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IEEE TRANSACTIONS ON POWER DELIVERY, VOL. 15, NO. 3, JULY 2000

Application of Staggered Undersampling to Power

Quality Monitoring
Hanoch Lev-Ari, Senior Member, IEEE, Aleksandar M. Stankovic, and S. Lin

AbstractIn this paper we address the issue of monitoring

power quality with limited sensing and computational resources. The proposed staggered sampling methodology utilizes
close-to-periodic nature of waveforms of interest. The sparse
samples obtained in our scheme are reordered (shuffled) before
FFT is used for spectral calculations. We provide a complete
characterization of the shuffling process in time domain and of the
dual inverse shuffling in the frequency domain. The methodology
is applied to a number of cases of interest in power quality,
demonstrating that harmonics of the fundamental frequency can
be recovered without error, and that effective descriptions are
possible for flicker, sag, interharmonics, and noisy measurements.

technology, like multirate algorithms, have been fruitfully applied to power quality problems [5].
The rest of the paper is organized as follows: in Section II
we describe the algorithm and present a motivating example;
the analysis of the algorithm is presented in Section III, and
followed by applications to power quality monitoring in Section IV; the paper is briefly summarized in Section V, while the
analytical details are explained in the Appendix.

Index TermsComputational algorithms, power quality, signal

sampling and reconstruction.

The main idea in staggered sampling is to manipulate sparse

samples of a signal to obtain a set equivalent to the one obtained
by fast sampling. This equivalence is of course possible only
for some classes of waveforms such as periodic signals. Recall
that the Nyquist theorem states that a band-limited signal can be
exactly reconstructed from samples provided that the sampling
rate is greater or equal to twice the bandwidth of the signal.
Slower sampling would in principle introduce deviations in the
frequency content (aliasing); this can, however, be avoided if the
signal belongs to a restricted class (e.g., periodic) and samples
are 1) taken at instants that are carefully chosen, and 2) precisely
manipulated before applying the fast Fourier transform (FFT).
Assume that a signal is periodic with period , and that a total
in our examples). Also asof samples is taken (e.g.,
have been comsume that harmonics higher than
pletely eliminated before sampling by the anti-aliasing filter.
seconds, where and are
Samples are taken every
in our examples). It will thus take
mutually coprime (
periods of the signal to acquire
samples, but all samples
will be taken at distinct relative positions within a period (as we
prove later). If the samples are then reordered (shuffled), they
samples within
can equal the set of samples taken every
one period. Given such a set of samples, we can recover the
)st harmonic exactly via
harmonic content up to the (
FFT. In a sense, one is using the periodicity of the signal to reduce the sampling frequency times without causing aliasing
[recall that the signal is assumed to be bandlimited to
by anti-aliasing filtering]. The analysis presented here builds on
results from [6].
We show in the next section that the required shuffling maps
the th sample into the th position according to the following
formula

I. INTRODUCTION

HE IMPORTANCE of power quality is increasing with the

power market deregulation and with the emergence of new
electric loads that are sensitive to variations in power supply
waveforms. The ability to adequately measure and document
power quality is thus critical to the unimpeded functioning of
the market. Existing sensing and digital signal processing technologies are capable performing the required measurements in
real time, but at significant signal conversion (AID) and computational costs. In this paper we propose a sampling and signal
processing scheme that reduces these penalties by exploiting the
particular close-to-periodic nature of most variables of interest.
This makes our scheme relevant for even very cost-sensitive
equipment, such as digital wattmeters and similar apparatus.
Problems related to power quality are becoming prevalent in a
majority of power utilities, and are a rising concern to customers
[1]. Signal processing technology is of primary importance in
assessment and classification of disturbances in a power network. Many digital signal processing (DSP) schemes have been
applied successfully in power systems, and a number of specialized algorithms has been developed over the last three decades.
A comprehensive summary of power quality issues and related
computational requirements is presented in [2]. A number of
potential sources of errors (such as finite word length) that was
typical in the early DSP technologies is described in [3]. While
the impact of some of the difficulties has been mitigated by
advances in microcontroller technology, certain problems (like
synchronization errors) persist [4]. Recent developments in DSP

II. DESCRIPTION OF THE ALGORITHM

(1)
Manuscript received January 28, 1999. The work of A. M. Stankovic was
supported by the National Science Foundation under Grant ECS-9502636 and
the Office of Naval Research under Grant N14-97-1-0704.
The authors are with Northeastern University, Boston, MA.
Publisher Item Identifier S 0885-8977(00)07200-9.

denotes the reminder of after division by

where
we obtain
In particular, when

08858977/00$10.00 2000 IEEE

LEV-ARI et al.: APPLICATION OF STAGGERED UNDERSAMPLING TO POWER QUALITY MONITORING

865

presence of nonharmonic components, including interharmonics, modulated harmonics, and white or colored noise.
As described in the previous section, the staggered undersampling algorithm forms a sequence of sparse samples
, viz.,

where
denotes the continuous-time (analog) input
signal. In contrast, conventional (dense) sampling of the
,
same analog signal would have produced a sequence
where

Notice that the sparse sequence

can also be interpreted as
because
a subsampled version of
(2)
Fig. 1.

Unordered samples of a 3 tone signal.

and
is highlighted in a blockThis relation between
diagram representation of the staggered undersampling algorithm, in which we represent the sparse sampler as a cascade of a
dense sampler followed by a factor-of- downsampler (Fig. 3).
As explained in Section II, the harmonic content of a periodic
analog input signal can be determined without error from a fi, provided
nite segment of the densely sampled sequence
that the dense sampler is synchronized with the period of the
is periodic with
analog signal (see Fig. 3). Consequently,
.
period , i.e.,
Our first objective in this section is to demonstrate that, for a
, we can reconstruct a single period
periodic input signal
without error by applying the shuffling operation to
of
consecutive samples of the subsampled sequence
. Comwith (2) we find that
bining the periodicity of
(3)

Fig. 2.

Ordered samples of a 3 tone signal and harmonic content.

so that no shuffling is required. This special case of the algorithm presented here was used for sampling oscilloscopes
[7, pp. 463465].
Next, we show an example that illustrate the capabilities of
the method. We consider a sinusoidal waveform with 10% fifth
,
harmonic and 1% seventh harmonic added, and
and
. Fig. 1 shows unordered samples, while Fig. 2
shows ordered (shuffled) samples and the correctly determined
harmonic content (expressed in dB of the fundamental).
III. ANALYSIS OF THE ALGORITHM
In this section we introduce a multirate signal processing
characterization of the staggered sampling algorithm, and
we use it to analyze the performance of this algorithm in the

which establishes a correspondence between the finite set of

and the finite set
sparse samples
. We now show that
of dense samples
this correspondence is onto and one-to-one, so that the latter set
can be obtained by shuffling the elements of the former, using
the index mapping (1).
Result 1: If , are mutually coprime, than the index mapping (1) is onto and one-to-one.
Proof: See Appendix.
Thus shuffling the finite set of sparse samples
reproduces one complete period of the densely
. The harmonic content of the analog
sampled sequence
input signal is then obtained by applying a scaled FFT to the
shuffled set of samples, as explained in Section II.
Our next objective in this section is to determine the effect
on the harmonic content
of nonharmonic components of
as determined by the FFT block in
Fig. 3. Since the system described by Fig. 3 is linear (albeit
time-varying) we can analyze the effect of each component individually. For instance, a white noise component in the analog
input signal gives rise to an independent identically distributed
(i.i.d.) sequence component in both the densely sampled signal

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IEEE TRANSACTIONS ON POWER DELIVERY, VOL. 15, NO. 3, JULY 2000

Fig. 3. Block-diagram representation of the staggered undersampling algorithm.

Fig. 4.

Equivalent block-diagram representation of the staggered undersampling algorithm.

and its undersampled version

. The power (=vari, where
is the cutoff freance) of this component is
denotes the
quency of the anti-aliasing lowpass filter, and
(one-sided) power spectral density of the analog white noise
component. Since the FFT is a (scaled) orthogonal transformation, the end result is a random contribution to every harmonic
. The power of that contribution is the same for
component
.
all harmonics, and equals
While the time-domain shuffle leaves the frequency content
of white noise unaltered, the same is not true for other types
of signals, such as colored noise or interharmonic sinusoldal
signals. As we shall presently show, the application of the algorithm of Fig. 3 to such signals results in mixing up their
frequency content. Our next result provides an explicit characterization of this frequency-domain effect.
Result 2: A shuffle operation [as given by (1)] followed by
an FFT is equivalent to an FFT operation followed by an inverse
shuffle.
Proof: See Appendix.
This result implies that, for any analog input signal, the
block-diagram of Fig. 3 is equivalent to the block-diagram
shown in Fig. 4. We can now use standard multirate analysis
techniques (see, e.g., [8]) to determine the discrete-time Fourier
, viz.,
transform (DTFT) of the sequence

The application Of an FFT to a finite segment of the sequence

results in a windowed (and frequency-sampled) version
of the DTFT, which is then mixed up by the inverse shuffle
operation.
The harmonic content after the inverse shuffle can be determined by using the following explicit expression.
Result 3: The inverse to the shuffle operation (1) is given by

where
is the smallest positive integer that satisfies the equa.
tion
Proof: See Appendix.

To demonstrate the utility of the equivalent block-diagram

representation of Fig. 4, consider a single (complex) interharmonic component at frequency , viz.,

where

. The DTFT of
is
, where
denotes the Dirac delta function. The
FFT block uses a windowed version (= a finite segment) of
, namely the FFT input is
, where
else
The DTFT of this windowed signal is a convolution between
and
, which equals
. The
output of the FFT operation is a sampled version of this DTFT,
, where
. Thus a single
namely
interharmonic complex sinusoid contributes a single smeared
, to the signal at the input of the inverse
peak, centered at
shuffle block in Fig. 4, while a real interharmonic sinusoid con, and the other at
.
tributes two smeared peaks, one at
For instance, a 30 Hz analog interharmonic component at the
input of a system designed to determine the harmonics of 60
,
and
) results in
Hz signals (with
. This
smeared peaks centered at
means that the strongest harmonics before inverse shuffling are
. Since in this case
, the inverse
located at
shuffle moves the strongest harmonics to
, which gets folded back to
, and
. Fig. 5 shows the ordered samples and
spectral content of a signal comprising the 60 Hz fundamental
and 10% of the 30 Hz interharmonic. Note how prominent are
13th and 14th harmonic in the lower panel.
IV. FURTHER APPLICATIONS IN POWER QUALITY MONITORING
We first consider the case of flicker [2], i.e., a 60-Hz sinusoid whose magnitude is modulated (10% in this case) with the
Hz). In that case the analog
frequency of (set to
Hz, and smaller (5%)
signal has the main component at
. The analysis from the previous
components at sidebands
section establishes that the main peak before inverse shuffling

LEV-ARI et al.: APPLICATION OF STAGGERED UNDERSAMPLING TO POWER QUALITY MONITORING

Fig. 5. Ordered samples of a 30 Hz interharmonic signal and harmonic

content.

Fig. 6.

Ordered samples of a flicker signal and harmonic content.

will be at
, which will be mapped to
after the inand
,
verse shuffling. The side components are at
and
and they are mapped by the inverse shuffling to
, respectively. Fig. 6 shows ordered samples and the harmonic content in the corresponding numerical simulation; note
and
which is
the magnitude of components at
dB. A family of spectral signatures (dependent
on ) can be used for detection of flicker in power system apparatus that uses staggered sampling.
Next, we consider the case of sag i.e., a 60-Hz sinusoid whose
magnitude is modulated (20% in this case) over several cycles
(3 in our case). Fig. 7 shows unordered samples, while Fig. 8
shows ordered samples and the harmonic content. In the case
of a sag, more effective detection methods are likely to be in

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Fig. 7. Unordered samples of a sag.

Fig. 8. Ordered samples of a sag signal and harmonic content.

time domaine.g., in the top panel of Fig. 8 samples obtained

during sag clearly differ from values obtained by a simple (even
linear) interpolation of the left and right neighboring values.
These simple calculations can be used in for fast on-line detection of sag.
Finally, we consider the case of noisy measurements, i.e., a
60-Hz sinusoid with added white noise (standard deviation is
5% of the magnitude of the fundamental); Fig. 9 shows ordered
samples and the harmonic content. As explained in the previous
section, we expect the noise to influence all harmonics, which is
the case in Fig. 9. In the case of white (or colored) noise detection, frequency domain criteria are likely to be most effective.
For example, a flat platform is expected in the case of additive
white noise, as explained in the previous section.

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IEEE TRANSACTIONS ON POWER DELIVERY, VOL. 15, NO. 3, JULY 2000

While the final choice depends on the application and characteristics of the employed hardware,
is reasonable with
present technology.
V. CONCLUSIONS

Fig. 9. Ordered samples of a noisy sinusoid and its harmonic content.

This paper introduces a signal processing methodology that

reduces the sampling frequency and data conversion requirements in power quality monitoring. The methodology utilizes
close-to-periodic nature of waveforms of interest. The sparse
samples obtained in our scheme have to be reordered (shuffled)
before FFT can be used for spectral characterization. A complete characterization of the shuffling process in time domain
and of the dual inverse shuffling in the frequency domain is a
central theme in the paper. Applications to a number of cases of
interest in power quality monitoring demonstrate that harmonics
of the fundamental frequency are recovered without error, and
that effective descriptions are possible for flicker, sag, interharmonics and noisy measurements (the last case was explored experimentally as well). Thus the staggered sampling procedure
allows equipment designers to extract more power quality information with limited signal processing resources.
APPENDIX
PROOF OF RESULTS 13
Proof of Result 1: To establish the one-to-one nature
of the index map (1) we need to show that two distinct
values of cannot be mapped into the same value
implies that
of . Indeed,
, and since
are coprime, we
. The only solution to this
deduce that
and
restricted to the range
equation with both
is
, which establishes the one-to-one property.
Since distinct values of map into distinct values of , the
is covered by the mapping (1) as
entire range
varies in the same range, namely the map is indeed onto.
Proof of Result 2: The shuffling relation (3) can also be
written in the form

Fig. 10.

Ordered samples of a measured voltage and its harmonic content.

The results for the case of (simulated) noisy measurements

are quite comparable with results obtained experimentally for
distribution voltage in our laboratory using a digitizing storage
oscilloscope (with 50 kHz sampling frequency, and 2% digitizing error, normalized by nominal voltage amplitude 170 V);
Fig. 10 shows ordered samples and the harmonic content.
Detection of brown-outs (i.e., sustained reductions in magnitude of the fundamental) is easy in frequency domain, as only
the first harmonic needs to be monitored.
In practical implementations the choice of is a very important issue: if is chosen too small, then we do not gain enough
in terms of reduced processing requirements from down sampling; if is too large, then it takes too long to collect samples, making our information about the sampled signal outdated.

where
, and
is the Kronecker delta. This
means that the mapping of the set
into the set
is a permutation, and that
. Thus,
the corresponding permutation matrix is
, where
is
the result we need to establish is
the DFT matrix, viz.,

We shall prove, instead, the equivalent statement

)th element of
is
Indeed, the (

LEV-ARI et al.: APPLICATION OF STAGGERED UNDERSAMPLING TO POWER QUALITY MONITORING

and, consequently, the (

)th element of

is

which establishes the result.

Proof of Result 3: From (1) we have

Since

, it follows that
and, therefore,
inverse of the map (1).

, which is the

REFERENCES
[1] E. W. Gunther and H. Mehta, A survey of distribution system power
quality, IEEE Trans. Power Delivery, vol. 10, pp. 322329, January
1995.
[2] G. T. Heydt, Electric Power Quality: Stars in a Circle Publications, 1991.
[3] K. Srinivasan, Errors of digital measurement of voltage, active and reactive powers and an on-line correction of frequency drift, IEEE Trans.
Power Delivery, vol. 2, pp. 7276, January 1987.
[4] X. Dai and R. Gretsch, Quasisynchronous sampling algorithm and its
applications, IEEE Trans. Instrument. Measure., vol. 43, pp. 204209,
April 1994.
[5] L. Toivonen and J. Morsky, Digital multirate algorithms for measurement of voltage, current, power and flicker, IEEE Trans. Power Delivery, vol. 10, pp. 116126, January 1995.
[6] S. Lin, Undersampling method and its application in developing a
digital spectral wattmeter with power quality monitoring, M.S. thesis,
Northeastern University, Boston, MA, May 1998.
[7] W. McC. Siebert, Circuits, Signals and Systems: The MIT Press, 1986.

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[8] J. G. Proakis and D. G. Manolakis, Digital Signal Processing: Principles, Algorithms, Applications, 3rd ed. Englewood Cliffs, NJ: Prentice-Hall, 1996.

Hanoch Lev-Ari (S78M84SM93) received the B.S., Summa Cum Laude,

in 1971, and the M.S. in 1978, both in electrical engineering from the Technion,
Israel Institute of Technology, Haifa, Israel; and the Ph.D. in electrical engineering from Stanford University, Stanford, CA, in 1984. During 1985 he held
a joint appointment as an Adjunct Research Professor of electrical engineering
with the Naval Postgraduate School, Monterey, CA and as a Research Associate
with the Information Systems Laboratory at Stanford; he stayed at Stanford as
a Senior Research Associate until 1990. He is currently an Associate Professor
with the Department of Electrical and Computer Engineering at Northeastern
University. During 19941996 he was also the Director of the Commununications and Digital Signal Processing (CDSP) Center at Northeastern University.
His present areas of interest include model-based spectrum analysis and estimation for nonstationary signals, scale-recursive (multirate) detection and estimation of random signals, and adaptive linear and nonlinear filtering techniques,
with applications to channel equalization, over-the horizon (OTH) radar, automatic target detection and recognition, and identification of time-variant systems. Dr. Lev-Ari served as an Associate Editor of Circuits, Systems and Signal
Processing, and of Integration, The VLSI Journal. He is a member of SIAM,
and a senior member of IEEE.

Aleksandar M. Stankovic obtained the Dipl.Ing. degree from the University of

Belgrade, Yugoslavia in 1982, the M.S. degree from the same institution in 1986,
and the Ph.D. degree from Massachusetts Institute of Technology in 1993, all in
electrical engineering. He has been with the Department of Electrical and Computer Engineering at Northeastern University, Boston since 1993, presently as
an Associate Professor. Dr. Stankovic is a member of IEEE Power Engineering,
Power Electronics, Control Systems, Industry Applications and Industrial Electronics Societies. He serves as an Associate Editor for the IEEE TRANSACTIONS
FOR CONTROL SYSTEM TECHNOLOGY.