Abstract:Beamforming is done with an array of sensors to achieve a directional or spatially-specific response by using a model of the arriving wavefront. Real acoustic sources may deviate from the conventional plane wave or monopole model, causing decreased array gain or a total breakdown of beamforming. An alternative to beamforming with the conventional source model is presented which avoids this by using a more general source model. The proposed method defines a set of “sub-beamformers,” each designed to respond to … Show more
“…[1][2][3][4] In particular, acoustic source localization based on microphone array signal processing can be used to locate gunfire or aircraft in defense and homeland-security applications, to localize noise sources in the design and manufacturing process of machines in the aerospace industry, 5 or to localize an unknown number of spatially distributed sound sources in spatially distributed noise. 6 All of these applications benefit from array designs and beamforming techniques that produce narrow beams.…”
Coprime linear microphone arrays allow for narrower beams with fewer sensors. A coprime microphone array consists of two staggered uniform linear subarrays with M and N microphones, where M and N are coprime with each other. By applying spatial filtering to both subarrays and combining their outputs, M+N−1 microphones yield M⋅N directional bands. In this work, the coprime sampling theory is implemented in the form of a linear microphone array of 16 elements with coprime numbers of 9 and 8. This coprime microphone array is experimentally tested to validate the coprime array theory. Both predicted and measured results are discussed. Experimental results confirm that narrow beampatterns as predicted by the coprime sampling theory can be obtained by the coprime microphone array.
“…[1][2][3][4] In particular, acoustic source localization based on microphone array signal processing can be used to locate gunfire or aircraft in defense and homeland-security applications, to localize noise sources in the design and manufacturing process of machines in the aerospace industry, 5 or to localize an unknown number of spatially distributed sound sources in spatially distributed noise. 6 All of these applications benefit from array designs and beamforming techniques that produce narrow beams.…”
Coprime linear microphone arrays allow for narrower beams with fewer sensors. A coprime microphone array consists of two staggered uniform linear subarrays with M and N microphones, where M and N are coprime with each other. By applying spatial filtering to both subarrays and combining their outputs, M+N−1 microphones yield M⋅N directional bands. In this work, the coprime sampling theory is implemented in the form of a linear microphone array of 16 elements with coprime numbers of 9 and 8. This coprime microphone array is experimentally tested to validate the coprime array theory. Both predicted and measured results are discussed. Experimental results confirm that narrow beampatterns as predicted by the coprime sampling theory can be obtained by the coprime microphone array.
“…Microphone arrays are powerful tools which can be used for spatial recording, isolating a signal from noise and reverberation, and locating sound sources. [1][2][3] Accuracy of source localization and array gain are improved with increasing number of array elements. Using acoustic vector sensors which combine sound pressure and particle velocity shows promise; [4][5][6] however, these sensors can be expensive.…”
Section: [Mrb]mentioning
confidence: 99%
“…There has been active research in the area of array implementation during the recent two decades which has seen progress regarding array configurations as well as processing algorithms. 1,2,[7][8][9][10] Random sampling is a type of sparse sensing that has been applied to array design, wherein irregular element distribution is used. However construction is non-trivial, as optimization is often computationally demanding, thus suboptimal solutions may be found using a number of heuristic algorithms such as the Monte Carlo, simulated annealing, golden section search, and conjugate gradient algorithms, or combinations thereof.…”
Section: [Mrb]mentioning
confidence: 99%
“…Recent developments in uniform linear array processing may still be leveraged by coprime array processing, due to the uniform nature of the coprime subarrays. These include investigations by Bouchard et al 1 which use a weighted combination of sub-beamformers, each using different spatial modes for the source model. Crocco and Trucco 7 propose a method of optimizing the mean adherence between the desired and actual directivity pattern (also known as beampattern or directional characteristic), that takes advantage of the relationship between the sampling interval and the element separation of a uniform linear array.…”
Coprime arrays represent a form of sparse sensing which can achieve narrow beams using relatively few elements, exceeding the spatial Nyquist sampling limit. The purpose of this paper is to expand on and experimentally validate coprime array theory in an acoustic implementation. Two nested sparse uniform linear subarrays with coprime number of elements ( M and N) each produce grating lobes that overlap with one another completely in just one direction. When the subarray outputs are combined it is possible to retain the shared beam while mostly canceling the other superfluous grating lobes. In this way a small number of microphones ( N+M-1) creates a narrow beam at higher frequencies, comparable to a densely populated uniform linear array of MN microphones. In this work beampatterns are simulated for a range of single frequencies, as well as bands of frequencies. Narrowband experimental beampatterns are shown to correspond with simulated results even at frequencies other than the arrays design frequency. Narrowband side lobe locations are shown to correspond to the theoretical values. Side lobes in the directional pattern are mitigated by increasing bandwidth of analyzed signals. Direction of arrival estimation is also implemented for two simultaneous noise sources in a free field condition.
“…Some of the most common methods that achieve this use a receiver that electronically or physically scans space to find the Direction of Arrival (DoA). Electronic systems usually incorporate an array of sensors [1,2,3]. …”
Time-Modulated Linear Arrays (TMLAs) offer useful efficiency savings over conventional phased arrays when applied in parameter estimation applications. The present paper considers the application of TMLAs to acoustic systems and proposes an algorithm for efficiently deriving the arrival angle of a signal. The proposed technique is applied in the frequency domain, where the signal and harmonic content is captured. Using a weighted average method on harmonic amplitudes and their respective main beam angles, it is possible to determine an estimate for the signal’s direction of arrival. The method is demonstrated and evaluated using results from both numerical and practical implementations and performance data is provided. The use of Micro-Electromechanical Systems (MEMS) sensors allows time-modulation techniques to be applied at ultrasonic frequencies. Theoretical predictions for an array of five isotropic elements with half-wavelength spacing and 1000 data samples suggest an accuracy of ±1∘ within an angular range of approximately ±50∘. In experiments of a 40 kHz five-element microphone array, a Direction of Arrival (DoA) estimation within ±2.5∘ of the target signal is readily achieved inside a ±45∘ range using a single switched input stage and a simple hardware setup.
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