Optimized microphone deployment for near-field acoustic holography: To be, or not to be random, that is the question

Arrays with sparse and random sensor deployment are known to be capable of delivering high quality far-field images without grating lobes. This raises the question of whether or not this idea can be applied to near-field imaging as well. To answer this question that has not yet been widely investigated in previous research, numerical simulations are undertaken in this paper to optimize the microphone deployment for both far-field and near-field arrays with the latter being the main focus. In the simulation, a recently introduced near-field equivalent source imaging (NESI) technique is employed for the near-field imaging. Global optimization techniques including the simulated annealing (SA) algorithm and the intra-block Monte Carlo (IBMC) algorithm are exploited to find the optimal microphone position efficiently. The combined use of the SA and the IBMC algorithms enables efficient search for satisfactory deployment with excellent beam pattern and relatively uniform distribution of microphones. In the near-field optimization, a special kind of beam pattern and cost function definition is used for the multiple-input-multiple-output (MIMO) imaging problem. As indicated by the simulation results, random deployment of microphones is necessary to avoid grating lobes in far-field imaging. In the near-field simulation, all results suggest that the optimal near-field array is the uniform rectangular array (URA) and the random deployment presents no particular benefit in near-field imaging.     

Kalman filter-based microphone array signal processing using the equivalent source model

This paper demonstrates that microphone array signal processing can be implemented by using adaptive model-based filtering approaches. Nearfield and farfield sound propagation models are formulated into state-space forms in light of the Equivalent Source Method (ESM). In the model, the unknown source amplitudes of the virtual sources are adaptively estimated by using Kalman filters (KFs). The nearfield array aimed at noise source identification is based on a Multiple-Input–Multiple-Output (MIMO) state-space model with minimal realization, whereas the farfield array technique aimed at speech quality enhancement is based on a Single-Input–Multiple-Output (SIMO) state-space model. Performance of the nearfield array is evaluated in terms of relative error of the velocity reconstructed on the actual source surface. Numerical simulations for the nearfield array were conducted with a baffled planar piston source. From the error metric, the proposed KF algorithm proved effective in identifying noise sources. Objective simulations and subjective experiments are undertaken to validate the proposed farfield arrays in comparison with two conventional methods. The results of objective tests indicated that the farfield arrays significantly enhanced the speech quality and word recognition rate. The results of subjective tests post-processed with the analysis of variance (ANOVA) and a post-hoc Fisher's least significant difference (LSD) test have shown great promise in the KF-based microphone array signal processing techniques.     

单层传声器阵列信号空间重采样的声波分离方法

为了重构非自由声场中目标声源的声场响应,提出单层传声器阵列信号空间重采样的声波分离方法.以球面波函数为基函数,建立由系列球面波函数叠加表达的声场数学模型.基于近场声全息原理,利用单层传声器阵列面上空间重采样形成的两组声压测量信号,求解基函数系数,并重构出传声器阵列两侧声源各自的声场响应,实现声波分离.使用脉动球和振动球共同作用的非自由声场,检验了数学模型以及传声器信号信噪比、传声器阵列形状和面积、声源中心位置、频率等关键参数对声波分离精度的影响,并在全消声室内进行了实验验证.最后,对单层传声器阵列重采样的声波分离方法的实施给出了建议.     

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.     

三维空间非同步测量高分辨率声成像方法

为解决三维空间中声源成像分辨率低的问题,提出一种基于非同步测量的三维空间声成像方法.该方法首先通过移动球形传声器阵列扫描空间分布的声源,然后利用非同步测量技术近似得到大孔径、高密度的传声器阵列测量结果,最后通过传统波束形成算法成像.仿真及实验结果表明,该方法与单次测量下的波束形成方法相比,聚焦性能更好,空间分辨率高,可...     

采用球谐分解的二范数广义逆波束形成

基于传声器阵列的声成像技术是解决噪声源识别的有效途径之一.该文提出了一种基于球谐分解的L2范数广义逆波束形成算法,并对此算法在分布式球形阵列布放方案下进行了定位精度及鲁棒性的对比分析研究.仿真结果显示,此算法对低频相干声源具有较高的空间定位精确度,且阵元位置误差对此算法性能的影响有限.通过在半消声室进行实验进一步证明了...     

Sound radiation from Caribbean steelpans using nearfield acoustical holography

The sound-field of a Caribbean low tenor steelpan is examined using nearfield acoustical holography (NAH). The particular instrument, a Bertrand Kelman low tenor, has been previously examined in a single plane using a four microphone sound intensity technique. This paper will examine the sound radiation in a cuboid space enclosing the instrument at three frequencies. Presented will be the results of sound pressure level (SPL), active acoustic intensity (AI), and reactive acoustic intensity (RI) obtained through application of the NAH technique. The limitations of NAH when applied to an irregularly shaped object such as a tenor steelpan will also be presented.     

提高成像分辨率的远场多孔声全息方法

针对远场声全息成像分辨率较差的问题,提出多孔声全息方法。利用多个小型传声器阵列进行优化布置,可以有效地提高分辨率,达到等效大阵列的声场可视化效果。首先推导了空间多孔加窗对远场Kirchhoff衍射积分声全息带来的干涉叠加效应;然后对主瓣宽度和旁瓣高度进行了定量计算,阐述了多孔优化布置方法,实现了旁瓣抑制条件下的分辨率最优;最后利用双音箱实验,验证了多孔声全息的可行性和理论分辨率,证明了多孔声全息方法的有效性。研究表明,多孔声全息的干涉叠加方法可以有效提高远场测量的重建分辨率,多孔声全息的优化设计方法可以实现旁瓣抑制条件下的分辨率最优。      

基于声全息方法和双目视觉实现运动声源声场可视化

运动声源声场的可视化是一种重要的运动声源定位的技术手段,利用双目视觉测量技术实现运动声源声场空间的自动测量,自动确定运动声源表面的空间位置,针对声源表面,利用传声器阵列,基于声全息方法实现运动声源声场的重建,建立视频图像与声场的空间映射,并建立视频与声场之间的时序,实现实景视频图像与声场重建结果的融合,可以自动生成声源运动过程的视频。基于该方法所开发了一套试验测量系统,对运动声源的测量试验结果表明,该方法可以有效实现运动声源的视频可视化,使人可以直接从视频中看到声源及其变化过程,使声源的定位和识别变得更加简单。      

Video visualization for moving sound sources based on binoculars stereo and acoustic holography

The video visualization methods for moving sound sources are very important. The spatial positions in the sound filed measurement of moving sources arc measured by an automated method based on the binoculars stereo technique,the spatial coordinates of the surface of the moving sound source are determined automatically.The sound field of the surface of the moving source is reconstructed with a microphone array using the acoustic holography method for moving sound sources.The spatial mapping between the sound field and the video images is constructed,and also the time sequence of the sound field and the images is founded. Then,the reconstructed sound field and the real-world images are merged,and the visualized sound field video of the moving source is generated automatically.A sound field measurement system based on this method is developed,the measurement results of the moving sound source with this system indicate that the sound field video visualization is realized effectively with the proposed method,the sound sources and their changing process can be seen directly from the generated video,the identification and localization for the moving sound sources can be easier with this method.     

吸顶式传声器阵列阵元坐标标定方法

提出了一种吸顶式传声器阵列阵元坐标的标定方法。针对在混响声场中,时延估计算法性能严重下降从而导致在标定传声器阵元坐标时产生较大误差的问题,提出了利用脉冲声源作为标定声源,并且截取脉冲源直达声的方法来抑制混响声场的影响,提高传声器阵元坐标标定的精度。建立了阵元坐标标定的误差分析模型,并以白噪声和脉冲声源作为标定声源进行数据仿真和对比分析。仿真结果表明,使用脉冲声源作为标定声源能有效地抑制混响声场的影响,获得传声器阵列阵元的准确坐标。同时,在封闭的房间内建立起孔径为3.5 m、64阵元的螺旋状吸顶传声器阵列进行了实验研究,实验结果验证了本文提出方法的有效性。      

球谐域自适应混响抵消与声源定位算法

提出了一种基于球谐域的自适应混响抵消与声源定位算法,该方法通过去混响处理改善语音质量,并提高球谐域定位算法在混响环境下的定位性能。推导了基于多通道线性预测的自适应混响抵消算法在球谐域的表达式,针对刚球模型提出分阶处理的去混响方法,并对去混响后的信号进行波达方向估计。采用32元球阵的仿真结果表明,相比于球谐域不分阶去混响方法,该方法最大可减少约2/3的运算量,同时语音PESQ得分及SRMR均显著提高。利用实验数据对算法性能进行测试,实验结果验证了该方法在实际声学环境中去混响和声源定位的有效性。      

球面近场声全息中的阵列传声器分布与球面数值积分

球形传声器阵列是实现封闭空间三维声场重构的测量前端,传声器在球形阵列面上的分布对三维声场重构的精度有重要影响。针对球面近场声全息方法,结合I.H.Sloan,Spherical t-designs,J.Fliege等球面数值积分法,检验传声器数目、传声器几何位置分布、积分权重、波数、基函数扩展项数、声场重构半径等参数对声场重构精度的影响。结果表明:Spherical t-designs球面数值积分法与其它积分法相比,在声场重构误差接近的情形下,允许传声器的数目和位置布置有更多的选择,并且在传声器数目一定时,各点积分权重相等,使得球面近场声全息中球面数值积分计算相对简单。      

Regularization for improving the deconvolution in real-time near-field acoustic holography

Near-field acoustic holography is a measuring process for locating and characterizing stationary sound sources from measurements made by a microphone array in the near-field of the acoustic source plane. A technique called real-time near-field acoustic holography (RT-NAH) has been introduced to extend this method in the case of nonstationary sources. This technique is based on a formulation which describes the propagation of time-dependent sound pressure signals on a forward plane using a convolution product with an impulse response in the time-wavenumber domain. Thus the backward propagation of the pressure field is obtained by deconvolution. Taking the evanescent waves into account in RT-NAH improves the spatial resolution of the solution but makes the deconvolution problem "ill-posed" and often yields inappropriate solutions. The purpose of this paper is to focus on solving this deconvolution problem. Two deconvolution methods are compared: one uses a singular value decomposition and a standard Tikhonov regularization and the other one is based on optimum Wiener filtering. A simulation involving monopoles driven by nonstationary signals demonstrates, by means of objective indicators, the accuracy of the time-dependent reconstructed sound field. The results highlight the advantage of using regularization and particularly in the presence of measurement noise.     

基于时间反转的复杂声场拾声传声器阵列性能研究

探讨时间反转技术在复杂声场传声器阵列拾声中应用的可行性及其机理,给出其一般规律和性能。研究表明:在自由空间中,其拾声性能与频率,阵列形状和半径有关,频率越高,半径越大,拾声效果越好。在普通房间中,在语音频段内,圆弧阵列在预定目标点处的阵列增益性能要比离预定目标点约25 cm远处的位置处大5 dB以上。在普通房间和混响室中的实验验证了上述结论。      

基于声传播算子的声源定位实验模拟方法研究

声传播算子是一种高效的时域声场计算方法,它能够很方便地计算出给定系统参数下任意时刻任意位置的声场变化情况,本文采用这种方法计算所得的二维房间声场信息进行传声器阵列的声源定位仿真实验。计算结果表明,用该方法获取的阵列数据能有效地应用于阵列信号处理算法中,准确地估计出初始高斯脉冲声源的方向。声传播算子声场计算方法能为传声器阵列声源定位的实验提供方便,使得传声器阵列声源定位算法在不同混响时间的鲁棒性实验研究变得更加简捷。     

球形传声器阵列下基于主导声源检测MUSIC群时延算法的多声源定位*

针对混响环境中,多径效应、散射、衍射等原因导致声源定位失败或分辨能力不足的现象,提出一种基于主导声源检测MUSIC群时延的邻近多声源定位方法。该方法采用球形传声器阵列,相比平面阵列可以捕获3D声场信息,利用球谐域下信号的频率分量与角度分量解耦的优势,从而可直接利用频率平滑技术处理宽带语声信号而不需要构造聚焦矩阵,并在球谐域下通过设置阈值对一组时频段进行主导声源检测,从而选择出包含直达声的一组时频块来构造MUSIC群时延空间谱。上述举措在提升波达方向估计在高混响环境下定位鲁棒性的同时,也提高了多个邻近声源的分辨能力。仿真实验结果表明,所提出的主导声源检测MUSIC群时延算法,在高混响和低信噪比条件下,仍具有更好的定位精度与更优的邻近多声源分辨效果。     

Beamforming correction for dipole measurement using two-dimensional microphone arrays

In this paper, a beamforming correction for identifying dipole sources by means of phased microphone array measurements is presented and implemented numerically and experimentally. Conventional beamforming techniques, which are developed for monopole sources, can lead to significant errors when applied to reconstruct dipole sources. A previous correction technique to microphone signals is extended to account for both source location and source power for two-dimensional microphone arrays. The new dipole-beamforming algorithm is developed by modifying the basic source definition used for beamforming. This technique improves the previous signal correction method and yields a beamformer applicable to sources which are suspected to be dipole in nature. Numerical simulations are performed, which validate the capability of this beamformer to recover ideal dipole sources. The beamforming correction is applied to the identification of realistic aeolian-tone dipoles and shows an improvement of array performance on estimating dipole source powers.     

基于遗传算法的传声器阵列位置有源校正研究

传声器阵列的位置误差会导致高精度定位应用场景中算法性能下降,为解决这一问题,该文提出了一种二维平面传声器阵列位置参数的有源校正方法.在校正声源位置已知的情况下,利用各阵元间的到达时间差均方误差之和最小化作为优化目标设计代价函数,根据遗传算法搜索阵列真实位置的全局最优解,从而估计得到传声器阵列的准确位置信息.仿真与半消声...