, we consider that the time-varying transmitter matrix in
its Hermitian transpose version is given by
⎡ ⎤
F0H (n − Q − B + 1) 0 ··· 0
⎢ .. ⎥
⎢ 0 . ··· 0 ⎥
F H (n) ⎢
⎢ .. .. ..
⎥ ∈ C(Q+B)M×(Q+B)N ,
⎥ (5.54)
⎣ . . F0H (n − 1) . ⎦
0 0 ··· F0H (n)
where we assume that F H (n) has full row rank, which means that the rank of F H (n) is (Q + B)M.
In addition, assume that, for each time index n, we have
IM F(n) ∈ CN ×M ,
F0 (n) = (5.55)
0K×(BN −L) (n)
where F(n) is an M × M matrix with full rank, whereas (n) is a given K × M matrix, with
M M − (BN − L) ≥ 0.9 Such structure for the transmitter matrices means that the redundant
elements are included at the end of each block to be transmitted and that these redundant elements
are linear combinations of the last M non-redundant elements.
Now, matrix F H (n) can be generated by discarding the first BN − L columns of F H (n).
Based on expressions (5.54) and (5.55), this means that we are removing BN − L linearly indepen-
dent columns of F H (n), thus reducing the rank from (Q + B)M to (Q + B)M − (BN − L) =
(Q + B − 1)M + M − (BN − L) = (Q + B − 1)M + M. This coincides with the proposed di-
mension for F H (n) described in the first paragraph of Subsection 5.3.2 (for T = 1). Therefore, we
8The term “achievable” was employed here since we do not know a priori if, for any channel model and for any choices of M and
T , we could use K = 1 and still satisfy the constraint: rank ZH (n) = L.
9This assumption (M ≥ 0) is not a very restrictive constraint, since if B = 1, then BN − L ≤ BN − K = N − K = M (assuming
L ≥ K), whereas if B > 1, then N can be chosen so that BN is close to L, implying that it is easy to satisfy BN − L ≤ M.
5.3. CONDITIONS FOR ACHIEVING ZF SOLUTIONS 165
have the following (Q + B)M × (QN + L) matrix:
⎡ H ⎤
F (n − Q − B + 1) 0 ··· 0
⎢ .. ⎥
⎢ 0 . ··· 0 ⎥
F (n) ⎢
H ⎢ ⎥
.. .. .. ⎥
⎣ . . F (n − 1)
H . ⎦
0 0 ··· H
F (n)
⎡ ⎤
0(BN −L)×M 0(BN −L)×K
⎢ 0 ⎥
⎢ IM H (n − Q − B + 1) ⎥
×⎢ 0 ⎥.
⎣ 0(BN −L)×K ⎦
0 diag IM
H (n − i) i=Q+B−2
(5.56)
From linear algebra, we know that the kernel of matrix F H (n) has dimension QN + L −
[(Q + B)M − (BN − L)] = (Q + B)K. Let (n) be a (QN + L) × (Q + B)K block-diagonal
matrix defined as
⎡ ⎤
(n − Q − B + 1) 0 ··· 0
⎢ .. .. .. .. ⎥
⎢ . . . . ⎥
(n) ⎢ ⎥, (5.57)
⎣ 0 · · · (n − 1) 0 ⎦
0 0 ··· (n)
where
⎡ ⎤
0(BN −L)×K
H (n − i) ⎦ ∈ CN ×K ,
(n − i) ⎣ − (5.58)
IK
α m = (n)α̂
α + γ̂γ m , (5.60)
which is a concrete example of Equation (5.36), where γ̂γ m is a particular solution of the linear system
F H (n)α α m = em . Observe that, in order to determine α m , it is only necessary to determine α̂
α , since
γ̂γ m and (n) are assumed to be already known, since they depend exclusively on matrix F (n). Let
us also simplify the forthcoming notation by omitting the index m of the related variables. Thus,
α = (n)α̂ α + γ̂γ must be a solution to the linear system e = F H (n)α α.
166 5. FIR LTV TRANSCEIVERS WITH REDUCED REDUNDANCY
As the solution to such linear system must also satisfy ZH α = 0L×1
, where Hthe L × (QN +
H
L) matrix Z is defined in Equation (5.47), then we have Z (n) α̂ H α = −Z γ̂γ . The former
equation can be rewritten in a much more convenient way by first defining vectors z̃l , for each
l ∈ {0, · · · , L − 1}, as
T
z̃lT 1 zl zl2 · · · zlN −1 , (5.61)
⎡ H ⎤
z̃0 (n − Q − B + 1) ··· (z0∗ )(Q+B−2)N z̃0H (n − 1) (z0∗ )(Q+B−1)N z̃0H (n)
⎢ z̃ H (n − Q − B + 1) ··· (z1∗ )(Q+B−2)N z̃1H (n − 1) (z1∗ )(Q+B−1)N z̃1H (n) ⎥
⎢ 1 ⎥
ZH (n)=⎢
⎢ .. .. .. ⎥
⎥
⎣ . . ··· . ⎦
H (n − Q − B + 1)
z̃L−1 ··· ∗
(zL−1 H (n − 1)
)(Q+B−2)N z̃L−1 ∗
(zL−1 ) (Q+B−1)N H
z̃L−1 (n)
(5.62)
0(BN −L)×K
(n − Q − B + 1) . (5.63)
(n − Q − B + 1)
/ 0
L
K≥ . (5.64)
Q+B
Consider that the global transmitter matrix was designed in such a way that there exist
N
δ ≤ Q + B matrices, let us say (n − δ + 1), · · · , (n − 1), (n) ∈ CN ×K , such that
⎧ ⎫
⎪
⎨ ⎪
⎬
rank (n − δ + 1) · · · (n − 1) (n) = δK, (5.66)
⎪
⎩ ⎪
⎭
N ×δK
the matrix product which defines Z H (n), and we already know that (Q + B)K ≥ L ≥ μ.
Therefore, using
<μ=
K≥ , (5.69)
δ
with
@ A
N
δ ≤ min Q + B, (5.70)
K
168 5. FIR LTV TRANSCEIVERS WITH REDUCED REDUNDANCY
guarantees that Z H (n) is full row rank, which indicates that it would be possible to satisfy the
proves that the lower bound K = 1 on the number of transmitted redundant elements is achievable.
Transmiter Receiver
Optimization Optimization
then the overall MSE of symbols is given by (see also Equation (5.23))
in which Rss (n) = E{ss (n)ss H (n)} and Rvv (n) = E{vv (n)vv H (n)}. In addition, we have also assumed
that s (n) and v (n) are uncorrelated, i.e., Rsv = E{ss (n)vv H (n)} = E{ss (n)}E{vv (n)}H = 0.
In summary, it assumed that the FIR MIMO channel transfer matrices are known and that
all the input signals to the system are uncorrelated with each other. In addition, we usually consider
that the second order statistics of all the input signals are known. The transceiver is represented by
FIR MIMO transfer matrices with given orders.
Our problem is to minimize MSE, usually subject to power constraints. The procedure to
find a solution is to derive analytical expressions for the MSE criterion and power constraint as
for example presented in [31]. For that we find the Karush-Kuhn-Tucker (KKT) conditions for
optimality and implement an iterative numerical algorithm based on those necessary conditions.
The basic strategy is to design a transmitter assuming a known initial receiver and then design
a receiver assuming the transmitter is the one previously designed, as Figure 5.1 illustrates. The
procedure can be repeated until no further reduction in the MSE is achieved.
5.5 EXAMPLES
This section presents three examples that illustrate some of the concepts developed in this chapter.
We will start with Example 5.4 in order to show a particular situation in which an FIR MIMO
transceiver outperforms a CP-OFDM system, considering the bit-error rate (BER) as figure of merit.
Then, Example 5.5 illustrates the BER performance of ZF- and MMSE-based systems satisfying or
not the zero-forcing conditions described in Section 5.3. After that, we explain how the theoretical
zero-forcing bounds derived here relate to the ZF detection of signals in long-code CDMA systems
in Example 5.6.
Example 5.4 (OFDM & FIR MIMO Transceivers) This example assesses the BER versus
signal-to-noise ratio (SNR) performance evaluated through a Monte Carlo simulation, where M = 8
5.5. EXAMPLES 171
BPSK symbols are transmitted per block through a channel whose transfer function is
thus implying that L = 3. The number of redundant elements is always K = L = 3, which means
that N = 11. The uncorrelated additive channel noise samples are complex, circularly symmetric,
and Gaussian distributed. Two distinct systems are considered in this example: the CP-OFDM and
the FIR MIMO transceiver proposed in [31], which optimizes the transmitter and receiver matrices
through an iterative MMSE-based approach, such as the one briefly described in Section 5.4.
Figure 5.2 depicts the obtained results, in which it is clear the striking advantages of using FIR
MIMO systems over OFDM-based systems in this particular setup. Of course, we are not taking
into account implementation issues, such as computational complexity, modularity, just to mention
a few. Indeed, a great advantage of OFDM is the low complexity implementation costs required to
implement the fast Fourier and inverse fast Fourier transforms.
0
10
−1
10
−2
10
BER
−3
10
−4
10
OFDM
FIR MIMO Transceiver
−5
10
0 2 4 6 8 10 12 14 16 18 20
SNR [dB]
10 0
ZF
MMSE
ZF (K = 5)
MMSE (K = 5)
ZF (Q = 1)
MMSE (Q = 1)
BER
10 −1
10 −2
−5 0 5 10 15 20 25 30 35 40
SNR [dB]
Example 5.5 (BER Performance & ZF Conditions) Consider the transmission of data blocks
containing N = 30 entries, including M symbols of a 16-QAM digital constellation as well as K =
30 − M redundant elements. Assume that the channel-transfer function has the following zeros:
2π 2π 3π 3π 4π 4π
1.8ej 5 , 1.8e−j 5 , 1.8ej 5 , 1.8e−j 5 , 1.8ej 5 , 1.8e−j 5 , 1.1ej 8 , 1.1e−j 8 , 0.9ej 8 , 0.9e−j 8 , ej 10 ,
7π 7π 7π 7π 5π
5π
e−j 10 , 0.8, which means that the channel order is L = 13, whereas the number of congruous zeros
with respect to N is μ = 6, corresponding to the first six zeros with magnitude 1.8.
In a time-invariant transmission, one must have K ≥ μ = 6 in order to achieve the ZF
solution (see Subsection 5.3.5). This means < =that M must be smaller than or equal to 24. In addition,
based on Equation (5.6), one has B = 30 13
= 1, while based on Equation (5.52), and considering
< =
that T = 1 and K = 6, one has Q ≥ 13 6 − 1 = 2. Figure 5.3 depicts some results obtained using
time-invariant ZF- and MMSE-based equalizers. The “ZF” and “MMSE” curves employ K = 6
redundant elements and Q = 2 as the order of the related receiver matrices. The “ZF (K = 5)” and
“MMSE (K = 5)” curves employ K = 5 and Q = 2, which means that the number of redundant
elements is violating the ZF constraint described in Section 5.3. The “ZF (Q = 1)” and “MMSE
5.5. EXAMPLES 173
(Q = 1)” curves employ K = 6 and Q = 1, which means that the orders of the related receiver
matrices are not satisfying the ZF constraint. The obtained results show that, by not following the
ZF conditions, either by choosing K = 5 or Q = 1, a floor in the BER curves appears, starting from
20 dB of SNR. On the other hand, for the designs following the ZF conditions, the BER tends to
zero as the SNR increases.
This example also illustrates that when alternative design criteria are used, such as the min-
imization of the MSE, the ZF conditions described in Section 5.3 are still useful in order to avoid
performance loss due to errors in the reconstruction of the signal. This eventually means that, even
for very high SNRs, it is still possible to have a “BER floor” due to the nonexistence of the ZF so-
lution. On the other hand, this “BER floor” does not appear by observing the conditions presented
in this chapter.
Example 5.6 (Long-Code CDMA Systems) As previously illustrated in Figure 1.21 of Chap-
ter 1, MIMO models encompass many block-transmission configurations, ranging from single-user
point-to-point communications employing multiple antennas, to multiple-access schemes in mul-
tiuser systems. This example illustrates an application of the theoretical results related to FIR LTV
transceivers with reduced redundancy within the framework of CDMA systems.
Indeed, CDMA systems can be described using the concept of MIMO transceivers. By using
such a description it is possible to derive some theoretical conditions for designing equalizers that
guarantee the perfect reconstruction of the transmitted signal at the receiver end. For example, we
can apply the theoretical analysis in order to obtain multiuser detection in CDMA systems with
long codes, i.e., codes which last for more than one symbol duration, as described in the following
reasoning.
Consider the TMUX structure of Figure 2.6 of Chapter 2. Assume that sm (n) is the symbol
associated with the mth user at the time instant n. It is possible to imagine that the impulse re-
sponses of the synthesis filters fm (k), with m ∈ M, that appear in that figure can play the role of a
spreading sequence of a CDMA system, whereas the impulse responses of analysis filters gm (k), with
m ∈ M, can be thought as the de-spreading codes, all of them associated with the mth user. If, in
addition, we assume that the synthesis and analysis filters are actually time-varying filters, then we
can consider that each subfilter implements a piece of a given CDMA spreading/de-spreading long
code associated with a user. In other words, CDMA with long codes can be interpreted as CDMA
with time-varying short codes.
As a result, it is possible to verify that CDMA with long codes can be represented by a
time-varying structure, i.e., it is an example of FIR LTV system. Each signal block faces a time-
varying code, which is implemented using time-varying transmit filters.Thus, by adapting the results
concerning the amount of redundancy required to allow a ZF solution, it is possible to establish the
conditions for the existence of zero-forcing multiuser detectors.
In this context, N represents the spreading factor, M denotes the number of codes that are
going to be used, whereas K represents the number of unused codes. The main conclusion of this
174 5. FIR LTV TRANSCEIVERS WITH REDUCED REDUNDANCY
analysis is that ZF equalization is always possible in CDMA systems using long codes as long as the
system is not at full capacity, i.e., one must have at least one unused code (K ≥ 1). In addition, it can
be shown that both the complexity of the receiver and its performance depend directly on the number
of unused codes. The conditions derived serve as useful guidelines for the design of communications
systems, allowing the tradeoff between performance and complexity of the receiver.
In conclusion, the existence of ZF equalizers is guaranteed if the amount of redundancy is
greater than or equal to the number of congruous zeros or if there are enough different transmission
filters at the transmitter. For practical channels, approximately congruous zeros may cause numerical
instability in equalizer design. A CDMA system with long codes may be interpreted as a transmul-
tiplexer with memoryless time-varying filters. In the uplink direction, the system can be modeled
as a transmultiplexer with time-varying filters with memory. In any case, for the CDMA system,
the block length is determined by the spreading factor, implying that the redundancy is equal to the
number of unused codes.
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Authors’ Biographies
PAULO S. R. DINIZ
Paulo S. R. Diniz was born in Niterói, Brazil. He received the
Electronics Eng. degree (Cum Laude) from the Federal Uni-
versity of Rio de Janeiro (UFRJ) in 1978, a M.Sc. degree from
COPPE/UFRJ in 1981, and a Ph.D. from Concordia University,
Montreal, P.Q., Canada, in 1984, all in Electrical Engineering.
Since 1979 he has been with the Department of Electronic
Engineering (undergraduate) at UFRJ. He has also been with the
Program of Electrical Engineering (the graduate studies dept.),
COPPE/UFRJ, since 1984, where he is presently a Professor. He
served as Undergraduate Course Coordinator and Chairman of the Graduate Department. He has
received the Rio de Janeiro State Scientist award from the Governor of Rio de Janeiro.
From January 1991 to July 1992, he was a visiting Research Associate in the Department of
Electrical and Computer Engineering of University of Victoria, Victoria, B.C., Canada. He also
held a Docent position at Helsinki University of Technology. From January 2002 to June 2002, he
was a Melchor Chair Professor in the Department of Electrical Engineering of University of Notre
Dame, Notre Dame, IN, USA. His teaching and research interests are in analog and digital signal
processing, adaptive signal processing, digital communications, wireless communications, multirate
systems, stochastic processes, and electronic circuits. He has published several refereed papers in
some of these areas and wrote the books Adaptive Filtering: Algorithms and Practical Implementation,
4th ed., Springer, NY, 2012, and Digital Signal Processing: System Analysis and Design, 2nd ed.,
Cambridge University Press, Cambridge, UK, 2010 (with E.A.B. da Silva and S.L. Netto).
He has served as the Technical Program Chair of the 1995 MWSCAS held in Rio de Janeiro,
Brazil. He was the General co-Chair of the IEEE ISCAS2011, and Technical Program co-Chair of
the IEEE SPAWC2008. He has been on the technical committee of several international conferences
including ISCAS, ICECS, EUSIPCO, and MWSCAS. He has served as Vice President for region
9 of the IEEE Circuits and Systems Society and as Chairman of the DSP technical committee of
the same Society. He is also a Fellow of IEEE. He has served as associate editor for the following
Journals: IEEE Transactions on Circuits and Systems II: Analog and Digital Signal Processing from
1996–1999, IEEE Transactions on Signal Processing from 1999–2002, and the Circuits, Systems and
Signal Processing Journal from 1998–2002. He was a distinguished lecturer of the IEEE Circuits and
Systems Society from 2000–2001. In 2004, he served as distinguished lecturer of the IEEE Signal
184 AUTHORS’ BIOGRAPHIES
Processing Society and received the 2004 Education Award of the IEEE Circuits and Systems
Society. He also holds some best-paper awards from conferences and from an IEEE journal.
WALLACE A. MARTINS
Wallace A. Martins was born in Brazil in 1983. He received
an Electronics Engineering degree (Cum Laude) from the Fed-
eral University of Rio de Janeiro (UFRJ) in 2007, and M.Sc. and
D.Sc. degrees in Electrical Engineering from COPPE/UFRJ in
2009 and 2011, respectively. He worked as a technical consul-
tant for Nokia Institute of Technology (INDT), Brazil, and for
TechKnowledge Training, Brazil. In 2008, he was a research vis-
itor at the Department of Electrical Engineering, University of
Notre Dame, Notre Dame, IN. Since 2010 he has been with the
Department of Control and Automation Industrial Engineering, Federal Center for Technological
Education Celso Suckow da Fonseca (CEFET/RJ – UnED-NI), where he is presently a Lecturer
of Engineering. His research interests are in the fields of digital communication, microphone ar-
ray signal processing, and adaptive signal processing. Dr. Martins received the Best Student Paper
Award from EURASIP at EUSIPCO-2009, Glasgow, Scotland.
MARKUS V. S. LIMA
Markus V. S. Lima was born in Rio de Janeiro, Brazil in 1984.
He received an Electronics Engineering degree from the Federal
University of Rio de Janeiro (UFRJ) in 2008, an M.Sc. degree in
Electrical Engineering from COPPE/UFRJ in 2009, and is cur-
rently pursuing his D.Sc. degree at COPPE/UFRJ. He has served
as a teaching assistant for the following undergraduate courses
taught at UFRJ: Digital Transmission, Digital Signal Process-
ing, and Linear Systems. His main interests are in adaptive signal
processing, microphone array signal processing, digital commu-
nications, wireless communications, statistical signal processing,
and linear algebra.