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I. INTRODUCTION
Source Localization and
Sensing: A Nonparametric
Iterative Adaptive Approach
BasedonWeightedLeast
Squares
The goal of array processing is to estimate the
locations and waveforms of sources by combining
the received data from multiple sensors so that the
desired signal is enhanced, while the unwanted
signals, such as interference and noise, are suppressed.
In active sensing applications such as radar/sonar
range-Doppler imaging, the aim is to find targets
present in a region of interest. For channel estimation
in communications, the aim is to estimate the non-zero
channel taps and their Doppler shifts, which are then
fed to the subsequent equalizer for symbol detection.
Low numbers of snapshots and low signal-to-noise
ratios (SNR) are among the many challenges that
array processing systems frequently face. Another
challenge is the presence of nearby sources, in terms
of location or Doppler, since closely spaced sources
are harder to discriminate. Herein both passive array
processing and active sensing are considered.
The most basic approach to array processing
is the classical delay-and-sum (DAS) method,
in which the received signal from each sensor is
weighted and delayed so as to focus on different
points in space. However, this method suffers from
low resolution and high sidelobe levels. There is a
vast amount of literature on methods that provide
superior performance over the DAS approach when
certain assumptions are met [1]. The well-known
standard Capon beamformer (SCB) [2] and multiple
signal classification (MUSIC) [3, 4] methods provide
superresolution when the sources are uncorrelated and
the number of snapshots is high. Many extensions
to these methods have been proposed to deal with
modelling errors, such as steering vector mismatches.
(See, e.g., [5]—[9].) However, none of these methods
is able to cope with very low snapshot numbers,
coherent or highly correlated sources, or severe noise.
Only a few snapshots are available when the
environment being sensed by the array is stationary
for a short duration of time. Moreover, to avoid
smearing, i.e., losing resolution because of wide
main beamwidths in the array response, averaging
can only be done over a small bandwidth [10, 11].
Therefore the number of available snapshots, which is
directly related to the time-bandwidth product, can be
very small, sometimes as small as 3, for applications
such as underwater array processing. Furthermore,
as discussed later, the data models for single-input
single-output (SISO) radar/sonar range-Doppler
imaging and multi-input single-output (MISO) channel
estimation in communication problems are similar
to the model used in array processing with a single
snapshot and an arbitrary array geometry (dictated by
the probing waveforms).
The array processing problem has been carried into
the sparse signal representation area by noticing that
the number of actual sources is usually much smaller
TARIK YARDIBI, Student Member, IEEE
JIAN LI, Fellow, IEEE
University of Florida
PETRE STOICA, Fellow, IEEE
Uppsala University
Sweden
MING XUE, Student Member, IEEE
University of Florida
ARTHUR B. BAGGEROER, Fellow, IEEE
Massachusetts Institute of Technology
Array processing is widely used in sensing applications
for estimating the locations and waveforms of the sources in
a given field. In the absence of a large number of snapshots,
which is the case in numerous practical applications, such
as underwater array processing, it becomes challenging to
estimate the source parameters accurately. This paper presents
a nonparametric and hyperparameter, free-weighted, least
squares-based iterative adaptive approach for amplitude and
phase estimation (IAA-APES) in array processing. IAA-APES can
work well with few snapshots (even one), uncorrelated, partially
correlated, and coherent sources, and arbitrary array geometries.
IAA-APES is extended to give sparse results via a model-order
selection tool, the Bayesian information criterion (BIC). Moreover,
it is shown that further improvements in resolution and accuracy
can be achieved by applying the parametric relaxation-based
cyclic approach (RELAX) to refine the IAA-APES&BIC
estimates if desired. IAA-APES can also be applied to active
sensing applications, including single-input single-output
(SISO) radar/sonar range-Doppler imaging and multi-input
single-output (MISO) channel estimation for communications.
Simulation results are presented to evaluate the performance of
IAA-APES for all of these applications, and IAA-APES is shown
to outperform a number of existing approaches.
Manuscript received December 11, 2007; revised July 21 and
November 4, 2008; released for publication November 16, 2008.
IEEE Log No. T-AES/46/1/935952.
Refereeing of this contribution was handled by W. Koch.
This work was supported in part by the Office of Naval Research
(ONR) under Grants N00014-07-1-0193, N00014-07-1-0293,
and N00014-01-1-0257, the Army Research Office (ARO)
under Grant W911NF-07-1-0450, the National Aeronautics and
Space Administration (NASA) under Grant NNX07AO15A, the
National Science Foundation (NSF) under Grants CCF-0634786,
ECS-0621879, and ECS-0729727, the Swedish Research Council
(VR), and the European Research Council (ERC).
Opinions, interpretations, conclusions, and recommendations are
those of the authors and are not necessarily endorsed by the United
States Government.
Authors’ addresses: T. Yardibi, J. Li, M. Xue, Dept. of Electrical
and Computer Engineering, University of Florida, PO Box 116130,
Gainesville, FL 32611, E-mail: (li@dsp.ufl.edu); P. Stoica, Dept.
of Information Technology, Uppsala University, Uppsala, Sweden;
A. B. Baggeroer, Depts. of Mechanical Engineering & Electrical
Engineering and Computer Science, Massachusetts Institute of
Technology, C ambridge, MA 02139.
0018-9251/10/$26.00 ° 2010 IEEE
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TABLE I
Notation used in the Text
k¢k 0 ` 0 -norm
k¢k 1 ` 1 -norm
k¢k 2 ` 2 -norm
k¢k F Frobenius norm of a matrix
¯ the Hadamard (elementwise) matrix product
tr( ¢ ) trace of a matrix
( ¢ ) T
transpose of a vector or matrix
( ¢ ) H
conjugate transpose of a vector or matrix
Fig. 1. Far-field linear array.
than the number of potential source points that can
be considered [12—29]. Sparsity-based techniques
have also been used in spectral estimation [30], image
processing, and array design [31—33], among many
other application areas. Sparse signal representation
algorithms can deal with a few snapshots (even one).
However, they may require large computation times
and the fine tuning of one or more hyperparameters.
Application areas, ranging from passive source
localization to active radar/sonar range-Doppler
imaging and channel estimation for communications
are addressed here. Section II considers passive
sensing applications and describes a weighted
least squares-based iterative adaptive approach
for amplitude and phase estimation (IAA-APES).
The algorithm is named IAA-APES herein since
its derivation resembles that of the amplitude and
phase estimation (APES) algorithm [34—36]. (See
Section IIC.) This name also distinguishes it from the
maximum likelihood (ML)-based iterative adaptive
approach (IAA-ML) discussed in the Appendix to
further motivate IAA-APES. Using the Bayesian
information criterion (BIC) [37, 38], IAA-APES can
be extended to yield point source estimates. (This
approach is referred to as IAA-APES&BIC.) Next, a
parametric relaxation-based cyclic approach, namely
RELAX [39, 40], is discussed as a way to further
refine the results of IAA-APES&BIC. (This approach
is referred to as IAA-APES&RELAX.) Section III
presents the data models for the aforementioned active
sensing applications and emphasizes the similarities
to the passive sensing case. IAA-APES is evaluated
via comprehensive simulations in Section IV, and the
IAA-APES’ performance is compared with that of a
number of existing approaches.
Notation : We denote vectors and matrices by
boldface lowercase and boldface uppercase letters,
respectively. The k th component of a vector x is
written as x k .The k th diagonal element of a matrix
P is written as P k . See Table I for other symbols and
their meanings.
used to estimate the desired source characteristics.
This section first introduces the data model for such
applications and then the IAA-APES algorithm in this
model.
A. Data Model
Consider the wavefield generated by K sources
located at Μ ,where Μ =[ Μ 1 , Μ 2 , ::: , Μ K ]and Μ k 1 are the
location parameters of the k th signal, k =1, ::: , K .In
the narrowband, multi-snapshot case, the 1 array
output vector of an M element array in the presence
of additive noise can be represented as [5, 43]
y ( n )= A ( Μ ) s ( n )+ e ( n ), n =1, ::: , N (1)
where N is the number of snapshots, A ( Μ )isthe
M£K steering matrix defined as A ( Μ )=[ a ( Μ 1 ),
a ( Μ 2 ), ::: , a ( Μ K )], and s ( n )=[ s 1 ( n ), s 2 ( n ), ::: , s K ( n )] T ,
n =1, ::: , N , is the source waveform vector at time n .
The array steering matrix has different expressions,
depending on the array geometry and on whether the
source is in the near-field or far-field of the array. For
instance, the steering vector corresponding to the k th
source for a far-field linear array (where Μ k represents
the impinging angle of source k in this case) is given
by
a ( Μ k )=[ e ¡j (2 ¼f=c 0 ) x 1 cos( Μ k ) , ::: , e ¡j (2 ¼f=c 0 ) x M cos( Μ k ) ] T
(2)
where f is the center frequency, c 0 is the wave
propagation velocity, and x m is the position of the
m th sensor, m =1, ::: , M . (See Fig. 1.) Note that for a
planar array or near-field sources, only the expressions
for the steering vectors have to be modified; the
algorithms that are presented can be applied without
any modifications, since a ( Μ ) is assumed to be a
known function of Μ . The number of sources, K ,is
usually unknown; hence, here, K is considered to be
the number of scanning points in the region. In other
words every point of a predefined grid that covers
the region of interest is considered as a potential
source whose power is estimated. Consequently,
II. PASSIVE SENSING
In passive sensing applications such as
aeroacoustic noise measurements [41] and underwater
acoustic measurements [42], an array of sensors is
1 With a slight abuse of notation, we do not use bold font for Μ k ,
k =1, ::: , K , which might be multi-dimensional, for simplicity.
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K will be much larger than the actual number of
sources present, and only a few components of
f s ( n ) g will be non-zero. This is the main reason why
sparse algorithms can be used in array processing
applications.
[26]. Fuchs [27, 28] uses a sparsity-constrained
deconvolution approach that assumes the sources
are uncorrelated and that the number of snapshots
is large. The sparsity-constrained solution is
obtained with a LASSO or BP type of algorithm.
Reference [29] introduces two hyperparameter
free deconvolution algorithms exploiting sparsity:
a sparsity-based extension to the deconvolution
approach for the mapping of acoustic sources
(DAMAS) [45] (which is similar to [27] and [28]
andwidelyusedinpractice)andasparsitybased
covariance-matrix fitting approach. Extensions
to the correlated source case are also provided.
However, the methods in [29] are based on the
sample covariance matrix, and hence, these
methods do not work well with a limited number of
snapshots.
B. Related Work
We focus our attention on array processing
algorithms exploiting sparsity, which have
gained noticeable interest recently. Sparse signal
representation aims at finding the sparsest s ,such
that y = As is satisfied, i.e., to minimize k s k 0 ,such
that y = As ,where A is known and y is measured.
The problem in its original form is a combinatorial
problem and is nondeterministic polynomial-time
(NP) hard, making it impractical [12]. Fortunately,
when s is sufficiently sparse [12—15], k s k 0 can
be replaced by k s k 1 , which leads to a convex
optimization problem that can be solved much more
easily by using, for instance, the least absolute
shrinkage and selection operator (LASSO) [16] or
basis pursuit (BP) [17] algorithms. Alternatively, the
focal underdetermined system solution (FOCUSS)
algorithm [18], which is derived using Lagrange
multipliers, can be used to iteratively solve the
sparse problem. A Bayesian approach, such as
sparse Bayesian learning (SBL) [19, 20] or the
approach in [21], can also be used to estimate s .
The algorithms mentioned up to this point are for
the single-snapshot case only. The extensions of
FOCUSS and SBL to the multiple-snapshot case
are M-FOCUSS [22] and M-SBL [23], respectively.
Another algorithm, the ` 1 -SVD (singular value
decomposition) algorithm [24, 25] is similar
to BP or LASSO, but this algorithm can work
with multiple snapshots. (For the single-snapshot
case, ` 1 -SVD becomes a LASSO and BP type
of method.) M-FOCUSS requires the tuning of
two hyperparameters, which might affect the
performance of the algorithm significantly. ` 1 -SVD
requires the tuning of a hyperparameter and an
estimate for the number of sources. Moreover,
implementing ` 1 -SVD requires convex optimization
software, such as SeDuMi [44]. M-SBL does
not require any hyperparameters. However,
M-SBL converges quite slowly in its original form
[19, 20, 23].
Besides the above algorithms, which are the
focus of our attention in the numerical examples,
there are other sparsity based approaches worth
mentioning. Reference [26] adds an additional spatial
sparsity regularizing term (an ` 2 -norm constraint)
to the ` 1 -norm constraint, and it minimizes a cost
function similar to that of ` 1 -SVD. However, this
method has two hyperparameters, assumes that
the source waveforms can be represented by a
sparse basis, and has high computational complexity
C. IAA-APES
IAA-APES is a data-dependent, nonparametric
algorithm based on a weighted least squares (WLS)
approach. Let P be a K£K diagonal matrix, whose
diagonal contains the power at each angle on the
scanning grid. Then P can be expressed as
P k = 1
X
js k ( n ) j 2 , k =1, ::: , K: (3)
N
n =1
Furthermore, define the interference (signals at angles
other than the angle of current interest Μ k ) and noise
covariance matrix Q ( Μ k )tobe
Q ( Μ k )= R ¡P k a ( Μ k ) a H ( Μ k )
( 4 )
where R = A ( Μ ) PA H ( Μ ). Then the WLS cost function
is given by (see, e.g., [34]—[36] and [43])
N
X
k y ( n ) ¡s k ( n ) a ( Μ k ) k 2 Q ¡ 1 ( Μ k )
(5)
n =1
a H ( Μ k ) Q ¡ 1 ( Μ k ) a ( Μ k ) , n =1, ::: , N: (6)
This looks like the result that would be obtained by
employing APES [34—36], but it is actually different
than APES since APES obtains Q ( Μ k ) from the data
by forming subapertures, while IAA-APES computes
Q ( Μ k ), as in (4). Moreover, IAA-APES is iterative, but
APES is not, and APES cannot be used with arbitrary
array geometries.
Using (4) and the matrix inversion lemma, (6) can
be written as
ˆ s k ( n )= a H ( Μ k ) R ¡ 1 y ( n )
a H ( Μ k ) R ¡ 1 a ( Μ k ) , n =1, ::: , N: (7)
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N
where k x k 2 Q ¡ 1 ( Μ k ) = x H Q ¡ 1 ( Μ k ) x and s k ( n ) represents the
signal waveform at angle Μ k and at time n . Minimizing
(5) with respect to s k ( n ), n =1, ::: , N , yields
ˆ s k ( n )= a H ( Μ k ) Q ¡ 1 ( Μ k ) y ( n )
641081594.006.png
TABLE II
The IAA-APES Algorithm
than necessary reduction of the number of degrees
of freedom when some of the interfering sources are
coherent since the cancellation of multiple coherent
interfering sources would require only one DOF if
the correct structure of P were known. However, we
do not assume that the true structure of P is known.
Moreover, it is the diagonal structure of P assumed by
IAA-APES that makes the algorithm work properly
even for low number of snapshot cases and coherent
sources.
X
1
( a H ( Μ k ) a ( Μ k )) 2 N
P k =
j a H ( Μ k ) y ( n ) j 2 , k =1, ::: , K
n =1
repeat
R = A ( Μ ) PA H ( Μ )
for k =1, ::: , K
ˆ s k ( n )= a H ( Μ k ) R ¡ 1 y ( n )
a H ( Μ k ) R ¡ 1 a ( Μ k ) , n =1, ::: , N
X
P k = 1
N
j ˆ s k ( n ) j 2
D. IAA-APES&BIC
n =1
end for
until (convergence)
In many applications it is desirable to obtain point
estimates rather than a continuous spatial estimate.
To achieve this sparsity we incorporate a model-order
selection tool, i.e., the BIC [37, 38], into IAA-APES.
Let P denote a set containing the indices of the peaks
selected from the IAA-APES spatial power spectrum
estimate. Also let I denote the set of the indices of
the peaks selected by the BIC algorithm so far. The
IAA-APES&BIC algorithm works as follows: first
the peak, from the set P , giving the minimum BIC is
selected. Then the second peak, from the set P¡I ,
which together with the first peak gives the minimum
BIC, is selected, and so on, until the BIC value
does not decrease anymore. 2 The IAA-APES&BIC
algorithm is summarized in Table III. BIC i ( ´ )is
calculated as follows (see [38])
TABLE III
The IAA-APES&BIC Algorithm
P : Set of peaks obtained from IAA-APES
I =Ø, ´ =1, quit=0, BIC old = 1
repeat
i 0 =argmin
i2P¡I
0
1
°
° 2
2
This avoids the computation of Q ¡ 1 ( Μ k ) for each
scanning point, i.e., K times. Moreover, f ˆ s k ( n ) g can
be computed in parallel for each scanning point,
which makes IAA-APES amenable to implementation
on parallel hardware. IAA-APES is summarized in
Table II. Since IAA-APES requires R ,whichitself
depends on the unknown signal powers, it has to
be implemented as an iteration. The initialization is
done by a standard DAS beamformer. Our empirical
experience is that IAA-APES does not provide
significant improvements in performance after
about 15 iterations. In IAA-APES, P and, hence
R , are obtained from the signal estimates of the
previous iteration and not from the snapshots as in
conventional adaptive beamforming algorithms, such
as SCB, which fails to work properly with coherent
or highly correlated sources, or few snapshots [43].
The Appendix provides an alternative derivation of
IAA-APES, based on the ML principle. IAA-APES
is shown to be an approximation of the IAA-ML
algorithm, which is locally convergent due to
cyclically maximizing the likelihood function. Hence,
the analysis in the Appendix provides an approximate
calculation that shows the local convergence of
IAA-APES.
Because P is assumed to be a diagonal matrix,
one degree of freedom (DOF) is needed to suppress
one interfering source. This may lead to a larger
@
X
X
A
BIC i ( ´ )=2 MN ln
y ( n ) ¡
a ( Μ j ) ˆ s j ( n )
n =1
j2fI[ig
+3 ´ ln(2 MN )
( 8 )
where ´ = jIj +1, jIj denotes the size of the set I , i is
the index of the current peak under consideration, and
f ˆ s j ( n ) g n =1 is the IAA-APES signal waveform estimate
corresponding to angle Μ j , j 2fI[ig . Note that the
second term on the right side of (8) does not matter
to peak selection; it matters only when (8) is used to
select the number of peaks to retain.
E. IAA-APES&RELAX
RELAX [39, 40] is a parametric cyclic algorithm
that requires an estimate of the number of sources in
the field. The results of IAA-APES&BIC can be used
to provide a good initial estimate for the last step of
RELAX. This approach is outlined in Table IV. The
RELAX iterations can be terminated when the norm
of the difference between two consecutive estimates
is smaller than a certain threshold (5 £ 10 ¡ 4 for the
examples considered herein). The maximization
2 Alternatively, the largest ´ peaks can be selected so that when
the ( ´ +1)st largest peak is added to I , the BIC value does not
decrease. This simplified version gives similar results to the one
described above in our numerical examples.
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N
N
BIC i ( ´ )
if BIC i 0 ( ´ ) < BIC old
I = fI , i 0 g
BIC old =BIC i 0 ( ´ )
´ = ´ +1
else quit=1
until (quit=1)
N
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TABLE IV
The IAA-APES&RELAX Algorithm
Μ 0 : Locations of the peaks obtained from IAA-APES&BIC
K 0 : Number of peaks obtained from IAA-APES&BIC
f ˆ s k ( n ) g : Corresponding waveforms obtained from
IAA-APES&BIC
repeat
for k =1, ::: , K 0
X
K 0
y k ( n )= y ( n ) ¡
a ( Μ 0 i ) ˆ s i ( n ), n =1, ::: , N
i =1
i6 = k
X
N
Μ 0 k =argmax Μ 0 2 R
j a H ( Μ 0 ) y k ( n ) j 2
n =1
M a H ( Μ 0 k ) y k ( n ), n =1, ::: , N
Fig. 2. Pulse compression for radar/sonar range-Doppler imaging.
end for
until (convergence)
step of the algorithm can be implemented without
much computational effort by means of a fine search
only around the peak values of the IAA-APES&BIC
result. 3 RELAX can be useful in estimating off-grid
sources accurately and for further improving the
IAA-APES waveform and angle estimates.
the Doppler interval of interest is divided into L bins,
and the Doppler frequency for the l th Doppler bin is
denoted as ! l . Then the M samples of the received
signal that is temporally aligned with the return from
the range bin of current interest r (see Fig. 2) can be
represented by
X
X
X
y r =
® r , l s ( ! l )+
® r + m , l J m s ( ! l )+ e r
III. ACTIVE SENSING
l =1
m = ¡M +1
m6 =0
l =1
(12)
for r =1, ::: , R ,where ® r , l (which is proportional to
the complex “voltage” radar-cross section (RCS)
of the corresponding target) denotes the complex
amplitude of the returned signal from the range bin
of current interest r and the l th Doppler bin m + r , l g
denote the complex amplitudes of the returned signals
from the adjacent range bins, and e r denotes the noise.
The reflections from nearby range bins are considered
to be clutter. The M£M shift matrix J m takes into
account the fact that the clutter returns from adjacent
range bins need different propagation times to reach
the radar/sonar receiver:
In active sensing applications, besides the
receiver, there are also one or more transmitters.
The radar/sonar range-Doppler imaging problem for
a SISO system is first investigated in this section.
Then the channel estimation problem for MISO
communications is discussed.
A. Range-Doppler Imaging
Pulse compression refers to the process of
transmitting a modulated pulse and then matched
filtering the returned signal, which arrives at the
antenna altered by complex coefficients that bear
target information [48].
1) Data Model : Consider a range-Doppler
imaging radar/sonar with a transmitted pulse
˜ s =[ ˜ s (1), ˜ s (2), ::: , ˜ s ( M )] T
2
3
á m ¡!
0 ¢¢¢ 1 ¢¢¢ 0
. . . . . . . . . . . .
0 . . . 1
. . . . . . . . . . . .
0 ¢¢¢ 0 ¢¢¢ 0
(9)
4
5 = J ¡m (13)
where M is the pulse length. Let
s ( ! l )= ˜ s ¯ d ( ! l )
J m =
(10)
be the Doppler shifted signal, where
d ( ! l )=[1, e j! l , ::: , e j ( 1) ! l ] T , l =1, ::: , L
for m =0, ::: , 1. Equation (12) can be written as
(11)
y r = S ® r + e r
(14)
3 The fine search can be implemented efficiently using the fast
Fourier transform (FFT) [39, 40] or a derivative-free uphill search
method, such as the Nelder-Mead algorithm [46, 47]. Note that the
latter method is available in the MATLAB optimization toolbox
with the name of “fminsearch.”
where
S =[ s ( ! 1 ), s ( ! 2 ), ::: , s ( ! L )] (15)
® r =[ ® r ,1 , ® r ,2 , ::: , ® r , L ] T
(16)
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ˆ s k ( n )= 1
L
1
L
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