Source 02

Real-Time Power System Disturbance
Identification and its Mitigation using

an Enhanced Least Squares Algorithm
Thip Manmek
A Thesis Submitted to
The University of New South Wales for the Degree of
Doctor of Philosophy
School of Electrical Engineering and Telecommunications
August 2006
CERTIFICATE OF ORIGINALITY
I hereby declare that this submission is my own work and to the best of my knowledge
it contains no material previously published or written by another person, nor material
which to a substantial extent has been accepted for the award of any other degree or
diploma at UNSW or any other educational institution, except where due
acknowledgement is made in the thesis. Any contribution made to the research by
others, with whom I have worked at UNSW or elsewhere, is explicitly acknowledged in
the thesis. I also declare that the intellectual content of this work is the product of my
own work, except to the extent that assistance from others in the project’s design and
conception or in style, presentation and linguistic expression is acknowledged.
(Signed)…………………………..
Thip Manmek
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ABSTRACT
This thesis proposes, analyses and implements a fast and accurate real-time power
system disturbances identification method based on an enhanced linear least squares
algorithm for mitigation and monitoring of various power quality problems such as
current harmonics, grid unbalances and voltage dips.
The enhanced algorithm imposes less real-time computational burden on processing the
system and is thus called “efficient least squares algorithm”. The proposed efficient
least squares algorithm does not require matrix inversion operation and contains only
real numbers. The number of required real-time matrix multiplications is also reduced in
the proposed method by pre-performing some of the matrix multiplications to form a
constant matrix. The proposed efficient least squares algorithm extracts instantaneous
sine and cosine terms of the fundamental and harmonic components by simply
multiplying a set of sampled input data by the pre-calculated constant matrix.
A power signal processing system based on the proposed efficient least squares
algorithm is presented in this thesis. This power signal processing system derives
various power system quantities that are used for real-time monitoring and disturbance
mitigation. These power system quantities include constituent components, symmetrical
components and various power measurements. The properties of the proposed power
signal processing system was studied using modelling and practical implementation in a
digital signal processor. These studies demonstrated that the proposed method is capable
of extracting time varying power system quantities quickly and accurately. The dynamic
response time of the proposed method was less than half that of a fundamental cycle.
Moreover, the proposed method showed less sensitivity to noise pollution and small
iii
variations in fundamental frequency. The performance of the proposed power signal
processing system was compared to that of the popular DFT/FFT methods using
computer simulations. The simulation results confirmed the superior performance of the
proposed method under both transient and steady-state conditions.
In order to investigate the practicability of the method, the proposed power signal
processing system was applied to two real-life disturbance mitigation applications
namely, an active power filter (APF) and a distribution synchronous static compensator
(D-STATCOM). The validity and performance of the proposed signal processing
system in both disturbance mitigations applications were investigated by simulation and
experimental studies.
The extensive modelling and experimental studies confirmed that the proposed signal
processing system can be used for practical real-time applications which require fast
disturbance identification such as mitigation control and power quality monitoring of
power systems.
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In memory of my farther…
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ACKNOWLEDGEMENTS
I would like to express my sincere gratitude to my supervisors, Professor Colin
Grantham, and Dr. Toan Phung, for giving me a good opportunity to work with them,
and stimulating suggestions and encouragement have helped me throughout my research
and the writing of this thesis. I would not have been able to complete this work without
their countless help and encouragement.
I gratefully acknowledge the full financial support from Mahanakorn University of
Technology (Thailand) throughout my PhD study in Australia. Furthermore, I am
deeply indebted to my supervisor for his financial support during my PhD study. Also, I
am thankful to Professor Trevor Blackburn and Professor Faz Rahman for their
financial support which enabled me to attend several international and local
conferences.
I am especially thankful to Dr. Baburaj Karanayil and Mr. Gamini Liyadipitiya for
kindly supporting and giving me many technical advices regarding hardware and
software needed for my project.
Also, I would like to express thanks to my friends in the Energy Systems research group
for the interesting discussions and sincere friendship. I particularly thank Huu Phuc To
for his kind support and valuable suggestions that I received during modeling and
experimenting of the active power filter.
Great thanks from my heart to my husband, Chathura, for his immense support,
unswerving love and encouragement which has enabled me to complete this work. I also
thank my sister and brother, for their never ending love. Finally, I would like to express
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my sincere gratitude to mother and late father, who have been giving me the best
throughout my life. They have encouraged me throughout my academic career, and I
certainly owe all of my success to their loving support and guidance.
Thip Manmek
Kingsford, August 2006
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CONTENTS
ABSTRACT.................................................................................................................... iii
ACKNOWLEDGEMENTS ............................................................................................ vi
CONTENTS.................................................................................................................. viii
LIST OF TABLES ...................................................................................................... xxix
LIST OF SYMBOLS AND ABBREVIATIONS ........................................................ xxx
CHAPTER 1 .................................................................................................................... 1
INTRODUCTION ........................................................................................................... 1
1.1 Overview ..................................................................................................................1
1.2 Classification of Power Quality Problems ...............................................................2
1.3 Importance of the Identification and Tracking of Disturbance Signal Quantities ...5
1.4 Existing Methods of Identification of Disturbance Quantities ................................6
1.5 Scope of this Thesis .................................................................................................9
1.6 Proposed Approach for Identification of Power System Disturbances..................10
1.7 Structure of Thesis .................................................................................................12
CHAPTER 2 .................................................................................................................. 14
REVIEW OF THE EXISTING TECHNIQUES FOR POWER SYSTEM
HARMONIC ESTIMATION ........................................................................................ 14
2.1 Overview ................................................................................................................14
2.2 Non-Recursive Harmonic Detection Methods .......................................................15
2.2.1 Discrete Fourier transform (DFT) and fast Fourier transform (FFT)
method……………………………………………………………………………15
2.2.2 Wavelet transform (WT) …………………………………………………..21
x Continuous wavelet transform (CWT)……………………………………..22

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x Discrete wavelet transform (DWT) ……………………………………….23
x Discrete wavelet packet transform (DWPT) ………………………………24
2.2.3 Conventional least squares (CLS) method ……………………………….29
2.3 Recursive Harmonic Detection Methods ...............................................................33
2.3.1 Kalman filtering (KF) method …………………………………………….33
2.4 Summary ................................................................................................................37
CHAPTER 3 .................................................................................................................. 39
PERFORMANCE OF THE EXISTING TECHNIQUES FOR POWER SYSTEM
HARMONIC ESTIMATION ........................................................................................ 39
3.1 Overview ................................................................................................................39
3.2 The Performance of DFT/FFT Method..................................................................41
3.3 The Performance of Wavelet Transform (WT) Method ........................................46
3.4 The Performance of Conventional Least Squares (CLS) Method .........................50
3.5 The Performance of a Kalman Filtering (KF) Method ..........................................55
x Performance and the order of Kalman filter 59
3.6 Comparison of the Studied Methods......................................................................61
3.6.1 Effect of noise distortion on estimation accuracy ………………………62
3.6.2 Effect of marginal fundamental frequency variation on the estimation
accuracy …………………………………………………………………………64
3.6.3 Transient response time …………………………………………………..65
3.7 Summary ................................................................................................................65
CHAPTER 4 .................................................................................................................. 67
PROPOSED EFFICIENT LEAST SQUARES TECHNIQUE FOR POWER
SYSTEM HARMONIC DETECTION.......................................................................... 67
4.1 Overview ................................................................................................................67
ix
4.2 Mathematical Derivation of the Proposed Efficient Least Squares Algorithm for
Harmonic Detection ........................................................................................................68
4.2.1 Singular value decomposition (SVD) based least squares method ………..69
4.2.2 Proposed efficient least squares technique ……………………………….72
4.3 Performance of the Proposed Efficient Least Squares Algorithm .........................81
4.3.1 Computational complexity ……………………………………………….82
4.3.2 Detection time ……………………………………………………………..83
4.3.3 Cramer-Rao Bound (CRB) analysis ………………………………………86
x CRLB for least squares ……………………………………………………86
x Comparison of simulated estimation performances of proposed
efficient least squares algorithm and CLS method with CRLB …………………87
4.4 Summary ................................................................................................................90
CHAPTER 5 .................................................................................................................. 92
PROPOSED EFFICIENT LEAST SQUARES ALGORITHM BASED POWER
SIGNAL PROCESSING SYSTEM............................................................................... 92
5.1 Overview ................................................................................................................92
5.2 Structure of the Power Signal Processing System .................................................95
5.3 Analysis of the Power Signal Processing System ..................................................97
5.3.1 Power signal processing in a single-phase system ……………………….97
x Instantaneous fundamental and harmonic components of voltage and
current signals ……………………………………………………………………99
x RMS values and phase angles of fundamental and harmonic
components of voltage and current signals ……………………………………..99
x Total harmonic distortion (THD) of voltage and current ………………..100
x
x Single-Phase power measurements ………………………………………101
5.3.2 Power signal processing in three-phase systems ………………………...103
x Voltage unbalance ……………………………………………………….108
x Stationary symmetrical component estimation ………………………….108
x Instantaneous symmetrical component estimation ………………………110
5.4 Modelling and Experimental Results for the Performance of the Power Signal
Processing System.........................................................................................................113
5.4.1 Performance of the proposed power signal processing system in
detecting fundamental and harmonic components …………………………….113
x Effect of sudden amplitude and phase changes …………………………116
x Effect of marginal fundamental frequency changes on performance ……119
x Effect of noise on the performance ………………………………………122
5.4.2 Performance of the power signal processing system in estimating
power and total RMS …………………………………………………………..123
5.4.3 Determining symmetrical components …………………………………127
x Stationary symmetrical components …………………………………….127
x Instantaneous symmetrical components …………………………………130
5.4.4 Experimental verification of the proposed efficient least squares
algorithm based harmonic detection ……………………………………………132
5.5 Summary ..............................................................................................................136
CHAPTER 6 ................................................................................................................ 138
PROPOSED SIGNAL PROCESSING SYSTEM FOR IDENTIFICATION OF
HARMONICS IN A THREE-PHASE ACTIVE POWER FILTER
APPLICATION ........................................................................................................... 138
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6.1 Overview ..............................................................................................................138
6.2 Three-Phase Active Power Filter System with the Proposed Efficient Least
Squares Algorithm ........................................................................................................142
6.2.1 System configuration ……………………………………………………142
6.2.2 Modelling of three-phase PWM converter……………………………….144
6.2.3 Design of dq- current controllers ………………………………………..147
6.2.4 Design of dc voltage controller ………………………………………….151
6.2.5 Current harmonic detection and controller phase lag correction based
on the proposed efficient least squares algorithm ……………………………..154
6.3 Modelling Results for Three-Phase Active Power Filter Based on the Proposed
Harmonic Detection Method.........................................................................................160
6.3.1 Modelling results for extracting individual current harmonic
components of a practical nonlinear load ………………………………………160
6.3.2 Modelling results for the selective harmonic compensation of active
power filter based on proposed harmonic extraction method ………………….163
6.4 Experimental Results of the Three-phase Active Power Filter Based on Proposed
Harmonic Detection Method.........................................................................................167
6.4.1 Experimental results for harmonics extraction using the proposed
method ………………………………………………………………………….169
6.4.2 Experimental results for the selective harmonic compensation of
active power filter using the proposed method …………………………………171
6.5 Summary ..............................................................................................................176
CHAPTER 7 ................................................................................................................ 178
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PROPOSED SIGNAL PROCESSING SYSTEM FOR VOLTAGE DIP
DETECTION IN A DISTRIBUTION STATIC SYNCHRONOUS
COMPENSATOR APPLICATION............................................................................. 178
7.1 Overview ..............................................................................................................178
7.2 Review of Grid Unbalance and Voltage Dip Detection Methods and the D-
STATCOM System.......................................................................................................179
7.2.1 Existing method of grid unbalance and voltage dip detection methods …179
x Grid unbalance detection methods ……………………………………..179
x Voltage dip detection methods …………………………………………..180
7.2.2 Brief review of the D-STATCOM system ………………………………182
7.3 Voltage Dip Detection Based on Proposed Efficient LS Method........................183
7.4 The D-STATCOM System with Proposed Voltage Dip Detection Method........189
7.4.1 System configuration ……………………………………………………189
7.4.2 Positive- and negative-synchronous reference frame controllers
(reactive power controller) ……………………………………………………..192
7.5 Modelling and Simulation Results of the D-STATCOM for Voltage Dip
Mitigation......................................................................................................................196
7.5.1 D-STATCOM system modelling ……………………………………….196
7.5.2 Simulation results for unbalanced grid voltage mitigation ……………..199
7.5.3 Simulation results for balanced voltage dip mitigation …………………202
7.5.4 Simulation results for unbalanced voltage dip mitigation ………………206
7.6 Experimental Results for the D-STATCOM with the Proposed Voltage Dip
Detection .......................................................................................................................209
7.6.1 Experimental results for unbalanced grid voltage mitigation ……………211
7.6.2 Experimental results for balanced voltage dip mitigation ……………….214
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7.6.3 Experimental results for unbalanced voltage dip mitigation …………….217
7.7 Summary ..............................................................................................................220
CHAPTER 8 ................................................................................................................ 223
CONCLUSIONS.......................................................................................................... 223
8.1 Conclusion of this Thesis .....................................................................................223
8.2 Suggestions for Future Work ...............................................................................230
8.2.1 Inter-harmonic detection …………………………………………………230
8.2.2 Voltage dip detection for dynamic voltage restorer (DVR) ……………..231
8.2.3 Application of more sophisticated control techniques for APF ………….231
REFERENCES............................................................................................................. 233
APPENDIX A .............................................................................................................. 241
A.1 Cramer-Rao Lower Bound (CRLB) for a Single Sinusoid ....................................241
A.2 Variance of Linear Least Squares Method.............................................................243
A.3 Cramer-Rao Lower Bound of the Linear Least Squares Technique ......................245
APPENDIX B .............................................................................................................. 250
B.1 Modelling Programs Using MATLAB ................................................................250
B.1.1 Proposed power signal processing system in detecting fundamental
and harmonic components………………………………………………………250
B.1.2 Proposed power signal processing system in estimating power and
total RMS values ……………………………………………………………….251
B.1.3 Proposed power signal processing system in determining symmetrical
components …………………………………………………………………….251
B.1.4 Proposed harmonic detection method in extracting current harmonic
components of a practical nonlinear load ………………………………………252
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B.1.5 Proposed harmonic detection method in selective harmonic
compensation of active power filter …………………………………………….252
B.1.6 Proposed voltage unbalance and dip detection method in D-
STATCOM system …………………………………………………………….253
B.2 Real-Time Programs used for Experiments .........................................................253
B.2.1 Proposed power signal processing system in estimating instantaneous
fundamental and harmonic component …………………………………………253
compensation of active power filter ……………………………………………254
B.2.3 D-STATCOM with the proposed voltage dip detection …………………255
APPENDIX C .............................................................................................................. 257
EXPERIMENTAL SETUPS........................................................................................ 257
C.1 Overview of Experimental Setups .......................................................................257
x Experimental verification of the proposed efficient least squares
algorithm ………………………………………………………………………257
x Active power filter (APF) system ………………………………………..258
x D-STATCOM system ……………………………………………………259
C.2 Description of dSPACE DS1104 Controller Board .............................................260
C.3 Description of Three-phase 35kW PWM Converter............................................263
C.4 External Voltage and Current Transducers ..........................................................271
APPENDIX D .............................................................................................................. 276
D.1 Papers Relevant to the Thesis Topic ....................................................................276
D.1.1 Refereed conference publications ………………………………………..276
D.1.2 Paper not relevant to the thesis topic ……………………………………278
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LIST OF FIGURES
Figure 1.1 Harmonic distortion example…………………………………………… 3
Figure 2.1 The first stage in the decimation-in-time algorithm................................19
Figure 2.2 Four-level DWPT decomposition tree for harmonic extraction scheme.26
Figure 2.3 Kalman filter harmonic extraction scheme. ............................................36
Figure 3.1 The postulated signal given in (3.1) with SNR of 17dB. ........................40
Figure 3.2 Steady-state amplitude error of DFT/FFT method for each harmonic
component: zero noise (dash) and with noise of SNR = 17dB (solid)...........................42
Figure 3.3 Phase angle error of DFT/FFT method for each harmonic component:
zero noise (dash) and with noise of SNR = 17dB (solid)................................................42
Figure 3.4 Detection time of DFT. ...........................................................................43
Figure 3.5 Fundamental amplitude and phase angle error of the DFT method for
sudden change in input signal at t=0.1 sec......................................................................44
Figure 3.6 Actual (solid) and estimated components using DFT method (dash) of a
signal (Top), and waveform error (bottom) for 1Hz increase in fundamental frequency
at t = 0.06 sec : (a) Fundamental component, (b) Fifth-order harmonic component and
(c) Seventh-order harmonic component..........................................................................45
Figure 3.7 First six decomposed components of postulated input waveform using
Daubechies 10 (db10) 4-level filter bank........................................................................48
Figure 3.8 First six decomposed components of postulated input waveform using
Symlets 10 (sym10) 4-level filter bank. ..........................................................................49
Figure 3.9 Steady-state amplitude error of CLS method for each harmonic
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Figure 3.10 Phase angle error of CLS method for each harmonic component: zero
noise (dash) and with noise of SNR = 17dB (solid). ......................................................51
Figure 3.11 Detection time of CLS method. ..............................................................52
Figure 3.12 Fundamental amplitude and phase angle error of the CLS method for
sudden change in input signal at t=0.1 sec......................................................................53
Figure 3.13 Actual (solid) and estimated components using CLS method (dash) of a
signal (top), and waveform error (bottom) for 1Hz increase in fundamental frequency at
t = 0.06 sec : (a) Fundamental component, (b) Fifth-order harmonic component and (c)
Seventh-order harmonic component. ..............................................................................54
Figure 3.14 Steady-state amplitude error of KF method for each harmonic
Figure 3.15 Steady-state phase angle error of KF method for each harmonic
Figure 3.16 Detection time of KF method..................................................................57
Figure 3.17 Actual (solid) and estimated components using KF method (dash) of a
signal (top), and waveform error (bottom) for 1Hz increase in fundamental frequency at
t = 0.06 sec: (a) Fundamental component, (b) Fifth-order harmonic component and (c)
Seventh-order harmonic component. ..............................................................................59
Figure 3.18 Transient response of the KF method: The input waveform with step
change at t=0.1sec (top), amplitude of fundamental (middle), fundamental amplitude
error (bottom). .................................................................................................................60
Figure 3.19 Transient response of the KF method to short duration disturbance: the
waveform of short duration disturbance (top) and estimated amplitude using KF
(bottom). .................................................................................................................61
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Figure 3.20 Comparison of estimated fundamental amplitude error for various
methods verses signal-to-noise ratio (SNR)....................................................................63
Figure 3.21 Comparison of estimated fundamental phase angle error for various
methods verses signal-to-noise ratio (SNR)....................................................................63
Figure 3.22 Comparison of the waveform errors against fundamental frequency
variation of the studied methods. ....................................................................................64
Figure 4.1 Detection time: actual (solid), proposed efficient least squares algorithm
(dash) and FFT method (dot). .........................................................................................84
Figure 4.2 Amplitude of the fundamental component from the proposed efficient
least squares method: at sampling frequency 3 kHz (dot), at sampling frequency 5 kHz
(dash), sampling frequency at 7 kHz (dash-dot) and actual value (solid).......................85
Figure 4.3 Phase angle of the fundamental component from the proposed efficient
least squares method: at sampling frequency 3 kHz (dot), at sampling frequency 5 kHz
(dash), at sampling frequency 7 kHz (dash-dot) and actual value (solid).......................85
Figure 4.4 Logarithmic variance of amplitude estimation vs. SNR: CRLB, proposed
efficient least squares method and conventional least squares method. .........................89
Figure 4.5 Logarithmic variance of phase angle estimation vs. SNR: CRLB,
proposed efficient least squares method and conventional least squares method...........89
Figure 5.1 General overview of the power signal processing system. .....................94
Figure 5.2 Structure of the proposed power signal processing system. ...................96
Figure 5.3 Estimation of instantaneous cosine and sine terms of fundamental and
harmonic components of voltage and current of a single-phase system using proposed
efficient least square algorithm. ......................................................................................98
xviii
Figure 5.4 Estimation of instantaneous cosine and sine of the fundamental and
harmonic components of a three-phase system using the proposed efficient least squares
algorithm. ...............................................................................................................105
Figure 5.5 Estimation of instantaneous cosine and sine of the fundamental and
harmonic components of a zero-sequence free three-phase system using the proposed
efficient least square algorithm. ....................................................................................107
Figure 5.6 The postulated voltage waveform. ........................................................114
Figure 5.7 Actual (solid) and extracted waveforms with the proposed method (dot):
instantaneous fundamental, 5th and 11th order harmonic components of the input voltage
waveform. ...............................................................................................................115
Figure 5.8 RMS values of the fundamental and harmonic components obtained
using the proposed method............................................................................................115
Figure 5.9 Phase angles of the fundamental and harmonic components obtained
using the proposed method............................................................................................116
Figure 5.10 Performance of the proposed method for step amplitude change at t = 0.1
sec. ...............................................................................................................117
Figure 5.11 Actual, and reconstructed voltage waveform using the proposed method
for step amplitude change at t = 0.1 sec. .......................................................................118
Figure 5.12 Performance of the proposed method for step phase change at t = 0.1
sec. ...............................................................................................................118
Figure 5.13 Actual, and reconstructed voltage waveform using proposed method for
step phase change at t = 0.1 sec. ...................................................................................119
Figure 5.14 Tracking performance of the proposed method with step frequency
change from 50 Hz to 51 Hz at t = 0.1 sec....................................................................120
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Figure 5.15 Comparison of the RMS errors of the proposed and the DFT methods
against the fundamental frequency change. ..................................................................121
Figure 5.16 Tracking performance of the reconstructed waveform using the proposed
method and the DFT method with frequency step changes from 50Hz to 51 Hz at t = 0.1
sec. ...............................................................................................................121
Figure 5.17 Steady state RMS error of reconstructed waveforms using proposed and
DFT methods at 51Hz fundamental frequency. ............................................................122
Figure 5.18 RMS error of reconstructed waveform vs. SNR for proposed and DFT
methods. ...............................................................................................................123
Figure 5.19 Postulated voltage and current waveforms given in and (5.38) with
100% step change at t = 0.1 sec. ...................................................................................124
Figure 5.20 Comparison of estimated total RMS values of voltage and current
waveforms using proposed method and DFT method, and actual values for 100% step
change at t = 0.1 sec. .....................................................................................................125
Figure 5.21 Comparison of steady state total RMS voltage error for proposed method
and DFT method. ..........................................................................................................126
Figure 5.22 Comparison of estimated total active power, total reactive power and
power factor using proposed method and DFT method, and actual values for 100% step
change at t = 0.1 sec. .....................................................................................................126
Figure 5.23 Postulated three-phase voltage waveforms. ..........................................128
Figure 5.24 RMS magnitudes of positive-, negative-, and zero-sequence determined
using proposed method. ................................................................................................129
Figure 5.25 Phase angles of positive-, negative-, and zero-sequence determined using
proposed method. ..........................................................................................................129
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Figure 5.26 Postulated three-phase voltage waveforms for instantaneous symmetrical
components estimation..................................................................................................130
Figure 5.27 Instantaneous positive-, negative-, and zero-sequence components
determined using proposed method. .............................................................................131
Figure 5.28 Comparison of amplitudes of positive-, negative- and zero-sequence
components determined using proposed and DFT methods. ........................................132
Figure 5.29 Overview of the experimental setup. ....................................................133
Figure 5.30 Captured three-phase load voltage waveforms. ....................................134
Figure 5.31 Experimental results: extracted instantaneous fundamental component
and its RMS value for phase- a , phase- b and phase- c ................................................135
Figure 5.32 Comparison of actual waveform and the reconstructed waveform with
proposed method: phase- a , phase- b and phase- c . .....................................................135
Figure 5.33 Percentage error in reconstructed voltage waveform for phase- a . ......136
Figure 6.1 Typical location for a shunt active filter based on a voltage source
converter. ...............................................................................................................140
Figure 6.2 Schematic diagram of the selective harmonic APF with proposed
harmonic detection method. ..........................................................................................142
Figure 6.3 Schematic diagram of three-phase voltage source converter................144
Figure 6.4 dq axis closed-loop current control diagrams. ...................................149
Figure 6.5 Bode diagram of the closed-loop dq current TF. ..............................151
Figure 6.6 Control block diagram of the voltage control loop. ..............................152
Figure 6.7 Bode diagram of the closed dc-link voltage control loop. ....................153
Figure 6.8 Proposed efficient least squares algorithm for identification of current
harmonics for APF. .......................................................................................................154
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Figure 6.9 Phase-lags of the current controller for various harmonics components. ...
...............................................................................................................157
Figure 6.10 Magnitude attenuation of the current controller for various harmonics
components. ...........................................................................................................159
Figure 6.11 MATLAB/SIMULINK model of three-phase thyristor rectifier load and
the harmonic current extraction using the proposed method. .......................................160
Figure 6.12 The load current waveform of phase-a. ................................................161
Figure 6.13 Extracted fundamental and individual harmonic components (i.e. 5th, 7th,
11th and 13th order harmonics) of phase – a current using the proposed method. ........162
Figure 6.14 Actual current waveform and the reconstructed waveform with the
proposed method of phase-a, -b and -c. ........................................................................162
Figure 6.15 MATLAB/SIMULINK model for the selective harmonic compensation
of active power filter based on the proposed harmonic detection method....................163
Figure 6.16 Simulation results of the proposed APF without current controller phase
lag compensation; (a) load current; (b) filter current; (c) source current; and (d) dc-link
voltage. ...............................................................................................................165
Figure 6.17 Simulation results of the proposed APF with current controller phase lag
compensation; (a) load current; (b) filter current; (c) source current; and (d) dc-link
voltage. ...............................................................................................................165
Figure 6.18 Performance of APF based on proposed method during step decrease in
load current together with fundamental frequency change from 50 to 52 Hz: (top) load
current; (middle) filter current; and (bottom) source current........................................166
Figure 6.19 Performance of APF based on DFT method during step decrease in load
current together with fundamental frequency change from 50 to 52 Hz: (top) load
current; (middle) filter current; and (bottom) source current........................................167
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Figure 6.20 Overview of experimental setup for APF system. ................................168
Figure 6.21 Photograph of experimental setup for APF system. .............................169
Figure 6.22 Extracted fundamental, 5th, 7th, 11th and 13th –order harmonic
components in load current for phase –a. .....................................................................170
Figure 6.23 Actual waveforms and the reconstructed waveforms with proposed
method for load current for phase - a , - b and - c .........................................................171
Figure 6.24 Response of APF with proposed selective harmonic compensation when
filtering enabled at t=0.035sec.: (Ch1) load current waveform (6.5 A/div), (Ch2) active
filter current (2 A/div), (Ch3) source current waveform (6.5 A/div) and (Ch4) dc-link
voltage (140 V/div). ......................................................................................................172
Figure 6.25 Oscillogram of (Ch1) load current (5 A/div) and (Ch2) source current (5
A/div) for APF with proposed harmonic detection method for D 0D of bridge rectifier
load. ...............................................................................................................173
Figure 6.26 Oscillogram of (Ch1) load current (5 A/div) and (Ch2) source current (5
A/div) for APF with proposed harmonic detection method for about 60 degrees firing
angle (i.e. D | 60D ) of bridge rectifier load. ..................................................................173
Figure 6.27 Harmonic constituent as a percentage of the fundamental component of
load current and source current with zero firing angle (i.e. D 0D )..............................174
Figure 6.28 Harmonic constituent as a percentage of the fundamental component of
load current and source current with about 60 degrees firing angle (i.e. D | 60D )........175
Figure 6.29 Dynamic response of APF with proposed selective harmonic
compensation: (Ch1) load current waveform (6.5 A/div), (Ch2) active filter current (2
A/div), (Ch3) filtered source current waveform (6.5 A/div) and (Ch4) dc-link voltage
(140 V/div). ...............................................................................................................176
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Figure 7.1 Estimation of instantaneous cosine and sine components of fundamental
voltage component using the proposed efficient least squares algorithm.....................184
Figure 7.2 Block diagram of voltage dip detection method based on proposed
efficient least squares algorithm....................................................................................189
Figure 7.3 Overall schematic diagram of D-STATCOM with proposed voltage dip
detection method. ..........................................................................................................190
Figure 7.4 Simulated three-phase unbalanced grid voltages. .................................193
Figure 7.5 Simulated direct dq components for three-phase voltages given in
Figure 7.4. ...............................................................................................................193
Figure 7.6 Simulated positive- and negative- sequence dq components for three-
phase voltages given in Figure 7.4. Top: positive-sequence in positive SRF; bottom:
negative-sequence in negative SRF. .............................................................................194
Figure 7.7 Detailed block diagram of reactive power controllers for D-STATCOM
given in Figure 7.3. .......................................................................................................195
Figure 7.8 MATLAB/SIMULINK model for D-STATCOM system with proposed
dip detection. ...............................................................................................................197
Figure 7.9 MATLAB/SIMULINK model of D-STATCOM controller. ................197
Figure 7.10 Simulated three-phase grid voltages. ....................................................200
Figure 7.11 Simulated grid voltages in the SRF dq components, before
compensation. Top: positive- sequence in positive SRF; bottom: negative-sequence in
negative SRF. ...............................................................................................................200
Figure 7.12 Compensated voltage waveforms and phase voltage waveform error in
the case of unbalance grid voltage. ...............................................................................201
xxiv
Figure 7.13 Compensated grid voltages in the SRF dq components in the case of
unbalanced grid voltage. Top: positive- sequence in positive SRF; bottom: negative-
sequence in negative SRF. ............................................................................................201
Figure 7.14 Simulated current injections and dc-link voltage during unbalanced
voltage compensation....................................................................................................202
Figure 7.15 Simulated three-phase grid voltage with 25% balanced voltage dip. ...203
Figure 7.16 Simulated grid voltages in SRF dq components in case of 25%
balanced voltage dip. Top: positive- sequence in positive SRF; bottom: negative-
sequence in negative SRF. ............................................................................................204
Figure 7.17 Compensated voltage waveforms at PCC and voltage waveform error in
case of 25% balance voltage dip. ..................................................................................204
Figure 7.18 Compensated grid voltages in SRF dq components in case of balanced
voltage dip. Top: positive-sequence in positive SRF; bottom: negative-sequence in
negative SRF. ...............................................................................................................205
Figure 7.19 Simulated current injections and dc-link voltage during balanced voltage
dip compensation. .........................................................................................................205
Figure 7.20 Simulated three-phase grid voltages during unbalanced voltage dip....207
Figure 7.21 Simulated grid voltages in SRF dq components in case of 25% voltage
dip with 20% amplitude unbalance. Top: positive- sequence in positive SRF; bottom:
negative-sequence in negative SRF. .............................................................................207
Figure 7.22 Compensated voltage waveforms at PCC and voltage waveform error in
the case of unbalance voltage dip..................................................................................208
xxv
Figure 7.23 Compensated grid voltages in SRF dq components in the case of 25%
voltage dip with 20% amplitude unbalance. Top: positive- sequence in positive SRF and
bottom: negative-sequence in negative SRF. ................................................................208
voltage dip compensation..............................................................................................209
Figure 7.25 Overview of the experimental setup of the D-STATCOM system.......210
Figure 7.26 A photograph of the experimental setup of the D-STATCOM.............211
Figure 7.27 Unbalanced phase voltages of phases-a, -b and -c (100 V/div)............212
Figure 7.28 Compensated voltage of phases-a, -b and -c using D-STATCOM with
proposed method (100 V/div). ......................................................................................212
Figure 7.29 Dynamic response when unbalance mitigation started at t = 0.06 sec.:
voltage of phases- a, -b and -c (100 V/div)...................................................................213
Figure 7.30 Dynamic response when unbalance mitigation started at t = 0.06 sec.:
(Ch1) d component of positive-sequence voltage in positive SRF (34.67 V/div); (Ch3)
d components of negative-sequence voltage in negative SRF (8.67 V/div); (Ch4)
q components of negative-sequence voltage in negative SRF (8.67 V/div)..............214
Figure 7.31 Balanced voltage dip of 17% occurred of at t = 0.06 sec. (100 V/div).......
...............................................................................................................215
Figure 7.32 Compensated voltage waveforms of phases-a, -b, and -c at PCC in case
of 17% balanced voltage dip at t = 0.06 sec. (100 V/div).............................................216
Figure 7.33 Dynamic response of balanced voltage dip compensation when dip
started at t = 0.06 sec.: (Ch1) d component of positive-sequence voltage in positive
SRF (34.67 V/div); (Ch3) d components of negative-sequence voltage in negative
SRF (8.67 V/div); (Ch4) q components of negative-sequence voltage in negative SRF
(8.67 V/div). ...............................................................................................................216

xxvi
Figure 7.34 Compensated steady-state voltage waveforms of phases-a, -b and -c at
PCC during 17% balanced voltage dip (100 V/div)......................................................217
Figure 7.35 Voltages of phases-a, -b and -c for unbalanced dip occurring at t = 0.06
sec. (100 V/div).............................................................................................................218
Figure 7.36 Detected dq symmetrical components for unbalanced voltage dip
(8.67 V/div). ...............................................................................................................219
Figure 7.37 Compensated voltage waveforms of phases-a, -b, and -c at PCC in case
of unbalanced voltage dip at t = 0.06 sec. (100 V/div). ................................................219
Figure 7.38 Dynamic response of unbalanced voltage dip compensation when dip
(8.67 V/div). ...............................................................................................................220
Figure C.1 Experimental verification of the proposed efficient least squares
algorithm. ...............................................................................................................258
Figure C.2 Experimental setup for active power filter system................................259
Figure C.3 Experimental setup for D-STATCOM system......................................260
Figure C.4 A photograph of CP1104 connector panel. ...........................................262
Figure C.5 Pin mapping of interconnection cable between CP18 Sub-D of connector
panel CP1104 and the IGBT converter. ........................................................................263
Figure C.6 IGBT converter power circuit. ..............................................................266
xxvii
Figure C.7 Schematic diagram of the interface PCB inside the IGBT converter. ..267
Figure C.8 Schematic diagram of the current transducer PCB (residing inside the
converter). ...............................................................................................................268
Figure C.9 Schematic diagram of the voltage transducer PCB (residing inside the
converter). ...............................................................................................................269
Figure C.10 Interconnection cable inside the converter (for PWM signals).............270
Figure C.11 A photograph of the IGBT converter. ...................................................270
Figure C.12 Schematic diagram of the current transducer PCB. ..............................273
Figure C.13 Schematic diagram of the voltage transducer PCB...............................274
Figure C.14 A photograph of external voltage and current transducers. ..................275
xxviii
LIST OF TABLES
Table 2.1. Comparison of computational complexity of the DFT versus FFT
algorithm. .................................................................................................................20
Table 3.1 Decomposed components, frequency sub-bands and fundamental and
harmonic components .....................................................................................................47
Table 3.2 Estimated amplitude and phase angle values using db10 4-level filter
bank. .....................................................................................................................48
Table 3.3 Estimated amplitude and phase angle values using Sym10 4-level filter
bank. .....................................................................................................................50
Table 4.1 Comparison of the real-time computational cost (number of calculation
operations) .................................................................................................................83
Table 6.1 System parameters for modeled APF....................................................164
Table 7.1 Design specifications and circuit parameters of the proposed D-
STATCOM....................................................................................................................198
xxix
LIST OF SYMBOLS AND ABBREVIATIONS
CF Filter capacitance
Cdc dc-side capacitor of VSC
erms Root mean squares error
esa esb esc Source phase voltages of phases -a,-b and -c
esd d axis component of source voltage in SRF
f sw Converter Switching frequency
fs Sampling frequency
GTI Current transfer function
GCI Transfer function of PI current controller
Gcloop Closed-loop transfer function of dq current feedback loops
Gdc _ cloop Closed-loop transfer function of outer voltage loop
I Identity matrix
I I Fisher information for vector I
i Instantaneous current
idc The dc side current
iL Load current
iF Converter current
iS Source current
iH Harmonic current
iCHai ѽ iCHbi Phase lag corrected instantaneous harmonic signal
xxx
iFd , iFq dq axes components of VCS current in SRF
Ii Peak amplitudes of the i th component of current
I iRMS RMS of the i th component of current
K Total number of harmonic components
K Pi Proportional gain of current controller
K Ii Integral gain of current controller
K Pv Proportional gain of dc voltage controller
K Iv Integral gain of dc voltage controller
L Number of the captured samples
LS Source inductance
LF Inductance of the ac-side of VCS
Lline Line inductance
Ltr Transformer leakage inductance
m Time-step in the discrete sample sequence
ma , mb and mc Modulation signals of phases -a,-b and -c
md , mq dq axes components of modulation signals.
Magi Magnitude correction factor
n Discrete present time step
Piactive Active power
active
Ptotal Total active power
P reactive Reactive power
PF Power factor
RS Source resistance
xxxi
Rline Line resistance
Rdc dc-side resistance of VCS
RF ac-side resistance of VCS
S Apparent power
t Continuous time
T Sampling time
THDV THD of the voltage
THDI THD of the current
v Instantaneous voltage
Vi Peak amplitudes of the i th component of voltage
Vi RMS RMS of the i th component of voltage
Vdc dc-link voltage
X Amplitude
Z Angular frequency of the supply
Zni Natural frequency of closed-loop current transfer function
Znv Natural frequency of closed-loop dc voltage transfer function
T Phase angle
V2 Noise variance
D Firing angle
Superscripts
+ Positive-sequence component
- Negative-sequence component
0 Zero-sequence component
* Reference or command value
xxxii
T Transpose
Complex number
sin Sine component of a signal
cos Cos component of a signal
Subscripts
i i th harmonic component
d d axis component
q q axis component
img Imaginary component
real Real component
Abbreviation
ADC Analogue to digital converter
APF Active power filter
BLUE Best linear unbiased estimator
CWT Continuous wavelet transform
CLS Conventional least squares
CRB Cramer-Rao Bound
CRLB Cramer-Rao Lower Bound
DFT Discrete Fourier transform
DSP Digital Signal Processor
dB Decibel
D-STATCOM Distribution static synchronous compensator
DWT Discrete wavelet transform
FFT Fast Fourier transform
HP High pass filter
ITAE Integral of time multiplied by absolute magnitude of the error
xxxiii
KVL Kirchoff’s voltage law
KCL Kirchoff’s current law
KF Kalman filter
LP Low pass filter
LS Least squares
PI Proportional plus integral
PLL Phase-locked-loop
PCC Point of common coupling
PCB Printed circuit board
PWM Pulse width modulated
RMS Root-mean-square
RL Resistance and Inductance
STFT Short-time Fourier transform
SVD Single value decomposition
SPWM Sinusoidal pulse width modulation
SRF Synchronous reference frame
SNR Signal-to-noise-ratio
THD Total harmonic distortion
TF Transfer function
VSC Voltage source converter
WT Wavelet transform
WGN White Gaussian noise
xxxiv
Chapter 1: Introduction
CHAPTER 1
INTRODUCTION
1.1 Overview
Fast and accurate measurement of fundamental and harmonic quantities is extremely
important at all levels of the electrical power system, and is of value for both power
distributors and power consumers. The design of most conventional equipment such as
lights, constant speed motors and power systems are based on the assumption that the
voltage sources are sinusoidal and the loads are linear, so that the resulting load currents
are also sinusoidal. As demands on accuracy have increased and non-linear loads have
become more common, this approximation is often no longer valid. Much effort is made
to investigate the influence of non-linear loads on measurements and new
instrumentations are developed to cope with non-sinusoidal conditions on the power
systems. One part of this effort, which is described in this thesis, is the development and
verification of a novel signal processing algorithm, for measuring time-varying
individual harmonic quantities of single-phase and three-phase waveforms. This
algorithm is non-recursive, and is based on a linear least squares algorithm. The
proposed method can identify each harmonic component with a fast response and high
accuracy in real-time. The important features of the proposed approach are its
sensitivity to marginal variations in the centre frequency and high immunity to noise
present in the signals which are normally encountered in practice. The proposed method
is suitable for environments where small frequency excursions are experienced and the
conventional discrete Fourier transform -based methods do not provide satisfactory
results. In this study, it is also confirmed that the proposed method is able to extract the
1
harmonics quantities in real-time, and its effectiveness is substantiated with a DSP-
based experimental three-phase shunt active power filter (APF) and distribution static
synchronous compensator (D-STATCOM) application.
1.2 Classification of Power Quality Problems
Power quality problems are classified in two main groups as steady-state and transients.
Power system harmonics and voltage unbalance can be given as examples of power
quality problems in the steady state. Voltage dips, spikes and surges are examples which
tend to occur over short time intervals, and classified as transients. Of all these,
harmonic distortion and voltage dips are the most important power quality problems
facing industrial and large commercial customers [1, 2]. A brief description of the
characteristics and the causes of typical power quality problems are given next.
Harmonic distortion: Harmonics can be understood as different frequency periodic
components that are superimposed on the main frequency waveform. Harmonics usually
have frequencies that are integer multiples of the fundamental frequency. In power
systems, existing harmonics are mostly odd integer multiple of the power frequency.
The 3rd, 5th, 9th, 7th, 11th and 13th orders can be identified as most the common
harmonics. In addition to these common harmonics, it is possible to face signal
components that are not integer multiples of the fundamental. Such components are
called as “inter-harmonics” and they are usually encountered while dealing with non-
periodic signals.
In recent years a rapid growth in harmonic voltages and currents injected into power
systems has been observed due to the increased utilisation of non-linear loads. In
addition, devices employing high-frequency switching, such as switch mode power
2
supplied used in televisions, computers and compact fluorescent lighting have added an
additional level of harmonics to the supply system. Propagation of current harmonics
through the distribution system usually leads to unwanted waveform distortions in the
system voltages. As an example, the illustration of up to ninth harmonic distortion can
be seen in Figure 1.1.
Harmonic voltages and currents can cause many serious operational problems to both
energy suppliers and users. Overheating of motors and transformers, malfunction of
control devices, protective relays, and iron losses in transformers can be listed as typical
detrimental effects of harmonics voltages and currents along the power network [3].
Figure 1.1 Harmonic distortion example.
Voltage unbalance: Voltage unbalances occur in three-phase systems as a result of
unbalanced load or source operations. Improper grounding and untransposed over head
transmission lines also lead to unbalance between phases in a three-phase system.
Analyses of three-phase systems are usually carried out by symmetrical sequence
3
components and the proportion of the negative- and positive-sequence components
determines the degrees of unbalance.
A possible effect of voltage unbalance is significant differences in three-phase currents.
Particularly, electric machines connected to unbalance power systems draw currents
with a degree of unbalance that is several times higher than that of the supply voltage.
This current unbalance results in loss of efficiency and temperature rise in the machine.
Unbalanced supplies also affect the operation of ac-dc converters. Ripple on the dc-side,
and non-characteristic harmonics on the ac-side are possible consequences of voltage
unbalance [4].
Voltage dips: According to IEEE Std. 1159-1995 [5], a voltage dip is defined as a
reduction in the voltage magnitude at the power frequency of short duration between 1
cycle and 6 cycles. Voltage dips are considered one of the most serious power quality
problems. Disruptive voltage dips are mainly caused by short-circuit faults. Voltage
dips causes problem on various types of utilisation equipment [1], [6]. Especially
computers, adjustable-speed drives and process-control equipment are well-known for
their sensitivity to supply voltage dips. Equipment used in modern industrial plants are
actually becoming more sensitive to voltage dips as the complexity of the equipment
increases and the equipments are interconnected in sophisticated processes.
Power interruptions are also common in power systems. Interruptions mostly occur
when a protective device operates and isolates the circuit serving a user. This normally
occurs only when there is a fault on a circuit that supplies a particular user. Short
interruptions are also known to be troublesome and costly type of power quality
problem for most customers. However, occurrence of voltage dips are much more
frequent compared to interrupts since voltage dip can occur due to the faults in a wide
4
part of power system whereas, interrupt usually occur only when there is a fault on a
particular circuit. If equipment is sensitive to these dips, the frequency of problems
experienced will be much higher than if the equipment would be only sensitive to
interruptions [7]. Over the last ten years, voltage dips have become one of the main
topics concerning power quality among utilities, customers and equipment
manufacturers. Several international standards and working group documents have been
produced to improve the understanding of voltage dip problems [8], [9] ,[10],[11],[12].
1.3 Importance of the Identification and Tracking of Disturbance

Signal Quantities
The analysis and mitigation of power quality problems are becoming an integral part of
power system studies. Reference [13], suggested a cost estimate of $25 billions per year
is spent in the U.S. for correcting power quality problems. Power quality problems can
impact customer operations, causing malfunctions and cost on lost production. To avoid
these malfunctions and unnecessary costs, many facilities have been required to
implement a mitigation system. Essentially, mitigation requires two components: the
method of extraction of the disturbance signal and the mitigating control strategy to
generate the control signal for the mitigating device to cancel the disturbance, or reduce
its level. A mitigation controller system should have characterized fast response to cope
with the dynamic nature of the power system disturbances. A fast and accurate
measurement technique is an essential to estimate the accurate compensating signals for
the conditioning task. The features of a good disturbance identification method for
mitigation purposes can be listed as below:
(i) An ability to extract, to detect and to track the distorted signal and disturbance
signals accurately and fast.
5
(ii) A performance rate that facilitates the tracking of phenomena such as harmonics
and symmetrical components without requiring a large Digital Signal Processing (DSP)
programming code and execution time.
(iii) Stability for the operation range of the disturbance.
(iv) Robustness against marginal variation in centre frequency.
1.4 Existing Methods of Identification of Disturbance Quantities
This section briefly explains the main existing methods of identification of amplitude
and phase-angle of the fundamental and its harmonic components used in power
systems applications. The principles of identifying harmonics using the root-mean-
square (RMS), discrete Fourier transform (DFT) based algorithms including fast Fourier
transform (FFT), conventional least squares (CLS), wavelet transforms (WT) and
Kalman Filter (KF) methods are explained and their qualities and shortcomings are
mentioned. These methods are studied in more detail in the next chapter to establish the
desired features for a new improved method.
A simple approximation of the amplitude of the sinusoidal signal can be obtained by
using the root mean square (RMS), which is evaluated over a cycle window. A great
advantage of this method is its simplicity, speed of calculation and less requirement for
memory, because the RMS value can be stored periodically instead of after every
sample [14]. However, this method’s dependency on the measurement window length is
considered as a disadvantage. A higher window length results in better accuracy at the
cost of a lower time-resolution. Moreover, the RMS value does not distinguish between
fundamental frequency, harmonics or noise components.
6
The most common category of harmonic estimation methods are based on the discrete
Fourier transform (DFT) and fast Fourier transform (FFT) [15], [16], [17], [18]. The
DFT is a discrete complex valued series for calculating the Fourier coefficients. The
DFT method can be used to estimate amplitudes and phase angles of the fundamental
and harmonic components. The FFT algorithm is computationally an efficient
implementation of the DFT and it reduces the computational cost required for
evaluation of the DFT by several orders of magnitude. However, the DFT/FFT methods
still need data points sampled over one fundamental cycle of signal to accurately
calculate the harmonic components. As a consequence, these methods are known to be
unsuitable for sinusoidal analysis of highly time-varying signals. Also, any corruption
of the DFT/FFT would yield inaccurate results due to leakages (occur if the number of
samples is not an integer) and picket-fence (occur if the analysed waveform includes
inter-harmonics). In some cases, results of the estimation can be improved with
windowing (e.g. Hanning, Hamming, Kaiser Windows) or filtering (e.g. low-pass or
high-pass filters) [19], [20]. Window length dependency resolution is another
disadvantage of the DFT/FFT methods. For example, the shorter the data window the
worse is the frequency resolution. Another shortcoming of the DFT/FFT methods is
their high sensitivity to noise [19].
One alternative to the DFT is the short-time Fourier transform (STFT), which is a
sliding window version of the DFT and performs the DFT on each windowed signal
[21]. This provides some time-information in addition to the frequency-information.
However, because of its constant time window (i.e., constant time resolution), the
frequency resolution must also be constant. The STFT uses a window of constant width
that is not adjusted to individual frequency components. One can intuitively see that
7
high frequency components of a signal may benefit from windows of short duration,
whereas low frequency components of a signal generally benefit from windows of
longer duration.
Since 1994, the use of Wavelet transform (WT) theory has been introduced to identify
particular harmonics or harmonic sub-bands of interest [22], [23]. The use of the WT
has emerged as an alternative to both the DFT and the STFT for the analysis of non-
stationary phenomena because WT is used to decompose the signal in different
frequency bands and characterise them separately. As described in [24], [25], wavelets
perform better with non-periodic signals that contain short duration impulse components
as is typical in power system transients. Furthermore, wavelet based techniques have
been proposed for detection and measuring of power system disturbances [26].
However, the estimation efficiency and accuracy of the wavelet transform depends on
the choice of the mother wavelet and the wavelet type should be chosen accordingly to
the specific event being studied. This algorithm also needs expensive computation
which restricts their real-time application in power systems. As a consequence, WT are
more commonly used in off-line inter-harmonic detection applications [23],[25],[26].
Conventional least squares parameter estimation techniques have a long and successful
history since Gauss’ first formulation. The basic idea is to choose the parameter
estimate that minimizes the sum of squared-error criterion that is related to a set of over
determined linear equations, the coefficients of which are unknown parameters. The
advantage of such an algorithm has been enhanced by the adoption of matrix notation
and the use of digital computers for performing the computations. The least squares
computations yield updated estimates of all model variables as well as their respective
cofactor or covariance matrices. As a consequence, this method is suitable for
8
identifying disturbances in time-varying harmonic signals. Methods to measure
harmonics in power systems using the least square method have been proposed in [27]
and [28]. However, the early idea of measuring with a conventional least square method
was discarded due to its immense mathematical complexity.
A digital recursive measurement scheme for on-line tracking of power system
harmonics proposed by Girgis et al [29] uses a Kalman filter (KF) to estimate the
harmonic components. The main feature of the KF is the recursive processing of the
noise measurement data. This technique is defined as a state-model, and can track
amplitudes and phase-angles of the fundamental and harmonics in real-time under noisy
environments [29]. However, the KF is generally sensitive to the initial condition of the
state variable covariance matrix and the noise variance which have been arbitrarily fixed
in [29]. There are no systematic methods of determining these two matrixes and a trial
and error approach is used to apply them in practice. Moreover, internal calculations of
the KF procedure also need relatively long time [30], [31],[32].
1.5 Scope of this Thesis
The aim of this research is first to propose a novel identification disturbances signal
quantities technique and then to verify the performance of the proposed identification
technique by applying it to a suitable practical mitigation application. The scope of this
research is outline as follow:
1. To propose a novel power system disturbance identification technique with the
following features.
(i) Fast and accurate detection of time-varying fundamental and harmonic
components in power system quantities.
9
(ii) Maintain better accuracy in detecting amplitude and phase angle value compared
to conventional methods such as DFT or FFT, when system frequency is slightly
changed.
(iii) Capability of coping with the unbalances in three-phase systems.
(iv) Low computational complexity so that it is more suitable for real-time
applications.
(v) Structural simplicity for efficient implementation with digital signal processors.
2. To design a power system signal processing system that estimates the constituent
components, symmetrical components, total harmonic distortions (THD) and power
factor based on the proposed identification method.
3. To evaluate performance of the proposed power signal processing system by applying
it to a three-phase active power filter application. In this application, the proposed
power signal processing system is used to identify the harmonics in the current in order
to generate compensation currents.
4. Finally, to apply the proposed power signal processing technique to identify grid
unbalances and voltage dips, and employ the identified information to a D-STATCOM
application to mitigate them.
1.6 Proposed Approach for Identification of Power System

Disturbances
The advantages of the conventional least square algorithm for disturbance identification
were mentioned in Section 1.4. This method is very suitable to extract disturbances in
time-varying harmonic signals. However, complex number arithmetic and inversion
10
matrix transformation require a significantly greater number of computational steps.
Therefore, the concept of the proposed method is to solve the system equations in an
alternative way without inverting any matrices and with less number of real-time matrix
multiplications. From the results, the proposed algorithm is shown to be
computationally efficient, since it only performs one matrix multiplication per time
sample. The size of the matrix is 2 K u L and hence only 2 K u L multiplication and
addition operations are required, where K is the number of the required harmonics of
the signal and L is number of sample data. Compared to this, the conventional linear
3
least square algorithms requires 2K 8LK 2 L number of multiplications and
additions. Therefore, the proposed method will be referred to as “Efficient least squares
algorithm” throughout this thesis. The proposed efficient least square algorithm can be
applied to extract the harmonic components in a single-phase and three-phase system
with the following advantages. (1) The proposed method extracts the fundamental and
harmonic components of distorted signals fast and accurately. (2) The proposed method
directly estimates the instantaneous fundamental and harmonic components. (3) The
proposed method is relatively less sensitive to the marginal changes in centre frequency.
(4) The proposed method has less transient time and therefore, the tracking capabilities
are superior to those of the conventional harmonic detection methods. (5) The proposed
method has relatively less computation complexity and therefore is suitable for real-
time applications.
The proposed efficient least squares method can be augmented with a simple calculation
unit to extract the instantaneous positive-negative-and zero-sequence components of the
power system signal. Also, the proposed efficient least squares method can be extended
11
to obtain other electrical power quantities such as power factor, active power, reactive
power and total harmonic distortion.
The theoretical, modelling and experimented studies presented in this thesis show that
the proposed efficient least squares method successfully satisfies all the requirements
mentioned in the previous section. The application of the proposed efficient least
squares method to an active power filter and D-STATCOM further confirms the
practicability of this method.
1.7 Structure of Thesis
This thesis is organized as follows. Chapter 2 considers the existing methods of
harmonic detection and describes the principles of operation of each method. Chapter 3
illustrates performances of the non-recursive and recursive existing harmonic
identification methods using computer simulations. The accuracy in detecting
harmonics of each method is compared with regard to (1) noise distortion, (2) effect of
the frequency variation, and (3) detection and transient response time. Chapter 4
proposes the efficient least squares algorithm. The mathematical derivation for the
proposed algorithm that begins with the linear least squares algorithm is presented. Its
properties and performances are examined. The power system signal processing system
based on the proposed efficient least squares algorithm is introduced in Chapter 5. The
mathematical derivation of various power system measurements as well as detailed
simulation and experimental studies of the proposed power system signal processing
system are presented in Chapter 5. The performance of the proposed power system
signal processing system with respect to noise, time-varying amplitudes and phase
angles and marginal fundamental frequency variations is demonstrated. Chapter 5 also
12
presents comparison performance of the proposed method with the conventional
methods, namely DFT and FFT.
Applicability of the proposed efficient least squares algorithm based power signal
processing methods in two real-life power system applications are studied in Chapters 6
and 7.
In Chapter 6, the proposed efficient least squares algorithm based harmonic detection
method is applied to insert individual harmonic components into the control system of a
three-phase active power filter (APF). Extensive simulations studies are conducted to
investigate the performance of the harmonic detection method as well as the harmonic
compensation performance of the APF. The experimental studies are carried out on a
three-phase laboratory APF setup. This chapter also discusses the detail control design
of the APF. The APF controller and the proposed harmonic detection algorithm were
implemented on a dSPACE DS1104 R&D board.
Chapter 7 presents an application of the proposed signal processing system for voltage
dip and unbalance detection to generate the current reference signal for a D-STATCOM
in order to compensate for voltage dips and unbalances. This chapter also describes the
control strategy that is based on the separation of positive- and negative-sequence
voltages and controlling them separately in order to handle both balanced and
unbalanced voltage dips. The operation and performance of the proposed D-STATCOM
system are verified via simulation and experimental studies.
Conclusions and suggestions for future work are given in Chapter 8.
13
Chapter2: Review of the Existing Techniques for Power System Harmonic Estimation
CHAPTER 2
REVIEW OF THE EXISTING TECHNIQUES

FOR POWER SYSTEM HARMONIC
ESTIMATION
2.1 Overview
This chapter reviews and studies various existing methods in the literature for harmonic
identification and enumerates advantages and drawbacks of each method. This is
achieved under the broad categorisation of the methods as non-recursive and recursive.
The non-recursive strategies possess simple and straightforward techniques to obtain
amplitudes and phase-angles of the power system signal. These methods capture a
number of samples of the input signal and process them to estimate the harmonic
components in feed forward manner. Recursive methods have more complicated
structure and can be used to achieve lower noise sensitivities. Recursive methods of
harmonic estimation assume that the error criterion is related to an under determined set
of linear functions of the unknown harmonic amplitudes and phase angles. These
methods then attempt to minimize the error criterion recursively in order to estimate the
amplitudes and phase angle of the harmonics.
In this thesis, starting from the simplest intuitive structure, three non-recursive and one
recursive structures are studied. The discrete Fourier transforms (DFT)/ fast Fourier
transform (FFT), the wavelet transform (WT), conventional least squares (CLS), and the
Kalman Filtering (KF) estimation methods are investigated.
14
Equation (2.1) represents a general power system signal which consists of a
fundamental and harmonics and is used as the input signal for the various methods to be
studied in the following literature review.
K
y t X 1 cos Z1t T1 ¦ X i cos Zi t Ti e t (2.1)
i 2
Where, X i is the unknown magnitude of the signal; where subscript 1 refers to the
fundamental component; Zi t Ti is the unknown instantaneous angle of the i th
component; and Z1 is a known fundamental angular frequency, usually fixed at 50/60
Hz. The i is an integer value and usually Zi iZ1 , where Zi represents the frequency of
the i th component. However, in some power systems, inter-harmonics exist and for
inter-harmonics Zi z iZ1 . K is the total number of harmonic components. The variable
e t represents the additive a stationary noise possesses of zero mean with variance V 2 .
2.2 Non-Recursive Harmonic Detection Methods
2.2.1 Discrete Fourier transform (DFT) and fast Fourier transform (FFT)
method
As mentioned earlier in Section 1.3, the discrete Fourier transform (DFT) is a discrete
complex valued series for calculating the Fourier coefficients. The DFT is the most
well-known tool for estimation of amplitude and phase angle of the fundamental and
harmonic components in a power system signal [16],[33],[34].
The basic equation that describes the DFT is
L 1
jZi n
x i ¦ y n e (2.2)
n 0
15
where
n is the present data point,
i is order of each harmonic component,
L is the window length (i.e. number of captured data points),
2S i ½
Zi ® ¾ is a set of fixed and equally spaced frequencies, 0 d i d L 1 ,
¯ L ¿
y n is the input signal at point n , and
x i is the DFT evaluated at frequency Zi .
Although (2.2) is described as complex series, real valued series can be obtained by
setting the imaginary part to zero. In general, the transform into the frequency domain
will be a complex valued function that includes magnitude and phase angle. The
magnitude and phase angle for the i th harmonic can be given as follows.
magnitude Xi x (i ) 2real x (i )imag

2
(2.3)
§ x (i )imag ·
phase angle Ti tan 1 ¨ ¸ (2.4)
© x (i ) real ¹
The Fast Fourier Transform (FFT) is an algorithm that efficiently computes the DFT,
and is much faster for higher number of samples (i.e. large L ). The FFT algorithm takes
advantage of the symmetry in the exponential functions e jZi n to reduce the number of
computations while computing the DFT. The idea behind the FFT is the divide and
conquer approach, to break up the original L point sample into two L data sequences.
2
16
This is because a number of smaller problems are easier to solve than one large one.
The derivation of the FFT algorithm is briefly explained below [35].
A DFT with L 2v points is considered for the divide and conquer approach. The L -
point data sequence can be split into two L data sequences f1 n and f 2 n ,
2
corresponding to the even-numbered and odd-numbered samples of y n , respectively,
that is
f1 n y 2m (2.5)
f2 n y 2m 1 , m 0,1,..., L 1 (2.6)
2
Thus f1 n and f 2 n are obtained by decimating y n by a factor of 2, and hence
the resulting FFT algorithm is called a decimation-in-time-algorithm. Now the L -point
DFT can be expressed in terms of the DFT’s of the decimated sequences as follows:
L 1
x i ¦ y n W
n 0
L
in
, i 0,1,..., L 1
¦ y n W
n even
L
in
¦ y n W
n odd
L
in
L 2 1 L 2 1
¦
m 0
y 2m W L2 mi ¦ y 2m 1W
m 0
L
2 m 1i
(2.7)
2S
j
where W L e L
.
By substituting exponential relationship, W L2 W L 2 into (2.7),
17
L 2 1 L 2 1
x i ¦ f1 m W Lim2 W Li ¦ f 2 m W Lim2
m 0 m 0
F1 i W Li F2 i , i 0,1,..., L 1 (2.8)
where F1 i and F2 i are the L -point DFTs of the sequence f1 m and f 2 m ,
2
respectively.
F1 i and F2 i are periodic with period L and can be represented as
2
F1 i L 2 F1 i and F2 i L 2 F2 i . In addition to this, the symmetry property
(i.e. W Li L 2 W Li ), can be exploited. Hence, (2.8) can be represented as (2.9) and
(2.10) below.
L
x i F1 i WLi F2 i , i 0,1,..., 1 (2.9)
2
§ L· L
x ¨ i ¸ F1 i WLi F2 i , i 0,1,..., 1 (2.10)
© 2¹ 2
2
It may be observed that the direct computation of F1 i requires L 2 complex
multiplications. The same applies to the computation of F2 i . Furthermore, there are
L additional complex multiplications required to compute W Li F2 i . Hence the

2
2
§L· L L2 L
computation of x i requires 2 ¨ ¸ complex multiplications. This first
©2¹ 2 2 2
L2 L 2
step results in a reduction of the number of multiplications from L to , which is
2 2
about a factor of 2 for large L .
18
Thus, the fundamental of the FFT algorithm described above can be depicted as in
Figure 2.1. Recombine Algebra shown in the diagram combines the samples again in
the correct order by utilizing (2.9) and (2.10).
The decimation of the data sequence can be repeated again and again until the resulting
sequences are reduced to one-point sequences. For L 2v , this decimation can be
performed v log 2 L times. Thus the total number of complex multiplications and
L
additions are reduced to log 2 L and L log 2 L respectively.
2
It can be seen that the number of operations have reduced dramatically. For example, a
calculation of the DFT requires L2 multiplications whereas the basic FFT algorithm
L
requires approximately log 2 L multiplications. Table 2.1 shows a comparison of the
2
number of complex multiplication and complex addition in the direct computation of the
DFT and in the basic FFT algorithms.
F1 0
y 0 L x 0
2 F1 1
y 2 Point x 1
DFT
§L ·
F1 ¨ 1¸
©2 ¹ L Point §L ·
y L 2 x ¨ 1¸
y(t) Recombine ©2 ¹ x Z
Algebra
F2 0 §L·
x ¨ ¸
y 1 L ©2¹
2 F2 1
y 3 §L ·
Point x ¨ 1¸
©2 ¹
DFT
§L ·
F2 ¨ 1¸
y L 1 ©2 ¹ x L 1
Figure 2.1 The first stage in the decimation-in-time algorithm.
19
Table 2.1. Comparison of computational complexity of the DFT versus FFT algorithm.
Complex multiplication Complex addition

Number of
data points, DFT, FFT, DFT, FFT,
(L) ( L2 ) L 2 log 2 L) ( L( L 1) ) ( L log 2 L )
8 64 12 56 24
16 256 32 240 64
32 1,024 80 992 160
64 4,096 192 4,032 384
128 16,384 448 16,256 896
256 65,536 1,024 65,280 2048
However, the FFT method needs data points sampled over one cycle of signal to
calculate the harmonic components accurately. In other words, the minimum possible
detection time of the FFT is one fundamental cycle. Therefore, the FFT based
algorithms are known to be unsuitable for harmonic analysis with time-varying signals
[36], [20]. Also, the corruption of the DFT and the FFT would yield inaccurate results
due to leakages/ or resolution limit of the DFT (leakages occur if the number of periods
sample is not an integer) and picket-fence (occur if the analysed waveform includes
inter-harmonics) [36], [37]. Another shortcoming of the DFT/FFT-based methods is
their sensitivity to noise [36]. Moreover, the DFT and the FFT result in significant
estimation errors for changes in fundamental frequency, even for slight changes. To
help solve this problem, recent studies have reported improvements with the DFT based
procedures which first estimate the frequency and then apply the DFT using a window
20
which is a multiple of the estimated period in order to obtain accurate harmonic
magnitudes [38]. However, these methods have more complicated structures due to the
frequency estimation requirement.
In addition, the FFT can be considered as a more general approach to harmonic
extraction, where all the harmonic frequency components within the band-width of the
signal are calculated. However, many applications, such as harmonic monitoring, may
require the extraction of a limited number of individual harmonics. For an example, in
some cases only fifth-order and seventh-order harmonics are of interest as the levels of
other harmonic components are significantly low. For such applications, the FFT
algorithm may not result in significant reduction of computational cost since it
calculates all the harmonics of the input signal.
2.2.2 Wavelet transform (WT)
To overcome some limitations in the DFT/FFT methods, a wavelet transform (WT), is
applied to the harmonic analysis [25]. A WT expands a signal not in terms of a
trigonometric polynomial but by wavelet, generated using the translation (shift in time)
and dilation (compression in time) of a fixed wavelet function called the mother
wavelet. These dilating and shifting mechanisms are desirable for analysing waveforms
containing non-stationary events. The transform coefficient magnitude indicates the
similarity in frequencies between the harmonic component of the signal and the dilated
wavelets. By continuously dilating the wavelet, harmonics frequencies in the signal can
be identified. The wavelet transform approach gives much more compact support for the
analysis of signals with localized transient components than the DFT method. This
makes the wavelet –based signal processing technique more suitable than the DFT in
21
some power system applications. Moreover, a method can also be developed to
accurately quantify the harmonic amplitudes and phases from the transform coefficients.
For practical implementation of the WT for harmonic identification, discrete wavelet
transform (DWT) or discrete wavelet packet transform (DWPT) can be employed. In
[25], the discrete wavelet packet transform (DWPT) is applied to extract harmonic
components of the power system signal. Here, high order Daubechies, a smooth
orthogonal wavelet packet, is chosen as the mother wavelet packet. This wavelet packet
is more suitable for harmonic analysis purposes than other existing wavelet packets
because they are smoother and the magnitude response of their corresponding filters is
less distorted. The DWPT decomposes the signal into harmonic trends and these
decomposed components are then analysed by the continuous wavelet transform
(CWT). In [25], a modulated Gaussian wavelet so called Morlet wavelet is applied as
the mother wavelet in the CWT to estimate the harmonic frequencies, amplitudes and
phase angles from the decomposed components.
Brief theoretical background of the method proposed in [25] is presented as follows.
x Continuous wavelet transform (CWT)
The CWT of signal y t can be defined as follows
f
CWT a, b ³ y t \ t dt
*
a ,b (2.11)
f
§ t b ·
Where \ a ,b t \ ¨ a is a translated and dilated version of the mother
© a ¸¹
wavelet \ t . The parameter ‘ a ’ corresponds to the scale (i.e. dilation) and the
22
parameter ‘ b ’ corresponds to the translation (i.e. related to the location of the window,
as the window is shifted through the signal) of \ t . Scaling, as a mathematical
operation, either dilates or compresses a signal. Larger scales corresponding to dilated
signals and small scales corresponding to compressed signals. 1 a is the
normalization factor. \ * t denotes the complex conjugate of \ t .
x Discrete wavelet transform (DWT)
The main idea of the discrete wavelet transform (DWT) is the same as that of the CWT,
but it employs a time-scale representation of a digital signal using digital filtering
techniques.
By choosing a a0p , b na0p b0 , and t kT in equation (2.6), where T =1.0 and
k , p , n are integer values, the discrete wavelet transform formula is given by
*
1 §¨ ª k na0p b0 º ·
DWT p, n
p ¨
¦k y > k @\ «« a p »» ¸¸ (2.12)
a0 © ¬ 0 ¼¹
For computational efficiency, a0 and b0 are set to 2 and 1 respectively resulting in a
binary dilation of 2 p and a dyadic translation of 2 p n .
The inverse discrete wavelet transform (IDWT) can also be given as below
1 § ·
y k ¨ ¦¦ DWT p, n \ ª¬ k na0 b0 / a0 º¼ ¸
p p
(2.13)
C\ © p n ¹
The IDWT given in (2.13) is compared with the general equation of finite impulse
response (FIR) digital filter given in (2.14), to understand the characteristics of\ k .
23
1
x > n@ ¦ y[k ]h[n k ] (2.14)
c k
It may be seen that \ k is the impulse response of a low-pass (LP) digital filter with
transfer function < Z [39]. If a0 2 , each dilation (i.e., p 1, 2,... ) of \ k
effectively halves the bandwidth of < Z . A multilevel DWT filter bank can be
implemented using (2.12) (i.e., DWT), in the forward transform stage and (2.13) (i.e.
IDWT) in the inverse transform stage.
x Discrete wavelet packet transform (DWPT)
Discrete wavelet packet transform (DWPT) is defined as
2 N 1
\ 2 n t 2 ¦ h0 k \ n 2t k (2.15)
k 0
2 N 1
\ 2 n 1 t 2 ¦ h1 k \ n 2t k (2.16)
k 0
The wavelet packets \ n , are generated form a linear combination of the scaled and
translated versions of the mother wavelet \ 1 t and its scaling function \ 0 t . h0 and
h1 are low-pass (LP) and high-pass (HP) filter of length 2 N , corresponding to the
mother wavelet.
A DWPT filter bank is implemented using (2.15) and (2.16) in a similar way that the
DWT filter bank is implemented in (2.12). Unlike the DWT filter bank where only the
(LP) output component is split, in the DWPT filter bank both the LP and HP are split at
24
each level. This results in a more general decomposition structure and can be used to
extract harmonics effectively.
A four-level DWPT filter bank structure to decompose signal y n into 16
components, d1 to d16 is shown in Figure 2.2 [25]. Parameters p and j are the level
number and decomposed output component index, respectively. H 0 and H1 represent
LP and HP filters respectively, and the coefficients of these filters are calculated using
the Daubechies wavelet packet. This DWPT gives a uniform sub-band frequency and is
used for harmonic studies. However, the amplitude of the signal attenuates when it
passes through a filter. The attenuation factor depends on the harmonic frequency and
the band width of the filter. The error due to the attenuation of the filter can be corrected
by multiplying the output by a scaling constant, which is the reciprocal of the filter gain
at each harmonic frequency fi . The DWPT filter bank contains cascaded LP and HP
filter combinations in each branch of the tree and therefore the total attenuation factor
must be considered for correction. In [25], this total attenuation factor for each
harmonic frequency is obtained by multiplying the gains of the filters at corresponding
harmonic frequency fi . The reciprocal of attenuation factor is referred as the overall
anti-distortion scaling constant ( C j f i ) in [25].
After obtaining the harmonic trends from the decomposed output sub-bands from the
DWPT, CWT can be employed to obtain the amplitudes and phase angles of the
harmonic components.
25
15 16
14 16 H1 d16
H1 14 15
12 16 H0
13 14
H1
8 16
12 14
H0
H1
12 13
#
H1 H0
11 12
10 12 H1 dj
H1 10 11
8 12 H0
H0 9 10
8 10 H1
H0 89
0 16 H0
7 8
y n 68
H1 67
H1
#
48 H0
H1 56
46 H1
H0 45
08 H0
H0 3 4
24 H1
H1 23
04 H0
H0 1 2
02 H1 d2
H0 0 1
H0 d1
p 1 p 2 p 3 p 4
Figure 2.2 Four-level DWPT decomposition tree for harmonic extraction scheme.
Equation (2.11) can be rewritten as
t2
CWT a, b ³ d (n)\ n dt
j
*
a ,b (2.17)
t1
26
where d j (n) is the decomposed output of the j th brunch. t1 and t2 are the starting and
ending times of the analysed section of the signal. The Gaussian function so called
Morlet wavelet given in (2.18) is chosen as the mother wavelet in the CWT.
2
( t b ) § t b ·
( j 2S f 0 0.5¨ ¸ )
\ a ,b t e a © a ¹
(2.18)
The parameter ‘ a ’ is chosen such that \ a ,b always covers the entire event. This fixes
the value of ‘ a ’ for a given duration. The frequency of \ a ,b can be varied by changing
f 0 in (2.18). f 0 is varied from the minimum frequency of f 0min to the maximum
frequency of f 0max with suitable increments. The value of the increment decides the
frequency resolution of the algorithm. The f 0min and f 0max are given below.
f 0 min af lj and f 0 max af hj (2.19)
where flj and f hj are the lowest and highest frequencies of d j (n) .
The parameter b in (2.18) is given by t1 and it coincides the wavelet with the analysed
section.
The harmonic frequency corresponding to the transform coefficient CWT a, b is given
by
fo
fi (2.20)
a
where fi is the ith harmonic frequency
27
The harmonic amplitude is indicated by the coefficient CWT a, b . The amplitude and
phase angle of the i th harmonic can be obtained by the coefficient of the CWT a, b as
follow:
Xi C\ a CWT a, b i T (2.21)
Im CWT a, b i
Ti tan 1 T 0i (2.22)
Re CWT a, b i
where C\ is a constant. The value of C\ can easily be found by transforming a
synthesized sinusoidal waveform [40]. X i is then multiplied by the anti-distortion
scaling constant C j f i to give the correct harmonic amplitude. T 0i is the initial phase
shift caused by the shifting of the wavelet at corresponding harmonic frequency fi by
‘ b ’.
The above described method alleviates the problem of images produced by the filter
bank, and therefore is more suitable for identification of harmonics in power system
waveforms.
However, the real-time implementations of the WT-based techniques are often
computationally prohibitive. This is due to the computational requirement of the inner
product calculations between the basic functions and the signal which are needed in
order to find the wavelet coefficients. Moreover, the efficiency and accuracy of the
wavelet transform depends on the choice of the mother wavelet and the wavelet type
should be chosen accordingly for the specific event to be studied.
28
2.2.3 Conventional least squares (CLS) method
The term least squares describes a frequently used approach to solving over determined
or inexactly specified systems of equations in an estimation. Instead of solving the
equations exactly, minimization of the sum of the squares of the residuals is only
attempted in the least squares algorithm. This makes the least squares technique a very
powerful estimation tool and in estimation theory the least square estimator is called the
“best linear unbiased estimator” (BLUE) since it minimizes the variance and does not
have bias [41].
In harmonic analysis, the least squares estimation of harmonic amplitude and phase-
angle is formulated as the solution of a set of linear equations which minimizes the
mean square error. The least squares technique possesses the advantages of simplicity
in its underlying structure, and robustness. Owing to these features, the power system
harmonic estimation can be improved dramatically by using the least square algorithm
and it has found applications in areas such as power quality monitoring [27], [28], [42].
A brief theoretical description on the conventional least squares technique in harmonic
estimation is presented below.
From the signal (2.1), when T is the sampling interval of the data acquisition system
that captures the power system waveform, the discrete form of the waveform model is
given as
K
y m ¦X i ª¬cos Zi mT cos Ti sin Zi mT sin Ti º¼ (2.23)
i 1
Equation (2.23) can be rearranged as below
29
K
y m ¦ A cos Z mT B sin Z mT
i i i i (2.24)
i 1
where Ai X i cos Ti and Bi X i sin T i ,
m - time step in the discrete sample sequence,
i - ith harmonic component ( i = 1 corresponding to the fundamental),
K - total number of required harmonic components.
The discrete-time version of (2.24) is given in matrix notation as
y Ax (2.25)
ª A1 º
ª y n º «B »
« » « 1»
« y n 1 » «# »
« » « »
# « Ai » ,
where y « », x
« y m » «B »
« » « i»
«
#
» «# »
«¬ y n L 1 »¼ «A »
« K»
«¬ BK »¼
ª cos Z1nT -sin Z1nT ! ! ! ! cos ZK nT -sin ZK nT º

«cos Z n -1 T -sin Z n -1 T -sin ZK n -1 T »»
« 1 1 ! ! ! ! cos ZK n -1 T
« # # ! ! ! ! # # »
A « »,
« # # ! cos Zi mT -sin Zi mT ! # # »
« # # ! ! ! ! # # »
« »
«¬ cos Z1 n -1 T -sinZ 1 n -1 L T ! ! ! ! cos ZK n -1 L T -sin ZK n -1 L T ¼»
n - present time step in the discrete sample sequence,
L - number of captured samples,
m (n L 1),..., n
30
and i 1,..., K .
In (2.25), the raised dot denotes matrix multiplication, A is a time-varying matrix, x is
a column vector and y is a input vector. Here, L is the number of measured samples
which is typically higher than the total number of harmonics, K . According to the
Nyquist theory, L should be greater than two times K (i.e., L ! 2 K ).
The actual data recording of the power system quantity for which harmonic analysis is
to be carried out can be denoted by sequence y m , (where m n L 1,...n for data
recording at time step n ). Now, the requirement is to find the amplitude and the phase
angle of each harmonic component such that the actual data sequence, y m , that
matches the postulated sequence, s m , as close as possible. To achieve this matching
between s m and y m , the difference between them is formed and minimised.
Using a squared error form for the purpose, the error function to be minimized is
expressed as
n 2
E ¦ ª¬ y m s (m) º¼
m n L 1
(y Ax)T y Ax
y T y 2y T Ax xT A T Ax (2.26)
where E is the error function to be minimized.
The minimum of (2.26) occurs where the derivative of E with respect to Ai and Bi can
be found as follows:
31
wE
2A T y 2A T Ax 0 (2.27)
wx
The above equation can be written in matrix form as
A A x
T
AT y (2.28)
The inverse matrix of A A

T
is closely related to the probable uncertainties of the
estimated parameters in vector x and it can be found as follows:
1
x A A T
AT y (2.29)
1
where A A
T
A T is the pseudo-inverse of matrix A .
Equation (2.29) represents the linear least squares algorithm and the amplitude and
phase angle for each harmonic component can thus be obtained from the x vector as
follows:
ª A1 º ª X 1 cos T1 º
«B » « X sin T »
« 1» « 1 1 »
« # » « # »
« » « »
x « Ai » « X i cos Ti » (2.30)
« Bi » « X i sin Ti »
« » « »
« # » « # »
«A » « X cos T »
« K» « K K
»
«¬ BK »¼ «¬ X K sin T K »¼
The amplitude X i and phase angle Ii of the i th order signal harmonic is thus given by
§B ·
Xi Ai2 Bi2 Ti tan 1 ¨ i ¸ (2.31)
© Ai ¹
32
In addition to the advantages mentioned earlier, the CLS approach also offers fast
tracking of the time-varying individual harmonic components including fundamental
component in the time-domain. In spite of the advantages, there are several performance
limitations with the CLS technique. The most prominent limitation is the high
computation cost of performing the calculation given in (2.29) due to the matrix
3
inversion and number of matrix multiplications. The CLS requires 2 K 8LK 2 L
number of multiplications /additions operations to be performed. Consequently, the
processing requirement for real-time implementation of the CLS algorithm is very high.
Moreover, real-time implementation of the conventional least squares algorithm given
in (2.29) is rather susceptible to round-off error, which is a characteristic of computer
hardware [43], due to the matrix inversion operation. This round-off error may cause
failure in the numerical process.
As a result of the above disadvantages, the CLS technique is inappropriate for efficient
real-time implementation.
2.3 Recursive Harmonic Detection Methods
2.3.1 Kalman filtering (KF) method
The Kalman filter is a set of mathematical equations that provides an efficient
computational (recursive) means to estimate the state of a process, in a way that
minimizes the mean of the squared error. The Kalman filter (KF) can be used as an
estimator to find the amplitude, frequency and phase angle of the power system signals
[44], [29], [45], [46]. KF -based algorithms have the ability to identify and track time
varying harmonic components. A KF can be used to identify harmonic components in a
noise polluted power system signal. In addition to harmonic amplitude and phase angle
33
identification, a modified KF technique that can track the system frequency, has also
been proposed [45].
A Kalman filter algorithm for harmonic detection can be briefly described as follows
[46].
The state space model and measurement equation of the power system harmonic signal
is given in (2.32) and (2.33) respectively.
x n 1 M n x n wn (2.32)
yn H n x n vn (2.33)
where
n - present time step,
x n is the 2 K u1 state vector at present time step n ,
M n is the 2 K u 2 K state transition matrix,
wn represents the discrete time variation of the state variables due to an input noise
sequence which is characterized by a covariance matrix Q n ,
yn is a scalar input signal measurement at time step n ,
H n is the 1u 2 K connection matrix between measurement and state vector;
vn is a scalar noise signal, characterized by its variance R .
The KF recursive estimation is described by the following algorithm;
34
T
Initialize the estimates xˆ n- and the state covariance matrix Pn- E ª« x n xˆ -n x n xˆ -n º» .
¬ ¼
Calculate the Kalman gain K n and update the estimates xˆ n :
1
Kn Pn- H Tn H n Pn- H Tn Rn (2.34)
xˆ n xˆ n- + K n yn H n xˆ n- (2.35)
Pn Pn- I K n H n (2.36)
xˆ -n+1 = M n xˆ n (2.37)
- T
Pn+1 = M n PnM n Q n (2.38)
If wn and vn are white Gaussian noise with zero mean value and are uncorrelated, then
the KF provides an optimal estimate of the real-time signal parameters.
The harmonic components of the signal can be identified in either the stationary or the
rotating reference frame. Here, the rotating reference frame is utilized by applying the
following conditions to the KF algorithm.
Hn ª¬cos Z nT sin Z nT ! cos Z KnT sin Z KnT º¼ (2.39)
Mn = M I (2.40)
where K is the total number of required harmonic components,
T is the sampling time period and
I is the 2 K u 2 K identity matrix.
35
In the rotating reference frame, the state variables xˆ n represent the in-phase and in-
quadrature components of the harmonics. Therefore, the magnitude and phase angle of
each harmonic component can be obtained as follows.
2 2
Magnitude X 2i 1 X 2i
n n (2.41)
X n 2i 1
Phase angle tan 1 (2.42)
X n 2i
where i is the i th harmonic component ( i =1 corresponds to the fundamental) (i.e. range
of i 1...K )
The block diagram of the above described KF algorithm for harmonic detection is
illustrated in Figure 2.3.
1
Hn Pn z
-+
+
+
+ ÷
H Tn
Qn
Kn
R
yn xn
+- ++
1
z
Figure 2.3 Kalman filter harmonic extraction scheme.
The KF starts with the initial condition of x̂0 , and P0- and then recursively calculates xˆ n .
Covariance matrices Q n and R play an important role in the KF performance. These
36
two covariance matrices should be able to successfully model variations in the system.
They should be selected based on the type of distortions and disturbances which are
present in the system. There are no systematic methods of determining these two
matrices and the trial and error approach is used to adjust them in practice. KF methods,
in general, depend strongly on the initial conditions x̂0 and P0- . If the initial conditions
are chosen incorrectly the performance of the estimator is poor, resulting in biased
estimation. Another shortcoming of the KF is its calculation time which is relatively
long [30], [31],[32].
2.4 Summary
The principles of operation of major harmonic measurement methods were explained in
this chapter. The DFT/FFT, the WT, and the CLS were studied as non-recursive
schemes. A KF-based harmonic identification method was studied as a recursive
scheme. Advantages and disadvantages of each method were discussed.
The DFT and FFT are most popular among the above methods and have simple
structure. The FFT is computationally more efficient than the DFT method. However,
both methods do not perform well in identifying time-varying harmonics. Due to this,
these methods are not suitable mitigation applications where time-varying harmonics
are present. In addition, the FFT may be considered inefficient for application where
only a limited number of harmonics are to be identified, since it estimates all the
harmonic components regardless of the specific required harmonics. The DFT/FFT
method is able to detect integer harmonics only and cannot identify inter-harmonics.
These methods are highly sensitive to noise and system frequency variation.
37
The WT method is applied with view of overcoming some of the limitations in the
DFT/FFT methods. However, the estimation efficiency and accuracy of the wavelet
transform depends on the choice of the mother wavelet and the wavelet type should be
chosen accordingly to the specific event to be studied. Also, this algorithm needs
expensive computational requirements that restrict their real-time application. This latter
shortcoming is common for the CLS method. This is due to the matrix inversion
operation and the matrix multiplications requirements of the least squares algorithm.
The computer implementation of matrix inversion may also cause round-off error which
could lead to failure in numerical process. In spite of these disadvantages, the CLS
method offers the fast tracking of the time-varying individual harmonic components
including the fundamental component in the time-domain. The CLS method provides
robust performance with respect to noise pollution.
The main advantage of the KF-based algorithm is its ability to track harmonics in highly
noise polluted power system signals. This method can also detect time-varying
harmonics. However, the potential of the KF as a tool for harmonic analysis, practically
has been limited by implementation difficulties. The response of the filter is governed
by the error covariance matrices Q n and R , which act as “tunning” parameters for the
estimation, balancing between accuracy, speed of tracking and filter divergence. In
practice, choosing appropriate parameters for desired filter operation can be an arduous
task, limiting the success of the application. Also, the KF has significantly high
computational complexity.
The next chapter compares performances of the above discussed existing harmonic
identification methods using computer simulation case studies.
38
Chapter3: Performance of the Existing Techniques for Power System Harmonic Estimation
CHAPTER 3
PERFORMANCE OF THE EXISTING

TECHNIQUES FOR POWER SYSTEM
HARMONIC ESTIMATION
3.1 Overview
This chapter evaluates the performance of the various power system harmonic
estimation methods described in the previous chapter. These methods are implemented
in MATLAB/SIMULINK computational software in order to investigate the
performances. The first part of the studies investigates effects of noise distortion,
transient response and sensitivity to marginal fundamental frequency variation for each
method individually. Then, all the methods are compared in terms of accuracy of
estimation and transient response time.
In this study, a postulated signal which consists of 50 Hz unity fundamental, 3rd, 5th, 7th,
9th and 11th order harmonic components is used to investigate the performance of each
method. This signal is given below as
y (t ) 1.00sin Z1t 45D 0.4sin 3Z1t 50D

0.3sin 5Z1t 150D 0.2sin 7Z1t 36D (3.1)
0.1 9Z1t 30D 0.05 11Z1t 120D e(t )
where Z1 = 314 rad/s, e t is additive noise distortion and the amplitudes of
fundamental and harmonics are represented as p.u. values.
39
Throughout the following study, the noise distortion is assumed to be white Gaussian
noise (WGN) of zero mean with a variance ( V 2 ). The intensity of the noise can also be
represented in terms of Signal-to-Noise-Ratio (SNR). The relationship between random
noise variance ( V 2 ) and SNR in dB is defined as
X1
SNR 20 u log (dB) (3.2)
2u V2
where X 1 is the magnitude of the signal fundamental component.
Figure 3.1 shows the postulated signal including noise distortion of SNR=17dB. This
signal is used in most of the investigations throughout this chapter and therefore, this
signal is assumed unless otherwise stated.
Figure 3.1 The postulated signal given in (3.1) with SNR of 17dB.
40
3.2 The Performance of DFT/FFT Method
This section presents the results of digital simulation case studies carried out to evaluate
the performance of the DFT/FFT method for harmonic estimation. In this example, the
input signal given in (3.1) is chosen. A sampling frequency of 3 kHz is chosen for
simulations.
The performance of the DFT/FFT against noise is studied by adding WGN to the
postulated signal. The Random number generation in SIMULINK and Randn command
in MATLAB are used to generate a noise with Gaussian distribution. The simulations
are conducted for two cases (i) polluted postulated waveform with signal to noise ratio
(SNR) of 17dB (ii) zero noise.
The performance of this method for exacting amplitude and phase angle of each
harmonic component with respect to noise is show in Figure 3.2 and Figure 3.3
respectively. The results show that the steady-state amplitude error has a maximum
value of about 0.15 pu with the 3rd harmonic and minimum of about 0.05pu with the
fundamental component under the noise pollution whereas zero error for the noiseless
signal. The steady-state phase angle error increased with the harmonic order from 5
degrees in the fundamental to a maximum of 100 degrees in 11th order harmonic.
Similar to the amplitude, the phase angle error is zero for zero noise.
The detection time of the DFT is illustrated in Figure 3.4. The top graph shows the
postulated waveform. As can be seen in this figure the DFT method takes one full
fundamental cycle for the detection of the fundamental component of the postulated
waveform.
41
Figure 3.2 Steady-state amplitude error of DFT/FFT method for each

harmonic component: zero noise (dash) and with noise of SNR =
17dB (solid).
Figure 3.3 Phase angle error of DFT/FFT method for each harmonic
component: zero noise (dash) and with noise of SNR = 17dB
(solid).
42
Figure 3.4 Detection time of DFT.
Figure 3.5 shows the transient error of the DFT method for a sudden 50% increase in
amplitudes and 20 degrees in phase angles of all the components of the postulated
waveform at t=0.1 sec. In this simulation, the increase in amplitudes and phase angles of
all the components (i.e. fundamental and harmonics) occurred simultaneously and zero
noise is assumed. As can be seen the DFT requires at least one cycle to identify this
change. No steady-state error is observed when there is zero noise presence in the
signal.
The effect of marginal fundamental frequency variation on the accuracy of the DFT
method of harmonic detection is observed. Figure 3.6 (a),(b)and (c) show the actual and
estimated fundamental, fifth and seventh-order harmonic components of a signal with a
fundamental frequency increase of 1 Hz ( i.e. from 50Hz to 51Hz) occurring at t = 0.06
sec. These figures indicate that the DFT method results in significantly high estimation
errors in the fundamental and other harmonic components for marginal changes in
fundamental frequency.
43
Figure 3.5 Fundamental amplitude and phase angle error of the DFT method
for sudden change in input signal at t=0.1 sec.
(a)
44
(b)
(c)
Figure 3.6 Actual (solid) and estimated components using DFT method (dash)
of a signal (Top), and waveform error (bottom) for 1Hz increase in
fundamental frequency at t = 0.06 sec : (a) Fundamental
component, (b) Fifth-order harmonic component and (c) Seventh-
order harmonic component.
45
3.3 The Performance of Wavelet Transform (WT) Method
This section presents the performance of the wavelet transform based harmonic
detection method. The MATLAB/Wavelet Toolbox commands are used to implement
the wavelet transform. Two different mother wavelet packets, namely Daubechies 10
(db10) and Symlets 10, (sym10) are used in a discrete wavelet packet filter bank with 4
levels. The different mother wavelet allows for the demonstrating effect of the type of
the mother wavelet on estimation accuracy. The input waveform given in (3.1) is
sampled at 3200 Hz and decomposed at 4th level (i.e. 24 16 sub-bands). Thus, each
frequency sub-band has a bandwidth of 3200/16 = 200Hz. Table 3.1 shows the
decompose component, corresponding sub bands and the harmonic order to be
estimated using a 4 level filter bank. The fundamental frequency of the given input
signal is 50 Hz. The amplitude and phase angle values of the fundamental and harmonic
components are computed from the CWT of corresponding sub-band as explained in
(2.21) and (2.22) of Chapter 2. As explained in Chapter 2, only the decomposed
components at 4th level are used to calculate the amplitude and phase angle values of the
components because each frequency sub-band at level 4 completely covers the
information of the respective harmonic component.
46
Table 3.1 Decomposed components, frequency sub-bands and fundamental and

harmonic components
Sub-bands Frequency Band (Hz) Harmonic Component
d6 500-600 11th
d5 400-500 9th
d4 300-400 7th
d3 200-300 5th
d2 100-200 3rd
d1 0-100 Fundamental
The decomposed components of the waveform given in (3.1) with noise (SNR= 17dB)
for db10 filter bank are shown in Figure 3.7. Table 3.2 shows the estimated values and
errors of amplitude and phase angles for the db10 filter bank. According to this table,
the maximum amplitude error of 0.0323 pu occurs in the 11th order harmonic and
minimum of 0.0055pu occurs in the fundamental. The maximum phase angle error
occurs in the 9th order harmonic which is 28.2 degrees. The minimum error of 1.89
degrees occurs in the estimated fundamental component.
47
Figure 3.7 First six decomposed components of postulated input waveform

using Daubechies 10 (db10) 4-level filter bank.
Table 3.2 Estimated amplitude and phase angle values using db10 4-level filter
bank.
Amplitude Phase angle

Harmonic
DWT Error DWT Error
Order
(db10) (pu) (db10) (degrees)
11th 0.0177 0.0323 -137.3 17.3
9th 0.1026 0.026 -55.19 28.2
7th 0.225 0.025 15.47 20.53
5th 0.327 0.027 146.6 3.4
3rd 0.3757 0.0242 -55.64 5.64
Fundamental 0.9945 0.0055 43.11 1.89
48
The decomposed components for the filter bank with the Symlets 10 (sym10) mother
wavelet packet is shown in Figure 3.8. Table 3.3 shows the estimated and error values
of the amplitudes and phase angles for the sym10 filter bank. The maximum amplitude
error for this case occurs in the estimated 3rd order harmonic and the value is 0.0525pu.
The minimum amplitude error of 0.007pu occurs in the 9th order harmonic component.
A maximum of 42 degrees and minimum of 1.27degrees phase angle errors occur in the
11th and 3rd order harmonic components respectively.
It is clear from the Table 3.2 and Table 3.3 that different mother wavelet packets results
in different estimation errors in amplitude and phase angle for the same input signal.
Thus, a poor selection of the mother wavelet may lead to inaccuracy in estimations.
Figure 3.8 First six decomposed components of postulated input waveform

using Symlets 10 (sym10) 4-level filter bank.
49
Table 3.3 Estimated amplitude and phase angle values using sym10 4-level filter
bank.
Amplitude Phase angle

Harmonic
DWT Error DWT Error
Order
“sym10” (pu) “sym10” (degree)
11th 0.035 0.015 -109.6 10.4
9th 0.1007 0.007 -72 42
7th 0.1676 0.0324 32.89 3.11
5th 0.2951 0.0049 135.5 14.5
3rd 0.4525 0.0525 -51.27 1.27
Fundamental 1.034 0.034 46.65 1.65
3.4 The Performance of Conventional Least Squares (CLS) Method
This section investigates the performance of the CLS-based harmonic detection method.
A sampling frequency of 3 kHz and 30 number of samples (i.e. L =30) have been chosen
for the simulation study. The same postulated waveform given in (3.1) is used as the
input signal.
Figure 3.9 and Figure 3.10 show the steady-state amplitude and phase angle error
respectively for the CLS method. This simulation is conducted for a noise polluted input
signal with SNR of 17dB and for unpolluted (i.e. zero noise) input signal. As can be
seen in Figure 3.9 the amplitude error for the polluted signal stays below 0.02 pu in the
fundamental and all harmonic components. The maximum steady-state phase angle
50
error of 15 degrees and minimum of 1 degree occur in the 11th order harmonic and the
fundamental component respectively. These can be observed in Figure 3.10.
Figure 3.9 Steady-state amplitude error of CLS method for each harmonic
(solid).
Figure 3.10 Phase angle error of CLS method for each harmonic component:
zero noise (dash) and with noise of SNR = 17dB (solid).
51
The detection time of the CLS is illustrated in Figure 3.11. The top graph shows the
postulated input waveform. As can be seen in this figure the CLS method takes less than
half of the fundamental cycle to initiate the detection of the fundamental component of
the postulated waveform.
Figure 3.11 Detection time of CLS method.
Figure 3.12 shows the transient error of the CLS method for a sudden 50% increase in
amplitude and 20 degrees in phase angles of all the component of the postulated
waveform at t=0.1 sec. This figure shows that the CLS method takes less than half of a
fundamental cycle transient time (i.e. less than 10ms).
Figure 3.13 (a),(b) and (c) show the actual and estimated fundamental, fifth-order and
seventh-order harmonic components of the signal with 1 Hz fundamental frequency
increase at t= 0.06 sec. As can be seen in these figures, the CLS methods are capable of
damping transient errors within 10 ms (i.e. less than half fundamental cycle) and the
steady-state error is about 5% for the fundamental component. However the steady-state
52
errors for the harmonic components have significantly high value as can be seen in
Figure 3.13 (a) and (b).
Figure 3.12 Fundamental amplitude and phase angle error of the CLS method
for sudden change in input signal at t=0.1 sec.
(a)
53
(b)
(c)
Figure 3.13 Actual (solid) and estimated components using CLS method (dash)
of a signal (top), and waveform error (bottom) for 1Hz increase in
fundamental frequency at t = 0.06 sec : (a) Fundamental
54
3.5 The Performance of a Kalman Filtering (KF) Method
The performance of the Kalman filter method of harmonic estimation given in Chapter
2, under different operating conditions, is investigated in this section. For this
investigation, the sampling frequency of the KF algorithm is selected to be 3200Hz (i.e.
64 u 50 Hz). The following initial conditions and parameters for the KF method are
selected for the simulation.
(i) Zero initial process xˆ 0- vector.
(ii) Null initial covariance matrix ( P0- ).
(iii) Constant noise variance ( R ) of 0.0005.
(iv) Constant diagonal state variable covariance matrix ( Q ) with each diagonal
element equal to 6 u105 .
The performance of the KF method in estimating amplitude and phase angle of the
fundamental and harmonic components is studied with respect to noise by utilizing the
6th order KF. The same postulated input signal given in (3.1) is used with (i) noise of
SNR = 17 dB and (ii) zero noise. Figure 3.14 and Figure 3.15 show the estimation error
of amplitude and phase angles respectively of the fundamental and harmonic
components. The maximum amplitude error of 0.07pu is observed in the 3rd and 5th
order harmonic components. The maximum phase angle error occurs in the 11th
harmonic and the actual value is about 65 degrees. However, it should be noted that the
estimation error of the KF method depends on noise variance ( R ), state variable
covariance matrix ( Q ) and the order of the KF. Thus, choice of proper KF parameters is
essential for good estimation of amplitudes and phase angles.
55
Figure 3.14 Steady-state amplitude error of KF method for each harmonic

(solid).
Figure 3.15 Steady-state phase angle error of KF method for each harmonic
(solid).
56
The detection time of the above KF is illustrated in Figure 3.16. The top graph shows
the postulated waveform. As can be seen in the bottom graph, it takes 15 ms to detect
the amplitude for the KF with the given initial conditions and parameters described
earlier.
Figure 3.16 Detection time of KF method.
The performance of the KF method with respect to the marginal fundamental frequency
changes is now investigated. Figure 3.17 (a),(b) and (c) show the actual and estimated
fundamental, fifth-order and seventh-order harmonic components of the signal with 1
Hz fundamental frequency increase at t = 0.06 sec. As can be seen in these figures, the
transient error is damped within 10ms. In this example the KF shows slightly less
steady-state errors in fundamental and harmonics compared to the CLS method
discussed in the previous section for fundamental frequency increase. It should be noted
again that the KF method performance depends on the KF parameters and thus, the
steady-state error values may vary for different selection of parameters.
57
(a)
(b)
58
(c)
Figure 3.17 Actual (solid) and estimated components using KF method (dash)
of a signal (top), and waveform error (bottom) for 1Hz increase in
fundamental frequency at t = 0.06 sec: (a) Fundamental
x Performance and the order of Kalman filter
In harmonic detection, the order of the KF must be chosen according to the highest
harmonic order to be estimated and selecting a lower order KF results in a larger error in
the estimated amplitudes and phase angles. Thus, higher estimation accuracy can be
achieved by choosing a higher order KF. However, the transient time of the KF
increases with the order of the KF and a higher order KF provides slower dynamic
response compared to that of a lower order KF. There is a trade-off between estimation
accuracy and transient time and therefore the selection of the order of the KF is based
on the specification for the application.
59
In order to show the effect of the order of the KF, simulations are conducted for the 1st
and 6th order Kalman filters. These Kalman filters are applied to identify two cases: (i)
step change in waveform and (ii) short duration disturbance.
Figure 3.18 shows that the KF estimation example for 100% step change in input
waveform consists of 1 pu fundamental and 0.1 pu 5th order harmonic components. As
can be seen in this figure, the 1st order KF is faster in tracking the change than is the 6th
order. The reduction in tracking speed (i.e. higher transient time) is due to the fact that
the energy of the sudden change in the amplitude is spread over all frequencies in the
higher order KF case. The 6th order KF takes approximately half a cycle before the
estimated amplitude converges to true value. Comparatively, a very short time is
required for the 1st order KF. However, the 1st order KF estimation is affected by the
presence of the harmonic element unlike the 6th order KF that gives almost negligible
steady-state error.
Figure 3.18 Transient response of the KF method: The input waveform with
step change at t=0.1sec (top), amplitude of fundamental (middle),
fundamental amplitude error (bottom).
60
Figure 3.19 shows the performance of the 1st and 6th order KFs in identifying a short
duration amplitude disturbance. For this simulation a fundamental frequency waveform
(i.e. harmonic free) of amplitude 1pu is chosen and the amplitude is increased to 1.5pu
for one cycle duration (i.e. for 20ms). The magnitude estimated by the 1st order KF is
closer to the actual amplitude during the event than is that from the 6th order KF. This is
due to the slower response of the 6th order KF. The steady-state error of the 1st order KF
estimation is negligible due the harmonic free input waveform.
Figure 3.19 Transient response of the KF method to short duration disturbance:

the waveform of short duration disturbance (top) and estimated
amplitude using KF (bottom).
3.6 Comparison of the Studied Methods
This section compares the performance of the above studied methods for power system
harmonic estimation. The comparisons are made in terms of effect of noise and
marginal variation in the fundamental frequency on the accuracy of estimation and the
transient response time of the methods.
61
3.6.1 Effect of noise distortion on estimation accuracy
The level of error incurred in estimated amplitude and phase angle versus SNR is
depicted for the DFT, WT, CLS and KF methods. A range from a highly noise polluted
signal (SNR=10dB) to a low noise polluted signal (SNR=50dB) is covered and the error
is measured in the steady-state. The maximum steady-state amplitude and phase angle
errors of the fundamental component for varying SNR are shown in Figure 3.20 and
Figure 3.21, respectively. The same sampling frequency and number of samples are
used in order to compare various methods. The KF estimation error results given in
Figure 3.20 and Figure 3.21 correspond to the 6th order KF.
Figure 3.20 shows the WT method has the maximum level of amplitude noise immunity
among the studied methods. The DFT/FFT is the most sensitive method to noise. The
level of amplitude noise immunity of the KF and CLS methods are in between the two
extremes and both methods show almost the same amplitude error for the range of SNR
from 27dB to 50dB. The noise immunity of the KF depends on the KF parameters as
explained earlier and proper parameters must be selected to reduce the estimation error.
Figure 3.21 shows that the WT method has the smallest phase angle error throughout
the given SNR range. The CLS method has almost the same phase angle error as the KF
method for high signal to noise ratios: from SNR of 35 to 50dB. The KF shows less
phase error for lower SNRs (i.e. highly noise polluted waveforms) than that of the CLS.
The DFT has the highest phase angle error among all the methods under noise pollution.
62
Figure 3.20 Comparison of estimated fundamental amplitude error for various

methods verses signal-to-noise ratio (SNR).
Figure 3.21 Comparison of estimated fundamental phase angle error for

various methods verses signal-to-noise ratio (SNR).
63
3.6.2 Effect of marginal fundamental frequency variation on the estimation

accuracy
This section investigates the robustness of the studied methods against marginal
variations in the fundamental frequency. The percentage waveform errors of the
fundamental, 5th and 7th order harmonic components for varying frequencies from 48-
52 Hz are given in Figure 3.22.
As can be seen from Figure 3.22, the KF method and CLS method have the smallest
level of error relative to the fundamental frequency change while the DFT method is the
most sensitive to fundamental frequency variations. The percentage errors for harmonic
components are significantly higher than that of the fundamental for all the methods.
Figure 3.22 Comparison of the waveform errors against fundamental frequency

variation of the studied methods.
64
3.6.3 Transient response time
Among the studied methods the CLS method takes transient response time of less than
half a cycle of the fundamental frequency. The 6th order KF, which is given above, also
responds to the amplitude and phase angle change within a half a cycle. However, the
KF transient response time depends on the parameters Q and R , and the order of the
KF as described earlier in Section 3.5. The DFT/FFT methods have slower response
compared to the CLS and KF methods and take at least 1 cycle to detect the amplitudes
and phase angles. The wavelet transforms do not usually find practical application in
dynamic real-time harmonic detection due to reasons discussed in Section 2.2.2. As a
consequence, the transient response of the WT has not been investigated in this thesis.
3.7 Summary
This chapter extensively investigated the performances of the existing harmonic
detection techniques using computer simulation. The investigation focused on studying
the effect of noise distortion, sensitivity to marginal fundamental frequency variations
and transient response of each method, namely DFT/FFT, WT, CLS, and KF.
The DFT/FFT methods take more than one cycle transient time to estimate the
amplitude and phase angle. These methods show high estimation error for noise
distorted waveforms. Thus, DFT/FFT methods are practically suited only for estimating
harmonic components of low noise distorted stationary signals in power systems.
Moreover, they have the highest sensitivity to fundamental frequency variations among
all the discussed methods.
The wavelet transform has the highest noise immunity. However, the type of mother
wavelet has great impact on the estimation accuracy and prior knowledge about the
65
measurement signal is essential for proper selection of mother wavelet. Thus, this
method is not suitable for on-line harmonic estimation of time varying signals, even
though it can provide excellent results in off-line power system disturbance analysis.
The Kalman filter method has good noise immunity and better estimation accuracy
against marginal variations of the fundamental frequency. However, there is a trade-off
between the estimation accuracy and transient response time. The performance of the
KF method depends on parameter matrices Q and R , and the order of the KF. The
higher order KF has slower transient response. Even though the lower order KF
provides fast transient response, it cannot cover higher order harmonics in the
measurement waveforms; this results in significant estimation errors in the steady-state.
The CLS-based method shows good estimation accuracy in detecting amplitudes and
phase angles under noise distortions conditions. The transient response time of the CLS
is less than half a cycle. This method also has a better estimation accuracy compared to
the DFT/FFT in identifying fundamental and harmonics under condition of marginal
variations in fundamental frequency. However, as discussed in chapter 2, this method
has the drawback of high computational complexity.
The next chapter proposes an enhancement to the liner least squares algorithm that
overcomes the main drawbacks of the CLS method.
66
Chapter4: Proposed Efficient Least Squares Technique for Power System Harmonic Detection
CHAPTER 4
PROPOSED EFFICIENT LEAST SQUARES

TECHNIQUE FOR POWER SYSTEM
HARMONIC DETECTION
4.1 Overview
After providing a rigorous definition of the harmonic extraction problem in the previous
chapters, this chapter introduces and studies the proposed development of a harmonic
identification method based on a linear least squares algorithm for real-time applications
such as monitoring and disturbance mitigation. A harmonic identification method that is
suitable for real-time monitoring and disturbance mitigation should satisfy the following
criteria:
(i) Mathematical simplicity so that it can be practically implemented
(ii) Fast detection and tracking of disturbance
(iii) Accuracy in extracting disturbance signal
(iv) Robustness against marginal changes in the system frequency
These conditions are imposed by practical requirements and each practical application
dictates the degree of speed, accuracy and robustness which it demands.
Various schemes described in Chapters 2 and 3 have been proposed in the literature to
give a solution to this problem. However, the research in this area still continues and the
67
need for faster, more precise and more robust algorithms having simple structure for
implementation purposes still exists.
In the rest of this chapter, an improvement for the conventional least squares harmonic
detection technique, which has already been described in chapter 2 and 3, is proposed
with regard to the overall problem. This improved method is referred to in this thesis as
the Efficient Least Squares algorithm. The mathematical governing equations of the
proposed method are provided and the features are illustrated. The proposed method is
also compared with existing methods in terms of the performance with respect to the
harmonic detection issue.
4.2 Mathematical Derivation of the Proposed Efficient Least Squares

Algorithm for Harmonic Detection
This section presents the proposed efficient least squares algorithm with the goal of
eliminating the problems of the conventional least squares algorithm in Chapter 2,
particularly, reducing the computational requirement so that it can be used in real-time
monitoring and disturbance mitigation applications. The intuitive foundation of the
proposed approach lies in the application of the singular value decomposition (SVD) to
the least squares algorithm. The SVD is a powerful and computationally stable
mathematical tool for solving rectangular matrices which has found many applications
in numerical computing [47],[48],[49]. It has been shown that application of the SVD
eliminates the matrix inversion problem of the conventional least squares algorithm and
also provides better noise immunity for estimation. Successful off-line harmonic
detection using the SVD based least squares algorithms have been demonstrated in the
68
literature [28], [50]. Even though these approaches eliminate the matrix inversion
problem, the computational burden remains high for effective real-time application.
In the proposed approach, real-time computational burden is reduced by rearranging and
pre-calculating matrices of the conventional least squares algorithm. The proposed
method is capable of measuring the harmonic signals accurately while requiring less
real-time computations compared to the conventional least squares and direct SVD
based methods. The analysis and implementation of the proposed method is carried out
in the time-domain. This allows convenient application of the method to real-time
power system harmonic identification for monitoring and mitigation applications.
As mentioned earlier in this section, the intuition for the proposed approach has been
drawn from the SVD based least square algorithm. Therefore, the mathematical
description for the proposed approach begins with a brief introduction to the SVD based
least squares method, even though the proposed method is not directly derived from this
latter method.
4.2.1 Singular value decomposition (SVD) based least squares method
The general power system signal given in (2.1), can be written in complex form as
shown below.
K
y t ¦X i cos Zi t T i
i 1
1 K
¦
2i1

Xi e i i e i i
j Z t T j Z t T

69
1 K
¦
2i1

X i e jTi e jZit X i e jTi e jZi t (4.1)
The discrete-time version of (4.1) can be written in matrix term as
1 K
y m ¦
2i1

X i e jTi e jZi mT X i e jTi e jZi mT (4.2)
y x
A (4.3)
where
ª X 1e jT1 º
« jT1 »
ª y n º « X 1e »
« » «# »
« y n 1 » « »
jT
« # » « X ie i »
y « », x « jTi »
and
« y ( m) » « X ie »
« # » «# »
« » « »
«¬ y n L 1 »¼ jT
« XKe K »
« X e jT K »
¬ K ¼
ª e jZ1nT e jZ1nT " " " " e jZK nT e jZK nT º

« jZ1 n 1T jZ1 n 1T jZK n 1T jZK n 1T
»
«e e " " " " e e »
1« # # " " " " # # »
A « »
2« # # " e jZi mT e jZi mT " # # »
« »
« # # " " " " # # »
«e jZ1 n L 1T e jZ1 n L T
1
" " " " e K
jZ n L T 1
e jZK n L 1T »
¬ ¼
Application of singular value decomposition allows solving the system equation without
is represented as the
matrix inversion. In this approach, the rectangular matrix A
product of three matrices [28],
70

A T
USV (4.4)
is a column-orthogonal matrix of the dimensions L u 2K , V

where U is an orthogonal
matrix of the dimensions 2K u 2K , while S is a 2K u 2K matrix of singular values with
non zero diagonal entries, s1 , s2 ,! , s p ordered in a descending way,
(i.e. s1 t s2 t ... t s p t 0 ).
By substituting (4.4) into the conventional least square algorithm given in (2.24), the
algorithm for the SVD based least squares can be found as shown below.
S 1U
x V Ty (4.5)
r r r
ª1 1 1º
where S r 1 diag « , ,..., »
«¬ s1 s2 s p »¼
The amplitude and phase angle for each harmonic component can thus be obtained. The
and V
reduced size matrices U are created from the original matrices by taking the
r r
and the first rows from the matrix V

first columns from the matrix U , respectively.
ª A1 R j A1 I º ª X 1e jT1 º
« » « jT1 »
« B1 R j B1 I » « X 1e »
«! » «# »
« » « »
« Ai j Ai » jT
« X ie i »
R I
x « » « (4.6)
jTi »
« Bi R j Bi I » « X ie »
« » «# »
«! » « »
« A j A » jT
« XKe K »
« K R K I
» « X e jT K »
« BK j BK » ¬ K ¼
¬ R I¼
Thus, the amplitude X i and phase angle Ti of the i th signal harmonic are given below.
71
Xi Ai R Ai I Bi R Bi I and
§ Ai I · § Bi I ·
Ti tan 1 ¨ ¸ tan 1 ¨ (4.7)
¨A ¸ ¨ B ¸¸
© i R¹ © i R¹
The main advantage of the SVD based least square method is that it eliminates the
matrix inversion problem of the conventional least squares method. However, this SVD
based least squares algorithm has high computational complexity since it requires three
matrix multiplications to perform. Even though this algorithm has many advantages in
off-line applications, the high computational complexity makes it unsuitable for real-
time harmonic detection applications.
This discussion leads to the proposal of another matrix manipulation technique to avoid
the shortcoming of the SVD in real-time applications. This technique is described
below.
4.2.2 Proposed efficient least squares technique
By applying the exponential relationship e jZiT n L in (4.3),

e jZi LT e jZi nT to matrix A
as shown in (4.8).
can be decomposed into matrix P and Q
matrix A

A
P Q
ª 1 1 " 1 1 º ªe jZ1nT 0 " 0 0 º

« e jZ1T e jZ1T
" e jZK T
e jZK T »» « 0 jZ1nT »
e " 0 0
1 «« j 2Z1T « »
A e e j 2Z1T " e j 2Z K T e j 2Z K T » « # # " # # »
2« »« »
« # # " # # »« 0 0 # e jZK nT
0 »
«e j L 1Z1T e j L 1Z1T " e K
j L 1 Z T j L 1ZK T » «
" e jZ K nT »
¬ e ¼¬ 0 0 0 ¼
(4.8)
72
The significance of (4.8) is that the matrix P is now a constant matrix ( L u 2 K ) (i.e.
is a time varying diagonal matrix ( 2 K u 2 K ).

time invariant). The Q
of (4.8) into the conventional least square algorithm given in

Substituting matrix A
(2.24),
1 T 1 T
ª PQ º
x A A
T Ty
A
«¬
PQ
»¼ PQ
y
1
x Q P PQ
T
T T P T y
Q
1 1
x P PQ
T Q
T T P T y
Q
-1
x Q
-1 P T P -1QP
Q Ty
-1
x Q
-1 P T P P T y (4.9)
-1

Let C P P
T 1
is a constant matrix, and Q
P T , C * (complex conjugate), since
Q
is diagonal matrix. The Q

Q * is shown below.
ªe jZ1nT 0 " 0 0 º

« jZ1nT »
« 0 e " 0 0 »
*
Q « # # " # # » (4.10)
« jZK nT »
« 0 0 # e 0 »
« 0 0 " 0 e jZK nT »¼
¬
Now, (4.9) can be written as follows:
*Cy
x Q (4.11)
73
The amplitude and phase angle for each harmonic component can thus be obtained.
ª X 1e jT1 º
« jT1 »
« X 1e »
« # »
« jT
»
« X ie i »
x « jTi »
(4.12)
« X ie »
« # »
« jT
»
« XKe K »
« X e jT K »
¬ K ¼
The amplitude X i and phase angle T i of the i th order harmonic can be obtained as
described earlier in (4.7).
is time invariant (i.e. constant) and can be pre-calculated.

As may be seen, the matrix C
Equation (4.11) shows that only two real-time matrix multiplications are required since
is pre-calculated. This algorithm also does not require matrix inversion.

the matrix C
This algorithm shows some improvement over the SVD based least squares method
with respect to the computational complexity in real-time. For example, the number of
real-time multiplication/addition operations required for this improved method is
2
L 2 K L(2 K ) and this is lower compared to that of the SVD based least squares
3 3
method: 2 K L(2 K ) 2 L(2 K ) and CLS method: 2 K 2 L(2 K ) 2 L . However,
complex arithmetic and matrix operations in the above improved method given in (4.11)
still requires a significantly large number of computational steps for real-time
applications.
The computational cost of the method given in (4.11) can be further reduced by
representing it in real number matrices using a rotational matrix. Furthermore, the real
74
number matrix equation will be rearranged to include time varying components to the
output matrix so that the algorithm can be used efficiently in harmonic detection for
mitigation applications. The following describes these developments.
The Euler's equation e jT cos T j sin T can be represented as a 2 u 2 rotation matrix
as shown below [51].
ª cos T sin T º
e jT cos T j sin T
o R (T ) « sin T (4.13)
¬ cos T »¼
in (4.8) are in the form of complex exponentials and

The elements of matrix P and Q
this can be represented in rotation matrix form with real numbers by letting the matrix
(4.13) substitute the elements of (4.8) as follows
ª R (0) R (0) " " " " R (0) R (0) º

« R (Z T ) R (Z1T ) " " " " R (ZK T ) R (ZK T ) »»
« 1
1« # # " " " " # # »

P « »
2« # # " R (Zi mT ) R (Zi mT ) " # # »
« # # " " " " # # »
« »
¬« R (Z1 ( L 1)T ) R (Z1 ( L 1)T ) " " " " R (ZK ( L 1)T ) R (ZK ( L 1)T ) »¼
(4.14)
ª R Z1nT 0 " " " " 0 0 º

« »
« 0 R Z1nT " " " " 0 0 »
« # # " " " " # # »
« »
« 0 0 " R Zi nT 0 " 0 0 » (4.15)
Q « »
0 0 " 0 R Zi nT " 0 0
« »
« # # " " " " # # »
« 0 0 " " " " R ZK nT 0 »
« »
«¬ 0 0 " " " " 0 R ZK nT ¼»
respectively.
where matrices P and Q are real number version of matrices P and Q
75
R (Zi mT ) and R (Zi mT ) elements of matrix P in (4.14) can be rearranged as
m
ª cos(Zi mT ) sin(Zi mT ) º ª cos(ZiT ) sin(ZiT ) º
R (Zi mT ) « sin(Z mT ) cos(Z mT ) » « sin(Z T ) cos(Z T ) » H im (4.16)
¬ i i ¼ ¬ i i ¼
Similarly,
R (Zi mT ) H -im (4.17)
where H -im is the transpose of H im .
By applying H im and H -im representation to (4.14),
ª H 0-1 H10 " " " " 0

H -K H 0K º
« 1 »
« H -1 H11 " " " " H 0-1 H10 »
1« # # " " " " # # »
P « » (4.18)
2« # # " H -im H im " # # »
« # # " " " " # # »
« (L-1) »
«¬ H -1 H1(L-1) " " " (L-1)
" H -K H (L-1)
K »¼
The real number version of (4.9) can be written as
-1
X Q-1 P T P P T Y (4.19)
76
ª X 1R (T1 ) º
« X R (T ) »
« 1 1 »
« X 2 R (T 2 ) » ª y n R (0) º
« » « »
« X 2 R (T 2 ) » « y n 1 R (0) »
« # » « # »
where X « » and Y « »
« X i R (Ti ) » « y m R (0) »
« X R (T ) » « # »
« i i
» « »
« # » ¬« y n L 1 R (0) ¼»
« X R (T ) »
« K K »
«¬ X K R (T K ) »¼
X is real number matrix version of complex vector x . Y is real number matrix version
real number vector y . It may be noted that each sampled data in Y is multiplied by
rotational matrix R (0) since the input data is considered as the reference. This also
result in an appropriate dimensions for output matrix Y . Similar to the complex
representation in (4.9) the time invariant matrix (i.e., constant matrix) in real number
-1
representation C = P T P P T is defined.
Substituting C into (4.19) and rearranging,
QX CY (4.20)
Lets define a real number matrix Xo as below
ª R Z1nT 0 " " " " 0 0 º ª X 1R (T1 ) º

« »« »
« 0 R Z1nT " " " " 0 0 » « X 1R (T1 ) »
« # # " " " " # # »« # »
« »« »
« 0 0 " R Zi nT 0 " 0 0 » « X i R (Ti ) »
X o = QX « » « X R (T ) »
0 0 " 0 R Zi nT " 0 0
« »« i i
»
« # # " " " " # # »« # »
« 0 0 " " " " R ZK nT 0 » « X R (T ) »
« »« K K
»
¬« 0 0 " " " " 0 R ZK nT ¼» «¬ X K R (T K ) »¼
77
ª X 1R (T1 ) u R Z1nT º
« »
« X 1R (T1 ) u R Z1nT »
« # »
« »
Xo « X i R (Ti ) u R Zi nT »
« X R (T ) u R Z nT »
« i i i
»
« # »
« X R (T ) u R Z nT »
« K K K »
«¬ X K R (T K ) u R ZK nT »¼
ª X 1R (Z1nT T1 ) º
« X R (Z nT T ) »
« 1 1 1 »
« # »
« »
Xo « X i R (Zi nT Ti ) » (4.21)
« X i R (Zi nT Ti ) »
« »
« # »
« X R (Z nT T ) »
« K K K
»
X
¬« K R ( Z K nT T K )¼
»
In the above matrix manipulation the time-varying terms in matrix Q are moved into
matrix X .
By substituting Xo = QX into (4.20),
Xo CY (4.22)
Here, Xo is the output matrix of dimension 4 K u 2 , which contains information of K
number of harmonics including the fundamental. Matrix C is time invariant (i.e.
constant matrix) and its dimensions are 4 K u 2 L . The input sampled data matrix Y has
dimensions 2 L u 2 .
78
However, (4.22) defined above contains repetitive information due to the rotational
matrix representation. As a result, the size of matrices in (4.22) are larger than required
for identifying K number of harmonics including the fundamental.
By removing the repetitive rows and columns from these matrices the following more
compact matrix relationship can be obtained. This matrix equation is given below.
xc Cc y (4.23)
where xc is a output vector containing 2K elements,
Cc is the compact version of matrix C .
Equation (4.23) can be elaborated with its element as given below.
ª X 1 cos Z1nT T1 º
« »
« X 1 sin Z1nT T1 » ª c1,1 " " " º ª y n º
c1, L
« » « # " " " # » « y n 1 »
# « »« »
« » « »
« X i cos Zi nT Ti » « # " c 2i ,m " # »
«
#
» (4.24)
« X sin Z nT T » « »
« i i i
» « # " c 2i+1,m " # » « y m »
« # » « # " " " # »« #
»
« X cos Z nT T » « »« »
«¬c 2K,1 " " " c 2K,L »¼ y n L 1 »
«
« K K K
» ¬ ¼
«¬ X K sin ZK nT T K »¼
Equation (4.24) characterises the proposed Efficient Least Squares algorithm for
harmonic identification. The proposed method samples the measuring signal y(t) at a
sampling period, T , and the discrete-time sequence y(m) is obtained
( m = n,(n - 1),...(n - L +1) ), where n corresponds to the present sample and L is the
number of samples (i.e., window size). This sampled data is represented as a vector y
of length L as shown in (4.24). Cc is a time invariant, 2K u L matrix. All the elements

79
of this matrix (i.e. c1,1 , c 2,1 …etc) can be pre-calculated given that the frequencies of the
required harmonic components ( Zi ), number of sample points ( L ) and sampling period
(T ) are known. xc is a vector of length 2K containing instantaneous cosine and sine
terms of the fundamental and harmonic components as shown in (4.24).
The proposed efficient least squares algorithm calculates instantaneous cosine and sine
terms of the fundamental and harmonic components by simply multiplying a set of input
data samples by a pre-calculated constant matrix. The algorithm does not require matrix
inversion and it contains only real numbers. This algorithm performs only one matrix
multiplication per sample time. One matrix multiplication corresponds to only 2K u L
multiplication/addition operations, since the size of the constant matrix in the efficient
least squares algorithm is 2K u L . These features make this algorithm computationally
efficient and therefore, it is very suitable for real-time harmonic detection. The number
of computations to be solved in this method depends upon the number of harmonics
required to be identified. This allows for saving in processing power when the numbers
of required harmonics are less than the total number, even if the required harmonics are
of higher order.
Generally, a minimal value for L is desired, since it will determine the time taken to
extract the harmonic. However, it should not be less than double of the total number of
harmonic components to be extracted, 2K . This requirement is to ensure that with the
proposed method there are a sufficient number of inputs and to maintain the required
accuracy. The response time of the proposed efficient least squares algorithm depends
on the sampling period (T ) and the number of captured samples ( L ), and it is given by
T u L . In order to improve the response time a smaller sampling period ( T ) can be
80
used. However, this will increase the sensitivity of the matrix, which may lead to lower
identification accuracy under noise conditions.
As noted earlier in this section, xc in (4.24) that the proposed efficient least squares
algorithm determines the instantaneous cosine and sine terms instead of the amplitudes
and the phase angles of the fundamental and harmonic components. This is
advantageous in mitigation applications as the detected instantaneous components can
be used directly. Nevertheless, it is also possible to calculate the values of the
amplitudes and phase angles of each harmonic component as follows.
The amplitude X i of the i th order component is given by
2 2
Xi ª¬ X i cos Zi nT Ti º¼ ª¬ X i sin Zi nT Ti º¼ ѽ (4.25)
If a reference time is defined, the phase angle T i of the i th order component can be
calculated by
§ X sin Zi nT Ti ·
Ti tan 1 ¨¨ i ¸¸ Zi nT ҏ (4.26)
© X i cos Zi nT Ti ¹
4.3 Performance of the Proposed Efficient Least Squares Algorithm
This section analyses computational complexity, detection time and the estimator
performance using Cramer-Rao Lower Bound (CRLB) for the proposed efficient least
squares algorithm.
81
4.3.1 Computational complexity
The computational complexity of an algorithm can be evaluated in terms of the
additions and multiplications required. Higher number of additions and multiplications
result in higher computational complexity. Among these operations, multiplication
requires significantly higher execution time compared to addition in microprocessor
based calculations.
Table 4.1 illustrates the real-time computational complexity of the proposed efficient
least squares algorithm and various other existing harmonic identification methods. In
order to facilitate the comparison with the existing algorithm, it will be assumed that the
number of captured sampling data points is 32 (i.e. L = 32); the number of harmonic
components to be extracted is 5 (i.e. K =5); and the number of harmonic components
present in the input signal is 7 (i.e., M =7).
Table 4.1 shows, from top to bottom, the computation complexity of the conventional
DFT [46], FFT method [35], conventional least square method (CLS), Kalman filter
method [46] and the proposed efficient least squares algorithm. The computational
complexity of the FFT method depends only on the number of captured sampling data
points ( L ). The computational requirements of the DFT, CLS and the proposed
methods depend on the number of captured sampling data points ( L ) as well as the
number of harmonic components to be extracted ( K ). For the KF method, the number
of operations depends on the number of harmonic components present in the input
signal ( M ).
Considering the number of calculation operation, the FFT is the most efficient; the
proposed efficient least squares algorithm also requires few calculations and becomes
82
more advantageous than the FFT method when only the fundamental components is to
be extracted (i.e., K =1). The Kalman filter is somewhat disadvantageous because its
model should include not only the harmonic components that must be detected, but also
the other significant harmonic components of the input signal. The DFT and CLS
method require a very high number of calculation operations; this confirms the
difficulty in real-time implementation of the DFT and CLS methods.
Table 4.1 Comparison of the real-time computational cost (number of calculation

operations)
Method Multiplication Addition Total
4 LK 6 K 2 LK 1
Basic DFT
670 319 989
L log L L log 2 L
2 2
Basic FFT
160 240
80
Conventional 2K
3
8LK 2 L 2K
3
8LK 2 L
Least Square (CLS) 14864
7432 7432
20M 2 8M 14 M 2 6M
Kalman Filter 1764
1036 728
Proposed efficient least 2LK 2LK

squares 320 320 640
4.3.2 Detection time
The detection time can be defined as the time that the estimated signal takes to approach
the actual value within an acceptable confidence interval and remain there for a
specified time. There is a trade-off between confidence interval and the detection time,
83
the more confidence the longer the detection time. Here, the detection time is defined
as the time taken to reach 99% confidence of the actual value after the start of the
measurement process. Figure 4.1 shows comparisons of detection time for the proposed
efficient least squares algorithm and the FFT method in identifying amplitude at 4 kHz
sampling frequency. The values of other parameters such as L and K were chosen to
be same as in Section 4.3.1. It is clear from Figure 4.1 that the proposed algorithm has
significantly less detection time compared to that of the FFT method.
Figure 4.1 Detection time: actual (solid), proposed efficient least squares
algorithm (dash) and FFT method (dot).
Figure 4.2 and Figure 4.3 show response of the proposed method to step changes in
amplitude and phase angle of the fundamental component for three different sampling
periods. These figures confirm that the response time can be improved by a reduced
sampling period (T ), as mention in Section 4.2. However, lower sampling time causes
increase in identification errors. This problem may be observed in Figure 4.2 and Figure
4.3 with 7 kHz sampling frequency. As a consequence, there is tradeoff between
detection time and identification errors.

84
Figure 4.2 Amplitude of the fundamental component from the proposed

efficient least squares method: at sampling frequency 3 kHz (dot),
at sampling frequency 5 kHz (dash), sampling frequency at 7 kHz
(dash-dot) and actual value (solid).
Figure 4.3 Phase angle of the fundamental component from the proposed
efficient least squares method: at sampling frequency 3 kHz (dot),
at sampling frequency 5 kHz (dash), at sampling frequency 7 kHz
(dash-dot) and actual value (solid).
85
4.3.3 Cramer-Rao Bound (CRB) analysis
The Cramer-Rao Lower Bound (CRLB) is a measure for evaluating the performance of
estimators under noisy conditions. The CRLB is defined to be a limit on the best
possible performance achievable for parameter estimation given a dataset. Cramer-Rao
Bound (CRB) analysis for the proposed efficient least squares algorithm and the
conventional least squares method are carried out to compare the performances of the
estimations.
The CRB theoretical analysis is conducted to set a lower bound on the variance of the
least squares as a function of noise variance. This CRLB is compared with simulated
variances of the proposed efficient least squares algorithm and the conventional least
squares method to assess the closeness to the best possible estimation. In this thesis
CRB analysis are carried out for the two main estimation parameters, namely, amplitude
and phase angle.
In the next section, the CRLB of amplitude and phase angle are found for the least
squares estimation.
x CRLB for least squares
If, X , T are the amplitude and phase angle to be estimated, the parameter vector of the
T
proposed technique can be defined as [ > X ,T @ . The Cramer-Rao Lower Bound
(CRLB) for least squares algorithm can be given in vector form as below. (Section 15.7
of [41])
1
var [ˆ t ª¬ I [ º¼ ij
i
(4.27)
86
where I [ is the complex Fisher information matrix and can be defined as
ª wC [ 1 wC [ º
ª¬ I [ º¼ ij tr «Cx 1 [ x Cx [ x »
«¬ w[i w[ j »¼
(4.28)
ª wP H [ 1 wP [ º
2 Re « Cx [ »
¬« w[i w[ j ¼»
The covariance matrix, Cx 1 [ for the least squares algorithm is obtained as explained
in Appendix A.2. By applying this covariance matrix to the Fisher information matrix
(4.28) and simplifying (Appendix A.3)
1 ª2L 0 º
I [ (4.29)
V ¬ 0 LX 2 »¼
2 «
From matrix(4.29), the CRLB for estimated amplitude and phase angle can be written as
follows:
2
V

var Xˆ t
2L
(4.30)
V2
ˆ
var T t
LX 2
(4.31)
x Comparison of simulated estimation performances of proposed
efficient least squares algorithm and CLS method with CRLB
This section analyzes the estimation performance of the proposed efficient least squares
method and the conventional least squares technique. The error variances of amplitude
and phase angle of each method are compared to CRLB for various signals to noise
87
ratios (SNR). The error variances are obtained via simulations and the parameters used
for simulation are described below.
A single-phase noise polluted waveform is sampled at 3000 Hz. This waveform has
unity amplitude and 50Hz fundamental component. A sampling window of 30 samples
is chosen (i.e. L =30) for the simulations. The noise pollution is modelled as White
Gaussian Noise (WGN) and the simulations are conducted for varying the signal to
noise ratio (SNR) from 10 to 50 dB. (i.e. the noise variance V 2 from 0.05 to
0.5 u 105 pu). For each SNR value an average of 1000 independent variance realizations
is used.
Figure 4.4 and Figure 4.5 show the CRLB and the bounds reached by both methods
when estimating amplitudes and phase angles under different noise conditions. The
results show that the proposed efficient least squares algorithm approaches the CRLB
better than does the CLS method in both amplitude and phase angle estimations. Thus
the proposed efficient least squares algorithm provides better estimation accuracy under
noise pollution condition.
88
Figure 4.4 Logarithmic variance of amplitude estimation vs. SNR: CRLB,

proposed efficient least squares method and conventional least
squares method.
Figure 4.5 Logarithmic variance of phase angle estimation vs. SNR: CRLB,
proposed efficient least squares method and conventional least
squares method.
89
4.4 Summary
This chapter introduced and systematically discussed the proposed power system
harmonic detection technique. The proposed technique is based on the least squares
algorithm and the intuition for the proposed method is drawn from the principle of
singular value decomposition (SVD). The derivation of the proposed method is carried
out using complex number representation. In the proposed method, the matrix inversion
operation which is needed in the CLS is removed using an approach similar to the SVD.
) of the signal is separated so that some

The time varying terms in the main matrix ( A
matrix operations can be pre-performed and thereby reduce the real-time computational
burden. Next, the complex representation is transformed into real number representation
using the rotational matrices. The time-varying terms are incorporated to the output
matrix in order to reduce a real-time matrix multiplication. The resultant matrix
equation is further simplified by removing the repetitive rows and columns. This
simplified matrix equation characterizes the proposed technique and it was named as the
proposed efficient least squares algorithm. The proposed efficient least squares
algorithm calculates the instantaneous cosine and sine terms of the fundamental and
harmonics components by simply multiplying a set of input data samples by a pre-
calculated constant matrix. The proposed algorithm does not require any matrix
inversion operation and contains only real numbers. This algorithm performs only one
matrix multiplication per sample time which corresponds to only 2K u L
multiplication/addition operations. The comparison shows that the proposed efficient
least squares algorithm has significantly low real-time computational complexity
compared to most of the existing harmonic detection techniques.
90
The proposed efficient least squares method retains all the good features of the CLS
method, such as fast transient response and less sensitivity to marginal changes in
fundamental frequency. In addition, Cramer-Rao bound analysis shows better
estimation performance for the proposed efficient least squares algorithm compared to
the CLS method under noise conditions.
The next chapter discusses the application of the proposed efficient least squares
algorithm for power system signal processing that is required for various real-time
monitoring and disturbance mitigation applications.
91
Chapter 5: Proposed Efficient Least Squares Algorithm based Power Signal Processing System
CHAPTER 5
PROPOSED EFFICIENT LEAST SQUARES

ALGORITHM BASED POWER SIGNAL
PROCESSING SYSTEM
5.1 Overview
This chapter presents an application of the proposed efficient least squares algorithm
which has been discussed in the previous chapter. The application covers various power
system signal processing requirements, particularly for real-time monitoring and
disturbance mitigation in power systems. The main power system quantities that need to
be identified for the monitoring, disturbance mitigation and various other purposes can
be classified as follows.
Constituting components: Identification of the fundamental and harmonic components
of the currents and the voltages in a power system is important for the disturbance
mitigation as well as the power quality monitoring. In mitigation applications such as
active power filters, the harmonic components must be identified rapidly and accurately
for the successful compensation.
Total harmonic distortion (THD) is a popular power quality measure that represents the
level of harmonic content in a power system signal. However, identification of
individual harmonic amplitudes is required for some power quality monitoring
applications. The accuracy of the estimation is the main criteria for the power quality
monitoring. In many other power system applications, the instantaneous signal (i.e.,
92
sinusoid) of a specific component or its basic information such as amplitude and phase
angle is required.
Symmetrical components: Stationary symmetrical components are used for fault
detection and evaluating the degree of unbalance in a power system. The stationary
symmetrical components can be calculated from the amplitudes and phase angles of
three-phase signals.
Instantaneous symmetrical components are used for synchronization; dynamic grid
unbalance and voltage dip mitigation. The fast estimation of these components is
essential for such voltage unbalance or dip compensation applications.
Power components: In a power system, different measures of power such as active,
reactive, harmonic, apparent powers, and power factor are defined. These measures are
required in many applications and should be calculated from the information embedded
in the voltage and the current signals.
The signal processing methods discuss in this chapter are capable of deducing the above
mentioned power system quantities from the sampled instantaneous current and voltage
signals with greater accuracy and fast response due to the application of proposed
efficient least squares algorithm. In addition the various other advantages of the
proposed efficient least squares algorithm described in the previous chapter, such as
simple structure, high noise immunity and less sensitivity to marginal changes in
fundamental frequency are also apparent in these signal processing methods. As a result,
the proposed efficient least squares algorithm can be successfully applied not only for
the harmonic detection but also for the real-time identification of various other time-
varying power system quantities.
93
Figure 5.1 depicts a general overview of the power signal processing system discussed
in this chapter. The system receives the sampled voltage and current signals as inputs
and it outputs various power system quantities. In a single-phase system the inputs are a
single pair of voltage and current signals whereas a three-phase system usually needs
three pairs of voltage and current signals which correspond to three-phases. However, in
a zero-sequence free three-phase power system, only two pairs of voltage and current
signals are required as the third pair is dependent on the other two. The outputs of the
signal processing system are the instantaneous fundamental, harmonic and symmetrical
components; RMS values and phase angles of the fundamental and the harmonic
components; the stationary symmetrical components; total harmonic distortions; and
various power measurements including power factor.
Instantaneous fundamentals of voltages and currents
Instantaneous harmonics of voltages and currents
Instantaneous symmetrical components of voltages and currents
RMS of fundamental components of voltages and currents

Voltage
Signals RMS of harmonic components of voltages and currents
Phase angles of fundamental components of voltages and currents
POWER SYSTEM Phase angles of harmonic components of voltages and currents

SIGNAL
PROCESSING Stationary Symmetrical Components
Current
Signals THD of voltages and currents
Apparent power
Active power
Reactive power
Power factor
Figure 5.1 General overview of the power signal processing system.

94
5.2 Structure of the Power Signal Processing System
Figure 5.2 shows the structure of the power system signal processor based on the
proposed efficient least squares algorithm. The structure of this power signal processing
system can be understood by considering four stages. As discussed in the previous
chapter, the proposed efficient least squares algorithm outputs the instantaneous cosine
and sine terms and the first stage of the signal processing system directly use them. The
first stage consists of identifying instantaneous fundamental and harmonic components
of the voltages and the currents. The outputs of the second stage of the signal processing
system are estimated by performing additional processing operations on the
instantaneous cosine and sine terms. The outputs of the second stage are the
instantaneous symmetrical components; RMS values and phase angles of the
fundamental and harmonic components; and active power of the fundamental and
harmonic components. The third stage consists of estimating the stationary symmetrical
components; total harmonic distortions (THD) of the voltages and the currents; and the
apparent power. The power system quantities in the third stage are obtained by
performing additional processing on the outputs of the second stage. The reactive power
and the power factor in the fourth stage are calculated from the active power and the
apparent power inputs. All these steps are illustrated in Figure 5.2.
95
Instantaneous fundamental and harmonic

components of voltages and currents
Instantaneous Instantaneous symmetrical components

Symmetrical
Components
Processing
RMS of fundamental and harmonic

Amplitude
and
Phase angles of fundamental and harmonic
Phase angle
Processing
Amplitudes of
Voltage symmetrical components
Signals Stationary
The Symmetrical
Phase angles of
Proposed Components
symmetrical components
Efficient Processing
Least
Squares
Current
Signals Algorithm
THD of voltages and currents
THD
Processing
Apparent Apparent power

Power
Processing
Active power of fundamental and

Active harmonic components
Power
Processing
Reactive Reactive power

Power
and
Power factor
Processing Power factor
Figure 5.2 Structure of the proposed power signal processing system.
96
5.3 Analysis of the Power Signal Processing System
This section analyses the powers signal processing system based on the proposed
efficient least squares algorithm. The mathematical derivations for each of the power
system quantities mentioned in the previous section will be discussed for single-phase
and three-phase power systems.
5.3.1 Power signal processing in a single-phase system
The single-phase power signal processing system requires a pair of sampled phase
voltage and current signals to estimate various power system quantities. All the
quantities given in Figure 5.1 except symmetrical components can be estimated for a
single-phase system.
As described in the previous chapter, the proposed efficient least squares algorithm
estimates the instantaneous cosine and sine terms of the fundamental and harmonic
components simply by multiplying the sampled input data (i.e. sampled voltage or
current) with a pre-calculated constant matrix. This constant matrix is common for the
both current and voltage estimations since it depends only on the sampling period and
the frequency values of the components to be identified. A common constant matrix
reduces memory requirement for the real-time digital signal processor implementation.
Figure 5.3 illustrates the application of the proposed efficient least squares algorithm in
estimating the instantaneous cosine and sine terms of the fundamental and the
harmonics of both the current and voltage signals. In this figure, the instantaneous
cosine and sine terms are represented using a different notation as given below.
97
ª v1cos º ª V1 cos Z1nT TV 1 º ªi1cos º ª I1 cos Z1nT T I 1 º

« sin » « » « sin » « »
« v1 » « V1 sin Z1nT TV 1 » « i1 » « I1 sin Z1nT T I 1 »
« # » « # » « # » « # »
« cos » « » « cos » « »
« vi » « Vi cos Zi nT TVi » and «ii » « I i cos Zi nT T Ii » (5.1)
« v sin » « V sin Z nT T » « i sin » « I sin Z nT T »
« i » « i i Vi
» «i » « i i Ii
»
« # » « # » « # » « # »
« v cos » «V cos Z nT T » «i cos » « I cos Z nT T »
« K » « K K VK
» «K » « K K IK
»
«¬ vKsin »¼ «¬VK sin ZK nT TVK »¼ «¬ iKsin »¼ «¬ I K sin ZK nT T IK »¼
where Vi and I i are the peak amplitudes of the i th component of voltage and current
respectively,
and TVi and T Ii are the phase angles of the i th component of voltage and current
respectively .
Constant Matrix ªv1cos º

« sin »
Cc « v1 »
« # »
« cos »
>v(n), v(n 1),..., v(n L 1)@

T
u «vi »
« visin »
« »
« # »
«v cos »
« Ksin »
¬« vK ¼»
ªi1cos º
« sin »
« i1 »
« # »
« cos »
T
>i(n), i(n 1),..., i(n L 1)@
u «ii »
« iisin »
« »
« # »
«i cos »
« Ksin »
¬« iK ¼»
Figure 5.3 Estimation of instantaneous cosine and sine terms of fundamental

and harmonic components of voltage and current of a single-phase
system using proposed efficient least square algorithm.
98
x Instantaneous fundamental and harmonic components of voltage and

current signals
The instantaneous fundamental and harmonic components of the voltage and the current
signals are directly given by the instantaneous cosine terms of the output matrices. This
can be represented as follows:
ª v1 º ªv1cos º
«#» « »
« » « # »
« vi » «vicos » (5.2)
« » « »
«#» « # »
«¬ vK »¼ «v cos »
¬ K ¼
ª i1 º ªi1cos º
«#» « »
« » « # »
« ii » «iicos » (5.3)
« » « »
«#» « # »
«¬iK »¼ «i cos »
¬K ¼
where vi and ii are the i th instantaneous harmonic component of the voltage and
current respectively. Usually, the fundamental is given by subscript i =1.
x RMS values and phase angles of fundamental and harmonic

components of voltage and current signals
The RMS value and the phase angles of the fundamental and each harmonic component
can be calculated using instantaneous cosine and sine terms of the proposed efficient
least squares algorithm. The RMS value of the i th component of voltage, Vi RMS is given in
(5.4).
1 cos 2 sin 2
Vi RMS
2
v v
i i (5.4)
99
The phase angle of the i th component of voltage, TVi is given in (5.5).
§ v sin ·
TVi tan 1 ¨ icos ¸ Zi nT (5.5)
© vi ¹
The RMS values and the phase angles of the fundamental and harmonic components of
the current can be determined similarly.
RMS RMS
The total RMS value of the voltage signal ( Vtotal ) and the current signal ( I total ) can be
obtained as follows:
K K
2 RMS 2
RMS
Vtotal ¦ Vi RMS
i 1
and RMS
I total ¦I
i 1
i (5.6)
x Total harmonic distortion (THD) of voltage and current
Total harmonic distortion (THD) is a measure representing the level of total harmonic
contamination in a signal and is defined as the ratio of the RMS value of the all
harmonic components to the RMS value of the fundamental component. The THD of
the voltage ( THDV ) and the current ( THDI ) can be represented mathematically as given
in (5.7) and (5.8) respectively.
K
RMS 2
¦ V
i 2
i
THDV u 100 % (5.7)
V1RMS
K
RMS 2
¦I
i 2
i
THDI u100 % (5.8)
I1RMS
100
where Vi RMS and I iRMS are the RMS values of the i th component of voltage and current
respectively.
x Single-Phase power measurements
Active power: The active power in a power system with harmonic distortion occurs
only due to interaction of the voltage and the current components that have the same
frequency. Interaction of voltage and current components that have two different
frequencies contributes only to the reactive and the apparent power.
The single-phase active power corresponding to each frequency (i.e. fundamental and
harmonics) can be efficiently estimated using the instantaneous cosine and sine terms of
the proposed efficient least squares algorithm. The active power corresponding to the
frequency of the i th component, Piactive is given in (5.9),
1 cos cos sin sin

Piactive
2
vi ii vi ii (5.9)
Proof of (5.9) is given below,
1 cos cos sin sin

Piactive
2
vi ii vi ii
1
2
Vi cos Zi nT TVi Ii cos Zi nT T Ii Vi sin Zi nT TVi Ii sin Zi nT T Ii
Vi I i
2
cos Zi nT TVi cos Zi nT T Ii sin Zi nT TVi sin Zi nT T Ii
Vi I i
4
cos 2Zi nT TVi T Ii cos TVi T Ii cos 2Zi nT TVi T Ii cos TVi T Ii
101
Vi I i
cos TVi T Ii Vi RMS I iRMS cos TVi T Ii
2
where Vi RMS I iRMS cos TVi T Ii is the familiar definition for the active power in a single-
phase system.
The total active power can be obtained by summing the active powers corresponding to
the each frequency components as given below.
K
active
Ptotal ¦P
i 1
i
active
(5.10)
Apparent power: The apparent power of a single-phase system can be obtained by
simply multiplying the total RMS values of the voltage and the current together as
shown below.
§ K
RMS 2
·§ K
RMS 2
·
S V I
RMS
total
RMS
total ¨¨ ¦I i ¸¨
¸¨ ¦ V i ¸¸ (5.11)
© i 1 ¹© i 1 ¹
Reactive power: The reactive power of a single-phase system can be obtained from the
apparent and the active power as given in (5.12).
2
P reactive S2 Ptotal
active
(5.12)
Power factor: The power factor of a power system that contains harmonics is defined
as the ratio of total active power to the apparent power. Thus, the power factor can be
written as,
active
Ptotal
PF (5.13)
S
102
5.3.2 Power signal processing in three-phase systems
Signal processing for three-phase power systems requires three pairs of sampled phase
voltage and current signals which corresponds to the three phases, to estimate the power
system quantities. All the quantities given in Figure 5.1 can be determined for three-
phase systems.
Figure 5.4 illustrates the application of the proposed efficient least squares algorithm in
estimating the instantaneous cosine and sine of the fundamental and harmonics of the
current and voltage signals in a three-phase system. As can be seen in this figure, the
three-phase system is a straightforward extension of the single-phase system described
in the previous section. The constant matrix of the proposed efficient least square
algorithm is common and therefore, a single matrix can be used as shown in Figure 5.4.
Most of the processing methods discussed for the single-phase system are common to
the three-phase system since it is a direct extension of the single-phase system. These
power quantities include the instantaneous fundamental and harmonic components, the
RMS values, phase angles and the THD. The power measurements of the three-phase
system are obtained by summing the individual phase powers as described below.
The active power corresponding to each frequency component in the three-phase can be
obtained as follows,
Pi
active
abc
P i
active
P
a i P
active
b i
active
c
1 cos cos sin sin cos cos sin sin cos cos sin sin
P i
active
abc 2
vai iai vai iai vbi ibi vbi ibi vci ici vci ici (5.14)
103
where Piactive is the three-phase active power corresponding to the i th frequency

abc
component.
The total active power Ptotal

active
can be calculated by summing the active power of all
abc
the components as given below.
K
P total
active
abc ¦P i
active
abc
(5.15)
i 1
The three-phase apparent power ( S abc ) is given by
S abc S a Sb S c V I V I V I
RMS
total a
RMS
total a
RMS
total b
RMS
total b
RMS
total c
RMS
total c (5.16)
where Sa , Sb and Sc are the apparent powers of phases- a ,- b and - c respectively.
The three-phase reactive power P reactive is defined similar to that of the single-phase
abc
system and it is given below.
2
P reactive
abc
S2abc Ptotal
active
abc
(5.17)
104
ªvacos1
º
Constant Matrix « sin »
Cc v
« a1 »
« # »
« cos »
>va (n), va (n 1),..., va (n L 1)@

T u «vai »
« vaisin »
« »
« # »
«v cos »
« aK sin
»
«¬ vaK »¼
ªvbcos 1
º
« sin »
v
« b1 »
« # »
« cos »
«vbi »
>vb (n), vb (n 1),..., vb (n L 1)@
T
u « vbisin »
« »
« # »
«v cos »
« bK sin
»
¬« vbK ¼»
ª vccos
1
º
« sin »
« »vc 1
« # »
« cos »
« vci »
> vc (n), vc (n 1),..., vc (n L 1)@
T
u « vcisin »
« »
« # »
« v cos »
« cK sin
»
¬« vcK ¼»
ªiacos 1
º
« sin »
i
« a1 »
« # »
« cos »
«iai »
>ia (n), ia (n 1),..., ia (n L 1)@
T
u « iaisin »
« »
« # »
«i cos »
« aK sin
»
«¬ iaK »¼
ªibcos 1
º
« sin »
« ib1 »
« # »
« cos »
«ibi »
>ib (n), ib (n 1),..., ib (n L 1)@
T
u « ibisin »
« »
« # »
«i cos »
« bK sin
»
¬« ibK ¼»
ªiccos1
º
« sin »
« ic1 »
« # »
« cos »
«ici »
>ic (n), ic (n 1),..., ic (n L 1)@

T
u « icisin »
« »
« # »
«i cos »
« cKsin
»
¬« icK ¼»
Figure 5.4 Estimation of instantaneous cosine and sine of the fundamental

and harmonic components of a three-phase system using the
proposed efficient least squares algorithm.
105
Zero-sequence free voltages and currents in three-phase system
In some three-phase power systems there exists a special condition where the
summation of three voltage or current signals corresponding to three phases is zero.
Such three-phase voltages or currents are called “zero-sequence free”, and these
conditions can be written as
va (t ) vb (t ) vc (t ) 0 (5.18)
ia (t ) ib (t ) ic (t ) 0 (5.19)
If the voltage and the current of a system satisfy (5.18) and (5.19) conditions, the signal
processing system based on the proposed efficient least squares algorithm requires only
two pair of input voltage and current signals, since the third phase is dependent on the
other two. Thus, it allows saving two matrix multiplications compared to the general
three-phase system. Figure 5.5 shows the application of the efficient least square
algorithm to a zero-sequence free three-phase power system.
The conditions (5.18) and (5.19) result in following relationships
vaicos vbicos vcicos 0

(5.20)
vaisin vbisin vcisin 0
iaicos ibicos icicos 0

(5.21)
iaisin ibisin icisin 0
Equation (5.20) and (5.21) can be used to estimate the instantaneous cosine and sine
terms corresponding to the third phase and hence all the power system quantities can be
calculated similar to a general three-phase system.
106
ªvacos 1
º
Cc « va1 »
« # »
« cos »
>va (n), va (n 1),..., va (n L 1)@

T
u «vai »
« vaisin »
« »
« # »
«v cos »
« aK sin
»
«¬ vaK »¼
ªvbcos 1
º
« sin »
« vb1 »
« # »
« cos »
«vbi »
>vb (n), vb (n 1),..., vb (n L 1)@
T
u « vbisin »
« »
« # »
«v cos »
« bK sin
»
«¬ vbK »¼
ªiacos 1
º
« sin »
« ia1 »
« # »
« cos »
«iai »
>ia (n), ia (n 1),..., ia (n L 1)@
T
u « iaisin »
« »
« # »
«i cos »
« aK sin
»
«¬ iaK »¼
ªibcos1
º
« sin »
« ib1 »
« # »
« cos »
«ibi »
>ib (n), ib (n 1),..., ib (n L 1)@
T
u « ibisin »
« »
« # »
«i cos »
« bKsin
»
¬« ibK ¼»
Figure 5.5 Estimation of instantaneous cosine and sine of the fundamental

and harmonic components of a zero-sequence free three-phase
system using the proposed efficient least square algorithm.
107
x Voltage unbalance
In three-phase power systems, there are situations where the three-voltage phasors are
not symmetrical. In these cases, the three-voltage phasers do not have equal amplitude,
nor are they shifted exactly with respect to each other. This is called unbalanced
voltages. Asymmetrical three-phase load currents are one of the many reasons that
causes voltage unbalance. The voltage unbalances are identified as a power quality
problem and a measure representing the level of the unbalance can be defined as
max . deviation from average voltage

Voltage unbalance u100 % (5.22)
average voltage
where, average voltage =

sum of voltage of each phase
3
All the voltage values in (5.22) are the RMS values of the fundamental components of
each phases.
x Stationary symmetrical component estimation
The stationary symmetrical components technique presents a mathematical approach for
the analysis of an asymmetrical three-phase system (under steady-state conditions) by
transforming it into a set of symmetrical sequence components called positive-,
negative-, and zero-sequence components. The stationary symmetrical components
analysis was first proposed in [52] and finds its main application in unbalanced fault
calculations of three-phase systems.
The RMS values and phase angles calculated using the proposed efficient least squares
algorithm can be used to estimate the stationary symmetrical components for the
108
fundamental and each harmonic frequency. The following describes the estimation of
stationary symmetrical components.
The stationary symmetrical component of the i th frequency can be defined as,
ªVai0 º ª1 1 1 º ªVai TVai º

RMS
« » 1« »« »
«Vai » 1 a a 2 » «Vbi TVbi »
RMS
(5.23)
« » 3 «« « »
«¬Vai »¼ ¬1 a
2
a »¼ «¬Vci TVci »¼
RMS
1 3
where a j 1120D ,
2 2
Vai TVai , Vbi TVbi and Vci TVci are phasor representations of the stationary voltages
RMS RMS RMS
using RMS values and phase angles at i th frequency for phase - a , - b and - c
respectively,
Vai0 , Vbi and Vci are the zero-, positive- and negative-sequence components given as
complex numbers.
By simplifying (5.23), the stationary symmetrical sequence components can be obtained
as follows,
1
Vai0
3
Vai TVai Vbi TVbi Vci TVci
RMS RMS RMS
^ ` ^ `
Re Vai0 j Im Vai0 (5.24)
1
Vai
3

Vai TVai Vbi TVbi 120D Vci TVci 240D
RMS RMS
RMS
(5.25)
^ `
Re V j Im V
ai ^ ` ai
109
1
Vai
3

Vai TVai Vbi TVbi 240D Vci TVci 120D
RMS RMS RMS
(5.26)
^ ` ^ `
Re V j Im V
ai ai
Equations(5.24), (5.25) and (5.26) are used to derive the stationary zero-, positive- and
negative- sequence phasors from the phase voltage phasors that have been obtained
using the proposed efficient least squares algorithm.
The amplitude ratio of the fundamental positive- and negative- sequence stationary
symmetrical components is another index used to indicate the degree of unbalance in a
three-phase system [4]. This is called the voltage unbalance factor (VUF) and is given
in (5.27) as
Va1
%VUF u 100 (5.27)
V
a1
where Va1 and Va1 are the magnitude of the positive- and negative- stationary
symmetrical components of the three-phase system.
x Instantaneous symmetrical component estimation
The concept of symmetrical components described in the previous section deals with
phasors and hence it is applicable only to the steady-state. This concept is extended to
accommodate dynamic and transient conditions [53]. The instantaneous symmetrical
components are mainly used in voltage dip and unbalance mitigation applications. Fast
and accurate extraction of the instantaneous symmetrical components is essential in
such applications and the proposed efficient least squares algorithm successfully
110
satisfies these requirements. The derivation of the instantaneous symmetrical
components is now described.
The instantaneous symmetrical components of voltage at i th frequency can be written as
follows:
ª vai0 º ª1 1 1 º ªvai º
« » 1« »
« vai » «1 D D 2 » «« vbi »» (5.28)
« » 3«
«¬ vai »¼ ¬1 D
2
D »¼ «¬ vci »¼
2S
j 1 3 2S
where D e 3
j is phase shift in the time domain,
2 2 3
vai vaicos jvaisin , vbi vbicos jvbisin and vci vcicos jvcisin are instantaneous i th voltage
components of phase- a ,- b and- c respectively, represented as complex numbers, and
0 cos sin cos sin cos sin

vai0 v
ai j vai0 , vai v ai
j vai and vai v
ai
j vai are zero,
positive- and negative- sequence components of the instantaneous i th voltage
component, represented as complex numbers.
By substituting the real and the imaginary components of vai , vbi , vci , vai0 , vai and vai into
(5.28), and simplifying,
0 cos sin 1
vai0 v ai
j vai0
3
vai vbi vci
(5.29)
1 cos cos cos 1
3

vai vbi vci
j vaisin vbisin vcisin
3

111
cos 1 sin
vai v
ai
j vai
3

vai D vbi D 2vci
cos cos
1 § cos vbi vci 3vbisin 3vcisin ·
¨ vai ¸ (5.30)
3 ¨© 2 2 2 2 ¸¹
1§ vsin vsin 3vbicos 3vcicos ·

j ¨ vaisin bi ci ¸
3 ¨© 2 2 2 2 ¸¹
and
cos 1 sin
vai v
ai
j vai
3

vai D 2vbi D vci
cos cos
1 § cos vbi vci 3vbisin 3vcisin ·
v
¨ ai ¸ (5.31)
3 ¨© 2 2 2 2 ¸¹
1§ vsin vsin 3vbicos 3vcicos ·

j ¨ vaisin bi ci ¸
3 ¨© 2 2 2 2 ¸¹
The instantaneous symmetrical sequence component are given by the real parts of (5.29)
, (5.30) and (5.31), thus
0 cos 1 cos cos cos

vai0 v
ai
3

vai vbi vci (5.32)
cos 1 § cos vbicos vcicos 3vbisin 3vcisin ·

vai
vai v
¨ ai
3 ¨©

2

2

2

2
¸¸ (5.33)
¹
cos 1 § cos vbicos vcicos 3vbisin 3vcisin ·

vai
vai v
¨ ai
3 ¨©

2

2

2

2
¸¸ (5.34)
¹
where, vai , vai and vai0 are i th order instantaneous positive- negative- and zero-
sequences component of phase-a respectively.
As may be seen, equations (5.32), (5.33) and (5.34) directly represent the instantaneous
zero-, positive- and negative- symmetrical components using instantaneous cosine and
112
sine terms obtained by the proposed efficient least square algorithm. This approach
allows derivation of the instantaneous symmetrical component individually at
fundamental and various harmonic frequencies.
5.4 Modelling and Experimental Results for the Performance of the

Power Signal Processing System
5.4.1 Performance of the proposed power signal processing system in detecting
fundamental and harmonic components
The basic performance of the efficient least squares algorithm based power signal
processing system is initially investigated through a set of computer simulations. This
section presents the results of digital simulation case studies carried out to evaluate the
harmonic estimation performance of the power system signal processing system based
on the proposed efficient least squares algorithm. Effects of varying the amplitude,
phase jumps, marginal fundamental frequency variation and noise are studied. A
sampling frequency of 4 kHz and 40 data samples (i.e., L =40) have been chosen for the
proposed efficient least squares algorithm. The postulated input voltage waveform given
in is chosen for the simulations. This waveform is heavily distorted with
harmonics as shown in Figure 5.6.
v(t ) 1.00 cos Z1t 50D 0.7 cos 3Z1t 135D

0.6 cos 5Z1t 40D 0.5cos 7Z1t 60D

0.4 cos 9Z1t 80D 0.3cos 11Z1t 75D
0.2 cos 13Z1t 18D e(t )
where e t is additive white Gaussian noise of zero mean with variance V 2 = 0.00005,
which corresponds to the random noise of SNR= 40dB.
113
Figure 5.6 The postulated voltage waveform.
Figure 5.7 shows extraction of the instantaneous fundamental, 5th and 11th order
harmonic components of the input voltage signal given in . The results show that
the extracted waveforms track the actual waveform after half a fundamental cycle
detection time.
Figure 5.8 and Figure 5.9 show RMS values and phase angles of the fundamental and
harmonic components of the postulated input voltage waveform. A detection time of
half a fundamental frequency (i.e., 10ms) is apparent again in the both RMS and phase
angle estimations.
114
Figure 5.7 Actual (solid) and extracted waveforms with the proposed method
(dot): instantaneous fundamental, 5th and 11th order harmonic
components of the input voltage waveform.
Figure 5.8 RMS values of the fundamental and harmonic components

obtained using the proposed method.
115
Figure 5.9 Phase angles of the fundamental and harmonic components

obtained using the proposed method.
x Effect of sudden amplitude and phase changes
This section investigates the tracking capability of the proposed method during sudden
changes in amplitude and phase angle. Occasionally, sudden changes in the amplitude
and phase of the signals can occur in power systems due to the switching or faults.
In order to investigate the tracking performance of the proposed method during
amplitude and phase change, the postulated voltage waveform is subjected to step
amplitude and phase changes. A 100% amplitude step increase is applied to the
fundamental, 5th and 7th order harmonic components of the input voltage waveform at t
= 0.1 sec. Figure 5.10 shows the performance of tracking the fundamental, 5th and 7th
order harmonics during the step change. As may be seen, the tracking is regained within
half cycle of the fundamental. This is further verified by the Figure 5.11 which gives
the comparison of the actual, and the reconstructed waveform obtained by summing all
instantaneous fundamental and harmonic components extracted using the proposed

116
method. Figure 5.12 shows the response of the proposed method for sudden phase
changes of 60D degrees in fundamental, 5th and 7th order harmonic components in the
input voltage waveform. This figure shows transient time similar to that for the step
amplitude change. Figure 5.13 shows the actual and the reconstructed voltage waveform
for the step phase change.
Figure 5.10 Performance of the proposed method for step amplitude change at
t = 0.1 sec.
117
Figure 5.11 Actual, and reconstructed voltage waveform using the proposed
method for step amplitude change at t = 0.1 sec.
Figure 5.12 Performance of the proposed method for step phase change at t =
0.1 sec.
118
Figure 5.13 Actual, and reconstructed voltage waveform using proposed

method for step phase change at t = 0.1 sec.
x Effect of marginal fundamental frequency changes on performance
This section studies the sensitivity of the proposed method with respect to the marginal
variations in the fundamental frequency. A step fundamental frequency change of 1 Hz
(i.e., frequency increased from 50 to 51 Hz) is applied to the input voltage waveform at
t = 0.1 sec. in order to study the effect. Figure 5.14 shows the tracking performance of
the proposed method for this step frequency change. As may be seen, the tracking is
regained within half a fundamental cycle. However, small steady state error in the phase
can be observed after the step frequency change. The steady state error can be
effectively measured using the root mean squares error (RMS) and it is defined as
follows:
erms
³ actual extracted u100% (5.36)
actual (rms )
119
Figure 5.15 compares the steady state RMS errors of the proposed method and the DFT
method for marginal fundamental frequency changes. The proposed method shows
superior estimation accuracy compared to the DFT method. For both methods, the least
error appears in the fundamental component and the significance of the error increases
as the harmonic order increases. A comparison of tracking capability of the proposed
and the DFT methods for the step increase in fundamental frequency is shown in Figure
5.16. As may be seen, the reconstructed waveform using the DFT method takes one
complete fundamental cycle to regain accurate tracking whereas the proposed method
takes only half a fundamental cycle. In addition the reconstructed waveform using the
DFT method has higher steady state phase delay compared to the proposed method.
Figure 5.17 shows the steady state RMS errors of the reconstructed waveforms using the
proposed and the DFT methods for the above described frequency change. The
proposed method shows a lower RMS waveform error of about 12.5% compared to
about 32% for the DFT method.
Figure 5.14 Tracking performance of the proposed method with step frequency
change from 50 Hz to 51 Hz at t = 0.1 sec.
120
Figure 5.15 Comparison of the RMS errors of the proposed and the DFT
methods against the fundamental frequency change.
Figure 5.16 Tracking performance of the reconstructed waveform using the

proposed method and the DFT method with frequency step
changes from 50Hz to 51 Hz at t = 0.1 sec.
121
Figure 5.17 Steady state RMS error of reconstructed waveforms using

proposed and DFT methods at 51Hz fundamental frequency.
x Effect of noise on the performance
The effect of noise on the estimation accuracy is investigated by carrying out
simulations with various SNRs. The same input voltage waveform given in is
selected and the RMS errors of the reconstructed waveform using the proposed and the
DFT methods are compared. Figure 5.18 shows this comparison. As may be seen, the
proposed method has superior noise immunity compared to the DFT method.
122
Figure 5.18 RMS error of reconstructed waveform vs. SNR for proposed and
DFT methods.
5.4.2 Performance of the power signal processing system in estimating power and
total RMS
This section discusses the performance of the proposed power signal processing system
in estimating the total RMS and the power quantities. The following postulated voltage
and current waveforms are used for the simulations.
v(t ) 1.00 cos Z1t 50D 0.7 cos 3Z1t 135D

0.6 cos 5Z1t 40D 0.5cos 7Z1t 60D
(5.37)
0.4 cos 9Z1t 80D 0.3cos 11Z1t 75D
0.2 cos 13Z1t 18D e(t )
i (t ) 0.5cos Z1t 30D 0.4 cos 3Z1t 55D

0.3cos 5Z1t 10D 0.2 cos 7Z1t 30D
(5.38)
0.1cos 9Z1t 45D 0.08cos 11Z1t 60D
0.05cos 13Z1t 10D g (t )
123
where
e t is white Gaussian noise of zero mean with variance V 2 = 0.00005 that corresponds
to voltage SNR of 40dB random noise,
and g t is white Gaussian noise of zero mean with variance V 2 = 0.00001 that
corresponds to current SNR of 40dB random noise.
The above voltage and current waveforms contain 50Hz fundamental and the 3rd, 5th,
7th, 9th, 11th and 13th order harmonics. A step amplitude of 100% is applied to both
current and voltage waveforms at t = 0.1sec in order to study the transient response. The
postulated waveforms with step changes are shown in Figure 5.19. A sampling
frequency of 4.0 kHz and 40 samples (i.e. L =40) are chosen for the proposed efficient
least squares algorithm.
Figure 5.19 Postulated voltage and current waveforms given in and
(5.38) with 100% step change at t = 0.1 sec.
124
Figure 5.20 compares the total RMS values of the voltage and the current calculated
using the proposed method and the DFT method with the actual values. As may be seen,
the proposed method has a faster transient response compared to that of the DFT
method. The steady state RMS estimation error comparison for this case is shown in
Figure 5.21. According to this figure, the proposed method has superior accuracy
compared to the DFT method.
Figure 5.22 shows comparison of estimated total active power, reactive power and
power factor using the proposed method and the DFT method. Similar to the previous
cases, the transient time for the estimation using the proposed method is less than that
for the DFT method. The total active power shown in this figure is obtained by
summing the individual active powers correspond to each harmonic component as
described in Section 5.3.1.
Figure 5.20 Comparison of estimated total RMS values of voltage and current
waveforms using proposed method and DFT method, and actual
values for 100% step change at t = 0.1 sec.
125
Figure 5.21 Comparison of steady state total RMS voltage error for proposed
method and DFT method.
Figure 5.22 Comparison of estimated total active power, total reactive power
and power factor using proposed method and DFT method, and
actual values for 100% step change at t = 0.1 sec.
126
5.4.3 Determining symmetrical components
This section presents the simulation results for calculation of symmetrical components
using the power signal processing system based on the proposed efficient least squares
algorithm. Results for stationary symmetrical components as well as instantaneous
symmetrical components are presented in this section.
x Stationary symmetrical components
The stationary symmetrical components are obtained using the calculated RMS values
and the phase angles as discussed in Section 5.3.2. The following postulated unbalanced
three-phase voltage waveforms are used for the simulation that evaluates stationary
symmetrical components.
ª1cos 2S 50t 20D 0.6 cos 2S 150t 45D º

« »
va t « 0.4 cos 2S 250t 90D 0.2 cos 2S 350t 30D » (5.39)
« »
«¬ 0.1cos 2S 450t 60D e(t ) »¼
ª1.1cos 2S 50t 45D 0.8cos 2S 150t 25D º

« »
vb t « 0.4 cos 2S 250t 95D 0.2 cos 2S 350t 15D » (5.40)
« »
«¬ 0.3cos 2S 450t 65D e(t ) »¼
ª 0.9 cos 2S 50t 30D 0.45cos 2S 150t 60D º

« »
vc t « 0.3cos 2S 250t 80D 0.2 cos 2S 350t 150D » (5.41)
« »
«¬ 0.05cos 2S 450t 45D e(t ) »¼
where the fundamental frequency is 50 Hz and the waveforms contain 3rd, 5th, 7th and
9th -order harmonics.
127
The postulated three-phase voltage waveforms given in (5.39), (5.40) and (5.41) are
shown in Figure 5.23. A sampling frequency of 4 kHz and 40 samples (i.e. L =40) are
chosen for the proposed efficient least squares algorithm in this simulation.
Figure 5.23 Postulated three-phase voltage waveforms.
Figure 5.24 shows RMS amplitudes of the stationary positive-, negative- and zero-
sequence symmetrical components of the fundamental determined by the proposed
method. Figure 5.25 shows corresponding phase angles for each sequence component.
The detection time is observed to be about half a fundamental cycle for both the
magnitudes and the phase angles.
128
Figure 5.24 RMS magnitudes of positive-, negative-, and zero-sequence

determined using proposed method.
Figure 5.25 Phase angles of positive-, negative-, and zero-sequence determined

using proposed method.
129
x Instantaneous symmetrical components
The simulation results present in this section deal with the determination of the
instantaneous symmetrical components. The postulated input three-phase voltage
waveform that is used for the simulation is shown in Figure 5.26. The postulated
waveform consists of three parts: (i) balanced unity amplitude three-phase voltages for
the time period from t = 0 to 0.1 sec.; (ii) an amplitude unbalanced (i.e., step reduction
in phase - b amplitude to 0.4 pu) three-phase voltage for the time period from t = 0.1 to
0.2 sec.; and (iii) both amplitude and phase unbalanced (i.e., step change in phase - a
amplitude from 1.0 to 1.5 pu and phase angle from 0D to 60D ), and distorted (i.e.,
contain 3rd and 5th order harmonics) three-phase voltage for the time period from t =0.2
to 0.3 sec.
Figure 5.26 Postulated three-phase voltage waveforms for instantaneous

symmetrical components estimation.
130
Instantaneous positive-, negative- and zero- sequence components of the fundamental
estimated using the proposed method are shown in Figure 5.27. As may be seen from
Figure 5.27, the perfectly balanced three-phase voltage results in only the positive-
sequence and both the zero and the negative-sequence component are absent. The
detection time is apparent in the estimated negative-and positive-sequence components.
All the sequence components appear in the unbalanced parts of the waveform. It is also
worth noticing that the proposed method is capable of extracting the instantaneous
fundamental symmetrical components from a distorted waveform. This is apparent in
the third time period of the Figure 5.27.
Figure 5.28 compares the amplitudes of the instantaneous symmetrical components
estimated using the proposed method and the DFT method. As may be seen, the
proposed method has superior transient response compared to that of the DFT method.
Figure 5.27 Instantaneous positive-, negative-, and zero-sequence components

determined using proposed method.
131
Figure 5.28 Comparison of amplitudes of positive-, negative- and zero-

sequence components determined using proposed and DFT
methods.
5.4.4 Experimental verification of the proposed efficient least squares algorithm

based harmonic detection
Experimental verification of the proposed efficient least squares algorithm is conducted
by applying it for real-time determination of artificially generated voltage harmonics.
Figure 5.29 illustrates the experimental setup used for this investigation. The
experimental setup consists of a three-phase IGBT inverter which is supplied by a three-
phase variac via a diode bridge rectifier; inductive load and a dSPACE DS1104 R&D
board hosted on a personnel computer. The dSPACE DS1104 R&D board generates the
PWM (i.e., pulse width modulated) signals which drive the IGBTs in the inverter. The
output side of the inverter is passed through an inductive filter which filters the
switching pulses in the voltages. The line voltages of the load are measured with a fast
and accurate voltage transducer board, which at the same time provides complete
isolation. The measured voltage signals are digitized with 12 bit ADCs that are located
132
on the dSPACE DS1104 R&D board. A complete description of the experimental setup
including detailed specifications of the IGBT inverter, dSPACE DS1104 R&D board
and voltage transducer board is given in Appendix C.
The automatic code generation facility provided with the dSPACE1104 development
environment is used to generate real-time C- programs from the MATLAB/SIMULINK
model. The program used in this investigation consists of two independent modules: the
module that generates PWM signals for the reference signal which contains the
fundamental and harmonics (i.e., 5th and 7th order harmonics) components; and the
module that contain the digital implementation of the proposed efficient least squares
algorithm. The sampling frequency and switching frequency for the PWM generation
program module is 9 kHz. The sampling frequency of the proposed algorithm is chosen
to be 3 kHz. The real-time waveforms are captured as a data file using capture facility
provided with the dSPACE DS1104 development environment and plotted using
MATLAB.
Variac DIODE BRIDGE IGBT INVERTER

3-Phase AC Supply
LF
+ RL
Cdc LOAD
Voltage
PWM signals Sensors
PC WITH dSPACE 1104

R&D BOARD vab
vbc
vca
Figure 5.29 Overview of the experimental setup.
133
Figure 5.30 shows the captured three-phase load voltage waveforms which contains
harmonics. Figure 5.31 shows the extracted instantaneous fundamental and its RMS
values for phases - a , - b and - c . In order to investigate the accuracy of extracted
values, the voltage waveforms are reconstructed by summing the extracted
instantaneous fundamental and the instantaneous individual harmonic components.
Figure 5.32 compares the reconstructed voltage waveform with the actual voltage
waveforms for phases- a , - b and - c . As may be seen, the reconstructed voltage
waveforms follow the actual waveform very closely. The errors in the reconstructed
waveforms are obtained by the difference between the reconstructed and the actual
waveforms. Figure 5.33 shows the error in the reconstructed voltage waveform for
phase- a . The lower error in the reconstructed waveform further confirms the accuracy
of the estimation for the proposed efficient least squares algorithm.
Figure 5.30 Captured three-phase load voltage waveforms.
134
Figure 5.31 Experimental results: extracted instantaneous fundamental

component and its RMS value for phase- a , phase- b and phase- c .
Figure 5.32 Comparison of actual waveform and the reconstructed waveform

with proposed method: phase- a , phase- b and phase- c .
135
Figure 5.33 Percentage error in reconstructed voltage waveform for phase- a .
5.5 Summary
This chapter discussed the power signal processing system based on the proposed
efficient least square algorithm. The power signal processing system is capable of
deducing a number of power system quantities which are required for various
applications, from measured instantaneous voltage and current signals. The proposed
efficient least squares algorithm directly provides the instantaneous fundamental and the
harmonic components. Various other power systems quantities, namely instantaneous
and stationary symmetrical components, RMS values and phase angles of the
fundamental and harmonic components, total harmonic distortion (THD), active power
of the fundamental and harmonics components, apparent power, reactive power and
power factor are estimated by performing additional processing on the output
instantaneous cosine and sine terms of the proposed efficient least squares method.
Extensive modelling and simulation studies were conducted in order to investigate the
performance of the proposed signal processing system. These studies show that the
136
proposed signal processing system offers many advantages such as simple structure,
high estimation accuracy, short detection time and fast transient response. In addition
the proposed method has higher noise immunity and less sensitivity to marginal
fundamental frequency changes compared to that of the basic DFT/FFT method.
Finally, the experimental verification of the proposed method is conducted using an
experimental setup that generates know harmonics artificially. The experimental results
demonstrate the applicability of the proposed efficient least squares algorithm for real-
time environments. The extensive study in this chapter indicates that the proposed
signal processing system is a potential candidate for identifying harmonics for real-time
disturbance mitigation applications.
In the next two chapters of this thesis, the application of the proposed efficient least
square algorithm based signal processing is applied to two real-life power system
applications, namely, an active power filter(APF) and a distribution static synchronous
compensator (D-STATCOM).
137
Chapter 6: Proposed Signal Processing System for Identification of Harmonics in a Three-phase Active
Power Filter Application
CHAPTER 6
PROPOSED SIGNAL PROCESSING SYSTEM

FOR IDENTIFICATION OF HARMONICS IN
A THREE-PHASE ACTIVE POWER FILTER
APPLICATION
6.1 Overview
This chapter presents an application of the proposed efficient least squares algorithm in
detection of the instantaneous current harmonics for an active power filter (APF).
Nonlinear loads such as thyristor rectifiers draw fluctuating harmonic currents from the
ac mains. The injection of harmonics causes supply voltage distortion, poor power
factor and low system efficiency. The effect of the harmonic generation on power
systems has become a serious power quality problem due to the increase use of
nonlinear loads [6], [54], [55]. Thus, it is essential to reduce the harmonic distortions in
a power system to maintain the power quality.
Harmonic currents are traditionally compensated with passive LC filters. However, the
main drawbacks of passive filtering are; inability to compensate for harmonics in the
load currents with varying magnitudes, tuning problems and the large weight of the
filtering system. To solve these problems, active power filters (APF) have been
proposed and considered as a more practical solution for reducing the current
harmonics. This latter method provides an easy tuning facility and stable operation. The
principle of the operation of an APF is to generate compensating currents into the power
138
system in order to cancel the current harmonics that are contained in the distorted load
current. The performance of the APF depends on the method used to obtain the current
reference and the control method that is used to generate the compensation currents. In a
conventional APF, the fundamental component of the source current is detected using a
method that is theoretically based on a high pass filter or a band pass filter or a low pass
filter. Then, the distortion waveform is obtained by subtracting the detected
fundamental component form the actual source current waveform. This distortion
waveform corresponds to the total harmonic content and is used to generates the
compensation currents [56], [57], [58], [59]. This method is usually called “total
harmonic compensation” since it does not require identifying each harmonic component
separately.
The main shortcomings of total harmonic compensation methods are identified as their
performance problems due to the delays of the controller and inverter phase lags. These
delays cause incorrect compensation, especially for the higher order harmonics. In
addition the system cannot effectively compensate the unbalances present in the three-
phase load currents because of the phase shift caused by the filters. This problem could
be solved by choosing a larger time constant for the filter, but this would lead to a
longer response time. With a view to solving the above mentioned problems, selective
harmonic compensation methods have been presented [60], [61], [38]. In selective
harmonic compensation, the most significant low order harmonics in a power system
are usually selected to be compensated, for example the 5th, 7th, 11th and 13thorder
harmonics. Most of the higher order harmonics are effectively attenuated by the line
impedance of the power system. Selective harmonic compensation allows for effective
139
compensation of the controller and inverter delays of the active power filter for
individual harmonic components.
vsa is Rs Ls PCC iL
vsb NON LINEAR
LOAD
vsc
LF
Voltage Source
RF Converter
iF +
Cdc
Vdc
-
Figure 6.1 Typical location for a shunt active filter based on a voltage source
converter.
Figure 6.1 shows the electrical circuit for the shunt active filter investigated in this
chapter. Shunt active filters are connected in parallel with the nonlinear loads in order to
reduce the injection of unwanted harmonic components from the load current. The main
component of a shunt APF is the voltage source converter (VSC) with dc-link
capacitors. The VSC is connected to the point of common coupling (PCC) via a
decoupling inductor that is usually the leakage inductance of a transformer. The
commands for the semiconductor switches are usually generated using the pulse width
modulation (PWM) technique.
As discussed above, the purpose of the APF is to compensate the harmonic components
of the load current iL so that only the fundamental frequency components remain in the
grid current iS . For selective harmonic compensation, the most significant active
140
individual harmonic components are extracted from the measured current and the
summation of those harmonic components is the reference for the VSC compensation
current iF .
Fast and accurate harmonic detection is very important for the selective harmonic
compensation as slow and inaccurate detection significantly affects the performance of
the APF. The discrete Fourier transform (DFT) method has been widely applied to
detect the selective order harmonic components in steady-state conditions, but one
complete fundamental cycle delay is unavoidable for this algorithm [61]. Hence the
DFT technique is not very suitable for detection of the varying load current harmonics,
especially when tracking sudden changes in the waveforms [36].
This chapter evaluates the performance of the proposed efficient least squares algorithm
in extracting harmonics for a three-phase active power filter. The rest of the chapter is
structured as follows. In Section 6.2, the system configuration of the APF including
converter modelling and controller design are discussed. The application of the
proposed efficient least squares algorithm for extracting harmonics and correcting the
current controller phase lags will be discussed in this section. In Section 6.3, the
modelling and simulation results are presented and the performance of the proposed
harmonic detection based APF is investigated. Section 6.4 provides experimental results
to demonstrate the real-time application of the proposed harmonic detection method for
the harmonic extraction for a prototype APF. A summary of the chapter will be
presented in section 6.5.
141
6.2 Three-Phase Active Power Filter System with the Proposed

Efficient Least Squares Algorithm
6.2.1 System configuration
The schematic diagram for the active power filter system is illustrated in Figure 6.2.
isa Rs Ls a iLa
isb b iLb NON LINEAR

LOAD
isc c iLc
esab esab esab

esa esb esc Harmonic detection
using the proposed
method
sin(Zt )
Z PLL cos(Zt )
abc iHa iHb
dq
LF
esd
Current controller iHa
phase lag correction
iHb RF
iFa
iCd iFb
abc
dq iCq
iFc Rdc Cdc
Vdc* 2
Vdc2 ++
*
iFd id Switching signals
Vdc2 Controller + controller
- PWM
Geneartion
iFd ma mb mc
iFa , iFb , iFc sin(Zt )
abc abc dq
iFq cos(Zt )
sin(Zt ),cos(Zt )
dq ud md mq
* -
iCq i Fq 0+ iq Decoupling
controller uq
iFd iFq Z esd
Figure 6.2 Schematic diagram of the selective harmonic APF with proposed
harmonic detection method.
142
The APF consists of a three-phase voltage source inverter, a dc-side capacitor Cdc with
its leakage resistance Rdc , and a filter inductance LF on the ac-side of the inverter. The
resistor RF represents the resistance of the cables that connects the APF to the PCC. A
three-phase controlled rectifier with a DC load is used as the nonlinear load. The three-
phase supply is represented as an ideal voltage source with an inductance LS and a
resistance RS , which characterizes the transformer and line impedances of the supply.
The analysis and design of the APF controller are conducted in the rotating reference
frame which is synchronized with the supply voltage vector. This reference frame is
usually represented in the two axes dq reference frame and all the electrical quantities
are transformed into the dq axes for design and analysis [59] [62] [63] [64].
As may be seen in Figure 6.2, the APF controller consists of two cascaded control
loops. The outer voltage control loop regulates the dc-link voltage to a required level.
The two inner dq current control loops force the APF currents iFd and iFq to follow
* * *
the command currents iFd and iFq respectively. The command iFd for the d-axis current
loop is obtained by summing the voltage controller output and the d axis component
*
of phase lag compensated harmonic current iCd . The command iFq for the q axis
current loop is obtained from the q axis component of the phase lag compensated
harmonic current iCq . These phase lag compensated harmonic currents are obtained by
correcting the current controller phase lag of the harmonic currents of phase-a and-b
(i.e. iHa and iHb ) that are obtained using the proposed harmonic detection method.
ma , mb and mc are the modulation signals for the PWM generator. The decoupling
terms are added to the output of the current controllers in order to remove the coupling
143
and then transformed to abc reference frame to obtain the modulation signal for the
converter. All these details are indicated in Figure 6.2.
6.2.2 Modelling of three-phase PWM converter
This section discusses the modelling and analysis of the three-phase PWM voltage
source converter used in the APF. Figure 6.3 shows a schematic diagram of the model
for the IGBT voltage source converter. The three-phase voltage source converter
consists of six IGBTs, and a dc-link capacitor bank ( Cdc ) to filter out the switching
ripple. The source voltages are esa , esb and esc . The ac-side of the converter is modelled
as an inductor LF and a resister RF . The terminal voltages of the converter are vFa , vFb
and vFc . The ac-side currents are iFa , iFb and iFc . Rdc represents the equalising and
leakage resistances of the dc-link capacitor bank Cdc . The dc-side voltage and current
are Vdc and idc respectively. All of these components are indicated in Figure 6.3.
idc
+
esa RF LF iFa
vFa
esb Rdc
iFb Vdc
vFb
esc iFc Cdc
vFc
Figure 6.3 Schematic diagram of three-phase voltage source converter.
The mathematical representation of the converter model can be obtained as follows.
144
By applying Kirchoff’s voltage law (KVL) to the ac-side of the voltage source
converter,
d
esa vFa RF iFa LF iFa (6.1)
dt
d
esb vFb RF iFb LF iFb (6.2)
dt
d
esc vFc RF iFc LF iFc (6.3)
dt
By assuming the power loss in the converter is negligible, the power balance of the ac-
and dc-sides can be written as follows:
vFa iFa vFb iFb vFc iFc Vdc idc (6.4)
By applying Kirchoff’s current law (KCL) to the dc-side of the converter,
d V
idc Cdc Vdc dc (6.5)
dt Rdc
By substituting (6.5) into (6.4),
d V2
vFa iFa vFb iFb vFc iFc CdcVdc Vdc dc (6.6)
dt Rdc
The voltage source converter in this thesis uses sinusoidal pulse width modulation
(SPWM) and therefore, the relationship between the terminal voltages (i.e. vFa , vFb
and vFc ) and the modulation signals (i.e. ma , mb and mc ) of phase - a , - b and - c can be
written as follows:
145
Vdc Vdc Vdc

vFa ma , vFb mb and vFc mc (6.7)
2 2 2
The complete model of the voltage source converter can be explained by (6.1), (6.2),
(6.3), (6.6) and (6.7). However, as mentioned earlier in Section 6.2.1, the analysis
provided in this chapter is conducted in the synchronously rotating reference frame
(SRF) and thus the model needs to be represented in the reference frame that rotates
synchronously with the supply frequency as described below.
By transforming (6.1), (6.2) and (6.3) into the SRF, the following equations which
describe the current dynamics in the dq axis, can be written as follows:
d
esd vFd RF iFd LF iFd Z LF iFq (6.8)
dt
d
0 vFq RF iFq LF iFq Z LF iFd (6.9)
dt
where iFd , iFq are the dq axes components of the ac-side current in the SRF,
esd is the d axis component of the source voltage in SRF and
Z is the angular frequency of the supply in elec rad/s.
The modulation signals given in (6.7) can be transformed into the SRF as below.
Vdc Vdc
vFd md and vFq mq (6.10)
2 2
where md , mq are the dq axes components of the modulation signals.
By substituting (6.10) into (6.8) and (6.9), and rearranging,
146
d R 1 V
iFd F iFd esd md dc ZiFq (6.11)
dt LF LF 2 LF
d R Vdc
iFq F iFq mq ZiFd (6.12)
dt LF 2 LF
The ac-side power of the converter can be written in the SRF as follows.
3
vFa iFa vFbiFb vFc iFc
2
vFqiFq vFd iFd (6.13)
By assuming the voltage drops in the ac-side resistance RF and inductor LF are
negligible, (6.13) can be rewritten as:
3 3
vFa iFa vFbiFb vFc iFc
2
vFq iFq vFd iFd | esd iFd
2
(6.14)
By substituting (6.14) into (6.6) and rearranging
d 1 3esd
Vdc Vdc Vdc2 iFd (6.15)
dt Rdc Cdc 2Cdc
The complete model of the three-phase converter in the SRF is given by (6.11), (6.12)
and (6.15) above.
6.2.3 Design of dq- current controllers
Equations (6.11) and (6.12) describe the dynamics of the dq axes currents in the SRF.
As may be seen in these equations, the dq axes current dynamics in the SRF are
coupled to the each other (i.e. iq depends on id and visa versa). It is necessary to
decouple them for proper control design. The decoupling can be achieved by
147
introducing new input terms ud and uq to (6.11) and (6.12) respectively. Then, the time
domain transfer functions (TF) of dq axes current can be given in (6.16) and (6.17).
d R
iFd F iFd ud (6.16)
dt LF
d R
iFq F iFq uq (6.17)
dt LF
Vdc 1 V
where uq mq ZiFd and ud esd md dc ZiFq .
2 LF LF 2 LF
As can be seen in (6.16) and (6.17), both dq axes has the same current transfer
function (TF) and these time domain TFs can be represented in Laplace domain as:
I Fd ( s ) I Fq ( s ) 1
GTI ( s ) (6.18)
U d (s) U q ( s) s RF LF
Figure 6.4 shows the closed-loop current control block diagram in the dq axes. The
decoupled current transfer function ( GTI ) and the transfer function of the PI
(proportional plus integral) current controller ( GCI ) are indicated in this figure. The
closed-loop transfer function of the dq current feedback loops can be obtained as
below.
GCI GTI K Pi s K Ii
Gcloop ( s ) (6.19)
1 GCI GTI §R ·
s 2 ¨ F K Pi ¸ s K Ii
© LF ¹
148
PI Controller
*
iFd + ud 1 iFd
K Ii GTI
- GCI K Pi s RF LF
s
(a) d axis
*
PI Controller
iFq + K Ii uq 1 iFq
GCI K Pi GTI
- s s RF LF
(b) q axis
Figure 6.4 dq axis closed-loop current control diagrams.
Design of gains of the PI-controller is conducted using the ITAE index. ITAE stands
for “integral of time multiplied by absolute magnitude of the error” and it is defined as
follows:
T
ITAE ³ t e(t ) dt
0
(6.20)
where e(t ) is the absolute error and t is the time.
The coefficients of closed-loop transfer function that minimizes the ITAE performance
criterion for a ramp input have been used for determining the gains of the PI controller.
By comparing the denominator of (6.19) with the optimum coefficients of the ITAE for
a ramp input for 2nd-order transfer function given in [65], the parameters of the PI
current controller ( GCI ) can be selected as follows:
149
RF
K Pi 3.2Zni and K Ii Zni2 (6.21)
LF
where Zni is the natural frequency of the closed-loop current TF.
K Pi and K Ii are the proportional and integral gains respectively. The dynamic response
of the closed-loop current TF depends on the natural frequency ( Zni ) and the value of
Zni is chosen for the desired dynamic response. Figure 6.5 shows the frequency
response (i.e. bode plot) of the closed-loop current TF. The natural frequency is chosen
to be 1257 rad/s. The values of RF and LF are 0.078 : and 6mH respectively. As may
be seen, -3dB cut-off frequency of the closed-loop is 4630rad/sec. This value is larger
than the frequency of the 13th order harmonic component which is the highest order
harmonic to be compensated with this APF. It may be also noted from the bode diagram
that the current controller has larger phase lag for the higher order harmonics despite of
the lower attenuation in the magnitude. For example at 4380rad/sec the phase lag is 50.1
degrees although the magnitude attenuation is only -3dB. As discussed in the
introduction to this chapter, controller phase lag affects the performance of the APF in
compensating higher order harmonics. The proposed signal processing method can be
effectively utilized to compensate the controller phase lag. This will be discussed in
detail in a later section of this chapter.
150
Figure 6.5 Bode diagram of the closed-loop dq current TF.
6.2.4 Design of dc voltage controller
The voltage control loop regulates the dc-link voltage to a set level that is large enough
to generate the compensation currents. Equation (6.15) which describes the relationship
between Vdc and iFd , is a non-linear equation. This equation is rearranged so that it can
be treated using linear control theory.
Multiplying (6.15) by 2 and rearranging,
d 2
(Vdc ) 2 (Vdc ) 2 p (6.22)
dt Rdc Cdc
3esd
where p iFd .
Cdc
Equation (6.22) can be written in Laplace domain as:
151
Vdc2 ( s) 1
GTV (6.23)
P( s) s 2 RdcCdc
As may be seen in (6.23), Vdc2 instead of Vdc has a linear transfer function and can be
used for control design. This does not cause any technical problems since Vdc is
unidirectional [64]. The block diagram of the outer voltage control loop is illustrated in
Figure 6.6. The dc-link voltage is regulated via d axis current and the d axis inner
current loop is assumed to be very fast compared to the outer voltage loop and
consequently, the inner d axis current loop can be replaced with unity gain during the
design of the dc-link voltage controller.
PI Controller
*2 *
V dc + K iFd iFd GTv
1 Vdc2
GCV K Pv Iv 1
s 2
- s RdcCdc
Figure 6.6 Control block diagram of the voltage control loop.
The closed-loop transfer function of the outer voltage loop ( Gdc _ cloop ) is shown below in
(6.24).
GCV GTV K Pv s K Iv
Gdc _ cloop ( s ) (6.24)
1 GCV GTV 2 § 2 ·
s ¨ K Pv ¸ s K Iv
© Rdc Cdc ¹
This closed-loop transfer function has 2nd order characteristics. The proportional and
integral constants (i.e. K Pv and K Iv ) of the PI controller can be obtained by comparing
152
the denominator of this transfer function to the ITAE-criterion for a ramp input for 2nd-
order transfer function [65] as follows:
2
K Pv 3.2Znv , K Iv Znv2 (6.25)
Rdc Cdc
where Znv is the natural frequency of the closed voltage loop.
The natural frequency of the outer loop ( Znv ) is chosen to be considerably lower than
the natural frequency of the inner current controller ( Zni ) to avoid any possibility of
formation of higher order systems. Figure 6.7 shows the frequency response of the outer
closed-loop dc-link voltage transfer function. This bode diagram corresponds to the
natural frequency ( Znv ) of 31.4 rad/s. The values of Cdc and Rdc are 8250 P F and
6000 : respectively.
Figure 6.7 Bode diagram of the closed dc-link voltage control loop.
153
6.2.5 Current harmonic detection and controller phase lag correction based
on the proposed efficient least squares algorithm
The proposed efficient least squares algorithm is next utilized to extract the harmonic
current components to be compensated using the APF as illustrated in Figure 6.2. As
discussed earlier in Section 6.1, the main advantage of the selective harmonic
compensation is that only the selected harmonic components are chosen to be
compensated. The proposed method extracts the instantaneous individual current
harmonic components and these harmonic components are summed up to generate the
compensation command. The APF described in this chapter deals only with the zero-
sequence free currents and thus the measuring of two currents instead of three are
adequate. Figure 6.8 shows the application of the proposed efficient least squares
algorithm in extracting individual current harmonics.
ªiacos1
º
Cc « ia1 »
« # »
« cos »
«iai »
T
>ia (n), ia (n 1),..., ia (n L 1)@
u « iaisin »
« »
« # »
«i cos »
« aKsin
»
«¬ iaK »¼
ªibcos1
º
« sin »
« ib1 »
« # »
« cos »
«ibi »
T
>ib (n), ib (n 1),..., ib (n L 1)@
u « ibisin »
« »
« # »
«i cos »
« bKsin
»
¬« ibK ¼»
Figure 6.8 Proposed efficient least squares algorithm for identification of

current harmonics for APF.
154
As described in the previous chapter, the instantaneous harmonic components can be
directly obtained from the instantaneous cosine terms. In this APF, the 5th, 7th, 11th and
13th order harmonics are chosen to be compensated. These harmonic components for
phase-a and -b can be obtained as shown below.
ª iHa 5 º ªiacos
5
º
«i » « cos »
« Ha 7 » «ia 7 » (6.26)
«iHa11 » «iacos
11
»
« » « cos »
¬iHa13 ¼ ¬«ia13 ¼»
ª iHb 5 º ªibcos
5
º
«i » « cos »
« Hb 7 » «ib 7 » (6.27)
«iHb11 » «ibcos
11
»
« » « cos »
¬iHb13 ¼ ¬«ib13 ¼»
where
iaicos I ai cos Zi t T ai ,
ibicos Ibi cos Zi t Tbi and
iHai and iHai are the instantaneous i th harmonic components of the phase-a and b currents
respectively.
The compensation command for phase -a and -b can be obtained by summing the
individual instantaneous components in (6.26) and (6.27) respectively as given below.
iTHa ¦
i 5,7,11,13
iHai ¦
i 5,7,11,13
iaicos (6.28)
155
iTHb ¦
i 5,7,11,13
iHbi ¦
i 5,7,11,13
ibicos (6.29)
where iTHa and iTHb are the compensation commands for phase-a and -b respectively.
As discussed earlier in this chapter, the SRF rotating at fundamental frequency has been
utilized for the analysis and control design of the APF. The currents at the fundamental
frequency become dc-signals after transforming them into the SRF, whereas the
harmonic components remain as ac-signals. The frequency of a balanced three-phase
harmonic component in the SRF fixed to the fundamental frequency can be obtained by
(i 1) f1 , where i is the harmonic order and f1 is the fundamental frequency. For
example the 13th order harmonic has a frequency of 600Hz in the SRF that is fixed to
the fundamental frequency of 50Hz.
The current controllers introduce a phase delay (i.e. phase-lag) for ac components.
These phase-lags increase with the frequency and seriously affect the overall
performance of the APF, particularly the compensation of the high order harmonic
currents. Figure 6.9 shows the phase-lags for the 5th, 7th, 11th and 13th order harmonics
for the current controller that is designed in Section 6.2.3.
156
Figure 6.9 Phase-lags of the current controller for various harmonics

components.
As mentioned earlier in Section 6.1, one advantage of using the selective harmonic
components for the APF application is that it allows for correcting the phase lag with
each the harmonic components. The correction for the current controller phase lag can
be achieved by adding the phase angles (i.e., introducing phase lead) given in Figure 6.9
to each harmonic component of the reference. These phase leads can be easily
introduced to the reference signal using the proposed method as follows.
From (6.26), the i th instantaneous harmonic component of the phase-a can be written
as:
iHai iaicos I ai cos Zi t T ai (6.30)
157
By introducing phase lead, Ii to the instantaneous harmonic component iHai given in
(6.30), the phase lag corrected instantaneous harmonic signal iCHai can be written as
follows.
iCHai I ai cos Zi t T ai Ii (6.31)
By expanding the right hand side of (6.31),
iCHai I ai cos Zi t T ai cos Ii I ai sin Zi t T ai sin Ii

(6.32)
iaicos cos Ii iaisin sin Ii
The instantaneous cosine and sine terms (i.e. iaicos and iaisin ) in (6.32) are directly obtained
form the proposed efficient least squares algorithm as shown in Figure 6.8. The phase
lag Ii is obtained from the closed-loop frequency response of the phase angle as shown
in Figure 6.9.
Now, the phase lag corrected compensation command for phase-a can be obtained by
summing the individual instantaneous components in (6.32) as given below.
iTCHa ¦
i 5,7,11,13
iCHai ¦ i
i 5,7,11,13
cos
ai cos Ii iaisin sin Ii (6.33)
Similarly for phase-b
iTCHb ¦
i 5,7,11,13
iCHbi ¦ i
i 5,7,11,13
cos
bi cos Ii ibisin sin Ii (6.34)
where iTCHa and iTCHb are the phase lag corrected compensation commands for phase-a
and -b respectively.
158
The current controllers also cause magnitude attenuation as shown in Figure 6.10. As
may be seen, these attenuations are not very significant. Nevertheless, the magnitude
attenuation can be corrected simply by multiplying each harmonic component by a
factor that is calculated from the frequency response. The magnitude correction factor
for the i th harmonic component ( Magi ) can be obtained as follows.
1
Magi § dBi ·
(6.35)
¨ ¸
© 20 ¹
10
where dBi is the magnitude attenuation of the current controller for the i th harmonic
component.
Figure 6.10 Magnitude attenuation of the current controller for various

harmonics components.
159
6.3 Modelling Results for Three-Phase Active Power Filter

Based on the Proposed Harmonic Detection Method
6.3.1 Modelling results for extracting individual current harmonic

components of a practical nonlinear load
The simulated performance of the proposed method in extracting individual current
harmonic components of a three-phase full bridge thyristor rectifier is presented in this
section. MATLAB/SIMULINK software tools have been used for the modelling and
simulations. In this simulation example, the proposed harmonic detection method is
utilized to extract the fundamental, 5th, 7th, 11th, and 13th order harmonics of the load
current. A sampling frequency of 3 kHz and 30 number of samples (i.e., L =30) is
chosen for the proposed efficient least squares algorithm. The line voltage of the source
is maintained at 300V, 50Hz during the simulations. The MATLAB/SIMULINK model
for this case study is illustrated in Figure 6.11.
Figure 6.11 MATLAB/SIMULINK model of three-phase thyristor rectifier

load and the harmonic current extraction using the proposed
method.
160
The system is simulated for time varying loading conditions. Figure 6.12 shows phase-
a of the input current of the three-phase full bridge thyristor rectifier. As may be seen,
the current corresponds to three different load conditions: rectifier operates with zero
firing angle (i.e. D 0D ) from t = 0.02 to 0.08 sec.; zero firing angle (i.e. D 0D ) with
increased current from t = 0.08 to 0.14 sec.; and firing angle of 60 degrees (i.e. D 60D )
from t = 0.14 to 0.22 sec.
Figure 6.13 shows the extracted fundamental, 5th, 7th, 11th and 13th order harmonic
components using the proposed efficient least squares algorithm. In order to evaluate
the identification accuracy the current waveforms are reconstructed by summing the
extracted fundamental and harmonic components. Figure 6.14 compares the actual
current waveform to the reconstructed current waveforms of phase-a, -b and -c. This
figure demonstrates the tracking capability of the proposed least squares algorithm and
its accuracy in extracting harmonics of practical nonlinear load currents under dynamic
loading conditions.
Figure 6.12 The load current waveform of phase-a.

161
Figure 6.13 Extracted fundamental and individual harmonic components (i.e.

5th, 7th, 11th and 13th order harmonics) of phase – a current using
the proposed method.
Figure 6.14 Actual current waveform and the reconstructed waveform with the
proposed method of phase-a, -b and -c.
162
6.3.2 Modelling results for the selective harmonic compensation of active

power filter based on proposed harmonic extraction method
The active power filter based on the proposed harmonic extraction method described in
section 6.2.1 is modelled in MATLAB/SIMULINK. This model is shown in Figure
6.15. The parameters of the modelled system are listed in Table 6.1. The modelled
system was simulated for two cases: (i) with current controller phase lag correction and
(ii) without current controller phase lag correction.
Figure 6.15 MATLAB/SIMULINK model for the selective harmonic

compensation of active power filter based on the proposed
harmonic detection method.
163
Table 6.1 System parameters for modeled APF
Three-phase supply, line voltage: (Vs) 300V, 50Hz
Sampling frequency (fs) 6800 Hz
dc-link voltage (Vdc) 700 V
Filter inductance (LF) 6 mH
Filter resistance (RF) 0.078 :
Natural frequency dq control loop ( Zni ) 1257 rad/s
Natural frequency Vdc2 control loop ( Znv ) 31.42 rad/s
Converter Switching frequency ( f sw ) 6800 Hz
Figure 6.16 and Figure 6.17 show the performance of the APF with the proposed
individual harmonic detection method applied. Figure 6.16 corresponds to the APF that
does not utilize the individual phase angle compensation for the current controller phase
lags. Figure 6.17 shows improved performance with the current controller phase lag
compensation for the individual harmonic components. However, both cases show good
transient response for the sudden change in the nonlinear load current. This is due to the
fast dynamic response of the proposed harmonic extraction method.
164
Figure 6.16 Simulation results of the proposed APF without current controller
phase lag compensation; (a) load current; (b) filter current; (c)
source current; and (d) dc-link voltage.
Figure 6.17 Simulation results of the proposed APF with current controller
phase lag compensation; (a) load current; (b) filter current; (c)
source current; and (d) dc-link voltage.
165
Figure 6.18 and Figure 6.19 show the performance of the proposed method and for
comparison the DFT method when a step decrease in the load current occurs together
with +2 Hz supply frequency change at t = 0.3 sec. The transient time of the DFT
method is longer than the proposed method. The fundamental frequency change does
not have significant effected on the harmonic compensation with the proposed method.
In contrast, the DFT method shows poor compensation performance due to the
inaccuracies of harmonic detection of the DFT method caused by the fundamental
frequency change.
Figure 6.18 Performance of APF based on proposed method during step

decrease in load current together with fundamental frequency
change from 50 to 52 Hz: (top) load current; (middle) filter
current; and (bottom) source current.
166
Figure 6.19 Performance of APF based on DFT method during step decrease in
load current together with fundamental frequency change from 50
to 52 Hz: (top) load current; (middle) filter current; and (bottom)
source current.
6.4 Experimental Results of the Three-phase Active Power

Filter Based on Proposed Harmonic Detection Method
The above discussed three-phase active power filter, based on the proposed harmonic
detection method, is next implemented in the laboratory. The APF consists of a three-
phase IGBT converter and ac-side inductors. A three-phase controlled rectifier (i.e. full
bridge thyristor rectifier) is selected as the non-linear load which draws high levels of
harmonic currents. A variable voltage supply is used as the source for the load. Four
current sensors are used for measuring the load current and the APF current. Three
voltage sensors are used for measuring source voltage and dc-link voltage. An overview
of the experimental setup is illustrated in Figure 6.20. The proposed harmonic detection
method and the APF controller are implemented on the dSPACE DS1104 R&D board
167
hosted on a personal computer. A complete description of the experimental setup
including detailed drawings and specifications of the IGBT inverter is given in
Appendix C.
Three-phase
supply
voltage
Non-linear
load
3-PHASE
CONVERTER LF
+
Current
_ sensors
Current
Voltage sensors sensors
I Fa PC WITH dSPACE 1104
I Lb
CONTROL CARD
I Fb ILa
PWM Signals
Vdc
eSca
eSbc
eSab
Active Power Filter
Figure 6.20 Overview of experimental setup for APF system.
Figure 6.21 is a photograph of the experimental setup in the laboratory. The parameters
of the experimental APF are similar to those of the modelled APF system. These
parameters are given in Table 6.1.
168
Figure 6.21 Photograph of experimental setup for APF system.
6.4.1 Experimental results for harmonics extraction using the proposed

method
The first part of the experimental study is conducted in order to investigate the real-time
performance of the proposed method in detecting harmonics in the nonlinear load
current. Figure 6.22 shows the real-time extraction of fundamental, 5th, 7th,11th and 13th
order harmonic components in the input current of the full thyristor rectifier for phase-a.
The experimental results in Figure 6.22 correspond to zero degrees firing angle
(i.e. D 0D ) of the bridge rectifier.
169
Figure 6.22 Extracted fundamental, 5th, 7th, 11th and 13th –order harmonic
components in load current for phase –a.
The load current waveforms are reconstructed by summing the extracted fundamental
and harmonic components of individual phases in order to investigate the accuracy of
detection. Figure 6.23 compares the actual and the reconstructed load current
waveforms for phases -a, -b and -c. The reconstructed waveforms closely follow the
actual waveforms. A detection time of about 0.01 sec. is apparent in the beginning of
the identification.
In the next section the experimental results for the selective harmonic compensation of
the APF based on the proposed harmonic detection method will be presented.
170
Figure 6.23 Actual waveforms and the reconstructed waveforms with proposed
method for load current for phase - a , - b and - c .
6.4.2 Experimental results for the selective harmonic compensation of active

power filter using the proposed method
This section presents the experimental results for the selective harmonic compensation
of the APF based on the proposed method. The response of the APF is recorded when
the filtering command is issued at 0.035 sec. as shown in Figure 6.24 in order to
investigate the response time of the APF with the proposed selective harmonic
compensation. These results correspond to zero firing angle of the thyristor bridge
rectifier (i.e. D 0D ). As may be seen, the filtering starts quickly after issuing the
command at t = 0.035 sec and the dc-link voltage is unaffected during the transients.
Figure 6.25 and Figure 6.26 show the oscillograms for load current and source current
with the APF based on the proposed harmonic detection method for the firing angles of
zero and about 60 degrees respectively.
171
Figure 6.24 Response of APF with proposed selective harmonic compensation

when filtering enabled at t=0.035sec.: (Ch1) load current
waveform (6.5 A/div), (Ch2) active filter current (2 A/div), (Ch3)
source current waveform (6.5 A/div) and (Ch4) dc-link voltage
(140 V/div).
172
Figure 6.25 Oscillogram of (Ch1) load current (5 A/div) and (Ch2) source
current (5 A/div) for APF with proposed harmonic detection
method for D 0D of bridge rectifier load.
Figure 6.26 Oscillogram of (Ch1) load current (5 A/div) and (Ch2) source
current (5 A/div) for APF with proposed harmonic detection
method for about 60 degrees firing angle (i.e. D | 60D ) of bridge
rectifier load.
173
Figure 6.27 shows the harmonic constituents of the load current and source current for
the above case with the zero degree firing angle (i.e. corresponds to Figure 6.25). As
may be seen, the most significant 5th order harmonic in the source current is reduced by
81.72%. The other significant harmonics, 7th, 11th and 13th are reduced by 90.19%,
38.36% and 33.26% respectively. In Figure 6.28, the harmonic analysis for the second
case with firing angle of about 60 degrees (i.e. D | 60D ) also shows reductions in the
significant harmonics in the source current. Reductions of 89.5%, 91.08% and 15.9%
can be observed in the 5th 7th and 11th order harmonics respectively. However, there is
no significant reduction in the 13th order harmonic components.
Figure 6.27 Harmonic constituent as a percentage of the fundamental

component of load current and source current with zero firing
angle (i.e. D 0D ).
174
Figure 6.28 Harmonic constituent as a percentage of the fundamental

component of load current and source current with about 60
degrees firing angle (i.e. D | 60D ).
The dynamic performance of the APF with the proposed harmonic detection method is
investigated by applying a step increase from 9A to 15A in the dc-side load current of
the thyristor bridge. The firing angle is kept at zero degree (i.e., D 0D ). The results
obtained in this test are shown in Figure 6.29. This figure shows the experimental
results for the load current, filtered current, source current and dc-link voltage during
this test. The APF shows good response in the filtered source current during the step
increase. Furthermore, it shows good regulation in dc-link voltage during the step
increase of the load.
175
Figure 6.29 Dynamic response of APF with proposed selective harmonic

compensation: (Ch1) load current waveform (6.5 A/div), (Ch2)
active filter current (2 A/div), (Ch3) filtered source current
waveform (6.5 A/div) and (Ch4) dc-link voltage (140 V/div).
6.5 Summary
This chapter discussed an application of the proposed efficient least squares algorithm
in an active power filter (APF). The active power filter application discussed in this
chapter is based on a shunt connected voltage source converter which injects harmonic
currents into the power system in order to cancel out the harmonic currents that are
generated by the nonlinear load. The selective harmonic compensation strategy is
utilized for the APF due to its advantages over total harmonic compensation. The main
advantage is that the selective harmonic compensation allows correcting the delays of
the current controllers for individual harmonic components. The detailed modelling of
the three-phase voltage source converter that is used for the APF application is
presented. The APF consists of a voltage controller that regulates dc-link voltage and
synchronous reference frame dq axes current controllers to force desired harmonic
176
currents into the power system. The systematic design of these controllers based on the
mathematical model is discussed in detail. The harmonics of the nonlinear load current
are extracted using the proposed least squares algorithm and used for deriving the
commands for the current controllers. The phase lags (i.e. delays) of the designed
current controllers for each harmonic component are calculated and the proposed
method is used to introduce phase leads that cancel the current controller phase lags.
A power system that consists of a voltage source and a practical nonlinear load (i.e. full
bridge thyristor rectifier) is modelled in MATLB/SIMULINK. This power system
model is simulated with the proposed method to investigate the performance of the
harmonic detection technique. The APF is then added to the model and the performance
of the harmonic compensation is investigated using the simulation results. The
simulations for the APF are conducted with and without the current controller phase lag
correction to demonstrate the effects of the phase lags correction on the performance.
A laboratory experimental setup for the APF is built using an IGBT voltage source
converter in order to verify the performance. The proposed harmonic detection method
and the APF control algorithm are implemented on the dSPACE DS1104 R&D board.
The first part of the experimental study is conducted to investigate the real-time
operation of the proposed harmonic detection method. The results of this study confirm
the accuracy of the harmonic detection technique. Then, the APF is utilized to
compensate the load current harmonics. The experimental performance of the APF is
extensively investigated for various operating conditions.
In the next chapter, the proposed efficient least squares algorithm will be applied for
detecting power system voltage dips and unbalances for mitigation using a distribution
static synchronous compensator (DSTATCOM).

177
Chapter 7: Proposed Signal Processing System for Voltage Dip Detection in a Distribution Static
Synchronous Compensator Application
CHAPTER 7
PROPOSED SIGNAL PROCESSING SYSTEM

FOR VOLTAGE DIP DETECTION IN A
DISTRIBUTION STATIC SYNCHRONOUS
COMPENSATOR APPLICATION
7.1 Overview
This chapter presents the application of the proposed efficient least squares algorithm in
power supply voltage dip and unbalance detection for mitigation using a distribution
static synchronous compensator (D-STATCOM). The D-STATCOM systems usually
consists of a voltage source converter which dynamically injects a current of desired
amplitude, frequency and phase into the grid line in order to mitigate dips or unbalances
present at the point of common coupling (PCC). Both balanced and unbalanced dips
occur in power systems. In order to handle both balanced and unbalanced voltage dips
effectively, the control strategy that is based on separating the positive- and negative-
sequence and controlling them separately is preferred. In this chapter, application of the
proposed efficient least squares algorithm in detecting the positive- and negative-
sequence components and then using those sequence components for mitigating voltage
unbalances and dips will be discussed in detail. The remainder of this chapter is
organized as follows.
The next section presents a brief review of the D-STATCOM system. The principle of
estimation of the positive- and negative- sequence components using the proposed
178
efficient least squares algorithm is discussed in Section 7.3. In Section 7.4, the system
configuration of the D-STATCOM with the proposed dip detection method and its
control strategy are discussed. Section 7.5 presents the modelling and simulation results.
The performance of the proposed dip detection method and D-STATCOM are
investigated for various disturbance conditions in the Section 7.5. In Section 7.6,
experimental results are provided to demonstrate that the proposed efficient least square
algorithm is a suitable tool for detecting voltage dips and to verify the simulation results
for voltage mitigation. A summary of this chapter is presented in Section 7.7.
7.2 Review of Grid Unbalance and Voltage Dip Detection Methods

and the D-STATCOM System
7.2.1 Existing method of grid unbalance and voltage dip detection methods
x Grid unbalance detection methods
Grid unbalances are usually determined via symmetrical components of the supply
voltage. It is essential to accurately track the symmetrical component on-line, since
these components significantly change their values during an unbalanced disturbance.
Accurate identification of the symmetrical components is required not only for
mitigating grid unbalances, but also for other important applications such as the
protection of power systems and digital relays[66], [67].
Many techniques are available for measuring symmetrical components of an unbalanced
three-phase system. One existing technique is based on the Park transform coupled with
a notch filter which is widely used to track the symmetrical components. However,
these filters are characterized by low detection precision, since they are sensitive to any
change in the power supply frequency, and variation in the distribution system
179
parameters [68]. The FFT is proposed in [69] and [70] in order to overcome the
previous problem. However, its main drawback is that the accuracy of the results
depends on the width of the data window. Furthermore, the accuracy of the results
significantly deteriorates under dynamically changing unbalance conditions. The
Kalman Filter, which is a recursive estimator that can be used to process noisy
measurements, is introduced as a powerful tool in estimating the time varying
parameters of symmetrical components. Although the Kalman Filter gives a least square
optimal estimate [29] ,[71] the algorithm entails bulk calculations which require a large
memory capacity and a high-speed microprocessor for on-line implementation.
x Voltage dip detection methods
Different techniques have been used in the literature to detect and estimate the voltage
magnitude during a voltage dip event. The most common technique is to calculate the
root-mean-square (RMS) value of the voltage. The main drawback of this method is its
dependency on the window length. The RMS magnitude of the voltage can be
calculated over a window length of any number of cycles using either non-overlapping
or overlapping windows. The voltage dips cannot be immediately detected using the
RMS calculation. It is necessary that the new value of the voltage after the change be
entirely within the sampling window to obtain its correct magnitude. Thus, depending
on the instant when the voltage dip event begins, magnitude and duration of the dip, and
type of the window employed, the detected RMS magnitude of the voltage dip can be
very different [72], [73].
Fourier transform, wavelet analysis and Kalman filtering have been proposed in the
literature as alternative signal processing tools to monitor voltage events in power
systems. The Fourier transform technique can return information regarding the state of
180
system supply. The advantage of this method is that it can return magnitude and phase
of the fundamental and harmonics component of the supply voltage. However, it takes
at least one cycle of the fundamental when a dip has commenced before information
regarding the magnitude and phase angle can be assumed accurately [74].
The wavelet transform has proven to be a powerful processing tool in power systems,
where its time-frequency characteristics permit detecting and identifying different type
of power quality disturbances [75], [76]. The selection of the mother wavelet function
plays an important role in the detection of disturbances. Several studies have been made
on the performance of different mother wavelets to accurately detect and characterize
several parameters of voltage and current disturbance in power systems [77]. The RMS
voltage magnitude of the voltage during a dip can be obtained using the coefficients of
the wavelet analysis as proposed in [78]. In all of these cases, the wavelet analysis is
always applied over power quality records previously acquired in a power system. Until
now there are no references in the literature to the real-time applications of the wavelet
analysis to the detection and analysis of power quality disturbances. Another problem
associated with the use of the wavelet analysis is observed in the case of multiple-step
voltage dips. In such a case, the detailed coefficients of the high-frequency scale detect
the different steps of the voltage dip, making it difficult to ensure which one exactly
corresponds to the end of the voltage dip.
Kalman filters have also been used for detection of sudden changes in voltage
magnitude. The performance of the Kalman filter of different order to detect the
beginning of a voltage dip and to estimate the voltage magnitude during the dip can be
seen in [79], [80]. However, the practical application of the Kalman filter as a tool for
181
voltage dip detection has been limited by implementation difficulties as it needs high
computational complexity.
A voltage dip detection technique that utilizes the phase-locked- loop (PLL) to each
supply phase independently has been introduced in [81], [82]. This technique can be
combined with any other technique to detect the magnitude of the dip voltage. The PLL
technique does not give good results if the voltage dip are associated with a phase angle
jump such as in unbalance voltage dip [83].
7.2.2 Brief review of the D-STATCOM system
Recently, the distribution static synchronous compensator (D-STATCOM) has been
introduced to distribution networks to manage the system reactive power and regulate
the voltage at the distribution buses. A D-STATCOM usually consists of a shunt
connected voltage source converter (VSC) [84]. The benefits of using a VSC are
sinusoidal currents, high current bandwidth, controllable reactive power to regulate bus-
voltage level and to minimize the resonances between the grid and the converter. A
system with these characteristics can be used to inject a controllable current into the
grid. By injecting a current into the point of common coupling (PCC), a shunt-
connected VSC can boost the voltages at that point during a voltage dip. Even though
the theory, control and modelling of conventional static compensator (STATCOM) have
been broadly discussed in the literature, more preference is given to the D-STATCOM
due to its simple connection requirement [85], [86]. Furthermore, an unbalance
correction can also be added to the functions of the D-STATCOM [87].
The extraction and tracking technique of voltage dip is the core of the D-STATCOM
mitigating control strategy. In order to obtain the required information to control the
182
D-STATCOM further processing during voltage unbalance or dip is required and the
processed information is required to be updated as fast as possible. Moreover, the
choice of techniques for the voltage dip detection is highly dependent on the real-time
implementation, the available computational hardware, and the amount of
computational effort.
The typical standard information tracking or detection methods such as the Fourier
transform or the practical digital implementation of it, the discrete Fourier transform
(DFT) or the Fast Fourier transforms (FFT) [88] and the phase-locked-loop (PLL) [53]
are generally used in D-STATCOM systems. The main drawback of the DFT and the
FFT methods is less efficient in tracking the signal dynamics. The DFT method is not a
fast technique since it needs at least one cycle of the fundamental when dip has
commenced before information regarding the magnitude and phase can be determined
accurately. The PLL also has the same problem that it is slow in returning the
information of the voltage disturbances.
In the next section the voltage dip detection and unbalances detection based on the
proposed efficient least squares algorithm for using in D-STATCOM applications is
introduced.
7.3 Voltage Dip Detection Based on Proposed Efficient LS Method
Voltage dips, both balanced and unbalanced, in three-phase system can be effectively
identified for mitigation purposes using instantaneous symmetrical components. The
proposed efficient least squares method as described in Section 5.3.2 is capable of
identifying the instantaneous symmetrical components of the fundamental frequency
accurately even though the point of common coupling voltage is strongly corrupted by
183
the voltage harmonics. Also, it fulfils the specific requirements of the fast transient
response, accuracy and robustness in order to ensure the satisfactory performance of the
mitigation system.
Identification of the instantaneous positive- and negative-sequence components of the
fundamental is usually adequate to mitigate both the balanced and unbalanced dips. The
size of the required constant matrix in this case is 2 u L , since only the fundamental
component is required to be identified (i.e., K 1 ). Figure 7.1 illustrates the
identification of the instantaneous cosine and sine components of the fundamental
voltage using the proposed method.
Constant Matrix
Cc
ªvacos º
>va (n), va (n 1),..., va (n L 1)@

T
u 1
« sin »
¬ va1 ¼
ªvbcos º
>vb (n), vb (n 1),..., vb (n L 1)@

T
u 1
« sin »
¬ vb1 ¼
ªvccos º
T
>vc (n), vc (n 1),..., vc (n L 1)@
u 1
« sin »
¬ vc1 ¼
Figure 7.1 Estimation of instantaneous cosine and sine components of

fundamental voltage component using the proposed efficient least
squares algorithm.
184
As described in Chapter 5, the instantaneous symmetrical components can be obtained
from the outputs of the proposed efficient least squares algorithm (i.e. the instantaneous
cosine and sine terms).
By taking the fundamental component (i.e. i 1 ) of (5.30) and (5.31), the instantaneous
positive- and negative- sequence components of phase-a can be written in complex
notation as follows:
cos 1
sin
va1 v
a1
j va1
3

va1 D vb1 D 2 vc1
1 § cos vbcos
1 3vbsin
1 vccos
1 3vcsin
1
·
v
¨¨ a1 ¸ (7.1)
3© 2 2 2 2 ¸¹
1§ 3vbcos v sin 3vccos v sin ·

j ¨ vasin1 1
b1 1
c1 ¸
3 ¨© 2 2 2 2 ¸¹
cos 1
sin
va1 v
a1
j va1
3

va1 D 2vb1 D vc1
1 § cos vbcos 3vbsin v cos 3vcsin ·
¨¨ va1
1
1
c1 1
¸ (7.2)
3© 2 2 2 2 ¸¹
1§ 3vbcos v sin 3vccos v sin ·

j ¨ vasin1 1
b1 1
c1 ¸
3 ¨© 2 2 2 2 ¸¹
The positive-sequence component of phase- b and - c can be obtained from (7.1) as
shown below.
cos sin
vb1 v
b1
j vb1 D 2va1
1 § vacos
1 3vasin1 cos vccos
1 3vcsin
1
·
¨¨ vb1 ¸¸ (7.3)
3© 2 2 2 2 ¹
1§ 3vacos vsin 3vccos vcsin ·
j ¨¨ 1
a1 vbsin 1
1
¸
2 ¸¹
1
3© 2 2 2
185
cos sin
vc1 v
c1
j vc1 D vc1
1 § vacos 3vasin1 vbcos 3vbsin ·
¨¨ 1
1
1
vccos
1 ¸ (7.4)
3© 2 2 2 2 ¸
¹
1 § 3vacos vasin1 3vbcos vbsin ·
j ¨¨ 1
1
1 vcsin
1 ¸
3© 2 2 2 2 ¸
¹
As described in Section 5.3.2, the real parts of va1 , vb1 and vc1 represent the
instantaneous positive-sequence components corresponding to phases- a ,- b and - c . The
real parts of (7.1), (7.3) and (7.4) can be represented in matrix form as follows:
cos
ª 1 3 1 3 º ªva1 º
« 1 0 » « sin »
« 2 2 2 2 » « va1 »
ª va º
1« 1 3 » «vb1 »
cos
« » 3 1
« vb » « 1 0 » « » (7.5)
3« 2 2 2 2 » « vbsin1 »
« vc »
¬ ¼ « 1 3 1 3 » «v cos »
« 1 0 » « csin 1
»
¬« 2 2 2 2 ¼» ¬« vc1 ¼»
where va , vb and vc are the fundamental instantaneous positive- sequence components
of phases - a ,- b and - c respectively.
Similarly, the negative- sequence components of phases - b and - c can be obtained by
considering the negative rotation as follows:
cos sin
vb1 v
b1
j vb1 D va1 (7.6)
cos sin
vc1 v
c1
j vc1 D 2va1 (7.7)
186
The matrix equation which corresponds to negative-sequence can be obtained by
substituting (7.2) into (7.6) and (7.7), and then taking the real part of the complex
numbers, as given below.
cos
ª 1 3 1 3 º ªva1 º
« 1 0 » « sin »
« 2 2 2 2 » « va1 »
ª va º
1« 1 3 » «vb1 »
cos
« » 3 1
« vb » « 1 0 » « » (7.8)
3« 2 2 2 2 » « vbsin1 »
« vc »
¬ ¼ « 1 3 1 3 » «v cos »
« 1 0 » « csin 1
»
¬« 2 2 2 2 ¼» ¬« vc1 ¼»
where va , vb and vc are the fundamental instantaneous negative-sequence components
of phases - a ,- b and - c respectively
As may be seen from the (7.5) and (7.8), the instantaneous positive- and negative-
sequence components can be easily obtained using the instantaneous cosine and sine
terms of the proposed efficient least squares algorithm. The D-STATCOM application
that is discussed in the next section uses synchronously rotating reference frame
controllers (i.e. dq controllers). Therefore, transformation of the instantaneous
positive- and negative- sequence voltage components into the dq axes is required. The
transformation of the positive- and negative- sequence components into the
dq reference frame is given in (7.9) and (7.10) respectively.
1
ªvd º
ª
2«
sin Zt
2

sin Zt 3 cos Zt 12 sin Zt º

3 cos Zt » ªva º
« »
« » « » «vb » (7.9)
¬«vq ¼» 3«
cos Zt
1

cos Zt 3 sin Zt 12 cos Zt
3 sin Zt » «¬vc »¼
¬« 2 ¼»
187
ª 1 1
ªvd º 2 « sin Zt
2

sin Zt 3 cos Zt 2

sin Zt 3 cos Zt º» «ªv »º

a

« » « » «v »
b (7.10)
«¬vq »¼ 3« 1 1
«¬
cos Zt
2
cos Zt 3 sin Zt
2
cos Zt 3 sin Zt »¼» «¬v »¼

c
where
vd and vq are the fundamental instantaneous dq components of the positive-sequence
voltage, and
vd and vq are the fundamental instantaneous dq components of the negative-
sequence voltage.
The overview of the complete dip detection method based on the proposed efficient
least squares algorithm is illustrated in Figure 7.2. As described earlier in this section,
the negative- and positive- sequence components at fundamental frequency are obtained
from the instantaneous cosine and sine terms. Then, the dq transformation is carried
out to obtain the d and q components for each sequence. The advantages of
obtaining the dq components via the symmetrical components instead of the direct
dq transformation will be discussed in Section 7.4.2.
188
cos(Z1t ) sin(Z1t )
Positive-sequence dq vd
vSa (t ) estimator
Proposed (Eq. 7.5) vq
abc
Efficient
vSb (t ) Least Squares
Algorithm
vSc (t ) Negative-sequence dq vd
estimator
(Eq. 7.8) abc vq
cos(Z1t )sin(Z1t )
Figure 7.2 Block diagram of voltage dip detection method based on proposed
efficient least squares algorithm.
7.4 The D-STATCOM System with Proposed Voltage Dip Detection

Method
7.4.1 System configuration
The configuration of the D-STATCOM with the proposed voltage dip detection method
is illustrated in Figure 7.3. The D-STATCOM consists of a three-phase voltage source
converter (VSC), a dc-side capacitor Cdc with its leakage resistance Rdc , and an
inductance LF on the ac-side of the converter. The resistance RF represents the cable
resistance of the ac-side of the converter. A shunt filter capacitor with capacitance CF
and an inductance Ltr that represents transformer leakage inductance are added to the ac-
side of the voltage source converter that forms a LCL filter. This filter helps in
effectively filtering out the switching ripple in the output voltage waveform.
189
Grid Line impedance

ea PCC a
eb Rline Lline
b
LOAD
ec c
vsab vsab vsab
vsa vsb vsc Ltr
CF
sin(Zt )
abc Voltage dip
Z PLL cos(Zt ) detection based
dq
on proposed
method
vsd (Figure 7.2)
LF
vd , vq
Reactive
RF
Power vd , vq
Controller
iFa
Rdc
*
iCq
*
iCd iFb
iFc Cdc
Vdc* 2
*
Vdc2 ++ iFd id Switching signals
Vdc2 Controller + controller PWM
-
iFd Geneartion
iFa , iFb , iFc ma mb mc
abc sin(Zt )
cos(Zt )
abc dq
sin(Zt ),cos(Zt ) dq
iFq md mq
i* - ud Decoupling
Cq
+ iq
controller uq
iFd iFq Z vsd
Figure 7.3 Overall schematic diagram of D-STATCOM with proposed

voltage dip detection method.
The grid is represented by using an ideal voltage source and impedance. This impedance
consists of an inductance Lline and a resistance Rline which characterizes the transformer
and power line respectively. The analysis and design of the D-STATCOM controller are
conducted in the rotating reference frame which is synchronized to the voltage vector.
190
As explained earlier in Section 7.1, the D-STATCOM mitigates the voltage dips by
dynamically injecting a current of desired amplitude and phase angle into the grid line.
A schematic diagram of the D-STATCOM with the proposed voltage dip detection
method is illustrated in Figure 7.3. The two inner dq current regulators in Figure 7.3
* *
force the converter currents iFd and iFq to follow the command currents iFd and iFq
*
respectively. The command iFd to the d axis current loop is obtained by summing the
dc-link Vdc2 -controller output and d axis component of the reactive power controller
*
output iCd . The command iFq is obtained from the q axis component of the reactive
power controller output iCq . The purpose of the outer loop Vdc2 -controller is to regulate
the dc-link voltage to a required level. The modulation signals ma , mb and mc for the
PWM generator are derived from the output of the current controllers. The decoupling
terms are added to the output of the current controllers in order to remove the coupling
and are then transformed into the abc stationary reference frame to obtain the
modulation signal for the VSC. The sin Z1t and cos Z1t terms required for the
transformation between the abc and dq reference fames are obtained via a phase
locked loop (PLL) which is synchronized to the fundamental component of the voltage.
All of the above are illustrated in Figure 7.3.
The inner loop dq current controller and the outer loop dc-link voltage controller
design for this D-STATCOM are similar to the controller design presented in the
Chapter 6, since they have the same structure and serve the same purpose. Nevertheless,
the dq current controllers in this application regulates only the fundamental frequency
components which are DC in the synchronous reference frame during the balanced dips
and 100Hz during the unbalance dips, and hence do not need to have a high bandwidth
191
* *
as was the case with the APF. The command currents iCd and iCq which are needed to
be injected into the power grid in order to compensate the voltage dips and unbalances,
are obtained via a reactive power controller. The structure and the purpose of the
reactive power controller will be discussed in the next section.
7.4.2 Positive- and negative-synchronous reference frame controllers

(reactive power controller)
In the case of balanced three-phase voltages, the direct transformation of abc voltages
into the dq reference frame will result in dc-quantities. Hence, the D-STATCOM can
use a conventional PI-controller to control the injected reactive currents. However, if the
grid voltage or the load voltages are unbalanced, a ripple of double the grid frequency
will occur in the dq reference frame. In the case of unbalanced three-phase voltages,
breaking the voltage signals into positive- and negative-sequence components and then
transforming into the dq synchronous reference frames (SRF) results in dc-quantities.
These dq feedback signals allow flexibility in reactive power control. Figure 7.5 and
Figure 7.6 show simulation results for the dq components estimation for the three-
phase voltages given in Figure 7.4. The simulated three-phase voltages are balanced
until t = 0.2 sec. and then have an unbalance of 10% from t = 0.2 to 0.4 sec. As may be
seen in Figure 7.5, the direct dq components are dc during the condition of balanced
voltage and become ac for the case of unbalanced voltage, whereas, the
dq components of the positive- and negative- sequences are dc for both the balanced
and unbalanced three-phase voltages as shown in Figure 7.6.
192
Figure 7.4 Simulated three-phase unbalanced grid voltages.
Figure 7.5 Simulated direct dq components for three-phase voltages given

in Figure 7.4.
193
Figure 7.6 Simulated positive- and negative- sequence dq components for

three-phase voltages given in Figure 7.4. Top: positive-sequence in
positive SRF; bottom: negative-sequence in negative SRF.
As may be seen in the top and bottom graphs of Figure 7.6, that the transformation of
the balanced three-phase voltages with unity amplitude into positive- and negative-
sequence dq components results in dc-quantities with the following values.
vd 1 pu vq 0 pu
(7.11)
vd 0 pu vq 0 pu
(7.12)
Note from Figure 7.6 that under unbalanced conditions (7.12) no longer holds zero for
vq which becomes 0.05 p.u for the case considered. Similarly, balanced and unbalance
voltage dips do not satisfy one or more of the conditions given in (7.11) and (7.12).
Thus, the voltage dips (i.e. both the balanced and unbalanced) and grid unbalance in
power systems can be corrected by regulating the positive- and negative-sequence dq
194
voltage components to the values corresponding to the balanced three-phase values
given in (7.11) and (7.12). Three controllers are utilized to regulate these dq
components of the positive- and negative- sequence voltages as illustrated in Figure 7.7.
Vd -Controller icd* 0

abc
Vd* 1 pu+ K IV icq dq
K PV d
- d
s
vd sin Zt cos Zt
icd*
Vd -Controller + dq
* abc icq*
V *
0pu + K IV icq
d +
K PV d
- d
s
sin Zt cos Zt
abc
v d Vq -Controller dq
Vq* 0 pu K IV icd*
+ K PV q
- q
s sin Zt cos Zt

v q
Figure 7.7 Detailed block diagram of reactive power controllers for D-

STATCOM given in Figure 7.3.
As shown in Figure 7.7, vd is regulated to 1.0 pu via iq* in the positive-sequence
synchronous reference frame. vd and vq are regulated to zero via the iq* and id* in the
negative-sequence synchronous reference frame. These dq - current components in the
positive- and negative- synchronous reference frames are converted into the abc axes
components and then added together to obtain the three-phase currents that need to be
injected to the line in order to compensate for the voltage dips and unbalances. These
abc current commands are then transformed into the positive- sequence synchronous
* *
reference frame dq components to generate the current commands (i.e. iCd and iCq )
for the current controllers. All of these quantities are indicated in Figure 7.7.
195
7.5 Modelling and Simulation Results of the D-STATCOM for

Voltage Dip Mitigation
7.5.1 D-STATCOM system modelling
In this section, the simulation results for the D-STATCOM with the proposed voltage
dip detection method will be presented. The configuration of the D-STATCOM system
with the proposed efficient least squares algorithm based voltage dip detection that is
shown in Figure 7.3 is modelled using the MATLAB/SIMULINK software package.
The model utilizes SIMULINK control and SimPower blocks and is shown in Figure
7.8 and Figure 7.9. Figure 7.9 shows the model of the D-STATCOM controller. The
model reproduces a prototype D-STATCOM system based on a shunt-connected VSC
that has been built in the Electrical Energy Systems Laboratory at the Department of
Electric Power Engineering of the University of New South Wales. The parameters of
the prototype D-STATCOM system are listed in Table 7.1.
The modelling and simulations of the D-STATCOM have been carried out for three
different cases: unbalanced grid voltage, balanced voltage dip and unbalanced voltage
dip.
196
Figure 7.8 MATLAB/SIMULINK model for D-STATCOM system with

proposed dip detection.
Figure 7.9 MATLAB/SIMULINK model of D-STATCOM controller.
197
Table 7.1 Design specifications and circuit parameters of the proposed D-STATCOM.
210V=1 pu,
Phase-voltage of three-phase supply: (Vs) (rms)
50Hz
Sampling frequency of D-STATCOM control algorithm ( f s ) 8000 Hz
Converter switching frequency ( f sw ) 8000 Hz
dc-link voltage (Vdc) 700 V
Line inductance ( Lline ) 11 mH
ac-side inductance ( LF ) 6 mH
Filter capacitance ( CF ) 8 PF
Transformer leakage inductance ( Ltr ) 2 mH
Natural frequency of dq- current control loops ( Zni ) 942.47 rad/s
Natural frequency of Vdc2 -control loop ( Znv ) 15.7 rad/s
Proportional gain of the positive-SRF d voltage controller 0.3
Integral gain of the positive-SRF d voltage controller 20
Proportional gain of the positive-SRF dq voltage controllers 0.12
Integral gain of the positive-SRF dq voltage controllers 10
Sampling frequency of the proposed efficient LS algorithm 2000 Hz
Number of samples of the proposed efficient LS algorithm ( L ) 20
198
7.5.2 Simulation results for unbalanced grid voltage mitigation
The D-STATCOM system has been tested for the performance in mitigation of the
unbalanced grid voltages. The simulated phase-voltages of the grid are shown in Figure
7.10. As may be seen, the amplitude of the phase voltage is 10% unbalanced from t =
0.05 to 0.35 sec. time period. The sequence components of the grid voltage is extracted
using the proposed efficient least squares method and are then transformed into the
corresponding SRFs in order to obtain the dq components. Figure 7.11 shows those
positive- and negative-sequence dq components before the mitigation. The proposed
voltage unbalance detection method is capable of tracking symmetrical components
accurately within less than half a cycle of a fundamental period. Figure 7.12 shows the
compensated phase voltages and the voltage waveform error for an unbalanced phase.
As may be seen in this figure, the grid voltages at the point of common coupling (PCC)
are clearly constant during the occurrence of the unbalanced grid voltage and the
maximum transient voltage error is about 2 %. Also, the steady-state errors during the
unbalanced compensation are negligible. Figure 7.13 shows the dq components of the
positive- and negative-sequence voltages during the compensation. The top plot of
Figure 7.14 shows the three-phase currents that are injected into the power grid in order
to mitigate the voltage unbalance. The second and third plots of this figure show the
injected currents in the positive-sequence SRF dq axes. It can be noted that there is a
ripple of 100 Hz contained in the dq current components during the mitigation of
voltage unbalance. As can be seen in the bottom plot, the dc-link voltage is almost
unaffected during the mitigation.
199
Figure 7.10 Simulated three-phase grid voltages.
Figure 7.11 Simulated grid voltages in the SRF dq components, before

compensation. Top: positive- sequence in positive SRF; bottom:
negative-sequence in negative SRF.
200
Figure 7.12 Compensated voltage waveforms and phase voltage waveform

error in the case of unbalance grid voltage.
Figure 7.13 Compensated grid voltages in the SRF dq components in the

case of unbalanced grid voltage. Top: positive- sequence in
positive SRF; bottom: negative-sequence in negative SRF.
201
voltage compensation.
7.5.3 Simulation results for balanced voltage dip mitigation
The grid voltage was affected by a 25% symmetrical voltage dip as shown in Figure
7.15. Figure 7.16 shows the SRF dq components extracted using the proposed voltage
dip detection method. As may be seen from Figure 7.16, the amplitude of the
d component of the positive-sequence is reduced by 0.25 pu which corresponds to the
voltage dip, and the q component of positive-sequence is zero. In the negative-
sequence dq components, the short duration transients are observed at the beginning
and at the end of the dip which is caused due to the rapid variation of the grid voltage.
The steady-state amplitudes of the both dq components of the negative-sequence
remain zero, since the voltage dip is balanced.
Figure 7.17 shows that the steady-state voltage at the PCC is nearly constant during the
occurrence of the voltage dip. The steady-state waveform error of the voltage shown in
202
Figure 7.17 remains almost zero, while the maximum transient error of only 0.15 pu
occurs at the start and the end of the dip for short time only. Figure 7.18 shows the
dq components of the positive- and negative-sequence voltages.
The top plot of Figure 7.19 shows the three-phase currents injected into the power
system in order to mitigate the balanced voltage dip. The second and third plots of this
figure show the injected currents in the positive-sequence SRF dq axes. These
components have dc values during the steady-state period of the dip, since the dip is
balanced. As may be seen in the bottom plot, the dc-link voltage is almost unaffected
during the voltage dip mitigation.
Figure 7.15 Simulated three-phase grid voltage with 25% balanced voltage dip.
203

balanced voltage dip. Top: positive- sequence in positive SRF;
bottom: negative-sequence in negative SRF.
Figure 7.17 Compensated voltage waveforms at PCC and voltage waveform

error in case of 25% balance voltage dip.
204
Figure 7.18 Compensated grid voltages in SRF dq components in case of

balanced voltage dip. Top: positive-sequence in positive SRF;
bottom: negative-sequence in negative SRF.
Figure 7.19 Simulated current injections and dc-link voltage during balanced
voltage dip compensation.
205
7.5.4 Simulation results for unbalanced voltage dip mitigation
The system has been tested for mitigation of the unbalanced voltage dips. Three-phase
grid voltages with 25% voltage dip and 20% amplitude unbalance are shown in Figure
7.20 and the corresponding positive- and negative-sequence dq components are
shown in Figure 7.21. As shown in Figure 7.21, the amplitude of the positive- sequence
d component decreases to 0.75 pu and the q component remains zero. In negative-
sequence, the d and q components have amplitudes of 0.08 pu and -0.04 pu
respectively during the unbalanced voltage dip. Figure 7.22 shows the voltage
waveform after compensation by the D-STATCOM with the proposed dip detection
method in the case of unbalanced voltage dip. As expected, the PCC voltage is constant
at 1 pu before, during and after the voltage dip, apart from the transient at the beginning
and at the end of the voltage dip. The error in the voltage waveform is also shown in
Figure 7.22. As may be seen, the error is very small in the steady-state and the transient
errors of less than 0.2 pu appear for short durations. Figure 7.23 shows the positive- and
negative-sequence dq components of the voltage during the unbalanced dip
compensation.
The top plot of Figure 7.24 shows the three-phase currents that are injected into the
power grid in order to mitigate the unbalanced voltage dip. The second and third plots
of this figure show the injected currents in positive-sequence SRF dq axes
respectively. As may be seen in the bottom plot, the dc link voltage is almost unaffected
during the voltage dip mitigation.
206
Figure 7.20 Simulated three-phase grid voltages during unbalanced voltage

dip.

voltage dip with 20% amplitude unbalance. Top: positive-
sequence in positive SRF; bottom: negative-sequence in negative
SRF.
207
Figure 7.22 Compensated voltage waveforms at PCC and voltage waveform

error in the case of unbalance voltage dip.
Figure 7.23 Compensated grid voltages in SRF dq components in the case of

25% voltage dip with 20% amplitude unbalance. Top: positive-
sequence in positive SRF and bottom: negative-sequence in
negative SRF.
208
voltage dip compensation.
7.6 Experimental Results for the D-STATCOM with the Proposed

Voltage Dip Detection
A prototype D-STATCOM system with the proposed efficient least squares algorithm
based voltage dip detection has been set up in the laboratory. The overview of this setup
is given in the block diagram shown in Figure 7.25. The same three-phase IGBT voltage
source converter that has been used for the APF is utilized in the experimental setup. As
may be seen in the Figure 7.25, the experimental setup consists of an additional resistive
load bank (i.e. load bank-A) and a switch. The arrangement is used for generating
voltage dips and unbalances that are required for the experimental studies. The load
current drawn into the load bank-A causes voltage drop in the line inductance and
thereby generates voltage dip in the PCC. The system parameters of the prototype D-
STACOM are similar to the modeled parameters which are given in Table 7.1. The
proposed efficient least square algorithm based dip detection method and the control
209
algorithm required for the D-STATCOM system have been implemented on the
dSPACE 1104 R&D board. A photograph of the experimental set-up for the D-
STATCOM is shown in Figure 7.26. Further details of the experimental setup are given
in Appendix-C.
Switch
Load
bank-A
Variac
Lline
3-Phase AC Supply
PCC
Load
Ltr
CF
IGBT INVERTER
LF
+
-
Cdc
Voltage
sensors
Current
PWM Signals
sensors
Vdc PC WITH dSPACE 1104
vab
R&D BOARD iFb
vbc
iFa
vca
D-STATCOM
Figure 7.25 Overview of the experimental setup of the D-STATCOM system.
210
Figure 7.26 A photograph of the experimental setup of the D-STATCOM.
7.6.1 Experimental results for unbalanced grid voltage mitigation
The D-STATCOM system is tested for the mitigation of unbalanced grid voltages. The
unbalanced grid voltage in the experimental setup is generated by applying an
unbalanced load at the load bank-A. Figure 7.27 shows the unbalanced phase voltages
of phases-a, -b and -c. As may be seen, the voltages of phase -a and -b are reduced by
about 15% and 7% respectively. Figure 7.28 shows the steady-state voltages of the grid
after the compensation using the D-STATCOM with the proposed method. The
unbalances are not apparent in the compensated voltages.
211
Figure 7.27 Unbalanced phase voltages of phases-a, -b and -c (100 V/div).
Figure 7.28 Compensated voltage of phases-a, -b and -c using D-STATCOM

with proposed method (100 V/div).
Figure 7.29 shows the dynamic response of the D-STATCOM during the grid unbalance
mitigation. The unbalance mitigation is enabled at t = 0.06 sec. Figure 7.30 shows the
corresponding d component of the positive-sequence voltage in the positive SRF (i.e.
vd ) and the dq components of the negative-sequence voltage in the negative SRF (i.e.
212
vd and vq ). During the unbalanced period, the vd , vd and vq have values of about 0.9
p.u., 0.1p.u. and 0.03 p.u. respectively (1.0 p.u. = 210 V). As may be seen, the values of
vd , vd and vq are regulated to 1.0 p.u., 0 p.u. and 0 p.u. by the reactive power
controllers when the unbalance mitigation is enabled. The vd shows fast transient
response. The vd and vq have slower transient response compared to that of the vd .
Figure 7.29 Dynamic response when unbalance mitigation started at t = 0.06

sec.: voltage of phases- a, -b and -c (100 V/div).
213
Figure 7.30 Dynamic response when unbalance mitigation started at t = 0.06

sec.: (Ch1) d component of positive-sequence voltage in positive
SRF (34.67 V/div); (Ch3) d components of negative-sequence
voltage in negative SRF (8.67 V/div); (Ch4) q components of
negative-sequence voltage in negative SRF (8.67 V/div).
7.6.2 Experimental results for balanced voltage dip mitigation
A second set of experiments is carried out when the grid voltage is affected by a
balanced voltage dip. The balanced voltage dip is generated by applying the same load
to each phase of the resistive load bank-A shown in Figure 7.25. The line inductance
Lline causes the same voltage drop in each phase due to this balanced load. The resistive
load bank-A is switched on at t = 0.06 sec. The generated balanced voltage dip is shown
in Figure 7.31. The magnitude of this voltage dip is about 17%. Figure 7.32 shows the
compensated voltage waveforms of phases-a, -b, and -c using the D-STATCOM with
the proposed voltage dip detection method. Figure 7.33 shows the corresponding
d component of the positive-sequence voltage in the positive SRF (i.e. vd ) and the
dq components of the negative-sequence voltage in the negative SRF (i.e. vd and vq ).
214
The value of vd is regulated to 1.0 p.u. after a short transient to result in balanced three-
phase voltages. The values of vd and vq are zero for any balanced three-phase voltages
including balanced dips. As may be seen in Figure 7.33, vd and vq have settled back to
zero after the transient disturbance. Figure 7.34 shows the compensated steady-state
voltage waveforms of the phases-a, -b and -c during the 17% balanced voltage dip.
Figure 7.31 Balanced voltage dip of 17% occurred of at t = 0.06 sec.

(100 V/div).
215
Figure 7.32 Compensated voltage waveforms of phases-a, -b, and -c at PCC in

case of 17% balanced voltage dip at t = 0.06 sec. (100 V/div).
Figure 7.33 Dynamic response of balanced voltage dip compensation when dip
started at t = 0.06 sec.: (Ch1) d component of positive-sequence
voltage in positive SRF (34.67 V/div); (Ch3) d components of
negative-sequence voltage in negative SRF (8.67 V/div); (Ch4)
q components of negative-sequence voltage in negative SRF
(8.67 V/div).
216
Figure 7.34 Compensated steady-state voltage waveforms of phases-a, -b and -

c at PCC during 17% balanced voltage dip (100 V/div).
7.6.3 Experimental results for unbalanced voltage dip mitigation
This section presents the results for the unbalanced dip mitigation using the D-
STATCOM. The unbalanced voltage dip for this experiment is generated by switching
on an unbalanced load in the resistive load bank-A shown in Figure 7.25. The switch is
turned on at t =0.06 sec. Figure 7.35 shows the generated unbalanced voltage dip using
the experimental setup. As may be seen, the voltages of the phases-a, -b and -c are
reduced by 28%, 15% and 8% respectively. Figure 7.36 shows the detected vd , vd and
vq components corresponding to the unbalanced voltage dip given in Figure 7.35. The
unbalance voltage dip is apparent in all the vd , vd and vq components. The small
positive value of vd before the dip started is due to the slight unbalance in the original
voltage waveform. Figure 7.37 shows the compensated voltage waveforms of phases -a,
-b and –c using the D-STATCOM with the proposed voltage dip detection method.
217
Figure 7.38 shows the vd , vd and vq components corresponding to the compensated
waveform given in Figure 7.37.
Figure 7.35 Voltages of phases-a, -b and -c for unbalanced dip occurring at t =

0.06 sec. (100 V/div).
218
Figure 7.36 Detected dq symmetrical components for unbalanced voltage dip

started at t = 0.06 sec.: (Ch1) d component of positive-sequence
voltage in positive SRF (34.67 V/div); (Ch3) d components of
negative-sequence voltage in negative SRF (8.67 V/div); (Ch4)
q components of negative-sequence voltage in negative SRF
(8.67 V/div).
Figure 7.37 Compensated voltage waveforms of phases-a, -b, and -c at PCC in

case of unbalanced voltage dip at t = 0.06 sec. (100 V/div).
219
Figure 7.38 Dynamic response of unbalanced voltage dip compensation when

dip started at t = 0.06 sec.: (Ch1) d component of positive-
sequence voltage in positive SRF (34.67 V/div); (Ch3)
d components of negative-sequence voltage in negative SRF
(8.67 V/div); (Ch4) q components of negative-sequence voltage
in negative SRF (8.67 V/div).
7.7 Summary
This chapter has shown that the proposed efficient least squares algorithm can be used
for voltage dip and unbalanced mitigation in the D-STATCOM. The discussed D-
STATCOM consists of a voltage source converter which injects reactive current into the
grid in order to mitigate the voltage dips and unbalances. The D-STATCOM consists of
positive- and negative-synchronous reference frame controllers (i.e. reactive power
controller) that regulate the dq components of the positive and negative voltage to
result in balanced three-phase voltages at the point of common coupling. The dc-link
voltage is regulated to a set value via the d axis current. The synchronous reference
frame dq axes current controllers are utilized to force the currents into the grid.
220
The proposed efficient least squares algorithm outputs the instantaneous cosine and sine
terms of the fundamental component. In the voltage dip and unbalance detection
methods, the size of the constant matrix (i.e. Cc ) required is only 2 u L , since only the
fundamental component needs to be identified (i.e. K 1 ). The instantaneous positive-
and negative- sequence components of the voltages are obtained using the instantaneous
cosine and sine terms of the fundamental components of phases -a, -b and -c. The
identified positive- and negative-sequence components are then represented in the
positive- and negative- synchronous reference frames respectively (i.e. vd , vq , vd and
vq ). These voltage components are regulated to set values that correspond to the three-
phase balance voltages in order to mitigate the voltage dips and unbalances. The
proposed voltage dip and unbalance detection method allows for extraction of these
sequence component within a half fundamental cycle. In addition the proposed method
is capable of identifying the voltage dips and unbalances accurately even though the
point of common coupling is strongly corrupted by the voltage harmonics.
A prototype D-STATCOM system is modelled and simulated using
MATLB/SIMULINK. The modelled system is tested for identification and mitigation of
the grid voltage unbalances, balanced and unbalanced voltage dips. Extensive
simulation studies showed that the proposed voltage unbalance and dip detection
method allows fast and accurate extraction of information that is required for successful
operation of the D-STATCOM system.
A laboratory experimental setup for the prototype D-STATCOM is built using an IGBT
voltage source converter in order to verify the performance. The proposed voltage dip
detection method and the D-STATCOM control algorithm are implemented on the
221
dSPACE 1104 R&D board. The prototype D-STATCOM system is tested for the
mitigation of grid unbalance, balanced and unbalanced voltage dips. The experimental
results verified the successful real-time application of the proposed voltage unbalance
and dip detection method in extracting information for the D-STATCOM application.
222
Chapter8: Conclusions
CHAPTER 8
CONCLUSIONS
8.1 Conclusion of this Thesis
This thesis presents an enhanced linear least squares algorithm for real-time power
system disturbance identification and its mitigation. The proposed method was named
“efficient least squares algorithm” in the thesis and it determines the power system
disturbance via instantaneous constituent components (i.e. instantaneous fundamental
and harmonic components). The extensive investigations that comprised analytical,
modelling and experimental implementation clearly indicated the real-time applicability
and the superior performance of the proposed power system disturbance identification
method when compared with existing identification methods.
In order to evaluate the standing of the proposed efficient least squares method, the
existing schemes of power system harmonic identifications were first researched and
discussed elaborately in Chapter 2 and 3. The schemes were broadly classified into non-
recursive and recursive identification methods. In the non-recursive class, the discrete
Fourier transform (DFT) including fast Fourier transform method (FFT), wavelet
transform method (WT) and conventional least squares algorithm (CLS) based method
were studied. The Kalman filter was studied under the recursive estimation method.
Chapter 2 provided review including brief discussion on principle of operation of each
method mentioned above. Chapter 3 investigated the performance of each method and
compared them in terms of accuracy of estimation and transient response time. This
investigation was conducted using computer models and their simulations for each
223
method. The effects of noise distortion and marginal fundamental frequency variations
on the estimation accuracy of each method were studied and compared. The
computational complexity of these harmonic identification schemes were also discussed
in Chapter 3. The investigations in Chapter 2 and 3 showed that the conventional least
squares method has good identification accuracy and offers fast tracking of time-
varying harmonic components. However, this method was identified to be unsuitable for
real-time applications due to its high computational burden. This is due to the fact that
the CLS algorithm requires a matrix inversion and several matrix multiplication
operations. In addition the computer implementation of matrix inversion may cause
round-off error which could lead to failure in the numerical process.
Having identified the shortcomings of the CLS method, Chapter 4 proposed
improvements to the CLS algorithm based harmonic identification method, particularly,
for reducing the computational requirement so that it can be used in real-time mitigation
applications. The proposed method was named as “efficient least squares algorithm” in
this thesis due to its reduced computational complexity. In the proposed method, the
matrix inversion operation which is needed in the CLS method was removed using an
approach similar to the singular value decomposition (SVD) technique. The several
matrix multiplications that are required in the CLS method were simplified to a single
real-time matrix multiplication. The derivation of the proposed method was conducted
using complex number representation and then it was transformed into real numbers
representation. The complete mathematical derivation of the proposed method was
elaborately described in this chapter.
The proposed efficient least squares algorithm has a simple structure. It calculates
instantaneous cosine and sine terms of fundamental and harmonic components by
224
simply multiplying a set of sampled input data by a pre-calculated constant matrix. This
algorithm performs only one matrix multiplication per sampling time which corresponds
to only 2 K u L ( K and L are the number of harmonics to be identified and number of
samples respectively) multiplication and addition operations. The comparison showed
that the proposed efficient least squares algorithm has significantly low real-time
computational complexity compared to most of the existing harmonic estimation
methods. The proposed method operates in the time domain and directly outputs
instantaneous cosine and sine terms of fundamental and harmonic components instead
of amplitudes and phase angles of each component. This allows convenient signal
processing in real-time power system disturbance mitigation applications.
The simulation and experimental results presented in this thesis clearly indicated that
the proposed efficient least squares algorithm offers fast tracking of the time-varying
individual harmonic components including the fundamental component. The transient
response time of the proposed method is less than half a cycle of the fundamental
period. The transient response time of the proposed method is superior to those of the
conventional harmonic identification methods such as DFT/FFT, in which the transient
response time is at least one fundamental cycle.
The estimation performance of the proposed efficient least squares algorithm under
noisy conditions was investigated using the Cramer-Rao bound (CRB) analysis. The
result of this analysis showed that the proposed method approaches the CRLB (i.e.
Cramer-Rao Lower Bound) better than does the conventional least squares algorithm.
This indicates the better estimation accuracy of the proposed method. In addition the
identification accuracy of the proposed method and the DFT method under noisy
225
conditions was compared using computer simulation case studies. The results indicated
that the proposed method has superior identification accuracy over the DFT method.
In the proposed algorithm, as with conventional non-recursive harmonic detection
methods, the fundamental power system frequency must be a known value and it is
assumed to be a constant. However, slight changes in fundamental frequency occur
occasionally in a real power system. The effect of small changes in fundamental
frequency on the accuracy of identification for the proposed method and the DFT
method was compared using computer simulation case studies. The results showed that
the proposed method has better transient response and steady state accuracy compared
to those of the DFT method.
As mentioned earlier in this section, the proposed efficient least squares algorithm
calculates the instantaneous cosine and sine terms of the fundamental and harmonic
components. In Chapter 5 these instantaneous cosines and sines were used to obtain
various power system measurements that are required for real-time monitoring and
disturbance mitigation in power systems. In addition to the instantaneous fundamental
and harmonic components, this power signal processing system can identify the
instantaneous and stationary symmetrical components, RMS values and phase angles of
the fundamental and harmonic components, total harmonic distortions (THD), active
power of the fundamental and harmonics components, apparent power, reactive power
and power factor. The detailed mathematical derivations of each of these quantities
were presented. Extensive modelling and simulation studies were conducted in order to
investigate the performance of the proposed signal processing system. The simulation
results clearly indicated that the signal processing system retains all the good features of
the proposed efficient least squares method discussed earlier. The experimental
226
verification of the proposed efficient least square algorithm based harmonic detection
was conducted using an experimental setup that generates known harmonic artificially.
The identification accuracy was established by comparing the actual waveform and the
reconstructed waveform by summing the individual fundamental and harmonic
components identified using the proposed method. Using this method, the percentage
error remained less than 3% for all conditions.
The extensive studies presented in Chapter 4 and 5 indicated that the proposed efficient
least squares algorithm based signal processing is a potential candidate for identifying
disturbances in power systems for real-time monitoring and mitigation applications.
In order to investigate the practicability, the proposed power signal processing method
was applied to two real-life disturbance mitigation applications namely, an active power
filter (APF) and a distribution synchronous static compensator (D-STATCOM). Chapter
6 presented the application of the proposed method in detecting individual harmonics
for an active power filter (APF). The active power filter discussed was based on a shunt
connected voltage source converter which injects harmonic currents into the power
system in order to cancel out the harmonic currents that are generated by the nonlinear
load. The selective harmonic compensation strategy was utilized for this APF due to its
advantages over total harmonic compensation. The detailed modelling and analysis of
the APF system and the systematic design of the current and voltage controllers were
presented. The extraction of the harmonic components of the nonlinear load and the
current controller phase lag compensation using the proposed method were discussed.
Various simulations for the APF system were conducted with and without the current
controller phase lag correction. The simulation results demonstrated that phase-lag
correction of current controllers significantly improves the harmonic compensation
227
performance. The simulation results showed that the APF has good transient response
for a sudden change in nonlinear load current. The simulated performance of the APF
with the proposed method incorporated and the DFT method were compared for a step
increase in supply frequency. The APF with the proposed method showed superior
transient and steady state filtering performance which demonstrated less sensitivity to
small system frequency changes. A laboratory experimental setup for the APF was built
using an IGBT voltage source converter in order to verify the performance. The
experimental performance of the APF is extensively investigated for various operating
conditions. The results confirmed the short response time and good filtering
performance of the APF. In the case of thyristor full bridge load with zero degree firing
angle, the most significant 5th order harmonic was reduced by 81.72% after filtering.
The other significant harmonics 7th, 11th and 13th were reduced by 90.19%, 38.36% and
33.26% respectively. Nevertheless, the harmonic compensation performance of the
active power filter should be able to be improved further by applying more sophisticated
control techniques and this is referred to in the next section.
An application of the proposed efficient least squares method in detecting voltage
unbalances and dip for a distribution synchronous static compensator (D-STATCOM)
was discussed in Chapter 7. In the proposed voltage dip and unbalance detection
method, the instantaneous cosine and sine terms of the fundamental voltage component
were identified using the proposed method and these cosine and sine terms were utilized
to obtain the instantaneous positive- and negative- sequence components. The identified
instantaneous positive- and negative- sequence components were then transformed into
the positive- and negative- synchronous reference frame dq components respectively
in order to facilitate fast and convenient voltage dip and unbalance mitigation. The size
228
of the constant matrix (i.e. Cc ) in the proposed least squares algorithm required for this
D-STATCOM application is 2 u L , since only the fundamental component needs to be
identified (i.e. K 1 ). This resulted in low computational burden in microprocessor or
digital signal processing (DSP) based implementation. In addition the proposed method
is capable of identifying the voltage dips and unbalances accurately even for cases
where the point of common coupling is strongly corrupted by the voltage harmonics.
The simulation results showed that the proposed voltage dip detection method is able
to identify the dq components of voltage dips and unbalances within a half
fundamental cycle.
The complete D-STATCOM system was modelled and simulated for grid unbalance
and voltage dip mitigation. Extensive simulation studies showed that the proposed
voltage unbalance and dip detection method allows fast and accurate extraction of
information that is required for successful operation of the D-STATCOM system. A
laboratory prototype D-STATCOM system was built using an IGBT voltage source
converter in order to verify the operation of the D-STATCOM system. The proposed
voltage dip detection method and the D-STATCOM control algorithm were
implemented on the dSPACE DS1104 R&D board. The prototype D-STATCOM
system is tested for the mitigation of grid unbalances, together with balanced and
unbalanced voltage dips. The experimental results verified the successful real-time
application of the proposed voltage unbalance and dip detection method in extracting
information for the D-STATCOM application.
229
8.2 Suggestions for Future Work
8.2.1 Inter-harmonic detection
The term “inter-harmonic” refers to those frequencies that are not integer harmonics of
the fundamental frequency of the power supply [89]. The usual origins of inter-
harmonics are periodically time-varying loads such as cycloconverters and arcing loads.
Inter-harmonic currents present the same problems with heating and inductive
interference as do integer harmonic currents [89], [90]. Moreover, inter-harmonic
currents cause inter-harmonic voltage distortion in the same manner as for harmonics
and create similar concerns. Electrical power engineers investigating current and
voltage waveform distortion in power systems have been placing interest in inter-
harmonics [91], [89], [90], [92]. Even though the DFT method is used to detect the
inter-harmonic present in a signal, it faces various difficulties due to leakage and picket-
fence effects [89].
The proposed efficient least squares method may be able to be applied for inter-
harmonic detection successfully. The frequencies of harmonic components included in
which has been discussed in chapter 4, does not necessarily have to be

the matrix A
integer multiples of the fundamental frequency. Thus, the constant matrix (i.e., Cc ) of
the proposed algorithm can be included with elements that are not integer multiples of
fundamental frequency. This is needed to be further investigated, possibly using
modeling and computer simulations. Successful application may allow real-time
instantaneous inter-harmonic detection and mitigation similar to the case for integer
harmonics.
230
8.2.2 Voltage dip detection for dynamic voltage restorer (DVR)
The purpose of dynamic voltage restorer (DVR) is similar to D-STATCOM and used to
protect sensitive loads from the effects of voltage dip on the distribution feeder. The
DVR which is placed in series with a sensitive load must be able to respond quickly to a
voltage dip if end users of sensitive equipment are to experience no voltage dip. It is
necessary for the DVR control system to not only detects the start and end of a voltage
dip but also to determine the dip depth and any associated phase shift. A typical
standard detection method such as the DFT/FFT is too slow in supplying this
information.
The proposed efficient least square algorithm may be applied for voltage dip detection
for use in conjunction with the main control system of a DVR. The proposed technique
is able to compute the phase shift and the reduction in the supply voltage much quicker
than typical DFT/FFT methods.
8.2.3 Application of more sophisticated control techniques for APF
As discussed in Section 6.2.5, in order to cope with the delay of the current control loop,
pre-determined phase leads that are calculated from the frequency response are
introduced to each of the individual harmonic components. Even though with this
technique the delay of the current controllers are theoretically compensated, this
approach is essentially based on feedforward open-loop compensation and is sensitive
to parameter mismatches and relies on the ability to accurately predict the current
controller performance. To overcome this problem, researchers have proposed a closed-
loop compensation control of selected source current harmonics [60] and a closed-loop
selective harmonic control based on repetitive techniques [38]. In the first method, the
selective harmonics in both the load current and source current are required to be
231
identified. The latter method needs individual harmonics of only the source current to be
identified and in order to improve the performance of selective compensation an
adjustable leading phase current control algorithm is proposed [5]. The harmonic
identification in this method is based on the discrete Fourier transform (DFT). The
proposed efficient least square algorithm in this thesis has various advantages over the
DFT method and an application of the proposed efficient least square algorithm for the
APF with feedback (i.e closed-loop) selective harmonic compensation may be
investigated in the future with advantage.
232
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240
Appendix A: The Cramer-Rao Lower Bound of the proposed method
APPENDIX A
THE CRAMER-RAO LOWER BOUND
The Cramer-Rao Lower Bound is a measure for evaluating performance of estimator.
As a background, Section A.1 determines CRLB for a single sinusoid. Section A.2 and
A.3 describe the determination of CRLB for linear least squares.
A.1 Cramer-Rao Lower Bound (CRLB) for a Single Sinusoid
The data of single sinusoid can be assumed to be
y > m@ X cos Z m T e m $
where X is the amplitude of the sinusoid, T is the phase angle, Z is the known angular
frequency, and e m is a Gaussian white noise (WGN) with zero mean and variance
V2.
T
The vector, y ª¬ y 0 ,! , y L 1 º¼ is defined for L samples m 0,1,! , L 1 . If
X and T are parameters to be estimated, the parameter vector is defined as
T
I >X T @ . Then CRLB for a single sinusoid in vector form follows as (Theorem 3.2
of [41])
i
1
var Iˆ t ª¬ I I º¼ ii $
241
As the signal is corrupted by Gaussian white noise, (A.2) is simplified to (Section 3.9 of
[41]):
1 L 1
wy > m; I@ wy > m; I@
ª¬ I I º¼ ij 2 ¦ $
V m 0 wIi wIj
where ª¬I I º¼ ij is a element of Fisher information matrix.
In evaluating the CRLB, let r Z m T and use certain simplification based on
following approximations [93]
L 1
¦ sin 2r | 0
m 0
$
L 1
¦ cos 2r | 0
m 0
$
By substituting (A.4) and (A.5) into elements of Fisher information matrix in (A.3),
L 1 L 1
1 1 §1 1 · L
ª¬ I I º¼11 2 ¦ cos2 r 2 ¦ ¨© 2 2 cos 2r ¸¹ | 2V 2
$
V m 0 V m 0
L 1
1 X
ª¬I I º¼12 ª¬ I I º¼ 21 2 ¦ X cos r sin r sin 2r | 0 $
V m 0 V2
L 1 L 1 2
1 1 §1 1 · LX
ª¬ I I º¼ 22 ¦X 2
sin 2 r ¦X 2
¨ sin 2 r ¸ | $
V2 m 0 V2 m 0 ©2 2 ¹ 2V
2
Then, the Fisher information matrix becomes
242
ªL º
0 »
1 «2
I I « » $
V2 « LX 2 »
«¬ 0 2 »¼
Therefore
ª2 º
« 0 »
V2 «L
1
I I » $
«0 2 »
¬« LX 2 ¼»
From (A.10) the CRLB for each estimated parameter of a single sinusoidal is obtained
as follows:
2V 2

var Xˆ t
L
$
2V 2

var Tˆ t
LX 2
$
A.2 Variance of Linear Least Squares Method
In matrix notation of the system equation from (4.3) with noise is given below.
y Ax e $
where
y [ y[0], y[1],! , y[ L 1]]T ,
x [ X , T ]T ,
matrix A is the main matrix with known of dimension L u 2 K ,

243
e [e[0], e[1],! , e[ L 1]]T and this noise vector has the statistical characterization
e N (0,V 2 I ) .
From Section 4.3 of [41], the partial derivative of logarithmic likelihood function
p y; x can be written as follows:
w ln p y; x w ª L
1 º
« ln 2SV 2 y - Ax y - Ax »
2 2 T

wx wx ¬ V ¼
1w T
ª y y - 2y T Ax + x T A T Ax º¼ $
2V wx ¬
2
Using matrix identities and simplifying,
w ln p y; x 1
ª A T y - A T Ax º¼ $
wx V2 ¬
Assuming that A T A is invertible
w ln p y; x A T A ª T -1 T
wx V 2 ¬«
A A A y - x º¼» $
Therefore, the minimum variance unbiased (MVU) estimator of x is given as
-1
xˆ A A
T
AT y $
I x
A A T
$
V2
The covariance matrix of x is found as
244
1
Cxˆ = I -1 x V 2 A T A $
1
C-1xˆ = I $
V2
A.3 Cramer-Rao Lower Bound of the Linear Least Squares Technique
The complex form of (A.13) can be written as
y x e
A
where
y [ y[0], y[1],! , y[ L 1]]T ,
x [ X , T ]T ,
is the main matrix with known of dimension L u 2 K ,

matrix A
e [e[0], e[1],! , e[ L 1]]T and this noise vector has the statistical characterization
e N (0,V 2 I ) .
If X ,T , are the parameters to be estimated the complex amplitude and phase angle, the
T
parameter vector can be defined as [ > X ,T @ . The Cramer-Rao Lower Bound for a
complex signal, in vector form, can be written as follows (Section 15.7 of [41])
i
1
var [ˆ t ª¬I [ º¼ ij $
245
where I [ is the complex Fisher information matrix and can be defined as
ª wC [ 1 wC [ º
ª¬ I [ º¼ ij tr «Cx 1 [ x Cx [ x »
«¬ w[i w[ j »¼
$
ª wP H [ 1 wP [ º
2 Re « Cx [ »
¬« w[i w[ j ¼»
T
As can be seen from (A.20), Cx is not a function of [ (i.e. [ > X ,T @ ). Therefore,
wCx [ wCx [
0 and 0 . As result, the first term of (A.22) is zero. Then, the
w[i w[ j
Fisher information matrix can be written as follows:
ª wP H [ 1 wP [ º
ª¬ I [ º¼ ij 2 Re « 2
I » $
¬« w[i V w[ j ¼»
P [ and P + [ for (A.13) can be written as below.
T
ª y (0) º ª y * (0) º
« y (1) » « * »
« » « y (1) »
« # » « # »
P [ « » DQG P [
+
« * » $
« y (m) » «
y ( m ) »
« # » « # »
« » « * »
¬ y ( L 1) ¼ «¬ y ( L 1) »¼
By applying (A.24) to (A.23) and simplifying,
2 ª L 1 wy * (m) wy (m) º

ª¬ I [ º¼ ij Re «¦ » $
V 2 ¬« m 0 w[i w[ j ¼»
246
X j 2S fm T j 2S fm T
The partial derivatives of the sampled input signal y (m)
2
e e and
X j 2S fm T j 2S fm T
y * (m)
2
e e with respect to amplitude and phase angle can be
written as follows
wy (m) 1 j 2S fm T j 2S fm T
w[1 2

e e
$
wy (m) jX j 2S fm T j 2S fm T
w[ 2 2
e e $
wy * (m) 1 j 2S fm T j 2S fm T
w[1 2
e e $
wy * (m) jX j 2S fm T j 2S fm T
w[ 2 2
e e $
By substituting partial derivatives given in (A.26),(A.27),(A.28) and (A.29) into (A.25),
2 ª L 1 wy * (m) wy (m) º

ª¬ I [ º¼11 Re ¦
V 2 «¬ m 0 w[ w[ ¼
»
$
2 ª L 1 1 j 2S fm T j 2S fm T º
Re ¦ e
V 2 «¬ m 0 4
e
j 2S fm T
e
j 2S fm T
e »
¼
2 ª L 1 wy * (m) wy (m) º

ª¬ I [ º¼12 Re ¦
V 2 «¬ m 0 w[ w[ »¼
$
2 ª L 1 jX j 2S fm T j 2S fm T º
Re ¦
V 2 «¬ m 0 4
e e
j 2S fm T
e
j 2S fm T
e »
¼

247
2 ª L 1 wy * (m) wy (m) º

ª¬ I [ º¼ 21 Re ¦
V 2 «¬ m 0 w[ w[ ¼
»

2 ª L 1 jX j 2S fm T j 2S fm T º
Re ¦
V 2 «¬ m 0 4
e e
j 2S fm T
e
j 2S fm T
e »
¼

$
2 ª L 1 wy * (m) wy (m) º

ª¬ I [ º¼ 22 Re ¦
V 2 ¬« n 0 w[ w[ ¼»
2
2
Re ª¬ Xje j 2S fm T Xje j 2S fm T º¼
V
2 ª L 1 X 2 j 2S fm T º
V 2
Re « ¦
¬m 0 4
e
e j 2S fm T e j 2S fm T e j 2S fmT »
¼

$
By simplifying (A.30),(A.31),(A.32) and (A.33) ,
2L
ª¬ I [ º¼11 | 2 $
V
ª¬ I [ º¼12 | 0 $
ª¬ I [ º¼ 21 | 0 $
LX 2
ª¬ I [ º¼ 22 | 2 $
V
So that, the Fisher information matrix becomes
1 ª2L 0 º
I [ $
V ¬ 0 LX 2 »¼
2 «
Therefore,
248
ª 1 º
« 0 »
I -1 [ V 2 « 2L » $
« 0 1 »
¬« LX 2 ¼»
The CRLBs for the estimated amplitude and phase angle are obtained as
2
V

var Xˆ t
2L
$
V2

var Tˆ t
LX 2
$
249
Appendix B: Modelling and Real-Time Control Programs
APPENDIX B
MODELLING AND REAL-TIME CONTOL

PROGRAMS
B.1 Modelling Programs Using MATLAB
The simulation programs for this thesis have been developed using MATLAB/
SIMULINK software. The simulation programs included in the thesis are developed in
MATLAB 6.5.1. The simulation programs are organized as directories. Each directory
contains an initialization file init_matrixC_xxxxx.m which has to be run first to load the
constant matrix for the proposed efficient least squares method. Some directories
contain another initialization file init_xxxx.m which contains the controller parameters.
This file also has to be run prior to the main program. After initialization, the specific
SIMULINK model file can be opened and run.
The simulation programs can be broadly classified under the following:
B.1.1 Proposed power signal processing system in detecting fundamental and

harmonic components
These programs are used to test the performance of the proposed power signal
processing system in detecting fundamental and harmonic components described in
Chapter 5, Section 5.4.1.
Directory: MODELING \ Chapter5\ single
Matlab/Simulink files:
250
1. Initialization file: init_matrixC_7K.m
2. Simulink model file: ELS_1p.mdl
B.1.2 Proposed power signal processing system in estimating power and total
RMS values
These programs are used to test the performance of the proposed signal processing
system in estimating various power measurements and total RMS values discussed in
Chapter 5, Section 5.4.2.
Directory: MODELING \ Chapter5\ single
2. Simulink model file: ELS_1p_power.mdl
B.1.3 Proposed power signal processing system in determining symmetrical

components
These programs are used to test the performance of determination of symmetrical
components using the proposed power signal processing system described in Chapter 5,
Section 5.4.3.
Directory: MODELING \ Chapter5\ threephase
2. Simulink model file: ELS_3p_symm.mdl
251
B.1.4 Proposed harmonic detection method in extracting current harmonic

components of a practical nonlinear load
These programs are used to test the performance of the proposed individual harmonic
detection method in extracting individual current harmonic components in input current
of a full bridge thyristor rectifier load described in Chapter 6, Section 6.3.1.
Directory: MODELING \ Chapter6\ harmonic
1. Initialization file: init_matrixC_5K_APF.m
2. Simulink model file: APF_ELS_harmonic.mdl

compensation of active power filter
These programs are used to test the performance of the harmonic detection method in
selective harmonic compensation for an active power filter described in Chapter 6,
Section 6.3.2.
Directory: MODELING \ Chapter6\APF
1. Initialization file: init_matrixC_5K_APF.m
2. Initialization file: init_APF.m
3. Simulink model file: APF_ELS_system.mdl
252
B.1.6 Proposed voltage unbalance and dip detection method in D-STATCOM

system
These programs are used to test the performance of the D-STATCOM with the
proposed voltage dip detection method in described in Chapter 7, Section 7.5.
Directory: MODELING \ Chapter7\DSTATCOM
1. Initialization file: init_matrixC_1K_DSTATCOM.m
2. Initialization file: init_DSTATCOM.m
3. Simulink model file: DSTATCOM_ELS_system.mdl
B.2 Real-Time Programs used for Experiments
The real-time programs have been developed under the dSPACE DS 1104 environment.
These real-time programs are organized as directories and each directory contains
number of files including main ControlDesk experiment file that links all the relevant
files. In addition the directories contain SIMULINK model file and the corresponding
initialization files.
The real-time programs can be mainly classified under the following categories:
B.2.1 Proposed power signal processing system in estimating instantaneous

fundamental and harmonic component
These programs are used for experimental verification of the proposed efficient least
squares algorithm described in Chapter 5, section 5.4.4.
253
Directory: EXPERIMENT \ ThreePhase
File Name Description
Pwmgeneration_withmesurin.MDL SIMULINK Model
yuu.LAY Layout file of the ControlDesk
ControlDesk experiment file contains links to all

PWMgeneration.CDX
files related to the experiment
PPC file is the executable file for the real-time

Pwmgeneration_withmesurin.PPC
processor.
pwmgeneration_withmesurin.SDF System description file.
This file provides information on the available

Pwmgeneration_withmesurin.TRC
variables of the real-time interface.
init_matrixC_5K.m Constant matrix initialization file.

compensation of active power filter
These programs are used to conduct experiments for active power filter based on the
proposed harmonic detection method described in Chapter 6, section 6.4.
254
Directory: EXPERIMENT \ APF
ActiveRec.MDL SIMULINK Model
main.LAY Layout file of the ControlDesk

Active_rec.CDX

ActiveRec.PPC
processor.
ActiveRec.SDF System description file.

ActiveRec.TRC
Init_APF.m Control Parameter initialization file for APF.
init_matrixC_5K_APF.m Constant matrix initialization file.
B.2.3 D-STATCOM with the proposed voltage dip detection
These programs are used to conduct experiments for the D-STATCOM system with the
proposed efficient least squares algorithm based voltage dip detection described in
Chapter 7, Section 7.6.
255
Directory: EXPERIMENT \ DSTATCOM
ActiveRec.MDL SIMULINK Model
main.LAY Layout file of ControlDesk

Active_rec.CDX

Active_rec.PPC
processor.
Active_rec.SDF System description file.

Active_rec.TRC
Control parameter initialization file for the

Init_DSTATCOM.m
DSTATCOM.
init_matrixC_1K_Dstatcom.m Constant matrix initialization file.
256
Appendix C: Experimental setups
APPENDIX C
EXPERIMENTAL SETUPS
C.1 Overview of Experimental Setups
Three different experimental setups are used to obtain results presented in this thesis.
The first setup is used for experimental verification of the proposed efficient least
squares algorithm. The second and third setups are a prototype active power filter
system and a prototype D-STATCOM system respectively. The core components of all
these experimental setups are a three-phase 35kW IGBT converter and a dSPACE
DS1104 controller board (i.e. R&D board). A PC with Pentium III 800MHz is used to
host the dSPACE controller board and the software development throughout the thesis.
The presumed advantage of this system is the rapid development speed of programs
through the use of the dSPACE Real-time Interface (RTI). This feature enabled
simulated systems developed in Matlab/Simulink software to be automatically encoded
into C-code for downloading to the DS1104 DSP system. Detailed descriptions of the
three-phase IGBT converter and the DS1104 controller board will be given later in this
appendix.
x Experimental verification of the proposed efficient least squares algorithm
The experimental verification of the proposed efficient least squares algorithm is
conducted using the setup illustrated in Figure C.1. In this setup, the IGBT converter is
supplied by a diode bridge rectifier connected to a variable voltage supply. This
arrangement is similar to an inverter that is used in three-phase dive applications. The
257
ac-side of the converter is connected to an inductive load (RL load) via filter
inductances. These filter inductances are used for filtering the PWM pulses in the
converter output voltages (i.e. ac-side voltages). The DS1104 controller board generates
the PWM signals for a waveform that contains known harmonic components. The
filtered line-voltages at the load are measured using three voltage sensors and taken into
the DS1104 via the analogue inputs. Theses signals are then treated with the proposed
efficient least squares algorithm which is implemented on the DS1104 controller board.
Variac DIODE BRIDGE IGBT INVERTER

3-Phase AC Supply
LF
+ RL
Cdc LOAD
Voltage
PWM signals Sensors
PC WITH dSPACE 1104

R&D BOARD vab
vbc
vca
Figure C.1 Experimental verification of the proposed efficient least squares

algorithm.
x Active power filter (APF) system
The APF system is set up utilizing the three-phase IGBT converter with ac-side
inductors as shown in Figure C.2. A three-phase controlled rectifier (i.e. thyristor bridge
rectifier) is chosen as the non-linear load that draws a high level of harmonic and
reactive currents. A variable three phase supply is used to power the nonlinear load.
Four current sensors and three voltage sensors were used for measuring load currents,
258
APF currents, source voltages and dc-link voltage as indicated in Figure C.2. These
measured currents and the voltages are taken into the DS1104 via analogue inputs. The
control algorithm including the proposed harmonic detection method implemented on
the DS1104 processes these inputs and generates the PWM outputs to switch the IGBTs
in the converter.
Variac
3-Phase AC Supply
NON-LINEAR
LOAD
IGBT CONVERTER
LF
Cdc
Voltage
Current sensors
sensors
PWM Signals
Vdc PC WITH dSPACE 1104 iFb

R&D BOARD
vab iFa
vbc iLa
vca iLb
Figure C.2 Experimental setup for active power filter system.
x D-STATCOM system
The D-STATCOM system is set up utilizing the three-phase IGBT converter with two
sets of inductors and a set of filter capacitors on the ac-side as shown in Figure C.3. A
variable voltage supply and another set of inductors are used to emulate the power grid.
Two resistive three-phase load banks are used in the system. The load bank -1 with a
switch is used to generate voltage dips and unbalances in the emulated grid. The system
is able to generate both balanced and unbalanced voltage dips. In the case of unbalanced
259
dip, the load bank-1 is set to be unbalanced load. Two current sensors and four voltage
sensors are used for measuring converter currents and line voltages at the point of
common coupling (PCC) together with dc-link voltage as indicated in Figure C.3.
Switch
Load
bank-A
Variac
Lline
3-Phase AC Supply
PCC
Load
Ltr
CF
IGBT CONVERTER
LF
+
-
Cdc
Voltage
sensors
Current
PWM Signals
sensors
Vdc PC WITH dSPACE 1104
R&D BOARD iFb
vab
vbc
iFa
vca
Figure C.3 Experimental setup for D-STATCOM system.
C.2 Description of dSPACE DS1104 Controller Board
The DS1104 Controller Board is specifically designed for the development of high-
speed multivariable digital controllers and real-times simulations in various fields. It is a
complete real-time control system based on a 603 Power PC floating point processor
260
running at 250 MHz. For advanced I/O purposes, the board includes a slave-DSP
subsystem based on the TMS320F240 DSP microcontroller. The DS1104 DSP board
also provides an on–board slave processor for producing PWM control signals. This
feature means that a signal DSP card can be used to control the converter,
measurements and process the captured data in real-time.
The board is provided together with the software “ControlDesk”, which allows real-time
management of the inputs and outputs by providing a virtual control panel with
instruments and scopes and which allows modification of controller parameters during
the experiment.
Manufactuer: dSPACE GmbH
Technologiepark 25
33100 Pederborn
Germany
Processor: Main processor operating at 250MHz, 64-bit floating point.
Memory: Global memory: 32 MB SDRAM and flash memory 8MB.
ADC: The board has 4 multiplexed channels with 16-bit resolution; 2Psecond
conversion time. It also has 4 A/D channel with 12-bit resolution and 800Ksec.
conversion time.
Slave DSP subsystem: Texas Instruments TMS320F240 DSP @ 20MHz clock
frequency. It has 1x 3 phase PWM output and 4x1 phase PWM outputs.
261
For the complete specification and detail of the ds1104 R&D Controller board see [94]
and [95].
Connector panel for DS1104: The connector panel provides easy connections between
the DS 1104 and the converter and transducers. A photograph of connector panel
CP1104 is shown in Figure C.4. The PWM signals can be connected using a CP18 Sub-
D connector. The pin mapping of the interconnection cable between the connector panel
CP1104 and the IGBT converter is shown in Figure C.5. The currents and voltage
signals are taken into the DS1104 using BNC connectors CP1-CP8 on the CP1104
connector panel.
Figure C.4 A photograph of CP1104 connector panel.
262
PWM Control Signals to 3 Phase

CP18-DS1104 Semikron IGBT Inverter Signal Interface
3 Channel PWM PCB inside Inverter
(7) (1) PWM_1
(8) (2) PWM_3
(9) (3) PWM_5
(12) (15) GND
Figure C.5 Pin mapping of interconnection cable between CP18 Sub-D of

connector panel CP1104 and the IGBT converter.
C.3 Description of Three-phase 35kW PWM Converter
The IGBT converter was designed and built in-house at the University of New South
Wales [96] [97]. The power circuit diagram of a 35kW IGBT converter and converter
interfacing circuit are shown in Figure C.6 and Figure C.7 respectively. The power
circuit given in Figure C.6 consists of a diode-bridge rectifier which was used only in
the first part of the experimental study where the converter was used as an inverter to
generate known harmonics. This diode bridge rectifier was removed in the rest of the
experimental study (i.e. active power filter and D-STATCOM). The dc-link capacitors
are rated to 1100V and a large capacitance value allows effective filtering of the
switching ripples and helps stiffening the dc-link voltage. The output is produced by
switching a Semikron three-phase IGBT smart module. The dc-link voltage, line-line
voltages and line currents are measured by isolated voltage and current transducers as
shown in the diagram in Figure C.6. The control signals are produced in the dSPACE
263
DS1104 DSP board and connected to the converter through a series of crossover
protection modules shown in Figure C.7. The DSP board is used to provide only three
PWM signal lines which correspond to three IGBTs on the upper arm of the converter
legs. The control signals for the lower three IGBTs are derived from the three upper
IGBT signals in the crossover protection circuitry that resides inside the converter. The
IGBT switching deadband was set at 4Ps. Figure C.10 shows the mapping of the
interconnection cable that connects the interface PCB to the Semikron IGBT module. A
photograph of the three-phase IGBT converter is shown in Figure C.11.
The component specifications are:
1. Semikron SKiip IGBT module
- 192GDL170-475CTV
- SK No.: 20226332
2. Semikron Semipont rectifier
- SKD 160/12
3. dc-link capacitors
Ten 3300 PF, 550V each.
Five pairs of series connected capacitors were connected in parallel for a maximum
dc-link voltage of 1100V. A 10k:, 10W resistor was connected across each
capacitor for dividing the voltage equally between the series connected capacitors.
264
4. Current transducers
The isolated LEM LTA 100P/SP1 current sensors are used to feedback the converter-
currents to the DSP controller. These current sensors are Hall Effect current sensors
with a frequency range of 0-150 kHz. The current sensor amplifier output is limited
to 9.1V as a measure of protecting the analogue inputs of the DSP board. The
schematic diagram of current transducer PCB is shown in Figure C.8. The gains of
the current sensors that are resided inside the converter are given below.
- I Fa (Red phase): 80.5A/10V
- I Fc (Blue phase): 78.3A/10V
5. Voltage transducers
An isolated LEM LV100 voltage sensor is used to measure the dc-link voltage and
feed them into the DSP controller. The voltage sensor amplifier output is limited to
9.1V as a measure of protecting the analogue inputs of the DSP board. The schematic
diagram of voltage transducer PCB is shown in Figure C.9. The gain of the dc-link
voltage sensors that reside inside the converter is given below.
- dc-link voltage: -794.1V/10V
265
1 2 3 4
ERROR HB1 OUT

HB1T
ERROR HB2 OUT
ERROR HB3 OUT
HB1B
D OVERTEMP OUT D
HB2T
9
8
7
6
5
4
3
2
1
SKD 160/12 5X
26
25
24
23
22
21
20
19
18
17
16
15
14
13
12
11
10
P HB2B
J1
INPUT CON26
HB3T
100W
HB3B
10K,10W
R
3300uF
550VDC
100W
Y
B B
C C
3300uF
550VDC
10K,10W
192GDL 170-475 CTV
U V W
V UV
SENSOR
E
V VW
SMPS
SENSOR
+15V
B L POWERBOX B
oV
230V
1 -15V
I U
PU65-32C
2
SENSOR
N +5V
230VAC
0V
CONTROL
I W
SUPPLY L
SENSOR
230VAC FA N
N
SEMIKRON
U V W
GD1 33-2K-40206
OUTPUT
Title
A 35 KW I GBT C ONV ER TE R A
V V I I
UV VW U W Size Number Revision
A4 R0
Date: 22-Aug-2006 Sheet1 of 2
File: G:\PCB Design\Schematic Diagrams\INVCKT.DDB
Drawn By: KVB
1 2 3 4
Figure C.6 IGBT converter power circuit.
266
1 2 3 4 5 6
+5V +5V
16
D1 +5V U?
1N4148 +5V 4 3
R1 I0 O0
2 U?A
O0
VddI
1K U? 5
D2 I1
1 18 6 3 2
R Vcc O1
D 1N4148+5V 11 7 D
I2 O1
2 17
ENAR RU CD4050
12 10 U?B J?
I3 O2
3 16 9
D3 S RL O2 1
5 4
1N4148 2
4 IXYS 15 15 13
ENAS SU EO O3 3
IXDP630 8 14
VddO
R2 GND O3 CD4050 4
5 14
D4 T SL 5
VCC 1K HEF4104
1N4148 6
6 13
1
ENAT TU U?C 7
8
7 12 7 6
OUTENA TL 9
FAULT +15V
10
+5V 8 11
RESET OSCOUT CD4050 11
+5V
J? 12
9 10 +15V
GND RCIN 13
8 R5 U?D 14
16
15 D5 IXDP630
1N4148 15
7 47K U? 9 10
16
14 4 3
R3 I0 O0 17
6 R6 2
O0 CD4050 18
VddI
13 1K D6 5
I1 19
5 1N4148+5V 6
O1 20
12 11 7 U?E
C I2 O1 21 C
4
22
11 PWM3 20K 12 10 11 12
D7 I3 O2 23
3 9
1N4148 O2 24
10 PWM2
CD4050 25
2 15 13
15 WAY D-CONNEC TOR

R4 EO O3 26
9 8 14
VddO
GND O3
1 1K +15V 26PIN IDC
D8 C1 HEF4104
PWM1 1N4148 0.0047nF
1
1
U?F
+5 1N4148 +15V
14 15
VCC
U1F
GND
K
+5V
R19 CD4050 +15V
14
D1
CN2 100K 8 R1 R2 R3 R4
74HC04 U1B
+5V 10K 10K 10K 10K
A
RESET
13 12 3 4
VCC
GND
K
8
7
PB1 74HC04 U?B 12

U?A
C11 D2 11
7
B B
1.0uF 2 3 13
GND
GND
10
A
VCC
VCC
9
CD4050
1N4148 CD4002
1
14
D3 +15V
LED
+5
A K
5V
R16
15V 680R
J?
U1A
3
2
2 1
1 SKIIP CONTROL PCB
CON3 R17
1
4.7K MM7 4HC04
THREE PWM INPUT AND DEAD TIME GENERATED ON BOARD
A 2 T1 A
BC549 Title
R18
3
10K Size Number Revision
B
Date: 22-Aug- 2006 Sheet of
File: G:\PCB Design\Schematic Diagrams\SKI IPControl.ddb
Drawn By:
1 2 3 4 5 6
Figure C.7 Schematic diagram of the interface PCB inside the IGBT converter.
267
1 2 3 4
D D
P2
R3 20K
33K
C1 330pF
C C
+15
R2 +15
U1C
6 5K6 4 U1D R4 CN1
10
1 8 12 1K CON2
9 14 2
13
4 R1 11 1
5K6 LT1058CN D1
-15 LT1058CN 9.1VZ
2
-15 LA100-P/SP13 D2
B
9.1VZ B
CN2
CON3 +15
+1 5V 1
0V 2 C5 C3 C4
3
47uF 10nF 100pF
-1 5V
-15
Title
A IN TE RN AL CU RRE NT S ENS OR BO AR D A
Size Number Revision

A4 R0
Date: 22-Aug- 2006 Sheet1 of 1
File: G:\PCB Design\Schematic Diagrams\MI NI CT.DDB
Drawn By: KVB
1 2 3 4
Figure C.8 Schematic diagram of the current transducer PCB (residing inside the converter).
268
1 2 3 4
+15
D D
R5 J1
10K JUMPER
P1
2K P2
R3 20K
1
2
22K
CN4 C1 330pF
CON2 R6
1 10K
C
R8 2 C2 10nF C
47K -15
25 W
LV 10 0 R2 +15
+15 U1C
+ +1 5V 5K6 4 U1D R4 CN1
CN3 10
8 12 1K CON2
CON2 M 9 14 2
+V 1 13
100R R1 LT1058CN 11 1
2 - -1 5V R7 5K6 LT1058CN
-V -15 9.1VZ D1
-15
B
9.1VZ D2 B
CN2
CON3 +15
+1 5V 1
0V 2 C5 C3 C4
-1 5V 3 47uF 10nF 100pF
-15
Title
A IN TE RN AL VO LTA GE S ENS OR BO AR D A

A4 R0
Date: 22-Aug- 2006 Sheet
1 of 1
File: G:\PCB Design\Schematic Diagrams\MINIVT.DDB
Drawn By: KVB
1 2 3 4
Figure C.9 Schematic diagram of the voltage transducer PCB (residing inside the converter).
269
Appendix C: Experimental setup
SIGNAL INTERFACE CARD SKIP CONTROL CARD TYPE II

red
15 15 GND
yellow
4 4 FAULT
blue
3 3 PWM3
violet
2 2 PWM2
black
1 1 PWM1
Figure C.10 Interconnection cable inside the converter (for PWM signals).
Figure C.11 A photograph of the IGBT converter.
270
C.4 External Voltage and Current Transducers
The hardware of the system consists of the external voltage and current transducer
board for fulfilling the additional current and voltage signal measurement requirements.
This transducer board contains three current and three voltage sensors. The voltage and
current sensors are manufactured by LEM and produce very linear characteristics over
their signal ranges. A photograph of the external voltage and current transducer board is
shown in Figure C.14. The specification of current and voltage transducers are given
below.
1. Current transducers
Isolated LEM LA 100-P/SP13 current sensors are used to feedback the load currents
to the DSP controller. These current sensors are Hall Effect current sensors with a
frequency range of 0-150 kHz. The current sensor amplifier output is limited to 9.1V
as a measure of protecting the analogue inputs of the DSP board. The schematic
diagram of current transducer PCB is shown in Figure C.12. The gains of the
external current transducers are given below.
- I La : -40.2A/10V
- I Lc : -82.1A/10V
2. Voltage transducer
Isolated LEM LV2-P voltage sensors are used to measure the line voltages and feed
them into the DSP controller. The voltage sensor amplifier outputs are limited to
9.1V as a measure of protecting the analogue inputs of the DSP board. The schematic
271
diagram of the external voltage transducer PCB is shown in Figure C.13. The gains
of the external voltage transducers are given below.
- Vab : -690.7/10V
- Vbc : -702.3/10V
- Vca : -690.4/10V
272
1 2 3 4
D D
P2
R3 20K
33K
C1 330pF
C C
+15
R2 +15
U1C
6 5K6 4 U1D R4 CN1
10
1 8 12 1K CON2
9 14 2
13
4 R1 11 1
5K6 LT1058CN D1
-15 LT1058CN 9.1VZ
2
-15 LTA100P/SP1 D2
B
9.1VZ B
CN2
CON3 +15
+1 5V 1
0V 2 C5 C3 C4
3
47uF 10nF 100pF
-1 5V
-15
Title
A EX TE RN AL CU RRE NT S ENS OR BO AR D A

A4 R0
Date: 22-Aug- 2006 Sheet1 of 1
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Drawn By: KVB
1 2 3 4
Figure C.12 Schematic diagram of the current transducer PCB.
273
1 2 3 4
D D
P2
R3 20K
22K
CN4 C1 330pF
CON2
1
C
R8 2 C2 10nF C
47K
25 W
LV 20 -P R2 +15
+15 U1C
+ +1 5V 5K6 4 U1D R4 CN1
CN3 10
8 12 1K CON2
CON2 M 9 14 2
+V 1 13
100R R1 LT1058CN 11 1
2 - -1 5V R7 5K6 LT1058CN
-V -15 9.1VZ D1
-15
B
9.1VZ D2 B
CN2
CON3 +15
+1 5V 1
0V 2 C5 C3 C4
-1 5V 3 47uF 10nF 100pF
-15
Title
A EX TE RN AL VO LTA GE S ENS OR BO AR D A

A4 R0
Date: 22-Aug- 2006 Sheet
1 of 1
File: G:\PCB Design\Schematic Diagrams\MI NI VT.DDB
Drawn By: KVB
1 2 3 4
Figure C.13 Schematic diagram of the voltage transducer PCB.
274
Appendix C Experimental setup
Figure C.14 A photograph of external voltage and current transducers.
275
Appendix D: List of Publications
APPENDIX D
LIST OF PUBLICATIONS
D.1 Papers Relevant to the Thesis Topic
D.1.1 Refereed conference publications
[1] Thip Manmek, Chathura P. Mudannayake, and Colin Grantham, “Voltage Dip
Detection Based on an Efficient Least Squares Algorithm for D-STATCOM
Application,” Proceeding of the 2006 CES/IEEE – PELS International Power
Electronics and Motion Control Conference (IPEMC06), Shanghai, P.R. China
on the 13th Aug – 16th Aug 2006.
[2] Thip Manmek, Colin Grantham and Toan Phung, “Real Time Tracking of RMS
Quantities in Three-Phase Systems under Nonsinusoidal Conditions,”
Proceeding of the 6th International Conference of the IEEE on Power
electronics and Drive systems (PEDS05), Kuala Lumpur, Malaysia on the 28th
Nov - 1st Dec 2005, pp.557-562.
[3] Thip Manmek, Chathura P. Mudannayake, and Colin Grantham, “Robust Signal
Processing System for Identification of Harmonics in an Active Power Filter
Application,” Proceeding of the 6th International Conference of the IEEE on
Power electronics and Drive systems (PEDS05), Kuala Lumpur, Malaysia on the
28th Nov - 1st Dec 2005, pp.313-318.
[4] Thip Manmek, Huu – Phuc To ,Colin Grantham and Toan Phung, “Real Time
Individual Harmonic Measuring Method for Active Harmonic Compensation,”
276
Proceeding of Australasian Universities Power Engineering Conference
(AUPEC 2005), Hobart, Australia, vol.1,25 Sept. - 28 Sept. 2005, pp.260-265.
[5] Thip Manmek, Colin Grantham and Toan Phung, “A New Efficient Algorithm
for Online Measurement of Power System Quantities” Proceeding of the 30th
Annual Conference of the IEEE Industrial Electronic Society (IECON 2004),
Busan, Korea, November 2004.
[6] Thip Manmek, Colin Grantham and Toan Phung, “A Real Time Power
Harmonics Measuring Technique under Noise conditions”, Proceeding of
Australasian Universities Power Engineering Conference (AUPEC 2004),
Brisbane, Australia, 26 Sept. - 29 Sept. 2004. ISBN 1-864-99775-3 (CD ROM),
Paper No. 146.
[7] Thip Manmek, Colin Grantham and Toan Phung, “Novel and Accurate
Instantaneous Power Measurements”, Proceeding of Australasian Universities
Power Engineering Conference (AUPEC 2004), Brisbane, Australia, 26 Sept. -
29 Sept. 2004. ISBN 1-864-99775-3 (CD ROM), Paper No. 179.
[8] T. Manmek, C. Grantham and B. T. Phung “A novel algorithm for identification
and tracking of power system harmonics”, IEEE Proc. International Power
Electronics and Motion Control Conference, (IPEMC’04), Xian, China, 14-16
Aug. 2004, pp.1346-1350.
[9] T. Manmek, C. Grantham and B. T. Phung “A new efficient algorithm for real
time harmonics measurement in power systems”, The first Electrical
Engineering/Electronics, Computer, Telecommunications and Information
277
Technology association of Thailand annual conference (ECTI-CON 2004),
Amari Orchid Resort Hotel, Pattaya, Thailand, 13-14 May 2004,p
[10] T. Manmek, C. Grantham and B. T. Phung, “ A novel technique real time
voltage and current harmonic estimations based on the singular value
decomposition method”, Proceeding of Australasian Universities Power
Engineering Conference (AUPEC 2003), Christchurch, New Zealand, 28 Sept. -
1 Oct. 2003. ISBN 0-473-09867-9 (CD ROM).
D.1.2 Paper not relevant to the thesis topic
[1] W. Ngamkham, T. Manmek and C. Wongtaychatham, “Translinear Peak
Detector Circuit for Sinusoidal Signal”, The first Electrical
Engineering/Electronics, Computer, Telecommunications and Information
Technology association of Thailand annual conference (ECTI-CON 2004),
Amari Orchid Resort Hotel, Pattaya, Thailand, 13-14 May 2004.
278

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Real-Time Power System Disturbance

Identification and its Mitigation using

School of Electrical Engineering and Telecommunications

it contains no material previously published or written by another person, nor material

diploma at UNSW or any other educational institution, except where due

acknowledgement is made in the thesis. Any contribution made to the research by

others, with whom I have worked at UNSW or elsewhere, is explicitly acknowledged in

conception or in style, presentation and linguistic expression is acknowledged.

system disturbances identification method based on an enhanced linear least squares

current harmonics, grid unbalances and voltage dips.

the proposed method by pre-performing some of the matrix multiplications to form a

multiplying a set of sampled input data by the pre-calculated constant matrix.

mitigation. These power system quantities include constituent components, symmetrical

proposed method under both transient and steady-state conditions.

processing system was applied to two real-life disturbance mitigation applications

(D-STATCOM). The validity and performance of the proposed signal processing

system in both disturbance mitigations applications were investigated by simulation and

disturbance identification such as mitigation control and power quality monitoring of

I would like to express my sincere gratitude to my supervisors, Professor Colin

and stimulating suggestions and encouragement have helped me throughout my research

their countless help and encouragement.

I gratefully acknowledge the full financial support from Mahanakorn University of

Technology (Thailand) throughout my PhD study in Australia. Furthermore, I am

financial support which enabled me to attend several international and local

software needed for my project.

experimenting of the active power filter.

throughout my life. They have encouraged me throughout my academic career, and I

certainly owe all of my success to their loving support and guidance.

Kingsford, August 2006

LIST OF TABLES ...................................................................................................... xxix

LIST OF SYMBOLS AND ABBREVIATIONS ........................................................ xxx

1.1 Overview ..................................................................................................................1

1.2 Classification of Power Quality Problems ...............................................................2

1.4 Existing Methods of Identification of Disturbance Quantities ................................6

1.5 Scope of this Thesis .................................................................................................9

1.6 Proposed Approach for Identification of Power System Disturbances..................10

1.7 Structure of Thesis .................................................................................................12

REVIEW OF THE EXISTING TECHNIQUES FOR POWER SYSTEM

HARMONIC ESTIMATION ........................................................................................ 14

2.1 Overview ................................................................................................................14

2.2 Non-Recursive Harmonic Detection Methods .......................................................15

2.2.2 Wavelet transform (WT) …………………………………………………..21

x Continuous wavelet transform (CWT)……………………………………..22

x Discrete wavelet packet transform (DWPT) ………………………………24

2.2.3 Conventional least squares (CLS) method ……………………………….29

2.3 Recursive Harmonic Detection Methods ...............................................................33

2.3.1 Kalman filtering (KF) method …………………………………………….33

2.4 Summary ................................................................................................................37

PERFORMANCE OF THE EXISTING TECHNIQUES FOR POWER SYSTEM

HARMONIC ESTIMATION ........................................................................................ 39

3.1 Overview ................................................................................................................39

3.2 The Performance of DFT/FFT Method..................................................................41

3.3 The Performance of Wavelet Transform (WT) Method ........................................46

3.4 The Performance of Conventional Least Squares (CLS) Method .........................50

3.5 The Performance of a Kalman Filtering (KF) Method ..........................................55

x Performance and the order of Kalman filter 59

3.6 Comparison of the Studied Methods......................................................................61

3.6.1 Effect of noise distortion on estimation accuracy ………………………62

3.6.2 Effect of marginal fundamental frequency variation on the estimation

3.6.3 Transient response time …………………………………………………..65

3.7 Summary ................................................................................................................65

PROPOSED EFFICIENT LEAST SQUARES TECHNIQUE FOR POWER

SYSTEM HARMONIC DETECTION.......................................................................... 67

4.1 Overview ................................................................................................................67