Ali Hazmi
Studies in Digital TV Signal Processing: Impulse Noise Mitigation,
Repeater Loop Interference Cancellation, and DVBT Transmission
in CATV Networks
Tampere 2007
Tampereen teknillinen yliopisto. Julkaisu 668
Tampere University of Technology. Publication 668
Ali Hazmi
Abstract
The Digital Video Broadcasting Project (DVB) is a multinational initiative to standardize digital
broadcasting worldwide. It produces different system specifications including satellite: DVBS,
cable: DVBC, terrestrial: DVBT, and others. The DVBT system for terrestrial broadcast
ing is probably the most complex DVB delivery system. It is proven a worldwide success. It
has become the de facto world standard for transmitting digital terrestrial television. Origi
nally, the DVBT standard was created for fixed and portable reception as the main application
areas. However, to remain successful, continual assessment and enhancement are needed to mit
igate any perceived deficiencies in the performance of the standardized techniques. In addition,
competition with other Terrestrial Digital TV standards (ATSC8VSB and DiBEGISDBT), and
aims to target new potential services based on handheld battery powered devices such as mo
bile telephones, PDAs, etc., have motivated further improvements of the standard. This thesis
addresses enhancements of the DVBT system. Three main issues are considered.
In the first part, we investigate the tolerance of the DVBT system to impulsive noise. Impulse
interferences can be produced by ignition sparks from vehicles or various household appliances.
DVBT has been shown to have reception difficulties in the presence of impulse noise, mainly
when using higher constellations for high data rates. A new method for compensating the ef
fects of impulse noise in the OFDM based DVBT systems is described. The scheme uses
channel estimation pilots for the estimation and cancellation of impulse noise. The results show
that the system performance can be improved significantly using the introduced method. In the
timefrequency selective fading channel, further measures are needed. We present an enhanced
channel estimation scheme for the developed impulse noise cancellation method to overcome its
sensitivity to the Doppler spread and fading impairments. Additionally, a simple and practical
impulse burst position detection method is presented. Combinations of the introduced methods
with other existing techniques are studied. Effects of impulse interferences on DVBT receiver
synchronization are also discussed.
In the second part, we introduce loop interference cancellation algorithms to cope with cou
pling problems in gapfillers. In the terrestrial digital video broadcasting system, the gapfillers
are used to insure sufficient coverage to shadowed users. However, there are a number of limi
tations to be overcome in order to maintain a good quality of service. One of the most serious
limitations is the loop interference, due to coupling between the transmitter and receiver anten
nas at the relay stations. Existing techniques for loop interference cancellation are reviewed and
new measures are introduced. Firstly, a channel estimation based frequencydomain adaptive
cancellation algorithm is described and its limitations and performance are simulated. Possible
improvements of the algorithm are presented. Secondly, an autocorrelation based cancellation
ii
technique is developed. Extensions of the method to deal with a specific multipath loop inter
ference with exponential profile are analyzed. Finally, an adaptive LMS based loop interference
cancellation algorithm is investigated.
In the last part, interoperability issue of the DVBT standard is considered. We study the
quality of the terrestrial digital video broadcasting (DVBT) transmission over the cable TV net
work. Usually, when terrestrial digital TV signals are distributed in the cable TV network, a
conversion is needed in the headend from DVBT to DVBC. We study the possibility of using
the DVBT signal without any conversion. We demonstrate the sensitivity of the OFDM system
for the phase noise effects by using a dynamic model for CATV channel. Then, we conclude by
giving the specifications which a CATV network should satisfy to allow DVBT transmission
with sufficient quality.
Preface
The research work for this thesis has been carried out during the years 20012007 at the Institute
of Communications Engineering of Tampere University of Technology, Tampere, Finland. It
was funded by the RTT Oy, which is a national organization that contributes to the research
and development of new radio and television technologies in Finland. It was also funded by the
Celtic WingTV project.
I wish to express my sincere and deep gratitude to my supervisor Professor Markku Renfors
for his invaluable guidance, continuous support, and infinite tolerance during the course of this
work and throughout my studies.
I would like to thank Dr. Slimane Ben Slimane, Associate Professor at KTH in Stockholm,
Sweeden, and Dr. Jussi Vesma Senior Research Engineer, at Nokia Technology Platforms,
Turku, Finland, for reviewing my thesis, and for their constructive feedback and comments on
the manuscript.
I owe special thanks to my colleague and coauthor Lic. Tech. Jukka Rinne for numerous
hours of working together and for fruitful technical discussions.
I wish to thank also my coauthor Dr. Jukka Henriksson at Nokia Research Center, Helsinki,
Finland, for his invaluable information as well as for his kind advises, discussions and assistance.
I am indebted to all my present and former colleagues and friends at the Institute of Commu
nications Engineering for the pleasant work environment and for the help I have received during
my work. I would like to thank Dr. Ridha Hamila, Dr. Abdelmonaem Lakhzouri, M.Sc. Tuomo
Kuusisto, M.Sc. Mikka Tupala, Dr. ElenaSimona Lohan, Dr. Mikko Valkama, Msc. Yang
Yuan, and Msc. Tero Ihalainen, for their cooperation, suggestions, and for the fruitful techni
cal discussions. In particular special thanks are due to Tarja Eralaukko, Sari Kinnari and Elina
Orava.
I wish to express my gratitude to all my friends in Finland for their support and care. Im
very obliged to Mohamed Maala, family Gabbouj, family Hammouda, to Faouzi Alaya Cheick,
Fehmi Chebil, Mejdi Trimeche, Yacine Madene, Khaled Zbaida, Ahmad Iftikhar, and to the
small Tunisian community in Tampere.
Finally, I wish to express my deepest gratitude to my parents, to my brothers and sisters for
their love and encouragement.
ALI HAZMI
Tampere, June 1, 2007
iii
Contents
Preface iii
List of Acronyms ix
1 Introduction 1
1.1 History of digital terrestrial broadcasting . . . . . . . . . . . . . . . . . . . . 1
1.2 Organization of the thesis . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2
1.3 Authors contribution . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
9 Conclusions 101
9.1 Future Research . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102
References 105
Appendix A 111
List of Figures
2.4 Typical delay profile for a channel consisting of the natural and the artificial
delay spread due to SFN. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
3.7 Inner interleaving (bit and symbol levels) and mapping of bits onto modulation
symbols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
4.5 BER vs. Blanking window length in AWGN channel: SNR=20 dB (16QAM). 32
ix
x LIST OF FIGURES
6.2 A model for the broadcastwave relay SFN used in our basic study. . . . . . . 57
6.7 BER performance of the frequency domain adaptive algorithm for singletap
loop with LAR of 1dB. AWGN channel is assumed between the relay and the
receiver. 64QAM and a code rate of 2/3 are considered. . . . . . . . . . . . 62
6.10 BER for adaptive LMS loop interference cancellation with spectrum whitening
method: Ricean channel case and different noise bandwidths and power levels.
Singletap loop, LAR = 1 dB, SNR=25 dB, 64QAM and a code rate of 2/3 are
considered. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
6.11 BER for adaptive LMS loop interference cancellation with spectrum whitening
method: Rayleigh channel case and different noise bandwidths and power
levels. Singletap loop, LAR = 1 dB, SNR=25 dB, 64QAM and a code rate of
2/3 are considered. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
7.5 BER in the Ricean channel case with LAR of 1dB using the basic
autocorrelation method, 64QAM , and a code rate of 2/3 . . . . . . . . . . . . 70
7.6 BER in the Rayleigh channel case with LAR of 1dB using the basic
autocorrelation method, 64QAM, and a code rate of 2/3. . . . . . . . . . . . 71
7.12 BER performance of the DVBT repeater with diversity: DVBT, 2k mode,
16QAM, 2/3 code rate, variable LAR and Doppler frequency cases. . . . . . . 78
7.13 BER performance of the enhanced frequency domain adaptive algorithm for
singletap loop with LAR of 1dB. Rayleigh channel is assumed between the
main transmitter and the relay. 64QAM and a code rate of 2/3 are considered. 79
7.14 The absolute value of the error function e(n): different LMS loop interference
cancellation schemes, one tap loop interference model with LAR of 2dB,
64QAM, and a code rate of 2/3. . . . . . . . . . . . . . . . . . . . . . . . . 80
7.15 BER in adaptive LMS loop interference cancellation case: one tap loop
interference model with LAR of 1dB, 64QAM, and a code rate of 2/3. . . . . 81
7.16 BER in adaptive LMS loop interference cancellation case: multitap loop
interference model with LAR of 2dB, 64QAM, and a code rate of 2/3. . . . . 81
8.1 Cable TV system headend. . . . . . . . . . . . . . . . . . . . . . . . . . . 85
8.2 Cable TV system distribution plant. . . . . . . . . . . . . . . . . . . . . . . 85
8.3 Gaussian noise channel in OFDM based system. . . . . . . . . . . . . . . . . 86
8.4 OFDM with nonlinearity. . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
8.13 Sensitivity of DVBT to the phase noise with different levels of scaling
parameters: QPSK, 2/3 Code rate. . . . . . . . . . . . . . . . . . . . . . . . 97
8.14 Sensitivity of DVBT to the phase noise with different levels of scaling
parameters: 16QAM, 2/3 Code rate. . . . . . . . . . . . . . . . . . . . . . 97
8.15 Sensitivity of DVBT to the phase noise with different levels of scaling
parameters: 64QAM, 2/3 Code rate. . . . . . . . . . . . . . . . . . . . . . 98
ACRONYMS
xiii
xiv List of Acronyms
MUX Multiplexer
MPEG Moving Picture Expert Group
MSE Mean Square Error
NLOS NonLine Of Sight
NTSC National Television Standards Committee
PRBS Pseudo Random Binary Sequence
OFDM Orthogonal Frequency Division Multiplexing
PAL Phase Alternating Line
PC Personnel Computer
PDA Personnel Digital Assistant
PDP Power Delay Profile
PES Packetised Elementary Stream
PM Phase Modulation
QAM Quadrature Amplitude Modulation
QEF Quasi Error Free
QPSK Quadrature Phase Shift Keying
RF Radio Frequency
RS Reed Solomon
SECAM Squentiel Couleur A Mmoire
SFN Single Frequency Network
SPS Six Pilot Spacing
SNR SignaltoNoise Ratio
TDMA Time Division Multiple Access
TPS Twelve Pilot Spacing
TS Transport Stream
UHF Ultra High Frequency
VCO VoltageControlled Oscillator
VHF Very High Frequency
VLSI Very Large Scale Integration
VRGC Variable Relay Gain Control
WLAN Wireless Local Area Network
WiMAX Worldwide Interoperability for Microwave Access
Chapter 1
Introduction
The mass communications media of television is one of the most significant technical accom
plishments of the 20th century. The ability of persons across the world to see each other, to
communicate with each other, and experience each others cultures and ideas is a monumental
development. However, the technology that we enjoy today required many decades to mature.
The recent development in TV broadcasting has been characterized by the increasing role that
digital signal transmission has played. Combinations of several driving forces have led to the
digitization of the television image. The key driving forces include the revolutionary advances
in microelectronic integrations and data compression techniques, in addition to the motivation
of introducing highdefinition television (HDTV).
The history of digital television transmission began in the mid 1970s when the European
analog PAL and SECAM systems were supplemented by the teletext system. Improvement
and modification of sound transmission began in 1980 with the introduction of the dualsound
carrier method. Further development of picture transmission began in the mid 1980s with the
Japanese MUSE method and continued in Europe with the so called MAC series. In 1993, Digi
tal television started officially in its specification phase in Europe. In the framework of the DVB
Project, a markedled initiative to standardize digital broadcasting worldwide, three different
systems have already been specified: satellite system DVBS [1], cable system DVBC [2], and
terrestrial system DVBT [3].
Being the most complex DVB delivery system, the DVBT standard is well established now,
and has already proved its capability to support very different types of networks, since it of
fers a high degree of flexibility. Digital terrestrial television is currently replacing the analog
television distribution in many countries.
Soon after a stable draft of the standard was available, suggestions to enhance the quality of
services delivered using DVBT stepped into the scene. The DVBT standard was originally de
veloped for stationary reception with a directional rooftop antenna or a nondirectional antenna
on a portable receiver. In 1998, a research project named MOTIVATE was started, as part of the
research program of the European Commission. MOTIVATE concentrates on the exploitation
of the features of the existing DVBT standard for mobile reception.
DVBT was shown to be perfectly suited for reception by mobile receivers even at high
drivingspeeds. However, new emerging services scenarios are appearing these days. These
1
2 INTRODUCTION
services use small handheld devices that may include integrated mobile phones and provide
reception inside buildings, and cars, etc. Hence, new challenges facing the DVBT standard
become more and more frequent. Questions of whether there is a need to change anything in
DVBT concerning the emerging new service scenarios were asked. In the end of the year 2000
a small group of interested parties started to study and try to answer this question. The con
clusion was that DVBT has excellent performance in many aspects and the vast majority of
the commercial requirements are directly fulfilled as such. However some enhancements of the
DVBT standard are needed. They have been partly implemented in the DVBH system [4].
The work for this thesis was carried out while participating in projects contributing to the
DVBT standard revision in order to fully support the new service scenarios.
The thesis considers three main issues. The first issue, treated mainly in Chapters 4 and 5,
is the tolerance of the DVBT receiver to impulsive interference. The second topic is the loop
interference due to coupling between transmitter and receiver antennas at the relay stations in
single frequency network. This issue is discussed in Chapter 6 and 7. Finally in Chapter 8 we
study the quality of DVBT transmission over the cable TV network.
The general conclusions of the thesis are drawn in Chapter 9.
The thesis consists of nine chapters. The first two chapters after the introduction discuss the ba
sics of the orthogonal frequency division multiplexing (OFDM) and the DVBT system. Specif
ically, Chapter 2 provides an overview on the OFDM technique, presenting its principles and
outlining the generic OFDM system model. Chapter 3 gives an overview of the DVBT trans
mission system, the simulation environment and the channel models used in the performance
evaluations.
The main results of the thesis are presented in Chapters 48.
In Chapter 4, we start by presenting practical impulse noise models. Sources of the impul
sive noise are described. Some general statistical models are investigated and then the effects
of the impulse noise on the DVBT signal are studied. Secondly, the existing techniques for im
pulsive noise cancellation are presented. In the third section of the Chapter 4, we introduce a
new algorithm [P1] that uses pilots in the DVBT signal to mitigate the effects of the impulsive
noise. The performance analysis of the new algorithm is presented. Finally, an example system
is studied and simulated.
In Chapter 5, we present new techniques [P2] for enhancing the pilot based impulse noise
canceller introduced in Chapter 4. Additionally, the performance resulting from combining this
algorithm with other existing impulse noise mitigation schemes is investigated [P3][P4]. An en
hanced burst position estimation algorithm is also developed. Finally the effects of the impulsive
noise on the time synchronization in DVBT systems are addressed [P5].
In Chapter 6, we introduce the problem of the loop interference due to coupling between
transmitter and receiver antennas at the relay stations. Then, we give a review of the main
existing algorithms dealing with loop interference cancellation in the DVBT system.
In Chapter 7, we investigate some extensions of these algorithms to improve their function
alities. New algorithms for loop interference cancellation are also presented [P6][P7][P8].
In Chapter 8, interoperability issue of the DVBT system is addressed. In particular, the
compatibility of DVBT and DVBC is investigated. We study the quality of DVBT transmis
sion over the cable TV network. Generally, when terrestrial digital TV signals are distributed
in the cable TV, a conversion is needed in the headend, from DVBT to DVBC (single carrier
AUTHORS CONTRIBUTION 3
system). In order to avoid this costly conversion and to transmit DVBT signal directly in the
existing cable channel, various requirements have to be satisfied. It is known that phase noise
represents one of the main limitations for OFDM based systems. In this chapter we demonstrate
the sensitivity of the OFDM system to the phase noise effects by using a dynamic model for
CATV channel [P9]. We conclude by giving the specifications which a CATV network should
satisfy to allow DVBT transmission with sufficient quality [P10][P11].
In Chapter 9, we draw the conclusions and the main results of the thesis.
The thesis is a monograph where the main material is based on the following publications that
are referred to as [P1],[P2],..,[P11].
[P1] J. Rinne, J. Henriksson, and A. Hazmi, "Impulse noise canceller for OFDM system uti
lizing pilots," in Proc. 7th International OFDMWorkshop, Hamburg, Germany, pp.
183187, Sept. 2002.
[P2] J. Rinne and A. Hazmi, "Impulse burst position detection and channel estimation schemes
for OFDM systems," in IEEE International Conference on Consumer Electronics and
IEEE transactions on Consumer Electronics, Vol. 49, Issue 3, pp. 539545, June 2003.
[P3] A. Hazmi, J. Rinne, M. Renfors,"An enhanced impulse burst cancellation method using
pilots and soft bits in OFDM based systems," in the Fifth IEEE Signal Processing Work
shop on Signal Processing Advances in Wireless Communications SPAWC04, Lisbon,
Portugal, pp.373376, July, 2004.
[P4] A. Hazmi, J. Rinne, and M. Renfors, "Combination techniques for impulse burst noise mit
igation in static and mobile channels for OFDM based systems," in Proc. 9th International
OFDMWorkshop, Dresden, Germany, pp. 6266, Sept. 2004.
[P5] A. Hazmi, J. Rinne, and M. Renfors, "Performance evaluation of symbol synchronization
in OFDM Systems over impulsive noisy channels," in Proc. IEEE Vehicular Tech. Conf.,
Milan, Italy, Vol. 3, pp. 17821786, May 2004.
[P6] A. Hazmi, J. Rinne, and M. Renfors, "A new algorithm for loop interference cancellation
in SFN for digital terrestrial broadcasting," in Proc. 7th International OFDMWorkshop,
Hamburg, Germany, Sep. 2002.
[P7] A. Hazmi, J. Rinne, and M. Renfors, "Cancellation of loop interference with exponen
tial profile using autocorrelation method in OFDM based systems," in Proc. Ninth IEEE
International Conference on Communications Systems, Singapore, pp. 140144 Sept.
2004.
[P8] A. Hazmi, J. Rinne, and M. Renfors, "Diversity based DVBT indoor repeater in slowly
mobile loop interference environment," in Proc. 10th International OFDMWorkshop,
Hamburg, Germany, Sep. 2005.
[P9] A. Hazmi, J. Rinne, M. Renfors, "DVBT signal in cable TV network: Advantages and
limitations," in Proc. of ICC2001, Helsinki, Finland, Vol. 7, pp. 22812285, June 1114,
2001.
[P10] A. Hazmi, J. Rinne, M. Renfors," DVBT signal over cable TV network and phase noise
requirements," in IEICE Trans. on Fundamentals of Electronics, Communications and
Computer Sciences, Vol. E84A, NO. 4, pp. 966970, April 2001.
4 INTRODUCTION
[P11] A. Hazmi, J. Rinne, M. Renfors," A simple cable TV network channel modeling and sim
ulation," in Proc. of the IASTED International Conference Modeling, Identification, and
Control, MIC2001, Innsbruck, Austria, pp. 1922, Feb. 2001.
The research work reported in this thesis was carried out at the Institute of Communications
Engineering (formerly Telecommunications Laboratory), Tampere University of Technology.
The author is an active member of a research group lead by Lic. Tech. Jukka Rinne and Prof.
Markku Renfors. Many of the ideas that have been discussed in this thesis have originated in
informal discussion within the group. The authors contribution to all the publications used in
this thesis has been essential. Beside [P1] where the main idea was due to the second author
and partly [P2] where the MSE analysis was done by the first author, the ideas and techniques
in all the remaining papers [P2P11] were primary originated and analyzed by the author of
this thesis. Naturally the coauthors contributed to the final appearance of each paper. All the
simulations and writing of these papers were done primarily by the author of the thesis.
As we consider three main issues related to DVBT system, the main results of the thesis can
be also classified into three parts that can be summarized as follows:
In [P1] a new method for compensating the effects of impulse noise in OFDM sys
tems is described. The scheme uses channel estimation pilots for the estimation and
cancellation of impulse noise. The results show that the system performance can
be improved significantly mainly in AWGN environment. The main idea has been
suggested by the second author.
In [P2], P[3] and [P4] we present new algorithms to enhance the performance of the
method presented in [P1] in timefrequency selective channels. In [P2] an enhanced
channel estimation scheme to be used in conjunction with the method of [P1] is in
troduced. The scheme uses the impulse free pilots to estimate the corrupted ones.
Three different alternatives are analyzed. Their performance in mean squared er
ror (MSE ) sense versus Doppler frequency are studied analytically. Bit error rate
(BER) simulation results are given, in the cases of SFN hilly terrain static and mobile
channels. Additionally, in [P2] we show results for a simple and practical impulse
burst position detection method. Our mitigation algorithms are based on estimat
ing the subcarrierwise deviations caused by the presence of interference samples.
In [P3] we consider the effects of using quantized values of the subcarrier deviation
estimates to weight the soft bits in a DVBT receiver. Hence increased reliability
against the impulse burst is achieved. The performance of this approach is compared
to conventional receiver structures in different fading environments. Static Ricean
and Rayleigh channel cases are used to evaluate the system performance in bit er
ror rate (BER) sense. Also a mobile channel case with moderate Doppler spread is
considered. In [P4] a novel combination technique for impulse noise mitigation in
OFDM based systems is described. The scheme uses a cascade of a convolutional
interleaver, adaptive impulse burst detector and the pilot based impulse canceller of
[P1]. The results show that the system tolerability to impulse noise can be improved
significantly.
In [P5] we study the effects of impulsive noise on the performance of the maximum
likelihood symbol synchronization (MLSS) algorithm in OFDM systems. Limita
tions of the MLSS algorithm are shown. Simulation results for the MSE of the timing
AUTHORS CONTRIBUTION 5
offset in static and mobile channels are provided. Ideas to enhance the performance
of the algorithm in these environments are also proposed.
Loop interference cancellation
In [P6] we present a new method to cancel the loop interference in digital terrestrial
broadcasting via single relay system in single frequency network (SFN) configura
tion. The distortion is caused by the coupling between the transmitting and receiving
antennas. It is shown that it is possible to use the method with different kinds of
channel profiles that can occur between the host station and the relay station. Ricean
and Rayleigh cases are studied. The performance of the method is evaluated in the
bit error rate sense.
In [P7] we discuss the use of the autocorrelation method introduced in [P6] to cancel
multitap loop interference with exponential profile in digital terrestrial broadcast
ing using relays. The method presented in [P6] can cancel efficiently the coupling
between the transmitter and the receiver antennas in the relay station when the one
tap loop model is considered. In [P7] we modify the autocorrelation method to deal
with multitap loop with exponential profile. Analytical and experimental results
are presented to verify the advantages and the limitations of the method in such an
environment. Bit error rate simulation results are also considered in AWGN and
Rayleigh static channels cases.
In [P8] we investigate the use of antenna diversity in DVBT indoor repeaters. We
use diversity to reduce the loop interference that may occur because of transmitter
receiver antenna coupling in the repeater. The case of slowly mobile loop interfer
ence environment is studied.
DVBT transmission in cable TV networks
In [P9] we study the quality of DVBT signal transmission over the cable TV net
work. The possibility of using the DVBT signal in cable TV network without any
conversion is investigated. A simple model for the channel is developed taking into
account the main impairments. Secondly, the performance of the DVBT signal is
simulated and compared to that of digital cable TV standard DVBC that is using
single carrier transmission techniques. The comparison is based on the bit error ratio
that each system can achieve in the same channel environment. Channel interference
problems are also simulated.
In order to avoid the costly conversion and to transmit DVBT signal directly in the
existing cable channel, many requirements have to be satisfied. It is known that
phase noise represents the main limitation for OFDM based systems. In [P10] and
[P11] we demonstrate the sensitivity of the OFDM system to the phase noise effects
by using a dynamic model for CATV channel. Then, we conclude by giving the
specifications which a CATV network should satisfy to allow DVBT transmission
with sufficient quality.
Chapter 2
OFDM: Orthogonal Frequency
Division Multiplexing
2.1 INTRODUCTION
After more than thirty years of research and development, orthogonal frequency division mul
tiplexing (OFDM, or COFDM for coded OFDM) has been widely applied in highspeed digital
communications [5] [6] [7]. Due to the recent advances of digital signal processing and very
largescale integrated circuit technologies, the initial obstacles of OFDM implementation, such
as massively complex computation and need for highspeed memory no longer exist. OFDM has
already been implemented in digital audio broadcasting and is applied to terrestrial digital tele
vision and HDTV broadcasting. Various projects and prototypes of OFDM systems are widely
used these days, including DVBT (digital video broadcasting for digital terrestrial television)
by the European Broadcasting Union, and WLAN, WiMAX, etc.
OFDM is a form of modulation that is particularly well suited to the needs of the terres
trial broadcasting channel [8]. OFDM can cope with high levels of multipath propagation, with
a wide spread of delays between the received multipath signals [9]. The special performance
of OFDM with respect to multipath and interference effects is only achieved through a careful
choice of the system parameters.
In the case of conventional data transmission systems, the frequency spectrum of each data sym
bol is allowed to occupy the entire available bandwidth [10]. The parallel data transmission
system offers possibilities for alleviating many of the problems encountered with the conven
tional serial systems [11]. A parallel system is one in which several sequential streams of data
are transmitted simultaneously, so that at any instant many data elements are being transmitted.
In such a system, the spectrum of an individual data element normally occupies only a small
part of the available bandwidth.
When dividing an entire channel bandwidth into many narrow subbands, the frequency re
sponse over each individual subband is relatively flat. Since each subchannel covers only a
7
8 OFDM: ORTHOGONAL FREQUENCY DIVISION MULTIPLEXING
e j2f0t
e j2f1t
X0(n)
X1(n)
Xk (n)
e j2fkt
Xk (n)
e j2f N1t
XN1(n)
small fraction of the original bandwidth, equalization is potentially simpler than in serial systems.
A simple equalization algorithm can minimize meansquared distortion on each subchannel,
and the implementation of differential encoding may make it possible to avoid equalization
altogether.
where Xk (n) are the QAM symbols transmitted on the k th subcarrier in the nth block.
In Figure 2.1, the principle of an OFDM transmitter is shown. N symbols are collected in the
serial to parallel converter and then transmitted over N complex carriers. After multiplexing,
the summation signal in the baseband can be exactly interpreted as an inverse Npoint discrete
Fourier transform (IDFT). Thus the system complexity, e.g., the use of oscillators and subfilters,
can be greatly reduced.
On the receiver side, the received signal y(t) is demodulated, sampled at the block rate 1/Tu ,
and passed to a discrete Fourier transform (DFT) operator which converts the signal back to the
frequency domain.
OFDM SIGNAL 9
/.!0
123.04 Ts
!
, .
g
"#$%&#' (&)*+$
Fig. 2.2 The concept of adding a cyclic prefix in case of a multipath channel.
ZTu
1
Yk (n) = y(t)ej2fk t dt. (2.4)
Tu
0
Due to recent advances of digital signal processing (DSP) and integrated circuit technologies,
a cost effective implementation of an OFDM system is possible thanks to the available efficient
IFFT/FFT algorithms.
same geographical area, we do not require the transmitters to use different frequencies, as in
the analog systems. Instead, we could form a Single Frequency Network (SFN) as in Figure
2.3, where all transmitters use identical signals, occupying exactly the same frequency block.
Such a procedure will create severe artificial multipath propagation conditions. However, us
ing cyclic prefix, receivers will be able to receive signals on the same frequency from different
transmitters as long as the delay between the first and last signal to arrive falls within the cyclic
prefix duration. Consequently, signals from transmitters whose signals are delayed relative to
the signals from a closer transmitter are treated as "artificial" multipath (Figure 2.4). Therefore,
this will result in substantial improvements in frequency economy and improved coverage for
mobile reception due to diversity gains. In addition, few low power transmitters can be used as
opposed to having one very high power transmitter. Overall, the power required using the SFN
concept is lower for transmitting signals to a given area.
Transmitter 2
Transmitter 1
Receiver
Time
T1 T2
Fig. 2.4 Typical delay profile for a channel consisting of the natural and the artificial delay spread due
to SFN.
Chapter 3
Digital Video Broadcasting:
Technical Overview
3.1 INTRODUCTION
This chapter gives an overview of the digital video broadcasting technology. We will detail the
basic parameters of the DVBT system and emphasize its distinguishing features with respect
to presentday broadcasting services. The principles and implementations of the DVBT trans
mitter and receiver with OFDM modulator and demodulator will be explained. Advantages and
limitations of the DVBT technology will be described. This chapter will present essential back
ground information for the following chapters, where enhancements to the discussed techniques
will be addressed in detail.
Studies on digital systems for television broadcasting have been carried out since the late 1980s.
A large group of interested partners started the Digital Video Broadcasting project. Within this
project, a set of specifications was developed for the delivery of digital television over satellites,
cable and through terrestrial transmitters.
As illustrated in Figure 3.1, the basic principles of terrestrial digital television transmission
can be divided into two main parts. The first part includes video and audio encoding (source
encoding) and the MPEG2 transport stream. This part is the same for all transmission media
(terrestrial, cable, satellite). The second main part deals with terrestrial transmission, specifi
cally, with channel coding and OFDM modulation. In this chapter we will focus mainly on the
second part, i.e., on channel coding and modulation. A brief introduction of the main blocks of
the first part will be given.
11
12 DIGITAL VIDEO BROADCASTING: TECHNICAL OVERVIEW
Programme MUX
Video Coder
Transport MUX
Audio Coder 1
Mux Adaptation
Outer Coder Outer Interleaver Inner Coder
Energy Dispersal
Mux Adaptation
Outer Coder Outer Interleaver Inner Coder
Energy Dispersal
To Aerial
During the first steps in the DVB development, a fundamental decision was taken. It consisted
of the selection of the MPEG2 standard to be the basic platform for source coding for both
audio and video in DVB. The description of the MPEG2 systems is documented in the follow
ing international standards: [ISO 13818  1] , [ISO 13818 2] and [ISO 13818 3]. However
these are generic and too wide to be applied to DVB directly. The document [ETA 154]] was
created by the DVB project to provide guidelines and specific choices of the MPEG2 system
parameters for use in the DVB applications.
DVBT standard essentially defines a way of transmitting MPEG2 source coded data (i.e.,
video, audio, or pure data) terrestrially. It includes channel coding, which is protecting the
system against disturbances occurring in the transmission channel, and frame adaptation.
14 DIGITAL VIDEO BROADCASTING: TECHNICAL OVERVIEW
Then, a scrambling process is carried out to ensure the adequate random bit transitions,
which are needed for synchronization purposes at the receiver side. This scrambling is based on
a Pseudo Random Binary Sequence (PRBS) generator using the polynomial 1 + X 14 + X 15 .
3.4.1.2 Outer coding Once the multiplex adaptation and randomization for energy disper
sal has been carried out, the outer coding and outer interleaving are the next steps to be taken. As
the outer coding scheme, the system has a ReedSolomon code with a length of 204 bytes, and
a dimension of 188 bytes (1 sync byte + 187 data bytes), which is usually referred as RS(204,
188, t=8). The last part t=8, means that this error correction code is able to correct up to 8 erro
neous bytes in arbitrary positions within a received block of 204 bytes. This shortened code is
derived from the RS(255, 239, t=8) code by inserting 51 null bytes before the data block. These
bytes set to zero must be discarded after the RS coding operation, giving the final block length
of 204 bytes (Figure 3.3).
3.4.1.3 Outer interleaving The outer interleaving process is done as convolutional byte
wise interleaving, with depth I=12. As shown in Figure 3.4, the 12 branches are depicted with
their corresponding FIFO (First In First Out) shift register. Every single shift register is 17 bytes
long. It is interesting to note that the SYNC byte is always going through the branch number
0. The corresponding deinterleaving process is exactly the same, but the indices are reversed,
in order to recover the original order of the packets.
3.4.1.4 Inner coding In the DVBT system, a group of punctured convolutional codes is
defined. These are derived from the mother code which has a code rate of CR = 1/2 with 64
states. The other specified code rates are 2/3, 3/4, 5/6, and 7/8. This feature allows the ability
to select the error correction capability of the system depending on the concrete circumstances
of the actual transmission. These codes derived from the mother code with CR=1/2 through
puncturing. The mother code has different generator polynomials, defined as G1 = 171OCT and
G2 = 133OCT for the outputs 1 and 2, as it can be seen in Figure 3.5.
DVBT: PHYSICAL LAYER 15
0 0
17 x 11
1 M=17 1 17 x 3
1 byte 1 byte
per 2 per 2
17 x 2 17 x 2
position position
3 3
17 x 3 M=17
11 11
17 x 11
Fig. 3.4 Functional block diagrams of the outer interleaver and deinterleaver
3.4.1.5 Inner interleaving In inner interleaving, two steps are considered. First, there is
a bitwise interleaving, and it is followed by a symbol interleaving. In the first one, the input
bits are demultiplexed into 2, 4, or 6 bit substreams, for QPSK, 16QAM and 64QAM modu
lation schemes, respectively. The resultant substreams are interleaved in different ways, so that
the bits forming a QAM subsymbol are mixed. This is important for the performance of the sys
tem, because it ensures the same treatment for every bit of the subsymbol. The size of the bit
interleaving block is 126 bits. The structure of the inner coding and inner interleaving blocks is
shown in the Figure 3.6.
In the symbol interleaving block the bits are interleaved as QAM subsymbols, depending on
the transmission mode. They can be QPSK, 16QAM, or 64QAM subsymbols (2, 4, or 6 bits),
and once these subsymbols are formed they are placed into blocks of 1512 or 6048 subsym
bols for 2k or 8k transmission modes respectively. These 1512 or 6048 subsymbols forming an
Output 1
Output 2
Inner Coder
X
Puncturing
Convolutional
Y with Inner Interleaver
Encoder
serial output
OFDM symbol are then interleaved. The symbol interleaving is also called frequency interleav
ing. This allow the system to be more robust under multipath fading channel conditions. Figure
3.7 shows the structure of this inner interleaving process, and its link to the mapping operation.
3.4.1.6 Mapping The active subcarriers are modulated depending on the transmission
mode. The possibilities are QPSK, 16 QAM, and 64QAM. In Figure 3.8, the basic constella
tions for uniform QPSK, 16QAM, and 64QAM are shown. In genera1, the more complex the
constellation, the more power is needed to reach the same BER. The mapping is done following
the Gray coding structure, in which different neighbor symbols only have one bit difference in
the whole symbol word. It is known that the Gray coding results in a lower bit error rate than
other possible mappings, for a given transmission. In the DVBT system, as possible transmis
sion mode, nonuniform mapped constellations are also supported in the cases of 16QAM and
64QAM.
TBC`
ABC DECFGHFIJFG K
ABC DECFGHFIJFG L
BEabC TBC`
ABC DECFGHFIJFG M
\F]^_
QRSTUH
<=>?@ DECFGHFIJFG VWXXYZ[
ABC DECFGHFIJFG N D S]^_
ABC DECFGHFIJFG O
ABC DECFGHFIJFG P
Fig. 3.7 Inner interleaving (bit and symbol levels) and mapping of bits onto modulation symbols
Im(z)
Im(z)
Im(z)
QPSK
16QAM
64QAM
As it can be observed from the table, the bandwidth of the transmission channel is the same
in both modes, 7.61M Hz approximately.
estimation technique. The location of the scattered pilots follows the pattern described in the
Figure 3.9, which a1so illustrates the structure of the frame.
cdijgklhmnocp
cdefgh cdijgqrkqnrcp
Fig. 3.9 Location of the scattered pilots in the OFDM frame structure.
Figure 3.9 illustrates how the scattered pilots are located. Every 12th subcarrier in the fre
quency direction, and every fourth in the time direction is a scattered pilot. Nevertheless, in
the frequency direction the distance between two scattered carriers belonging to consecutive
symbols is just 3. These are very important facts to be taken into account for the channel esti
mation. The location of the continua1 and transmission parameter signalling pilots is defined in
the system specifications [3]. An OFDM symbol has 68 transmission parameter signaling pilots
subcarriers in the 8K mode, and 17 subcarriers in the 2K mode. Every transmission parameter
signaling pilot subcarrier of a given symbol conveys the same differentially encoded bit. The
transmission parameter signaling pilot information is defined in 68 consecutive OFDM symbols
(frame). The modulation of these transmission parameter signalling pilots is simple. Again, the
imaginary part is zero, and the rea1 part is modulated in DBPSK (Differentia1 Binary Phase
Shift Keying). The power level of the transmission parameter signaling pilots is not boosted as
for scattered and continua1 pilots.
A simulation model is needed to assess the performance of the introduced algorithms. Therefore
we need to describe the simulation environment. We will focus mainly on the channel models
that have been used in the performance evaluation.
block starts to produce output data as soon as it is possible, and therefore all the blocks can be
operating simultaneously, which reduces the simulation time.
The CCSS DVBT simulation model assumes ideal time synchronization, and it does not con
sider any RF impairments like carrier frequency offset or I/Qimbalance. Otherwise the model
includes the functions of a practical DVBT receiver. The basic functionality of the simulation
model has been verified by comparisons with other project partners.
3.5.2.1 Static channels The static channel profiles are based on the DVBT specifica
tions [3]. They have been generated from the following equations where x(t) and y(t) are input
and output signals respectively:
N
i eji x(t i )
P
0 x(t) +
i=1
y(t) = s , (3.1)
N
2i
P
i=0
where 0 represents the line of sight ray. The number of echoes N is equals to 20. i is the
phase shift from scattering of the ith path, i is the attenuation of the ith path and i is the
relative delay of the ith path. The phases, delays and attenuations are listed in Table A.1 in the
Appendix A.
The Ricean factor K (the ratio of the power of the direct path (the line of sight ray) to the
reflected paths) is
20
K= s (3.2)
N
2i
P
i=1
In the simulations a Ricean factor K = 10 dB has been used. This is refereed in the specifi
cations as fixed reception. When there is no line of sight, i.e., when 0 = 0 we have a Rayleigh
profile modeling a portable reception scenario. In the following chapters, whenever static Ricean
or Rayleigh channels are used, we mean fixed and portable reception respectively. To simulate
the case of high frequency selective channel, we considered static SFN that were constructed
by combining two static Rayleigh channels. The delay between the two paths was selected to
be variable parameter ranging between Tu /32 to Tu /8. The power of the second path is set to
0 dB.
3.5.2.2 Mobile channels Modeling mobile channels is more complicated since the path
delay, gain, arrival angle, and the receiver speed are changing. Mobile channels are usually
modeled by timevariant, i.e., dynamic FIR filters.
To evaluate the performance of the introduced algorithms in mobile environment we consid
ered mainly the typical urban channel model (TU6). We used both cases of mildly and heavily
frequency selective channels. The delay profile of the TU6 channel model follows the COST
20 DIGITAL VIDEO BROADCASTING: TECHNICAL OVERVIEW
207 specifications [17]. The Doppler spectra are selected to be classical. The description of
TU6 channel taps are listed in Table A.2.
The SFN channel models were created by combining two TU6 channels. The delay between
the two impulse responses was selected to be variable parameter ranging between Tu /32 to
Tu /8. The power of the second path is set to 0 dB.
3.6.3 Gapfiller
If gaps exist in a service area, as may be encountered in deep valleys, tunnels, or inside houses,
the multipath capability of DVBT enables these gaps to be filled in a very efficient way [3].
The principle is as follows: outside the gap or the uncovered subarea, the DVBT signal is
picked up by a directional antenna. After filtering and amplification, the signal is retransmitted
(at the same frequency) into the uncovered area by a socalled gapfiller (also called as repeater
or relay). This is usually refereed as a direct or a nonregenerative repeater. Sometimes it is
also possible to decode the received signal in the repeater and encode it again before retrans
mission. This is performed in the regenerative repeaters. The choice between regenerative and
nongenerative gap fillers depends on many factors. In the scope of DVBT, a nonregenerative
BASIC ASPECTS OF DVBT NETWORKS 21
gap fillers are usually preferred because of the limitation for the processing delay due to the
guard interval duration.
The most important precondition for the application of a gapfiller is sufficient isolation be
tween transmitter and receiver antennas. To prevent the retransmitter from oscillating, the gain
of the retransmitter has to be less than the feedback attenuation (see Figure 3.10).
A gapfiller should have sufficient transmission power to provide coverage for the uncovered
area. The maximum possible radiated power depends on both the isolation between the recep
tion antenna and the transmitting antenna and the performance of the power amplifier of the
repeater. The antenna isolation depends on:
The height and dimension of the tower or building where the repeater is located.
The location of the area which should be covered in relation to the direction to the main
transmitter.
The environment around the repeater (buildings or other objects which could cause re
flections).
In addition to the general problem of isolation explained above, even if the feedback attenua
tion is higher than the transmitter gain, a decrease in the system performance has to be expected.
Among all reflections, there will be one dominating path coming either from the limited isola
tion between the antennas and/or the feedback from reflectors around the repeater station. In
general, there is a time delay between the input and the output of a gapfiller, mainly due to
filtering within the device. This will cause frequency selective attenuation of the retransmitted
signal similar to the characteristic of a twopath or multipath reception, resulting in a degrada
tion of system performance. As mentioned before, the isolation depends on the overall design
of the place where the repeater is installed [3].
22 DIGITAL VIDEO BROADCASTING: TECHNICAL OVERVIEW
3.7.1 DVBH
The digital video broadcasting for handheld devices(DVBH) [4] [18] can be seen as an expan
sion of the DVBT standard. It contains few additional features.
The main additional elements in the link layer are time slicing and additional forward error
correction (FEC) coding. The time slicing reduces significantly the average power consumption
in the receiver frontend and also enables smooth and seamless frequency handover. The use of
time slicing is mandatory in DVBH. The FEC for multiprotocol encapsulated data (MPEFEC)
gives an improvement in Doppler performance and increases tolerance to impulse interference.
The use of MPEFEC is optional for DVBH.
The physical layer has four extensions to the existing DVBT physical layer. First, the bits
in transmitter parameter signaling have been upgraded to include two additional bits to indicate
the presence of DVBH services and the possible use of MPEFEC. Second, a new 4k mode
is adopted. This gives additional flexibility for the network design. The 4k mode is an option
for DVBH complementing the 2k and 8k. Third, a new way of using the symbol interleaver
of DVBT has been defined. For 2k and 4k modes, the operator may select (instead of native
interleaver that interleaves the bits over one OFDM symbol) the option of an indepth interleaver
that interleaves the bits over four or two OFDM symbols, respectively. This approach brings
the basic tolerance to impulse noise of these modes up to the level attainable with the 8K mode
and also improves the robustness in mobile environment. Finally, the fourth addition to DVBT
physical layer is the 5MHz channel bandwidth to be used in nonbroadcast bands . The basic
parameters of DVBH physical layer can be found in [4].
Because of the similarity between DVBT and DVBH in the physical layer, all the algorithms
and techniques introduced by the thesis are applicable for both standards.
3.7.2 DVBS
The DVBS satellite modulator has a similar block diagram to the DVBT modulator except that
the system only uses singlecarrier modulation with QPSK [1]. Multipath is not a problem with
satellite transmission and the nonlinearity of the satellite transmitter power amplifier (due to
hard powerefficiency constraints) at present limits the modulation choice to QPSK modulation.
The system uses the same concatenated convolutional and ReedSolomon error correcting codes
but the bitwise inner interleaver is not used. Since satellite channel bandwidths are much larger,
the source bitrate can be made to vary from 2 Mbits/s up to around 60 Mbits/s. The system is
suitable for broadcast television transmission within the 12 GHz satellite bands.
3.7.3 DVBC
The DVBC cable modulator is similar to the single carrier DVBS modulator except that the
modulation scheme on the single carrier is either 16QAM, 32QAM or 64QAM, [2]. Fur
thermore, the convolutional encoder is omitted, leaving the outer interleaver and ReedSolomon
channel coding of DVBT in place. The cable receiver does, however, contain an adaptive equal
izer in order to deal with the short multipath propagation which exists in VHF and UHF cable
systems.
Chapter 4
Impulsive Noise Modeling, Effects
and Mitigation in DVBT systems
4.1 INTRODUCTION
This chapter focuses on the effects of impulsive noise on DVBT signals. It provides new algo
rithms for impulsive noise mitigation. At first, impulse noise models are discussed. Sources of
the impulsive noise are described, some general statistical models are investigated, and then the
effects of the impulse noise on the DVBT signal are studied. Secondly, the common techniques
for impulsive noise cancellation are presented. In the third section of this chapter we introduce
a new algorithm that uses pilots in the DVBT signal in order to mitigate the effects of the im
pulsive noise. Performance analysis of the new algorithm is also presented. Finally, an example
system is studied and simulated.
The performance of any communication system is dependent on the channel characteristics and
it can often be improved by the use of techniques which successfully exploit these characteris
tics. Identifying the corrupting noise distribution is an important requirement for most system
design problems because it leads to the development of methods based on which the effects of
noise are minimized. Also, it allows us to predict the performance of the system. The most
widely used noise model is the Gaussian random process. This is because if the noise results
from a random combination of a large (infinite in theory) number of independent energy sources,
then the Central Limit Theorem (CLT) applies. The use of the Gaussian model is also convenient
since it often leads to analytically tractable results. However, in some specific environments the
Gaussian noise model may not be appropriate.
23
24 IMPULSIVE NOISE MODELING, EFFECTS AND MITIGATION IN DVBT SYSTEMS
Common sources of impulse noise include lightning, industrial machines, car starters, faulty
or dusty insulation of high voltage powerlines, and various unprotected electric switches. This
interference may be produced also by various household appliances like hairdryers, vacuum
cleaners, drilling machines, etc... In addition, single or multiple bursts of pulses occur while
switching on or off any device connected to the power line. These noise sources will generate
high energy pulses which block the regular TV signal for very short time durations, resulting
in annoying spots on the TV screen and sharp click sounds in the audio.
P
X 1
ni (m) = h(k)n(m k)b(m k), (4.1)
k=0
where b(m) is a binaryvalued random sequence model of the time of occurrence of impulsive
noise, n(m) is a continuousvalued random process model of impulse amplitude, and h(m) is
the impulse response of a filter that models the duration and shape of each impulse. The dis
tributions of the random processes modeling the impulse amplitude, duration, and shape are a
priori unknown. The establishment of the statistical distribution of these parameters will fully
define the model characterizing the impulse noise.
Figure 4.1 shows the impulsive noise model given in equation (4.1).
In the literature, there are two important statistical processes for modeling impulsive noise
as an amplitude modulated binary sequence, BernoulliGaussian process and PoissonGaussian
process [19] [20]. In the first model, the random time of occurrence of the impulses is modeled
by a binary Bernoulli process b(m) and the amplitude of the impulses is modeled by a Gaussian
process n(m).
The probability mass function of a Bernoulli process is given by
b(m) = 1
PB (b(m)) = (4.2)
1 b(m) = 0
IMPULSE NOISE MODELING 25
In the second model, the probability of occurrence of an impulsive noise event is modeled by
Poisson process, and the distribution of the random amplitude of impulsive noise is modeled by
a Gaussian process. In a Poisson model, the probability of occurrence of k impulses in a time
interval of T is given by
(T )k T
P (k, T ) = e , (4.3)
k!
P rob(one impulse in a small time interval t) = t
(4.4)
P rob(no impulse in a small time interval t) = 1 t
Burst 1 Burst 2
Burst Duration
Pulse Duration
10 ms (fixed) 250 ns (fixed)
Test No. Pulses per Burst Min. Pulse Spacing(s) Max. Pulse Spacing(s) Max. Burst Duration(s)
1 1 N/A N/A 0.25
2 2 1.5 45.0 45.25
3 4 15.0 35.0 105.25
4 12 10.0 15.0 165.25
5 20 1.0 2.0 38.25
6 40 0.5 1.0 39.25
dk =
X
cj,k pkj , (4.5)
j
where cj,k are the unknown estimator coefficients and pkj represent the deviations at the pilots
subcarriers.
The M SE for the estimator, J, can be written as:
2
2
X
J = E{dk dk } = E{dk cj,k pkj }, (4.6)
j
28 IMPULSIVE NOISE MODELING, EFFECTS AND MITIGATION IN DVBT SYSTEMS
where E{.} refers to ensemble averaging operation. Minimization of J with respect to the
weights {cj,k }corresponds to forcing the error k = dk dk to be orthogonal to the samples
pl , (l = 0, m, 2m, ..., where m is the pilot spacing) and yields the following
E{k pl } = 0. (4.7)
By expanding this, it is possible to write
k = Rk ck , (4.11)
where the covariance vector is
k = E{dk pl }, (4.12)
and autocorrelation matrix is given by
R = E{pk j pl }. (4.13)
Subsequently in the absence of additive noise, the structure of the covariance vector and
autocorrelation matrix will look like the following
E{dk p0 }
E{dk pm
}
k = E{dk p2m
(4.14)
}
...
and
r0 rm r2m ...
rm
r0 ...
R=
r2m
(4.15)
rm r0
... r0
where ri = E{pi p } . The dimensions of these are determined by how many pilots are used
for the deviation estimation. It can be noted that R is independent of k, which improves the
efficiency of the determination of error deviates. Based on the blanking position information, it
is possible to calculate k and Rk using FFT of the blanking window. The optimum estimator
coefficients can be calculated from (4.11) in a straightforward manner
ck = R1 k . (4.16)
Now using these coefficients, it is possible to estimate the deviations at the datacarriers as
given in (4.5). These deviations can then be subtracted from the demultiplexed subcarrier value,
resulting in mitigation of the effects of the impulse noise.
IMPULSE NOISE CANCELLATION UTILIZING PILOTS 29
The estimation procedure described above requires some knowledge of the covariance val
ues of the deviation process. There are several approaches to get such knowledge. First, one
may derive the theoretical covariance functions taking into account the modulation parameters,
blanking window length and shaping, etc. This might be feasible at least if some approximation
and simplifying assumptions are made. The next approach is to run computer simulations for
the required system parameters and thus get reliable estimates for the covariance values. This
might give the best results relatively simply. The third method could be based on measuring
some prototype receiver to get the covariance values.
As an example, we derive an approximation for the autocorrelation function of the deviation
process for a DVBT like signal. We assume, like in the DVBT standard, that the real signal is
given by
N 1
X t
s(t) = Re{eic t bk ei2k Tu }, (4.17)
k=0
where c is the center angular frequency and k = k N/2, with k is the carrier index. bk is a
complex coefficient representing the modulated bits at carrier k. Tu is the duration of the useful
OFDM symbol (without guard interval). N is the FFT size of the OFDM modulation. For the
following we use complex envelope notation and calculate the lth sample taken at intervals T
(the DVBT elementary period), where
N
X 1 l
sl = bk ei2k N . (4.18)
k=0
In order to determine the autocorrelation function of the deviation process, we must first cal
culate the discrete Fourier transform of samples sl over the interval [u, u + L] where u denotes
the starting point of the blanking window and L is the window length. The deviation value at
carrier index is calculated as
N 1 u+L
1 X X i2q k
dv = bk e N . (4.19)
N q=u
k=0
In the derivation we have assumed that the modulation values are zero mean and statistically
independent and that the blanking length, L, is small, e.g. less than 10% in the relation to whole
symbol length N . The autocorrelation of (4.20) indeed has the desired property that the value
of r(v, w) depends only through the difference w on the index values w and . Now the re
quired covariances in (4.14) and (4.15) are directly given by the equation above for the noiseless
cases as pilot values p are just special d values for certain indexes (multiples of m).
takes place in every OFDMsymbol. By supposing that the burst occurs at samples [u, u + L]
in OFDM symbol, the elements in the 2 2 autocorrelation matrix may be constructed simply
as follows
L
1 X i2(u+j)k/N
rk = e , (4.21)
N j=0
where k = 0 or m. The effect of possibly different subcarrier amplitudes in (4.20) has been
averaged out. The covariance vector can be expressed as
r mod (k,m)
k = (4.22)
r mod (k,m)m
and hence the deviation at any k th carrier is given by
dk = cTk p, (4.23)
where ck can be calculated using (4.16) and
pk mod (k,m)
p= (4.24)
pk mod (k,m)+m
which consists of pilot deviates from two neighboring pilots.
0
10
1
10
2
BER after Viterbi 10
3
10
4
10 Blanking, L=700
Blanking, L=1000
Blanking, L=1300
Blanking+canceller, L=700
5
10 Blanking+canceller, L=1000
Blanking+canceller, L=1300
Pure AWGN
6
10
10 12 14 16 18 20 22 24 26
SNR (dB)
Fig. 4.3 Performance of impulse noise reduction algorithm in AWGN channel (16QAM).
0
10
1
10
2
10
BER after Viterbi
3
10
4
10
Blanking, L=700
Blanking, L=1000
5
10 Blanking, L=1300
Blanking+canceller, L=700
Blanking+canceller, L=1000
Blanking+canceller, L=1300
Pure AWGN
6
10
10 12 14 16 18 20 22 24 26 28 30
SNR (dB)
Fig. 4.4 Performance of impulse noise reduction algorithm in AWGN channel (64QAM).
32 IMPULSIVE NOISE MODELING, EFFECTS AND MITIGATION IN DVBT SYSTEMS
0
10
1
10
2
10
BER after Viterbi
3
10
4
10
None
Blanking
Blanking+canceller
5
10
0 500 1000 1500
Blanking window width
Fig. 4.5 BER vs. Blanking window length in AWGN channel: SNR=20 dB (16QAM).
Chapter 5
Pilot Based Impulse Noise Canceller
Enhancements
5.1 INTRODUCTION
A new method for compensating the effect of impulse noise in OFDM systems has been de
scribed in the previous chapter [P1]. The scheme uses channel estimation pilots for the esti
mation and cancellation of impulse noise. The performance of the new method has been tested
for the AWGN case with relatively large impulse burst window length. It was shown that in
this case, the performance of the DVBT system can be improved significantly. However, addi
tional measures are needed to handle the timefrequency selective channel cases with impulse
noise interference. In this chapter we present new techniques to enhance the pilot based impulse
noise canceller. Additionally, performance of combining this algorithm with other standard im
pulse noise mitigation schemes is investigated. An enhanced burst position estimation algorithm
is also studied. Finally, effects of the impulsive noise on the time synchronization in DVBT
systems are addressed.
One problem in the channel estimation with impulse noise is that an impulse burst can induce
error dispersion due to errors in channel estimation pilots. The errors are spread in time depend
ing on the length of the time domain channel estimator. Our aim here is to present methods that
provide good enough channel estimates so that blanking compensation algorithm will operate
successfully [P2]. All the methods are based on the assumption that impulse noise corrupts all
the channel estimation pilots in the effected OFDM symbols.
33
34 PILOT BASED IMPULSE NOISE CANCELLER ENHANCEMENTS
When impulse noise is considered, this approach would be preceded by preestimation [P2]:
1. Carry out preestimation of pilots of the impulse burst contaminated symbol in time or
frequency domains using impulse noise free symbols.
2. Use conventional estimation.
In Figure 5.1 the sixth (DVBT) OFDM symbol is supposed to be corrupted by impulse burst.
In the preestimation, three different methods are suggested here:
5.2.1.1 Four away OFDM symbols (FOA) Use pilots from the previous symbol with
the same pilot pattern (e.g., in DVBT four symbols back) (Figure 5.1 (a)).
5.2.1.2 Six pilots spacing (SPS) Use pilots from neighboring symbols, combine the pi
lots and produce channel estimates for the pilots of corrupted symbol using interpolation (Figure
5.1 (b)).
5.2.1.3 Twelve pilots spacing (TPS) Use pilots from neighboring symbols, make the
interpolation first (both symbols) and then calculate the estimates for the pilots of corrupted
symbol (Figure 5.1 (c)).
1
2
3
4
5
6
Combination
Interpolation
Estimated Pilots
Interpolation Interpolation
5 7
Average
Estimated Pilots
(c) Twelve Pilots
Spacing Method
Fig. 5.1 Description of the enhanced channel estimation methods in impulsive noise environment.
36 PILOT BASED IMPULSE NOISE CANCELLER ENHANCEMENTS
For the SPS method, the estimate at pilots (of both neighboring symbols) is calculated first by
where R() is the autocorrelation function of timevariant channel and refers to time in OFDM
2 2 2
symbol lengths, H = R(0) and the variances awgn and ICI are the AWGN and ICI powers,
respectively. Here it has been assumed that all the pilots contain the same amount of estimation
errors. In practice this is not true, hence the result is somewhat pessimistic.
Similarly, for the SPS method, the MSE at the pilots is given by
2 2 2
J2 = 2H 2R(1) + awgn + ICI , (5.6)
and then frequencydomain preestimation is used (e.g., in DVBT, interpolator with 6 carrier
pilot spacing). The total MSE in the pilot estimates after preestimation can be found from
2
J2 = H cH H H T
opt copt + copt copt , (5.7)
where and are the crosscovariance vector and autocovariance matrix (including the effect
of J2 ), respectively. The optimum preestimation filter coefficients are given by
copt = 1 . (5.8)
For TPS method, the MSE at the pilots for each neighboring symbol after the frequency
domain preestimation (for DVBT interpolator with 12 carrier pilot spacing), can be found
from
2
J = H cH H H T
opt copt + copt copt , (5.9)
where and are the crosscovariance vector and autocovariance matrix (including the effect
of J2 ), respectively. Note that the pilot pattern is different from the one used with SPS method,
and hence the variance vector and matrix differ from those used in SPS method. The total MSE
for TPS method after combining the pilots is given by
3 2 1 1 2
2
J3 = H R(1) + R(2) + [awgn + ICI ]. (5.10)
2 2 4
The ICI power used in the above equations can be expressed in the case of wide sense sta
tionary uniformly distributed uncorrelated scatters as [26]
ENHANCED CHANNEL ESTIMATION SCHEMES FOR IMPULSIVE NOISE ENVIRONMENT 37
N
" #
2 1 X Tu
ICI =1 2 N +2 (N i)J0 2fd i , (5.11)
N i=1
N
where N is the number of carriers, J0 (.) is zeroth order Bessel function of the first kind, fd is
the Doppler frequency, and Tu (useful) OFDM symbol duration.
The theoretical MSE values based on the above analysis versus Doppler frequencies for pre
estimation methods are shown in Figure 5.2. It has been assumed that the guard interval Tu /4
is used. The power of the channel is set to unity and the design power delay profile (PDP) is
the same as the test PDP (i.e., channel PDP is known to the receiver). Uniform PDP of given
maximum length is used. SNR is set to 30 dB. Impulses are assumed to occur quite infrequently,
i.e, consecutive OFDM symbols are not contaminated. The interpolator length for SPS and TPS
methods is 91. The maximum delay in the channel is Tu /16.
The results show an optimistic bound for the FOA method where it has been assumed that
only the pilots (from symbol located four symbols back) used for channel estimation are noisy
and contain estimation error. The other pilots are assumed to have negligible amount of noise
(AWGN and possible ICI). It can be seen that in this case, for small Doppler frequency, all the
systems give an MSE of around 103 or less. As the Doppler frequency is increased, the esti
mation performance is degraded. The TPS method is the best method for all the Dopplers. This
is due to the fact of using the interpolated pilot from the neighboring symbols.
1
10
FOA : optimistic
FOA : pessimistic
SPS
0 TPS
10
1
10
MSE
2
10
3
10
4
10
0 10 20 30 40 50 60 70 80 90 100
Doppler Hz
Fig. 5.2 Theoretical MSE vs. Doppler frequency for different preestimation methods. The interpolator
length is 91 and maximum delay of the channel is Tu /16.
The aim here is to study the performance of the new channel estimation scheme for pilot
based impulse noise canceller in the case of timefrequency selective channel. Therefore, we
need to have an impulse noise and multipath channel models that will be used in the simulation
chain.
5.2.3.1 Impulse noise model Because of the blanking operation used in the pilot based
impulse noise canceller presented in the previous chapter, only the determination of the location
of the impulse noise burst within the OFDMsymbol is needed in the derivation of the impulse
noise canceller algorithm. Therefore, in the impulse noise modeling, beside the burst location
and length L, we do not need to include other parameters that refer, for example, to the proba
bility of occurrence of the impulse burst or the impulse amplitude.
Generally in the literature, a standard BernoulliGaussian (B.G) process is used to model
impulsive noise [19] [27]. Here the value of L = 500 samples is used for the burst length, corre
sponding to 54.5 s duration in the case of 8k mode in DVBT [3]. For easier implementation,
we suppose that the impulse burst occurs exactly once in every 8 OFDM symbols. This hypoth
esis can be considered as the worst case if we compare it to the DTG models that we discussed
earlier.
5.2.3.3 Results and discussion The performance of the three methods was tested by
using the 8k DVBT system mode. The submodulation was 64QAM and the code rate was
either 1/2 or 2/3. The impulse noise burst length, L, was fixed in all simulations to 500 samples.
The interpolators used in the SPS and TPS methods are of length 125. They were designed for
the maximum delay spreads of Tu /8 and Tu /16 respectively, and SNR of 30 dB.
Figures 5.3, 5.4, and 5.5 show the BER performance of the three methods in the static SFN
cases with different code rates (1/2 and 2/3), for maximum delay spread in samples s =512,
s =1024, and s =2048, respectively. It can be seen that the three methods perform well in
smallest delay spread case (s =512), but as the delay spread is increased, only the FOA method
can keep a satisfactory performance in the BER sense.
Figures 5.6, 5.7 and 5.8 show the BER performance of the three methods in the mobile case
with different code rates (1/2 and 2/3), for maximum delay spread in samples s =512, s =1024,
and s =2048, respectively. It can be seen that the QEF criterion (BER= 2.x104 after Viterbi),
can be achieved only for TPS and SPS methods when 1/2 code rate is used and that the delay
spread s has to be smaller than 512 for the TPS method. Larger delay spread is tolerated for
the SPS method. In the 2/3 case, only small Dopplers are tolerated in order to achieve the QEF
performance. For a Doppler larger than 35 Hz, only the TPS method can pass the QEF test, with
a s =512.
ENHANCED CHANNEL ESTIMATION SCHEMES FOR IMPULSIVE NOISE ENVIRONMENT 39
0
10
1
10
2
10
3
10
4
10 FOA: CR=1/2
FOA: CR=2/3
SPS: CR=1/2
5
10 SPS: CR=2/3
TPS: CR=1/2
TPS: CR=2/3
6
10
10 15 20 25 30 35 40
SNR in dB
Fig. 5.3 BER performance for the three methods in the static SFN channel: 64QAM and s =512 sam
ples.
0
10
1
10
2
10
BER after Viterbi
3
10
4
10 FOA: CR=1/2
FOA: CR=2/3
SPS: CR=1/2
5
SPS: CR=2/3
10
TPS: CR=1/2
TPS: CR=2/3
6
10
10 15 20 25 30 35 40
SNR in dB
Fig. 5.4 BER performance for the three methods in the static SFN channel: 64QAM and s =1024
samples.
40 PILOT BASED IMPULSE NOISE CANCELLER ENHANCEMENTS
0
10
1
10
2
10
FOA: CR=1/2
FOA: CR=2/3
SPS: CR=1/2
4
10 SPS: CR=2/3
TPS: CR=1/2
TPS: CR=2/3
5
10
10 15 20 25 30 35 40
SNR in dB
Fig. 5.5 BER performance for the three methods in the static SFN channel: 64QAM and s =2048
samples.
0
10
1
10
2
10
BER after Viterbi
3
10
FOA: CR=1/2
FOA: CR=2/3
SPS: CR=1/2
4
10
SPS: CR=2/3
TPS: CR=1/2
TPS: CR=2/3
5
10
0 10 20 30 40 50 60 70
Doppler Frequency in Hz
Fig. 5.6 BER performance for the three methods in the mobile SFN channel: 64QAM and s =512
samples.
ENHANCED CHANNEL ESTIMATION SCHEMES FOR IMPULSIVE NOISE ENVIRONMENT 41
0
10
1
10
2
10
3
10 FOA: CR=1/2
FOA: CR=2/3
SPS: CR=1/2
4
10 SPS: CR=2/3
TPS: CR=1/2
TPS: CR=2/3
5
10
6
10
0 10 20 30 40 50 60 70
Doppler Frequency in Hz
Fig. 5.7 BER performance for the three methods in the mobile SFN channel: 64QAM and s =1024
samples.
0
10
1
10
2
10
BER after Viterbi
FOA: CR=1/2
3
FOA: CR=2/3
10
SPS: CR=1/2
SPS: CR=2/3
TPS: CR=1/2
TPS: CR=2/3
4
10
5
10
0 10 20 30 40 50 60 70
Doppler Frequency in Hz
Fig. 5.8 BER performance for the three methods in the mobile SFN channel: 64QAM and s =2048
samples.
42 PILOT BASED IMPULSE NOISE CANCELLER ENHANCEMENTS
The impulsive noise canceller presented in the previous chapter [P1] is using the FFT of blanked
signal to estimate the subcarrier deviation induced by the blanked burst. These subcarrier de
viations bk are subtracted from subcarrier samples values resulting from the FFT of the signal
samples with blanking. The impulsive bursts are supposed to be ideally detected. The subcarrier
deviation values bk depend on the burst position, duration, and amplitude. In practice and de
pending on the impulse burst detection algorithm accuracy, a false detection of some impulsive
samples, and hence miss blanking, will induce a different subcarrier deviation pattern.
As mentioned earlier, we focus on the case where the impulse burst position is ideally de
tected. Therefore the amplitude of the burst does not affect the results. The subcarrier deviation
will be used as additional channel state information in the DVBT receiver. We consider more
reliable, the received samples that include less deviation due to the blanking and vice versa. We
highly weight received sample with corresponding small deviation. The weighting operation
can be combined with the channel weighting operation already used in the soft bits generation
block in the conventional DVBT receiver [P3]. The next section will present in more detail,
how this can be implemented. Figure 5.9 shows a typical case of quantized subcarrier devia
tions. In the Figure we consider for illustrative purpose, only 200 samples of the OFDM symbol
including blanked burst.
1.8
1.6
1.4
0 20 40 60 80 100 120 140 160 180 200
subcarrier index
3
Absolute value
0
0 20 40 60 80 100 120 140 160 180 200
subcarrier index
1
Absolute value
0.5
0
0 20 40 60 80 100 120 140 160 180 200
subcarrier index
1
Quantized levels
0.5
The basic Viterbi decoder for AWGN channel calculates the path metric dj using the follow
ing equation
n
X
dj = yi aij 2 , (5.12)
i=1
where yi is a detected sample, aij is ith symbol value from the j th code word and n is the length
of the code word. In the frequency selective multipath channel, the path metric can be written
as follows
n
X
dj = vi i aij 2 , (5.13)
i=1
where vi represent the unequalized signal and i is a channel complex gain factor parameter
during the ith symbol. Therefore, vi = i yi where yi is ideally equalized symbol and the path
metric can be written as
n
X
dj = i 2 yi aij 2 . (5.14)
i=1
This equation shows that the branch metrics computed using the channel state information
can be done by weighting each local metric by the corresponding squared channel gain factor.
High channel gain factor results in high reliability and small factor is associated to low local
metrics with low reliability in Viterbi decoder.
When considering the presence of impulsive interference, we can intuitively exploit the sub
carrier deviation to enhance the CSI. High carrier deviation signal implies a bad sample and
small carrier deviation means a more reliable sample. Therefore, we define an amplification
factor
i = 1/bi . (5.15)
where bi is the subcarrier deviation level defined earlier. The path metric can now be computed
as follows:
X n
dj = i 2 i2 yi aij 2 . (5.16)
i=1
Therefore, the branch metrics computation can now be done by weighting each local metric by
the corresponding squared channel attenuation factor and the squared inverse subcarrier devia
tion level.
From the implementation point view, and because the subcarrier deviation levels bi could
present a high dynamic range, we will consider quantized levels and we limit the i2 factor to a
finite set of values. Figure 5.9 presents the case of 10 levels quantizer.
In practice, when the burst are occurring seldomly, a two value quantizer will be enough to
give an acceptable BER performance. For example, if the magnitude of the subcarrier deviation
bi is larger than a threshold T1 then i is set to zero, and if bi is smaller than the threshold T1
than i is set to one. Here, the used threshold value is determined by extensive simulations.
When focusing on a certain range of impulse burst durations, it is shown that the position of the
burst is not a critical parameter in defining the carrier deviation levels. In Figure 5.10, a block
diagram for modified soft bit generation in the presence of impulsive interference in a DVBT
receiver is shown. If using more than two levels, the quantizer needs to know the number of
levels and the maximum and minimum allowed levels. If we consider that the minimum level
44 PILOT BASED IMPULSE NOISE CANCELLER ENHANCEMENTS
will be set to zero, the maximum level needs to be determined by averaging simulation results.
It is shown that the value of 0.5 gives good system performance. The number of levels is a free
parameter, which compromises between complexity and sufficient performance [P3].
Extract Channel
pilots estimation
Subcarrier
deviation Quantization 1/.
estimation
Fig. 5.10 Block diagram of a modified soft bit generation in DVBT system when using a pilot based
impulse noise canceller.
0
10
1
10
2
10
BER after Viterbi
3
10
4
10
5
10 Blanking: CR=2/3
Blanking: CR=1/2
Blanking + Softbits scaling: CR=2/3
6
10 Blanking + Softbits scaling: CR=1/2
Blanking + Softbits scaling + Compensation: CR=2/3
Blanking + Softbits scaling + Compensation: CR=1/2
7
10
10 15 20 25 30 35 40
SNR in dB
Fig. 5.11 BER performance with different impulse interference cancellation methods in static Rayleigh
channel case.
0
10
1
10
2
10
BER after Viterbi
3
10
4
10
Blanking: CR=2/3
Blanking: CR=1/2
5 Blanking + Softbits scaling: CR=2/3
10
Blanking + Softbits scaling: CR=1/2
Blanking + Softbits scaling + Compensation: CR=2/3
Blanking + Softbits scaling + Compensation: CR=1/2
6
10
0 10 20 30 40 50 60 70
Doppler Frequency in Hz
Fig. 5.12 BER performance with different impulse interference cancellation methods in TU6 mobile
channel case.
46 PILOT BASED IMPULSE NOISE CANCELLER ENHANCEMENTS
The pilot based impulse noise canceller method can cancel the effects of impulse noise very
effectively [P1]. However, the efficiency depends on the reliability of the burst detection algo
rithm. In the earlier discussion, we always assumed ideal detection of the burst position. In
practical burst position estimation algorithms, false alarms because of small amplitude impulse
cases appear and degrade seriously the performance of the pilot based impulse noise canceller
method. Therefore, care has to be taken to improve the detection of the impulse bursts. In addi
tion, it is known that the use of interleavers and coding algorithms are also effective in impulsive
environments [31]. Interleavers are usually used to destroy memory effects of the communica
tion channel or to disperse error burst caused by the channel. In this section, we focus on the
use of hybrid approaches where combinations of these techniques are considered [P4]. We also
introduce a practical burst position detection algorithm.
ter about 0 dB, the blanking receiver starts to perform better than the conventional receiver. It
is also interesting to see how the perfect position information affects the performance of the im
pulse noise compensation schemes. It can be seen that for low impulse amplitude levels, the
system with practical position detection performs better than with perfect position knowledge.
This is because it is difficult to locate the burst in these conditions. It seems that the effects
of blanking may be more severe than the effects of the impulsive noise burst in low impulse
amplitude levels case.
20
Blanking and cancellation using detection
18
Blanking and cancelling with
known burst position
16
14
Conventional receiver
SIR dB
10
8
Blanking with known burst position
4
10 8 6 4 2 0 2 4 6
Impulse amplitude dB
Fig. 5.13 Typical SIRbehavior after different receiver structures for 16QAM, code rate 2/3, and 20 dB
SNR.
In extensive simulations, the above mentioned settings of the impulse burst detection algo
rithm have turned out to be optimum in the AWGN and static multipath channel cases. However,
in severely frequency selective mobile channel cases with variable impulse burst amplitude and
duration, the pilot based impulse canceller needs to be enhanced. Additionally, the detection of
the impulse burst at low amplitude levels (e.g., 3 dB) is very difficult, which results on many
false alarms that cause consequently a degradation of the performance of the pilot impulse can
celler. Therefore, we need to deal with these cases in a special way. Fortunately, when the
amplitude of a burst is relatively small, the spreading of the burst after FFT operation will have
a minor effect on the data carriers, and interleaving and coding will be enough to compensate
the impulse effects.
1
h1 (n) = 1 , n = 1, 2, ....1 .
1 (5.18)
h2 (n) = 2 , n = 1, 2, ....2 .
where h1 refers to the same local power estimator used earlier. The other filter h2 , is a longer
moving average FIR filter meant to estimate the average power of the received signal. The length
2 of h2 is chosen to be of the order of the OFDM symbol length (useful symbol + guard in
terval). The threshold is then determined by adding a fixed margin that is sufficient to avoid
unnecessary blanking. A margin of 1.5 dB is found to be suitable for 8k DVBT system, when
moderate Doppler is considered. Figure 5.14 shows a block diagram of the proposed adaptive
impulse burst detection algorithm. The amplitude square of the received signal, possibly con
taminated by impulse noise, is first calculated. The output signal is filtered by the two FIR filters
h1 and h2 . The output of h2 is first scaled by a proper scaling margin and then used as threshold
for the output of h1 . If the output sample of h1 is larger than the threshold, then the received sig
nal at this sample index is blanked, otherwise it is kept the same. Figure 5.15 shows the output
signals of h1, h2 , and the used threshold device. As can be seen, the threshold can follow the
signal amplitude without any error in the detection of the impulse burst, despite the relatively
severe Doppler spread (50 Hz) and the small impulse burst amplitude ( 3dB).
Received signal
r(n)
2
(.) Threshold level
Fig. 5.14 Block diagram for adaptive burst detection and blanking.
1
The short fillter (h 1) output
2
Amplitude (dB)
7
0 1 2 3 4 5 6 7 8
OFDM symbol time index x 10
4
Multipath
channel
Output bit
Adaptive stream
impulse burst
detection
of interleaving only and pilot based canceller only. The interleaver parameter are such that the
number of branches J = 16 and the FIFO shift register length F = 6048. The FIR filters h1 and
h2 have 150, and 10000 taps respectively. As can be seen in Figure 5.17, the hybrid approach
improves the system tolerability to impulse noise in the BER sense. For high burst amplitudes
(larger than 0 dB) the use of interleaver only, cannot sufficiently compensate for the effects of
the impulse bursts. However, when used with pilot based impulse canceller approach, a BER
smaller than the QEF (2.104 ) criterion can be achieved. When the burst amplitude is smaller
than 0 dB, we can se how false detection can affect the performance of the pilot based canceller.
Fortunately, when both techniques are combined, the QEF criterion is again achieved. In Figure
5.18, results for mobile channel [17] case with moderate Doppler (25 Hz) are shown. As can be
seen, the hybrid approach gives always better performance than the conventional methods.
0
10
Interleaver : number of branches J = 16
FIFO register length F = 6048
Burst length = 500 samples (8k mode).
1 Static Rayleigh channel (DVBT Spec.).
10
AWGN : SNR = 25 dB.
2
10
BER after Viterbi
3
10 BER = 2.104
4
10
Interleaver
5
10 Pilot based canceller
Interleaver+Pilot based canceller
6
10
10 5 0 5 10 15 20
Impulse burst amplitude in dB
Fig. 5.17 BER simulation results in static Rayleigh channel case for 64QAM, code rate 2/3, and SNR
= 25 dB.
0
10
Interleaver : number of branches J = 16
FIFO register length F = 6048
Burst length = 500 samples (8k mode)
1 Mobile TU6 Channel
10
Doppler Frequency = 25Hz
AWGN : SNR = 25 dB
2
10
BER after Viterbi
3
10
BER = 2.104
4
10
5
Interleaver
10
Pilot based canceller
Interleaver + Pilot based canceller
6
10
10 5 0 5 10 15 20
Impulse burst amplitude in dB
Fig. 5.18 BER simulation results in mobile TU6 channel for 64QAM, code rate 2/3, Doppler = 25 Hz,
and SNR = 25 dB.
IMPULSIVE NOISE EFFECTS ON TIME SYNCHRONIZATION 51
In this section we investigate the effects of impulsive noise on symbol synchronization in DVB
T receiver [P5]. Specifically, we focus on the performance of the maximum likelihood symbol
synchronization (MLSS) algorithm in OFDM systems [32]. Limitations of the MLSS algorithm
are investigated. Simulations results for the timing offset mean squared error (MSE) in static
and mobile channels are shown. Ideas to enhance the performance of the MLSS algorithm in
these environments are also proposed.
here is the magnitude of the correlation coefficient between r(k) and r(k + N ) and * refers
to the complex conjugate operator. As can be seen in Equation (5.19), the symbol timing es
timate depends on the received signal. Therefore, care has to be taken when the observation
interval includes also impulse noisy samples. If those samples are blanked, only the position
and the duration of the burst will affect the symbol timing estimate. In AWGN channel, the
MLSS is shown to have good MSE performance. Further improvement of the MLSS algorithm
performance can be achieved by averaging the MSE over several OFDM symbols.
5.5.1.1 Mean loglikelihood timing offset estimation In [32] it is shown that the es
timation methods can be improved if the parameter can be considered constant over several
OFDM symbols. This can happen in static and slowly changing environments. If the obser
vation interval contains M complete OFDM symbols, the average loglikelihood function of
given this observation interval is shown to be
M1
1 X
() = m (), (5.20)
M m=0
where m ()is the loglikelihood function of given that only mth frame is observed. Averag
ing operation improves the result significantly.
the result when compared to the no averaging case. Here we consider a different approach. The
ensemble loglikelihood is computed using the median (nonlinear filtering) and not the mean
(linear filtering)[P5]. This is mainly motivated by the observation that only few symbols are af
fected by the impulse burst in M complete symbols. Hence, the median will simply filter out the
erroneous timing offsets estimates considered as outliers in the total M observation symbols.
The median loglikelihood is then given by:
() = M EDIAN {0 , 1, . . . , M1 } (5.21)
In static channel case this will secure very good performance in the MSE sense.
5.5.1.3 Hybrid mean and median timing offset estimation In the mobile channel
case, the assumption that the parameter is constant over several OFDM frames is not valid
anymore. It is possible to improve the MLSS performance for high Dopplers by combining
both linear and non linear operations. As an example of combining median and mean opera
tions for signal processing, we can cite the modified trimmed mean (MTM) filter solution [33].
The MTM selects the median from a moving signal window M and averages only those points
inside the window whose values are close to the median. The MTM can be defined by two
parameters, M , the total number of used symbols and W , the number of closest estimates to
the median. Although it is more complex, the hybrid approach can assure good performance
in various environments. We used here the average of 4 closest estimates to the median, i.e.,
W = 4.
0.14
MSE without average, SFN delay=300 samp.
MSE without average, SFN delay=700 samp.
MSE without average, SFN delay=1000 samp.
0.12
MSE with average, SFN delay=300 samp.
MSE with average, SFN delay=700 samp.
MSE with average, SFN delay=1000 samp.
0.1 MSE with median, SFN delay=300 samp.
MSE with median, SFN delay=700 samp.
MSE with median, SFN delay=1000 samp.
0.08
MSE
0.04
0.02
0
0 200 400 600 800 1000 1200 1400
Burst duration in samples
Fig. 5.19 Timing offset MSE performance in Rayleigh static SFN channel.
0.09
0.08
0.01
0
0 200 400 600 800 1000 1200 1400
Burst duration in samples
Fig. 5.20 Timing offset MSE performance in Mobile SFN channel based on TU6 profile, with
MTM(16,4) filter and Doppler frequency = 100 Hz.
Chapter 6
GapFiller and Common Techniques
for Loop Interference Cancellation
6.1 INTRODUCTION
The terrestrial digital video broadcasting (DVBT) [3] is a well established communication sys
tem where broadcasters launch their networks with high power transmitters in order to quickly
insure an attractive coverage to TV operators. However, there are number of problems to over
come in order to maintain a good quality of services in a substantially growing network [29],
[34], [35]. One of the most serious problems is service deterioration caused by shadowing. A
possible answer to the shadowing problem is the installation of gap fillers [36], [37], [38].
In DVBT system, a gap filler scheme without any regeneration of the received signal (direct
relay) is preferable. If regeneration is used, a long delay of more than the guard interval will take
place and intersymbol interference (ISI) will occur in the received OFDM signals. Therefore,
we focus on the direct relay system in SFN operation. Figure 6.1, shows a simple configuration
of a DVBT system using a single relay station.
A particular problem with the use of gap fillers in DVBT systems is related to the coupling
(loop interference) between transmitter and receiver antennas at the relay station [36] [39]. The
coupling may cause oscillations in the repeater [39], [36] and distorts the repeated signal. The
loop interference must be reduced to a tolerable level in order to avoid distortion and oscillation
problems. Conventional alternatives to limit the coupling are based on spatial positioning of
the transmitter and receiver antennas in the relay. These solutions, are however expensive and
not always satisfactory. Beside spatial solutions, not many measures have been proposed in the
literature.
In this chapter we aim to give a review of the main existing algorithms which have been
introduced for dealing with loop interference cancellation in digital TV systems using gap fillers
for coverage extension and to study their performance and limitations.
55
56 GAPFILLER AND COMMON TECHNIQUES FOR LOOP INTERFERENCE CANCELLATION
In order to evaluate the performance of the loop interference canceller algorithms, we introduce
the channel model that the DVBT signal will face while propagating from the transmitter to the
relay station. We also introduce the loop interference model, which the repeated signal, in the
relay, will obey. These models will be used in the analytical and simulation studies. We assume
that the channel from the repeater to the user receiver is AWGN channel. The user is usually
close to the repeater so that we can neglect the multipath effects in this part of the link. The
dominant frequency selectivity effects are then present mainly between the main transmitter and
the repeater.
A complex base band model of DVBT is used in the loop interference cancellation algorithms
studies.
p L 1 L 1
p L 1
p
where (.) is the Dirac delta function, l is the tap index, [l ]l=0 , [l ]l=0 , and [l ]l=0 are
the random channel amplitudes (following either Rayleigh or Ricean distribution), phases, and
delays, respectively. Lp is the number of resolvable paths. The first path being the reference
path whose delay 0 = 0. Lp is related to the ratio of the maximum delay spread of the channel
max to the system sampling period T (in the DVBT system, T = 7/64 s for 8 M Hz channels)
[3] as follows:
max
Lp = . (6.2)
T
Using the slowfading assumption [40], Lp is considered to be constant over the symbol in
terval. When Ricean environment is considered, the Ricean factor P 1 (the ratio of the power of
the direct path, the lineofsight ray, to the power of the reflected paths) is defined as follows:
20
P 1 = PLp 1 . (6.3)
k=1 k 2
In the system performance evaluation, a variable Ricean factor P 1 will be used. When Rayleigh
environment is considered, P 1 is simply set to zero, by nullifying the line of sight component
[12], [40]. The static Rayleigh profile is using the relative power, phases, delays listed in Table
A.1 in the Appendix A.
Coupling
Relay broadcast
station
Fig. 6.1 Simple configuration of DVBT system using single relay station.
electrically coupled via antennas. The transmitter and receiver antennas having different di
rectivities must be fixed so as to prevent the transmitted signal from interfering the received
desired signal. Traditionally, in order to suppress the coupling, these antennas must be spatially
separated, which requires a large occupancy area in the relay station construction.
A model for the broadcastwave relay SFN used in our basic study is shown in Figure 6.2.
Coupling path
C(z)
X(z)
Multipath
Channel H(z) g
R(z) S(z)
Amplifier
AWGN AWGN
Nb(z) Nc(z)
Fig. 6.2 A model for the broadcastwave relay SFN used in our basic study.
The aim is to cancel the loop interference caused by coupling. Let X(z) be the transmitted
OFDM signal. H(z) is the channel between the transmitter and the relay station, as described
earlier. R(z) is the received signal at the relay end, and Nb (z) is AWGN added to the received
signal at the relay station. C(z) is the transfer function of the coupling path. G(z) is the gain
of the amplifier at the relay station.
In order to evaluate the performance of the loop interference canceller algorithms, we need
to have a general model for the coupling path. Here we consider the following assumptions:
The coupling path has either one or multitap profile with a maximum delay T2 .
58 GAPFILLER AND COMMON TECHNIQUES FOR LOOP INTERFERENCE CANCELLATION
The coupling path tap coefficients follow a Rice distribution with a Ricean factor P2.
Given these assumptions, the coupling path transfer function C(z) can be described as fol
lows:
N
X
C(z) = k ejk z k . (6.4)
k=M
where
 M = T1 /T .
 N = (T1 + T2 /T ).
 k is the gain of the kth path.
 T is the DVBT system sampling period.
2M
P 2 = PN . (6.5)
k=M+1 k 2
In Figure 6.3 we show an example of the normalized impulse response amplitude of the coupling
path versus time normalized to the sampling period T .
0.9 T T
1 2
0.8
Normalized amplitude of the coupling
0.7
0.6 M
0.5
0.4
0.3 , ,...,
M+1 M+2 N
0.2
0.1
0
0 10 20 30 40 50 60 70 80 90
Time in 1/T
Fig. 6.3 An example of the impulse response amplitude of the coupling path.
Generally the gain of the repeater, G(z), is assumed to be frequency independent. Therefore,
in the following we will use a constant value g to model the gain parameter.
To simulate the severity of the loop, we introduce the loop amplitude ratio parameter, LAR,
as the ratio of the power of the main (useful) signal to coupling signal (coupled back to the relay
receiving antenna). If the power of the main signal is normalized to unity, then LAR can be
described by the following equation:
1
LAR = qP . (6.6)
N 2
g k=M k
SIGNAL AND CHANNEL MODELS 59
q
PN 2
For the system to be stable [41], [42], [43], the denominator, g k=M k in equation (6.6)
needs to be less than one. For instance, if the gain is unity then the LAR should be larger than
one. A higher LAR value means a more severe coupling.
Another way to express the severity of the loop interference is by using the gain margin pa
rameter [44], which can be defined as the difference between the antenna isolation and the gap
filler gain:
N
X
Gain margin(dB) = 20log10 k g(dB) (6.7)
k=M
T2=0
Magnitude
a0
response
T1=M
Samples
y(n) s(n)
g
1
impulse response magnitude
Input signal
0.9
Gain margin
0.8
0.7
0.6
Magnitude
0.5
0.4
0.3
Delay M
0.2
0.1
0
0 5 10 15 20 25 30 35
Samples
Besides spatial solutions, not many techniques for loop interference cancellation have been
published in the literature. Some of the existing methods use the frequency domain adaptive
schemes to estimate and cancel the loop interference. They benefit from the known carriers that
are transmitted within the OFDM symbol. Other technique use the LMS adaptive algorithm. In
the following, we will review the basics of these techniques and we will show their performance
and limitations.
S(z) 1
F (z) = = (6.8)
X(z) 1 gC(z) + W (z)
+ + +

+
After taking its IFFT, Er (n), the impulse response of the residual loop interference, is used in
updating the coefficients of W (z). Even though the algorithm was designed for ISDBT based
systems [45], [46], [47], we can extend its use to the DVBT system. Because of the slight
differences in the pilot patterns in ISDBT and DVBT standards, we use a slightly different
approach in the estimation of the error function Er (z). In order to estimate F (z) and therefore
Er (z) we utilize mainly the same conventional channel estimation block in the DVBT receiver
[48]. We tune our general model of the loop interference defined in Section 6.2 to fit the same
model used in [36]. It is a one tap loop model where T2 = 0 and N = M . The gain of the
repeater, g, is set to unity. Other parameters are shown in Table 6.1. Simulations results to
evaluate the BER at the receiver for a DVBT system were carried out. Figure 6.7, shows the
BER versus SN R at the receiver.
Three curves for the cases without canceller, with canceller, and no loop interference are
shown. It can be seen that for these parameters, the frequencydomain adaptive canceling
scheme can effectively cancel the loop interference.
However, when considering the presence of, e.g., a Rayleigh channel profile between the
transmitter and the relay, the algorithm fails. This can be proven analytically by reproducing the
same analysis, and including H(z) in the model shown in the Figure 6.2. It can be shown that
because of the deep fades that the Rayleigh type of channel has the Er (z) defined in Equation
(6.9) will tend to infinity at these frequencies and that will make the estimation of C(z) difficult.
62 GAPFILLER AND COMMON TECHNIQUES FOR LOOP INTERFERENCE CANCELLATION
0
10
1
10
No loop interference
With Canceller
No Canceller
3
10
4
10
10 12 14 16 18 20 22 24 26 28 30
SNR at the DVBT receiver ( dB )
Fig. 6.7 BER performance of the frequency domain adaptive algorithm for singletap loop with LAR
of 1dB. AWGN channel is assumed between the relay and the receiver. 64QAM and a code rate of 2/3
are considered.
In the next Chapter we will propose an enhanced version of this algorithm where this limitation
is overcome.
6.3.2.1 LMS loop interference cancellation with variable relay gain Obviously,
the coupling effects can be reduced to a tolerable level by reducing the relay gain. However,
this will also reduce its coverage. The idea in this algorithm is to set the gain at an initial stage
to a low level, which allows adaptive tracking of the loop interference parameters without being
disturbed by system oscillation. During the second stage, the gain is increased gradually, until
the aimed relay amplification is reached [49](Figure 6.8). We refer to this as variable relay gain
control operation (VRGC). The adaptive tracking of the loop parameters can be done using the
LMS algorithm, where the coefficients of the canceller are updated by minimizing the power of
the signal at the input of the power amplifier of the repeater. The power at this point is at its
minimum, when there is no coupling.
Figure 6.9, shows the block diagram of the relay station with the LMS adaptive loop inter
ference cancellation scheme.
The main drawback of this algorithm is in its limited stability [49]. Additionally, the algo
rithm does not tolerate a high repeater gain. In the next Chapter we will present an enhanced
version of the algorithm so that it can track the loop parameters in a more stable manner.
COMMON TECHNIQUES FOR LOOP INTERFERENCE CANCELLATION 63
Gain of the
relay
Final Gain
Initial Gain
A1 A2 Time
To the DVBT
Coupling path receiver
c(n)
r(n)
i(n) s(n)
g(n) w(n)
From the
transmitter
AWGN
Nb(n) Error LMS
Loop+canceler: P = 50 dB
Loop+canceler: P = 30 dB
Loop+canceler: P = 20 dB
Loop+canceler: P = 10 dB
Loop+canceler: P = 0 dB
5
10
0 100 200 300 400 500 600 700
Number of noisy samples in the guard band
Fig. 6.10 BER for adaptive LMS loop interference cancellation with spectrum whitening method: Ricean
channel case and different noise bandwidths and power levels. Singletap loop, LAR = 1 dB, SNR=25
dB, 64QAM and a code rate of 2/3 are considered.
2
10
BER after Viterbi
3
10
4
10
0 100 200 300 400 500 600 700
Number of noisy samples in the guard band
Fig. 6.11 BER for adaptive LMS loop interference cancellation with spectrum whitening method:
Rayleigh channel case and different noise bandwidths and power levels. Singletap loop, LAR = 1
dB, SNR=25 dB, 64QAM and a code rate of 2/3 are considered.
Chapter 7
Enhanced Algorithms for Loop
Interference Cancellation
7.1 INTRODUCTION
In this chapter we aim to introduce new algorithms dealing with loop interference cancellation
in DVBT systems using gap filler for coverage extension and to show their performance and
limitations. In addition, we will develop some extensions of the common techniques presented
in the previous chapter to improve their functionality and to overcome their limitations.
1
X
Q(l) = s(n)s (n + l) l = 1, 2, ..., (7.1)
n
where refers to the number of samples of the signal s(n) used in the autocorrelation compu
tation.
Figures 7.2 and 7.3 show the magnitude and the angle of the autocorrelation sequence Q(l)
respectively. It can be seen that if we pick the second local largest maximum sample in the
67
68 ENHANCED ALGORITHMS FOR LOOP INTERFERENCE CANCELLATION
autocorrelation sequence for the positive lags, we get an estimate of the loop tap a0 and delay
M.
The same approach can be generalized for DVBT signals. The loop interference cancellation
can be seen as a parameter estimation where the parameters are :
One tap complex coefficient a0 = M ejM with M defining the ratio between the trans
mitted and coupled signal in the relay.
0.9
Magnitude of a
0
0.8
0.7
Delay of M sample
0.6
Magnitude
0.5
0.4
0.3
0.2
0.1
0
0 5 10 15 20 25
Samples
1
Amplitude of the autocorrelation sequence Q(l)
M
0.9 Amplitude of the loop taps
0.8
0.7
0.6
Amplitude
0.5
0.4
0.3
0.2
0.1
0
0 5 10 15 20 25 30 35 40 45 50
Autocorrelation index
4
Angle of the autocorrelation sequence Q(l)
Angle of the loop taps
3
1
Angle
4
0 5 10 15 20 25 30 35 40 45 50
Autocorrelation index
Coupling path
M
a0 z To the
X(z)
S(z) receiver
Multipath channel g
H(z)
R(z)
AWGN
Canceller Nc (z )
AWGN W(z)
Nb (z )
Q( k M )
Delay
Figure 7.4 shows the general block diagram of the autocorrelation method for loop interference
cancellation in DVBT system. The total delay introduced by the multipath channel and the in
terference loop is assumed to be smaller than the OFDM system guard interval. We also assume
that the loop interference is invariant in time and the system is ideally synchronized.
An observation length of two guard intervals is used in the autocorrelation computations.
An estimate of the loop interference parameters (complex tap coefficient a0 and delay M ) are
found by picking the second largest sample value in the normalized computed autocorrelation
function. Once the complex tap coefficient and the delay parameters are estimated, the canceller
parameters will be initialized by this first estimate. The algorithm steps are repeated for every
received OFDM symbol and the canceller parameters will be updated iteratively by taking the
average of the previous estimate and the current one.
The performance of the autocorrelation method was tested through simulations. The condi
tions associated with the loop interference are assumed to be the same as in [36] and given in
Table 6.1.
We assume the presence of multipath propagation channel defined by the Ricean or the
Rayleigh models where the coefficients are given in Table A.1 [3]. In the Ricean case, the value
of P 1 = 10 is used for the Ricean factor and in the Rayleigh case, P 1 = 0.
The results of BER versus SNR at the receiver are shown for both Ricean and Rayleigh chan
nel cases in Figure 7.5 and 7.6, respectively. It can be seen that the BER is significantly improved
by the use of the new loop interference canceller.
0
10
1
10
No loop interference
With canceller
No canceller
2
10
BER after Viterbi
3
10
4
10
5
10
10 12 14 16 18 20 22 24 26 28 30
SNR at the DVBT receiver ( dB )
Fig. 7.5 BER in the Ricean channel case with LAR of 1dB using the basic autocorrelation method,
64QAM , and a code rate of 2/3
0
10
1
10
2
10
No loop interference
4
With canceller
10 No canceller
5
10
10 12 14 16 18 20 22 24 26 28 30
SNR at the DVBT receiver ( dB )
Fig. 7.6 BER in the Rayleigh channel case with LAR of 1dB using the basic autocorrelation method,
64QAM, and a code rate of 2/3.
7.2.2.1 Multitap loop with exponential profile In practice, the feedback signal due to
coupling will face a multipath propagation channel. A general coupling path transfer function
C(z) is given in Equation (6.4). In order to extend the use of the autocorrelation method for
multitap loop model, we assume that the channel magnitudes l are given by the following
equation:
l+1 = l e , M l N1 (7.2)
The parameter defines the relation between the consecutive tap magnitudes. This is a par
ticular case of the general profile commonly used for modelling terrestrial channels [51] [40].
As is increased, the main feedback energy is concentrated in the first few taps, for instance, if
approaches + we will have a one tap model as used in [36].
7.2.2.2 Analysis The aim here is to estimate the loop parameters, which means the coef
ficients (a1 , ..., aM , ...aN1 ) where ak = k ejk , k = 1, ..., N1 . For simplicity, the gain g of
the repeater is set to unity. In Figure 6.2, the repeater transmitted signal, s(n), (we neglect the
effect of the AWGN Nb (n)) is given by:
N1
X
s(n) = r(n) + ak s(k n) (7.3)
k=1
where r(n) is the received signal from the main transmitter. Based on the central limit theo
rem, we know that if the number of OFDM carriers is sufficiently high, r(n) can be considered
asymptotically Gaussian with zero mean and variance r2 . In our analysis, we will start from
the assumption that the received sequence r(n) is also white and we will simulate in the sequel
the effects of the virtual carriers, cyclic prefix and the multipath channel on the cancellation
performance.
Given the earlier assumptions, the sequence s(n) is known as an autoregressive process (AR)
of order N1 , and the AR parameters consist of the FIR filter coefficients (a1 , ..., aM , ...aN1 )
and the driving white noise process with variance r2 [52].
72 ENHANCED ALGORITHMS FOR LOOP INTERFERENCE CANCELLATION
In general, it is known that the autocorrelation function (ACF) of the sequence s(n) satisfies
the recursive difference equation,
K
P
al ss [k l] 1kK
l=1
ss [k] = K
(7.4)
al ss [l] + r2
P
k=0
l=1
where K is the number of samples used to compute the ACF (K N1 ). Giving that l = 0
for l < M , and l > N1 , Equation (7.4), can be rewritten as follows:
N1
P
al ss [k l] 1 k N1
l=M
ss [k] = N1
(7.5)
al ss [l] + r2
P
k=0
l=M
By substituting the loop interference model parameters introduced in Equation (7.2) in the
Equation (7.5), and using the fact that M 1 (actually M can be controlled by the system
designer and chosen to be high enough to assure uncorrelated data for larger lag), ss [0] r2
and the ACF at k = M + 1 is given by
N1
l e(lM) ejl ss [M + 1 l]
P
ss [M + 1] =
l=M
N1 (7.6)
= M+1 ejM +1 ss [0] + l e(lM) ejl ss [M + 1 l]
P
l=M
l6=M+1
As can be seen, in the second term of the right side of Equation (7.6), the autocorrelation lag
is always different from zero; therefore this term is small compared to the first term. Hence we
can write
ss [M + 1]
M+1 ejM +1 = aM+1 (7.7)
ss [0]
Using the same analysis we can find the other coefficients an by
ss [n]
n ejn = an f or M n N1 (7.8)
ss [0]
Obviously, for n = M, ...., N1 the following inequalities hold,
ss [N1 ] ss [N1 1] ss [M + 1] ss [M ]
   .....    (7.9)
r2 r2 r2 r2
and M can be determined by taking the ACF maximum absolute value, ss [k]  , for 1 k N1
and considering the corresponding index k. Despite N1 is an unknown parameter, it is possible
to estimate only the N1 first taps that present an amplitude higher than a certain threshold.
Using the earlier analysis, the loop parameters can be estimated using the ACF and Equa
tions (7.67.8), when the loop coefficient amplitudes satisfy the relations of Equation (7.2). In
practice, we have to take into account also other factors that affect the results.
Firstly, we have to consider that the received signal, s(n) , in DVBT and ISDBT systems
[3] [45] is not actually white because of the cyclic prefix and the virtual carriers. Additionally,
AUTOCORRELATION BASED CANCELLER ALGORITHM 73
the signal r(n) usually faces multipath propagation that has a Ricean or a Rayleigh profile [3]
which could change the statistical properties of the transmitted signal and destroy its possible
whiteness. AWGN, always present in any communication systems, could also affect the accu
racy of the estimation of the loop parameters. In the earlier analysis, the received signal variance
r2 is assumed to be known and constant. However, we need to estimate it and the estimation
error will also influence the performance of the autocorrelation method.
To take into account all these additional factors, and to minimize their effects on the algo
rithm performance, we try to proceed iteratively. In the beginning we estimate the power of the
received signal r(n) while keeping the repeater off. This operation can use 2 to 10 OFDM sym
bols ( 2 20 ms in 8k mode). Here we are considering a slowly changing environment which
is normally justified in such scenarios. The second step consists of turning the repeater on and
estimating the first tap as according to Equation (7.8). The FIR filter w(z) having coefficients
(b0 , ..., bM , ...bN1 ) set initially to zero, is updated iteratively. At the first iteration, we compute
the ACF and we set bM = ssss[M] [0] where M is determined by taking the index corresponding to
the maximum ACF amplitude for lag k = 1..., N1 . ACF is estimated using a block of OFDM
symbols. The next tap bM+1 is similarly estimated in the next iteration while the first tap (bM )
is kept constant.
b0 z 1
b1
bM
z 1
bN 1
1 z 1
bN 1
Fig. 7.7 Loop interference cancellation based on the iterative autocorrelation method.
Figure 7.7 shows a block diagram of the iterative autocorrelation method for loop interfer
ence cancellation with exponential delay profile. Here we are estimating the loop taps one by
one, to avoid the possible estimation error that could occur if we compute all the coefficients at
once according to Equations (7.67.8). By estimating every tap while cancelling the effects of
the preceding loop taps, the accuracy of the autocorrelation algorithm will be enhanced. This
will be proven in the sequel when we consider realistic cases.
To evaluate the performance of the enhanced autocorrelation algorithm we consider an 8k
mode DVBT system that transmits 64QAM modulated symbols with error correction at the
receiver. We consider the case of AWGN, Ricean and Rayleigh channels. The parameters asso
74 ENHANCED ALGORITHMS FOR LOOP INTERFERENCE CANCELLATION
ciated with the loop are varied from typical to worst case conditions. As a reference we consider
an ideal case, where the transmitted signal at the repeater, i.e., s(n) is assumed to be white and
Gaussian distributed. Mean squared error between the exact feedback loop coefficients and the
estimated ones will be used as the performance metric, considering both typical DVBT signal
and the Gaussian signal model assumed in the derivation of the algorithm. The MSE is defined
as follows:
2
M SE = E (an an ) , n = M.....N (7.10)
where an and an are the nth ideal and estimated feedback loop coefficient, respectively. E is the
statistical average operator. M SE versus LAR with different values are investigated. The
BER performance at the receiver is also simulated for static AWGN and static Rayleigh cases.
1
10
Standard DVBT signal, = 0.5
WGN signal, = 0.5
Standard DVBT signal, = 1
WGN signal, = 1
2
10 Standard DVBT signal, = 2
WGN signal, = 2
3
10
MSE
4
10
5
10
6
10
1 2 3 4 5 6 7 8 9 10
LAR in dB
Fig. 7.8 MSE performance of loop amplitude estimation using enhanced autocorrelation method with
different LAR and .
Case 1 : = 0.5 and LAR = (1, 2, ...10 dB) This setting simulates the worst case, where
the loop amplitudes are slowly decaying. It can be seen that when the LAR is smaller than 4 dB,
the MSEs in cases of standard DVBT signal and WGN signal are very similar. As the LAR is
getting higher, the MSE in the case of WGN signal is decaying more rapidly to a smaller values.
The MSE in the case of DVBT signal when LAR is larger than 7 dB is also getting smaller but
not as well as in the WGN case (Figure 7.8).
Case 2 : = 1.0 and LAR = (1, 2, ...10 dB) Here a moderate case is considered. The loop
amplitude is now more rapidly decaying and the feedback energy is carried by the first few taps.
When the LAR is smaller than 3 dB, the MSE is similar in cases of standard DVBT signal and
WGN signal. It can be seen that the MSE in the WGN case is converging to very small values
when LAR is getting higher than 4 dB. The MSE in the case of the DVBT signal, when the
LAR is larger than 5 dB, is also getting smaller (Figure 7.8).
Case 3 : = 2.0 and LAR = (1, 2, ...10 dB) This setting simulates the standard case, where
the loop is modeled mainly by a single tap. The MSE of the DVBT signal and the WGN signal
cases are converging more rapidly to small values for LAR higher than 4 dB. The MSE perfor
AUTOCORRELATION BASED CANCELLER ALGORITHM 75
mance of the WGN case is always better than the DVBT case mainly for high LAR (Figure
7.8).
In Figure 7.9, and 7.10, the BER is simulated for AWGN and Rayleigh channels [3]. As
can be seen, the autocorrelation method can cancel the loop interference effectively in case of
exponentially decaying multipath loop channel model.
0
10
No loop interference
1 Loop interference + canceller
10
Loop interference + no canceller
2
10
BER after Viterbi
3
10
4
10
8k mode DVBT system
64 QAM, 2/3 code rate
AWGN channel.
Channel loop : LAR (dB)= 3dB, = 1.
5
10
6
10
0 5 10 15 20 25 30
SNR at the DVBT receiver(dB)
Fig. 7.9 BER performance of the enhanced autocorrelation method in the AWGN channel case with
exponentially decaying multipath loop channel model.
0
10
1
No loop interference
10
Loop interference + canceller
Loop interference + no canceller
2
10
BER after Viterbi
3
10
4
10
6
10
0 5 10 15 20 25 30
SNR at the DVBT receiver(dB)
Fig. 7.10 BER performance of the enhanced autocorrelation method in the Rayleigh channel case with
exponentially decaying multipath loop channel model.
76 ENHANCED ALGORITHMS FOR LOOP INTERFERENCE CANCELLATION
c(n)
h(n)
x(n)
y(n) z(n)
h(n) f( y(n), y(n) ) g
y(n)
c(n)
Channel Relay
7.3.1 Introduction
We investigate here the use of a simple antenna diversity scheme in order to overcome the loop
interference problem [P8]. The idea is to select the branch that has less coupling. Frequency
domain analysis of the received signal (coupling + received signal from the transmitter) is first
performed. Instantaneous SNR at each branch is computed and selection of the received carrier
that correspond to the higher SNR is performed. Obviously, because of using frequency domain
analysis, a fixed processing delay that exceeds the system guard interval has to be tolerated.
Therefore, this repeater is intended to be used only for indoor channel environment where the
signals received by the user are only coming from the repeater. The focus is on slowly fading
loop interference environment.
diversity scenario and without any cancellation methods. Some simulation results are given in
Figure 7.12. As can be seen, the use of diversity improves the system performance in the BER
sense. We can also notice that in the case of fading feedback loop, the tolerance of the DVBT
system to loop interference is more limited when compared to the static case.
0
10
1
10
2
10
No diversity, fd= 2 Hz
6
10
7
10
20 18 16 14 12 10 8 6 4
Average loop amplitude ratio
Fig. 7.12 BER performance of the DVBT repeater with diversity: DVBT, 2k mode, 16QAM, 2/3 code
rate, variable LAR and Doppler frequency cases.
In the subsection 6.3.1 we presented the frequency domain adaptive cancellation algorithm [36].
We noticed that the algorithm is limited in the presence of severe channel frequency selectivity
that may occur between the main transmitter and the repeater.
We propose here an enhanced version of the algorithm. We divide the algorithm in two parts.
First, we start with an initialization step at the startup of the the repeater operation. During
the the initialization period, the repeater is turned off (g = 0), and an estimate of the multipath
channel between the transmitter and the relay is carried out. The multipath channel estima
tion is carried out, like in standard DVBT receiver, using scattered pilots transmitted within the
OFDM symbols. Performing the estimation of the channel while the repeater is turned off will
assure an accurate estimate due to the absence of the coupling effects. During the initialization
step the entire system transfer function (Figure 6.6) is given by
S(z)
= H(z). (7.17)
X(z)
The actual estimate of the channel will be saved in order to be used in the normal working
state of the repeater. Here, we are assuming that the multipath channel is static or slowly chang
ing in time. Therefore, only a small number of OFDM symbols are needed in the start up to
perform the channel estimation.
ENHANCED LMS LOOP INTERFERENCE CANCELLATION 79
Secondly, when the initialization period is completed, we switch to the standard mode of
the algorithm as described in the subsection 6.3.1. Therefore, the transfer function of the entire
system will be given by
F (z) = F (z)H(z). (7.18)
Here, before we estimate the error function Er (z), we first equalize F (z) with the estimate
of the channel H(z) so that we have the same form of the error function as in the subsection
6.3.1. Figure 7.13 shows the performance of the enhanced frequency domain adaptive algorithm
versus the standard one. The case of one tap loop with LAR = 1dB was considered. As it
can be seen, the enhanced algorithm can cancel the loop effectively even in the presence of a
Rayleigh channel between the main transmitter and the repeater.
0
10
1
10
2
10
BER after Viterbi
3
10
4
10
No loop interference
With Canceller: Enhanced algorithm
5
10 With Canceller: Standard algorithm
6
10
10 12 14 16 18 20 22 24 26 28
SNR at the DVBT receiver (dB)
Fig. 7.13 BER performance of the enhanced frequency domain adaptive algorithm for singletap loop
with LAR of 1dB. Rayleigh channel is assumed between the main transmitter and the relay. 64QAM
and a code rate of 2/3 are considered.
The main drawback of the LMS based loop interference canceller [49] presented in the subsec
tion 6.3.2 was in its limited stability. We propose here an enhanced version of the algorithm to
overcome this limitation.
The steady state of the LMS loop interference canceller algorithm can be enhanced by using
both power and phase information of the signal at the output of the repeater in the cost function
to be minimized.
In the following the main equations for LMS coefficient updating are given:
w(n) = w(n 1) + e(n)s (n) (7.19)
2 jarg(s(n))
e(n) = (s(n) Pe )e (7.20)
where s(n) is the repeater transmitted signal, w(n) is the vector of the LMS coefficients, n is
the time index, is the step size (0 < 1) and Pe is the expected received power.
80 ENHANCED ALGORITHMS FOR LOOP INTERFERENCE CANCELLATION
Figure 7.14 shows the absolute value of the error e(n) versus time, in the case where Pe is
measured and when it is assumed to be constant of unit power. In these two cases, the phases
of the signal s(n) is used in the error signal e(n). We also included in the Figure 7.14 the case
where only the power of the signal s(n) is used as in the standard LMS interference canceller
algorithm. As can be seen, in both cases where the phase information is used, the steady state
of the LMS algorithm presents smaller absolute error values which means better stablity of the
algorithm.
1.4
Old LMS cancellation: Only the power of the signal at the output of the repeater is used
New LMS cancellation: Constant power and phase information are used
1.2
New LMS cancellation: measured received power and phase information are used
0.8
0.6
0.4
0.2
0
0 50 100 150 200 250 300 350 400 450
Time in samples
Fig. 7.14 The absolute value of the error function e(n): different LMS loop interference cancellation
schemes, one tap loop interference model with LAR of 2dB, 64QAM, and a code rate of 2/3.
To evaluate the BER performance of the enhanced LMS adaptive loop interference cancella
tion algorithm, we consider the case of Rayleigh channel (P 1 = 0) and a one tap loop model.
The relay gain parameters are set as follows (we refer here to the Figure 6.8) :
Initial gain(dB) = 0 (0 time A1)
F inal gain(dB) = 15 (A2 time)
. (7.21)
A1 = 150 ms (150 OF DM symb.)
A2 = 250 ms (250 OF DM symb.)
Again, we use parameter values of Table 6.1 for the loop interference model. Here we as
sume that Pe is measured. The BER simulation results are depicted in Figure 7.15. A multitap
loop model case is also simulated. We use the profile of fixed reception defined in [3] to model
the loop interference. Therefore, the taps k in C(z), are following the Ricean distribution with
Ricean Factor P 2 = 10 (Table A.1). The LAR value of 2 dB is simulated. The BER per
formance is shown in Figure 7.16. As can be seen, in both cases the enhanced adaptive LMS
algorithm cancels effectively the loop interference.
7.6 CONCLUSIONS
Three new algorithms for loop interference cancellation have been introduced. Their perfor
mance and limitations were discussed. Extensions and enhancements of some existing ap
CONCLUSIONS 81
0
10
1
10
2
10
3
10
No loop interference
With canceller
No canceller
4
10
5
10
5 10 15 20 25 30 35
SNR at the DVBT receiver ( dB )
Fig. 7.15 BER in adaptive LMS loop interference cancellation case: one tap loop interference model
with LAR of 1dB, 64QAM, and a code rate of 2/3.
0
10
No loop interference
With canceller
No canceller
1
10
2
10
BER after Viterbi
3
10
4
10
5
10
5 10 15 20 25 30 35
SNR at the DVBT receiver ( dB )
Fig. 7.16 BER in adaptive LMS loop interference cancellation case: multitap loop interference model
with LAR of 2dB, 64QAM, and a code rate of 2/3.
82 ENHANCED ALGORITHMS FOR LOOP INTERFERENCE CANCELLATION
proaches were also presented. The algorithms are characterized by different structures and
implementations, depending on the loop interference model, i.e., one tap loop, or multitap loop,
and on the channel frequency selectivity between the transmitter and the repeater (Ricean or
Rayleigh).
These algorithms were studied and simulated for different channel and loop interference pa
rameters. The autocorrelation based algorithm is very simple and efficient in the case of one
tap loop model. It can cancel the loop interference effectively even if a severe frequency se
lective channel is present between the main transmitter and the receiver. This is an advantage
when compared to the existing adaptive frequency domain loop canceller algorithm. However
for a multitap loop model case, the performance of the autocorrelation method is limited. An
extension version of the aucorrelation method was developed to deal with the exponential mul
titap loop model case. The enhanced algorithm cancels the loop interference effectively in this
condition. The autocorrelation based algorithm is cancelling the loop interference in time do
main. Due to its small processing delay this algorithm is recommended to be used for indoor
and outdoor gap fillers where the loop can be modeled by one tap loop or exponential delay
profile.
A new frequency domain algorithm was introduced. It uses the diversity technique to se
lect the branch presenting less coupling. The BER performance of this method shows that it
can improve the loop cancellation effectively when compared to no diversity case. Due to its
frequency domain processing, a larger delay is needed when compared to the autocorrelation
method. This technique is therefore recommended to be used only for indoor gap filler where
the signal at the receiver side is assumed to be coming only from the repeater and not from the
main transmitter.
The enhanced versions of the existing algorithms like the LMS based algorithm improve
substantially the stability and the performance of the existing methods.
Chapter 8
DVBT Signal in Cable TV
Networks: Limitations and
Requirements
8.1 INTRODUCTION
Following its revolutionary steps towards digital technology, the further evolution of the DVB is
aimed at digital integrated broadcasting, in which video and audio signals and data services are
transmitted via terrestrial, satellite, and cable communication networks with open, transparent
interfaces [53] [54]. Therefore, interoperability issues of DVBT system should be examined
and compatibility of the existent standard with other DVB broadcasting services has to be fur
ther investigated. As a case study, compatibility of the DVBT signal and DVBC is addressed
[55]. We study the quality of DVBT signal transmission over the cable TV network. Gener
ally, when terrestrial digital TV signals are distributed in the cable TV, a conversion is needed
in the headend, from DVBT to DVBC. DVBC uses conventional single carrier QAM trans
mission, so the signal processing functionalities are quite different from DVBT; however, the
error control coding and higher layers have a lot of commonality with DVBT. In order to avoid
this costly conversion and to transmit DVBT signal directly in the existent cable channel, var
ious requirements have to be satisfied. It is known that phase noise represents one of the main
limitation for OFDM based systems compared to singlecarrier QAM transmission. We demon
strate the sensitivity of the OFDM system for the phase noise effects by using a practical model
for CATV channel [P9]. Then, we conclude by giving the specifications for the CATV network
to allow DVBT transmission with sufficient quality [P10][P9].
Cable television, originally known as Community Antenna Television or CATV, originated in the
late 1940s as a way to bring television broadcasts to people in remote areas where overtheair
reception was poor [56] [57]. Pay television was introduced in the early 1970s, and the cable
TV industry that we know today was born. In the following years, the use of satellites to trans
83
84 DVBT SIGNAL IN CABLE TV NETWORKS: LIMITATIONS AND REQUIREMENTS
mit television programs made more content available to cable network operators. Operators
expanded their services to include new channels dedicated to music, sports, news, childrens
programming, and other specialized content. Governmental regulation shaped the development
of the cable television industry. In 1992 , after a revision of some FCC technical rules, a pres
sure for the industry has been made to improve quality of service and control prices. Then the
wireline and wireless communication service markets were opened allowing cable TV operators
to become players in more than just video entertainment. Today, cable network operators face
intense competition from other industries, including direct satellite system (DSS) television and
telecommunications. In response, cable operators are evolving their businesses to supply more
than just packaged entertainment [57]. Cable operators are upgrading their coaxbased facili
ties with fiberoptic technology and introducing digital communication servicesincluding high
speed Internet access, telephony, data, and interactive television. The combination of broadband,
hybrid fibercoax infrastructure and highspeed cable modems is enabling cable TV operators
to offer subscribersaccess to digital and interactive services at speeds hundreds of times faster
than traditional telephone lines [56].
A cable TV operator receives (via satellite dish, antenna, or fixed broadband networks) many
signals containing programmes that are transmitted, e.g., by the national broadcast networks,
by large independent stations, by the many cable networks and services, and by local televi
sion studios [57]. These signals are received at the cable systems headend, where they are
amplified, processed, and converted to the signal formats and frequencies used by the cable TV
operator to carry the programmes through a network of coaxial or fiberoptic cables to a sub
scribers television or settop box. A variety of equipment is used at the headend, including
antennas and satellite dishes, preamplifiers, frequency converters, demodulators, and modula
tors, processors, and scrambling and descrambling equipment. The cable distribution system
carries the multiple channels of programmes to the network subscribers.
As Figures 8.1 and 8.2 illustrate, this system generally consists of a trunk cable or fiber, which
carries signals from the cable headend to the center of a town or urban area; feeder cables or
fibres, which carry the signals from the trunk out into the neighborhoods; taps, which draw off
a portion of the feeder line cable for the subscriber; and drop cables, which connect individual
subscribers television sets to the feeder lines. Equipment used in the cable distribution system
includes trunk amplifiers (used to maintain signal strength and compensate for cable losses),
bridger amplifiers, power supplies, and other electronic and fiberoptic components. At the sub
scriber premises, the drop cable may be connected directly to a "cable ready" television set or
connected first to a settop box. Settop boxes provide frequency and channel conversion for
older television sets and descrambling of pay service signals. The drop cable may also be con
nected to other interactive equipment, including telephony terminal equipment or highspeed
cable modems with return capability via a telephone line or twoway cable.
In the following we will focus on the cable TV channel modeling. The impairments that
can affect the OFDM signal will be analyzed. A model will be finally provided, to be able to
evaluate the performance of DVBT signal in a CATV network.
CABLE DISTRIBUTION SYSTEM 85
Satellite Reception
Trunk Cables
Local Signals
HeadEnd
Trunk Amplifiers
Distant Signals
Line Extender
Amplifiers
Drop
Bridger Amplifier
Taps
8.4.1.1 Gaussian noise In CATV systems, the thermal noise, which is caused by the
random motion of the charged particles, is considered to be Gaussian distributed.
.
.
. .
+
. .
. .
. .
To analyze the effect of the Gaussian noise on the OFDM signal [58], lets consider noisy
samples, x(n), of a received baseband OFDM signal defined by
x(n) = IDF T {X(k)} + d(n). (8.1)
By using the linearity property of the Fourier transform, the demodulated noisy signal, X (k),
is given by
n o
X (k) = DF T x(n) = X(k) + D(k), (8.2)
where D(k) is the DFT of d(n) which is also Gaussian with the same variance.
A simple model for the AWGN noise impairment is shown in Figure 8.3.
8.4.1.2 Linear distortion The CATV system is simply a cascade of three main parts:
Trunk, distribution and drop parts. Each of these parts is characterized by different cable pro
prieties. Homes are connected to the cable system by flexible drop cables, typically 50 meters
long. In the distribution part, the cable is tapped so that flexible drop cables can be connected
to it and routed to the residences.
Signal reflections occur through the cable plant and are called microreflections. They are
caused by individual slight errors in impedance match. The severity of the mismatch is measured
by the magnitude of the returnloss ratio.
In general, the cable TV system is invariant in time and a finite impulse response linear sys
tem can be used to model the cable attenuations and reflections. Therefore, we can investigate
the OFDM signal performance through a channel having discrete transfer function of the form:
GENERIC CATV CHANNEL MODEL 87
M
X
H(z) = am z m , (8.3)
m=0
where am are the channel tap coefficients and M is the order of the FIR filter. If the transmitted
signal is
N
X 1
x(n) = IDF T (X(k)) = X(k)ei2kn/N , (8.4)
k=0
For error free transmission, and no equalization the following equation must hold [58].
M
dmin X
i2mk/N
> max(X(k) am e ), (8.6)
2
m=1
where a0 is normalized to 1, and dmin is the minimum distance of the used constellation. This
expression can be manipulated further to give us the sufficient condition for error free trans
mission in the noise free case. By applying the triangle inequality we will have the following
sufficient condition
M
dmin X
> max(X(k)) am . (8.7)
2 m=1
8.4.1.3 Nonlinear distortion The OFDM signal carried by a large number of subcarriers
is very sensitive to nonlinear distortion because of its greatly variable envelope, which depends
on the instantaneous phase value of each carrier [59] [60] [61]. Due to the central limit theo
rem, the complex baseband OFDM signal can be modeled (for a high number of independently
modulated carriers) with a complex Gaussian process having Rayleigh envelope distribution
[6]. This allows the analytical treatment of nonlinear OFDM systems making use of the more
general results on the effects of nonlinear distortions of Gaussian signals [62] [63].
Analogtodigital (A/D) converters, mixers, and power amplifiers in the CATV systems are
usually the major sources of nonlinear distortions due to the limited range that they allow for
signal dynamics. It is possible to distinguish between two different classes of nonlinear distor
tion: the first, named cartesian, acts separately on the real and imaginary baseband components
of the complex signal, while the second acts on the envelope of the complex signal. The socalled
cartesian clipping belongs to the first class [57]. While AM/AM (amplitude distortion which
depends on the amplitude of the input) and AM/PM (phase distortion which depends on the am
plitude of the input) introduced by power amplifiers belong to the second class [60] [59]. In the
CATV system, mainly the nonlinear distortion due to power amplifiers is present, so in the fol
lowing sections we will concentrate our analysis on the second class of the nonlinear distortion
[57].
We consider the modulation of a complex symbols X(k) by Npoint inverse discrete Fourier
transform (IDFT). The result, x(n), is distorted by the transfer characteristic function g(x) that
88 DVBT SIGNAL IN CABLE TV NETWORKS: LIMITATIONS AND REQUIREMENTS
models the nonlinearity. Using a simplified model of Figure 8.4, the distorted samples are
demodulated using FFT to get the subcarrier samples Y(k).
. .
. .
. .
. .
. .
To analyze the effect of the nonlinearity on the subcarrier samples Y(k), we consider the
error caused by the channel nonlinearity that can be defined by:
N 2 2
2 = (r + i2 ) = N tot . (8.9)
2 2
A polar representation of the complex signal x(n) is given as follow
where the phase, arg(x(n)), is uniformly distributed between 0 and 2. The magnitude of
x(n) is Rayleigh distributed. Lets consider the softlimiting nonlinearity that can be defined as
follows:
L if x L
g(x) = x if L x < L (8.11)
L if xL
and
x x2 /22
p(x) = e , x0 . (8.15)
2
The error variance can then be expressed as
2 2 L
E e2 (n) = 2 2 eL /2 2 2LQ( ),
(8.16)
where we have used the Qfunction notation. In the receiver, the pdf of the symbol error ap
proaches again, zero mean normal distribution with error variance [58],
2 1 2
= E E 2 (k) = E e2 (n) = tot
2 2
eb 2 btot
E Q( 2b), (8.17)
N
here b it is the input backoff factor
L
b= , (8.18)
tot N
which defines the ratio of the minimum input signal power to yield amplifier saturation level
and the average input power.
The equivalent system model for a soft limiting channel can be simplified to be as addition
2
of the input signal with a Gaussian distributed noise having E as variance.
The power amplifiers in general introduce nonlinear distortions in both amplitude (AM/AM
conversion) and phase (AM/PM conversion) [59]. The effect of phase distortion has been in
vestigated in [59] [66] and found to be significant only at very low power levels. The noise due
to nonlinearity is primarily determined by the amplitude distortion. Thus, the earlier analysis of
the soft limiter is valid for modeling the intermodulation effects due to amplifier nonlinearity
in CATV systems.
8.4.1.4 Phase noise For conventional Analog TV systems (like PAL, SECAM and
NTSC), the phase noise of the frontend oscillators does not represent a big problem [67] [68]
[69]. With the introduction of the OFDM system, the significance of the phase noise increases
very strongly.
The main difference between OFDM and other digital modulation methods is that the OFDM
signal consists of many low rate modulated carriers that are separated from each other with the
aid of DFT. The low symbol rate makes the synchronization more difficult when fast phase dis
turbances occur. On the other hand, the phase noise leads to a nonorthogonality of the carriers
because of the leakage of the DFT. The origin of the phase noise is mainly due to the use of
practical oscillators that suffer from a random perturbation of the phase of the steady sinusoidal
waveform [70] [71]. Practical modulators and demodulators usually work either at baseband or
at a convenient intermediate frequency (IF) [67]. As we must transmit our signal at some allo
cated radio frequency (RF), it follows that in practice we must shift our modulated signal up to
RF in the transmitter, and down from RF to IF or baseband in the receiver. To do this, we must
use practical oscillators, whose phase noise will be imparted to the signal we convey.
90 DVBT SIGNAL IN CABLE TV NETWORKS: LIMITATIONS AND REQUIREMENTS
To analyze the effects of the phase noise on the OFDM signal [72] [70], lets suppose that
the complex envelope of the transmitted OFDM signal, for a given OFDM symbol, is:
N
X 1
x(n) = X(k)ei2kn/N , n = 0, 1, ....., N 1 (8.19)
k=0
This symbol is actually extended with a cyclic prefix in order to cope with multipath delay
spread. For the sake of simplicity, we will not consider this prefix since it is eliminated in the
receiver.
If we assume that the channel is flat, the signal is only affected by phase noise impairment
(n) at the receiver. The received signal r(n) can be expressed as follows:
NP
1
1
Y (k) = N r(m)ej2km/N =
m=0
NP
1 NP
1
1
= N ej(m) X(r)ej2rm/N ej2km/N = (8.21)
m=0 r=0
NP
1 NP
1
1
= N X(r) ej(m) ej2(rk)m/N .
r=0 m=0
In order to separate the signal and noise terms, let us suppose that (m) is small, so that
ej(m) 1 + j(m)
In this case:
NP
1 NP
1 NP
1 NP
1
1 j
Y (k) = N X(r) ej2(rk)/N + N X(r) (m)ej2(rk)m/N =
r=0 m=0 r=0 m=0
N 1 NP
1
(8.22)
j P j2(rk)m/N
= X(k) + N X(r) (m)e = X(k) + e(k).
r=0 m=0
We obtain an error term e(k) for each subcarrier, which results from some combination of
all carriers and is added to the useful signal. If we analyze more deeply this noise contributions,
we can observe that:
1. If r = k
N 1 N 1 N 1
j X X X(k) X
e(k) = X(r) (m) = j (m) = jX(k)0 . (8.23)
N r=0 m=0
N m=0
So we have a common error added to every subcarrier, which is proportional to the symbol
value multiplied by a complex number j0 that is a rotation of all the carriers by the same angle
simultaneously. This means that if the rotation in a given symbol can be measured using some
carriers, which bear reference information, it is then possible to correct the remaining carriers
in the symbol. This part of the phase noise is known as common phase error [73].
2. If r 6= k
GENERIC CATV CHANNEL MODEL 91
N 1 N 1
j X X
e(k) = X(r) (m)ej2(rk)m/N . (8.24)
N m=0
r=0
r 6= k
This term corresponds to the summation of the information of the N 1 subcarriers each
multiplied by some complex number, which comes from an average of phase noise with a spec
tral shift. The result is also a complex number, which is added to each subcarriers useful signal
and has the appearance of white noise. It is normally known as intercarrier interference (ICI) or
loss of orthogonality [73].
To model the phase noise impairment, a phase noise mask of typical oscillators is usually
needed. Based on the given mask, the simulation of the system performance in the presence of
the phase noise is easily carried out [72]. In the Figure 8.5 an example of a phase noise mask,
which is derived from a typical oscillator, is shown. It is generally plotted along the offset fre
quency. The amplitude is given in dBc/Hz and means the noise power in 1 Hz bandwidth related
to the total carrier power. It is possible to observe the common phase error behaviour and the
white noise like behaviour of the phase noise, in the low and high frequency values of the phase
noise mask, respectively
10
20
50
60
70
80
90
0 0.5 1 1.5 2 2.5
Frequency offset (Hz) 4
x 10
To study the performance of the OFDM transmitted signal over the CATV system, and based
on the earlier analysis, a generic channel model is built, including, almost all the possible im
pairments that can affect the transmitted signal. A practical CATV system includes a number of
cable and fiber sections with amplifiers, multiple mixers for up/downconversion, etc. Thus, a
complete model would include multiple instances of each distortion type, in a more or less arbi
trary order. However, in a practical system, each of the effects is relatively mild, and coupling
92 DVBT SIGNAL IN CABLE TV NETWORKS: LIMITATIONS AND REQUIREMENTS
E2
+
FIR = (a0 , a1 ,.., aM )
2
D
effects of different types of distortions are not expected to be significant. Thus we can simplify
the model to include a single instance of each of the distortion types, and basically analyze each
distortion type independently of the others. This is summarized in Figure 8.6.
Table 8.1 The multipath taps that model the linear distortion in the CATV channel
20
25
30
35
40
0 50 100 150 200 250 300 350 400 450 500
Delay with respect to the main impulse (ns)
10
20
40
50
60
70
80
0 1 2 3 4 5 6 7 8 9 10
Frequency Hz 4
x 10
Fig. 8.8 The measured spectrum of the signal coming out from the actual converter used in CATV
network.
here SPdB is the scaling parameter in dB, LP SdBc is the level of the phase noise used in the
simulation model in dBc and LP MdBc is the level of the phase noise concluded by the mea
surements in dBc (fixed value). Therefore, increasing the scaling parameter will be understood
as using a better converter.
When the measurement results are shown on a logarithmic scale in frequency domain, as
illustrated in Figure 8.9, it is possible to see that the nearly 0 dB power level extends up to about
1 kHz. This implies that there is a frequency drift in the converter during the measurement. Of
course this problem does not necessarily harm the operation of the system in practice, since the
drift can be followed, at least partly, by the carrier synchronization circuitry of DVBT receivers.
However, the measurement results are affected by this phenomenon. This can be taken into
account by just correcting the spectrum at small frequency offsets by limiting it to the maximum
value of 30 dBc. .
Our aim here is to study the quality of the DVBT transmission over the cable TV network.
Usually, DVBT signals are converted in the headend to DVBC format [2] when distributed in
the cable TV channel.
We consider the possibility of avoiding this costly conversion and to transmit DVBT signal
directly in the existent cable channel. Firstly, we study the sensitivity of the DVBT for each
impairment parameters characterizing the CATV channel, like linear distortion and phase noise.
Secondly, a performance comparison against DVBC is also carried out. There are two major
differences between DVBT and DVBC. The DVBC [2] (see Figures 8.10 and 8.11) is us
ing a single carrier technique based on Quadrature Amplitude Modulation (QAM). The second
difference is that DVBC is not using inner coding. To achieve the appropriate level of error
SIMULATION OF THE DVBT SIGNAL OVER THE CATV CHANNEL 95
10
20
40
50
60
70
80 2 3 4 5
10 10 10 10
Frequency Hz
protection required for the transmission of a digital signal in cable TV channel, a forward error
correction based only on ReedSolomon encoding is used. In most cases, the DVBC system
utilizes 64QAM, but lowerlevel systems, such as 16QAM and 32QAM, and higherlevel sys
tems such as 128QAM and 256QAM can also be used. In each case, the data transmission
capacity of the system is traded against robustness.
Because of these differences, and to assure a fair comparison between the two systems we
will try to choose the systems, parameters so that the capacity is almost the same.
0
10
1
10
2
10
BER after Viterbi
3
10
4
10
AWGN, QPSK
AWGN, 16QAM
5 AWGN, 64QAM
10
CATV channel, QPSK
CATV channel, 16QAM
CATV channel, 64QAM
6
10
2 4 6 8 10 12 14 16 18 20 22
SNR in dB
Fig. 8.12 Sensitivity of DVBT to the linear distortion with 2/3 code rate.
Based on Figure 8.12, we can state that the linear distortion in the CATV channel degrades
the performance of the DVBT system mildly when compared to the AWGN channel. About 3
dB higher SNR is needed for each DVBT system to achieve a sufficient performance for each
submodulation.
2
BER after Viterbi 10
3
10
4
10
5
10
0 1 2 3 4 5 6 7 8
SNR in dB
Fig. 8.13 Sensitivity of DVBT to the phase noise with different levels of scaling parameters: QPSK,
2/3 Code rate.
2
10
BER after Viterbi
3
10
4
10
5
10
6
10
8 9 10 11 12 13 14 15 16
SNR in dB
Fig. 8.14 Sensitivity of DVBT to the phase noise with different levels of scaling parameters: 16QAM,
2/3 Code rate.
98 DVBT SIGNAL IN CABLE TV NETWORKS: LIMITATIONS AND REQUIREMENTS
1
10
2
10
4
10
Scaling = 2 dB
Scaling = 4 dB
5 Scaling = 6 dB
10
Scaling = 8 dB
Scaling = 10 dB
AWGN channel
6
10
14 15 16 17 18 19 20 21 22
SNR in dB
Fig. 8.15 Sensitivity of DVBT to the phase noise with different levels of scaling parameters: 64QAM,
2/3 Code rate.
1
10
2
10
DVBC 16QAM
BER
3
10
CR=3/4, Scaling = 8 dB
CR=3/4, Scaling = 10 dB
4
10
CR=2/3, Scaling = 8 dB
CR=2/3, Scaling = 10 dB
5
Scaling = 6 dB
10
Scaling = 8 dB
Scaling = 10 dB
6
10
5 10 15 20 25 30
SNR in dB
Fig. 8.16 Sensitivity of DVBT and DVBC to the phasenoise scaling parameter.
DVBT signal over the current CATV networks without any conversion from DVBT to single
carrier DVBC system. The results help in the design and the optimization of efficient OFDM
systems for cable transmission.
Chapter 9
Conclusions
The DVBT system for terrestrial broadcasting is probably the most complex DVB delivery sys
tem. Originally, the DVBT standard was created for fixed and portable reception. As expected
it has proven a worldwide success. However, to remain successful, continual assessment and
enhancement are needed to mitigate any perceived deficiencies in standard performance.
The thesis addressed enhancements of the DVBT system. Its results come as a participation
of the DVBT standard revision in order to fully support the new service scenarios. The thesis
considered three main issues.
In Chapters 2 and 3 we overviewed the principles of OFDM technique and DVBT technol
ogy.
In Chapter 4 we began the first issue. The tolerance of the DVBT signal to impulsive in
terference was studied and various techniques to mitigate impulse noise were introduced. We
started by discussing the main impulse noise models and sources. Many existing approaches
to combat the impulsive noise have been presented. Generally, those approaches are based on
clipping the impulsive samples. The clipping methods are very simple and useful but they leave
moderate impulsive levels untouched which mean that their capabilities are quite limited. Other
approaches closely related to clipping are based on blanking all impulsive samples which known
to be corrupted. Clearly, the detection of the position of the impulsive samples is needed. In
Chapter 4 we introduced a new method to compensate for impulsive noise in the frequency do
main. The performance analysis of the new algorithm was presented. An example system was
studied and simulated. It was shown that the algorithm can cancel effectively the impulsive
interference mainly in AWGN channels.
In Chapter 5 we tried to enhance the performance of the algorithm introduced in Chapter 4
in timefrequency selective channels. The performance of combining this algorithm with other
existing impulse noise mitigation schemes was investigated. An enhanced channel estimation
scheme that is used with the pilot impulse noise canceller was introduced. The scheme used
the impulse free pilots to estimate the corrupted ones. The bit error rate performance showed
that those techniques improve the tolerability of the DVBT signal to impulsive noise even in
severe channel conditions. In Chapter 5 we also studied the effects of impulsive noise on the
performance of the maximum likelihood symbol synchronization (MLSS) algorithm in OFDM
systems. The limitations of the MLSS algorithm were shown. The simulations results of the
timing offset mean squared error (MSE) in static and mobile channels were considered. New
ideas to enhance the performance of the algorithm in these environments were also proposed.
101
102 CONCLUSIONS
In Chapter 6 we reviewed the main existing algorithms dealing with loop interference can
cellation in digital TV systems using gap fillers for coverage extension and we studied their
performance and limitations. A particular problem with the use of gap fillers in DVBT sys
tems, is related to the coupling (loop interference) between transmitter and receiver antennas at
the relay station. The coupling causes oscillations in the repeater and distorts the repeated sig
nal. The loop interference must be reduced to an allowable level in order to avoid distortion and
oscillation problems. Beside spatial solutions, not many measures have been proposed in the
literature.
In Chapter 7, new techniques have been introduced. A simple time domain canceller based
on the autocorrelation method was designed. The algorithm shows to cancel very effectively
the loop interference when only one tap loop model is assumed. An enhanced version of the
same algorithm was also suggested. The multitap loop case having an exponential profile was
considered. The Bit error rate results show that the enhanced version cancel efficiently the loop
interference in these conditions. In addition we investigated the use of antenna diversity in
DVBT indoor repeaters. The case of slowly mobile loop interference environment was stud
ied. The diversity approach seems to relatively improve the performance of the DVBT system
compared to the case where no diversity is used. In Addition we considered some enhance
ments of the existing algorithms. We showed that these enhancements improve substantially the
stability and the performance of these methods.
The third issue addressed in this thesis was the transmission of the DVBT signal in CATV
networks. Following its revolutionary step towards digital technology, the further evolution of
the DVB is aimed at digital integrated broadcasting in which video and audio signals and data
services are transmitted via terrestrial, satellite and cable communication networks with open,
transparent interfaces. Therefore interoperability issue of DVBT system should be examined
and compatibility of the existent standard with other DVB broadcasting services has to be further
investigated.
In Chapter 8, the interoperability issue was addressed. As an example, the compatibility of
the DVBT and DVBC was investigated. We studied the quality of DVBT transmission over
the cable TV network. Generally, when terrestrial digital TV signals are distributed in cable
TV, a conversion is needed in the headend, from DVBT to DVBC (single carrier). In order to
avoid this costly conversion and to transmit DVBT signal directly in the existent cable channel,
many requirements have to be satisfied. It is known that phase noise represents one of the main
limitations for OFDM based systems. In this chapter we demonstrated the sensitivity of the
OFDM system for the phase noise effects by using a dynamic model for the CATV channel. We
concluded by giving the specifications which a CATV network should satisfy to allow DVBT
transmission with sufficient quality.
The emergence of new consumer applications like mobile TV and the convergence of various
wireless technologies are leading the DVB community to continually evaluate the suitability
of the existing DVBT system to accommodate such emerging situations and to consider the
benefits that new stateoftheart technologies could bring to DVBT.
In this thesis we studied some aspects related to the effects of impulse noise and possible
solutions to such problem. However, more robust methods that can deal with this impairment
and improve the reception difficulties of DVBT in mobile environments are still needed. For
instance, these difficulties can occur when Hybrid DVBH/GSM mobile broadcast system is
FUTURE RESEARCH 103
considered. For the DVBT(H) receiver the interference caused by the GSM power amplifier
noise looks like very high power and long duration interference. Therefore, new techniques and
algorithms need to be designed for such scenarios.
The gap filler issue remains also one of the main challenges facing the design of DVBT(H)
networks. The feedback loop interference needs to be studied further and low complexity loop
interference cancellers need to be developed.
The DVBT standard enhancement is still going on. We already start discussing about the
DVBT2 second generation. Questions about new techniques and targets that the new standard
will achieve are being studied. A large number of techniques have been proposed. The intended
enhancements will improve capacity, robustness and flexibility of the DVBT.
In our group, we are also studying the enhancement of the DVBT receiver from the mobility
sense. New ICI canceller schemes are being proposed. Additionally, optimization of the pilot
pattern in DVBT is also under consideration. The optimization is aimed at higher capacity and
larger Doppler tolerance.
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Appendix A
Definition of the Channel Models
Used in Performance Evaluations
Table A.1 Relative power, phase and delay values for static Ricean and Rayleigh channels
111
112 DEFINITION OF THE CHANNEL MODELS USED IN PERFORMANCE EVALUATIONS
Table A.2 Characteristics of the 6tap typical urban (TU6) channel model