Functional generalized inverse beamforming with regularization matrix applied to sound source localization

2016 ◽  
Vol 23 (18) ◽  
pp. 2977-2988 ◽  
Author(s):  
Shu Li ◽  
Zhongming Xu ◽  
Zhifei Zhang ◽  
Yansong He ◽  
Jin Mao
Keyword(s):  
Source Localization ◽  
Dynamic Range ◽  
Generalized Inverse ◽  
Free Field ◽  
Computation Time ◽  
Side Lobe ◽  
Sound Sources ◽  
Low Frequencies ◽  
Common Application ◽  
Regularization Matrix

Microphone arrays have become a popular technique to identify sound sources. They can be utilized to localize the sources for various applications. The most common application is the conventional beamforming that provides the source maps with strong side lobes and poor spatial resolution at low frequencies. To overcome these problems, the focus is set on deconvolution and generalized inverse techniques such as a deconvolution approach for the mapping of acoustic sources (DAMAS) and generalized inverse beamforming (GIB). Although the source maps are clearly improved, these methods have the shortcomings of expensive computing and limited dynamic range. In this paper, we propose a source localization method called functional generalized inverse beamforming with regularization matrix (FGIBR) based on an inverse problem. Compared with GIB, the accuracy of FGIBR could be improved by introducing a new beamforming regularization matrix and a scaling parameter c0. Also the dynamic range of the source maps can be increased by applying FGIBR with an exponent parameter called order v. Several simulated examples are given to illustrate that the side lobes are suppressed and the main lobe becomes much narrow; moreover, if order v is increased, the beamforming side lobes can be sharply reduced and the actual position of the noise source can be precisely located. Then FGIBR is implemented to deal with experimental data in the free field. In the case of the experiment, the source is correctly located. The proposed FGIBR demonstrates a good performance in terms of resolution and side lobe rejection compared with other beamforming methods. Furthermore, the computation time is shown to be low if the iteration and order are reasonable.

Functional Generalized Inverse Beamforming Based on the Double-Layer Microphone Array Applied to Separate the Sound Sources

2016 ◽  
Vol 138 (2) ◽  
Author(s):  
Shu Li ◽  
Zhongming Xu ◽  
Yansong He ◽  
Zhifei Zhang ◽  
Shaoyu Song
Keyword(s):  
Double Layer ◽  
Dynamic Range ◽  
Generalized Inverse ◽  
Microphone Array ◽  
A Priori ◽  
Sound Sources ◽  
Popular Method ◽  
Array Measurements ◽  
Two Sides ◽  
Regularization Matrix

Beamforming based on microphone array measurements is a popular method for identifying sound sources. However, beamforming has many limitations that limit their precision. These limitations are addressed in research. To separate the contributions which come from two sides of the microphone array more accurately, an innovative beamforming method based on a double-layer microphone array, called functional generalized inverse beamforming (FGIB), is proposed to improve beamforming performance. This method, which involves the use of a priori beamforming regularization matrix and a matrix function to redefine the inverse problem, is combined with the advantages of both generalized inverse beamforming (GIB) and functional beamforming. Compared with GIB, with reduced iterations, the computational efficiency of FGIB is greatly improved. The dynamic range of the proposed method can be modestly improved as order v increases. Furthermore, the sidelobes gradually disappear and the mainlobes become narrower. Both simulations and experiments have shown that the sources are correctly located and separated. The proposed FGIB demonstrates the good performance when compared to other beamforming methods in terms of resolution and sidelobes level.

scholarly journals Acoustic source localization using the open spherical microphone array in the low-frequency range

2019 ◽  
Vol 283 ◽  
pp. 04001
Author(s):  
Boquan Yang ◽  
Shengguo Shi ◽  
Desen Yang
Keyword(s):  
Source Localization ◽  
Generalized Inverse ◽  
Microphone Array ◽  
Low Frequency ◽  
Poor Performance ◽  
Microphone Arrays ◽  
Three Dimension ◽  
Frequency Range ◽  
Array Aperture ◽  
Regularization Matrix

Recently, spherical microphone arrays (SMA) have become increasingly significant for source localization and identification in three dimension due to its spherical symmetry. However, conventional Spherical Harmonic Beamforming (SHB) based on SMA has limitations, such as poor resolution and high side-lobe levels in image maps. To overcome these limitations, this paper employs the iterative generalized inverse beamforming algorithm with a virtual extrapolated open spherical microphone array. The sidelobes can be suppressed and the main-lobe can be narrowed by introducing the two iteration processes into the generalized inverse beamforming (GIB) algorithm. The instability caused by uncertainties in actual measurements, such as measurement noise and configuration problems in the process of GIB, can be minimized by iteratively redefining the form of regularization matrix and the corresponding GIB localization results. In addition, the poor performance of microphone arrays in the low-frequency range due to the array aperture can be improved by using a virtual extrapolated open spherical array (EA), which has a larger array aperture. The virtual array is obtained by a kind of data preprocessing method through the regularization matrix algorithm. Both results from simulations and experiments show the feasibility and accuracy of the method.

scholarly journals DESIGNING PLANAR MICROPHONE ARRAY FOR SOUND SOURCE LOCALIZATION

Akustika ◽  
2019 ◽  
Vol 32 ◽  
pp. 123-129 ◽  
Author(s):  
Victor Ershov ◽  
Vadim Palchikovskiy
Keyword(s):  
Inclination Angle ◽  
Source Localization ◽  
Sound Source ◽  
Dynamic Range ◽  
Microphone Array ◽  
Sound Source Localization ◽  
Maximum Diameter ◽  
Design Parameters ◽  
Sound Sources ◽  
The Individual

Mathematical background for designing planar microphone array for localization of sound sources are described shortly. The designing is based on optimization of objective function, which is maximum dynamic range of sound source localization. The design parameters are radial coordinates (distance along the beam from the center of the array) and angle coordinates (beam inclination) of the microphones. It is considered the arrays with the same radial coordinates of the microphones for each beam and the independent radial coordinates of each microphone, as well as the same inclination angle for all beams and the individual inclination angle of each beam. As constraints, it is used the minimum allowable distance between two adjacent microphones, and minimum and maximum diameter of the working area of the array. The solution of the optimization problem is performed by the Minimax method. An estimation of the resolution quality of designed arrays was carried out based on localization of three monopole sources. The array of 3 m in diameter without inclination of the beams and with different radial coordinates of the microphones on each beam was found to be the most efficient configuration among the considered ones.

scholarly journals Strongly directional responses to tones and conspecific calls in the auditory nerve of the Tokay gecko, Gekko gecko.

2021 ◽  
Author(s):  
Jakob Christensen-Dalsgaard ◽  
Paula T. Kuokkanen ◽  
Jamie Emoto Matthews ◽  
Catherine E. Carr
Keyword(s):  
Auditory Nerve ◽  
Dynamic Range ◽  
Free Field ◽  
High Capacity ◽  
Auditory Pathway ◽  
Sound Sources ◽  
Gekko Gecko ◽  
Sound Direction ◽  
Tokay Gecko ◽  
Field Sound

The configuration of lizard ears, where sound can reach both surfaces of the eardrums, produces a strongly directional ear, but the subsequent processing of sound direction by the auditory pathway is unknown. We report here on directional responses from the first stage, the auditory nerve. We used laser vibrometry to measure eardrum responses in Tokay geckos, and in the same animals recorded 117 auditory nerve single fiber responses to free-field sound from radially distributed speakers. Responses from all fibers showed strongly lateralized activity at all frequencies, with an ovoidal directivity that resembled the eardrum directivity. Geckos are vocal and showed pronounced nerve fiber directionality to components of the call. To estimate the accuracy with which a gecko could discriminate between sound sources, we computed the Fisher information (FI) for each neuron. FI was highest just contralateral to the midline, front and back. Thus, the auditory nerve could provide a population code for sound source direction, and geckos should have a high capacity to differentiate between midline sound sources. In brain, binaural comparisons, for example by IE neurons, should sharpen the lateralized responses and extend the dynamic range of directionality.

scholarly journals Towards Robust Multiple Blind Source Localization Using Source Separation and Beamforming

Sensors ◽  
2021 ◽  
Vol 21 (2) ◽  
pp. 532
Author(s):  
Henglin Pu ◽  
Chao Cai ◽  
Menglan Hu ◽  
Tianping Deng ◽  
Rong Zheng ◽  
...  
Keyword(s):  
Source Localization ◽  
Sound Source ◽  
Indoor Localization ◽  
Weighting Function ◽  
Signal To Noise Ratio ◽  
Source Separation ◽  
Sound Source Localization ◽  
Angle Of Arrival ◽  
Sound Sources ◽  
Localization Algorithms

Multiple blind sound source localization is the key technology for a myriad of applications such as robotic navigation and indoor localization. However, existing solutions can only locate a few sound sources simultaneously due to the limitation imposed by the number of microphones in an array. To this end, this paper proposes a novel multiple blind sound source localization algorithms using Source seParation and BeamForming (SPBF). Our algorithm overcomes the limitations of existing solutions and can locate more blind sources than the number of microphones in an array. Specifically, we propose a novel microphone layout, enabling salient multiple source separation while still preserving their arrival time information. After then, we perform source localization via beamforming using each demixed source. Such a design allows minimizing mutual interference from different sound sources, thereby enabling finer AoA estimation. To further enhance localization performance, we design a new spectral weighting function that can enhance the signal-to-noise-ratio, allowing a relatively narrow beam and thus finer angle of arrival estimation. Simulation experiments under typical indoor situations demonstrate a maximum of only 4∘ even under up to 14 sources.

Preamplifier With Ultra Low Frequency Cutoff for Infrasonic Condenser Microphone

2012 ◽  
Author(s):  
Rasmus Trock Kinnerup ◽  
Arnold Knott ◽  
Ole Cornelius Thomsen ◽  
Kresten Marbjerg ◽  
Per Rasmussen
Keyword(s):  
Input Impedance ◽  
Dynamic Range ◽  
Cutoff Frequency ◽  
Low Frequency ◽  
Input Resistance ◽  
Bias Current ◽  
Microphone Signal ◽  
Low Frequencies ◽  
Pass Filter ◽  
Condenser Microphone

Measuring infrasonic sound sets high requirements on the instruments used. Typically the measurement chain consists of a microphone and a preamplifier. As the input resistance of the preamplifier forms a high pass filter with the capacitance of the microphone in the picofarad range, measuring ultra low frequencies becomes a challenge. The electric preamplifier presented in this paper together with a prepolarized condenser microphone form a measurement system. The developed preamplifier connects the microphone signal directly to the input of an operational amplifier with ultra high input impedance. The bias current for the preamplifier further complicates the signal amplification. A configuration of two diode-connected FETs provide the input bias current. The resulting input impedance of nearly 1 TΩ yields a total lower limiting −3 dB cutoff frequency of 8 mHz and a dynamic range of 95 dB. Being able to measure down to ultra low frequencies in the infrasonic frequency range will aid actors in the debate on wind turbine noise. Sonic booms from supersonic flights include frequencies down to 10 mHz and the preamplifier proposed in this paper will aid scientists trying to modify the N-shaped shock wave at high level which prohibits flights in land zones.

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