Multi-Speaker Localization by Central and Lateral

Microphone Arrays Based on the Combination of

2D-SRP and Subband GEVD Algorithms
Ali Dehghan Firoozabadi1,*, Pablo Irarrázaval2,3,4, Pablo Adasme5, David Zabala-Blanco6, Pablo Palacios-Játiva7,8, Hugo Durney1, Miguel
Sanhueza Olave1, Cesar Azurdia7
2022 8th International Conference on Signal Processing and Communication (ICSC) | 978-1-6654-5430-8/22/$31.00 ©2022 IEEE | DOI: 10.1109/ICSC56524.2022.10009454

1Department of Electricity, Universidad Tecnológica Metropolitana, Av. José Pedro Alessandri 1242, Santiago 7800002, Chile
2Electrical Engineering Department, Pontificia Universidad Católica de Chile, Santiago 7820436, Chile
3Biomedical Imaging Center, Pontificia Universidad Católica de Chile, Santiago 7820436, Chile
4Institute for Biological and Medical Engineering, Pontificia Universidad Católica de Chile, Santiago 7820436, Chile

5Electrical Engineering Department, Universidad de Santiago de Chile, Av. Victor Jara 3519, Santiago 9170124, Chile
6Department of Computer Science and Industry, Universidad Católica del Maule, Talca 3466706, Chile
7Department of Electrical Engineering, Universidad de Chile, Santiago 8370451, Chile
8Escuela de Informática y Telecomunicaciones, Universidad Diego Portales, Santiago 8370190, Chile

Tel.: +56 (2) 2787 7117; E-mail: adehghanfirouzabadi@utem.cl, ORCID: 0000-0002-6391-6863

Abstract—In this paper, a novel speech source localization
(SSL) algorithm by the use of the central and lateral microphone
arrays is proposed with low complexity and high precision in
undesirable acoustical environments. Firstly, a proposed star-
shaped microphone array (SSMA) at the room’s center is
considered in combination with the improved 2D steered
response power (2D-SRP) algorithm for determining the
speakers’ directions, where the PENS algorithm is considered
for speakers counting. Then, the two closest cross-shaped arrays
(CSMA) to each speaker on the wall are selected, where these
arrays prepare the proper information with less microphones.
Each CSMA is considered in combination with subband
generalized eigenvalue decomposition (SB-GEVD) method on
various time frames for estimating the two others speaker’s
direction. Therefore, the 3D speakers’ positions are calculated
by the crossing of the estimated directions from SSMA and two
CSMAs. The proposed lateral-central microphone array-
multiple sound source localization method (LCMA-MSSL) is
compared with other research, which shows the superiority of
the proposed method in undesirable noisy and reverberant
scenarios.
Keywords— Speaker localization, microphone array, adaptive
processing, eigenvalue decomposition, time delay estimation.
I. INTRODUCTION
The multiple simultaneous speech source localization
(SSL) in undesirable environments is a challenge in the
implementation of signal processing systems. The SSL
algorithms are normally used as a pre-processing method for
other speech processing applications such as speech
enhancement [1], speaker tracking [2], and speech recognition
[3]. The microphone array (MA) is considered as a proper
instrument for increasing the precision of the SSL algorithms
in such applications as hearing aids [4], videoconferencing
[5], smart robots [6], and acoustical speech processing rooms.
The learning-based algorithms [7] localize the speakers more
accurately, but it is not possible to implement in the real
conditions based on the high computational complexity. In the
recent decade, various algorithms have been introduced for the
SSL, which are categorized to time-frequency (TF) signal
processing [8], and spatial spectrum analysis [9]. The other
category of the SSL algorithms calculates the steered response
power (SRP) for the candidate areas in the recording space by
considering the spatial diversity of acoustical environments
for evaluating the probability of existence the sound sources
in different positions.
The cross-correlation based methods are one of the
important categories in the two-step SSL algorithms [10]. In
contrary, the SRP-based methods are used for the condition,
where a large number of microphones is permitted [11].
Nikolas et al. presented a perpendicular cross-spectra fusion
(PCSF) method in 2017 as a robust algorithm for speaker
localization [12] that selects the mathematical formulas for
estimating the speakers’ positions. The subsystems in speaker
localization are parallelly implemented for preparing the
candidate DOA points in each TF bins. They proposed the
divergence feature of various DOA estimators for verifying
the reliability of estimated locations. The calculated DOAs are
considered as an input data for the presented classification
process, which is based on a match criterion. Ning et al. in
2021 proposed the morphological component analysis for the
sound field (SF-MCA) by alternating direction method of
multipliers (ADMM) for multiple SSL [13]. In the first step,
the Helmholtz formula is selected for analysis the speech
components. In the next step, the plane-wave and green
functions are considered for representing the sparse model of
the speech signal, where the decomposition parameters are
extracted by using of ADMM method. Inkyu et al. in 2022
presented a localization method based on Monte Carlo
criterion by using of diffraction and reflection features (DR-
MCL) [14]. The proposed method estimates the propagation
path based on the diffraction and reflection effects of sound
signals by using of background ray tracking system. For
estimating the diffraction of propagation path, the ray tracking
method is combined by uniform theory of diffraction model
based on the features around the scatter of obstacles. The 3D
scatter of obstacles is reconstructed in the computation phase
domain, where the obtained information is considered for
producing the reflection, primary, and diffraction sound rays
in the real-time phase domain.
In this research, a novel 3D muti-source localization
system is proposed based on the central and lateral MAs by
using of 2D-SRP and subband generalized eigenvalue
decomposition (SB-GEVD) methods on undesirable
environments. In the first step, a proposed central star-shaped
microphone array (SSMA) is considered for estimating the
direction of speakers. The obtained information by this SSMA

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 is used as an input for 2D-SRP algorithm. Implementing the
SRP function in 2D format reduces the computational
complexity. The peaks’ positions in 2D-SRP method, which
are the speakers’ directions, are extracted by PENS algorithm
[15]. Two closest lateral cross-shaped arrays (CSMA) to each
speaker are selected based on the speakers’ directions. In the
following, the SB-GEVD algorithm is implemented
adaptively on CSMA on various time frames. Finally, the
location of speakers are calculated by the intersection between
DOA of central MA and the DOAs of lateral MAs. This
algorithm is repeated for all speakers to calculate each 3D
speakers’ locations.
Section 2 shows the microphone signal model. The central
SSMA and lateral CSMAs are presented in section 3. Section
4 represents the proposed multi-speaker SSL method by using
of SB-GEVD method in combination with 2D-SRP
algorithms. Section 5 shows the results and discussion of the
simulations for multiple speakers. Finally, section 6 includes
some conclusions of the proposed SSL algorithms.
II. MICROPHONE SIGNAL MODEL IN ACOUSTICAL
ENVIRONMENT
The acoustic sensors specially MAs are the principal base
of SSL algorithms. Since the SSL methods are implemented
firstly on simulated data and then on real signals, the selected
model for simulations should be as similar as possible to the
real environments. The real model is considered for
microphone signal simulations, which is expressed as:
L
( computational complexity.
xm ( t ) =  sl ( t ) * γ ( d m ,l , t ) + vm ( t ) ( m = 1,..., M ) (1)
l =1
where in Eq. (1), xm ( t ) is the signal in m-th microphone, sl ( t )
is the signal in l-th speech source, M is the number of
microphones, L is the number of speech sources, γ ( d m,l , t ) is
the room impulse response (RIR) between the l-th speech
(
source and m-th microphone and, vm ( t ) is the white noise, and
* is the convolution operator. By considering the ideal model
with the noise and reverberation effects, the simulated signals
are highly similar to the captured speech in real scenarios.
III. THE PROPOSED STAR AND CROSS-SHAPED MICROPHONE
ARRAYS
The acoustical environments decrease the accuracy of the
localization algorithms because of the noise and
reverberation. Therefore, the MA is proposed as a proper
instrument for increasing the accuracy of localization
algorithms by the use of spatial layout and preparing the extra
information. Various structures were proposed for MAs [16]
based on the speech processing algorithms. In this paper, a
combination of central and lateral MAs with the SSL
algorithms is proposed for increasing the estimation accuracy
and covering a wide range of acoustical environments. The
proposed MA layout provides the proper information for the
algorithms by considering the sufficient microphone pairs. In
the first step, a SSMA is proposed for estimating the speakers’
directions. Figure 1(a) show the structure of the proposed
central SSMA with 10 microphones and the selected
microphone pairs for the SSL algorithm. This structure is
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considered for DOA estimation by using of 2D-SRP
algorithm. The microphone pairs (1,6), (1,7), (2,7), (2,8),
(3,8), (3,9), (4,9), (4,10), (5,10), and (5,6) are used for
calculating the cross-correlation in the 2D-SRP method.
Considering these microphone pairs not only decreases the
complexity, but also estimates the speakers’ directions (
αˆC ,l for l = 1,..., L ) with high accuracy. As mentioned, two
categories of lateral MAs in distributed format are considered
for accurate SSL. Figure 1 (b and c) shows the lateral CSMAs
for the second step of the localization algorithm. Each CSMA
contains of 5 microphones, which are installed in a proper
distance on the walls. These lateral MAs in combination with
central MA complete the localization process. Figure 1(b)
represents the microphone pairs in the first CSMA (LA1) for
horizontal DOA estimation ( αˆ LA1,l for l = 1,..., L ). Three
microphone pairs (m1, m2), (m2, m3), and (m3, m1) are
selected for considering in adaptive SB-GEVD method for
DOA estimation. As mentioned, two closest CSMAs are
considered for the speaker. In the next step, the second closest
CSMA to the speaker is considered for estimating the vertical
direction estimation ( αˆ LA 2,l for l = 1,..., L ). Figure 1(c) shows
the microphone pairs (m1, m2), (m2, m3), and (m3, m1) in the
second CSMA (LA2) by using of adaptive SB-GEVD
algorithm for vertical DOA estimation. The central and lateral
MAs (SSMA and CSMAs) prepare the best acoustical
information with minimal noise and reverberation effects,
which increases the reliability of estimated locations. In
addition, using of lateral and central MAs decrease the
Fig. 1. a) The central star-shaped array, b) the lateral cross-shaped array for
horizontal direction estimation, and c) the lateral CSMA for vertical DOA
estimation with selected microphone pairs.
IV. THE PROPOSED SSL METHOD BY 2D-SRP AND SUBBAND
GEVD ALGORITHMS
The multi-speaker localization in real conditions is a
challenge in speech processing specially for noisy and
reverberant environments. The one-step algorithms by the
energy calculation contain high computational complexity and
proper accuracy, but the time-delay estimation (TDE)-based
algorithms estimate the speakers’ locations faster with less
precision. Figure 2 shows the proposed 3D SSL system for
simultaneous speakers in acoustical environments. In this
system, the SSMA and CSMAs are proposed in combination
with 2D-SRP and adaptive SB-GEVD algorithms for multiple
SSL. The high accuracy in SRP algorithm [9] with low
complexity of GEVD [17] method is selected as a strategic
criterion in the proposed system. In addition, implementation
of the SRP function in two dimensions highly decreases the
complexity of the method. Also, the accuracy of the GEVD
method is increased by adaptive and subband implementation.
 Fig. 2. The block diagram of the presented localization system based on the central and latera MAs in combination with 2D-SRP and adaptive subband GEVD.

+∞
A. The 2D-SRP method in combination with central star-  2D 2D
  2D 2D
 ′  2D 2D

shaped microphone array Q  θ1 ,...,θ M  =  P  ω , θ1 ,..., θ M  P  ω , θ1 ,..., θ M  dω (5)
  −∞    
In this paper, a novel system is proposed for simultaneous
SSL. As shown in Figure 2, the first step is estimating the  2D 2D

where P  ω , θ1 ,..., θ M  is the FSB output in the limited two
directions of the speakers by the SSMA in combination with  
a 2D version of the SRP method for decreasing the complexity dimensions. By replacing Eq. (4) to Eq. (5) and changing the
with high accuracy. Based on the microphones signal model parameters in the sigma and integral, the 2D-SRP is
in Eq. (1), the delay-and-sum beamformer (DSB) is informed implemented as:
as:
 2D 
2D M M
Q  θ1 ,...,θ M  = ψ ab (ω ) X a (ω ) X b′ ( ω ) e ( a b )
jω θ −θ
 2D 2D

M (6)
P  t , θ1 ,..., θ M  =  xm ( t − θ m ) (2)   a =1 b =1
  m =1
2D 2D
where the weighting function is expressed as
where in Eq. (2), θ1 ,...,θ M are M steered delays for orienting ψ ab (ω ) = Ga (ω ) Gb′ (ω ) . In this paper, the phase transform
the array to the 2D candidate points in the acoustical 1
environments. Therefore, the DSB output is expressed as a (PHAT) function ( ψ abPHAT (ω ) = ) is selected
function of the microphone signals and steered delays as: X a ( ω ) X b′ (ω )
because of its performance in the reverberation environments,
 2D 2D

P  t , θ1 ,..., θ M = which is considered in combination with 2D-SRP algorithm
  (3)
M M
as:
1 1 (
s ( t − τ 0 ) *  γ ( d m ,l , t − τ 0 + τ m ) +  vm ( d m ,l , t − τ 0 + τ m )
r
m =1 m r
m =1 m  2D 2D

Q  θ1 ,..., θ M=
where in Eq. (3), τ 0 is phase center, τ m is the delay between m-  
M M (7)
th source and center of the array, and rm is the distance between 1
m-th source and center of the array. Since the reverberation 
a =1 b =1 X a ( ω ) X b′ ( ω )
X a ( ω ) X ′
b ( ω ) e
jω (θ a −θb )

contains the multi-path versions of speech signals, noise

( In the ideal conditions, the peak positions in this function
vm ( t ) is highly correlated with the source signal, which are the source locations. But the results contain big errors in
decreases the performance of DSB. The alternative method is the real scenarios with the high level undesirable factors.
considering the adaptive filters in this beamformer, which is Therefore, the number of local peak positions in this function
named filter-and-sum beamformer (FSB) for decreasing the is calculated based on the PENS algorithm (L first peaks) for
noise and reverberation effects, which is expressed in the decreasing the complexity as:
frequency domain as:
αˆ C ,1 = arg max Q  θ1 ,..., θ M 
2D 2D

 2D 
2D M
P  ω , θ1 ,...,θ M  =  Gm ( ω )X m (ω ) e − jωθm (4) θ1 ,...,θ M ∈R ( x,y )  
  m =1
. . . (8)
where in Eq. (4), X 1 (ω ) ,..., X M (ω ) are the Fourier transform
 2D 2D

of the signals in SSMA, and G1 (ω ) ,..., GM (ω ) are the Fourier αˆ C , L = arg max Q  θ1 ,...,θ M 
θ1 ,...,θ M ∈R ( x,y )  
αˆ C ,L ≠αˆ C ,1 ,...,αˆ C ,L −1
transform of the adaptive filters. The selection of these filters
depends on the source and the noise signals in the acoustical where in Eq. (8), αˆC ,l is the speaker’s direction for l-th sound
environments. The 2D-SRP is calculated based on the output source (l=1,…,L) based on the central SSMA with M
power of FSB as: microphones. In addition, the z distance is selected as a fixed
value for decreasing the computational complexity in the SRP

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 algorithm to be able for real implementations. Since the 2D-
SRP function is just considered for DOA estimations, the
limited 2D assumption does not affect the accuracy of the final
results. In the following, two closed latera CSMAs to each
speaker are selected by combination with adaptive SB-GEVD
algorithm for vertical and horizontal direction estimations.
B. The adaptive SB-GEVD algorithm in combination with
lateral MAs
In the real scenarios, the acoustical environments are
considered as a linear system. Therefore, the relation between
the microphone signal and RIR is explained as:
xi ( n ) q j = x j ( n ) qi
where xi ( n ) is the recorded signal by microphone including
the noise and reverberation effects. The RIR between the
source and each microphone is show as:
q j =  q j ,0 , q j ,1 ,...., q j ,B  (10)
where B is the length of RIR vector. Since three microphone
pairs in the CSMA is considered for the evaluations, variables
i and j are selected as 3. Therefore, the covariance matrix is
calculated as:
 hx1 x1 hx1 x2 hx1x3 
 
H =  hx2 x1 hx2 x2 hx2 x3  (11)
 
 hx3 x1 hx3 x2 hx3 x3 
where in Eq. (11), the covariance values in this matrix are
hxi x j = E { xi ( n ) xTj ( n )} for i,j=1,2,3. For solving this
covariance matrix, the impulse response vector
F =  q3 ; − q2 ; − q1  is designed as a series of impulse
responses obtained by three microphone pairs with length 3B.
Vector F includes the eigenvector for the eigenvalue 0. The
adaptive SB-GEVD algorithm considers the iterative random
gradient methods for calculating the eigenvector related to the
smallest eigenvalue of signal covariance matrix H Bx and noise
covariance matrix H Bv on continues time frames. The
generalized eigenvalue of covariance matrix H is obtained by
minimizing the cost function of F T H Bx F , which is considered
as minimizing the mean square error (MSE) as:
F T ( n ) xB ( n )
e (n) =
F T ( n ) H Bv F ( n )
which is implemented by least mean square (LMS) method as:
∂F ( n )
F ( n + 1) = F ( n ) − µ e ( n )
∂e ( n )
where µ is the adaptation step. To avoid the delay
propagation, an extra normalization step is implemented in
LMS algorithm, where finally vector F is expressed as:
F% ( n + 1)
F ( n + 1) =
F% T ( n + 1) H Bv F% ( n + 1)
where,
F% ( n + 1) = F ( n ) − µ e ( n ) { xB ( n ) − e ( n ) H Bv F ( n )}
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The impulse response between the sound source and 3
microphone pairs ( q1 , q2 , q3 ) is calculated by estimating
vector F . Finally, the DOA for each speaker (
αˆ LA1,l for l = 1,..., L ) is obtained by averaging of three DOAs
from each microphone pair. This process is repeated for the
other closest CSMA to obtain the vertical DOAs (
αˆ LA 2,l for l = 1,..., L ). In this paper, the SB-GEVD algorithm
is adaptively implemented on frequency bands b1 = 0 − 4KHz
and b2 = 4 − 8KHz on continues time frames of the recorded
signals. The aim of subband processing is to focus on low
(9) frequency components od recorded signal (more speech
information) and decrease the effect of high frequency
components (high noisy and less speech signals) (
αˆ LAi ,l ,b1 , αˆ LAi ,l ,b2 for i = 1, 2 ). This process is parallelly repeated
on two closest CSMAs to each speaker on continues time
frames. Therefore, the weights for each subband are calculated
as:
1 N (
max Hist αˆ LAi ,l ,b1 ) i = 1, 2
wi =  2
for  (16)
N l = 1,..., L
n =1
 max Hist (αˆ
j =1
LAi , l , b j )
The final histogram is calculated by affecting the weights
in each histogram for two different subbands, where the peak
positions are the DOA for each speaker.
1 2 i = 1, 2
αˆ LA ,l = arg max  (
wi Hist αˆ LAi ,l ,b j for  ) (17)
θ1 ,...,θ M ∈R ( x,y ) 2 j =1 l = 1,..., L
i
This calculation is performed on both closest CSMAs to
each speaker for estimating the horizontal ( αˆ LA1,l ) and vertical
( αˆ LA 2,l ) directions. Finally, three directions are calculated for
each speaker ( αˆC ,l , αˆ LA1,l , αˆ LA 2,l ), which are the DOAs in 3D
space. The intersection between these DOAs provides a point
(in the ideal case) and an area (in the real conditions), where
the 3D source location is selected as closest point in the area
to the three DOAs.
( xˆl , yˆ l , zˆl ) =
2 2 (18)
L2 2
min (x− x ) +(y− y ) +(z − z )
l
C
l
L1
l for l = 1,..., L
The various combinations of x, y, z are evaluated in this
(12) area for estimating the final 3D location ( xˆl , yˆl , zˆl ) for each
speaker. This estimation is repeated to calculate all 3D
speakers’ locations in the acoustical environment.
V. SIMULATIONS AND RESULTS
(13) The experiments are done on simulated data of TIMIT
dataset [18]. Based on the experiments in the real scenarios,
90% and 8% of the overlapping are between 2 and 3
simultaneous speakers, respectively [19]. Therefore, the
experiments are implemented for 2 and 3 overlapped speakers.
One female (X1) and one male (X2) speech signals are
selected for 2 simultaneous speakers. The third speaker (X3)
(14)
is a male in the 3 simultaneous speakers’ scenarios. Figure 3
shows the simulated room for the evaluations of the proposed
LCMA-MSSL algorithm in acoustical scenarios. As shown,
(15) the SSMA with 10 microphones is located at (230,175,95) cm.
In addition, two CSMAs with 5 microphones are installed on
each wall with 170 cm height of the ground. The room
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 dimension is (460,350, 270) cm, which is similar to the real
conditions. Also, the three speakers are located at
X1=(125,262,165) cm, X2=(141,74,175) cm, and
X3=(383,158,179) cm, respectively.
Fig. 3. The simulated room for speech signal recording in the multiple SSL
application by SSMA and CSMAs for 2 and 3 simultaneous speakers.
The Hamming window with 40 ms and 50% inter-window
overlap is selected in the simulations to use the maximum
recorded data and updating the noise covariance matrix. The
Gaussian noise is selected to be similar to the real scenarios.
In addition, the Image method [20] is considered for preparing
the reverberation in the acoustical room.
The proposed LCMA-MSSL algorithm is compared with
the PCSF [12], SF-MCA [13], and DR-MCL [14] methods by
the mean absolute estimation error (MAEE) for 2 and 3
simultaneous speakers on different noisy and reverberant
environments. In the first step, the signal-to-noise ratio (SNR)
is fixed and the reverberation time ( RT60 ) is variable, but in
the second step, the RT60 is fixed and SNR is variable. Figure
4 shows the results of the proposed LCMA-MSSL in
comparison with the PCSF, SF-MCA, and DR-MCL for 2
simultaneous speakers based on the averaged MAEE criteria.
Figure 4(a) represents the results of this comparison for fixed
SNR=5 dB and variable 0 ≤ RT60 ≤ 600 ms , where the aim is
evaluating the reverberation effects on the accuracy of
estimated locations. As shown, the PCSF and LCMA-MSSL
methods contain the lowest and highest localization accuracy,
respectively. For example, the MAEE results for the LCMA-
MSSL is 33 cm in comparison with PCSF by 51 cm, SF-MCA
by 47 cm, and DR-MCL by 40 cm for RT60 = 500 ms , which
shows the superiority of the presented algorithm. The
precision of all methods is decreased by increasing the
reverberation effects. Figure 4(b) shows the MAEE results of
the LCMA-MSSL in comparison with the PCSF, SF-MCA,
and DR-MCL algorithms for fixed RT60 = 500 ms and
−10 ≤ SNR ≤ 20 dB to evaluate the noise effects on the
localization accuracy. In SNR=10 dB the averaged MAEE
results for the LCMA-MSSL is 31 cm in comparison with
PCSF by 48 cm, SF-MCA by 43 cm, and DR-MCL by 38 cm.
As shown, the DR-MCL method contains the closest results
to the LCMA-MSSL algorithm, which shows that the
presented algorithm localizes the speakers more accurately. In
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addition, increasing the noise level decreases the accuracy of
all localization methods.
Fig. 4. The averaged MAEE for the LCMA-MSSL in comparison with the
PCSF, SF-MCA, and DR-MCL algorithms for 2 simultaneous speakers, a)
for fixed SNR=5 dB and variable 0 ≤ RT60 ≤ 600 ms , b) fixed
RT60 = 500 ms and variable −10 ≤ SNR ≤ 20 dB .
Figure 5 represents the results of the LCMA-MSSL
comparing with the PCSF, SF-MCA, and DR-MCL by the
MAEE for three overlapped speakers on noisy and reverberant
environments. As seen, the presented algorithm localizes the
speakers with more precision comparing with other works.
Figure 5(a) represents the averaged MAEE results for fixed
SNR=5 dB and variable 0 ≤ RT60 ≤ 600 ms . The proposed
LCMA-MSSL localizes the 3 simultaneous speakers with
more precision comparing with traditional works. For
example, in RT60 = 500 ms the proposed LCMA-MSSL
localize the speakers with 38 cm based on averaged MAEE
criteria in comparison with the PCSF by 58 cm, SF-MCA by
51 cm, and DR-MCL by 48 cm. Figure 5(b) represents the
MAEE for the LCMA-MSSL comparing with the previous
works for 3 simultaneous speakers on fixed RT60 = 500 ms
and variable −10 ≤ SNR ≤ 20 dB for verifying the noise
effects on the accuracy of the localization. As seen, the
proposed method has better results comparing with the
previous methods. In SNR=10 dB, the averaged MAEE
results for the proposed LCMA-MSSL method is 36 cm in
comparison with the PCSF by 54 cm, SF-MCA by 47 cm, and
DR-MCL by 42 cm, where the accuracy of all methods
decreases by the noise effects. Figures 4 and 5 show the
reliability of the presented method comparing with the other
algorithms.
VI. CONCLUSIONS
In this paper, a two-step 3D SSL system was proposed for
simultaneous speakers by the central and lateral MAs. The
central SSMA and lateral CSMAs were proposed for
increasing the localization accuracy. Firstly, the speakers’
directions are calculated by central SSMA in combination
with 2D-SRP method. Implementing the SRP method in 2D
 prepares the high accuracy for estimating the direction of
speakers. In the following, two closest CSMAs are considered
for each speaker. These CSMAs are for horizontal (L1) and
vertical (L2) directions’ estimations combining with adaptive
SB-GEVD method. Finally, the intersection between these
DOAs prepares an 3D area in the acoustical space, where the
closest area in this sector to all three DOAs is selected as the
3D speaker’s location. The proposed LCMA-MSSL methos
was compared with the PCSF, SF-MCA, and DR-MCL
algorithms based on the averaged MAEE criteria for 2 and 3
overlapped speakers. The results show the reliability of the
presented algorithm in comparison with the other SSL
algorithms.
Fig. 5. The averaged MAEE for the LCMA-MSSL in comparison with the
PCSF, SF-MCA, and DR-MCL algorithms for 3 simultaneous speakers, a)
for fixed SNR=5 dB and variable 0 ≤ RT60 ≤ 600 ms , b) fixed
RT60 = 500 ms and variable −10 ≤ SNR ≤ 20 dB .
ACKNOWLEDGMENT
The authors acknowledge financial support from:
ANID/FONDECYT Postdoctorado No. 3190147,
Competition for Research Regular Projects, year 2021, code
LPR21-02, Universidad Tecnológica Metropolitana, projects
DICYT 062117S, ANID FCHA/Beca de Doctorado
Nacional/2019 21190489, CODELCO: Concurso Piensa
Minería 2021, SENESCYT “Convocatoria abierta 2014-
primera fase, Acta CIBAE-023-2014”.
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