A covariance fitting approach for correlated acoustic source mapping

Microphone arrays are commonly used for noise source localization and power estimation in aeroacoustic measurements. The delay-and-sum (DAS) beamformer, which is the most widely used beamforming algorithm in practice, suffers from low resolution and high sidelobe level problems. Therefore, deconvolution approaches, such as the deconvolution approach for the mapping of acoustic sources (DAMAS), are often used for extracting the actual source powers from the contaminated DAS results. However, most deconvolution approaches assume that the sources are uncorrelated. Although deconvolution algorithms that can deal with correlated sources, such as DAMAS for correlated sources, do exist, these algorithms are computationally impractical even for small scanning grid sizes. This paper presents a covariance fitting approach for the mapping of acoustic correlated sources (MACS), which can work with uncorrelated, partially correlated or even coherent sources with a reasonably low computational complexity. MACS minimizes a quadratic cost function in a cyclic manner by making use of convex optimization and sparsity, and is guaranteed to converge at least locally. Simulations and experimental data acquired at the University of Florida Aeroacoustic Flow Facility with a 63-element logarithmic spiral microphone array in the absence of flow are used to demonstrate the performance of MACS.     

任意几何形状和阵元指向性的传感器阵列优化波束形成方法

针对常规波束运用到实际传感器阵列时旁瓣较高这一缺点,提出了适用于任意几何形状和阵元指向性的传感器阵列的两种旁瓣约束优化波束形成方法。一种是主瓣宽度约束条件下的最低旁瓣波束形成,另一种是在所能允许的最高旁瓣约束条件下的高增益波束形成。通过对加权向量范数进行约束,提高波束形成的稳健性。这两种优化波束形成问题都可以写成二阶锥(Second-OrderCone)约束优化的形式,然后利用已有的内点方法(Interior-PointMethods)求出其数值解。计算机仿真和湖上实验数据处理结果表明,该低旁瓣波束形成方法相比于常规波束和其它基于自适应波束形成的旁瓣控制方法,在同等主瓣宽度的情况下可以获得更低的旁瓣级;该高增益波束形成方法在同等旁瓣级约束条件下可以获得更高的阵增益,而且旁瓣级能够满足设定要求。     

水下声成像中旁瓣抑制方法及其实验研究

提出了一种近场条件下采用不等间隔阵并进行孔径变迹处理的水下声成像旁瓣抑制方法,并进行了理论和实验研究。考虑以球面波传播理论为基础的聚焦波束形成,首先通过阵元位置微调,设计了可实现低旁瓣的不等间隔阵,从单程波束响应上降低旁瓣;然后,将孔径变迹处理方法应用于水下声成像中,全部阵元用于接收,部分阵元用于发射,从双程波束响应上进一步降低旁瓣。通过水池实验对所提出的方法进行了验证。结果表明:采用不等间隔阵并进行孔径变迹处理可以更为有效地降低旁瓣,而主瓣仅有小量展宽,且该方法工程应用简便易行,在改善成像质量的同时降低了系统复杂度。     

二维解卷积波束形成水下高分辨三维声成像

针对水下三维成像的空间分辨率难以提高,且具有较高旁瓣级的问题,提出了一种二维解卷积波束形成高分辨三维声成像算法,该算法首先完成任意距离切片的平面阵波束形成,近场情况下采用菲涅尔近似实现近场平面阵波束形成,然后通过二维解卷积技术对任意距离切片的二维波束形成结果进行解卷积处理,去除阵列指向性函数的影响,改善波束响应非理想冲击函数所造成波束形成主瓣宽及旁瓣级高的问题。通过计算机仿真分析,新算法可以有效的提高水下三维成像的空间分辨率,抑制旁瓣级,并能够在较宽频带和不同阵列孔径内保持与常规波束形成相当的稳定性。通过试验研究,新方法比常规波束形成实际目标成像分辨率提高一倍,最高旁瓣级下降20 dB,验证了该算法在实际系统中的有效性.     

基于压缩感知的矢量阵聚焦定位方法

本文针对噪声源近场定位识别问题,利用声源分布在空间域具有稀疏性,在压缩感知理论框架下建立了新体系下的矢量阵聚焦波束形成方法,用于解决同频相干声源的定位识别问题.新方法可在小快拍下准确获得噪声源的空间位置,且不损失对噪声源贡献相对大小的评价能力.通过详细的理论推导、仿真分析和试验验证,证明了基于压缩感知的矢量阵聚焦定位新方法本质上实现了l1范数正则化求解下的波形恢复和空间谱估计,因此具有较高的定位精度,较强的相干声源分辨能力、准确的声源贡献相对大小评价能力以及较高的背景压制能力,可应用于水下复杂噪声源的定位识别.     

声矢量传感器阵宽带相干信号子空间最优波束形成

提出了基于矢量传感器阵的宽带相干信号子空间最优波束形成。建立了矢量传感器阵的宽带阵列信号模型,将宽带聚焦思想引入到了矢量传感器阵宽带处理,证明了矢量传感器阵与声压传感器阵宽带聚焦矩阵之间的关系,为矢量传感器阵宽带聚焦矩阵的构造提供了一种方法。推导了基于矢量传感器阵宽带最优波束形成的空间谱,通过空间重采样方法实现矢量传感器阵宽带聚焦,避免了对目标源方位的预估。通过仿真验证了矢量传感器阵具有更优越的性能:可以解宽带相干源;能够实现全空间无模糊定向;可以空间欠采样,因而在不增加计算量的基础上,能够提高阵列处理的性能;克服了传统声压传感器线阵端射方向上,目标方位估计性能下降的缺点。     

High-resolution deconvolved beamforming for three-dimensional imaging sonars

Three-dimensional(3D) imaging sonars based on the conventional beamforming(CBF) suffers from relatively wide main-lobes and high sidelobe level.To improve the spatial resolution of 3D imaging sonars,a deconvolved beamforming method is proposed with the iterative Richardson-Lucy algorithm in this paper.At first,each distance slice can be processed to obtain images by the CBF.For the near field,the Fresnel approximation is used.Then,the deconvolution technique is applied to the CBF outputs to obtain high-resolution images and suppress the sidelobe level.The simulations demonstrate that the proposed algorithm is able to improve the spatial resolution significantly and suppress the sidelobes for 3D imaging sonars.Meanwhile,the algorithm shows similar robustness with the CBF in the case of wideband and sparse array.The priority of the proposed algorithm is also validated by the tank experiment data.The presented results indicate that the spatial resolution is increased by one time and the average sidelobe level is reduced by 20 dB.     

近场反卷积聚焦波束形成声图测量

为了提高声图测量中对多个声源的分辨能力和定位精度,给出了一种近场二维反卷积聚焦波束形成声图测量方法。推导了水下声图测量的广义卷积模型,根据声图测量中点传播函数移变但可预测的特点,通过预存点传播函数字典的方式,将波束形成过程中的卷积问题转化成叠加积分问题,并应用二维Richardson-Lucy迭代算法实现了二维移变模型情况下的近场二维反卷积求解,从而实现高分辨声图测量。通过仿真和海试对比了反卷积、常规声图测量和MVDR声图测量的性能,结果表明反卷积算法在500次迭代情况下聚焦峰尺度小于另外两种算法的1/2,旁瓣级下降超过6 dB.     

Performance evaluation of optical beamforming-based wideband antenna array

A wideband-generalized pattern multiplication approach to evaluate the performance of an optical beamforming-based wideband antenna array is proposed and experimentally demonstrated, which enables the far-field measurement of a large wideband array with a small anechoic chamber. Because the optimum reception of a wideband microwave signal is highly related to the time-domain distortions of the beamforming system, a correlation-receiver-based radiation pattern is applied to take the fidelity of the wideband signals into account.A four-element optical beamforming system is built to verify the feasibility of the proposed method. The results achieved by the proposed method agree well with the conventional direct measurements.     

Beamformer performance with acoustic vector sensors in air

For some time, compact acoustic vector sensors (AVSs) capable of sensing particle velocity in three orthogonal directions have been used in underwater acoustic sensing applications. Potential advantages of using AVSs in air include substantial noise reduction with a very small aperture and few channels. For this study, a four-microphone array approximating a small (1 cm3) AVS in air was constructed using three gradient microphones and one omnidirectional microphone. This study evaluates the signal extraction performance of one nonadaptive and four adaptive beamforming algorithms. Test signals, consisting of two to five speech sources, were processed with each algorithm, and the signal extraction performance was quantified by calculating the signal-to-noise ratio (SNR) of the output. For a three-microphone array, robust and nonrobust versions of a frequency-domain minimum-variance (FMV) distortionless-response beamformer produced SNR improvements of 11 to 14 dB, and a generalized sidelobe canceller (GSC) produced improvements of 5.5 to 8.5 dB. In comparison, a two-microphone omnidirectional array with a spacing of 15 cm yielded slightly lower SNR improvements for similar multi-interferer speech signals.     

逆波束形成的旁瓣抑制

为了降低普通逆波束形成旁瓣,提出了一种新的旁瓣抵消技术。该技术利用指向性函数中的旁瓣部分去估计声场的响应,然后从基阵估计的声场减去这部分指向性函数旁瓣估计的声场,从而得到最终的声场估计。由于较多的旁瓣响应被估计的声场扣除,最终估计的声场中的旁瓣大大降低。利用该技术可有效地抑制干扰,同时又可保持普通逆波束形成的方位高分辨力。湖试结果显示了该技术的有效性和可行性。     

Automatic 3D scanning surface generation for microphone array acoustic imaging

This work presents a new technique for automatically generating the 3D scanning surface for acoustic imaging using microphone arrays. Acoustic images, or maps, of sound coming from spatially distributed sources, may be generated from microphone array data using algorithms such as beamforming. Traditional 2D acoustic maps can contain errors in the near-field if the object being imaged has a 3D shape. It has been shown that using the 3D surface geometry of an object as a scanning surface for beamforming can provide more accurate results. The methods used previously to generate this 3D scanning surface have either required existing CAD (Computer-Aided Design) models of the object being acoustically imaged or have required separate equipment which is generally bulky and expensive. The new method uses one or more cameras in the array, a data projector, and structured light code to automatically generate the 3D scanning surface. This has the advantage that it is inexpensive, can be incorporated as an add-onto existing microphone arrays, has short scan time, and is capable of being extended to imaging dynamic scenes. This technique is tested using beamforming and CLEAN-SC (CLEAN based on spatial Source Coherence) algorithms for a spherical array and an Underbrink multi-arm spiral array. For sound sources located about 1.2 m from the array, the mean position errors obtained are 6 mm. This is a quarter of the diameter of the mini-speakers being used as a sound sources.     

线阵波束形成声强缩放方法估算声源的辐射声功率

为了解决波束形成声源识别过程中声源辐射声功率定量计算的问题,给出了阵型简洁、便于组合的线阵声强缩放模型。通过推导线阵的声强缩放系数,建立起线阵波束输出结果与声源辐射声功率之间的换算关系。无论是线阵还是平面阵的声强缩放方法,对于偏离阵列中心位置较远处的声源进行辐射声功率估算时都存在较为明显的误差。通过理论推导和仿真模拟计算,研究了同一单极子点声源在不同位置处的声功率估算偏差随频率、幅度的变化规律,发现该估算偏差只与声源偏离位置有关,而与声源自身的强度信息无关的结论,据此给出了相应的声功率估算修正方法。半消声室实验结果和声压法测量结果对比表明:修正后的线阵声强缩放方法用于中高频声源的辐射声功率计算时,单频声源的估算误差不超过1.0 dB,宽带声源的估算误差不超过1.8 dB。     

Robust supergain beamforming for circular array via second-order cone programming

The phenomenon of supergain for a circular array and its robust beamforming are presented. The coplanar superdirective array gain of the circular array, although it is not so extreme as an endfire line array, outperforms a lot over that of a conventional delay-and-sum beamformer in isotropic noise fields when the inter-element spacings are much smaller than one-half wavelength. However, optimum beamforming algorithms can be extremely sensitive to slight errors in array characteristics. The performance are known to degrade significantly if some of underlying assumptions on the sensor array is violated. Therefore, white noise gain constraint is used to improve the robustness of the supergain beamformer against random errors. We show that the design of the weight vector of robust supergain beamformer can be reformulated as a form of second-order cone programming and resolved efficiently via the well-established interior point method. Results of computer simulation for a 24-element circular array confirm satisfactory performance of the approach proposed in this paper.     

A robust super-resolution approach with sparsity constraint in acoustic imaging

Acoustic imaging is a standard technique for mapping acoustic source powers and positions from limited observations on microphone sensors, which often causes an ill-conditioned inverse problem. In this article, we firstly improve the forward model of acoustic power propagation by considering background noises at the sensor array, and the propagation uncertainty caused by wind tunnel effects. We then propose a robust super-resolution approach via sparsity constraint for acoustic imaging in strong background noises. The sparsity parameter is adaptively derived from the sparse distribution of source powers. The proposed approach can jointly reconstruct source powers and positions, as well as the background noise power. Our approach is compared with the conventional beamforming, deconvolution and sparse regularization methods by simulated, wind tunnel data and hybrid data respectively. It is feasible to apply the proposed approach for effectively mapping monopole sources in wind tunnel tests.     

Fast implementation of sparse iterative covariance-based estimation for source localization

Fast implementations of the sparse iterative covariance-based estimation (SPICE) algorithm are presented for source localization with a uniform linear array (ULA). SPICE is a robust, user parameter-free, high-resolution, iterative, and globally convergent estimation algorithm for array processing. SPICE offers superior resolution and lower sidelobe levels for source localization compared to the conventional delay-and-sum beamforming method; however, a traditional SPICE implementation has a higher computational complexity (which is exacerbated in higher dimensional data). It is shown that the computational complexity of the SPICE algorithm can be mitigated by exploiting the Toeplitz structure of the array output covariance matrix using Gohberg-Semencul factorization. The SPICE algorithm is also extended to the acoustic vector-sensor ULA scenario with a specific nonuniform white noise assumption, and the fast implementation is developed based on the block Toeplitz properties of the array output covariance matrix. Finally, numerical simulations illustrate the computational gains of the proposed methods.     

A fast signal subspace approach for the determination of absolute levels from phased microphone array measurements

Phased microphone arrays are used in a variety of applications for the estimation of acoustic source location and spectra. The popular conventional delay-and-sum beamforming methods used with such arrays suffer from inaccurate estimations of absolute source levels and in some cases also from low resolution. Deconvolution approaches such as DAMAS have better performance, but require high computational effort. A fast beamforming method is proposed that can be used in conjunction with a phased microphone array in applications with focus on the correct quantitative estimation of acoustic source spectra. This method bases on an eigenvalue decomposition of the cross spectral matrix of microphone signals and uses the eigenvalues from the signal subspace to estimate absolute source levels. The theoretical basis of the method is discussed together with an assessment of the quality of the estimation. Experimental tests using a loudspeaker setup and an airfoil trailing edge noise setup in an aeroacoustic wind tunnel show that the proposed method is robust and leads to reliable quantitative results.     

声矢量圆阵宽带相干目标MVDR方位估计*

针对宽带相干目标的远程探测问题，本文提出一种基于声压振速联合处理和矢量重构的声矢量圆阵MVDR波束形成方法。该方法利用相位模态变换技术，将声矢量圆阵变换为与信号频率无关的虚拟线阵，并构建虚拟线阵声压与组合振速的互协方差矩阵，利用声压与振速各分量间的空间相关性有效地抑制各向同性环境噪声；并对宽带相干信号的互协方差矩阵进行矢量重构，即将最大特征值对应的特征向量划分为相互重叠的子向量，从而构建前/后向Hermitian矩阵；最后，基于MVDR波束形成器实现宽带相干目标的方位估计。仿真计算和实验数据处理结果表明，该方法具较强的解相干能力和噪声抑制能力以及较高的方位估计性能。     

利用拖船辐射噪声的深海声学参数近场匹配场反演方法

针对深海声学参数难以通过远距离合作声源反演获取的问题,提出了利用拖船低频噪声近场匹配场反演方法。首先,利用聚焦波束形成计算拖曳阵接收拖船噪声的方向性,获得传播路径特征;然后,构建多参数反演模型,由波数积分声传播模型计算拷贝场,采用遗传算法对多频匹配场目标函数进行反演。同时,采用蒙特卡罗方法分析参数后验概率密度。仿真与试验结果表明:深海环境中拖曳阵接收拖船噪声主要来自海底反射路径,利用该特性反演得到海水深度、噪声源距离、阵列深度、沉积层厚度等参数,多频联合反演可以提高沉积层厚度等参数反演准确性。宽带匹配场处理表明,利用反演最优参数模型能准确给出拖船噪声源的空间位置。     

Waveguide invariant focusing for broadband beamforming in an oceanic waveguide

The performance of broadband sonar array processing can degrade significantly in shallow-water environments when interference becomes angularly spread due to multipath propagation. Particularly for towed line arrays near endfire, elevation angle spreading of multipath interference often results in masking of weaker sources of interest. While adaptive beamforming in a series of narrow frequency bands can suppress coherent multipath interference, this approach requires long observation times to estimate the required narrowband covariance matrices. To form wideband covariance matrices which can be estimated with less observation time, plane-wave focusing methods have been used to avoid interference covariance matrix rank inflation. This paper extends wideband focusing to the case of coherent multipath interference. The approach presented here, called waveguide invariant focusing (WIF), exploits a robust relationship for the frequency dependence of horizontal wave number differences. Unlike matched-field methods, WIF does not model multipath wave fronts but rather makes the interference appear to occupy the same rank-one subspace across frequency. This permits formation of wideband covariance matrices without interference rank inflation. Simulation experiments in a realistic ocean environment indicate that adaptive beamforming using WIF covariance matrices can provide a significant array gain improvement over conventional adaptive methods with limited observation time.