DA-MUSIC: Data-Driven DoA Estimation via Deep
Augmented MUSIC Algorithm
Julian P. Merkofer, Student Member, IEEE, Guy Revach, Student Member, IEEE , Nir Shlezinger, Member, IEEE,
Tirza Routtenberg, Senior Member, IEEE, and Ruud J. G. van Sloun, Member, IEEE
Abstract—Direction of arrival (DoA) estimation of multiple
signals is pivotal in sensor array signal processing. A popular
multi-signal DoA estimation method is the multiple signal clas-
sification (MUSIC) algorithm, which enables high-performance
super-resolution DoA recovery while being highly applicable in
practice. MUSIC is a model-based algorithm, relying on an
accurate mathematical description of the relationship between
the signals and the measurements and assumptions on the signals
themselves (non-coherent, narrowband sources). As such, it is
sensitive to model imperfections. In this work we propose to over-
come these limitations of MUSIC by augmenting the algorithm
with specifically designed neural architectures. Our proposed
deep augmented MUSIC (DA-MUSIC) algorithm is thus a hybrid
model-based/data-driven DoA estimator, which leverages data
to improve performance and robustness while preserving the
interpretable flow of the classic method. DA-MUSIC is shown
to learn to overcome limitations of the purely model-based
method, such as its inability to successfully localize coherent
sources as well as estimate the number of coherent signal sources
present. We further demonstrate the superior resolution of the
DA-MUSIC algorithm in synthetic narrowband and broadband
scenarios as well as with real-world data of DoA estimation from
seismic signals.
I. INTRODUCTION
Source separation, localization, and tracking are crucial tasks
in sensor array processing. In particular, direction of arrival
(
DoA
) estimation of multiple, broadband, and possibly coherent
signal sources plays a key role in a wide range of applications,
including radar, communications, image analysis, and speech
enhancement [2]–[4]. Over the last decades, a multitude of
different
DoA
estimation algorithms have been proposed, and
the problem is an active area of research [5], [6].
A leading scheme employed in many
DoA
applications is the
popular multiple signal classification (
MUSIC
) algorithm [7],
which can provide asymptotically unbiased estimates of the
number of incident wavefronts present, their approximate fre-
quencies, and their
DoA
s.
MUSIC
and other classic approaches,
e.g., conventional beamforming [8] and
MVDR
beamform-
ing [9], are based on knowledge of the underlying statistical
model; this dependency induces several key limitations. Among
the drawbacks of the model-based (
MB
) approaches is the
inherent limitation of the signal model from which they are
Parts of this work were presented in the 2022 IEEE International Conference
on Acoustics, Speech, and Signal Processing (ICASSP) as the paper [1]. J. P.
Merkofer is with the EE Dpt., Eindhoven University of Technology, Eindhoven,
The Netherlands (e-mail: [email protected]), this work was partially done
while being with the ETH Z
¨
urich. G. Revach is with the D-ITET, ETH Z
¨
urich,
Switzerland, (email: grev[email protected]). N. Shlezinger and T. Routtenberg are
with the School of ECE, Ben-Gurion University of the Negev, Beer Sheva,
Israel (e-mail:
{
nirshl; tirzar
}
@bgu.ac.il). T. Routtenberg is also with the ECE
Dpt., Princeton University, Princeton, NJ, United States. R. J. G. van Sloun
is with the EE Dpt., Eindhoven University of Technology, and with Phillips
Research, Eindhoven, The Netherlands (e-mail: r.j.g.v[email protected]).
derived, which considers narrowband signals. This results, for
example, in the inability to consistently estimate the
DoA
of
correlated (coherent) signals, as well as a failure to resolve
closely spaced signals with an insufficient number of samples
or a low signal-to-noise ratio (SNR) [5], [7].
To extend narrowband models to broadband
DoA
esti-
mation, various extensions and alternative approaches have
been explored [10], because when the impinging signals
are broadband, multiple frequency ranges can carry different
information regarding the
DoA
angles at hand. Generally, these
extensions from narrowband to broadband can be subdivided
into coherent and incoherent processing methods. In coherent
processing methods, the covariance matrices of the observations
in different frequency bins are coherently combined using
certain transformation matrices. Most coherent techniques are
based on the coherent signal subspace (
CSS
) concept [11],
which focuses the transformations of the covariances at
different frequencies into a surrogate narrowband model, yet
utilizes different transformation matrices. Incoherent broadband
processing methods combine
DoA
estimates obtained separately
for each frequency bin [12].
The recent success of data-driven (
DD
) deep learning
(
DL
) across a wide range of disciplines gave rise to neural
network (
NN
)-aided
DoA
estimators [13]. The works [14]–[17]
implemented model-agnostic
DoA
estimation using a dense
NN
,
a convolutional
NN
(
CNN
), a sparse-connected
CNN
, and a U-
Net architecture, respectively. Other methods [18], [19] utilize
the knowledge of the spatial covariance matrices, where [18]
trained a
CNN
based on a classification task (allowing it to also
operate with unknown number of sources) and [19] analyzed
purely
DD
estimators and
DD
methods in combination with
maximum likelihood estimation (
MLE
). The works [20], [21]
also combined
DL
with
MLE
by using multiple dense
NN
s
and a ResNet, respectively, to estimate a subset of candidate
angles. [22] extends the architecture of [21] to estimate the
number of sources with the ResNet. While such black-box
NN
s enable handling array imperfections due to their model-
agnostic nature, they involve highly parameterized models that
may be computationally intensive and lack the interpretability
of MB methods.
Alternatively,
NN
s were used to robustify the
MB MUSIC
as a form of hybrid
MB
/
DD
system [23], [24]. Specifically,
the work [25]and [26] proposed to estimate the discretized
MUSIC
spectrum from the (spatially smoothed) covariance
matrix of the measurements through the utilization of multiple
convolutional
NN
s. While these methods are more robust
to model inaccuracies compared with the original
MUSIC
algorithm, utilizing the
MUSIC
spectrum as a label for
training causes them to experience the same drawbacks as
1
arXiv:2109.10581v5 [eess.SP] 11 Jan 2023
their
MB
counterpart. Another
DD
approach proposed in [27]
considered systems with subarray sampling and uses
NN
s to
obtain a single estimated covariance matrix from incoherent
subarray measurements. This
NN
-aided estimate is utilized
for
DoA
recovery via the subspace-based
MUSIC
algorithm.
The method addresses the fundamental dependency of
MUSIC
on the estimated covariance matrix, thereby robustifying the
MUSIC
algorithm, yet it does not fully exploit the
NN
s’
ability to improve
MUSIC
, as the
NN
is trained using the
true covariance matrix as a label, without considering its
downstream task. Furthermore, the work [28] proposed a hybrid
MB
/
DD
approach that includes an eigenvalue decomposition
(
EVD
) of full-row Toeplitz matrices reconstruction (
FTMR
)
matrices [29] and classification deep
NN
s for estimation of
a
MUSIC
-like spectrum and detection of number of sources
present via eigenvectors and eigenvalues
The works [30], [31] applied
CNN
s for broadband
DoA
estimation of a single source. The two approaches, however,
utilize very different preprocessing methods and are trained
as classification and regression tasks, respectively. While the
classification approach limits itself to a fixed resolution, the
regressor of [31] outputs the
DoA
angles directly. Therefore, it
can achieve an arbitrary resolution, yet it is limited to the
spatial covariance matrix as input. Another approach [32]
first decomposed the broadband signals into non-overlapping
narrowband components and used support vector regression to
obtain the DoA.
The limitations of both existing
MB
and
DD DoA
estimation
algorithms in resolving multiple signals that are possibly
broadband and coherent motivate the derivation of a
NN
-aided
DoA
estimator capable of leveraging data to enable
MUSIC
to successfully operate in such scenarios.
We introduced deep augmented
MUSIC
(
DA-MUSIC
) in [1],
a hybrid
MB
/
DD
implementation of
MUSIC
, which exploits the
structure of the classic subspace algorithm while augmenting
it with
NN
s to learn to enhance its performance. The proposed
architecture overcomes the fundamental limitations of
MUSIC
,
enabling it to accurately detect the locations of coherent sources.
Our design builds upon the insight that the sensitivity to model
mismatch and inability to handle coherent and broadband
sources of
MUSIC
lie in its estimation of the noise and
signal subspaces through an
EVD
of the empirical covariance
matrix. Accordingly, the proposed
DA-MUSIC
improves this
crucial step by obtaining a surrogate, pseudo covariance matrix
through a recurrent
NN
(
RNN
) from the measurements, which
is learned along with a
NN
that acts as a peak finder. This
work presents crucial extensions to
DA-MUSIC
allowing it
to operate with an unknown and varying number of signal
sources by successfully estimating the number of sources
for non-coherent and coherent signals. An additional neural
augmentation allows the categorization into noise and signal
eigenvectors to be obtained in a learned fashion by training a
classifier. Numerical results show an estimation accuracy of
98% in comparison to
MUSIC
with 89% accuracy in the same
non-coherent scenario. We further demonstrate
DA-MUSIC
s
capabilities in various
SNR
domains, three different broadband
scenarios, as well as real-world seismic signals.
DA-MUSIC
not only manages to focus frequency components but is also
able to concurrently focus an interdependent elevation angle
to receive a stable azimuth estimation.
The rest of the paper is organized as follows: Section II
describes the assumed system model and surveys some of
the related work of
DoA
estimation, as well as the technical
details of the
MB MUSIC
algorithm; Section III introduces
the
DA-MUSIC
algorithm; Section IV presents the results of
the simulations; and Section V provides concluding remarks.
Throughout the paper, we use boldface lower-case letters for
vectors; e.g.,
x
, and boldface uppercase letters, e.g., for
X
. The
i
th entry of a vector
x
is denoted by
x
i
and entries separated by
commas represent column vectors. We use calligraphic letters
to denote the discrete-time Fourier transform, e.g.,
X(ω)
is
the frequency representation of a signal
x(t)
. We use
(·)
>
and
(·)
H
to denote the transpose and Hermitian operators
respectively, and
k·k
and
k·k
F
to denote the
`
2
and Forbenius
norms respectively. Further,
U
and
CN
represent the uniform
and the complex normal distributions. The symbol
I
denotes
the identity matrix.
II. SYSTEM MODEL AND PRELIMINARIES
In this section, we detail the system model for which we de-
velop the proposed
DA-MUSIC
algorithm in Section III. To that
aim, we first discuss the signal model in Subsection
II-A
. Then,
we formulate the
DoA
estimation problem in Subsection
II-B
,
and survey some of the related literature in Subsection
II-C
.
Since our proposed solution builds upon the
MUSIC
algorithm,
we recall
MUSIC
in Subsection
II-D
and briefly discuss the
CSS method for broadband extension in Subsection II-E.
A. Signal Model
We distinguish the considered signal model into two cases:
the conventional narrowband setting and its more general
broadband setup. For more details regarding the two models,
we refer the reader to the textbooks [10], [33], and [34].
1) Narrowband: The most common setup in the array
signal processing literature concerning
DoA
estimation is the
narrowband setting. Here, the time it takes for the waves to
propagate the array is assumed to be negligible such that the
occurring delays are sufficiently small. Therefore, the signal
measurement model for an arbitrary array structure consisting
of
M
sensor elements measuring
D
impinging narrowband
signals takes the following form:
x(t) = A(θ) s(t) + v(t). (1)
In
(1)
, the measurements
x(t) C
M
at time instance
t
depend
on the signals
s(t) C
D
, which originate from the unknown
angles
θ = [θ
1
, . . . , θ
D
]
, while
v(t) C
M
is additive white
Gaussian noise. The matrix
A(θ) C
M×D
contains the
steering vectors {a(θ
d
)}, i.e.,
A(θ) =
a(θ
1
) . . . a(θ
D
)
, (2)
where
{θ
d
}
are the source directions. For example, the steering
vector
a(ψ)
for a uniform linear array (
ULA
) with an element
spacing of
m = `/2
, where
`
is the wavelength of the signals,
is defined for direction ψ as
a(ψ) =
1, e
jπ sin ψ
, . . . , e
jπ(M 1) sin ψ
. (3)
2
DoA
Estimator
EVD
min
multiplicity
Noise
subspace
MUSIC Spectrum
1
a
H
( ) E
N
E
H
N
a( )
Peak
finder
MUSIC
K
X
E
N
P ( )
ˆ
D
1:M
v
1:M
ˆ
x(t)
t=1,...,T
ˆ
N
Input
DoAs
{a( )}
Observations
EVD
min
multiplicity
Noise
subspace
MUSIC Spectrum
1
a
H
( ) E
N
E
H
N
a( )
Peak
finder
MUSIC
K
X
E
N
P ( )
ˆ
D
1:M
v
1:M
ˆ
x(t)
t=1,...,T
ˆ
N
Input
DoAs
{a( )}
DoA
Estimator
EVD
min
multiplicity
Noise
subspace
MUSIC Spectrum
1
a
H
( ) E
N
E
H
N
a( )
Peak
finder
MUSIC
K
X
E
N
P ( )
ˆ
D
1:M
v
1:M
ˆ
x(t)
t=1,...,T
ˆ
N
Input
DoAs
{a( )}
Observations
EVD
min
multiplicity
Noise
subspace
MUSIC Spectrum
1
a
H
( ) E
N
E
H
N
a( )
Peak
finder
MUSIC
K
X
E
N
P ( )
ˆ
D
1:M
v
1:M
ˆ
x(t)
t=1,...,T
ˆ
N
Input
DoAs
{a( )}
Fig. 1: DoA estimation illustration.
Consequently, the steering vectors (and, in turn, the matrix
A(θ)
) specify the underlying array geometry. The collection
of the measurements at the array elements over multiple time
instances is defined as
X =
x(1) . . . x(T )
, (4)
where T is referred to as the number of snapshots.
2) Broadband: In practice, many signals are broadband,
and the delay caused by propagating the array aperture needs
to be incorporated into the signal model. The following
notation models the measurements received at array element
m {1, ..., M}
x
m
(t) =
D
X
d=1
s
d
(t τ
md
) + v
m
(t), (5)
where
τ
md
represents the time delay of the
m
th array element
measuring source
d {1, ...., D}
compared to the measurement
at the reference element
m = 1
. Transformed to the frequency
domain, the relationship in (5) at frequency ω becomes
X
m
(ω) =
D
X
d=1
e
jωτ
md
S
d
(ω) + V
m
(ω), (6)
which can in turn be written in vector-matrix form analogous
to the narrowband system model (1) as follows:
X(ω) = A(ω, θ) S(ω) + V(ω). (7)
For a
ULA
, the broadband steering vectors are defined for
frequency ω and angle of interest ψ as
a(ω, ψ) =
1, e
jω
m
c
sin ψ
, . . . , e
jω(M 1)
m
c
sin ψ
, (8)
where c is the propagation velocity.
B. Problem Formulation
DoA
estimation is concerned with localizing signal sources
by determining their angles of incidence utilizing the measure-
ments of an array aperture [33]. Formally, this corresponds to
estimating the angles
θ = [θ
1
, . . . , θ
D
]
from the measurements
x(t) = [x
1
(t), . . . , x
M
(t)]
measured at multiple time instances
t {1, ..., T }. An illustration of such a system is depicted in
Fig. 1. The following additional assumptions are imposed upon
the observation model as well as the derived synthetic data.
The signals are uncorrelated to the noise (though signals may
possibly be correlated with each other) and generated in the far-
field region. And there is uniform propagation in all directions
in an isotropic and non-dispersive medium. We further assume
that we have knowledge (though possibly mismatched) of the
underlying array geometry, implying that we can compute an
approximation of
a(ψ)
or
a(ω, ψ)
. We consider a data-driven
setting where we have access to training data. This data is a set
of
U
pairs,
{(X
u
, θ
u
)}
U
u=1
, each comprising the observations
and
DoA
angles from where the signals originated. In many
scenarios; e.g., in wireless communications, a specific training
set can be developed before deployment. If the ground-truth
DoA
s are not obtainable at all, the data-driven
DoA
estimator
must be trained by utilizing synthetic data that closely describes
the real signals. Our goal is thus to leverage the available
domain knowledge and data to design a system for recovering
the
DoA
s
θ
from a corresponding observation matrix
X
whose
columns are
T
snapshots of the measured waveforms at the
M sensors.
C. Related Literature
We next provide an overview of relevant
DoA
estimation
methods based on the above problem formulation, motivating
the need for our proposed
DA-MUSIC
detailed in Section III.
An extensive overview of various
MB DoA
estimators can
be found in [5], [6], and a recent literature review of
DD
DoA
estimation approaches can be found in [35]. We thus
only discuss some representative
MB
methods, followed by
reviewing relevant broadband extensions, and conclude with
DD
architectures for narrowband as well as broadband signals.
1)
MB
Narrowband Estimators: The conventional beam-
former (i.e. maximizing the steered response) is a basic
approach to
DoA
estimation and an extension of classical
Fourier-based spectral analysis [8]. Various improvements and
alternative beamforming methods have been developed, such
as the minimum variance distortionless response beamformer
[9] and other adaptive beamformers [36]. An overview of
beamforming techniques can be found in [37].
An alternative family of
DoA
estimators is based on subspace
methods, which aim at recovering the
DoA
s by identifying
the noise and signal subspaces. The
MUSIC
algorithm [7]
is a highly popular subspace-based method and has been
researched extensively. Being the focus of this paper, more
information and a detailed description of the algorithm can
be found in Section
II-D
. Extensions of
MUSIC
include Root-
MUSIC [38], a polynomial-rooting version, as well as spatially
smoothed
MUSIC
, which removes the correlation between the
incident signals by dividing the receiver array into overlapping
subarrays [39]. In practice, however, the number of coherent
sources is mostly unknown, and therefore the decorrelation
effect of spatial smoothing is not obvious [40]. Another popular
subspace-based method for
DoA
estimation is estimation of
signal parameters via rotational invariance techniques (
ESPRIT
)
[41] and its variations [42]. These methods rely heavily on the
accuracy of the underlying model assumptions, are generally
sensitive to array aperture perturbations, and are inherently
derived for narrowband signals.
2)
MB
Broadband Estimators: Broadband
DoA
estimation
algorithms can be categorized into two groups: incoherent
methods and coherent methods. Generally, incoherent methods
3
Empirical
cov.
EVD
λ
min
multiplicity
Noise
subspace
MUSIC Spectrum
1
a
H
(ψ) E
ˆ
N
E
H
ˆ
N
a(ψ)
Peak
finder
MUSIC
K
X
E
ˆ
N
P (ψ)
ˆ
D
λ
1:M
v
1:M
ˆ
θ
x(t)
t=1,...,T
ˆ
N
Input
DoAs
{a(ψ)}
Fig. 2: Block diagram of the MUSIC algorithm.
use independent frequency bins (
IFB
s) to process the
DoA
information at every frequency separately and then combine the
results of these narrowband
DoA
estimations. There are many
different variations in the implementation of this approach [43]–
[45], which typically vary in the computation of the covariance
matrices. Unfortunately, the computational complexity increases
with each frequency bin.
Conversely, coherent methods combine the covariance matri-
ces estimated for
IFB
s in order to apply narrowband techniques
directly over a single, so-called focused covariance matrix. An
overview of coherent techniques can be found in [10], [46].
A leading approach for coherent broadband
DoA
estimation
is based on the
CSS
method [11], with example estimators
given in [47]–[49]. Many variations of the
CSS
method are
concerned with the crucial aspect of the focusing strategy,
proposing different techniques to combine the covariances at
each
IFB
to obtain a faithful narrowband formulation. See,
for example, [50], [51] and the more recent summary [52].
These methods, however, experience significant bias with larger
angular sectors of interest and can impose additional model
assumptions such as noise statistics.
3)
DD
Estimators: Inspired by the dramatic success of
DL
in computer vision and natural language processing, recent
years have witnessed a growing interest in the application of
NN
s for
DD DoA
estimation. A recent review of
DL DoA
estimation approaches can be found in [53] and essential
methods were covered in Section I.
DL
architectures presented
in the numerical evaluations or otherwise closely related to
this work are summarized and discussed below.
The work [18] implemented a
CNN
architecture for
DD DoA
estimation based on a multi-label classification task. Particularly,
the method takes real, imaginary, and phase information of
the sample covariance as a three-channel input and predicts a
probability grid of directions. Given the binary cross-entropy
loss for training, the method is capable of operating with a
potentially unknown and varying number of sources. However,
with a growing number of sources, the number of classes grows
exponentially.
Similarly to the above, the methods in [25], [26] propose
CNN
s for
DoA
estimation by taking the same channels of
the sample covariance matrix as input (spatially smoothed
covariance of single snapshot for [26]). However, the
CNN
s
are trained as a regression task estimating a discretized segment
of the
MUSIC
spectrum. Thereby, they not only inherit the
drawbacks and imitations of
MUSIC
but are, besides increased
robustness, dependant on the underlying model assumptions.
The sample covariance matrix estimate deviates from the
actual covariance, especially with multiple possibly coherent
broadband signals which affect
MB
as well as
DD
performance.
The work [27] considered systems with subarray sampling and
trained a
NN
for covariance matrix reconstruction. The method
takes subarray covariance matrices as input and predicts a full
covariance matrix. This
NN
-aided estimate is then utilized
for
DoA
recovery with the
MUSIC
algorithm. While this
approach addresses the fundamental dependency of
MUSIC
and other doa estimators on the estimated covariance matrix, it
is still dependent on their subarray estimate, and cannot further
influence or react to MUSIC’s performance.
The approach presented in [28] proposed a hybrid
MB
/
DD
framework with dense
NN
s for localization and estimation of
the number of sources. To reduce the mentioned issues of the
sample covariance matrix the framework computes a
FTMR
matrix from the measurements. Then an
EVD
decomposes
it into eigenvalues and eigenvectors which are utilized to
predict the number of sources present and produce a pseudo
MUSIC
spectrum, respectively. Trained similarly to [18], the
localization
NN
input, however, is directly dependent on the
selection of noise eigenvectors which makes this method less
robust to an incorrect estimation of the number of sources.
D. MUSIC Algorithm
As our proposed
DA-MUSIC
algorithm originates from the
MUSIC
algorithm, we next present this method in detail.
MUSIC
, originally proposed by Schmidt in [7], considers
the narrowband signal model with incoherent sources
(1)
,
where the signals in
s(t)
are mutually independent. Fig. 2
visualizes a simple outline of the
MUSIC
structure as a block
diagram. The approach takes the empirical covariance matrix
of the received measurements
X
, then conducts an
EVD
,
followed by categorizing the eigenvectors into signal and noise
subspaces. The orthogonality between the two subspaces allows
the formulation of a spatial spectrum, which contains peaks at
DoA angles.
1) Formal Derivation: With the assumption of the signal
and the noise being uncorrelated, the covariance matrix of
x(t)
is given by
K
X
= A(θ)K
S
A
H
(θ) + λK
0
X
, (9)
with
K
S
being the covariance of the incident signals
s(t)
.
The matrix
A(θ)K
S
A
H
(θ)
is singular and has a rank of less
4
RNN
EVD
NN
Selection
MUSIC Spectrum
1
a
H
(ψ)
˜
E
˜
E
H
a(ψ)
NN
Deep Augmented MUSIC
˜
K
˜
E
P (ψ)
ˆ
D
λ
1:M
v
1:M
ˆ
θ
x(t)
t=1,...,T
ˆ
N
Input
DoAs
{a(ψ)}
Z
1
Fig. 3: Block diagram of the DA-MUSIC algorithm.
than
M
when the number of array elements
M
is strictly
larger than the number of signals
D
. Therefore,
λ
is an
eigenvalue of
K
X
(in the metric of
K
0
X
, which takes the form
K
0
X
= σ
2
I
for additive white Gaussian noise (
AWGN
) with
variance
σ
2
). Further,
A(θ)K
S
A
H
(θ)
has to be non-negative
definite, because
A(θ)
has full rank and
K
S
is positive definite,
and consequently,
λ
in
(9)
is the minimal eigenvalue of
K
X
,
denoted
λ
min
. The multiplicity of
λ
min
corresponds to the
number of incident wavefronts, and equals N = M D.
MUSIC
builds upon this representation of the covariance
of the signals. The algorithm takes as input
T
snapshots of
the waveforms at
M
array elements, represented as
X
in
(4)
,
and uses them to obtain an empirical estimate of
K
X
via
ˆ
K
X
=
1
T
XX
H
. Then, the number of incident signals
D
is
estimated via
ˆ
D = M
ˆ
N, (10)
where
ˆ
N
is the estimated multiplicity of the minimal eigenvalue
of
ˆ
K
X
. The eigenvectors corresponding to the
ˆ
N
smallest
eigenvalues form the noise subspace
E
ˆ
N
, which is orthogonal
to the
D
dimensional signal subspace spanned by the incident
signal mode vectors. Consequently,
MUSIC
estimates the
DoA
s
by computing the spatial spectrum
P (ψ) =
1
a
H
(ψ)E
ˆ
N
E
H
ˆ
N
a(ψ)
, (11)
and the
ˆ
D
dominant peaks of
P (ψ)
are set as the estimated
DoA angles
ˆ
θ.
MUSIC
is a popular and highly applicable subspace-based
method that is reasonably efficient and statistically consis-
tent [5], [54]. When the signal model is adequately accurate,
it can achieve super-resolution and deliver a highly accurate
estimate of the number of signal sources present. Nevertheless,
the algorithm is sensitive towards the accuracy of the empirical
estimate of
K
X
, and cannot reliably estimate the
DoA
angles
as well as the number of sources of coherent signals. The
reason for this is that highly correlated signals cause zero
entries within the covariance matrix and can, therefore, become
indistinguishable from noise [55]. Furthermore, the
MUSIC
algorithm is inherently a narrowband approach due to the
assumptions imposed on the system model. Nonetheless, it can
be extended to broadband, i.e., signal models as in
(7)
, and so
we next describe the coherent method to achieve this, which
is also adopted in our proposed DA-MUSIC.
E. Coherent Broadband
The main concept behind coherent broadband
DoA
esti-
mation is to transform the different frequency covariance
matrices into a single covariance matrix at a focusing frequency.
Accordingly, coherent methods find appropriate transformations
for each frequency, transform the covariances, and obtain a
focused covariance matrix by some form of averaging. In
particular, the
CSS
method [11] aims at combining the spatial
signal subspaces to align the signal subspaces associated with
the
DoA
along all frequency bins. To formulate this, let
K(ω)
be the covariance of the frequency domain observations
(7)
,
and divide the spectrum into
B IFB
s with central frequencies
{ω
b
}
B
b=1
. The focused covariance matrix, which is used as the
input covariance for narrowband
DoA
recovery, is estimated
as
K =
B
X
b=1
α
b
T
b
K(ω
b
) T
H
b
, (12)
where
T
b
is the focusing matrix and frequency bins are
prioritized by the weighting
α
b
. The focusing matrices can be
determined by attempting to focus the spectral components at
some frequency ω
r
and some focusing angles ψ by solving
T
b
= arg min
T
kA(ω
r
, ψ) T A(ω
b
, ψ)k
F
. (13)
Coherent techniques have been shown to achieve a better
estimation accuracy and a smaller computational complexity
as well as a lower resolution threshold than non-coherent
methods [52]. However, the
CSS
methods require initial values
for the focusing matrix
T
, the reference frequency
ω
r
, and the
relevant focusing angles
ψ
to find the focusing matrices with
(13)
, and are typically sensitive towards these initial values.
Additionally, it is not guaranteed that the alignment of the
signal and noise subspaces exists to form a viable general
covariance matrix without disarranging the noise subspace
[51].
III. DEEP AUGMENTED MUSIC
DA-MUSIC
is a hybrid
MB
/
DD DoA
estimation algorithm
derived from the classic
MUSIC
algorithm by replacing crucial
and model mismatch sensitive elements of the model-based
structure with specific
NN
s. Fig. 3 outlines the resulting
structure of the
DA-MUSIC
architecture and highlights the
remaining similarities to the original
MUSIC
algorithm, de-
picted in Fig. 2.
5
Fig. 4: Detailed network structure of the
DA-MUSIC
algorithm.
Principally,
DA-MUSIC
builds upon the understanding
that the core challenges associated with the classic
MUSIC
algorithm can be tackled by providing a surrogate covariance
matrix. In particular, as the
CSS
method provides a surrogate
covariance
K
that transforms a broadband signal model into
a narrowband one, a similar approach can be employed to
handle coherent sources and array mismatches. Therefore, to
improve the categorization of the noise and signal subspaces,
the correlation of the received measurements is learned from
temporal data by employing a dedicated
RNN
, which is
augmented into the overall flow of
MUSIC
. We next elaborate
on the architecture in Subsection
III-A
, after which we
present the training method in Subsection
III-B
and provide a
discussion in Subsection III-C.
A. Architecture
The proposed
DA-MUSIC
algorithm preserves the structure
of the
MB MUSIC
while replacing certain critical aspects with
NN
s. Our neural augmentations aim to improve the crucial
steps of estimating the noise and signal subspaces from the
empirical covariance and the translation of the spatial spectrum
into
DoA
s via peak finding. By doing so,
DA-MUSIC
is not
constrained by the additional model assumptions imposed
in the derivation of
MUSIC
, and can, as we will show, e.g.
learn to successfully localize coherent signals. To present the
architecture of
DA-MUSIC
, we commence with the simple
case where the number of sources
D
is known, and then show
how its estimation is incorporated. Details of the
NN
s used in
our experimental study are reported in Section IV.
1) Known Number of Sources: Fig. 4 depicts a detailed
outline of the individual elements of the
DA-MUSIC
architec-
ture. The respective output dimensions of the corresponding
components are given in the bracket notation. First, the input
signal
x(t)
is transformed into the pseudo covariance matrix
˜
K
using a
RNN
implemented through a gated recurrent unit
(
GRU
). The final state of the
GRU
is passed to a dense layer
enabling a reshaping to the desired dimension of the pseudo
covariance matrix
˜
K
as well as the subsequent transformation of
the complex space. Then, through the continued use of the
EVD
,
the algorithm categorizes the subspaces using the eigenvectors.
Inserting the steering vectors
a(ψ)
allows to compute an
estimate of the spatial spectrum in
(11)
, denoted
P (ψ)
,
identically to
MB MUSIC
, by using the noise eigenvectors
selected from
˜
K.
Next,
DA-MUSIC
attains the
DoA
s from the spatial spec-
trum
P (ψ)
using an additional
NN
, comprised of a multi-
layer perceptron (
MLP
) of three fully connected dense layers
followed by a single dense layer with linear activation. The
input to the
MLP
are
R
samples of
P (ψ)
taken uniformly
in
[0, 2π)
. The output of the
MLP
is the set of estimated
DoA angles
ˆ
θ. The benefits of using a NN-based peak-finder
compared to a model-based one are two-fold. First, learning the
translation of the pseudo-spectrum into
DoA
s from data enables
achieving improved resolution compared to conventional peak
finding, since
ˆ
θ
d
[0, 2π)
instead of being dependent on the
number of angles
ψ
used to evaluate the spectrum. Furthermore,
peak finding is generally non-differentiable; thus, replacing
it with a
NN
facilitates training
DA-MUSIC
end-to-end. The
resulting architecture enables the application of gradient-based
optimization, by propagating through the
NN
s as well as the
EVD
operation, as done in [56]. Doing so allows us to jointly
tune the noise subspace recovery along with the translation of
the
MUSIC
spectrum into
DoA
s by comparing its estimated
DoAs with the true DoAs, as we detail in Subsection III-B.
2) Varying Number of Sources: Delivering unbiased esti-
mates of the number of signal sources as well as the ability
to successfully localize these sources makes
MUSIC
highly
applicable. The above-discussed DA-MUSIC architecture can
be extended to operate with a potentially unknown and varying
number of signal sources, despite the determinant nature of
the
NN
s. This is achieved by addressing three key aspects of
the algorithm: the estimation of the number of sources; the
selection process of the noise subspace from the eigenvectors;
and the adaptation of the output strategy to overcome the
varying number of DoA angles.
Estimating number of sources:
Our design allows
DA-MUSIC
to learn abstract mappings as pseudo covariance
features
˜
K
, which is geared towards end-to-end training and
is not restricted to a natural ordering of the model-based
covariances. Consequently, instead of estimating the number of
sources by inspecting the magnitude of its eigenvalues, we opt
a data-driven approach. We augment the process of estimating
the number of sources with a classifier, implemented through
a
MLP
, as a classification task. Since subspace methods with
M
inputs can resolve at most
M 1
signals, the classifier
has
M 1
classes. Fig. 5 depicts the
DA-MUSIC
architecture
with an added classifier taking the eigenvalues as an input and
outputting an estimate of the number of sources.
Computing noise subspace:
The noise subspace selector
needs to know the number of sources
D {1, ..., M 1}
to
classify the eigenvectors into signal and noise subspaces; i.e.,
by choosing the eigenvectors corresponding to the
N = M D
smallest eigenvalues. When the number of sources is not known,
we propose to weight each eigenvector by an estimate of it
belonging to the noise subspace. We do this by introducing an
additional neural augmentation, depicted in Fig. 6, which uses
a MLP to cluster the eigenvalues in a learned fashion.
In particular, the
MLP
maps the estimated
M
eigenvalues
into a vector
q =
q
1
, ..., q
M
whose entries hold the individual
probabilities of choosing the corresponding eigenvector as a
noise eigenvector. The selection is performed by computing
˜
E = Vdiag(q) =
q
1
v
1
. . . q
M
v
M
, allowing to learn a
suitable noise subspace. Note that this setting specializes
in the conventional approach of assigning based on the
multiplicity of the minimal eigenvalues; in the conventional
approach, the entries of
q
are either ones or zeros and
˜
E
coincides with the corresponding subspace. Consequently, the
6
GRU dense
reshape complex
EVD
selector
spectrum
MLP dense
MLP dense
x(t)
(2 · M, T )
a(ψ)
h
(2 · M )
(2 · M · M)
(2 · M, M )
˜
K
(M , M)
λ
(M)
λ
(M)
(2 · M
2
)
(M 1)
ˆ
D
V
(M , M)
˜
E
(M , M)
P (ψ)
(R)
(2 · M )
(M 1)
ˆ
θ
Fig. 5:
DA-MUSIC
algorithm with a separately trained (inter-
nal) classifier.
GRU dense
reshape complex
EVD
real MLP dense matrix
·
selector
spectrum
MLP dense
MLP dense
x(t)
(2 · M, T )
a( )
h
(2 · M )
(2 · M ·M)
(2 · M, M )
e
K
(M,M)
(M)
(M)
(2 · M
2
)
(M 1)
ˆ
D
(M)
V
(M,M)
(2 · M )
(2 · M )
q
(M)
diag(q)
(M,M)
(M,M)
˜
E
V
(M,M)
˜
E
(M,M)
P ( )
(R)
(2 · M )
(M 1)
ˆ
selector
Fig. 6: Detailed outline of the
DA-MUSIC
subspace selection
augmentation.
proposed approach provides additional flexibility in selecting
the noise subspace and facilitates coping with settings where
distinguishing between the eigenvectors is challenging due, for
example, to low SNRs.
Outputting varying number of DoAs:
The most common
strategy to overcome the dynamic output dimensions of
NN
s is
to scale the output dimension to the maximum occurring value.
In our case, since
MUSIC
cannot resolve more than
M 1
sources, the dimension of the last dense layer of
DA-MUSIC
is set to
M 1
. The only additional modification required with
this strategy compared with known
D
is the slight alteration
to the loss function discussed in Subsection
III-B
below. An
advantage of this strategy is that the approach allows to extract
up to
M 1 DoA
, where
θ
1
is most likely a true
DoA
angle,
θ
2
has a slightly lower probability to be a
DoA
angle, etc. The
final estimation is thus carried out by first taking
ˆ
D
from the
module that estimates the number of sources, and then using
the first
ˆ
D outputs as the recovered DoAs.
B. Training Procedure
DA-MUSIC
is trained end-to-end in a supervised setting as
a multiple regression problem. As detailed in Subsection
II-B
,
the training set is comprised of
U
tuples of sequences of
measurements and their corresponding
DoA
s; i.e., the
u
th
tuples
includes the
T
u
measurements
X
u
and their corresponding
D
u
DoA
angles
θ
u
. We first describe how this data is used for
training when
D
is known and fixed; i.e.,
D
u
D
. This acts
as a preliminary step for discussing how the training procedure
is carried out in the general case where the model does not
have knowledge D.
1) Known Number of Sources: Given a sequence of mea-
surements
X
as input, the model predicts the estimated
DoA
angles
ˆ
θ
that are compared to the true
DoA
angles
θ
. Gradient-
based optimization is possible because every element of the
architecture is differentiable, allowing backpropagation through
the complete structure. Derived from the root mean squared
periodic error (
RMSPE
) [57], [58] the following loss function
additionally compares all permutations of the predicted angles
with the true angles to capture all possible assignments of
the estimated
DoA
to the true
DoA
. Thereby the minimal
permutation
RMSPE
includes the permutation invariance of
the DoAs and is obtained as:
RMSPE(θ,
ˆ
θ) = min
P∈P
D
1
D
mod
β
(θ P
ˆ
θ)
2
1
2
, (14)
where
P
D
is the set of all
D × D
permutations and
mod
β
denotes the element-wise modulus operation regarding the
angle range of interest, e.g.
β = π
for
ψ [π/2, π/2)
or
β = 2π for ψ [0, 2π).
2) Unknown Number of Sources: As discussed in the
previous subsection,
DA-MUSIC
is designed to resolve a
varying and unknown number of sources by introducing an
additional
NN
classifier for the number of sources. To formulate
the training procedure of the overall system, we use
w
c
to
denote the trainable parameters of the classifier, while
w
d
represents the parameters of the remaining trainable modules
of
DA-MUSIC
(covariance estimator, peak finder, and subspace
selector). The loss used to train
w
d
is the
RMSPE
loss of
(14)
,
which is altered during training to account for varying sources
by computing,
L
RMSPE
θ
u
,
ˆ
θ(X
u
)
= RMSPE
θ
1:D
u
,
ˆ
θ
1:D
u
(X
u
)
. (15)
In
(15)
,
ˆ
θ(X
u
)
denotes the
M 1
outputs of
DA-MUSIC
applied to
X
u
, while
θ
1:D
u
= (θ
1
, ..., θ
D
u
)
. This means that
only the first
D
u
angles of the estimated
DoA
ˆ
θ
are compared
with the true
DoA θ
while the remaining angles of
ˆ
θ
are
completely ignored.
The separate classifier is trained using the categorical cross-
entropy of the classes as a loss function ensuring optimal
training. In particular, letting
λ
u
be the input to the
MLP
classifier when applying
DA-MUSIC
to
X
u
and letting
ˆ
D(λ
u
)
be the softmax output of the
MLP
applied to
λ
u
(with
ˆ
D
i
(λ
u
)
being its ith entry), the loss used for training w
c
is given by
L
CE
D
u
,
ˆ
D(λ
u
)
= log
ˆ
D
D
u
(λ
u
). (16)
It is noted that since
(16)
is used for training
w
c
, then one
should block the gradient during backpropagation from passing
from the classifier to the
EVD
, as indicated by the dashed
connections in Fig. 5. This ensures that the
MLP
learns from the
eigenvalues themselves and not by influencing and disrupting
the
GRU
. Further, the regressor is completely independent
of the classifier, which allows
DA-MUSIC
to operate with
different classifiers if needed or with any other desired scheme
delivering an estimate of D. The resulting training procedure
(employing mini-batch gradient descent with ADAM [59]) is
summarized as Algorithm 1.
C. Discussion
The design of the architecture of
DA-MUSIC
is derived from
the model-based
MUSIC
structure. This allows for exploiting
the successful aspects of the algorithm while improving
7
Algorithm 1: Training DA-MUSIC
Data: Data set {(X
u
, θ
u
)}
U
u=1
, learning rate µ = 0.001,
decay rates b
1
= 0.9, b
2
= 0.999, = 10
8
1 Initialize weights w
d
, w
c
;
2 Initialize moment vectors ν
d
, υ
d
, ν
c
, υ
c
;
3 for epoch = 1, 2, . . . do
4 for each batch do
5 Apply DA-MUSIC to {X
u
} for u batch;
6 Compute gradients g
d
via
g
d
w
d
P
ubatch
L
RMSPE
(θ
u
,
ˆ
θ(X
u
));
7 Compute gradients g
c
via
g
c
w
c
P
ubatch
L
CE
(D
u
,
ˆ
D(λ
u
));
8 Update biased first moment via
9 ν
d
b
1
ν
d
+ (1 b
1
)g
d
;
10 ν
c
b
1
ν
c
+ (1 b
1
)g
c
;
11 Update biased second raw moment via
12 υ
d
b
2
ν
d
+ (1 b
2
)g
2
d
;
13 υ
c
b
2
ν
c
+ (1 b
2
)g
2
c
;
14 Compute bias corrected moments
ˆ
ν
d
,
ˆ
ν
c
,
ˆ
υ
d
,
ˆ
υ
c
;
15 Update w
d
via w
d
w
d
µ
ˆ
ν
d
/(
ˆ
υ
d
+ );
16 Update w
c
via w
c
w
c
µ
ˆ
ν
c
/(
ˆ
υ
c
+ );
17 end
18 end
certain critical elements and alleviating important drawbacks.
Replacing the empirical covariance estimation with a
RNN
is the key neural augmentation of
DA-MUSIC
, enabling the
system to learn the pseudo covariance from the measurements
themselves such that the resulting surrogate model facilitates
subspace-based
DoA
recovery. Thereby, the performance of
DA-MUSIC
, for example, is not affected by coherent signals
and other related issues discussed in Subsection
II-D
. Further-
more, learning end-to-end allows
DA-MUSIC
to operate with
broadband signals, as it effectively learns to produce a focused
pseudo covariance, similarly to the
CSS
methods discussed in
Subsection
II-C
. It is noted that our augmentation approach
depends on the array geometry, as the steering vectors
a(·)
are used for computing the
MUSIC
spectrum. Nonetheless, as
we numerically demonstrate in Section IV,
DA-MUSIC
learns
to overcome mismatches in the array geometry from the data
without any alterations or renewed training.
Specifically, the
RNN
utilized by
DA-MUSIC
is able to learn
an appropriate focusing matrix while concurrently correlating
the measurements comparably to
(12)
. The usage of a
RNN
applied to the time-domain signal and only passing the last state
to the next layer allows
DA-MUSIC
to operate with different
signal durations and possibly cope in real-time with dynamic
variations in the
DoA
s, though investigation of the latter is left
for future work. Our experiments indicate that a deeper
RNN
aids the correlation aspect while a wider
RNN
allows a more
complicated mapping with this correlation.
Another important component of the architecture of
DA-MUSIC
is its incorporation of the
EVD
as a means for
division into subspaces. Though computationally expensive, the
internal
EVD
allows
DA-MUSIC
to not only classify signal
and noise subspaces with the eigenvalues, but also significantly
simplifies the estimation of the number of signal sources
present.
To enable training end-to-end from the errors in
(14)
, a
TABLE I: Simulation parameters.
Parameter Description Notation Default Value
Array geometry ULA
Number of array elements M 8
Element spacing m `
min
/2
SNR 10 dB
Snapshots T 200
Grid points of continuum R 360
Min. frequency f
min
0 [Hz]
Max. frequency f
max
999 [Hz]
Sampling frequency f
s
2(f
max
+ 1)
Time length T
samp
1 s
Fast Fourier transform (FFT) points N
f
T
samp
· f
s
NN
-based peak finder is used in form of a
MLP
. Model-based
peak-finding is generally non-differentiable; therefore, replacing
it with a
NN
enables gradient-based optimization through the
entire
DA-MUSIC
structure. Furthermore, improved resolution
can be achieved by extracting the
DoA
from the spatial
spectrum in a learned manner (i.e.,
ˆ
θ
d
[0, 2π)
and it is
not dependent on the number of angles
ψ
used to evaluate the
spectrum P (ψ)).
IV. NUMERICAL EVALUATIONS
In this section, we present our numerical evaluations of the
proposed
DA-MUSIC
algorithm
1
Our experimental study is
comprised of evaluations in a narrowband synthetic setting
(Subsection
IV-A
); a broad synthetic setup (Subsection
IV-B
);
and experiments with real-world data corresponding to azimuth
estimation in seismic arrays (Subsection IV-C).
A. Synthetic Narrowband
The numerical evaluations of synthetic data presented below
are obtained by simulating the measurements
x(t)
according
to the narrowband system model
(1)
. In particular, we simulate
a
ULA
with
M = 8
array elements that measure impinging
waveforms originating from the
DoA
angles
θ = [θ
1
, . . . , θ
D
]
,
which are separately drawn from the uniform distribution
U(π/2, π/2)
. The signals
s(t) = [s
1
(t) . . . s
D
(t)]
>
are
each drawn randomly from the complex Gaussian distribution
CN(0, 1)
for all
t
modeling random amplitudes and phases.
The noises measured at the
M
array elements
v(t) =
[v
1
(t) . . . v
M
(t)]
>
are also drawn from
CN(0, 1)
for all
t
,
followed by appropriate scaling to meet the constant
SNR
.
In the coherent cases, all signals have identical amplitudes
and phases, and when not stated otherwise, the simulation
parameters are set according to Table I.
1) Known Number of Sources: We first evaluate the
RMSPE
in [rad] achieved when the number of sources
D
is known in
the synthetic narrowband scenario described above. The results,
reported in Table II, compare the performance of the following
DoA estimators for a different number of sources D:
The
DA-MUSIC
architecture is implemented according
to Fig. 3 and trained separately for each case D.
The classic
MB
MUSIC algorithm, implemented as
described in Section
II-D
, utilizes the external knowledge
of D.
1
The source code used in our experiments can be found at https://github.
com/DA-MUSIC/TVT23.
8
Fig. 7: DoA estimation of D = 2 closely spaced sources.
Fig. 8: DoA estimation with mismatch in the array geometry
for D = 5 sources.
The
CNN
architecture of [18], with a reduced stride in
the first layer to account for
M = 8
, and an increased
grid-size for the last layer to achieve R = 360.
The
DD
DeepMUSIC proposed in [25], while incorporat-
ing minor alterations that were necessary to accommodate
for the difference in the setup, includes tuning of individual
hyperparameters to assure successful training of the
CNN
s.
The above algorithms are compared with a random guessing of
the
DoA
angles. The results in Table II show that the proposed
DA-MUSIC
notably outperforms all considered benchmarks,
notably surpassing the
MB
MUSIC algorithm not only for
coherent sources, but also for non-coherent ones, which is the
scenario for which the MB algorithm is designed.
TABLE II:
RMSPE
of different
DoA
estimation algorithms
with constant and known D for T = 200 snapshots.
RMSPE
[rad]
DA-MUSIC CNN
Deep-
MUSIC
Classic
MUSIC
Random
non-coherent
D = 2 0.0117 0.0237 0.0329 0.0336 0.6809
D = 3 0.0315 0.0464 0.1600 0.0841 0.6034
D = 4 0.0563 0.0841 0.2656 0.1459 0.5421
D = 5 0.0751 0.1319 0.2701 0.2008 0.4963
coherent
D = 2 0.0140 0.0274 0.5781 0.2350 0.6854
D = 3 0.0407 0.0528 0.4023 0.4819 0.6060
D = 4 0.0519 0.0859 0.2915 0.4522 0.5407
D = 5 0.0658 0.1254 0.3054 0.4401 0.5000
Fig. 9: DoA estimation of
D = 5
signals with different SNRs.
Fig. 10: DoA estimation of
D = 5
signals with different
number of snapshots T .
To compare the resolution of the algorithms, Fig. 7 shows
the
RMSPE
for localizing
D = 2
non-coherent signals, which
are located close together at a
θ
distance from each other. The
MB MUSIC
algorithm is shown to collapse when the angular
difference approaches
θ 0.1
radians, while
DA-MUSIC
demonstrates a constant low error for all
θ
, indicating its
improved resolution.
Next, we evaluate
DA-MUSIC
in the presence of a mismatch
in the array geometry. Fig. 8 depicts the
RMSPE
achieved
when each element of the steering vector
a(·)
is corrupted
with zero-mean Gaussian noise, leading to a mismatch from
the values used to compute the spatial spectrum. Indicating
improved robustness,
DA-MUSIC
is shown to overcome such
mismatches in the array geometry from the data. The
CNN
,
TABLE III:
RMSPE
of different
DoA
estimation algorithms
with varying and unknown D for T = 200 snapshots.
RMSPE
[rad]
DA-MUSIC CNN
Classic
MUSIC
Beamformer Random
non-coherent
D = 2 0.0430 0.0403 0.0428 0.1900 0.8318
D = 3 0.0705 0.0842 0.0917 0.2906 0.6981
D = 4 0.0894 0.1463 0.1489 0.3757 0.6021
D = 5 0.1222 0.1847 0.1856 0.4029 0.5357
coherent
D = 2 0.0383 0.0376 0.6139 0.1670 0.8243
D = 3 0.0688 0.0729 0.5743 0.2843 0.6962
D = 4 0.0869 0.1181 0.5258 0.3883 0.6051
D = 5 0.1046 0.1652 0.4744 0.4179 0.5435
9
Fig. 11: Accuracy of estimating
ˆ
D
for various
T
with
D {2, ..., 5} non-coherent signals.
Fig. 12: Accuracy of estimating
ˆ
D
for various
T
with
D {2, ..., 5} coherent signals.
deepMUSIC, and the Random algorithm remain unaffected as
they are independent of a(·).
Fig.9 depicts the performances differences when localizing
D = 5
non-coherent signals with
T = 200
snapshots
available for different
SNR
levels in the range of
[20, 20]
dB.
DA-MUSIC
shows a constant low
SNR
for positive dB
settings and without any fluctuations which slowly decreases
with increasing SNR.
We conclude the evaluation of
DA-MUSIC
with a known
number of sources by considering the case in which the
number of available snapshots
T
is varied. Fig. 10 depicts the
performance degradation of the estimators with fewer snapshots
available. The
DD
estimators are only trained for the case
T = 200
as indicated by the circle around the
T = 200
marker, yet manage to operate with shorter sequences during
inference due to the recurrent unit or by taking the covariance
matrix as input.
2) Unknown and Varying Number of Sources: Table III
shows the results obtained in the exact same narrowband
scenario introduced in Section
IV-A
above but with an
unknown and varying number of sources
D {2, ..., 5}
. Here,
during inference, the
DoA
estimation algorithms do not have
any knowledge of the varying number of sources present.
To compute the
RMSPE
in such settings (i.e., if the
DoA
estimators output the wrong number of sources
ˆ
D
), the
DoA
are
either truncated (least dominant peaks for the
MB
algorithms
and the
CNN
and highest indexed
DoA
angles for
DA-MUSIC
)
Fig. 13: RMSPE for varying and unknown number of
D {2, ..., 5} non-coherent signals.
Fig. 14: RMSPE for varying and unknown number of
D {2, ..., 5} coherent signals.
or padded with random
DoA
angels until
|
ˆ
θ| = |θ| = D
. We
compare the following DoA estimators:
The
DA-MUSIC
architecture is implemented according
to Fig. 5 with the classifier which predicts the number
of sources being trained along with the overall
DoA
estimation method.
The
DA-MUSIC
(RTC) variation is also implemented
according to Fig. 5, yet with a retroactively trained
classifier (RTC), i.e., we first train
DA-MUSIC
for a
specific scenario, and then we fix the
DoA
estimator
modules and train only the classification network for
alternate scenarios.
The Classic MUSIC algorithm is implemented as before
but determines the number of sources by estimating the
multiplicity of the smallest eigenvalue utilizing a pre-
determined threshold.
The CNN of [18] as introduced above.
A conventional beamformer [8], utilizing a peak-finder
to estimate the number of sources by determining the
number of dominant peaks.
The Random algorithm corresponds to the base perfor-
mance when choosing DoA angles at random.
Figs. 11 and 12 show the accuracy in identifying the number
of sources versus the number of snapshots
T
obtained by
the mentioned algorithms during the estimation of
ˆ
D
for
non-coherent and coherent signals, respectively. Again, the
DD
estimators are only trained for the case
T = 200
as
10
TABLE IV: Simulation parameters of the broadband scenarios.
Parameter Description Notation Value
Scenario 1 Modulation frequency f
c,d
0 999 [Hz]
Scenario 2
Number of subcarriers K 1000
Signal bandwidth f
d
1000 [Hz]
Scenario 3
Modulation frequency f
c,d
0 899 [Hz]
Number of subcarriers K 10
Signal bandwidth f
d
100 [Hz]
indicated by the circle. Unfortunately, the performance of the
internal classifier of
DA-MUSIC
is dependent on the number of
snapshots, and to be able to maintain a more constant accuracy
it must be trained for each case. Consequently,
DA-MUSIC
(RTC), which trains its classifier retroactively, requires separate
training for each number of snapshots, yet manages to achieve
the most accurate predictions.
The corresponding performances in localizing the varying
number of sources are depicted in Figs. 13 and 14 for
non-coherent and coherent signals, respectively. The shown
RMSPE
is an average over all considered
D {2, ..., 5}
. The
fundamental limitation of the
MB MUSIC
structure to estimate
the number of signal sources for coherent signals also severely
impacts the localization abilities of the algorithm.
B. Synthetic Broadband
We proceed to evaluate
DA-MUSIC
in a broadband setting.
The previously introduced synthetic environment requires
certain alterations and assumptions to account for broadband
signals. The sensor elements of the receiving array are ade-
quately spaced, and the element spacing is therefore assumed
to be
m =
`
min
2
=
1
2
c
f
max
, (17)
where
`
min
is the minimal wavelength corresponding to
the maximal occurring frequency
f
max
and the frequency
spectrum of interest is considered to be within
[f
min
, f
max
]
.
The measurements are simulated utilizing the broadband system
model
(7)
, where the elements of
S(ω)
and
W(ω)
are the
N
f
-
point
FFT
s of the elements of
s(t)
and
w(t)
. The parameter
values of the different broadband scenarios are specified in
Table IV if not specified otherwise. We simulate the following
DoA estimators:
The
DA-MUSIC
architecture is implemented according
to Fig. 3, but the
GRU
parameters are scaled (having
10 times more parameters available) to enable optimal
learning despite the more complex broadband scenarios.
The classic MUSIC algorithm is implemented in its
narrowband format as described in Section
II-D
and
utilizes steering vectors calibrated to the exact array
element spacing using `
min
/2.
Broadband
MUSIC
corresponds to an incoherent broad-
band extension of
MUSIC
[46] and is implemented using
10 [Hz]
per
IFB
; i.e.,
|ω
b
ω
b1
| = 10 [Hz]
for all
b {1, ..., B}.
The
DoA
estimators are again compared to choosing
DoA
angles at random.
We consider the following three different signal models of the
broadband signals s
d
(t) for d {1, ..., D}:
Fig. 15: RMSPE for known number of
D = 2
signals from
Broadband Scenario 1.
Fig. 16: RMSPE for known number of
D = 2
signals from
Broadband Scenario 2.
Fig. 17: RMSPE for known number of
D = 2
signals from
Broadband Scenario 3.
1) Broadband Scenario 1: A broadband signal is obtained as
narrowband signals modulated on different carrier frequencies,
i.e.,
s
d
(t) = ¯s
d
exp (2πjf
c,d
t), (18)
where for each
d {1, ..., D}
, both
¯s
d
and
f
c,d
are randomly
drawn from CN(0, 1) and U(f
min
, f
max
) respectively.
2) Broadband Scenario 2: Broadband signals are obtained
via orthogonal frequency division multiplexing (
OFDM
) [60],
which are modulated on the same carrier frequency. The signals
are considered in baseband and take the following form
s
d,OFDM
=
1
K
K1
X
k=0
¯s
k
exp (2πjkf
d
t/K), (19)
11
where
¯s
k
CN(0, 1)
is randomly drawn for each of the
K
subcarriers and the bandwidth is f
d
= f
max
f
min
.
3) Broadband Scenario 3: A combination of the two
previous scenarios and consists of
OFDM
signals modulated
on different carrier frequencies
s
d
(t) = exp (2πjf
c,d
t) s
d,OFDM
, (20)
where
f
c,d
is drawn randomly from
U(f
min
, f
max
f
d
)
to
account for the signal bandwidth f
d
.
Results:
Figs. 15, 16, and 17 present the
RMSPE
obtained
when localizing
D = 2
broadband signals from Broadband
Scenario 1, 2, and 3 respectively. The number of snapshots
goes as high as the sampling frequency
f
s
= 2000 [Hz]
and is
given logarithmically. This high number is suitable for the
MB
broadband
MUSIC
algorithm to achieve reliable transformation
from the time domain to the frequency domain.
DA-MUSIC
is again only trained for the case
T = 200
, as indicated by a
circle around the marker, yet manages to perform similarly well
with a higher number of snapshots. As expected, the classic
narrowband
MUSIC
algorithm completely fails to operate
with these broadband signals, while
DA-MUSIC
consistently
achieves the most accurate estimates, outperforming the
MB
broadband
MUSIC
algorithm, except for Broadband Scenario
3 with a very large number of snapshots
T > 10
3
, where
DA-MUSIC
trained with much shorter sequences is slightly
outperformed by the
MB
estimator. These results demonstrate
the suitability of
DA-MUSIC
for coping with broadband
scenarios with a limited number of observations.
Figs. 18, 19, and 20 analyze the performances with differently
sized frequency ranges
[f
min
, f
max
]
. It is noted that the
architecture of
DA-MUSIC
is almost invariant towards such
scalings and manages to handle signals during inference with
much larger bandwidths or modulated with higher carrier
frequencies than the signals of the training data. Specifically,
DA-MUSIC
is only trained for signals with carrier frequencies
and bandwidths within 0 to 1000
[Hz]
as indicated by the
circle around the marker. The results in Figs. 18-20 show that
DA-MUSIC
, whose complexity is fixed, learns to achieve the
most accurate estimates, outperforming the
MB
broadband
MUSIC
; the latter requires an increase of the number of
IFB
s
to overcome a larger frequency range and has a constant
10 [Hz]
per bin in the depicted results leading to a severe increase in
computational complexity.
C. Non-Synthetic Data: Azimuth Estimation in Seismic Arrays
In this section, we demonstrate the feasibility and the perfor-
mance of
DA-MUSIC
in processing non-synthetic seismic data.
The seismic data was recorded by the German Experimental
Seismic System (
GERES
) array located in the Bavarian Forest,
Germany.
GERES
is part of the Comprehensive nuclear-Test
Ban Treaty Organization (
CTBTO
) international monitoring
system, and is a well-maintained and calibrated station. Data
from
GERES
is continuously streamed to the International Data
Centre (
IDC
) of the
CTBTO
, where it is analyzed. The array is
composed of 25 vertical seismometers with a minimal distance
between the sites of 124 [m] and an aperture of approximately
2.13 [km]. The seismic signal at each sensor is sampled at
Fig. 18: RMSPE of varying frequency range for the carrier
frequencies of Broadband Scenario 1 signals.
Fig. 19: RMSPE of varying frequency range for the bandwidth
of Broadband Scenario 2 signals.
Fig. 20: RMSPE of varying frequency range for the carrier
frequencies of Broadband Scenario 3 signals.
40 [Hz]
. Details about the
GERES
arrays and the exact array
configuration can be found in [61].
We use data from October to December 2021, in which
GERES
detected arrivals for 2904 events of which 2816 were
used during this analysis, 2534 for training, and 282 for testing.
We employ sliding windows of length 100 seconds with a shift
of 25 seconds around the signal arrival time as designated by
the
IDC
analysis. The following parameters for each event are
obtained from the
IDC
s Reviewed Event Bulletin: the azimuth
DoA
angle
θ
, the slowness value
u
, and the sensor positions
r
1
, ..., r
M
. Using these parameters, the steering vectors take
the following form:
a(f, u, ψ, α)=
e
j2πf ur
1
k(ψ)
, . . . , e
j2πf ur
M
k(ψ)
,
12
TABLE V:
RMSPE
in [rad] of different
DoA
estimation
algorithms for seismic data.
DA-MUSIC
Broadband
MUSIC
Classic
MUSIC
Beam-
former
Random
RMSPE
0.6269 1.0075 1.2475 0.9383 1.7097
for some frequency
f
and with the wave vector for certain
elevation α and azimuth ψ of interest,
k(ψ, α) = [sin α cos ψ, sin α sin ψ, cos α].
We compare
DA-MUSIC
with following
DoA
estimators for
the setting settings α = π/4 and f = 1 [Hz]:
Broadband
MUSIC
, corresponding to an incoherent broad-
band extension of
MUSIC
[46], and instead of using the
constant
f = 1 [Hz]
it utilizes 10
IFB
with frequencies in
[0, 20] [Hz].
The classic MUSIC algorithm, implemented in its narrow-
band format as described in Section
II-D
with additionally
filtering the measurements with an experimentally cali-
brated low-pass filter, allowing only frequencies within
[0, 10] [Hz] to pass.
A conventional beamformer [8].
Choosing a DoA angle at random.
The results, reported in Table V, show that
DA-MUSIC
manages to outperform the
MB
estimators by not only focusing
the frequency component, but also by concurrently focusing
the interdependent elevation angle to receive a stable azimuth
estimation. On the other hand, the
MB
algorithms require
further knowledge of the elevation and frequency at hand to
operate reliably. While the errors in Table V may appear to be
relatively large, it is noted that the average error achieved via
expert analysis reported by the
IDC
Reviewed Event Bulletin
is 0.4243 [rad]. This indicates the ability of
DA-MUSIC
to
achieve comparable results and to notably outperform the
MB
estimators while operating with simplified and approximated
model parameters.
V. CONCLUSIONS
We presented a hybrid
MB
/
DD
implementation of the
MUSIC
algorithm for
DoA
estimation. The proposed
DA-MUSIC
was shown to mitigate some of the limitations and
drawbacks of the classic method.
DA-MUSIC
is operable with
an unknown number of sources and with broadband signals
while being adaptable to various scenarios and robust towards
severe mismatches in the array geometry. The proposed hybrid
MB
/
DD
approach provides a viable alternative in both low
and high snapshot domains and shows a remarkable resolution
capability compared to both
MB
and
DD
benchmarks in various
settings.
VI. ACKNOWLEDGMENT
We thank Dr. Yochai Ben Horin for constructive and valuable
joint discussions on the seismic data, and for providing the
data.
REFERENCES
[1]
J. P. Merkofer, G. Revach, N. Shlezinger, and R. J. G. van Sloun, “Deep
augmented music algorithm for data-driven DoA estimation, in IEEE
Internatioal Conference on Acoustics, Speech, and Signal Processing
(ICASSP), 2022, pp. 3598–3602.
[2]
I. Bilik, O. Longman, S. Villeval, and J. Tabrikian, “The rise of radar
for autonomous vehicles: Signal processing solutions and future research
directions, IEEE Signal Process. Mag., vol. 36, no. 5, pp. 20–31, 2019.
[3]
D. Rahamim, J. Tabrikian, and R. Shavit, “Source localization using
vector sensor array in a multipath environment, IEEE Trans. Signal
Process., vol. 52, no. 11, pp. 3096–3103, 2004.
[4]
J. Foutz, A. Spanias, and M. K. Banavar, “Narrowband direction of
arrival estimation for antenna arrays, Synthesis Lectures on Antennas,
vol. 3, no. 1, pp. 1–76, 2008.
[5]
H. Krim and M. Viberg, “Two decades of array signal processing research:
the parametric approach, IEEE Signal Process. Mag., vol. 13, no. 4, pp.
67–94, 1996.
[6]
Z. Ahmad and I. Ali, “Three decades of development in DoA estimation
technology, Indonesian Journal of Electrical Engineering and Computer
Science, vol. 12, pp. 6297–6312, 2014.
[7]
R. Schmidt, “Multiple emitter location and signal parameter estimation,
IEEE Trans. Antennas Propag., vol. 34, no. 3, pp. 276–280, 1986.
[8]
M. S. Bartlett, “Smoothing periodograms from time-series with continu-
ous spectra, Nature, vol. 161, no. 4096, pp. 686–687, May 1948.
[9]
J. Capon, “High-resolution frequency-wavenumber spectrum analysis,
Proc. IEEE, vol. 57, no. 8, pp. 1408–1418, 1969.
[10]
T. E. Tuncer, T. K. Yasar, and B. Friedlander, “Chapter 4 - narrowband
and wideband DoA estimation for uniform and nonuniform linear arrays,
in Classical and Modern Direction-of-Arrival Estimation. Boston:
Academic Press, 2009, pp. 125–160.
[11]
H. Wang and M. Kaveh, “Coherent signal-subspace processing for the
detection and estimation of angles of arrival of multiple wide-band
sources, IEEE Trans. Acoust., Speech, Signal Process., vol. 33, no. 4,
pp. 823–831, 1985.
[12]
G. Su and M. Morf, “The signal subspace approach for multiple wide-
band emitter location, IEEE Trans. Acoust., Speech, Signal Process.,
vol. 31, no. 6, pp. 1502–1522, 1983.
[13]
P.-A. Grumiaux, S. Kiti
´
c, L. Girin, and A. Gu
´
erin, A survey of sound
source localization with deep learning methods, The Journal of the
Acoustical Society of America, vol. 152, no. 1, pp. 107–151, 2022.
[14]
S. Chakrabarty and E. A. Habets, “Multi-speaker DoA estimation using
deep convolutional networks trained with noise signals, IEEE J. Sel.
Topics Signal Process., vol. 13, no. 1, pp. 8–21, 2019.
[15]
Z.-M. Liu, C. Zhang, and S. Y. Philip, “Direction-of-arrival estimation
based on deep neural networks with robustness to array imperfections,
IEEE Trans. Antennas Propag., vol. 66, no. 12, pp. 7315–7327, 2018.
[16]
L. li Wu, Z. Liu, and Z. Huang, “Deep convolution network for direction
of arrival estimation with sparse prior, IEEE Signal Process. Lett.,
vol. 26, pp. 1688–1692, 2019.
[17]
H. Hammer, S. E. Chazan, J. Goldberger, and S. Gannot, “Dynamically
localizing multiple speakers based on the time-frequency domain,
EURASIP Journal on Audio, Speech, and Music Processing, vol. 2021,
no. 1, pp. 1–10, 2021.
[18]
G. K. Papageorgiou, M. Sellathurai, and Y. C. Eldar, “Deep networks for
direction-of-arrival estimation in low SNR, IEEE Trans. Signal Process.,
vol. 69, pp. 3714–3729, 2020.
[19]
A. Barthelme and W. Utschick, A machine learning approach to DoA
estimation and model order selection for antenna arrays with subarray
sampling, IEEE Trans. Signal Process., vol. 69, pp. 3075–3087, 2020.
[20]
D. Hu, Y. Zhang, L. He, and J. Wu, “Low-complexity deep-learning-
based DOA estimation for hybrid massive MIMO systems with uniform
circular arrays, IEEE Wireless Commun. Lett., vol. 9, pp. 83–86, 2020.
[21]
M. L. L. de Oliveira and M. Bekooij, “Deep-MLE: Fusion between a
neural network and MLE for a single snapshot DOA estimation, IEEE
International Conference on Acoustics, Speech and Signal Processing
(ICASSP), pp. 3673–3677, 2022.
[22]
——, “Resnet applied for a single-snapshot DOA estimation, IEEE
Radar Conference (RadarConf22), 2022.
[23]
N. Shlezinger, J. Whang, Y. C. Eldar, and A. G. Dimakis, “Model-based
deep learning, arXiv preprint arXiv:2012.08405, 2020.
[24]
N. Shlezinger, Y. C. Eldar, and S. P. Boyd, “Model-based deep learning:
On the intersection of deep learning and optimization, IEEE Access,
2022.
[25]
A. M. Elbir, “DeepMUSIC: Multiple signal classification via deep
learning, IEEE Sensors Letters, vol. 4, no. 4, pp. 1–4, 2020.
13
[26]
C. Liu, W. Feng, H. Li, and H. Zhu, “Single snapshot DOA estimation
based on spatial smoothing MUSIC and CNN, IEEE International
Conference on Signal Processing, Communications and Computing
(ICSPCC), 2021.
[27]
A. Barthelme and W. Utschick, “DoA estimation using neural network-
based covariance matrix reconstruction, IEEE Signal Process. Lett.,
vol. 28, pp. 783–787, 2021.
[28]
D. T. Hoang and K. Lee, “Deep learning-aided coherent direction-of-
arrival estimation with the ftmr algorithm, IEEE Trans. Signal Process.,
vol. 70, pp. 1118–1130, 2022.
[29]
W. Zhang, Y. Han, M. Jin, and X. Qiao, “Multiple-toeplitz matrices
reconstruction algorithm for doa estimation of coherent signals, IEEE
Access, vol. 7, pp. 49 504–49 512, 2019.
[30]
S. Chakrabarty and E. A. Habets, “Broadband DoA estimation using
convolutional neural networks trained with noise signals, in Workshop
on Applications of Signal Processing to Audio and Acoustics (WASPAA),
2017, pp. 136–140.
[31]
W. Zhu and M. Zhang, A deep learning architecture for broadband DoA
estimation, in International Conference on Communication Technology
(ICCT), 2019, pp. 244–247.
[32]
L. li Wu and Z. Huang, “Coherent SVR learning for wideband direction-
of-arrival estimation, IEEE Signal Process. Lett., vol. 26, pp. 642–646,
2019.
[33]
B. Friedlander, “Chapter 1 - wireless direction-finding fundamentals, in
Classical and Modern Direction-of-Arrival Estimation, T. E. Tuncer and
B. Friedlander, Eds. Boston: Academic Press, 2009, pp. 1–51.
[34]
Z. Chen, G. Gokeda, and Y. Yu, Introduction to Direction-of-arrival
Estimation. Artech House, 2010.
[35]
S. Ge, K. Li, and S. N. M. Rum, “Deep learning approach in DoA
estimation: A systematic literature review, Mob. Inf. Syst., vol. 2021,
pp. 6 392 875:1–6 392 875:14, 2021.
[36]
O. Frost, “An algorithm for linearly constrained adaptive array processing,
Proc. IEEE, vol. 60, no. 8, pp. 926–935, 1972.
[37]
B. Van Veen and K. Buckley, “Beamforming: a versatile approach to
spatial filtering, IEEE ASSP Mag., vol. 5, no. 2, pp. 4–24, 1988.
[38]
B. Friedlander, “The root-music algorithm for direction finding with
interpolated arrays, Signal Processing, vol. 30, no. 1, pp. 15–29, 1993.
[39]
Tie-Jun Shan, M. Wax, and T. Kailath, “On spatial smoothing for
direction-of-arrival estimation of coherent signals, IEEE Trans. Acoust.,
Speech, Signal Process., vol. 33, no. 4, pp. 806–811, 1985.
[40]
Qing Chen and Ruolun Liu, “On the explanation of spatial smoothing
in music algorithm for coherent sources, in International Conference
on Information Science and Technology, 2011, pp. 699–702.
[41]
A. Paulraj, R. Roy, and T. Kailath, “Estimation of signal parameters
via rotational invariance techniques- esprit, in Nineteeth Asilomar
Conference on Circuits, Systems and Computers, 1985., 1985, pp. 83–89.
[42]
V. Vasylyshyn, “Direction of arrival estimation using esprit with sparse
arrays, in European Radar Conference (EuRAD), 2009, pp. 246–249.
[43]
M. Wax, T.-J. Shan, and T. Kailath, “Spatio-temporal spectral analysis by
eigenstructure methods, IEEE Trans. Acoust., Speech, Signal Process.,
vol. 32, no. 4, pp. 817–827, 1984.
[44]
S. Chandran and M. Ibrahim, “DoA estimation of wide-band signals
based on time-frequency analysis, IEEE J. Ocean. Eng., vol. 24, no. 1,
pp. 116–121, 1999.
[45]
S. Argentieri and P. Danes, “Broadband variations of the music high-
resolution method for sound source localization in robotics, in 2007
IEEE/RSJ International Conference on Intelligent Robots and Systems ,
2007, pp. 2009–2014.
[46]
Y.-S. Yoon, L. M. Kaplan, and J. H. McClellan, “DoA estimation of
wideband signals, Advances in Direction-of-Arrival Estimation, 2006.
[47]
J. Krolik and D. Swingler, “Focused wide-band array processing by spatial
resampling, IEEE Trans. Acoust., Speech, Signal Process., vol. 38, no. 2,
pp. 356–360, 1990.
[48]
T.-S. Lee, “Efficient wideband source localization using beamforming
invariance technique, IEEE Trans. Signal Process., vol. 42, no. 6, pp.
1376–1387, 1994.
[49]
B. Friedlander and A. Weiss, “Direction finding for wideband signals
using an interpolated array, in Asilomar Conference on Signals, Systems,
and Computers, vol. 1, 1991, pp. 583–587.
[50]
E. di Claudio and R. Parisi, “WAVES: weighted average of signal
subspaces for robust wideband direction finding, IEEE Trans. Signal
Process., vol. 49, no. 10, pp. 2179–2191, 2001.
[51]
Y.-S. Yoon, L. Kaplan, and J. McClellan, “TOPS: new DOA estimator
for wideband signals, IEEE Trans. Signal Process., vol. 54, no. 6, pp.
1977–1989, 2006.
[52]
F. Ma and X. Zhang, “Wideband doa estimation based on focusing signal
subspace, Signal, Image and Video Processing, vol. 13, 06 2019.
[53]
S. Ge, K. Li, and S. N. M. Rum, “Deep learning approach in DOA
estimation: A systematic literature review, Mob. Inf. Syst., vol. 2021,
pp. 6 392 875:1–6 392 875:14, 2021.
[54]
S. M. Kay, “Fundamentals of statistical signal processing: estimation
theory, Technometrics, vol. 37, p. 465, 1993.
[55]
H. Krim and J. Proakis, “Smoothed eigenspace-based parameter esti-
mation, Automatica, vol. 30, no. 1, pp. 27–38, 1994, special issue on
statistical signal processing and control.
[56]
O. Solomon, R. Cohen, Y. Zhang, Y. Yang, Q. He, J. Luo, R. J. van
Sloun, and Y. C. Eldar, “Deep unfolded robust PCA with application
to clutter suppression in ultrasound, IEEE Trans. Med. Imag., vol. 39,
no. 4, pp. 1051–1063, 2019.
[57]
T. Routtenberg and J. Tabrikian, “Bayesian parameter estimation using
periodic cost functions, IEEE Trans. Signal Process., vol. 60, no. 3, pp.
1229–1240, 2011.
[58]
——, “Non-bayesian periodic Cram
´
er-Rao bound, IEEE Trans. Signal
Process., vol. 61, no. 4, pp. 1019–1032, 2013.
[59]
D. P. Kingma and J. Ba, Adam: A method for stochastic optimization,
arXiv preprint arXiv:1412.6980, 2014.
[60]
S. B. Weinstein, “The history of orthogonal frequency-division multi-
plexing [history of communications], IEEE Commun. Mag., vol. 47,
no. 11, pp. 26–35, 2009.
[61]
H.-P. Harjes, “Design and siting of a new regional array in central europe,
Bulletin of the Seismological Society of America, vol. 80, pp. 1801–1817,
1990.
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