PHASOR ESTIMATION

Power system protection

• The rigid interconnectivity existing among modern power systems make them
highly unstable when, faults are not cleared rapidly.
• If a fault in an important transmission line is not identified and removed at the
shortest possible time, it might lead to a widespread damage in the power system.
• The rapid removal of transmission lines’ faults is one of the best measures used to
improve the power systems’ stability, and, implicitly, to ensure an adequate
reliability of the grid and the continuity into energy transmission.
• In order to accomplish its functions, a protection relay must distinguish the type of
the fault (which phases are involved) and also determine the exact component of
the power system to be isolated.
• When a fault occurs in a power transmission line, the voltage and current signals
are severely distorted.
• These signals may contain dc offsets, high frequency transients, and oscillation
components.
• The simplest method for fast fault clearance is that of the decrease of time of the
protection operation.
• In addition, especially in the competitive electricity markets, the rapid fault
restoration on transmission lines is faced with the quality of the utility’s power
service.
 Power system protection
• So, following the occurrence of a fault, the utility tries to restore power as quickly as
possible.
• In the aim of rapid and efficient service restorations, accurate fault location estimation and
fault discrimination methods are needed.
• The unpredictable nature of the power system signals during faults, make their extraction a
challenging job.
• In order to prevent the damages from spreading to the healthy parts of the power system,
the protective relaying algorithms need to detect the faults within sub cycles of the power
system frequency.
• Accuracy in estimating phasors is then a very important task of a protection relay to ensure,
for example, it is responding to the fundamental frequency components of both currents and
voltages and not to high frequency components generated during a switching transient or
inrush phenomenon.
• For this it is usually necessary to evaluate sequence components and/or harmonic
components of the three-phase voltages and currents separately.
• Pharos estimation algorithms for protective relaying are required to filter out unwanted
components from the input signals and retain only the components of interest.
• The components to be removed include harmonics and the decaying exponential transient
(dc offset) component.
• They affect the accuracy and the speed of convergence of the pharos estimation algorithms
to a great extent.
 Numerical relay

• The process of numerical relaying consists of several steps.

• Currents and voltages from instrument transformers are band-limited using low
pass anti-aliasing analog filters.
• The analog signals are then discritized and quantized to get digital signals.
• The acquired information is then processed by the relay algorithm.
• The algorithm uses signal processing techniques to estimate the magnitudes and
angles of the corresponding current and voltage phasor.
• These estimates are used to calculate other quantities, such as impedances.
• The computed quantities are compared with pre-specified thresholds to decide
whether the power system is experiencing a fault condition or not.
• If it is, relay sends a command to open one or more circuit breakers to isolate the
faulted section of power system .
 PHASOR ESTIMATION
• In the phasor estimation process, the desired frequency component of the signal
received is converted to a representative phasor.
• This process is called estimation because the true value of the desired component is
not known upfront; the quality of the phasor estimated depends on the method used.
• All the methods which attempt to estimate the parameters of the signal are based on
some assumptions and accuracy of phasor estimated depends on method used.
• Typically, the phasor estimation processes a time window of data to obtain the
desired phasor using the method selected.
• The data window is continuously updated including newer samples and discarding
older samples, the phasor estimation is performed for every new data window to
obtain updated phasors.
• An example for a fixed data window size of N = 8 samples is shown in Figure
below.
• The distance between samples in the time scale is the sampling period ∆t.
• The sampling frequency is FS = 1/∆t.
 PHASOR ESTIMATION
An example for a fixed data window size of N = 8 samples is shown in Figure below.
The distance between samples in the time scale is the sampling period ∆t.
The sampling frequency is FS = 1/∆t.
 PHASOR ESTIMATION
• For most relaying algorithms, the digital filters extract the fundamental frequency signal.
• This needs to be done quickly and accurately.
• The fault signal is normally composed of fundamental frequency component, harmonics and a
decaying dc component.
• The task of the digital filters is to extract only the fundamental frequency signal by filtering all
other unwanted signals.
• The time constant and amplitude of the decaying dc are unknown and are associated with the
fault resistance, fault position and the fault beginning time.
• The time constant of the decay is generally determined by the X/R ratio (inductive reactance to
the resistance ratio) of the system.
• For high resistance earth faults the decay rates are very high, sometimes less than half the
power system frequency cycle. Decaying dc is a non-periodic signal and its frequency
spectrum encompasses all frequencies.
• The decaying dc component therefore, seriously affects the accuracy and convergence of
digital filter algorithms such as Fourier, cosine, Walsh, Kalman and least-error-squares (LES)
filters .
• Due to this, the estimated pharos using these algorithms contain errors if decaying dc is
present in the input signal.
• Algorithms are programs used in microprocessors that manipulate the samples of voltages and
currents to produce parameters of interest.
 Phasor based models
• Most of the existing algorithms proposed for use in numerical relays can be
grouped in two categories.
• The first type is based on a model of the waveform itself.
• Phasor based models were the first to be widely used by industry and academics to
design relays and check their performance.
• The parameters of interest for the relaying application are contained in the
waveform description.
• Application of the waveform-model algorithms includes the following processes.
• The peak value of sinusoidal current for over current protection
• The fundamental frequency voltage and current phasors for distance relaying
• The magnitude of harmonics in waveforms of currents for harmonic restrain in
transformer protection
• The fundamental frequency of a periodic signal for frequency relays.
 Phasor based models

• The information necessary for waveform algorithms is taken from sampling the
signals at equal intervals over a pre-specified time period usually referred to as a
data window.
• After the required parameters are calculated, a new sample is incorporated to the
data window, and the oldest sample is discarded.
• The primary limitation of most phasor models is their inability to handle time-relate
consequences affecting fundamental frequency phasors.
• These time related consequences include the effects of DC offset, nonlinearities of
CTs and CVTs, and protective relay memory circuits losing stored voltage or
current data.
 Transient models

• The second type involves a model of the element being protected, such a
transmission line or a power transformer ---Transient models of relays
• These models take into account the presence of high frequency and DC
components in the relay inputs.
• Because the objectives for developing different transient models are not identical,
the complexity of the models varies substantially.
• It is also important to realize that it is easier to develop transient models for
computer-based relays than for electromechanical and solid-state relays.
 Transient models

• Some of the situations which require transient models for proper evaluation are
1) transformer or capacitor inrush,
2) CT or CVT transient, CT steady state saturation,
3) presence of harmonics,
4) presence of transient DC offset,
5) evolving faults,
6) power system swings and dynamics of rotating machines,
7) time varying machine impedances (from sub-transient through transient to steady
state),
8)one line to ground faults on resonant grounded systems,
9) and series capacitors and their protection
 Fundamental criteria describing requirements and
performance of digital filters applied in protections
• Accuracy in transmission of useful signal components and efficient rejection of
noise,
• Fast stabilizing of output signal after step change of input signal,
• Linear phase shift frequency response,
• Minimum burden during digital realization of given filter
• Some of the requirements are quite contradictory and that is why design of a
digital filter is a matter of compromise.
 Digital filter
• All analog filters have infinite impulse response
• Digital filters, on the other hand, can be divided for:
• filters similar to analog ones having infinite impulse response called IIR
• filters, (sometimes also called recursive filters)
• filters having finite impulse response called FIR filters (sometimes also called non-
recursive filters)
• realized in digital technique only
• For such filters one introduces the so called ‘‘window’’ of coefficients having a
length equivalent to N samples used in weighting sum
Digital filters
 Digital Relay Algorithms for Signal Estimation

• Phasor estimation algorithms can be divided into two categories:

• the full cycle algorithms and the half-cycle algorithms.
• The full cycle algorithms typically take one power system cycle to estimate the
voltage or current phasor.
• These algorithms may not be suitable for use in EHV/UHV system applications
where the decisions have to be taken in less than one cycle.
• Since the response speed is of paramount importance, half-cycle algorithms or
short window algorithms have been proposed whose convergence speed to the final
value are faster than the full cycle algorithms
Digital Relay Algorithms for Signal Estimation
• Algorithms can be divided in the following categories

Non-recursive
• These filters use finite length windows, and their outputs essentially depend
on the data contained within this window.
• Short window algorithms
• The short window technique computes the voltage and current phasors from
sampled data and then estimates the impedance as seen from the relay
locations.
• Short window algorithms can be adversely affected by the presence of
nonfundamental frequencies in the system voltages and currents.
• Miki and Mikano
• Mann and Morrison
 Digital Relay Algorithms for Signal Estimation

Long window algorithm

• Discrete Fourier Transform
• FFT
• Walsh Functions
• Least Square Error
• The long window algorithms, if designed properly, can adequately suppress the
effects of the presence of no fundamental frequencies in the fault waveforms.

• Recursive
• Kalman Filtering (state-space model)
• Recursive Least Square Error
 Non-recursive short window techniques
• Short window techniques make the assumption that the signals are sinusoids of the
nominal frequency, and that the system frequency is invariant.
• Depending on the technique, only two or three samples are necessary to estimate
phasors with short window algorithms.
• A phasor is a representation of a sinusoidal voltage or current of the nominal
frequency, f0, and its positive going zero crossing is θ radians ahead of the time
equal to zero.
• The mathematical representation of a phasor is as follows.

The magnitude and phase of the phasor can be calculated using the real and the
imaginary parts of the phasor as follows.
 Least Square Method for Estimation of
Phasors
• Two sample estimation
• In a numerical relaying setup,voltage and current signals would be sampled at
appropriate frequency and acquired by a micro processor or a DSP.
• For relaying decision making we need to estimate the voltage and current phasors.
• For simplicity, imagine a single phase circuit as shown and also assume that the

frequency of the supply is known .

• Figure shown here also introduces the concept of data window for estimation.
• This window contains the ‘active’ set of samples which are currently being processed for
phasor estimation.
• In the present case, we say that we are using a 2-sample window. Each consecutive window,
differs from the previous window by adding a new sample and by removing the oldest active
sample.
Result of estimation in presence of noise
• In real life, the voltage signal will not
be a perfect sinusoid.
• Further, transducers like voltage
transformer, A/D converter etc.
introduce inaccuracies which we can
model as noise.
• The noise has a zero mean, and it's
standard deviation measures the
accuracy of the meter.
• Typically, noise is modeled by zero
mean Gaussian distribution.
• Consequently, if a 0-100V voltmeter
has a standard deviation of 1%, then
it implies that a measurement of a
signal having magnitude of 100V will
be measured any where between 97-
103V, 99% of time.
Table 1 : Effect of Gaussian Noise on Estimation of Phasors using 2-Sample Window
Fig
• Figure shows the plot of estimated value of Vm as a function of sample number for
E = 0.5.
• From figure and table shown before, it is evident that As magnitude of zero mean
noise increases, the standard deviation associated with magnitude increase.
• Mean of Vm is nearly 10.
• Actual estimates seldom match with 10V.

• Summary
• Two sample technique utilizes minimum number of measurements to estimate the
phasor (Vm; Øv ).
• With bare minimum number of measurements, the noise affects the accuracy.
• Noise has to be filtered out to estimate Vm and v .
• To filter out noise, we need to consider redundant measurement.
• Redundancy in measurement is defined as ratio of actual number of measurement
used for estimation to minimum number of measurement required for estimation.
Three sample technique
• Two sample technique does not have
redundancy.
• Three sample technique has a redundancy of
1.5.
• Three sample technique estimates phasors
better than two sample technique.
 DFT

• The Discrete Fourier Transform is a discrete time version of the Fourier Transform. The
result of this algorithm is referred to as the phasor quantity.
• The equation for the DFT is:

• where n is the harmonic number,

• k is the sample, N is the number of samples per cycle, and j means it is an imaginary
number.
• Equation (1) can be simplified to:

• The magnitude of the DFT is found by squaring the real part,

• squaring the imaginary part, summing the squared values and taking the square root.
• The angle of the phasor is found by taking the arc tangent of the imaginary part over the
real part.
• At 5 amp input at 60 Hz is sampled at 4 samples per cycle (N = 4).
• We set n=l (to measure the fundamental), the samples k are then 0 through 3, as shown
in Fig.
The magnitude of the current is then 5 amps and the angle is -90 degrees.

The -90 degrees is not intuitively obvious, but results from the fact that the phasor
equation assumes the pharos quantities are based on cosines, and our waveform was a
sine starting at 0 degrees or a cosine starting at -90 degrees.
 RMS
• Root Mean Square is a method of calculating the total energy content of a waveform.
• The RMS algorithm does not differentiate as to the frequencies being measured and will thus
include all frequency components inclusive of DC.
• The RMS equation is:

For a digital system, (5) is converted to the discrete time equation:

The RMS algorithm yields the same

Inserting N equals 4 into (6) and expanding yields: result as the DFT when
the input is purely fundamental.

Inserting the samples from the above example into (7) yields:
• The DFT is a band pass filter around the harmonic being measured and will thus reject
harmonics where the RMS calculation will not.
• This is not easily demonstrated with a 4 point DFT since all harmonics alias back into the
fundamental.
• It can be demonstrated by adding a DC component to a 5 amp sine wave sampled at 4 samples
per cycle since a 4 point DFT does reject the DC component.
• Let's compare the 2 algorithms.
• For a 5 amp waveform with a 1 amp DC offset (not realistic,
• but for illustrative purposes only) the samples would then be
Equation (10) illustrates that the RMS algorithm does not reject any frequencies
(inclusive of DC), where as the DFT rejects harmonics as long as proper anti-aliasing
techniques are utilized.
• The Discrete Fourier Transform can extract any frequency from a signal.
• In protective relaying the extracted frequency is usually the fundamental though
there are occasions to measure the DC content or other harmonics.
• Since the DFT is capable of rejecting everything except the frequency being
measured, it has a much better response to transient overshoot than the RMS
algorithm.
• Since the RMS algorithm responds to all frequencies, the dc offset is added to the
measurement causing a large overshoot.
• Phasor estimation algorithms can be divided into two categories:
• The full cycle algorithms and the half-cycle algorithms.
• The full cycle algorithms typically take one power system cycle to estimate the
voltage or current pharos.
• These algorithms may not be suitable for use in EHV/UHV system applications
where the decisions have to be taken in less than one cycle.
• The most popular of the full cycle algorithms are the DFT algorithm and the LES
algorithm. .
• Full Cycle DFT can easily remove DC and integer harmonics and compute the
fundamental frequency phasor using one cycle samples or so.
• Since the response speed is of paramount importance, half-cycle algorithms or
short window algorithms have been proposed whose convergence speed to the
final value are faster than the full cycle algorithms
• The two such important half-cycle algorithms are the digital mimic filter plus
halfcycle DFT and half-cycle LES algorithm.
• These half-cycle algorithms suffer from some major disadvantages.
• The half-cycle DFT filters cannot eliminate even numbered harmonics and the
decaying dc present during fault conditions, hence affecting the accuracy and
convergence.
• This leads to a large amount of error in the estimated phasor of the current or
voltage signal if those harmonics are present in the input signal.
• To make these filters insensitive to the decaying dc component, the input signals
are pre-processed by a digital mimic filter before being processed by the half-cycle
DFT filter.
• Mimic filter removes the decaying dc if the time constant of the decaying dc is
known before hand, which is usually not the case in a real power system.
• In addition to that, the mimic filter is susceptible to noise.
• In the half-cycle least error squares technique, the decaying dc component is
included in its model.
• The decaying dc component can be mathematically expressed by a Taylor series
expansion.
• In the LES technique, the first two terms of the Taylor series expansion are used to
model the decaying dc component.
• The least error squares technique is then applied to estimate the fundamental
frequency phasor and other harmonics, provided they are modeled in the technique.
 Full Cycle Fourier Algorithm
• So far we have used number of sample points required in estimation method to define the
length of data window.
• Alternatively, length of data window can be characterized by it's time span.
• For example, for a 3-sample data window, time span of data window is , thus, higher the
sampling frequency, smaller the time span.
• We now consider the case when length of the data window is one cycle, though we have a
freedom to choose number of samples in a window subject to the constraint N > 2.

• Data window length characterized by:

• 1 Number of sample points
• 2 Time span of the window
• For example, a 3-sample data window spans

• Full cycle Fourier: Data window spans one cycle subject

• to N > 2

• Let the sampling frequency be such that (K>2) K samples be acquired in a cycle.
• For example, for the first cycle (samples 0, 1, 2……K-1), LS estimation model with K
samples per cycle in the data window is given by follows eqn.
Comparison of the Estimation Algorithms
• Table 1 illustrates the results of the estimation when full cycle data window is used
• . It can be seen that standard deviation associated with measurement reduces even
further to 1.3176 for E = 3.
• This should be contrasted with 3-sample data window where corresponding was
2.7638.
• This brings out an important aspect of relaying discussed earlier that accuracy of
estimation is improved by increasing the length of data window.

• accuracy versus speed

• for full Fourier improved accuracy
• 2, 3 sample algorithms faster performance
 Example 1: Harmonic and Noise Filtering Capability of
the Full Cycle Algorithm

• In this example, we evaluate the capability of full cycle Fourier algorithm to filter
out harmonics. Input signal corresponds to a 50 Hz square wave shown below. The
harmonic spectrum of such wave form is given by
 Illustration of the Harmonic and Noise Filtering
Capability

• This signal is sampled at a rate of 10 samples per cycle and full cycle Fourier
method is applied to estimate the fundamental.
• In addition noise is introduced using random number generator.
• The true value of fundamental component is = π /4*10 =12.7324.
 Half Cycle Fourier Algorithm
Example : 2
• To improve speed, we can even restrict the data window to half a cycle.
• When this is done, we get half cycle Fourier algorithm.
• With K number of samples per half cycle, the relevant equations are given by

Window length: Half cycle

Faster Estimation
• Notice that our convention is that the latest sample corresponds to the window
number.
• Therefore, first K - window are incomplete because K - samples are not available
with them.
• To complete the incomplete windows, adequate number of zeros are padded in
the beginning.
• Correct estimates are obtained only after L ≥K .
• The table in the next slide compares the performance of the half cycle Fourier
algorithm with the 2- and 3-point algorithms
Table summarizes the performance of half cycle algorithm for the standard sinusoidal signal
used in all our examples.
In presence of harmonics, it can be shown that the accuracy of the algorithm is not as good
as full cycle algorithm.
 Half Cycle vs. Full Cycle Fourier Calculation
• The Discrete Fourier Transform has the capability of working on different
sized "windows".
• TheFull Cycle window generates the sums using all the sampled data
collected over the last cycle.
• This means that the "window" includes the last full cycle's worth of data.
• The Half Cycle window generates the sums using the sampled data
collected in the last half cycle.
• Therefore, the data "window" is a half cycle.
• Using a Half Cycle window allows the Discrete Fourier Transform to
more quickly track a change in the sampled data than is possible with a
Full Cycle window.
• However, there are differences in the filtering actions of the Half and Full
Cycle filters.
• For example, the Half Cycle Fourier is subject to errors due to dc offset
and even harmonics of the fundamental frequency.
• Both the Full and Half Cycle Fourier may be implemented as either a
Recursive or Non-recursive filter.
 Issues Related to Fault Current Estimation

•Fig shows pre-fault to post-fault current

waveform.
•A 3-sample full cycle data window is
considered.
• The window W1 contains only pre-fault data.
•Thus it can be used to correctly estimate the
pre-fault current.
•The first post-fault sample is seen in data
window W2.
• Window W2 contains one post-fault current
sample and two pre-fault current samples.
•Hence it does not correspond to either pre
fault or post fault phasor.
•Hence, it's estimation is completely
erroneous.
• When, we reach window W4, we find that it is
Data window during fault populated completely with post fault data.
•Consequently, it's pharos estimated
corresponds to the post fault pharos.
• Thus, the delay introduced in measuring post-fault signal is
equal to the length of data window.
• Thus, 1 cycle data window introduces a delay of 1 cycle in
estimation.
• It is likely that CT may be driven into saturation by DC offset
current.
• While half cycle window will reduce accuracy of estimation,
with it's use one can strike a compromise between the problem
of CT saturation and improving accuracy of estimation.

The next example considers the effect of delaying DC offset

current of the fundamental on estimation.
Example 3: Comparison of DC Filtering by the Estimation Algorithms
Consider,

The figure in the following slide shows the estimated magnitude of Im, measured for 5-
fundamental cycles, using the 2-point, 3-point, half-cycle and full-cycle Fourier
algorithm.
• Fig show the estimated magnitude of Im, measured for 5-fundamental cycles using
2-point, 3-point cycle and full cycle Fourier algorithms. I
• t can be seen that, significant errors are seen in all estimation methods.
• Also, accuracy of full cycle fourier algorithm is seen to be the most accurate
algorithm.
• The reason is quite obvious.
• Even if we view DC offset current as noise, it is apparent that it does not have a
zero mean.
• Thus, least square based estimation algorithms are expected to fail under such
situations.
• Use some other filter for the DC offset current: mimic impedance.
 Mimic impedance :

• Mimic impedance is an impedance whose X/R ratio is identical to the X/R ratio
of transmission lines.
• In that sense, it mimics a transmission line.
• Fig shows a current source having sinusoidal component and dc offset current
connected to the R +j X impedance.
• The sinusoidal voltage developed across the impedance is given by
•This is the sinusoidal-steady response
•The current is scaled by magnitude of mimic impedance and in phase by θ
•Thus by an inverse operation, we get back the sinusoidal current waveform devoid of dc
offset component.
•Filtering algorithm discussed earlier will then give satisfactory results.
•Mimic impedances are routinely used in distance relays used for transmission line relaying
where the problem of decaying dc offset is most serious.
•Mimic impedance can also be implemented in software.
 Frequency response of Estimation
algorithms
• By now we have deduced that:
• 1. Full cycle fourier algorithm gives the best performance in filtering harmonics
and noise.
• 2. Half cycle does improve speed of response at the cost of accuracy.
• 3. Three sample algorithm is quite fast but the accuracy of estimation is poor.

• Any of the above estimation algorithms can be viewed as a digital filter whose job
is to extract fundamental in presence of harmonics and noise.
• The presentation so far was biased towards elimination of noise.
• Filtering of harmonic can be discussed more neatly by evaluating the frequency
response of the estimation algorithms.
•Input to the filter is stream of samples at frequency , mf0, m = (0; ±1; ± 2···· )
•Since, in relaying we are primarily interested in extracting the fundamental
component.
•The output of the estimation algorithm is viewed by the relay logic as the
fundamental component of the signal.
•Thus, if ,m= ±1 the output should follow input.
•On the other hand, if ,m ≠ ±1, ideally, the output should be zero.
• The frequency response can be evaluated by analytical tools.
• However, to simplify presentation, we restrict the treatment to experimental (by
simulation) evaluation of the frequency response.
• The frequency response for 3-sample, half cycle and full cycle algorithms are
shown in fig
 Salient observations
• Full-cycle algorithm: rejects DC component and all harmonics efficiently
• Half-cycle algorithm: rejects odd but not even harmonics efficiently
• 3-sample algorithm: poor harmonic rejection
• A characteristic frequencies are wrongly interpreted by all algorithms as
fundamental.
• In fact, the full cycle Fourier algorithm is identical to DFT.
 Transmission Line Model
• By assuming that the transmission line to which the relay is connected is
composed of a series resistance and inductance, the fundamental equation
is:

• where, R and L are the resistance and reactance of the fault loop (up to the
fault point respectively).
• Any sampled voltage and current signals taken at any time is considered to
obey above equation.
• To solve the equation and calculate R and L, two equations are required.
• This can be achieved by measuring v(t), i(t) and di/dt at two different
instants of time
By solving equations (10) & (11), R and L may obtained from the following matrix

where D is matrix determinant. The derivative of the current may be calculated

from difference formula,

It is obvious how sampled voltage and current signals can be combined to form the
resistance and inductance of the fault loop.
• The Kalman filter is a recursive filter whose output depends on the present inputs as well as
on all previous inputs.
• Effectively, the weight assigned to the latest input is maximum whereas the assigned weight
decreases as the input becomes older.
• The Kalman filter differs from the other filtering algorithms in that its gain coefficients vary
with time (the gains are nonstationary).
• The recursive form of the discrete fourier transform (DFT) on the other hand is simple and
needs fewer computations, but the presence of an exponentially decaying dc component in a
signal adversely affects the phasor estimates.