In this paper, we present a comparison among three algorithms for estimating positive sequence synchrophasor, frequency and ROCOF starting from a three-phase signal. The considered techniques are the SV filtering algorithm (SV-F) [
24], the SV Taylor-Fourier algorithm (SV-TF) [
26] and the SV IpDFT algorithm (SV-IpDFT) [
25]. All of them feature high flexibility: their parameters can be tuned in order to meet specific requirements in terms of accuracy and latency according to the application. In this respect, we focus on fast response, fixing the algorithm latency at about one and half nominal cycles. The P-class algorithm suggested by the IEC Std, which is used as reference to compare the performance obtained with the three aforementioned algorithms, has a different latency, but always below two cycles. In the following, the common ground and useful definitions are firstly introduced, then all the considered algorithms are explained in detail.
2.1. Common Background on Synchrophasor Measurements
Let us consider a three-phase power system characterized by the rated frequency
, corresponding to the angular frequency
. The synchrophasor approach is based on the following quasi-steady-state model of the generic phase
p signal
(with
), which may represent the voltage or current waveform measured in a node as a function of a shared time coordinate
t:
Basically, the generic phase
p waveform consists of a main term
, which is assumed to be a modulated sinewave whose peak amplitude
and phase angle
are slowly-varying with respect to the rated frequency.
represents a second contribution containing the other components that may be present in the signal, which are considered as disturbances (e.g., harmonics and interharmonics); its magnitude is supposed to be considerably lower than that of
, while its spectral content near
is assumed to be negligible. Starting from these considerations, the corresponding phase
p (dynamic) synchrophasor
is defined as:
In the following, overbar is used to highlight complex-valued quantities; for a lighter notation, a previously defined complex variable written without overbar denotes its magnitude. From an operative point of view, the synchrophasor is a possibly time-varying complex-valued number associated to an electric quantity of an AC power system that, under the previous assumptions, is able to carry its most relevant informative content. Using the Euler’s formula, an expression of
where the synchrophasor explicitly appears can be easily obtained:
where
denotes the complex conjugate operator. From another point of view,
has been decomposed into the sum of two counter-rotating vectors in the complex plane: one with positive, the other with negative angular speed, which is often known as image. It is worth noting that when assuming sinusoidal steady-state conditions at the rated frequency, the synchrophasor corresponds to the usual phasor. Considering once again the waveform model (
1), the frequency
(corresponding to the angular frequency
) and its rate of change ROCOF
are defined as:
is also introduced: it represents the frequency deviation with respect to rated conditions, which is the rotational speed of the synchrophasor in the complex plane divided by
. The target of a PMU is sampling the three-phase waveform of an electrical quantity and, by means of a proper algorithm, extracting the corresponding synchrophasors, frequency and ROCOF at discrete and fixed time instants
(with
i integer, see [
7]), hence multiples of the reporting interval
. Since
is generally an integer multiple of
,
when phase angle wrapping is considered. The algorithm should be able to track dynamic variations as well as rejecting disturbances.
The study of three-phase systems is more efficient when it is performed by using the symmetrical components. Having available the phase
a,
b and
c synchrophasors, the corresponding positive, negative and zero sequence synchrophasors (
,
and
, respectively) can be obtained as follows by means of the Fortescue transformation (here the unitary formulation is considered):
where
. Among the symmetrical components, the positive sequence term has by far the highest magnitude, since three-phase systems are weakly unbalanced during regular operation, in particular as far as the transmission grid is considered; for this reason, many applications rely on a positive sequence representation of the power system.
It is worth noting that processing the single-phase signals while using the definition (
4) may lead to different frequency measurements for each of the phases; conversely, as from [
7], a unique frequency (and ROCOF) value should be provided for each three-phase quantity. On the one hand, this can be obtained through additional processing (e.g., averaging) on the three frequency estimates [
27]; on the other hand, computing a shared frequency value reminds that, from a physical point of view, a three-phase quantity is not a mere set of three independent single-phase quantities. In this regard, a combined processing of the three signals would enable a direct estimate of frequency and ROCOF other than the symmetrical components of the synchrophasors, and in particular of the positive sequence term. Moreover, native three-phase PMU algorithms may exploit that, under regular operation, three-phase quantities are weakly unbalanced: this may be beneficial to improve the quality of the estimates while reducing the computational cost.
2.2. IEC/IEEE 60255-118-1 P-Class Reference Algorithm
The IEC Std suggests a measurement model for synchrophasor estimation and two PMU algorithms, one for class P, intended for applications requiring fast response, and another one for class M, designed for measurement applications. Both of them are based on the same concept of quadrature demodulation, which is briefly summarized in the following. Let us substitute (
3) in the decomposition (
1) of the phase
p waveform
and let us multiply by
, namely a unitary magnitude vector whose rotational speed in the complex plane corresponds to the rated angular frequency of the power system. The signal
is thus obtained, which results:
From the previous expression,
is a complex-valued signal consisting of three contributions. Except for a constant scale factor equal to
, the first term coincides with the quantity of interest, i.e., the synchrophasor, that—based on the assumptions introduced in
Section 2.1—is expected to have a spectral content restricted at very low frequencies. The second term is generated by the counter-rotating image component in (
3), and is represented by the same scale factor multiplied by the synchrophasor conjugate and an unitary vector, that is rotating clockwise in the complex plane with angular speed equal to
; therefore, its frequency content is confined around
. The last term is the disturbance multiplied by an unitary vector rotating in the complex plane with angular speed equal to
; reminding the hypothesis about
, the spectral content of this third contribution is virtually entirely located away from zero. Thanks to the spectral separation of these terms, the synchrophasor can be extracted from
through proper low-pass filtering. In this respect, it should be stressed that it is particularly important to somehow attenuate the second contribution (namely that involving the conjugate of the synchrophasor): it represents a very large unwanted component, since it has the same magnitude as the low-frequency term to be extracted.
After having explained the basic concepts, we focus on the details of the P-class algorithm reported in [
7] (from here on IEC-P algorithm for the sake of brevity), which will be considered as a benchmark for the comparison among the different low-latency techniques; its block diagram is reported in
Figure 1.
Let us suppose that the phase quantities have been sampled with rate
(the positive integer
is the number of samples per nominal cycle) corresponding to the sampling interval
; it is also assumed that
, with
Q positive integer. For each phase
p, a first estimate
of the synchrophasor in the generic reporting instant
is obtained by means of the following expression, which corresponds to multiplication by the rotating exponential and finite impulse response (FIR) filtering:
where * indicates the convolution sum; from here on, hat is used to denote estimated values. Filter coefficients
correspond to the
sample triangular window:
and the constant
G is its DC gain:
It is worth noting that the low-pass filter defined by (
9) is linear-phase, non causal, with zero group delay. Let us assume sinusoidal steady-state operation with frequency equal to
:
in (
7) is constant, while the term involving
rotates with angular speed
. It can be easily proven that the employed filter has frequency response zeros located at multiples of
. Hence, under these conditions it is able to completely cancel out the term rotating at
. Since the DC gain of the triangular window FIR filter defined as in (
9) is
, the estimate provided by (
8) is theoretically exact.
In this algorithm, a three-phase definition of frequency is employed in order to extract a unique value from the three phases. In particular, the frequency deviation
corresponds to the rotational speed (in hertz) of the positive sequence synchrophasor, whose estimate
is obtained by applying the unitary Fortescue transformation (
6):
From a physical point of view, this definition of frequency corresponds to the electrical speed of the air gap magnetic field in an ideal, symmetric three-phase machine whose winding magnetomotive force is purely positive sequence and expressed by the synchrophasor .
Frequency deviation and ROCOF measurements are obtained as first and second order discrete-time derivatives of the estimated phase angle
. For this purpose, let us firstly compute
in the time instants
and
, where
is a positive integer; after that, frequency and ROCOF estimates result:
Therefore, (
12) and (
13) correspond to the application of two linear-phase FIR filters
and
to
, respectively. Now, let us suppose that the input signals
are pure sinewaves whose frequency
f is different from
. In this case, the phase
p synchrophasor is:
where
represents the usual phasor at the frequency
f. Hence, the synchrophasor is a complex number having the same magnitude as the phasor, but its phase angle is time-varying, since it rotates in the complex plane with angular speed equal to the difference between
and
. In particular, the real and imaginary parts of
are quadrature sinewaves whose frequencies are equal to the frequency deviation
. From (
7), the signal to be filtered is made of two contributions:
Introducing
as the frequency response of the previously defined triangular window filter (real-valued and identical to the amplitude response), the evaluated synchrophasor results:
reminding that
. The first thing to be noticed is that the term proportional to
is attenuated but not completely rejected by the filter, since its rotational speed does not correspond to a zero of the filter. Furthermore, even neglecting this effect, the magnitude of the synchrophasor estimate is biased: in fact
when
. This effect is significant even for small values of the frequency deviation. However, having measured
, it can be compensated since the filter response is known; a more accurate synchrophasor estimate
can thus be obtained. In this respect, ref. [
7] adopts the following approximated expression instead of using the exact filter response:
As mentioned before, the positive sequence synchrophasor estimate has key importance in power systems. Let us obtain its expression under off-nominal frequency conditions; the substitution of (
16) into (
11) leads to:
As in (
16), also
is disturbed by the presence of a component which rotates clockwise in the complex plane with angular speed
. Based on the assumption of weakly unbalanced system, such disturbance has a much lower relative magnitude, since
) is close to
G, whereas
(often called unbalance factor) is well below one. Therefore, the positive sequence synchrophasor estimate is considerably less affected by this phenomenon. Finally, it is worth stressing that
does not depend on the zero sequence component.
The overall algorithm latency is given by , which keeps into account the filtering process and the phase-angle derivative computation. It is thus always , thus in order to comply with the latency limit for P class algorithms.
2.3. Space Vector Algorithm
PMU algorithms based on the SV transformation of the phase quantities and filtering of the resulting signal when decomposed in rectangular and polar coordinates have been recently proposed by the authors in [
23,
24]. The starting point is the usual measurement model (
1), here rewritten by using the vector notation in order to consider the three phases simultaneously:
where:
From the three-phase waveforms, it is possible to compute the SV
in a rotating reference frame whose instantaneous angular position is
; it results:
where
is the SV in a stationary reference frame characterized by
. Substituting (
19) into (
21) while using (
3) and reminding (
6) leads to:
Three terms appear: the first one is related to the positive sequence synchrophasor, the second depends on the negative sequence synchrophasor, the last is produced by disturbances. Of course, the term due to the positive sequence synchrophasor is expected to be considerably higher with respect to the others. Furthermore, it is worth noting that the zero sequence synchrophasor does not affect the SV signal. Now, let us choose , namely performing the SV transformation in a reference frame whose angular speed corresponds to the rated angular frequency. In this case, the positive sequence synchrophasor produces a very low frequency contribution; instead, the term related to the negative sequence synchrophasor rotates with an angular speed close to in clockwise direction. Reminding that the spectral content of is assumed to be located away from , the spectrum of is well separated from that of the positive sequence synchrophasor. This suggests that an estimate of can be extracted from by means of proper low-pass filtering, without explicitly computing per phase synchrophasors. Adopting the same three-phase definition of frequency deviation already employed by the IEC-P algorithm, and ROCOF represent (except for a scale factor ) the first and second order derivatives of the positive sequence synchrophasor phase angle. Therefore, their estimates can be obtained by filtering , computing its phase angle and performing numerical differentiations.
The architecture of this approach corresponds to the block diagram reported in
Figure 2; its practical implementation requires first of all to properly sample the phase signals with rate
. From the acquired data, applying (
21) with
the samples of the SV signal
are obtained. A first low-pass filtering stage
H with unit DC gain is applied to the real and imaginary parts of
in order to remove most of
as well as the term produced by the negative sequence. Both FIR and infinite impulse response (IIR) filters can be employed, even if best performance and greater flexibility is obtained with FIR design. The samples of
are thus obtained and they have to be further processed in order to compensate for the group and phase delay introduced by
H; it is extremely simple when a linear-phase FIR filter is adopted.
After that, is decomposed into its magnitude and argument . The estimate of the positive sequence synchrophasor phase angle in the generic reporting instant is obtained by applying to a linear-phase FIR low-pass filter P (characterized by unit DC gain, designed to attenuate the residual impact of and of the oscillating term produced by ) and compensating its delay. Its bandwidth should be sufficient to properly follow the phase angle dynamics, which are typically rather slow.
Similarly, frequency deviation and ROCOF measurements in the reporting instants ( and , ) are computed by filtering with linear-phase, partial band first and second-order FIR differentiators (F and R respectively) and compensating the introduced delays. Their frequency responses should match those of ideal first and second order differentiators only near zero (just slow frequency variations have to be accurately tracked), while they should have considerably lower magnitudes at higher frequencies in order to achieve good disturbance rejection and noise immunity.
An estimate of the positive sequence synchrophasor magnitude can be obtained by applying a low-pass filter M to while taking into account the introduced delay. Similarly to P, M should have unitary DC gain and should be designed to reject the infiltration of as well as the term proportional to . Its bandwidth should be large enough to properly follow amplitude modulations, which are fairly slow in practical power systems.
However, it should be noticed that
may be significantly biased under off-nominal frequency conditions, similarly to what happens with the IEC-P algorithm discussed in the previous subsection. Assuming perfectly sinusoidal, positive sequence input with angular frequency
,
results:
where
is the positive sequence phasor while filter
H is assumed to be a linear-phase FIR filter having amplitude response
For a linear-phase filter
H, the relationship between its frequency response
and amplitude response
is:
, where
is the group delay. According to the procedure described in the previous lines, it is easy to derive the magnitude estimate
, reminding that the DC gain of
M is unitary by assumption. Similarly to what explained in
Section 2.2, the effect of filter
H can be removed since its response is known and a measurement of the frequency deviation is available. Hence a better estimate
of the positive sequence synchrophasor magnitude is:
One of the main advantages of this SV-based approach is its flexibility: achieved performance strongly depends on the filters
H,
M,
P,
F and
R, which can be tuned in order to reach predetermined goals in terms of accuracy, latency and responsiveness. It is worth highlighting that closed-form expressions provided in [
24] allow predicting the results of the P- and M-class compliance tests prescribed by [
7], thus substantially helping the design of the aforementioned filters.
The algorithm latency in this case is , where , , , , are the number of taps of the filters H, M, P, F and R, respectively.
2.4. Space Vector Taylor-Fourier Algorithm
In [
13], the TF approach to synchrophasor estimation was introduced. Basically, it is based on a measurement model which is a truncated Taylor expansion of the phase
p synchrophasor around the generic reporting time
in order to model its time evolution. Model parameters are obtained through least-squares fitting of a sliding window of the collected samples, thus corresponding to FIR filtering. Synchrophasor, frequency and ROCOF estimates can be easily obtained from the model parameters. It is worth noting that a frequency estimate for each phase is obtained. Since according to [
7] a unique frequency value for each three-phase quantity must be provided, a possible solution is computing the average between the three different frequency measurements.
In [
26], the technique has been extended for being applied to the SV signal, thus directly estimating positive and negative sequence synchrophasors. This enables higher design flexibility with respect to a per phase approach, which results in better performance and lower computational burden. Furthermore, it directly provides frequency and ROCOF values according to the three-phase definition adopted by the previously presented algorithms.
The block diagram of the method is reported in
Figure 3; the starting point is the SV
on a stationary reference frame, which is obtained by applying the SV transformation (
21) with
to the three-phase signals. Under the usual assumptions, it results:
As mentioned above, the TF approach relies on a measurement model obtained through truncated Taylor expansions of the synchrophasors around the reporting instant. Therefore, considering the positive and negative sequence synchrophasors in the neighborhood of
:
where
and
are the
kth order derivatives of the positive and negative sequence synchrophasor model at the reporting instant
; considering
, they correspond to the two synchrophasors. Expansion orders
and
are in general different: this additional degree of freedom is enabled thanks to the SV-based approach, while it would have not been possible if the conventional implementation were adopted.
The expressions (
26) can be used to write a model for the SV signal in the neighborhood of the reporting instant:
Now, let us suppose that the phase signals have been sampled with rate
; the samples of the SV signal
can be easily computed. Furthermore, let us consider a sliding window made of
samples (with
odd) of
and centered on
; these samples can be arranged in a vector
. Using (
27) it is possible to write the corresponding vector of
samples obtained from the model, having assumed that its parameters (namely the synchrophasor derivatives) are constant within the window. Adopting vector notation, this leads to:
where:
and
is obtained from (
31) by replacing
with
; the superscripts
and
indicate the Hermitian transpose and the transpose operators, respectively.
is the vector of the model parameters in the reporting instant. It can be estimated by minimizing the Euclidean norm of the vector of the differences between the samples of the SV signal and those obtained from the model, hence:
which corresponds to an ordinary least squares (LS) problem whose solution is:
where
is a complex filter bank, namely its
hth row
contains the complex-valued coefficients of the FIR filter that permits obtaining the
hth element of
. In the same fashion,
denotes the vector-valued frequency response of the filter bank; its
hth element is the frequency response of the FIR filter defined by the coefficients
. Therefore, the estimated positive sequence synchrophasor is:
Estimates can also be obtained by using a weighted LS (WLS) method [
28] where the weights are given, for instance, by the squared coefficients of a cosine window. Up to now, the WLS estimator has never been applied in conjunction with the SV-TF approach.
As previously mentioned, the TF expansion applied to the complex-valued SV signal permits selecting different expansion orders for the positive and negative sequence synchrophasor, which is not possible when considering the conventional implementation in the real-valued phase signals: this additional degree of freedom in the design of TF filters enables better performance. In general, increasing the positive sequence synchrophasor expansion order is beneficial, since the model is able to better represent its dynamics. On the contrary, when the focus is measuring the positive sequence synchrophasor, choosing often leads to an overparametrized model. In this case, the best choice is , since it increases the robustness with respect to noise and disturbances (which are bettered filtered thanks to the “stiffer” underlying model), without significant drawbacks.
According to the three-phase definition of frequency deviation and ROCOF, they can be obtained, with the formulas reported in [
13], from the estimated derivatives of the positive sequence synchrophasor at
, that is as:
From the above equations, it is clear that an order is needed to estimate frequency and ROCOF.
The latency of the algorithm depends only on the length of the window and thus, in this case, is .
2.5. Space Vector IpDFT Algorithm
IpDFT algorithms based on proper weighting windows [
29,
30] have been widely used to measure spectral components and frequencies of electrical signals [
31]. More recently, several PMU techniques exploiting this approach have been proposed in the literature [
16,
32,
33]. In order to obtain a synchrophasor estimate, they require observing the phase signal
over a time interval centered around the reporting instant
and corresponding to an integer number
of rated cycles; the sampling frequency
is multiple of the rated frequency
, so that
samples per nominal cycle are collected. The underlying signal model is very similar to (
1), but in this case the amplitude and frequency (defined as in (
4)) are assumed to be constant within the analyzed time interval. Therefore, it results:
The first step of the IpDFT algorithm is applying a proper tapering window to the collected samples. Assuming that spectral interference is negligible, there should be at least two DFT bins produced by the signal component rotating with angular frequency (namely the one related to the synchrophasor to be estimated) which falls under the main lobe of the window. Since its shape is known, the ratio between the magnitudes of these components allows estimating the frequency deviation and, in turns, also the synchrophasor. In particular, the knowledge of allows compensating the effect of scalloping loss which may undermine the evaluated magnitude under off-nominal frequency conditions.
As mentioned above, the method properly works if the two considered DFT bins are not affected by spectral interference: this may be produced by the disturbance
, but also by the image component (namely the counter-rotating term appearing in (
38)): specific care has to be taken, since it has the same magnitude as the term to be evaluated. This effect can be reduced by selecting a proper window (e.g., Rife-Vincent type I, characterized by maximum asymptotic decay of the sidelobes) and increasing the observation interval, even if the latter expedient results in several drawbacks. First of all, algorithm latency and computational burden are increased; secondly, the assumption of having constant signal parameters over a longer time window becomes harder to be met, thus it may seriously jeopardize the achieved dynamic performance.
Having estimated the three synchrophasors of the phase quantities, the positive sequence synchrophasor
is evaluated through the Fortescue transformation (
6). From the previous considerations, the estimates
of the phase synchrophasors may contain disturbances due to the infiltration of the image components. Similarly to what explained in
Section 2.2, this effect is significantly attenuated in the positive sequence synchrophasor measurement
thanks to the weak unbalance level of real-world three-phase quantities. However, explaining how this cancellation occurs is troublesome, since the IpDFT algorithm is inherently nonlinear and closed-form expressions cannot be easily derived. Finally, it is also worth noting that applying the IpDFT algorithm to the phase signals produces three (different) frequency estimates, which have to be processed (averaged) in order to obtain a unique value.
Similarly to the TF approach, the IpDFT algorithm can be favorably applied to the SV signal
in a stationary reference frame, as in the block diagram reported in
Figure 4; this enables a direct estimation of the positive sequence synchrophasor. As in
Section 2.4,
is computed from the phase waveforms by means of the SV transformation (
21) with
. Its model, which also in this case is assumed to be valid over the
C nominal cycle interval centered around the reporting instant, results by applying to (
38) the SV transformation on the same reference frame, hence:
The next step is considering the
samples of the SV signal collected around
and, having applied a smoothing window defined by the sequence
(
), computing the DFT bins
whose corresponding frequencies are close to
, namely whose indexes
k are near to
C. Let us assume that, as typically happens in practical implementations,
and thus also
are even numbers.
The key underlying assumption of IpDFT algorithms is that long-range leakage is negligible for the considered indexes
k. When comparing (
39) with (
38), it is evident that this assumption is more easily met as long as the IpDFT algorithm is applied to the SV signal instead of a phase waveform. In fact, thanks to low unbalance level of power systems, in the first case the disturbance produced by the counter-rotating component is considerably lower in relative value.
Therefore, neglecting spectral interference, it results (
in the following):
where
is the discrete-time Fourier transform of
computed in the generalized bin
, while
. Now, let us suppose to have employed a periodic window having
: it is worth noting that this property holds true for many widely employed windows, even including those belonging to the Rife-Vincent class I family. In this case, it is possible to write:
where
is the amplitude response of the window. Let us assume that
is the closest integer to
, namely the index of the highest DFT component. Writing
(
),
can be estimated from the ratio between the magnitudes of the two largest bins with the following equations:
where
).
The frequency is then estimated as:
Using
, the positive sequence synchrophasor in the reporting instant is obtained through scalloping loss compensation and phase shift of the highest DFT component:
Finally, the ROCOF is obtained by using the previously described algorithm to perform frequency measurements in the time instants
and
while computing a discrete-time derivative, thus applying the linear-phase FIR filter
(introduced in
Section 2.2) to the frequency estimates:
As in
Section 2.2, the algorithm latency is given by half the window length plus the delay required by the discrete-time derivative used to compute ROCOF, that is
.