Hazmi

Julkaisu 668 Publication 668
Ali Hazmi
Studies in Digital TV Signal Processing: Impulse Noise Mitigation,
Repeater Loop Interference Cancellation, and DVB-T Transmission
in CATV Networks
Tampere 2007
Tampereen teknillinen yliopisto. Julkaisu 668
Tampere University of Technology. Publication 668
Ali Hazmi
Studies in Digital TV Signal Processing: Impulse Noise

Mitigation, Repeater Loop Interference Cancellation, and
DVB-T Transmission in CATV Networks
Thesis for the degree of Doctor of Technology to be presented with due permission for
public examination and criticism in Tietotalo Building, Auditorium TB111, at Tampere
University of Technology, on the 15th of June 2007, at 12 noon.
Tampereen teknillinen yliopisto - Tampere University of Technology

Tampere 2007
ISBN 978-952-15-1787-7 (printed)
ISBN 978-952-15-1878-2 (PDF)
ISSN 1459-2045
i
Abstract
The Digital Video Broadcasting Project (DVB) is a multinational initiative to standardize digital
broadcasting worldwide. It produces different system specifications including satellite: DVB-S,
cable: DVB-C, terrestrial: DVB-T, and others. The DVB-T 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 DVB-T 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 (ATSC-8VSB and DiBEG-ISDBT), and
aims to target new potential services based on hand-held battery powered devices such as mo-
bile telephones, PDAs, etc., have motivated further improvements of the standard. This thesis
addresses enhancements of the DVB-T system. Three main issues are considered.
In the first part, we investigate the tolerance of the DVB-T system to impulsive noise. Impulse
interferences can be produced by ignition sparks from vehicles or various household appliances.
DVB-T 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 DVB-T 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
time-frequency 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 DVB-T receiver
synchronization are also discussed.
In the second part, we introduce loop interference cancellation algorithms to cope with cou-
pling problems in gap-fillers. In the terrestrial digital video broadcasting system, the gap-fillers
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 frequency-domain 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 multi-path loop inter-
ference with exponential profile are analyzed. Finally, an adaptive LMS based loop interference
cancellation algorithm is investigated.
In the last part, inter-operability issue of the DVB-T standard is considered. We study the
quality of the terrestrial digital video broadcasting (DVB-T) 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 head-end from DVB-T to DVB-C. We study the possibility of using
the DVB-T 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 DVB-T transmission
with sufficient quality.
Preface
The research work for this thesis has been carried out during the years 2001-2007 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 co-author Lic. Tech. Jukka Rinne for numerous
hours of working together and for fruitful technical discussions.
I wish to thank also my co-author 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. Elena-Simona 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
2 OFDM: Orthogonal Frequency Division Multiplexing 7

2.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
2.2 OFDM principles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
2.3 OFDM signal . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 8
2.3.1 The definition of OFDM signal . . . . . . . . . . . . . . . . . . . . . . 8
2.3.2 Guard interval . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
2.3.3 Single frequency networks . . . . . . . . . . . . . . . . . . . . . . . . 9
3 Digital Video Broadcasting: Technical Overview 11

3.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
3.2 General considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
3.3 Source encoding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
3.3.1 Video encoding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
3.3.2 Audio encoding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
3.3.3 MPEG-2 transport stream . . . . . . . . . . . . . . . . . . . . . . . . 13
3.4 DVB-T: Physical layer . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
3.4.1 Multiplexing and channel coding . . . . . . . . . . . . . . . . . . . . . 14
3.4.2 OFDM frame structure . . . . . . . . . . . . . . . . . . . . . . . . . . 17
3.4.3 Reference signals . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
3.5 DVB-T receiver simulation model . . . . . . . . . . . . . . . . . . . . . . . . 18
3.5.1 Simulation environment . . . . . . . . . . . . . . . . . . . . . . . . . 18
v
vi CONTENTS
3.5.2 Channel modeling . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19

3.6 Basic aspects of DVB-T networks . . . . . . . . . . . . . . . . . . . . . . . . 20
3.6.1 Conventionally planned networks . . . . . . . . . . . . . . . . . . . . 20
3.6.2 Single frequency networks (SFN) . . . . . . . . . . . . . . . . . . . . 20
3.6.3 Gap-filler . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
3.7 Other DVB standards . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
3.7.1 DVB-H . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
3.7.2 DVB-S . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
3.7.3 DVB-C . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
4 Impulsive Noise Modeling, Effects and Mitigation in DVB-T systems 23

4.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
4.2 Impulse noise modeling . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
4.2.1 Impulse noise sources . . . . . . . . . . . . . . . . . . . . . . . . . . 23
4.2.2 Impulse noise statistical models . . . . . . . . . . . . . . . . . . . . . 24
4.2.3 Impulse noise effects on DVB-T signal . . . . . . . . . . . . . . . . . . 25
4.3 Common techniques for impulse noise cancellation . . . . . . . . . . . . . . . 26
4.3.1 Time domain clipping . . . . . . . . . . . . . . . . . . . . . . . . . . 26
4.3.2 Mean square error monitoring . . . . . . . . . . . . . . . . . . . . . . 27
4.3.3 Blanking method . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
4.3.4 Using guard band . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
4.4 Impulse noise cancellation utilizing pilots . . . . . . . . . . . . . . . . . . . . 27
4.4.1 Derivation of the impulse noise cancelling method . . . . . . . . . . . . 27
4.4.2 Example system . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
4.4.3 Simulation results . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
5 Pilot Based Impulse Noise Canceller Enhancements 33

5.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
5.2 Enhanced channel estimation schemes for impulsive noise environment . . . . . 33
5.2.1 Description of the proposed method . . . . . . . . . . . . . . . . . . . 34
5.2.2 Mean squared error performance . . . . . . . . . . . . . . . . . . . . . 34
5.2.3 Bit error ratio performance . . . . . . . . . . . . . . . . . . . . . . . . 37
5.3 Impulse burst cancellation method using pilots and soft bits in OFDM based
systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
5.3.1 Soft bits and carrier deviations weighting . . . . . . . . . . . . . . . . . 42
5.3.2 Simulation results and discussion . . . . . . . . . . . . . . . . . . . . . 44
5.4 Combination of interleaving and pilot based impulse noise canceller techniques 46
5.4.1 Burst position estimation method . . . . . . . . . . . . . . . . . . . . . 46
5.4.2 Results on the estimation of the burst position . . . . . . . . . . . . . . 46
5.4.3 Adaptive impulse burst detection . . . . . . . . . . . . . . . . . . . . . 47
5.4.4 Combination technique . . . . . . . . . . . . . . . . . . . . . . . . . . 49
CONTENTS vii
5.4.5 Simulation performance of the combination technique . . . . . . . . . . 49

5.5 Impulsive noise effects on time synchronization . . . . . . . . . . . . . . . . . 51
5.5.1 Considered symbol synchronization methods . . . . . . . . . . . . . . . 51
5.5.2 Simulation settings and results . . . . . . . . . . . . . . . . . . . . . . 52
6 Gap-Filler and Common Techniques for Loop Interference Cancellation 55

6.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
6.2 Signal and channel models . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
6.2.1 Channel model . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
6.2.2 Loop model . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
6.2.3 A simple gap filler configuration . . . . . . . . . . . . . . . . . . . . . 59
6.3 Common techniques for loop interference cancellation . . . . . . . . . . . . . 60
6.3.1 Channel estimation based frequency domain adaptive cancellation
algorithm . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60
6.3.2 LMS adaptive loop interference cancellation algorithm . . . . . . . . . . 62
6.3.3 LMS loop interference cancellation with spectrum whitening . . . . . . 64
7 Enhanced Algorithms for Loop Interference Cancellation 67

7.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
7.2 Autocorrelation based canceller algorithm . . . . . . . . . . . . . . . . . . . . 67
7.2.1 Description of the method . . . . . . . . . . . . . . . . . . . . . . . . 67
7.2.2 Enhanced autocorrelation method for multi-tap loop with exponential
profile . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 70
7.3 diversity method . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
7.3.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
7.3.2 Diversity scheme . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
7.3.3 Simulation results . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77
7.4 Enhanced version of the channel estimation based frequency domain adaptive
cancellation algorithm . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 78
7.5 Enhanced LMS loop interference cancellation . . . . . . . . . . . . . . . . . . 79
7.6 Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 80
8 DVB-T Signal in Cable TV Networks: Limitations and Requirements 83

8.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
8.2 History of cable television . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
8.3 Cable distribution system . . . . . . . . . . . . . . . . . . . . . . . . . . . . 84
8.4 Generic CATV channel model . . . . . . . . . . . . . . . . . . . . . . . . . . 86
8.4.1 CATV channel impairments . . . . . . . . . . . . . . . . . . . . . . . 86
8.5 Practical CATV channel model . . . . . . . . . . . . . . . . . . . . . . . . . 92
8.5.1 Linear distortion . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
8.5.2 Non-linear distortion . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
8.5.3 Phase noise . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
viii CONTENTS
8.6 Simulation of the DVB-T signal over the CATV channel . . . . . . . . . . . . 94

8.6.1 Simulation parameters and settings . . . . . . . . . . . . . . . . . . . . 96
8.6.2 Sensitivity to linear distortion . . . . . . . . . . . . . . . . . . . . . . 96
8.6.3 Sensitivity to phase noise . . . . . . . . . . . . . . . . . . . . . . . . . 96
8.6.4 DVB-T versus DVB-C in CATV channel . . . . . . . . . . . . . . . . . 98
9 Conclusions 101
9.1 Future Research . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 102
References 105
Appendix A 111
List of Figures
2.1 OFDM transmission principle. . . . . . . . . . . . . . . . . . . . . . . . . . 8
2.2 The concept of adding a cyclic prefix in case of a multipath channel. . . . . . . 9
2.3 Multipath propagation in SFN network. . . . . . . . . . . . . . . . . . . . . 10
2.4 Typical delay profile for a channel consisting of the natural and the artificial
delay spread due to SFN. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10
3.1 DVB-T functional block diagram. . . . . . . . . . . . . . . . . . . . . . . . 12
3.2 Packet coming from MPEG-2 transport MUX. . . . . . . . . . . . . . . . . . 14
3.3 Reed-Solomom RS(204,188) error protected data packet. . . . . . . . . . . . . 14
3.4 Functional block diagrams of the outer interleaver and de-interleaver . . . . . . 15
3.5 The inner coder with two outputs . . . . . . . . . . . . . . . . . . . . . . . . 15
3.6 Inner coding and inner interleaving block diagrams . . . . . . . . . . . . . . . 15
3.7 Inner interleaving (bit and symbol levels) and mapping of bits onto modulation
symbols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
3.8 Basic uniform constellations used in DVB-T system . . . . . . . . . . . . . . 16
3.9 Location of the scattered pilots in the OFDM frame structure. . . . . . . . . . 18
3.10 Principle of a Gap-filler. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
4.1 Impulse noise generation block diagram . . . . . . . . . . . . . . . . . . . . 24
4.2 DTG impulse noise model . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
4.3 Performance of impulse noise reduction algorithm in AWGN channel (16-QAM). 31
4.4 Performance of impulse noise reduction algorithm in AWGN channel (64-QAM). 31
4.5 BER vs. Blanking window length in AWGN channel: SNR=20 dB (16-QAM). 32
ix
x LIST OF FIGURES
5.1 Description of the enhanced channel estimation methods in impulsive noise

environment. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
5.2 Theoretical MSE vs. Doppler frequency for different pre-estimation methods.
The interpolator length is 91 and maximum delay of the channel is Tu /16. . . 37
5.3 BER performance for the three methods in the static SFN channel: 64-QAM
and s =512 samples. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
and s =1024 samples. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
and s =2048 samples. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
5.6 BER performance for the three methods in the mobile SFN channel: 64-QAM
and s =512 samples. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
and s =1024 samples. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
and s =2048 samples. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
5.9 Typical case of quantized carrier deviations . . . . . . . . . . . . . . . . . . . 42
5.10 Block diagram of a modified soft bit generation in DVB-T system when using a
pilot based impulse noise canceller. . . . . . . . . . . . . . . . . . . . . . . . 44
5.11 BER performance with different impulse interference cancellation methods in
static Rayleigh channel case. . . . . . . . . . . . . . . . . . . . . . . . . . . 45
5.12 BER performance with different impulse interference cancellation methods in
TU6 mobile channel case. . . . . . . . . . . . . . . . . . . . . . . . . . . . 45
5.13 Typical SIR-behavior after different receiver structures for 16-QAM, code rate
2/3, and 20 dB SNR. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 47
5.14 Block diagram for adaptive burst detection and blanking. . . . . . . . . . . . 48
5.15 Adaptive burst detection and blanking outputs. . . . . . . . . . . . . . . . . . 48
5.16 Modified OFDM system block diagram. . . . . . . . . . . . . . . . . . . . . 49
5.17 BER simulation results in static Rayleigh channel case for 64-QAM, code rate
2/3, and SNR = 25 dB. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 50
5.18 BER simulation results in mobile TU6 channel for 64-QAM, code rate 2/3,
Doppler = 25 Hz, and SNR = 25 dB. . . . . . . . . . . . . . . . . . . . . . . 50
5.19 Timing offset MSE performance in Rayleigh static SFN channel. . . . . . . . . 53
5.20 Timing offset MSE performance in Mobile SFN channel based on TU6 profile,
with MTM(16,4) filter and Doppler frequency = 100 Hz. . . . . . . . . . . . . 53
6.1 Simple configuration of DVB-T system using single relay station. . . . . . . . 57
LIST OF FIGURES xi
6.2 A model for the broadcast-wave relay SFN used in our basic study. . . . . . . 57
6.3 An example of the impulse response amplitude of the coupling path. . . . . . 58
6.4 A simple loop interference model. . . . . . . . . . . . . . . . . . . . . . . . 59
6.5 The gap filler as an echo generator. . . . . . . . . . . . . . . . . . . . . . . . 60
6.6 A model for the broadcast-wave relay SFN. . . . . . . . . . . . . . . . . . . . 61
6.7 BER performance of the frequency domain adaptive algorithm for single-tap
loop with LAR of 1dB. AWGN channel is assumed between the relay and the
receiver. 64-QAM and a code rate of 2/3 are considered. . . . . . . . . . . . 62
6.8 Variable relay gain. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
6.9 Standard adaptive LMS loop interference cancellation algorithm. . . . . . . . 63
6.10 BER for adaptive LMS loop interference cancellation with spectrum whitening
method: Ricean channel case and different noise bandwidths and power levels.
Single-tap loop, LAR = 1 dB, SNR=25 dB, 64-QAM 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. Single-tap loop, LAR = 1 dB, SNR=25 dB, 64-QAM and a code rate of
2/3 are considered. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
7.1 Impulse response of one-tap loop interference model. . . . . . . . . . . . . . 68
7.2 The magnitude of the sequence Q(l). . . . . . . . . . . . . . . . . . . . . . . 68
7.3 The angle of the sequence Q(l). . . . . . . . . . . . . . . . . . . . . . . . . . 69
7.4 Block diagram of the autocorrelation method diagram. . . . . . . . . . . . . . 69
7.5 BER in the Ricean channel case with LAR of 1dB using the basic
autocorrelation method, 64-QAM , 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, 64-QAM, and a code rate of 2/3. . . . . . . . . . . . 71
7.7 Loop interference cancellation based on the iterative autocorrelation method. . 73
7.8 MSE performance of loop amplitude estimation using enhanced autocorrelation

method with different LAR and . . . . . . . . . . . . . . . . . . . . . . . . 74
7.9 BER performance of the enhanced autocorrelation method in the AWGN

channel case with exponentially decaying multipath loop channel model. . . . 75
7.10 BER performance of the enhanced autocorrelation method in the Rayleigh

channel case with exponentially decaying multipath loop channel model. . . . 75
7.11 DVB-T repeater using diversity. . . . . . . . . . . . . . . . . . . . . . . . . 76

xii LIST OF FIGURES
7.12 BER performance of the DVB-T repeater with diversity: DVB-T, 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
single-tap loop with LAR of 1dB. Rayleigh channel is assumed between the
main transmitter and the relay. 64-QAM 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,
64-QAM, 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, 64-QAM, 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, 64-QAM, and a code rate of 2/3. . . . . 81
8.1 Cable TV system head-end. . . . . . . . . . . . . . . . . . . . . . . . . . . 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
8.5 Phase noise mask. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91

8.6 The generic CATV channel model . . . . . . . . . . . . . . . . . . . . . . . 92
8.7 Template for echoes provided by Nordic specification. . . . . . . . . . . . . . 93
8.8 The measured spectrum of the signal coming out from the actual converter used
in CATV network. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
8.9 Phase noise mask for modelling. . . . . . . . . . . . . . . . . . . . . . . . . 95
8.10 DVB-C system, transmitting side. . . . . . . . . . . . . . . . . . . . . . . . 95
8.11 DVB-C system, receiving side . . . . . . . . . . . . . . . . . . . . . . . . . 95
8.12 Sensitivity of DVBT to the linear distortion with 2/3 code rate. . . . . . . . . 96
8.13 Sensitivity of DVB-T to the phase noise with different levels of scaling
parameters: QPSK, 2/3 Code rate. . . . . . . . . . . . . . . . . . . . . . . . 97
parameters: 16-QAM, 2/3 Code rate. . . . . . . . . . . . . . . . . . . . . . 97
parameters: 64-QAM, 2/3 Code rate. . . . . . . . . . . . . . . . . . . . . . 98
8.16 Sensitivity of DVB-T and DVB-C to the phase-noise scaling parameter. . . . . 99

List of Acronyms
ACRONYMS
3G Third Generation Wireless System

3GPP 3rd Generation Partnership Project
8-VSB 8-level Vestigial Sideband modulation
ACI Adjacent Channel Interference
ACF Autocorrelation Function
A/D Analog-to-Digital converter
AGC Automatic Gain Control
AM Amplitude Modulation
ATSC Advanced Television Systems Committee
ARIB Association of Radio Industries and Businesses
AWGN Additive White Gaussian Noise
BER Bit Error Ratio
BPSK Binary Phase Shift Keying
CATV Community Antenna Television
CDMA Code Division Multiple Access
CLT Central Limit Theory
CP Cyclic Prefix
CR Code Rate
CSI Channel State Information
D/A Digital-to-Analog convertor
DBPSK Differential Binary Phase Shift Keying
DFT Discrete Fourier Transform
DiBEG Digital Broadcasting Experts Group
xiii
xiv List of Acronyms
DQPSK Differential Quadrature Phase Shift Keying

DTG Digital Television Group
DTV Digital TV
DTT Digital Terrestrial Television
DVB Digital Video Broadcasting
DVB-C Digital Video Broadcasting, Cable
DVB-H Digital Video Broadcasting, Handheld
DVB-T Digital Video Broadcasting, Terrestrial
DVB-S Digital Video Broadcasting, Satellite
DSP Digital Signal Processing
DSS Direct Satellite System
ETSI European Telecommunications Standards Institute
ES Elementary Stream
FCC Federal Communications Commission
FEC Forward Error Correction
FFT Fast Fourier Transform
FIR Finite Impulse Response
FIFO First In First Out
FOA Four OFDM Away
GSM Global System for Mobile Communication
HDTV High Definition Television
ICI Inter Carrier Interference
IDFT Inverse Discrete Fourier Transform
IF Intermediate Frequency
IFFT Inverse Fast Fourier Transform
ISDB-T Integrated Services Digital Broadcasting, Terrestrial
ISI Inter Symbol Interference
ISO International Standards Organization
ITU International Telcommunication Union
LAR Loop Amplitude Ratio
LOS Line Of Sight
LMS Least Mean Square
MAC Media Access Control
MFN Multi Frequency Network
MLSS Maximum Likelihood Symbol Synchronization
MPE-FEC Multi-Protocol Encapsulation Forward Error Correction
List of Acronyms xv
MUX Multiplexer
MPEG Moving Picture Expert Group
MSE Mean Square Error
NLOS Non-Line 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 Signal-to-Noise Ratio
TDMA Time Division Multiple Access
TPS Twelve Pilot Spacing
TS Transport Stream
UHF Ultra High Frequency
VCO Voltage-Controlled 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
1.1 HISTORY OF DIGITAL TERRESTRIAL BROADCASTING
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 high-definition 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 dual-sound
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 marked-led initiative to standardize digital broadcasting worldwide, three different
systems have already been specified: satellite system DVB-S [1], cable system DVB-C [2], and
terrestrial system DVB-T [3].
Being the most complex DVB delivery system, the DVB-T 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 DVB-T stepped into the scene. The DVB-T standard was originally de-
veloped for stationary reception with a directional roof-top antenna or a non-directional 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 DVB-T standard for mobile reception.
DVB-T was shown to be perfectly suited for reception by mobile receivers even at high
driving-speeds. However, new emerging services scenarios are appearing these days. These
1
2 INTRODUCTION
services use small hand-held devices that may include integrated mobile phones and provide
reception inside buildings, and cars, etc. Hence, new challenges facing the DVB-T standard
become more and more frequent. Questions of whether there is a need to change anything in
DVB-T 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 DVB-T has excellent performance in many aspects and the vast majority of
the commercial requirements are directly fulfilled as such. However some enhancements of the
DVB-T standard are needed. They have been partly implemented in the DVB-H system [4].
The work for this thesis was carried out while participating in projects contributing to the
DVB-T 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 DVB-T 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 DVB-T transmission over the cable TV network.
The general conclusions of the thesis are drawn in Chapter 9.
1.2 ORGANIZATION OF THE THESIS
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 DVB-T 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 DVB-T 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 4-8.
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 DVB-T 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 DVB-T 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 DVB-T 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 DVB-T 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, inter-operability issue of the DVB-T system is addressed. In particular, the
compatibility of DVB-T and DVB-C is investigated. We study the quality of DVB-T 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 head-end, from DVB-T to DVB-C (single carrier
AUTHORS CONTRIBUTION 3
system). In order to avoid this costly conversion and to transmit DVB-T 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 DVB-T transmission with sufficient quality [P10][P11].
In Chapter 9, we draw the conclusions and the main results of the thesis.
1.3 AUTHORS CONTRIBUTION
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 OFDM-Workshop, Hamburg, Germany, pp.
183-187, 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. 539-545, 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.373-376, 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
OFDM-Workshop, Dresden, Germany, pp. 62-66, 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. 1782-1786, 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 OFDM-Workshop,
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. 140-144 Sept.
2004.
[P8] A. Hazmi, J. Rinne, and M. Renfors, "Diversity based DVB-T in-door repeater in slowly
mobile loop interference environment," in Proc. 10th International OFDM-Workshop,
Hamburg, Germany, Sep. 2005.
[P9] A. Hazmi, J. Rinne, M. Renfors, "DVB-T signal in cable TV network: Advantages and
limitations," in Proc. of ICC2001, Helsinki, Finland, Vol. 7, pp. 2281-2285, June 11-14,
2001.
[P10] A. Hazmi, J. Rinne, M. Renfors," DVB-T signal over cable TV network and phase noise
requirements," in IEICE Trans. on Fundamentals of Electronics, Communications and
Computer Sciences, Vol. E84-A, NO. 4, pp. 966-970, 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. 19-22, 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 [P2-P11] were primary originated and analyzed by the author of
this thesis. Naturally the co-authors 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 DVB-T system, the main results of the thesis can
be also classified into three parts that can be summarized as follows:
Impulse noise effects and mitigation
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 time-frequency 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 subcarrier-wise 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 DVB-T 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
multi-tap 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 multi-tap 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 DVB-T 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.
DVB-T transmission in cable TV networks
In [P9] we study the quality of DVB-T signal transmission over the cable TV net-
work. The possibility of using the DVB-T 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 DVB-T signal is
simulated and compared to that of digital cable TV standard DVB-C 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 DVB-T 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 DVB-T 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 high-speed digital
communications [5] [6] [7]. Due to the recent advances of digital signal processing and very
large-scale integrated circuit technologies, the initial obstacles of OFDM implementation, such
as massively complex computation and need for high-speed 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 DVB-T (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.
2.2 OFDM PRINCIPLES
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 sub-bands, the frequency re-
sponse over each individual sub-band is relatively flat. Since each sub-channel 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 N-1t
XN-1(n)

Fig. 2.1 OFDM transmission principle.
small fraction of the original bandwidth, equalization is potentially simpler than in serial systems.
A simple equalization algorithm can minimize mean-squared distortion on each sub-channel,
and the implementation of differential encoding may make it possible to avoid equalization
altogether.
2.3 OFDM SIGNAL
2.3.1 The definition of OFDM signal

1
The spacing of the subcarriers is defined to be Tu , where Tu is the useful symbol duration.
Subcarrier frequencies are thus expressed as
k
fk = f0 + , k [0, N 1] (2.1)
Tu
where f0 represents the first subcarrier frequency and N is the total number of subcarriers.
Moreover, the orthogonal basis functions k (t) can be described as:
i2f t
e k
, 0 t < Tu
k (t) = (2.2)
0 , otherwise
The transmitted OFDM baseband signal in time domain is given as
+ N
X X 1
x(t) = Xk (n)k (t nTu ), (2.3)
n= k=0
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 N-point discrete
Fourier transform (IDFT). Thus the system complexity, e.g., the use of oscillators and sub-filters,
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
"#$%&#' (&)*+$
5+6& 78# (%)# (9:;%$ g
Fig. 2.2 The concept of adding a cyclic prefix in case of a multipath channel.
A reconstructed decision variable Yk on sub-channel k can be written as
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.
2.3.2 Guard interval

Due to multipath propagation in the transmission channel, many copies from the originally
transmitted signal will be obtained in the receiver side. Since a continuous sequence of OFDM
symbols is transmitted, this will introduce inter-symbol interference (ISI) between OFDM sym-
bols. The ISI can be eliminated by adding a guard interval g to each OFDM symbol. In this
way, the total duration of an OFDM symbol leaving the transmission antenna is Ts = g + Tu .
In order to be effective, the length of the guard interval should be longer than the delay spread of
the channel. The delay spread characterizes the maximum expected delay difference between
different significant multipath components reaching the receiver [12]. The guard interval could
be an all-zero sequence. In this case, however, the problem of intercarrier interference (ICI)
would arise, meaning that the subcarriers would no longer be orthogonal. To eliminate ICI, usu-
ally the OFDM symbol is cyclically extended [13] [14] [15]. A guard interval is copied from
the end of an OFDM symbol and placed as a prefix to the corresponding symbol. Figure2.2 rep-
resents an OFDM symbol with its delayed version. Because the delayed version occurs in the
guard period, ISI is not introduced to the useful part of the OFDM symbol after the cyclic prefix
(CP). The guard inteval periods will be deleted in the receiver side prior to the demodulation.
2.3.3 Single frequency networks

The capability of the OFDM system to overcome multipath interference has an interesting ad-
ditional advantage. When distributing the same program over a number of transmitters in the
10 OFDM: ORTHOGONAL FREQUENCY DIVISION MULTIPLEXING
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
Fig. 2.3 Multipath propagation in SFN network.
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 DVB-T system and emphasize its distinguishing features with respect
to present-day broadcasting services. The principles and implementations of the DVB-T trans-
mitter and receiver with OFDM modulator and demodulator will be explained. Advantages and
limitations of the DVB-T 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.
3.2 GENERAL CONSIDERATIONS
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 MPEG-2 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
Data Coder Splitter

2
MPEG-2 n
Source coding and
Multiplexing
Mux Adaptation
Outer Coder Outer Interleaver Inner Coder
Energy Dispersal
Mux Adaptation
Outer Coder Outer Interleaver Inner Coder
Energy Dispersal
To Aerial
Frame Guard Interval

Inner Interleaver Mapper OFDM D/A Front End
Adaptation Insertion
Pilot & TPS Terrestrial

Signals Channel Adapter
Fig. 3.1 DVB-T functional block diagram.

SOURCE ENCODING 13
3.3 SOURCE ENCODING
During the first steps in the DVB development, a fundamental decision was taken. It consisted
of the selection of the MPEG-2 standard to be the basic platform for source coding for both
audio and video in DVB. The description of the MPEG-2 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 MPEG-2 system
parameters for use in the DVB applications.
3.3.1 Video encoding

When digital TV signals are transmitted, the bandwidth has to be within the current analog TV
channels. Unfortunately, broadcast quality television, when digitized, results in a bit rate of
at least 216 Mbits/s, which is too large for the available bandwidth. Therefore, the compres-
sion of data is mandatory for the transmission and storage of digital video. The compression
scheme which has been adopted widely on all broadcast transmission media is the MPEG 2 IS0
standard. Specifically, the MPEG2 algorithm uses three main methods to remove the temporal,
spacial, and statistical redundancy which exists in all television images.
3.3.2 Audio encoding

The MPEG standard defines a method of compressing the audio using the MPEG layer II cod-
ing. Audio compression is achieved by first splitting the signal into a set of frequency sub-bands.
Based on the fact that the signal does not use all the audio bandwidth all the time, only useful
sub-band outputs are kept and quantized. In addition, the quantization distortion can be reduced
by using the psychoacoustic model. The model makes use of the fact that the human ear cannot
hear soft sounds which occur at the same frequency as loud sounds.
3.3.3 MPEG-2 transport stream

The MPEG2 systems parts specifies how the sources for video, audio, and data are combined
into a program stream. The process of combining the streams is known as multiplexing. The
most basic component in the multiplexed bit stream is known as an Elementary Stream (ES). A
programme (eg., a television programme) contains a combination of elementary streams. Each
ES is input to an MPEG-2 processor which accumulates the data into a stream of Packetised
Elementary Stream (PES) packets. A PES packet may be of fixed or variable size. A trans-
port stream consists of a sequence of fixed sized transport packets of 188 Bytes. Each packet
comprises 184 Bytes of payload and a 4 Bytes header.
3.4 DVB-T: PHYSICAL LAYER
DVB-T 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.
3.4.1 Multiplexing and channel coding

3.4.1.1 Transport multiplex adaptation and scrambling First we should rearrange
the input data stream coming from the source encoding layer as fixed length TS packets. The
TS packets are 188 bytes long, from which the first byte is a sync-byte. The byte is 01000111
in binary form. (Figure 3.2)
SYNC : 1 byte MPEG-2 transport MUX data : 187 bytes
Fig. 3.2 Packet coming from MPEG-2 transport MUX.
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 Reed-Solomon 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).
SYNC 1 / SYNC n Randomized data 187 Bytes 16 Parity bytes
Fig. 3.3 Reed-Solomom RS(204,188) error protected data packet.
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 DVB-T 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.
DVB-T: 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
(a) Interleaver (b) De-interleaver
Fig. 3.4 Functional block diagrams of the outer interleaver and de-interleaver
3.4.1.5 Inner interleaving In inner interleaving, two steps are considered. First, there is
a bit-wise 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, 16-QAM and 64-QAM 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, 16-QAM, or 64-QAM 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
Input 1-bit 1-bit 1-bit 1-bit 1-bit 1-bit

delay delay delay delay delay delay
Output 2
Fig. 3.5 The inner coder with two outputs
Inner Coder
X
Puncturing
Convolutional
Y with Inner Interleaver
Encoder
serial output
Fig. 3.6 Inner coding and inner interleaving block diagrams

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 64-QAM. In Figure 3.8, the basic constella-
tions for uniform QPSK, 16-QAM, and 64-QAM 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 DVB-T system, as possible transmis-
sion mode, non-uniform mapped constellations are also supported in the cases of 16-QAM and
64-QAM.
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)
Re(z) Re(z) Re(z)
QPSK
16-QAM
64-QAM
Fig. 3.8 Basic uniform constellations used in DVB-T system

DVB-T: PHYSICAL LAYER 17
3.4.2 OFDM frame structure

According to the specifications, the transmitted signa1 is organized in frames. Let us consider
TF as the duration of the frame, which consists of 68 OFDM symbols, that are numbered from
0 to 67. Each OFDM symbol has 6817 useful carriers, in the transmission mode 8k, and 1705 in
the transmission mode 2k. From this tota1 of 6817 useful carriers, 6048 convey the data, and the
rest carry reference information. In the 2k mode, the data is modulated onto 1512 carriers. In
both modes, additional carriers known as guard band or virtual carriers are placed at the upper
and lower extremes of the transmission channel spectrum, and they are zeroed, i.e., modulated
by zero, in order to reduce the ACI (Adjacent Channel Interference). After IFFT transforma-
tion, a guard interval of length is added to the useful symbol having a length of Tu . The guard
interva1 is inserted in the front as cyclic continuation of the data part. There is one more impor-
tant parameter to be considered, T = 7/64s = 0.109s. It is called the elementary period of
the system, and all the time and frequency related parameters of the system can be expressed as
multiples of this number. Table 3.1 shows the main OFDM parameters for DVB-T.
Table 3.1 Main OFDM parameters for DVB-T.
Parameter 8k mode 2k mode
Number of useful carriers 6817 1705

FFT size 8192 2048
Useful symbol duration Tu 896s 224s
Carrier spacing 1/Tu 1116Hz 4464Hz
Bandwidth 7.61M Hz 7.61M Hz
Cyclic prefix duration Tu /32, Tu /16, Tu /8, Tu /4 Tu /32, Tu /16, Tu /8, Tu /4
As it can be observed from the table, the bandwidth of the transmission channel is the same
in both modes, 7.61M Hz approximately.
3.4.3 Reference signals

In the previous section, reference information was mentioned. This reference information is con-
veyed in known subcarriers. There are three different types of reference subcarriers: scattered
pilots, continua1 pilots, and transmission parameter signalling pilots. Scattered and continua1
pilots can be used for frame synchronization, frequency synchronization, and also for chan-
nel estimation, while transmission parameter signalling pilots give information about the type
of modulation, the constellation characteristics, hierarchy information, selected guard interva1,
code rate, transmission mode, and even the frame number within the super- frame. The scat-
tered and continua1 pilots are modulated according to the data generated in a PRBS generator.
This PRBS generator is defined by the polynomia1 X 11 + X 2 + 1.
The scattered and continua1 pilots are transmitted at a "boosted" power level with respect to
the data carriers. These boosted subcarrier pilot symbols always have imaginary part zero, and
the rea1 part is given by 4/3 (1 wk ) equation where wk is the va1ue of the PRBS at the
instant k, which yields a va1ue of 4/3. As the PRBS sequence is a1so known by the receiver,
it is possible to know which va1ue is expected. This is the basis for the pilot based channel
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
siti uivvewv {|}uivvewv~

xeyzt uivvewv
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.
3.5 DVB-T RECEIVER SIMULATION MODEL
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.
3.5.1 Simulation environment

The used simulation tool is CoCentric System Studio (CCSS) [16]. It is a simulation tool from
Synopsys. The receiver and transmitter structures are represented by a data-flow graph, which
consists of functional blocks and connectors. The blocks can be hierarchical models containing
other blocks. Otherwise, the functionality of the block may be described by a C-code. The
advantage of the System Studio data-flow graphs is the flexible treatment of the input data. Each
DVB-T RECEIVER SIMULATION MODEL 19
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 DVB-T simulation model assumes ideal time synchronization, and it does not con-
sider any RF impairments like carrier frequency offset or I/Q-imbalance. Otherwise the model
includes the functions of a practical DVB-T receiver. The basic functionality of the simulation
model has been verified by comparisons with other project partners.
3.5.2 Channel modeling

The channels that have been used in the evaluation of the DVB-T receiver performance can be
classified in two categories: static and mobile channels.
3.5.2.1 Static channels The static channel profiles are based on the DVB-T 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 time-variant, 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
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 BASIC ASPECTS OF DVB-T NETWORKS
3.6.1 Conventionally planned networks

Conventionally planned DVB-T networks consist of transmitters with independent programme
signals and with individual radio frequencies. Therefore, they are also referred to as Multi Fre-
quency Networks (MFN). In order to cover large areas with one DVB-T signal, a certain number
of radio-frequency channels is needed. The number of channels depends on the robustness of
the transmission, i. e., the type of modulation associated with the applied channel code rate.
The frequency resource expressed as the number of channels needed to provide one signal at
any location is far higher with MFN than with Single Frequency Networks (SFN). The transmit-
ters in an MFN need not to obey rules of synchronous emissions. Therefore, no co-ordination
between transmitter operators is necessary. The installation of local or regional services is easy
with the MFN concept compared to the SFN concept.
3.6.2 Single frequency networks (SFN)

3.6.2.1 Principle In a SFN, all the transmitters are synchronously modulated with the same
signal and radiate on the same frequency. Due to the multi-path capability of the multi-carrier
transmission system (COFDM), signals from several transmitters arriving at a receiving antenna
may contribute constructively to the total wanted signal. With the SFN technique, large areas
can be served with a common multiplex at a common radio center frequency. Therefore, the
frequency efficiency of SFNs appear to be very high compared to MFNs . The SFN technique
is not only frequency efficient but also power efficient. This can be explained by considering
the strong local variations of the field strength of any given transmitter. SFNs can use lower
powered transmitters because in peripheral regions, multiple transmitters contribute to the over-
all received signal (macro diversity). The price to pay for frequency and power efficiency is the
synchronous operation of all transmitters in a given network.
3.6.3 Gap-filler
If gaps exist in a service area, as may be encountered in deep valleys, tunnels, or inside houses,
the multipath capability of DVB-T enables these gaps to be filled in a very efficient way [3].
The principle is as follows: outside the gap or the uncovered sub-area, the DVB-T 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 so-called gap-filler (also called as repeater
or relay). This is usually refereed as a direct or a non-regenerative 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
non-generative gap fillers depends on many factors. In the scope of DVB-T, a non-regenerative
BASIC ASPECTS OF DVB-T 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 gap-filler is sufficient isolation be-
tween transmitter and receiver antennas. To prevent the re-transmitter from oscillating, the gain
of the re-transmitter has to be less than the feedback attenuation (see Figure 3.10).
Fig. 3.10 Principle of a Gap-filler.
A gap-filler 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 position of the antennas on the tower or building.
The radiation diagram of the antennas.
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 gap-filler, mainly due to
filtering within the device. This will cause frequency selective attenuation of the retransmitted
signal similar to the characteristic of a two-path 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].
3.7 OTHER DVB STANDARDS
3.7.1 DVB-H
The digital video broadcasting for hand-held devices(DVB-H) [4] [18] can be seen as an expan-
sion of the DVB-T 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 front-end and also enables smooth and seamless frequency handover. The use of
time slicing is mandatory in DVB-H. The FEC for multiprotocol encapsulated data (MPE-FEC)
gives an improvement in Doppler performance and increases tolerance to impulse interference.
The use of MPE-FEC is optional for DVB-H.
The physical layer has four extensions to the existing DVB-T physical layer. First, the bits
in transmitter parameter signaling have been upgraded to include two additional bits to indicate
the presence of DVB-H services and the possible use of MPE-FEC. Second, a new 4k mode
is adopted. This gives additional flexibility for the network design. The 4k mode is an option
for DVB-H complementing the 2k and 8k. Third, a new way of using the symbol interleaver
of DVB-T 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 in-depth 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 DVB-T
physical layer is the 5-MHz channel bandwidth to be used in non-broadcast bands . The basic
parameters of DVB-H physical layer can be found in [4].
Because of the similarity between DVB-T and DVB-H in the physical layer, all the algorithms
and techniques introduced by the thesis are applicable for both standards.
3.7.2 DVB-S
The DVB-S satellite modulator has a similar block diagram to the DVB-T modulator except that
the system only uses single-carrier modulation with QPSK [1]. Multipath is not a problem with
satellite transmission and the non-linearity of the satellite transmitter power amplifier (due to
hard power-efficiency constraints) at present limits the modulation choice to QPSK modulation.
The system uses the same concatenated convolutional and Reed-Solomon error correcting codes
but the bit-wise inner interleaver is not used. Since satellite channel bandwidths are much larger,
the source bit-rate 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 DVB-C
The DVB-C cable modulator is similar to the single carrier DVB-S modulator except that the
modulation scheme on the single carrier is either 16-QAM, 32-QAM or 64-QAM, [2]. Fur-
thermore, the convolutional encoder is omitted, leaving the outer interleaver and Reed-Solomon
channel coding of DVB-T 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 DVB-T systems
4.1 INTRODUCTION
This chapter focuses on the effects of impulsive noise on DVB-T 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 DVB-T 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 DVB-T 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.
4.2 IMPULSE NOISE MODELING
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.
4.2.1 Impulse noise sources
Impulsive interference is usually described in the literature as a process characterized by bursts

of one or more short pulses whose amplitude, duration, and time of occurrence are random.
23
24 IMPULSIVE NOISE MODELING, EFFECTS AND MITIGATION IN DVB-T 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 hair-dryers, 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.
4.2.2 Impulse noise statistical models

An impulsive noise sequence ni (m) consists of short duration pulses with a random amplitude,
duration, and time of occurrence, and may be modeled as the output of a filter excited by an
amplitude-modulated random binary sequence as
P
X 1
ni (m) = h(k)n(m k)b(m k), (4.1)
k=0
where b(m) is a binary-valued random sequence model of the time of occurrence of impulsive
noise, n(m) is a continuous-valued 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).
Fig. 4.1 Impulse noise generation block diagram
In the literature, there are two important statistical processes for modeling impulsive noise
as an amplitude modulated binary sequence, Bernoulli-Gaussian process and Poisson-Gaussian
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!
where is a rate function with the following properties:

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
4.2.3 Impulse noise effects on DVB-T signal

The study of impulsive noise impairment in the area of digital terrestrial television (DTT) has
been of interest starting from the beginning of the launch of this service. While DVB-T signal is
more immune to this form of impairment because of the parallel transmission that characterizes
the OFDM modulation technique, the lack of a suitable time-interleaving scheme in the DVB-T
specification [3] makes the system sensitive to sources of interference of an impulsive nature.
As a result of an increased interest in the causes and effects of impulse noise, a number of
recent noise surveys have been conducted. Impulsive noise models for systems operating at low
frequencies have been proposed by the ITU [21]. These models are based on measurements
of median levels of interfering noise. However, at higher frequencies, high peak levels of in-
terference may adversely affect reception even though the measured median level does not rise
above the background noise. Recently, a working group within the Digital Television Group
(DTG), carried out a series of theoretical and practical studies to devise a representative set of
test waveforms for impulsive interference [22].
Based on the statistical analysis of a large amount of captures of impulsive interference, the
DTG concluded that a simplistic way to simulate impulsive noise was to use the well-known
gated Gaussian noise model. Another variation of this model was also introduced which is based
on the use of a train of gating pulses with constant amplitude and duration but whose spacing is
uniformly random.
The conclusion was to select a set of tests consisting of six waveform patterns. A number of
test signals have been designed, all based on the above idea. The diagram in Figure 4.2 below
illustrates the terminology.
Each burst is relatively short compared with the symbol period, so that most bursts only affect
a single OFDM symbol. The separation between bursts is sufficiently great for them to behave
as isolated events: any errors resulting from the first burst will have been flushed from the sys-
tem by the time the second burst is received. The number of pulses per burst is defined, but the
spacing between pulses is allowed to vary randomly between given maximum and minimum
values.
So far, six tests have been defined [22], as detailed in the Table 4.1.
Burst 1 Burst 2
Burst Duration
Pulse Duration
10 ms (fixed) 250 ns (fixed)
Fig. 4.2 DTG impulse noise model
Table 4.1 Impulse noise tests.
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
4.3 COMMON TECHNIQUES FOR IMPULSE NOISE CANCELLATION
4.3.1 Time domain clipping

A simple time domain approach which is based on the assumption that some of the impulses ex-
perienced will be of much larger magnitude than the received signal. To detect these impulses,
the decoder may monitor for sudden drops in the automatic gain control level [23]. If this oc-
curs for several time domain samples, the decoder declares that an impulse has occurred and
erases or clips the corresponding sample. In clipping, the samples may be given the value which
corresponds to the clipping level amplitude (and keeping the phases). Erasing means that the
impulse samples become zero-valued. Clipping and erasing methods above are somewhat use-
ful but they leave moderate impulse levels untouched which means that their capabilities are
quite limited.
IMPULSE NOISE CANCELLATION UTILIZING PILOTS 27
4.3.2 Mean square error monitoring

In this frequency-domain method another source of side information can be used in the decoder
[24]. It is the squared error between the received signal points and the sliced data points on each
sub-channel (a frequency domain approach). The magnitude of the squared errors across all
the data-carriers can be used to estimate the reliability of the sliced data points. We can simply
mark unreliable a symbol if the squared error signal exceeds a predetermined threshold.
4.3.3 Blanking method

This method differs from the clipping method by blanking (setting to zero) all samples which
are known to be corrupted, i.e., belonging to a burst period [23]. The knowledge of impulse
position and duration may be based, e.g., on monitoring instantaneous power within a sliding
window when exceeding certain clipping levels.
4.3.4 Using guard band

In order to make the performance of the blanking approach better, we can try to estimate the
samples that have been removed. If the noise burst is detected and the corresponding time sam-
ples blanked, it might be possible to use the information that there should be no signal on the
empty carriers (guard band) to restore the original post-FFT values [25]. Unfortunately the
method requires solution of a general complex system equations which can be quite difficult and
cumbersome.
4.4 IMPULSE NOISE CANCELLATION UTILIZING PILOTS
4.4.1 Derivation of the impulse noise cancelling method

When the impulsive samples are found, those samples will be blanked, i.e., their values are set
to zero. For computational simplicity it is assumed here that the blanking is done for consequent
samples. For the cancellation of the effects of blanking, a linear deviation estimator operating
in the frequency domain exploiting information of pilots (e.g., channel estimation pilots) can be
used [P1]. Here it is reasonable to assume that the deviations at each subcarrier due to impulsive
noise blanking are coming from a wide sense stationary random process. By supposing that the
estimator spans over a finite set of sub-carriers, the output of the deviation estimator at the k th
carrier may be expressed as:
dk =
X
cj,k pkj , (4.5)
j
where cj,k are the unknown estimator coefficients and pkj represent the deviations at the pilots
sub-carriers.
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
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
E{k pl } = 0 cj,k pkj )pl } = 0

P
E{(dk (4.8)
j
E{dk pl cj,k pkj pl } = 0
P
(4.9)
j
E{dk pl } = E{ cj,k pkj pl }.
P
(4.10)
j
This set of linear equations may be expressed in matrix equation form as
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 straight-forward manner
ck = R1 k . (4.16)
Now using these coefficients, it is possible to estimate the deviations at the data-carriers 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.
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 DVB-T like signal. We assume, like in the DVB-T 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 DVB-T 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
The autocorrelation for carrier deviations can now be derived to be approximately

N 1 u+L
1 X 2 w
r(, w) = E{dv dw }
X
= 2
E{|b k | } ei2q N . (4.20)
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).
4.4.2 Example system

For conciseness, we consider here a simplified cancelling scheme that uses only two neighbor-
ing pilots. It has been assumed that the impulses occur in a bursty manner so that one burst
takes place in every OFDM-symbol. 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.
4.4.3 Simulation results

The suggested impulse noise canceller was tested by using 8k DVB-T system model. The sub-
modulation was chosen to be either 16-QAM or 64-QAM and the code rate of the convolutional
code was set to 2/3. The impulse noise burst length, L, was 700, 1000 and 1300 samples, cor-
responding to 76.6, 109 and 142 s durations. Here we consider mainly the AWGN channel
in the evaluation of the impulse noise canceller. Frequency and time selective channels will be
considered in the next chapter.
Each symbol has one impulse burst, whose location was assumed to be determined perfectly.
Two neighboring pilots were used to estimate the distortion on the data-carriers located between
the pilots. In the calculation of distortion SNR was assumed to be large.
The results are shown in Figure 4.3 and Figure 4.4 for 16-QAM and 64-QAM, respectively.
For comparative purposes, the performance in pure AWGN conditions is shown. Also, the per-
formance of systems with blanking only are presented. As can be seen, systems with blanking
and compensation perform well for the two shortest blanking window widths. These blank-
ing lengths are still quite useful from the application point of view and, thus, the results are
encouraging.
Next, in order to find out the upper limit of the system performance we considered the case
where SNR is 20 dB. The results can be found in Figure 4.5 for 16-QAM case. In this simu-
lation, the mean impulse amplitude was set to about 39 dB larger than the mean amplitude of
the data. As shown in Figure 4.5, the system without any special means to combat against large
amplitude impulses fails to operate properly. Using only blanking, seems to operate satisfac-
torily when there are relatively small number of corrupted samples. However when using the
proposed canceller with blanking, the number of tolerated impulsive samples can be about 500
( 55 s) more than with blanking only.
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
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 (16-QAM).
0
10
1
10
2
10
BER after Viterbi
3
10
4
10
Blanking, L=700
Blanking, L=1000
5
10 Blanking, 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 (64-QAM).
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 (16-QAM).
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 DVB-T system can be improved significantly. However, addi-
tional measures are needed to handle the time-frequency 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 DVB-T
systems are addressed.
5.2 ENHANCED CHANNEL ESTIMATION SCHEMES FOR IMPULSIVE NOISE

ENVIRONMENT
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
5.2.1 Description of the proposed method

The aim is to use pilots of the other (previous and future) symbols to construct reliable channel
estimates. To make the results more practical, 8k DVB-T system is used here as an example [3].
Conventionally, the separable channel estimation in OFDM is done in three steps:
1. Carry out the channel estimation in time domain (e.g., in DVB-T this produces channel
estimates for every third subcarrier).
2. Carry out channel estimation in frequency domain (channel estimates for all the subcarriers
will be available).
3. Compensate channel on data-carriers.
When impulse noise is considered, this approach would be preceded by pre-estimation [P2]:
1. Carry out pre-estimation 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 (DVB-T) OFDM symbol is supposed to be corrupted by impulse burst.
In the pre-estimation, 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 DVB-T 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)).
5.2.2 Mean squared error performance

It is assumed that the bursts do not occur too often, in which case pilot response estimation is pos-
sible using these approaches. Also it is possible to use longer pre-estimation in time direction,
but the impulse burst should occur very rarely to avoid error spreading.
In the following analysis it is supposed that the mth symbol contains an impulse burst. Other
symbols are assumed to be impulse free. Mean Squared Error (MSE) is calculated analytically
after pre-estimation. The conventional separable estimator (time and frequency) will enhance
the MSE further, but the MSE after pre-estimation can be used in the fair comparison of different
schemes.
In general the MSE can be written as:
J = E{|H(k, m) H(k, m)|2 }, (5.1)

where E{.} is the average operation in time domain, k refers to subcarrier index, and m is the
OFDM symbol number. H(.) is the channel response and H(.) is the pre-estimated channel
response. Here, index k takes pilot position values defined by the pilot pattern.
In the case of FOA method,
H(k, m) = H(k, m 4) + nawgn + nICI . (5.2)

ENHANCED CHANNEL ESTIMATION SCHEMES FOR IMPULSIVE NOISE ENVIRONMENT 35
1
2
3
4
5
6
Pilots of the impulse corrupted symbol
(a) Four Away OFDM

Symbols Method
Combination
Interpolation
Estimated Pilots
(b) Six Pilots Spacing

Method
5 7
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.
For the SPS method, the estimate at pilots (of both neighboring symbols) is calculated first by
H(k, m) = H(k, m 1) + nawgn + nICI , (5.3)

and then pre-estimation interpolation is used. In the above equations, AWGN and ICI exist in
the pilots, i.e., before pre-estimation.
For TPS method, it can be found that
1
H(k, m) = [H(k, m 1) + nawgn + nICI ] +
2
1
[H(k, m + 1) + n,awgn + n,ICI ], (5.4)
2
where AWGN and ICI are defined after pre-estimation. MSE for FOA method can be calculated
in straight-forward manner:
J1 = E{|H(k, m) [H(k, m 4) + nawgn + nICI ]|2 }

2 2 2
= 2H 2R(4) + awgn + ICI , (5.5)
where R() is the autocorrelation function of time-variant 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 frequency-domain pre-estimation is used (e.g., in DVB-T, interpolator with 6 carrier
pilot spacing). The total MSE in the pilot estimates after pre-estimation 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 pre-estimation 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 pre-estimation (for DVB-T 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]
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 pre-estimation methods. The interpolator
length is 91 and maximum delay of the channel is Tu /16.
5.2.3 Bit error ratio performance

Unfortunately, the MSE cannot give the whole picture of the DVB-T receiver performance when
using these methods. To include in the results the effect of the Forward Error Correcting (FEC)
part of the DVB-T receiver (mainly the inner decoder part), bit error ratio (BER) after Viterbi
decoding will be considered in the following as a performance measure. Different Doppler
frequencies and code rates will be considered for each of the methods described above.
The aim here is to study the performance of the new channel estimation scheme for pilot
based impulse noise canceller in the case of time-frequency 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 OFDM-symbol 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 Bernoulli-Gaussian (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 DVB-T [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.
5.2.3.2 Time-frequency selective channel The channel impairment considered here

is due to multipath reflections and Doppler spread. A worst case condition where the chan-
nel presents high frequency selectivity and high Doppler spread is assumed. SFN channel is
such an example [28] [29]. Here, we consider the SFN configuration based on DVB-T channel
model for typical urban reception [17]. The considered SFN is based on two transmitters. The
strongest multipath components from both transmitters are assumed to be at the same level. The
channel model is mainly parameterized by the delay s that refers to the delay spread of the SFN
channels and fd , the Doppler frequency when we consider a mobile SFN channel(Appendix A).
5.2.3.3 Results and discussion The performance of the three methods was tested by
using the 8k DVB-T system mode. The sub-modulation was 64-QAM 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.
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
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: 64-QAM 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: 64-QAM and s =1024
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
10 15 20 25 30 35 40
SNR in dB
Fig. 5.5 BER performance for the three methods in the static SFN channel: 64-QAM 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: 64-QAM and s =512
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
6
10
0 10 20 30 40 50 60 70
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
samples.
5.3 IMPULSE BURST CANCELLATION METHOD USING PILOTS AND SOFT

BITS IN OFDM BASED SYSTEMS
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 DVB-T 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 DVB-T 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.
Transmited signal before blanking (Pilots+QPSK modulated data)

2
Absolute value
1.8
1.6
1.4
0 20 40 60 80 100 120 140 160 180 200
subcarrier index
3
Absolute value
Received signal after blanking

2
0
0 20 40 60 80 100 120 140 160 180 200
subcarrier index
1
Absolute value
absolute carrier deviation
0.5
0
0 20 40 60 80 100 120 140 160 180 200
subcarrier index
1
Quantized levels
0.5
quantized carrier deviation

0
0 20 40 60 80 100 120 140 160 180 200
subcarrier index
Fig. 5.9 Typical case of quantized carrier deviations
5.3.1 Soft bits and carrier deviations weighting

The signal-to-noise-ratio is subchannel dependent in multipath channels. In systems using con-
volutional error correcting coding, like in DVB-T systems, the information of the gains of
individual subcarriers, often referred to as the channel state information (CSI), can be utilized
in the convolutional Viterbi decoder. In coded OFDM (COFDM) systems, CSI can be used in
computing the branch metrics, as suggested, e.g., in [30].
IMPULSE BURST CANCELLATION METHOD USING PILOTS AND SOFT BITS IN OFDM BASED SYSTEMS 43
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 DVB-T
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
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].
Compensate Soft QAM

FFT
blanking detection
To inner
deinterleaver
and Viterbi
decoder
1/.
| .|^2
Extract Channel
pilots estimation
Subcarrier
deviation Quantization 1/.
estimation
Fig. 5.10 Block diagram of a modified soft bit generation in DVB-T system when using a pilot based
impulse noise canceller.
5.3.2 Simulation results and discussion

To evaluate the performance of the new method we consider the 8k DVB-T system mode using
64-QAM sub-modulation. We simulate the cases of 1/2 and 2/3 code rates. The impulsive noise
model is based on the gated complex Gaussian process with given impulse burst duration and
amplitude characteristics [20]. Here the impulse burst is assumed to occur in every symbol.
The length, L, was fixed to 500 samples. In the modified soft bit generation block, we used
a quantizer with 10 levels. The maximum value is set to 0.5, a value which is determined by
extensive simulations.
For performance comparisons, we simulate in addition to the introduced method, two other
possible receiver structures. The first one uses blanking only. In this structure the impulsive
bursty samples are first detected and their level are set to zero. In the second structure, blanking
and soft bit weighting is used. In the introduced method, we consider blanking, compensation,
and soft bit weighting.
Figure 5.11 shows the BER performance of the three receiver structures when static Rayleigh
channel is considered. As can be seen, the BER performance of the introduced method over-
comes the performance of other methods.
In the TU6 mobile channel case (Figure 5.12), it can be seen that the QEF performance is
achieved only for small Doppler frequencies when 2/3 code rate is used. Higher Dopplers can
be tolerated by the introduced method when 1/2 code rate is selected.
IMPULSE BURST CANCELLATION METHOD USING PILOTS AND SOFT BITS IN OFDM BASED SYSTEMS 45
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
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
6
10
0 10 20 30 40 50 60 70
Fig. 5.12 BER performance with different impulse interference cancellation methods in TU6 mobile
channel case.
5.4 COMBINATION OF INTERLEAVING AND PILOT BASED IMPULSE NOISE

CANCELLER TECHNIQUES
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.
5.4.1 Burst position estimation method

In practice, an OFDM symbol containing, e.g., an impulse burst of amplitude -3 dB compared to
undistorted signal level and duration of 900 samples ( 100 s) will cause problems in 8k DVB-T
reception with 64-QAM submodulation. Thus, the impulse burst detection method should be
able to operate in this kind of conditions. Here, the position of the impulsive samples are de-
tected by a local estimator of power of incoming signal samples. When the local power estimate
is larger than the predetermined threshold , the sample is declared to be impulse contaminated.
The local power is estimated using simple running-sum FIR-filter whose taps are given by
1
h(n) = , n = 1, 2, ....., . (5.17)

where is the total number of taps. It was found by extensive simulations that a suitable value
for would be in the neighborhood of 150 and for around 1.2. Additionally, it was found
that to avoid false detections, and hence unnecessary blanking, should be increased by about
20%. Thus, = 1.4 is relatively good starting point for 8k DVB-T.
5.4.2 Results on the estimation of the burst position

Three different receiver structures were tested in noise free conditions without any other dis-
tortion than the impulse noise. The considered methods were: 1. conventional receiver, 2.
receiver with impulse detection and blanking, and 3. receiver with impulse detection, blank-
ing, and compensation. Signal-to-Interference Ratio (SIR) in frequency domain data-carriers
was used as a performance measure in all the evaluations. This is the signal power to distor-
tion power ratio caused by the combination of impulsive noise and possible cancelling scheme.
The gated Gaussian impulse burst amplitude level was varied from -9 to +6 dB compared to the
average amplitude level of the OFDM signal. The impulse burst duration was fixed to 900 sam-
ples, which corresponds to about 100 s. Single burst per symbol was used. At most, DVB-T
can tolerate blanking of about 900 samples when using pilot assisted cancellation and 16-QAM
with 2/3 code rate in a case with 20 dB SNR, as shown in Chapter 4.
The simulated results are shown in Figure 5.13, where it can be clearly seen that blanking
and compensation with practical burst position detection produces the best SIR behavior. For
low impulse amplitude levels, conventional receiver is better than the blanking receiver, but af-
COMBINATION OF INTERLEAVING AND PILOT BASED IMPULSE NOISE CANCELLER TECHNIQUES 47
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
Blanking using detection

12
10
8
Blanking with known burst position
4
10 8 6 4 2 0 2 4 6
Impulse amplitude dB
Fig. 5.13 Typical SIR-behavior after different receiver structures for 16-QAM, 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.
5.4.3 Adaptive impulse burst detection

We mentioned in the previous section that the burst detection is performed by power estimation
using sliding window and thresholding. The choice of the threshold level is very critical, be-
cause we need to avoid false alarms, while detecting the impulse-noisy-samples. The threshold
depends mainly on the burst amplitude and the transmitted signal envelope. In the static channel
case, the power of the transmitted signal is assumed to be constant and, therefore, a fixed thresh-
old level will be a suitable choice. However in mobile and severe frequency selective channel
cases, the envelope will be changing in time, which requires a dynamic changing of the thresh-
old level. We present here a simple method where two FIR filters are considered in the detection
process [P4]. The taps are given by
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 DVB-T 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)
if (input (n) > threshold (n))

Input
s(n) = 0 ;
else
h1(n) s(n) = r (n) ;
2
(.) Threshold level
h2(n) Blanked signal

s(n)
Scaling margin
Fig. 5.14 Block diagram for adaptive burst detection and blanking.
1
The short fillter (h 1) output
Impulse bursts The long fillter (h ) output

2
0
The threshold
2
Amplitude (dB)
7
0 1 2 3 4 5 6 7 8
OFDM symbol time index x 10
4
Fig. 5.15 Adaptive burst detection and blanking outputs.

COMBINATION OF INTERLEAVING AND PILOT BASED IMPULSE NOISE CANCELLER TECHNIQUES 49
5.4.4 Combination technique

In the DVB-T system, the protection against impulse noise can be improved by increasing the
size of the interleaver associated with the inner convolutional code so that it acts over several
OFDM symbols, rather than entirely within a symbol as it is currently the case [3]. Another
alternative would be to include an additional convolutional interleaver, in addition to the existing
bit and symbol interleavers. The purpose of interleaving is to randomize the bursty channel
errors. This can greatly improve the performance of the error-correcting codes which are often
good at correcting random errors. In general, interleaving rearranges the order of symbols to
be transmitted according to a given rule. At the receiver, the reverse rule is used to restore the
original sequence. Here we consider the case of convolutional interleaver. It consists of a set of
J shift registers, each with a fixed delay F [31].
Figure 5.16 shows an OFDM system where the use of the convolutional interleaver, adap-
tive burst detection block, and pilots based impulse canceller are considered. The input bits
streams are first convolutionally coded and then mapped into symbols. The symbols are then
interleaved and modulated by inverse fast Fourier transform (IFFT). After adding cyclic prefix,
the OFDM symbols are then transmitted through a multipath fading channel, where AWGN and
impulse noise are also present. At the receiver, an adaptive burst detector block is used to esti-
mate the location of the bursts, blank the impulsive samples, and provides location information
to the pilot based impulse canceller which compensates the effect of the impulse burst in the re-
ceived signal. After deinterleaving, the received symbols are decoded with soft Viterbi decoder.
Shadowed blocks represent the modifications added to the standard DVB-T system. Here we
consider the case of additional time interleaver [P4]. In order to quantify the effects of the extra
interleaving, simulations will be conducted in the next section.
Input bit stream
Convolutional IFFT & Add

Modulator Interleaver
coder cyclic prefix
Multipath
channel
Impulsive noise AWGN Channel
Remove Pilot based

Viterbi
cyclic prefix impulse Deinterleaver Demodulator
decoder
& FFT canceller
Output bit
Adaptive stream
impulse burst
detection
Fig. 5.16 Modified OFDM system block diagram.
5.4.5 Simulation performance of the combination technique

The considered scheme was tested by using the 8k DVB-T system model. The submodulation
was fixed to 64-QAM and the code rate of 2/3 was used. The impulse noise burst length, L,
was fixed to 500 samples corresponding to 54.5 s and assumed to occur only once in every 8
OFDM symbols. We consider the case of a static Rayleigh channel as used in the DVB-T stan-
dards [3]. An SNR of 25 dB was assumed. For comparative purposes, we consider also the case
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 64-QAM, 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 64-QAM, code rate 2/3, Doppler = 25 Hz,
and SNR = 25 dB.
IMPULSIVE NOISE EFFECTS ON TIME SYNCHRONIZATION 51
5.5 IMPULSIVE NOISE EFFECTS ON TIME SYNCHRONIZATION
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.
5.5.1 Considered symbol synchronization methods

The MLSS algorithm presented in [32] is using the correlation proprieties of the received OFDM
signal. We consider an observation of 2N + consecutives samples of the received OFDM
signal r(k), where N and are the FFT length and the cyclic prefix length, respectively. Con-
sequently, these samples contain one complete N + sample OFDM symbol. It is shown that
the maximum likelihood estimate of the symbol timing is given by the following equation
= arg max (), (5.19)

where
+1 +1
X
2 2
X

() = 2 r(k)r (k + N ) (|r(k)| + |r(k + N )| ),

k= k=
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 log-likelihood 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 log-likelihood function of
given this observation interval is shown to be
M1
1 X
() = m (), (5.20)
M m=0
where m ()is the log-likelihood function of given that only mth frame is observed. Averag-
ing operation improves the result significantly.
5.5.1.2 Median log-likelihood timing offset estimation In impulsive environment

and because of the large timing offset error that the noisy OFDM symbol could generate, the
MSE timing offset will have an impulsive distribution, i.e., few frames will present a high MSE
and the other frames will provide a timing offset MSE around the optimum. Therefore, aver-
aging will spread the impulsive timing offset error over the other frames and will even worsen
the result when compared to the no averaging case. Here we consider a different approach. The
ensemble log-likelihood is computed using the median (non-linear 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 log-likelihood 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.
5.5.2 Simulation settings and results

We have performed simulations to evaluate the performance of the timing synchronization al-
gorithm in impulsive noisy channels. The performance in static and mobile channel cases were
considered [3] [17]. We used the 8k mode DVB-T system [3]. All sub-carriers are modulated
with complex signal points taken from a 16-QAM constellation. Different receiver structures
are investigated. Log-likelihood, average log-likelihood and median log- likelihood estimates
of are used. We evaluate the performance of the estimators for a range of burst duration val-
ues using the mean-squared error E{( )2 } criterion. We also vary the delay spread of the
channel.
The burst is assumed to occur once in the cyclic prefix of every 4 OFDM symbols. A cyclic
prefix of 2048 samples is used. A number of 10000 symbols are simulated for each burst du-
ration value, and 16 symbols are used to compute the average or the median likelihood estimate.
Figures 5.19 shows simulation results for Rayleigh static SFN channel case, for different receiver
structures. It can be seen that the presence of impulse noise could deteriorate the synchroniza-
tion performance severely. The median log-likelihood approach can improve significantly the
performance results. Figure 5.20 shows the MSE perfromance in the mobile SFN case when
MTM is used. It can be seen that using appropriate settings of the MTM filter, it is also possible
to improve the synchronization performance even in severe mobile environment.
IMPULSIVE NOISE EFFECTS ON TIME SYNCHRONIZATION 53
0.14
MSE without average, SFN delay=300 samp.
0.12
MSE with average, SFN delay=300 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
8k DVBT 1/4 GI, 16QAM

Rayleigh static SFN channel
AWGN, SNR = 0 dB
0.06
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
0.07 MSE with average, SFN delay=300 samp.

0.06
MSE with Median, SFN delay=300 samp.
MSE with Median, SFN delay=700 samp.
0.05 MSE with Median, SFN delay=1000 samp.
MSE with MTM(16,4), SFN delay=300 samp.
MSE
MSE with MTM(16,4), SFN delay=700 samp.

0.04 MSE with MTM(16,4), SFN delay=1000 samp.
0.03 8k DVBT 1/4 GI, 16QAM

Mobile SFN channel using
TU6 profile, fd = 100 Hz
0.02 AWGN, SNR = 0 dB
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
Gap-Filler and Common Techniques
for Loop Interference Cancellation
6.1 INTRODUCTION
The terrestrial digital video broadcasting (DVB-T) [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 DVB-T 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 inter-symbol 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 DVB-T system using a single relay station.
A particular problem with the use of gap fillers in DVB-T 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 GAP-FILLER AND COMMON TECHNIQUES FOR LOOP INTERFERENCE CANCELLATION
6.2 SIGNAL AND CHANNEL MODELS
In order to evaluate the performance of the loop interference canceller algorithms, we introduce
the channel model that the DVB-T 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 DVB-T is used in the loop interference cancellation algorithms
studies.
6.2.1 Channel model

Depending on the position of the relay relative to the main transmitter, the DVB-T signal will
face either Ricean or Rayleigh environment. Both environments are modeled as a linear fil-
ters [12], [40] characterized by a complex-valued low pass equivalent impulse response of the
following form:
Lp 1
X
h(t) = l ejl (t l ). (6.1)
l=0
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 DVB-T system, T = 7/64 s for 8 M Hz channels)
[3] as follows:
max
Lp = . (6.2)
T
Using the slow-fading 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 line-of-sight 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.
6.2.2 Loop model

Figure 6.1 shows the signal coupling in digital terrestrial broadcasting in single frequency net-
work (SFN) via a single relay system. The receiver and the transmitter in the relay station are
SIGNAL AND CHANNEL MODELS 57
Coupling
Host station wave

Transmission
antenna
Receiver
Host station Reception antenna
Relay broadcast
station
Fig. 6.1 Simple configuration of DVB-T 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 broadcast-wave relay SFN used in our basic study is shown in Figure 6.2.
Host station Relay station Receiver
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 broadcast-wave 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 signal processing in the relay system introduces a delay T1 .
The coupling path has either one or multi-tap profile with a maximum delay T2 .
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 DVB-T system sampling period.
The Ricean factor P 2 is given as follows :
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
6.2.3 A simple gap filler configuration

Lets consider an example of a simple gap filler where we neglect the effect of the channel be-
tween the main transmitter and the repeater. We also assume a one tap model of the loop, i.e.,
T2 = 0, and T1 = M .
Figure 6.4 shows a block diagram of the simplified gap filler configuration. As shown in Fig-
ure 6.4, the system due to the feedback operates as an "echo generator" [44]. The input signal,
y(n) is amplified, delayed, and received again by the repeater. That delayed signal is added to
the current received signal, and the process is repeated. Consequently, the impulse response of
the complete system is the sum of the one of the isolated system without feedback, and a "train"
of delayed replicas of the same response. Those replicas can be seen as echoes generated by the
gap-filler with feedback.
Figure 6.5 shows that situation in the time domain. It can be observed that the relative at-
tenuation between one pulse and its following replica is equal to the gain margin, and the delay
between replicas is the delay value M . Therefore, the higher the value, and the smaller the gain
margin, the longer the overall impulse response of the complete system.
In fact, the real situation is more complex, because the coupling mechanism is a combination
of different coupling paths, each one with its own delay, amplitude and phase, because of the
multipath environment near the antennas.
T2=0
Magnitude
a0
response
T1=M
Samples
y(n) s(n)
g
Fig. 6.4 A simple loop interference model.

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
Fig. 6.5 The gap filler as an echo generator.
6.3 COMMON TECHNIQUES FOR LOOP INTERFERENCE CANCELLATION
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.
6.3.1 Channel estimation based frequency domain adaptive cancellation

algorithm
This algorithm has been introduced in [36]. It is a channel estimation based frequency domain
adaptive canceling scheme. The canceller uses an adaptive FIR filter where the transfer function
is determined from the scattered pilot carriers that are contained in the QAM-OFDM signals.
The analysis in [36] was based on the assumption that the transmitter and the repeater have a
line of sight connection. Therefore, the analysis neglects the possible effect of the frequency
selective channel that the transmitted signal from the main transmitter to the repeater could face.
Figure 6.6 shows the model that has been used in [36]. It includes the canceling path W (z).
The repeater, through conventional channel estimation methods, evaluates the channel impulse
response for adaptation of the FIR filter W (z). Analytically by considering the model of Figure
6.6, if we neglect the noise terms we end up with the following equation:
S(z) 1
F (z) = = (6.8)
X(z) 1 gC(z) + W (z)
We define the error function Er (z) = gC(z) W (z).

COMMON TECHNIQUES FOR LOOP INTERFERENCE CANCELLATION 61

+ + +
-
+

Fig. 6.6 A model for the broadcast-wave relay SFN.
Consequently, the loop interference cancellation process is controlled by minimizing Er (z)

that can be described in terms of the transfer function F (z) as follows:
1
Er (z)
=1 . (6.9)
F (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 ISDB-T based
systems [45], [46], [47], we can extend its use to the DVB-T system. Because of the slight
differences in the pilot patterns in ISDB-T and DVB-T 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 DVB-T 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 DVB-T system were carried out. Figure 6.7, shows the
BER versus SN R at the receiver.
Table 6.1 Parameters of the loop interference.
Parameter descriptions Parameter values
Loop amplitude ratio LAR = +1 dB

Loop interference delay time T1 = 10.09 s
Loop interference phase M = 45 deg.
Relay station noise C/Nb = 40 dB
Three curves for the cases without canceller, with canceller, and no loop interference are
shown. It can be seen that for these parameters, the frequency-domain 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.
0
10
1
10
No loop interference
With Canceller
No Canceller
BER after Viterbi

2
10
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 single-tap loop with LAR
of 1dB. AWGN channel is assumed between the relay and the receiver. 64-QAM 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 LMS adaptive loop interference cancellation algorithm

Due to the need for OFDM demodulation and frequency-domain channel estimation, the com-
putational complexity and storage requirements of algorithms using scattered pilots [36] are in
general considerably high which limits the usability of these algorithms. Fortunately, iterative
adaptive algorithms can provide a simple and efficient structure for loop interference cancel-
lation. In the following, we review two approaches using the LMS adaptive algorithm [49]
[50].
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.
Gain of the
relay
Final Gain
Initial Gain
A1 A2 Time
Fig. 6.8 Variable relay gain.
To the DVB-T
Coupling path receiver
c(n)
r(n)
i(n) s(n)
g(n) w(n)
From the
transmitter
AWGN
Nb(n) Error LMS
Fig. 6.9 Standard adaptive LMS loop interference cancellation algorithm.

6.3.3 LMS loop interference cancellation with spectrum whitening

This method has been introduced in [50]. Its aim is to provide faster convergence and stable
operation than the standard LMS algorithm when used to cancel loop interference in OFDM
systems. The update of the LMS coefficients can be performed as follows:
w(n + 1) = w(n) + s(n)s (n)/Pe (6.10)

where w(n), n = 1, ..., N are the LMS filter coefficients, s(n) is the sampled data at the relay
amplifier input, is the step size, and Pe is the expected power of the received signal. This
alternative algorithm can also cancel the loop interference, but the cancellation ability is lim-
ited because OFDM signal is not ideally white. The cancellation ability can be improved by
modifying the algorithm. The modified adaptation equation is given as follows:
w(n + 1) = w(n) + s, (n)s, (n)/Pe (6.11)

where
s, (n) = s(n) + v(n)
Here v(n) is a random noise signal containing spectral components outside the OFDM band
such that the signal s, (n) appears to be white random noise (hence the name of this algorithm).
In DVB-T systems, a number of carriers at the bottom and top end of the OFDM spectrum
will be left out (set to zero) in order to allow for separation between channels (guard band). In
8k mode the width of the guard band is 687 samples.
To evaluate the performance of the modified LMS adaptive loop interference cancellation
algorithm, we consider the case of Ricean (P 1 = 10) and Rayleigh (P 1 = 0) channels, where
the relative power, phases and delays are given in Table A.1. Again we use the parameter values
of Table 6.1 for the loop interference model. We consider the case where the generated and
added random noise v(n) is white and has Gaussian distribution with variable variance P . The
LMS filter parameters are: 256 tap coefficients and a step size of = 104 . Different settings
for the modified LMS based loop interference cancellation are simulated. For the random noise
v(n), variable bandwidth, W1 , and different noise powers are considered. By W1 we mean the
number of samples in the OFDM guard band where v(n)is added. The length of W1 is varying
from 0 to 687.
In the case of Ricean channel (Figure 6.10), the algorithm cancels the loop interference ef-
fectively even with small noise amplitude, and limited window length (P =-20 dB, W1 = 400
samples), assuming a signal variance of 1. In the case of Rayleigh channel (Figure 6.11), the al-
gorithm can cancel the loop interference effectively only for high noise power levels and wider
bandwidths (P = 0 dB, W1 =687 samples).
LMS loop interference cancellation performance: Ricean channel

3
10
BER after Viterbi 4

10
Loop+canceler: P = 50 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. Single-tap loop, LAR = 1 dB, SNR=25
dB, 64-QAM and a code rate of 2/3 are considered.
LMS loop interference cancellation performance : Rayleigh channel

1
10
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. Single-tap loop, LAR = 1
dB, SNR=25 dB, 64-QAM 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 DVB-T 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.
7.2 AUTOCORRELATION BASED CANCELLER ALGORITHM
7.2.1 Description of the method

In this algorithm [P6], the idea is to consider the same model for the loop interference as used in
[36], i.e., a delayed one tap model. In addition, we consider the presence of a Rayleigh channel
between the relay and the main transmitter.
In the following, we present a simple example where the basics of the autocorrelation method
are described. Figure 6.4 shows a simple block diagram, where a received signal y(n) is as-
sumed to be Gaussian distributed with unit variance. s(n) is the amplified signal that will be
also coupled back to the receiver antenna through the coupling path. The aim is to estimate M
and a0 shown in Figure 7.1.
Lets take the autocorrelation of the signal s(n). This is given in the following equation:
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 DVB-T 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.
Delay of M taps with respect to the main impulse.
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
Fig. 7.1 Impulse response of one-tap loop interference model.
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
Fig. 7.2 The magnitude of the sequence Q(l).

AUTOCORRELATION BASED CANCELLER ALGORITHM 69
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
Fig. 7.3 The angle of the sequence Q(l).
Host station Relay station
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 )
Extract first 2*GI Q(k )

samples in each Second maximum
Autocorrelation | . |
OFDM symbol position
kM
Q( k M )
Delay
Fig. 7.4 Block diagram of the autocorrelation method diagram.

Figure 7.4 shows the general block diagram of the autocorrelation method for loop interference
cancellation in DVB-T 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
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
Fig. 7.5 BER in the Ricean channel case with LAR of 1dB using the basic autocorrelation method,
64-QAM , and a code rate of 2/3
7.2.2 Enhanced autocorrelation method for multi-tap loop with exponential

profile
The main limitation of the earlier discussed algorithm is that it is restricted to operate with one
path feedback loop interference model. Here we give an extension of this algorithm to deal with
multi-tap loop interference having exponential profile [P7].
0
10
1
10
2
10
BER after Viterbi

3
10
4
With canceller
10 No canceller
5
10
10 12 14 16 18 20 22 24 26 28 30
Fig. 7.6 BER in the Rayleigh channel case with LAR of 1dB using the basic autocorrelation method,
64-QAM, and a code rate of 2/3.
7.2.2.1 Multi-tap 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
multi-tap 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].
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 (7.6-7.8), 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 DVB-T and ISDB-T systems
[3] [45] is not actually white because of the cyclic prefix and the virtual carriers. Additionally,
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 (7.6-7.8). 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 DVB-T system that transmits 64-QAM modulated symbols with error correction at the
receiver. We consider the case of AWGN, Ricean and Rayleigh channels. The parameters asso-
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 DVB-T 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 DVB-T 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 DVB-T 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 DVB-T 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 DVB-T 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 DVB-T signal and the WGN signal
cases are converging more rapidly to small values for LAR higher than 4 dB. The MSE perfor-
mance of the WGN case is always better than the DVB-T 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
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
0
10
1
10
Loop interference + canceller
Loop interference + no canceller
2
10
BER after Viterbi
3
10
4
10
8k mode DVBT system

64 QAM, 2/3 code rate
5
10 Rayleigh channel.
Channel loop : LAR (dB)= 4dB, = 1.
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
7.3 DIVERSITY METHOD
c(n)
h(n)
x(n)
y(n) z(n)
h(n) f( y(n), y(n) ) g
y(n)
c(n)
Channel Relay
Fig. 7.11 DVB-T repeater using diversity.
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.
7.3.2 Diversity scheme

Figure 7.11 shows the block diagram of the DVB-T repeater where antenna diversity is con-
sidered. For simplicity we consider here only two branches. As shown in Figure 7.11, the
transmitted DVB-T signal x(n) from the main transmitter arriving at the receiver antennas of
the repeater will pass through different channels, h1 (n) and h2 (n) assumed to be uncorrelated
(this can be accomplished with a sufficient receiver antenna spacing in the repeater). The chan-
nels h1 (n) and h2 (n) are also assumed to be static Rayleigh channels (no line-of-sight between
the main transmitter and the repeater: indoor environment case). c1 (t) and c2 (t) are modelling
the couplings for different branches. As mentioned in the previous paragraph, the coupling
profiles are assumed to be slowly changing in time. Based on the signals y 1 (n) and y 2 (n),
a frequency domain analysis is performed in order to select the branch that provides higher
signal-to-interference ratio. Analytically, the frequency domain transforms of the signals y 1 (n)
and y 2 (n) are given by the following equations:
DIVERSITY METHOD 77
Y 1 (k) = H 1 (k)X(k) + C 1 (k)Z(k) + N 1 (k)

(7.11)
Y 2 (k) = H 2 (k)X(k) + C 2 (k)Z(k) + N 2 (k)
where k is the subcarrier index, X(k) is the transmitted symbol, H 1 (k) and H 2 (k) are the chan-
nel responses in different branches. N 1 (k) and N 2 (k) are additive noise assumed to be Gaussian
distributed and uncorrelated. C 1 (k) and C 2 (k) are the channel responses of the couplings. Z(k)
is the repeated signal to be transmitted to the users.
In order to estimate H 1 (k) and H 2 (k), we make an initialization step at the startup of the
repeater. In the initialization period, the repeater is turned off (g = 0). The multipath chan-
nel estimation is carried out, like in standard DVB-T receiver, using scattered pilots transmitted
within the OFDM symbols. In the initialization step the equation (7.11) is reduced to:
Y 1 (k) = H 1 (k)X(k) + N 1 (k)

(7.12)
Y 2 (k) = H 2 (k)X(k) + N 2 (k)
and the channel responses at each branch can be estimated as follows:
H 1 (k) = Y 1 (k)/X(k) = H 1 (k) + N 1 (k)/X(k)

(7.13)
H 2 (k) = Y 2 (k)/X(k) = H 2 (k) + N 2 (k)/X(k)
When the initialization period is completed, we switch to the standard mode of the repeater
operation (g 6= 0). Therefore, the system of equations in (7.11) can be written as in (7.14). Here
we substitute the channel response estimates H 1 (k) and H 2 (k) and we neglect the effect of the
additive Gaussian noise.
Y 1 (k) = H 1 (k)X(k) + C 1 (k)Z(k)

(7.14)
Y 2 (k) = H 2 (k)X(k) + C 2 (k)Z(k)
The couplings at each branch are then given by:
C 1 (k) = (Y 1 (k) H 1 (k)X(k))/Z(k)

(7.15)
C 2 (k) = (Y 2 (k) H 2 (k)X(k))/Z(k)
We can use equation (7.15) to estimate the amplitude of the coupling at each branch and we
select Y (k) corresponding to the branch that has less coupling. Without loss of generality, if we
set the gain of the repeater to unity (g = 1), the criteria that will be used to determine the best
branch is defined as follow:
if kC 1 (k)k kC 2 (k)k then Z(k) = Y 1 (k)
(7.16)
else Z(k) = Y 2 (k)
7.3.3 Simulation results

The considered scheme was tested by using the 2k DVB-T system model [3]. The sub-modulation
was fixed to 16-QAM and the code rate was 2/3. One tap loop interference case is considered
for both couplings. The LAR defined in Section 6.2 is chosen as a variable parameter and the
delay spread is fixed to 10 s. The fading of the feedback is simulated by considering a TU6
profile (Table A.2) with different Doppler values. We consider the case of a static Rayleigh
channel models for h1 (n) and h2 (n) as used in DVB-T standards (Table A.1). An SNR of 25
dB at the receiver side was assumed. For comparative purposes, we include also a case without
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 DVB-T
system to loop interference is more limited when compared to the static case.
0
10
1
10
2
10
No diversity, fd= 2 Hz
BER after Viterbi

3 No diversity, fd= 10 Hz
10
No diversity, fd= 20 Hz
With diversity, fd= 2 Hz

4
10

5
10
6
10
7
10
20 18 16 14 12 10 8 6 4
Average loop amplitude ratio
Fig. 7.12 BER performance of the DVB-T repeater with diversity: DVB-T, 2k mode, 16QAM, 2/3 code
rate, variable LAR and Doppler frequency cases.
7.4 ENHANCED VERSION OF THE CHANNEL ESTIMATION BASED

FREQUENCY DOMAIN ADAPTIVE CANCELLATION ALGORITHM
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 DVB-T 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
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 single-tap loop
with LAR of 1dB. Rayleigh channel is assumed between the main transmitter and the relay. 64-QAM
and a code rate of 2/3 are considered.
7.5 ENHANCED LMS LOOP INTERFERENCE CANCELLATION
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.
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
The absolute value of the error function

1
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, 64-QAM, 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
BER after Viterbi
3
10
With canceller
No canceller
4
10
5
10
5 10 15 20 25 30 35
Fig. 7.15 BER in adaptive LMS loop interference cancellation case: one tap loop interference model
with LAR of 1dB, 64-QAM, and a code rate of 2/3.
0
10
With canceller
No canceller
1
10
2
10
BER after Viterbi
3
10
4
10
5
10
5 10 15 20 25 30 35
Fig. 7.16 BER in adaptive LMS loop interference cancellation case: multitap loop interference model
with LAR of 2dB, 64-QAM, and a code rate of 2/3.
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
DVB-T 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, inter-operability issues of DVB-T 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 DVB-T signal and DVB-C is addressed
[55]. We study the quality of DVB-T 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 head-end, from DVB-T to DVB-C. DVB-C uses conventional single carrier QAM trans-
mission, so the signal processing functionalities are quite different from DVB-T; however, the
error control coding and higher layers have a lot of commonality with DVB-T. In order to avoid
this costly conversion and to transmit DVB-T 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 single-carrier 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 DVB-T transmission with sufficient quality [P10][P9].
8.2 HISTORY OF CABLE TELEVISION
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 over-the-air
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 DVB-T 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 coax-based facili-
ties with fiber-optic technology and introducing digital communication services-including high
speed Internet access, telephony, data, and interactive television. The combination of broadband,
hybrid fiber-coax infrastructure and high-speed 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].
8.3 CABLE DISTRIBUTION SYSTEM
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 head-end, 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 fiber-optic cables to a sub-
scribers television or set-top box. A variety of equipment is used at the head-end, including
antennas and satellite dishes, preamplifiers, frequency converters, demodulators, and modula-
tors, processors, and scrambling and de-scrambling 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 head-end 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 fiber-optic components. At the sub-
scriber premises, the drop cable may be connected directly to a "cable ready" television set or
connected first to a set-top box. Set-top boxes provide frequency and channel conversion for
older television sets and de-scrambling of pay service signals. The drop cable may also be con-
nected to other interactive equipment, including telephony terminal equipment or high-speed
cable modems with return capability via a telephone line or two-way 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 DVB-T signal in a CATV network.
CABLE DISTRIBUTION SYSTEM 85
Satellite Reception
Trunk Cables
Local Signals
Head-End
Trunk Amplifiers
Distant Signals
Studio for local Origination

Programming
Fig. 8.1 Cable TV system head-end.
Line Extender
Amplifiers
Drop
Bridger Amplifier
Taps
Trunk System Distribution Subscriber Drop

System
Fig. 8.2 Cable TV system distribution plant.

8.4 GENERIC CATV CHANNEL MODEL
8.4.1 CATV channel impairments

CATV impairments that affect the OFDM signal can be mainly classified into linear, non-linear
distortions and phase noise.
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.
.
.
. .

+
. .
. .
. .

Fig. 8.3 Gaussian noise channel in OFDM based system.
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 micro-reflections. They are
caused by individual slight errors in impedance match. The severity of the mismatch is measured
by the magnitude of the return-loss 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
then the received samples are given in the absence of noise by

M
X
X (k) = X(k) am ei2mk/N . (8.5)
m=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 Non-linear distortion The OFDM signal carried by a large number of subcarriers
is very sensitive to non-linear 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 non-linear OFDM systems making use of the more
general results on the effects of non-linear distortions of Gaussian signals [62] [63].
Analog-to-digital (A/D) converters, mixers, and power amplifiers in the CATV systems are
usually the major sources of non-linear distortions due to the limited range that they allow for
signal dynamics. It is possible to distinguish between two different classes of non-linear 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 so-called
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 non-linear distortion due to power amplifiers is present, so in the fol-
lowing sections we will concentrate our analysis on the second class of the non-linear distortion
[57].
We consider the modulation of a complex symbols X(k) by N-point inverse discrete Fourier
transform (IDFT). The result, x(n), is distorted by the transfer characteristic function g(x) that
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).
. .

. .

. .
. .
. .

Fig. 8.4 OFDM with nonlinearity.
To analyze the effect of the non-linearity on the subcarrier samples Y(k), we consider the
error caused by the channel non-linearity that can be defined by:
E(k) = X(k) Y (k). (8.8)

The transmitted symbols, X(k), are independent and identically distributed complex num-
bers with zero mean and variances r2 and i2 for real and imaginary parts, respectively [58]
[64]. After the N-point IDFT process, the real and imaginary variances are equal and given 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
x(n) = |x(n)| ei arg(x(n)) , (8.10)
where the phase, arg(x(n)), is uniformly distributed between 0 and 2. The magnitude of
x(n) is Rayleigh distributed. Lets consider the soft-limiting nonlinearity that can be defined as
follows:

L if x L
g(x) = x if L x < L (8.11)
L if xL

where L is the amplitude limiting level.

When applying x(n) to the soft limiter model, the output distorted signal y(n) is given by
y(n) = g(x(n)) = g(|x(n)|)ei arg(x(n)) . (8.12)

The total average noise power introduced by the limiter can be written as
Z
E e2 (n) = x2 pe (x)dx,

(8.13)
0
where pe (x) is defined by [65]

( )
Z L
pe (x) = (x) p(x)dx + p(x + L) , x 0 , (8.14)
0
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 Q-function 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 back-off 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 non-linear 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 non-linearity
in CATV systems.
8.4.1.4 Phase noise For conventional Analog TV systems (like PAL, SECAM and
NTSC), the phase noise of the front-end 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 non-orthogonality 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.
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:
r(n) = x(n)ej(n) , (8.20)
and it is demodulated by means of DFT :
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
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
The derived mask

30
The typical oscilator
Relative power (dB)
power spectral density

40
50
60
70
80
90
0 0.5 1 1.5 2 2.5
Frequency offset (Hz) 4
x 10
Fig. 8.5 Phase noise mask.
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/down-conversion, 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
E2

+
FIR = (a0 , a1 ,.., aM )
2
D
Fig. 8.6 The generic CATV channel model .
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.
8.5 PRACTICAL CATV CHANNEL MODEL
8.5.1 Linear distortion

It is possible to model this part with an FIR filter where the order and the coefficients are carefully
chosen. In our simulation model, the echo profile of the Nordig specification is considered [74].
Nordig is specifying a common platform for Digital Television to be used within the Nordic re-
gion (Denmark, Finland, Iceland, Norway and Sweden). It is suggested that 5 echoes should be
present simultaneously. A template for echoes provided by Nordig specification is given in Fig-
ure 8.7. The corresponding multipath model taps that will be used in the simulations are given
in Table 8.1. The phases are not specified in the specifications. In the simulations, we assume
that all the taps have the same phase of /4.
Table 8.1 The multipath taps that model the linear distortion in the CATV channel
Tap Delay in ns Tap amplitude in dB Phase in radians

a0 1 6 /4
a1 111 16 /4
a2 221 27 /4
a3 331 36 /4
a4 441 36 /4
PRACTICAL CATV CHANNEL MODEL 93
In DVBT system Ts= 7/64 microsecond

10
Amplitude with respect to the total signal power

15
20
25
30
35
40
0 50 100 150 200 250 300 350 400 450 500
Delay with respect to the main impulse (ns)
Fig. 8.7 Template for echoes provided by Nordic specification.
8.5.2 Non-linear distortion

It was shown that the equivalent system model for soft limiting channel could be simplified to an
2
additive white Gaussian noise with variance E . Based on [64] this is also a sufficient model for
the intermodulation interference between different channels of the CATV network. In [75], the
requirements referring to the performance limits which shall be obtained by the CATV network
are specified. The interference to television channels was classified into single frequency inter-
ference and multiple frequency intermodulation interference. The single frequency interference
may result from intermodulation or the presence of other interfering signals (ingress signals,..).
It was stated in [75] that at any system outlet the level of any unwanted signal within the system
shall be such that the lowest carrier to interference ratio within a wanted television channel be
not less than 57 dB for AM signals. Similarly, the level of the multiple frequency intermodula-
tion interference, in any wanted television channel, shall be such that the carrier to interference
ratio is not less than 57 dB. Consequently, in our CATV model, the level of white noise vari-
2
ance E that we will use in our simulations to model the non-linearity behavior is chosen to be
at least 57 dB below the main carrier.
8.5.3 Phase noise

To derive the characteristics of the phase noise mask that will be used in the simulation model,
some measurements were carried out based on a typical converter used generally in analog
CATV systems [76] [77]. The measured spectrum of the signal coming out from the actual con-
verter can be seen in Figure 8.8. The frequency was normalized so that 0 Hz corresponds to the
exact UHF output frequency of the converter. In the case of DVB-T, we are especially interested
in the noise power of the low frequency components. If the slowly changing phase is removed
from measurement results, the remaining phase noise spectrum can be modeled as shown in Fig-
ure 8.9. It includes a phase noise mask to be used in the simulation. Such mask represent the
long-time averaged power spectral densities of the frequency converters local oscillator.
To assess the effects of the phase noise, a variable scaling parameter used to scale the level
of the phase noise introduced by the oscillator at the 0 Hz carrier frequency will be used in the
10
20
Relative power, dBc

30
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.
simulation model. This scaling parameter can be explained as follows:
SPdB = LP MdBc LP SdBc (8.25)
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 DVB-T 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. .
8.6 SIMULATION OF THE DVB-T SIGNAL OVER THE CATV CHANNEL
Our aim here is to study the quality of the DVB-T transmission over the cable TV network.
Usually, DVB-T signals are converted in the head-end to DVB-C format [2] when distributed in
the cable TV channel.
We consider the possibility of avoiding this costly conversion and to transmit DVB-T signal
directly in the existent cable channel. Firstly, we study the sensitivity of the DVB-T for each
impairment parameters characterizing the CATV channel, like linear distortion and phase noise.
Secondly, a performance comparison against DVB-C is also carried out. There are two major
differences between DVB-T and DVB-C. The DVB-C [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 DVB-C is not using inner coding. To achieve the appropriate level of error
SIMULATION OF THE DVB-T SIGNAL OVER THE CATV CHANNEL 95
10
20
Relative power, dBc

30 The phase noise mask
40
50
60
70
80 2 3 4 5
10 10 10 10
Frequency Hz
Fig. 8.9 Phase noise mask for modelling.
protection required for the transmission of a digital signal in cable TV channel, a forward error
correction based only on Reed-Solomon encoding is used. In most cases, the DVB-C system
utilizes 64-QAM, but lower-level systems, such as 16-QAM and 32-QAM, and higher-level sys-
tems such as 128-QAM and 256-QAM 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.

Fig. 8.10 DVB-C system, transmitting side.
!"# $"% '( (,#./-"0 1 4-55$ $"#-,+

&
%$)!%*+,#-!" $2*,+-3,#-!" %$.!%$
67)8!+ #! 87#$ =!">!+*#-!",+ :$$%; 6!+!)!"

),99-"0 4$;-"#$ +$,>$ %$.!%$ <-", 7 %,#,
Fig. 8.11 DVB-C system, receiving side

8.6.1 Simulation parameters and settings

In the following simulations we will consider an 8k DVB-T system. Different sub-modulations
will be used, like QPSK, 16-QAM and 64-QAM. The code rate 2/3 will be simulated. A simple
channel estimation technique based on a frequency interpolation is used. Some parameters will
be varied depending on the impairments; others will be fixed for all the simulations.
8.6.2 Sensitivity to linear distortion

The BER performance of the DVB-T signal transmission in CATV network is shown in Figure
8.12. Here, only the linear distortion introduced by the echo profile of the channel is considered.
The guard interval of 256 samples, i.e., of Tu /16 is chosen.
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 DVB-T system mildly when compared to the AWGN channel. About 3
dB higher SNR is needed for each DVB-T system to achieve a sufficient performance for each
sub-modulation.
8.6.3 Sensitivity to phase noise

In Figures 8.13, 8.14, and 8.15, plots of the BER versus SNR for QPSK, 16QAM, and 64QAM
are given. Here also, a guard interval of Tu /16 s, is chosen in the simulation. Only phase
noise impairment is present.
In the case of CATV channel, the DVB-T signal is mainly affected by the phase noise distor-
tion. For lower constellation sizes, like QPSK, the system is not very sensitive to the phase noise
impairment. However, for higher-order constellations, like 64-QAM, it is clear that the DVB-T
system performance gets worse in the presence of phase noise. Using as reference the target
BER (2.104 ), we can conclude that the phase noise scaling value of about 10 dB is needed
to guarantee a sufficient performance of the DVB-T system when 64-QAM is used. Smaller
scaling values can be tolerated if smaller sub-modulations like QPSK and 16-QAM are used.
DVBT 8K mode, QPSK, 2/3 CR

0
10
Scaling = 2 dB
Scaling = 4 dB
Scaling = 6 dB
AWGN channel
1
10
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 DVB-T to the phase noise with different levels of scaling parameters: QPSK,
2/3 Code rate.
DVBT 8K mode, 16QAM, 2/3 CR

0
10
Scaling = 2 dB
Scaling = 4 dB
Scaling = 6 dB
1 Scaling = 8 dB
10
Scaling = 10 dB
AWGN channel
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 DVB-T to the phase noise with different levels of scaling parameters: 16-QAM,
2/3 Code rate.
DVBT 8K mode, 64QAM, 2/3 CR

0
10
1
10
2
10
BER after Viterbi

3
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 DVB-T to the phase noise with different levels of scaling parameters: 64-QAM,
2/3 Code rate.
8.6.4 DVB-T versus DVB-C in CATV channel

As mentioned in the earlier sections, the CATV channel is different from the terrestrial channel.
In addition, DVB-C doesnt use convolutional encoding, as it is the case for the DVB-T system.
For a fair comparison between the two systems in the presence of the CATV channel impair-
ments and, considering especially the performance of both systems to the phase noise distortion,
we need to tune the system parameters in such a way that about the same transmitted bit-rate is
achieved.
In [2], it is stated that the useful bit-rate for transparent transmission in the 16-QAM case,
when 8 MHz bandwidth is considered is of 25.2 Mbits/s. Similar useful bitrate can be achieved
with DVB-T system when 64-QAM is used. In [3], it is mentioned that if we use for example
2/3 code rate and Tu /32 as a guard interval, we can have a useful bit rate of 24.13 Mbits/s.
Alternatively, if we use 3/4 code rate and Tu /8 as guard interval we can achieve a useful bit rate
of 24.88 Mbits/s.
Others combinations of the code rates and guard intervals can be also found to assure a
comparable useful bit rate with DVB-C 16-QAM case.
For the sake of simplicity, we will use only these two cases to conduct the performance
comparison of both systems in BER sense.
Figure 8.16, shows the BER performance of the DVB-C 16-QAM, and DVB-T 64-QAM in
the presence of phase noise. Different phase noise scaling parameter values are simulated. .
It can be seen that the DVB-C 16-QAM system tolerates better the phase noise distortion than
DVB-T 64-QAM when 3/4 code rate is used. When using code rate of 2/3, DVB-T tolerability
to phase noise can be enhanced. Furthermore, DVB-T 64-QAM and 2/3 code rate can even
assure better performance than DVB-C system when the the phase noise scaling value of 10 dB
is used.
As a conclusion, the DVB-T system parameters can be chosen to allow the OFDM signal to
be transmitted successfully in the CATV even in the worst channel conditions. Special attention
must be paid to the phase noise distortion, which is the main impairment that can influence the
OFDM performance in the CATV channel. Simulations confirm the possibility to transmit the
DVBT & DVBC systems with simular user data rates

0
10
DVBT 64QAM
1
10
2
10
DVBC 16QAM
BER
3
10
CR=3/4, Scaling = 8 dB
4
10
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 DVB-T and DVB-C to the phase-noise scaling parameter.
DVB-T signal over the current CATV networks without any conversion from DVB-T to single
carrier DVB-C system. The results help in the design and the optimization of efficient OFDM
systems for cable transmission.
Chapter 9
Conclusions
The DVB-T system for terrestrial broadcasting is probably the most complex DVB delivery sys-
tem. Originally, the DVB-T 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 DVB-T system. Its results come as a participation
of the DVB-T 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 DVB-T technol-
ogy.
In Chapter 4 we began the first issue. The tolerance of the DVB-T 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 time-frequency 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 DVB-T 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 DVB-T 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 multi-tap 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
DVB-T in-door repeaters. The case of slowly mobile loop interference environment was stud-
ied. The diversity approach seems to relatively improve the performance of the DVB-T 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 DVB-T 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 inter-operability issue of DVB-T system should be examined
and compatibility of the existent standard with other DVB broadcasting services has to be further
investigated.
In Chapter 8, the inter-operability issue was addressed. As an example, the compatibility of
the DVB-T and DVB-C was investigated. We studied the quality of DVB-T transmission over
the cable TV network. Generally, when terrestrial digital TV signals are distributed in cable
TV, a conversion is needed in the head-end, from DVB-T to DVB-C (single carrier). In order to
avoid this costly conversion and to transmit DVB-T 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 DVB-T
transmission with sufficient quality.
9.1 FUTURE RESEARCH
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 DVB-T system to accommodate such emerging situations and to consider the
benefits that new state-of-the-art technologies could bring to DVB-T.
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 DVB-T in mobile environments are still needed. For
instance, these difficulties can occur when Hybrid DVB-H/GSM mobile broadcast system is
FUTURE RESEARCH 103
considered. For the DVB-T(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 DVB-T(H)
networks. The feedback loop interference needs to be studied further and low complexity loop
interference cancellers need to be developed.
The DVB-T standard enhancement is still going on. We already start discussing about the
DVB-T2 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 DVB-T.
In our group, we are also studying the enhancement of the DVB-T receiver from the mobility
sense. New ICI canceller schemes are being proposed. Additionally, optimization of the pilot
pattern in DVB-T is also under consideration. The optimization is aimed at higher capacity and
larger Doppler tolerance.
References
[1] ETSI EN 300 421: Digital Video Broadcasting (DVB); Framing structure, channel coding
and modulation for 11/12 GHz satellite services.
[2] ETSI EN 300 429: Digital Video Broadcasting (DVB); Framing structure, channel coding
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[3] ETSI EN 300 744: Digital Video Broadcasting(DVB);Framing structure, channel coding
and modulation for terrestrial television, Jan. 2004.
[4] ETSI EN 302 304: Digital Video Broadcasting(DVB);Transmission System for Handheld
Terminals (DVB-H), Nov. 2004.
[5] J. Bingham, Multicarrier modulation for data transmission: an idea whose time has come,
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[7] S. Weinstein, and P. Ebert, Data transmission by frequency-division multiplexing using

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[8] A. Ruiz, J. Cioffi, and S. Kasturia, Discrete multiple tone modulation with coset coding
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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
i i (lin) i (s) i (rad)

1 0, 057662 1, 003019 4, 855121
2 0, 176809 5, 422091 3, 419109
3 0, 407163 0, 518650 5, 864470
4 0, 303585 2, 751772 2, 215894
5 0, 258782 0, 602895 3, 758058
6 0, 061831 1, 016585 5, 430202
7 0, 150340 0, 143556 3, 952093
8 0, 051534 0, 153832 1, 093586
9 0, 185074 3, 324866 5, 775198
10 0, 400967 1, 935570 0, 154459
11 0, 295723 0, 429948 5, 928383
12 0, 350825 3, 228872 3, 053023
13 0, 262909 0, 848831 0, 628578
14 0, 225894 0, 073883 2, 128544
15 0, 170996 0, 203952 1, 099463
16 0, 149723 0, 194207 3, 462951
17 0, 240140 0, 924450 3, 664773
18 0, 116587 1, 381320 2, 833799
19 0, 221155 0, 640512 3, 334290
20 0, 259730 1, 368671 0, 393889
111
112 DEFINITION OF THE CHANNEL MODELS USED IN PERFORMANCE EVALUATIONS
Table A.2 Characteristics of the 6-tap typical urban (TU6) channel model
Tap Delay (s) Power (dB) Doppler spectrum

1 0 3.0 Classical
2 0.2 0 Classical
3 0.5 2.0 Classical
4 1.6 6.0 Classical
5 2.3 8.0 Classical
6 5.0 10.0 Classical
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Julkaisu 668 Publication 668

Studies in Digital TV Signal Processing: Impulse Noise

Tampereen teknillinen yliopisto - Tampere University of Technology

2 OFDM: Orthogonal Frequency Division Multiplexing 7

3 Digital Video Broadcasting: Technical Overview 11

3.5.2 Channel modeling . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19

4 Impulsive Noise Modeling, Effects and Mitigation in DVB-T systems 23

5 Pilot Based Impulse Noise Canceller Enhancements 33

5.4.5 Simulation performance of the combination technique . . . . . . . . . . 49

6 Gap-Filler and Common Techniques for Loop Interference Cancellation 55

7 Enhanced Algorithms for Loop Interference Cancellation 67

8 DVB-T Signal in Cable TV Networks: Limitations and Requirements 83

8.6 Simulation of the DVB-T signal over the CATV channel . . . . . . . . . . . . 94

2.1 OFDM transmission principle. . . . . . . . . . . . . . . . . . . . . . . . . . 8

2.2 The concept of adding a cyclic prefix in case of a multipath channel. . . . . . . 9

2.3 Multipath propagation in SFN network. . . . . . . . . . . . . . . . . . . . . 10

3.1 DVB-T functional block diagram. . . . . . . . . . . . . . . . . . . . . . . . 12

3.2 Packet coming from MPEG-2 transport MUX. . . . . . . . . . . . . . . . . . 14

3.3 Reed-Solomom RS(204,188) error protected data packet. . . . . . . . . . . . . 14

3.4 Functional block diagrams of the outer interleaver and de-interleaver . . . . . . 15

3.5 The inner coder with two outputs . . . . . . . . . . . . . . . . . . . . . . . . 15

3.6 Inner coding and inner interleaving block diagrams . . . . . . . . . . . . . . . 15

3.8 Basic uniform constellations used in DVB-T system . . . . . . . . . . . . . . 16

3.9 Location of the scattered pilots in the OFDM frame structure. . . . . . . . . . 18

3.10 Principle of a Gap-filler. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21

4.1 Impulse noise generation block diagram . . . . . . . . . . . . . . . . . . . . 24

4.2 DTG impulse noise model . . . . . . . . . . . . . . . . . . . . . . . . . . . 26

4.3 Performance of impulse noise reduction algorithm in AWGN channel (16-QAM). 31

4.4 Performance of impulse noise reduction algorithm in AWGN channel (64-QAM). 31

5.1 Description of the enhanced channel estimation methods in impulsive noise

6.3 An example of the impulse response amplitude of the coupling path. . . . . . 58

6.4 A simple loop interference model. . . . . . . . . . . . . . . . . . . . . . . . 59

6.5 The gap filler as an echo generator. . . . . . . . . . . . . . . . . . . . . . . . 60

6.6 A model for the broadcast-wave relay SFN. . . . . . . . . . . . . . . . . . . . 61

6.8 Variable relay gain. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63

6.9 Standard adaptive LMS loop interference cancellation algorithm. . . . . . . . 63

7.1 Impulse response of one-tap loop interference model. . . . . . . . . . . . . . 68

7.2 The magnitude of the sequence Q(l). . . . . . . . . . . . . . . . . . . . . . . 68

7.3 The angle of the sequence Q(l). . . . . . . . . . . . . . . . . . . . . . . . . . 69

7.4 Block diagram of the autocorrelation method diagram. . . . . . . . . . . . . . 69

7.7 Loop interference cancellation based on the iterative autocorrelation method. . 73

7.8 MSE performance of loop amplitude estimation using enhanced autocorrelation

7.9 BER performance of the enhanced autocorrelation method in the AWGN

7.10 BER performance of the enhanced autocorrelation method in the Rayleigh

7.11 DVB-T repeater using diversity. . . . . . . . . . . . . . . . . . . . . . . . . 76

8.5 Phase noise mask. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91

8.16 Sensitivity of DVB-T and DVB-C to the phase-noise scaling parameter. . . . . 99

3G Third Generation Wireless System

DQPSK Differential Quadrature Phase Shift Keying

1.1 HISTORY OF DIGITAL TERRESTRIAL BROADCASTING

1.2 ORGANIZATION OF THE THESIS

1.3 AUTHORS CONTRIBUTION

Impulse noise effects and mitigation

2.2 OFDM PRINCIPLES

Fig. 2.1 OFDM transmission principle.

2.3 OFDM SIGNAL

2.3.1 The definition of OFDM signal

5+6& 78# (%)# (9:;%$ g

A reconstructed decision variable Yk on sub-channel k can be written as

2.3.2 Guard interval

2.3.3 Single frequency networks

Fig. 2.3 Multipath propagation in SFN network.

3.2 GENERAL CONSIDERATIONS