The present invention relates generally to the field of signal analysis, and more specifically, to a synchrophasor estimation method characterized by a high level of accuracy and a fast response time, designed to analyze typical power grid operating conditions, including the case of signals corrupted by interharmonics, by means of Phasor Measurement Units (PMUs).
Phasor Measurement Units (PMUs) are devices that produce synchronized phasor, frequency and Rate-of-Change-of-Frequency (ROCOF) estimates from voltage and/or current signals and a time synchronizing signal. A phasor calculated from data samples using a standard synchronizing time signal as the reference for the measurement is called synchrophasor. PMUs typically embed (see
PMUs constitute the backbone of wide area monitoring, protection and control systems of modern power grids and provide a tool for system operators and planners to manage a power grid during both normal and fault conditions. PMUs are typically organized in synchrophasor networks (see
The synchrophasor estimation (SE) algorithm of a Phasor Measurement Unit (PMU) must be designed depending on the supplied power grid application [1]. Indeed, PMUs expected to supply protection applications must be characterized by fast response times, whereas PMUs expected to supply monitoring applications must be characterized by highly accurate measurements and resiliency against interfering spectral components. To be more specific, a PMU meant to produce precise measures must be capable to reject interharmonics close to the 50 or 60 Hz main tone [1]. These are defined as signals characterized by an amplitude that is 10% of the main tone and a frequency fi in the bands [10, fn−Fr/2] and [fn+Fr/2,2 fn], being fn the nominal power system frequency, Fr the PMU reporting rate and fi/fn∉ [1].
Recently, the idea of a single PMU capable of satisfying high accuracy and fast response time at the same time has become increasingly popular [2], [3]. The advantages are evident: from a cost perspective, an electrical utility interested to use PMUs to simultaneously supply monitoring and protection functionalities of situational awareness systems does not need to equip every measurement point with two different devices. From a measurement reliability perspective, the performance of fault management systems using PMU data to tune or back-up existing protection schemes, are not degraded by the presence of interharmonic tones [4].
In this respect, [3] proposes a hybrid PMU design based on tunable trigger thresholds to switch between two different frequency-tracking filters. The technique self-switches from a highly accurate algorithm to a fast response one in case a transient event is detected. Similarly, [4] presents and experimentally validates a SE process that implements in parallel two different Taylor Fourier Transform algorithms: the first one produces accurate measurements of steady state signals, while the second one is better suited to follow fast signal changes. In this case, the selection of the most appropriate algorithm is implemented by a detector that identifies the presence of power system transients.
Nevertheless, both approaches lack of a single design of the SE algorithm capable of satisfying both static and dynamic performance requirements. Consequently, the reliability of both approaches is based on the robustness of the adopted switching logic.
Among possible SE techniques, the interpolated discrete Fourier transform (IpDFT) has been widely used, in view of the optimal tradeoff between SE accuracy, response time and computational complexity [5] [9]. The IpDFT enables us to extract the parameters of a sinusoidal waveform by processing the highest DFT bins of the related DFT spectrum. However, the IpDFT results can be significantly corrupted by the spectral interference produced by the negative image of the fundamental tone of the signal or by eventual interfering tones. To cope with such phenomena, several IpDFT-based approaches proposed by the current literature rely on the possibility to compensate for the effects of the spectral interference produced by the negative image component of the fundamental tone [5][9]. Non-IpDFT-based methods have been investigated too, such as sine-fitting techniques [10], compressive sensing [11] and lookup tables [12].
In this regard, in [6] the formulation of an enhanced IpDFT-based SE algorithm that iteratively compensates the effects of the spectral interference produced by the negative image of the main spectrum tone is presented. However, the method presented in [6] does not account for the spectral interference produced by tones other than the fundamental one. As a consequence, it produces incorrect results in presence of interfering tones that are relatively close to the main one. In this respect, in [13] an IpDFT-based method that compensates the spectral interference generated by both the image component of the main tone and harmonic tones is presented. Although the technique proposed in [13] takes into account tones other than the fundamental one, such investigation disregards interharmonic tones.
Thus, there exists a substantial need for a single SE technique capable of accurately estimating the synchrophasors with a fast response time, even when analyzing signals corrupted by interharmonic tones close to the fundamental one.
In a first aspect the invention provides a method for performing synchrophasor estimation of an input signal, whereby the input signal is a sinusoidal power system voltage or current signal, comprising:
In a preferred embodiment, said periodic sampling, comprises:
In a further preferred embodiment, said transforming the discrete time-domain function to the discrete frequency-domain function, comprises:
In a further preferred embodiment, said estimating the instantaneous parameters of a synchrophasor comprises:
In a further preferred embodiment, the enhanced Interpolated DFT (e-IpDFT), comprises:
In a further preferred embodiment, said synchrophasor comprises a complex function characterized by an instantaneous frequency, amplitude and phase corresponding to the instantaneous frequency, amplitude and phase of the main tone of the input signal, referred to a common time reference.
In a second aspect, the invention provides a system for performing synchrophasor measurement comprising:
In a further preferred embodiment, the one or more processing elements comprise a programmable hardware element and/or circuitry.
In a further preferred embodiment, the one or more processing elements comprise a processor and memory.
The invention will be better understood though the detailed description of preferred embodiments and in reference to the drawings, wherein
table 1 contains i-IpDFT parameters;
table 2 relates to compliance with respect to static signals: maximum TVE, FE and RFE for the i-IpDFT and maximum limit allowed by [1]; and
table 3 relates to compliance with respect to dynamic signals: maximum TVE, FE and RFE for the i-IpDFT and maximum limit allowed by [1].
The invention concerns a synchrophasor estimation method characterized by a high level of accuracy and a fast response time, designed to analyze typical power grid operating conditions including the case of signals corrupted by interharmonics. The method is meant for Phasor Measurement Units (PMUs) expected to supply simultaneously protection applications, requiring fast response times, and measurement applications, requiring high accuracy and resiliency against interfering spectral components. The method, called iterative-Interpolated DFT (i-IpDFT), iteratively estimates and compensates the effects of the spectral interference produced by both a generic interfering tone, harmonic or interharmonic, and the negative image of the fundamental tone. The i-IpDFT technique applies for any number of IpDFT interpolation points, windowing function, window length and sampling frequency. The 3-points i-IpDFT technique for cosine and Hanning window functions is analytically presented. A procedure to select the i-IpDFT parameters is included. The performance of the i-IpDFT method is assessed with respect to all the static and dynamic operating conditions defined in the current IEEE Std. C37.118 for PMUs.
The present invention proposes a single SE method characterized by a high level of accuracy and a fast response time, designed to estimate the synchrophasors in typical power grid operating conditions, including the case of signals corrupted by interharmonics close to the main tone.
The method is meant for PMUs specifically designed to supply simultaneously protection applications, requiring fast response times, and measurement applications, requiring high accuracy and resiliency against interfering spectral components.
The technique is hereafter called iterative Interpolated Discrete Fourier Transform (i-IpDFT) and demonstrates how it is possible to develop a DFT-based SE algorithm capable of rejecting interharmonics, also when adopting relatively short window lengths (3 periods of a signal at fn). The i-IpDFT method, that represents an evolution of the one presented in [6], iteratively finds and compensates the effects of the spectral interference produced by both an interfering tone, harmonic or interharmonic, and the negative image of the fundamental tone. The i-IpDFT algorithm can be formulated for any window function, number of IpDFT interpolation points, window length and sampling frequency. In what follows, the i-IpDFT algorithm is analytically formulated for a 3-points IpDFT for the family of cosy window functions, with a particular focus on the cosine and the Hanning (or Hann) window, selected in view of their peculiar characteristics.
The main noticeable advantage of the i-IpDFT algorithm when comparing it with the method presented in [6] and other SE techniques, is that it is capable of correctly estimating the synchrophasors in all typical power system operating conditions without being characterized by a slow response time, even in presence of an interharmonic tone. More specifically, it satisfies all the accuracy requirements defined in the current standard for PMUs, the IEEE Std. C37.118 [1], for both dynamic and static operating conditions, for both M and P performance class PMUs as defined by the IEEE Std. 37.118 [1].
Within this context, Section III reviews the theoretical background about the IpDFT for cosy windows and presents the effects of spectral leakage on the IpDFT performance. Section IV formulates the i-IpDFT SE technique, with a specific focus on the selection of the algorithm parameters. Section V assess the algorithm performance in an offline simulation environment with respect to the all the static and dynamic testing conditions defined in the IEEE Std. C37.118 [1]. The results are presented in the case of a power system operating at a nominal frequency of 50 Hz, a reporting frequency of 50 fps and signals characterized by an SNR of 80 dB.
The IpDFT is a technique that enables us to extract the parameters of a sinusoidal waveform by processing the highest DFT bins of the related DFT spectrum. In particular, the IpDFT enables us to mitigate the effects generated by incoherent sampling by [14], [15]:
A. cosα Window Functions
The IpDFT solution can be analytically derived only for cosα windows (also known as sinα windows), that are defined as follows [16]:
being N the number of samples. In particular,
Generally, IpDFT algorithms were formulated for the so called Rife-Vincent class I windows, i.e., cosα windows characterized by a null or even value for α. The most elementary one is the rectangular (α=0), whose DFT is known as the Dirichlet kernel:
This function is characterized by the narrowest main lobe among the cosα windows but at the same time it exhibits the highest side lobes.
In order to reduce the effects of spectral leakage generated by the side-lobe levels, IpDFT algorithms typically adopt bell-shaped windows, obtained by increasing the value of α in (1). The most common is the Hanning window (α=2) defined as:
w
H(n)=0.5·(1−cos(2πn/N)), n∈[0,N−1] (3)
and characterized by the following DFT:
W
H(k)=−0.25·DN(k−1)+0.5·DN(k)−0.25·DN(k+1), k∈[0,N−1] (4)
that is known for the good trade-off between the main-lobe width and side-lobe levels [15].
More recent studies have derived the analytical solution of the IpDFT problem for cosα windows in the case of an odd value for α [17]. In applications where the main lobe width plays a crucial role in identifying relatively nearby tones, like in SE, the so-called cosine window, defined for α=1, is an extremely attractive solution as it represents a compromise between the rectangular and the Hanning window:
Its DFT can be derived as follows:
W
C(k)=−0.5jDN(k−0.5)+0.5jDN(k+0.5),k∈[0,N−1] (6)
However, for an odd value for α, the window is a weighted sum of Dirichlet kernels modulated by non-integers of the frequency resolution, therefore an intrinsic limitation arises: as it can be seen from
B. The Interpolated DFT (IpDFT)
As known, the IpDFT is based on a static signal model that, in general, is assumed to contain a single unknown frequency component. In this respect, let us consider a finite sequence of N equally spaced samples x(n) obtained by sampling an input continuous signal x(t) with a sampling rate F′S=1/TS, being TS the sampling time:
x(n)=A0 cos(2πf0nTs+φ0),n∈[0,N−1] (7)
where {f0, A0, φ0} are the signal frequency, amplitude and initial phase respectively. The signal is assumed to be windowed with a discrete function of N samples w (n), being T=N TS the window length. As discussed in [18], the mismatch between this signal model and real voltage or current waveforms measured by PMUs is the main source of error of DFT-based SE algorithms.
The DFT X(k) of the windowed input signal is:
where
is the DFT normalization factor, and WNe−j2π/N is the so called twiddle factor. The DFT returns a sequence of N samples (also called bins) of the theoretical Discrete Time Fourier Transform (DTFT) that are uniformly spaced by the DFT frequency resolution Δf=1/T. More specifically, based on the convolution theorem, the DFT of the windowed signal x(n) exhibits a pair of scaled, shifted and rotated versions of the DFT of the window function (see
X(k)=X0+(k)+X0−(k) (9)
where:
X
0
+(k)=A0e+jφ
X
0
−(k)=A0e−jφ
being W(k) the DFT of the adopted window function.
As shown in
f
0=(km+δ)Δf (11)
being −0.5≤δ<0.5 a fractional correction term and km the index of the highest bin. The IpDFT problem lies in finding the correction term δ (and, consequently, the fundamental tone's parameters {f0, A0, φ0}) that better approximates the exact location of the main spectrum tone.
The IpDFT problem solution has been originally provided as a 2-point interpolation scheme [14], [15]. More recently, multipoint interpolation schemes have demonstrated to inherently reduce the long-term spectral leakage effects, leading to more accurate interpolation results [7], [8]. In this respect, for the Hanning window, the fractional term g can be computed by interpolating the 3 highest DFT bins as [8]:
Similarly, for the cosine window, g can be obtained by interpolating the 3 highest DFT bins as [17]:
The fundamental tone's amplitude and phase can then be computed as (computational details are given in Appendix):
C. Spectral Leakage Effects on the IpDFT
The main assumptions behind the formulation of the IpDFT technique are the following [14]:
In order to satisfy the first two assumptions when applying the IpDFT to SE, sampling rates in the order of few kHz and window lengths containing few periods of a signal at the rated power system frequency must be adopted respectively [19]. This choice causes the energy of the DFT spectrum to be concentrated in the lower frequency range and the positive and negative image of the main tone of the spectrum to be relatively close.
In such conditions, in case of incoherent sampling, the tails of the negative image of the spectrum main tone (grey curve in
To cope with these conditions, in [6] a technique that mitigates the effect of the spectral leakage produced by the negative image of the spectrum is presented.
The method is called enhanced-IpDFT (e-IpDFT), and is described by:
The e-IpDFT is described in what follows, by making reference to the lines of Algorithm 1, also shown in
It starts with a preliminary estimation of the main tone parameters obtained by applying the IpDFT to the DFT spectrum X(k) (line 2). These values are used to estimate the main tone's negative image {circumflex over (X)}0−(k) (line 4), as in (10), that is then subtracted from X(k), to return an estimation of the main tone's positive image (line 5). The IpDFT is applied to the resulting spectrum, where the spectral interference produced by the negative image is considerably reduced (line 6). The compensation of the spectral interference produced by the negative image of the fundamental tone, can be improved by iterating the procedure a predefined number of times P (see Section IV.B). We summarize the method proposed in [6] by defining a single function e-IpDFT that accounts for lines 2-7 of Algorithm 1:
{{circumflex over (f)},Â0,{circumflex over (φ)}0}|P=e−IpDFT[X(k)] (18)
The method presented in [6] does not account for the spectral interference produced by tones other than the fundamental one. As a consequence, it produces incorrect results in presence of interfering tones that are relatively close to the main one, such those defined in the OOBI test.
This Section aims at presenting the i-IpDFT, that represents an enhancement of the method proposed in [6] as it takes into account the effects of the spectral interference generated by both the negative image of the main tone and a generic interfering one. More specifically, in Section IV.A we define the SE algorithm and in Section IV.B and IV.C we propose a procedure to tune the algorithm parameters.
A. The i-IpDFT Algorithm Formulation
Let us consider a steady-state discrete signal composed of two tones, a fundamental and an interfering tone (not necessary harmonic, i.e., fi/f0∈□), both unknown:
x(n)=A0 cos(2πf0nTs+φ0)+Ai cos(2πfinTs+φi) (19)
As shown in
X(k)=X0(k)+Xi(k)=X0+(k)+X0+(k)+Xi+(k)+Xi−(k) (20)
The proposed i-IpDFT algorithm iteratively estimates and compensates the effects of spectral leakage generated by an interfering tone and by the negative image of the main tone, such that the IpDFT is applied to a DFT spectrum that is only composed by the positive image of the main tone X0+(k).
The proposed i-IpDFT is described by:
The i-IpDFT is described in what follows, by making reference to the lines of Algorithm 2, also shown in
The first steps (lines 1-2) of the i-IpDFT algorithm exactly correspond to the e-IpDFT technique. Although, in presence of an interfering tone, the estimated main tone parameters might be largely biased, they can be used to approximate the positive and the negative image of the fundamental tone (line 3,
where K is the total number of computed DFT bins. In such a case, the e-IpDFT is applied to X(k)−{circumflex over (X)}0q-1(k), to estimate the parameters {fiq, Âiq, φiq} of the detected interharmonic tone (line 6). The latter, are used to evaluate both the positive and negative image of the interharmonic tone (line 8,
The whole procedure can be iterated a predefined number of times Q, leading to more and more accurate estimates as Q increases (see Section IV.B).
The presented i-IpDFT algorithm can be formulated for any window function, number of IpDFT interpolation points, window length and sampling frequency. Even though it has been formulated for a single interfering component, it can be easily extended to consider more than one interfering component. In this respect, it is worth mentioning that the amount of DFT bins to be calculated at line 1 depends on the highest frequency component that has to be compensated.
B. On the Tuning of the Number of Iterations P and Q
The performance of the i-IpDFT algorithm are mainly influenced by two parameters, and in what follows a procedure to select them is presented:
The effects of P can be evaluated when applying the e-IpDFT technique to a single-tone signal that is incoherently sampled. In this respect,
The effects of the overall number of iterations Q are evaluated when applying the i-IpDFT algorithm to a signal corrupted by an inter-harmonic tone. In particular, the case of a signal characterized by a main tone at 47.5 Hz and an inter-harmonic tone at 20 Hz is presented (similar results hold for all combinations of f0 and fi in the GOBI range). Again, an 80 dB SNR is considered [20].
C. On the Tuning of the Threshold λ
The threshold λ must be set so that the iterative compensation (i.e., lines 5-10 of Algorithm 2) is activated in presence of an inter-harmonic tone. Furthermore, the proposed technique can turn out to be extremely useful to compensate the spectral leakage generated by an harmonic tone when using the cosine window, which has demonstrated to generate spectral leakage also with coherent sampling.
In general, the normalized spectral energy En can contain contributions generated by an interfering tone (e.g., harmonic or interharmonic) as well as any spurious component generated by a dynamic event that is wrongly captured by the DFT (e.g., steps in amplitude and in phase or amplitude and phase modulations). In this respect,
The IEEE Std. C37.118 [1] and its latest amendment [21] have defined two performance classes and related measurement requirements, to which PMUs must comply with: P-class, intended for protection applications, and M-class, for measurement ones. The main difference between P and M-class requirements is represented by the Out-Of-Band Interference (GOBI) test, defined for M-class only, that is meant to assess the PMU capability to reject superposed interharmonics close to the fundamental tone. In this regard, we carry out the numerical validation of the i-IpDFT algorithm in a simulation environment, by making reference to the static and dynamic performance requirements dictated by [1] and following the testing procedures described in the IEEE Guide C37.242 [22]. In order to limit the number of tests, the nominal frequency and the reporting rate have been fixed to 50 Hz and 50 frames-per-second respectively. For each test, the results are presented by means of three graphs showing the maximum Total Vector Error (TVE), Frequency Error (FE) and Rate Of Change Of Frequency (ROCOF) Error (RFE) as a function of the independent variable of the specific test, together with the maximum limit allowed by [1] for both P and M-class PMU. Moreover, two tables summarize the maximum obtained TVE, FE and RFE and the maximum limit allowed by [1] in all tests. Although [1] does not provide any guidelines regarding the noise, additive white Gaussian noise with zero mean and variance corresponding to an 80 dB SNR is added to the various reference signals, in order to simulate more realistic conditions [20].
As resumed in Table 1 the i-IpDFT algorithm results are shown for both the Hanning (solid lines) and cosine windows (dotted lines), using a sampling rate of 50 kHz and a window containing 3 periods of a signal at the nominal power system frequency. The first 10 DFT bins of the spectrum are computed, in order to be able to compensate the effects of any interfering component up to the 3rd harmonic. The 3-points IpDFT is used to estimate the fractional correction term δ.
Finally, the ROCOF is computed by means of a classical backward first-order approximation of a first-order derivative:
ROCOF(n)=|{circumflex over (f)}0(n)−{circumflex over (f)}0(n−1)|·Fr (22)
where {circumflex over (f)}0(n) and {circumflex over (f)}0(n−1) represent the fundamental frequency estimations at two successive reporting times.
A. Static Conditions
Regarding the steady state conditions (see
During the signal frequency test (see
As far as the OOBI test is concerned (see
B. Dynamic Conditions
Regarding the dynamic conditions (see
During the measurement bandwidth test (see
The analytical formulation of the IpDFT correction term δ for the cosine window given in (15) is derived in [17]. Regarding the tone's amplitude given in (16), no derivation was found in the existing literature.
If the number of samples N is sufficiently large, the following approximation is valid [19]:
Moreover, sine functions are approximated by their arguments in the case of small angles. The Dirichlet kernel evaluated in k±0.5 can be then approximated as:
Therefore, the DFT of the cosine window in (6) can be approximated as:
The tone's amplitude A0 corresponds to the modulus of the DTFT of the signal evaluated at frequency f0=km+δ:
A
0
=|X(km+δ)| (26)
Its value can be derived as a function of the highest amplitude DFT bin:
leading to (16). The approximation sign is used instead of equality, because the ratios might differ due to spectral leakage [19].
Filing Document | Filing Date | Country | Kind |
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PCT/IB2017/053414 | 6/9/2017 | WO | 00 |