High-resolution plane-wave decomposition in an auditorium using a dual-radius scanning spherical microphone array

The spatial and temporal distribution of early reflections in an auditorium is considered important for sound perception. Previous studies presented measurement and analysis methods based on spherical microphone arrays and plane-wave decomposition that could provide information on the direction and time of arrival of early reflections. This paper presents recent results of room acoustics analysis based on a spherical microphone array, which employs high spherical harmonics order for improved spatial resolution, and a dual-radius spherical measurement array to avoid ill-conditioning at the null frequencies of the spherical Bessel function. Spatial-temporal analysis is performed to produce directional impulse responses, while analysis based on the windowed Fourier transform is employed to detect direction of arrival of individual reflections at selected frequencies. Experimental results of sound-field analysis in a real auditorium are also presented.     

A comparison of measured room acoustics metrics using a spherical microphone array and conventional methods

The traditional microphone configuration used to measure room impulse responses (IRs) according to ISO 3382:2009 is an omnidirectional and figure-8 microphone pair. IRs measurements were taken in a 2500-seat auditorium to determine how the results from a spherical microphone array (an mh acoustics Eigenmike-em32) compare to those from the traditional microphone setup (a Brüel & Kjær Type-4192 omnidirectional microphone and a Sennheiser MKH30 figure-8 microphone). Measurements were obtained at six receiver locations, with three repetitions each in order to first evaluate repeatability. The metrics considered in this study were: reverberation time (T30), early decay time (EDT), clarity index (C80), strength (G), lateral energy fraction (J_LF) and late lateral energy level (L_J). Before calculating these quantities, the IRs were filtered to equalize the frequency response of the microphones and sound source. For the spherical array measurements, the omnidirectional (monopole) and figure-8 (dipole) patterns were extracted using beamforming. In terms of repeatability, the average standard deviation of the three measurements at each receiver location averaged across all metrics, receivers, and octave bands was found to be 0.01 just noticeable differences (JNDs). The analysis comparing the measurements from the two microphone configurations yielded differences which were less than 1 JND for the majority of metrics, with a few exceptions of EDT and C80 slightly above 1 JND. Based on this case study, these results indicate that spherical microphone arrays can be used to obtain valid room IR measurements, which will allow for the development of new metrics utilizing the higher spatial resolution made possible with spherical arrays.     

Localization of distinct reflections in rooms using spherical microphone array eigenbeam processing

This paper presents an experimental and comparative study of several spherical microphone array eigenbeam (EB) processing techniques for localization of early reflections in room acoustic environments, which is a relevant research topic in both audio signal processing and room acoustics. This paper focuses on steered beamformer-based and subspace-based localization techniques implemented in the spherical EB domain, including the plane-wave decomposition, eigenbeam delay and sum, eigenbeam minimum variance distortionless response, eigenbeam multiple signal classification (EB-MUSIC), and eigenbeam estimation of signal parameters via rotational invariance techniques (EB-ESPRIT) methods. The directions of arrival of the original sound source and the associated reflection signals in acoustic environments are estimated from acoustic maps of the rooms, which are obtained using a spherical microphone array. The EB-domain-based frequency smoothing and white noise gain control techniques are derived and employed to improve the performance and robustness of reflection localization. The applicability of the presented methods in practice is confirmed by experiments carried out in real rooms.     

Rotating beamforming – motion-compensation in the frequency domain and application of high-resolution beamforming algorithms

A method is presented for the location of rotating sound sources by a microphone array. In contrast to other methods which are formulated in the time domain, this method works completely in the frequency domain and allows the application of advanced, high resolution beamforming techniques. Following the work by Lowis and Joseph for ducted sources, it is shown that a generalized cross-spectral matrix in the rotating frame of reference can be calculated which can serve as a starting point for advanced beamforming techniques. The Green's function of a moving point source under free space conditions is expressed in spherical coordinates, which are, for numerical reasons, more convenient than cylindrical coordinates. The microphones on the array have to be arranged in a ring. As a practical example high resolution beamforming deconvolution methods are applied to simulated data and measured data from rotating sources.     

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.     

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.     

Legendre函数加权的声纳宽频带恒定束宽波束特性分析*

为了处理水声信号,声纳阵列通常需要形成宽频带恒定束宽的波束。采用两种阵元加权方法分析二维圆弧形恒定束宽换能器声纳阵列:球面Legendre函数加权方法和柱面Legendre函数加权方法。分别对球面阵、柱面阵和平面阵这三种几何结构的声纳阵列进行分析,并且计算波束宽度和波束方向。结果表明,在宽频带范围内,除了柱面Legendre函数加权的球面阵之外,其余Legendre函数加权的声纳阵列均能利用简单的、不随频率变化的阵元权重和阵元延时,形成恒定束宽的波束,并且具有较小的旁瓣,此外波束方向与预设方向也较为一致。相对于其他恒定束宽波束形成方法,Legendre函数加权方法能利用较低的计算复杂度来实现良好的宽频带恒定束宽的波束特性。     

圆柱阵子阵分级处理的稳健超指向性波束形成方法

提出了圆柱阵子阵分级处理的稳健超指向性波束形成方法。首先建立了圆柱阵分两级子阵进行波束形成的信号模型,接着利用空间均匀噪声场中噪声互谱矩阵的循环特性得到基于特征波束分解与综合模型的圆柱阵超指向性的最优解,然后仿真研究了其误差敏感度函数、阵增益和波束图等性能指标,并与圆柱阵的传统全局处理方法进行了对比。提出的分两级子阵处理的超指向性方法与传统全局处理方法相比不仅降低了数据存储量和波束形成计算量,而且进一步提升了稳健性,并且在低频段的阵增益远远高于常规波束形成的值,对水下声呐阵列的设计具有一定的参考价值。      

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.     

Source excitation strategies for obtaining impulse responses in finite difference time domain room acoustics simulation

This paper considers source excitation strategies in finite difference time domain room acoustics simulations for auralization purposes. We demonstrate that FDTD simulations can be conducted to obtain impulse responses based on unit impulse excitation, this being the shortest, simplest and most efficiently implemented signal that might be applied. Single, rather than double, precision accuracy simulations might be implemented where memory use is critical but the consequence is a remarkably increased noise floor. Hard source excitation introduces a discontinuity in the simulated acoustic field resulting in a shift of resonant modes from expected values. Additive sources do not introduce such discontinuities, but instead result in a broadband offset across the frequency spectrum. Transparent sources address both of these issues and with unit impulse excitation the calculation of the compensation filters required to implement transparency is also simplified. However, both transparent and additive source excitation demonstrate solution growth problems for a bounded space. Any of these approaches might be used if the consequences are understood and compensated for, however, for room acoustics simulation the hard source is the least favorable due to the fundamental changes it imparts on the underlying geometry. These methods are further tested through the implementation of a directional sound source based on multiple omnidirectional point sources.     

基于波束形成的多类型多声源定位研究

为实现空压机多噪声源的准确定位,仿真对比了多种近场球面波多声源定位算法。基于时域波束形成,研究了相同声源平面、不同声源频率、不同声源纵向距离、不同声源强度下多声源定位以及声源频率、声源纵向距离和声源强度多因素联合的多声源定位仿真方法,模拟了更接近实际的噪声源类型。基于频域波束形成,仿真研究了1400 Hz,2400 Hz,3400 Hz,4400 Hz的多声源。分别利用互功率谱波束形成和除自谱的互功率谱波束形成,仿真研究了相干声源和不相干声源。开发了阵列声成像测试平台,运用频域波束形成和功率谱波束形成对空压机进行了定位试验研究。结果表明,1400 Hz下空压机的主要噪声源是气缸盖、空气滤清器和曲轴附近的机体,这可为空压机减振降噪改进设计提供依据。     

Maximization of the directivity ratio with the desired audible gain level for broadband design of near field loudspeaker arrays

This paper mainly addresses design methods for near field loudspeaker arrays. These methods have been studied recently since they can be used to realize a personal audio space without the use of headphones. From a practical view point, they can also be used to form a directional sound beam within a short distance from the sources especially using a linear loudspeaker array. In this regard, we re-analyzed the previous near field beamforming methods in order to obtain a comprehensive near field beamforming formulation. Broadband directivity control is proposed for multi-objective optimization, which maximizes the directivity with the desired gain, where both the directivity and the gain are commonly used array performance measures. This method of control aims to form a directive sound beam within a short distance while widening the frequency range of the beamforming. Simulation and experimental results demonstrate that broadband directivity control achieves higher directivity and gain over our whole frequency range of interest compared with previous beamforming methods.     

A directional acoustic array using silicon micromachined piezoresistive microphones

The need for noise source localization and characterization has driven the development of advanced sound field measurement techniques using microphone arrays. Unfortunately, the cost and complexity of these systems currently limit their widespread use. Directional acoustic arrays are commonly used in wind tunnel studies of aeroacoustic sources and may consist of hundreds of condenser microphones. A microelectromechanical system (MEMS)-based directional acoustic array system is presented to demonstrate key technologies to reduce the cost, increase the mobility, and improve the data processing efficiency versus conventional systems. The system uses 16 hybrid-packaged MEMS silicon piezoresistive microphones that are mounted to a printed circuit board. In addition, a high-speed signal processing system was employed to generate the array response in near real time. Dynamic calibrations of the microphone sensor modules indicate an average sensitivity of 831 microV/Pa with matched magnitude (+/-0.6 dB) and phase (+/-1 degree) responses between devices. The array system was characterized in an anechoic chamber using a monopole source as a function of frequency, sound pressure level, and source location. The performance of the MEMS-based array is comparable to conventional array systems and also benefits from significant cost savings.     

Design of robust broadband beamformers using worst-case performance optimization for time-domain spherical microphone arrays

Spherical array modal beamformers are known to be sensitive to microphone mismatches. To combat the problem, robust design of spherical broadband beamformers with the time-domain implementation structure using the Worst-Case Performance Optimization(WCPO) is studied. It is shown that the conservativeness of the upper error bound on beamforming response is the primary factor to degrade the performance of the existing WCPO-based approach. Then an improved design approach for spherical broadband beamformers is presented using a stricter upper error bound on beamforming response. Theoretical analysis shows that the cost function of the proposed design is smaller and the range of feasible set of the proposed design is also larger when compared with the existing design. Moreover, a scheme on constraint specification has been proposed to reduce fluctuation of beamforming response over frequency at steering direction to alleviate signal distortion. Simulation results show that the proposed design outperforms its existing counterparts and exhibits a lower sidelobe level under the same condition of microphone mismatches.     

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...     

时域实现结构球面阵的最差性能优化稳健宽带波束形成器设计

针对球面阵模态域波束形成器对传声器失配误差敏感的问题,研究了时域实现结构球面阵的最差性能优化稳健宽带波束形成器设计。揭示了波束响应误差上界的保守性是影响现有最差性能优化方法性能的主要原因,进而提出了一种波束响应误差上界更为紧凑的优化设计方法。理论分析表明,与现有最差性能优化方法相比,所提出方法的代价函数更小、且可行解的范围也更大。针对最差性能优化设计方法存在的波束指向上阵列响应随频率变化波动偏大而引起信号失真的问题,还给出了一种减小波束图在指向方向上受信号频率影响的约束方法。仿真分析结果表明:在相同传声器失配误差条件下,所提出的设计方法要优于现有方法,得到的波束图具有更低的旁瓣级。      

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。     

用于自适应波束形成语音增强的球谐域掩蔽函数估计方法

提出一种用于球形阵列自适应波束形成的掩蔽函数估计方法.该方法利用包含空间信息的球谐系数提取低维空间向量,并采用复高斯混合模型和深度学习两种方案来估计掩蔽函数,最终利用估计的掩蔽函数设计最小方差无失真响应波束形成器,以达到空域滤波的效果.理论分析和仿真实验证明,对于相同时长的声信号,球谐域掩蔽函数估计方法的计算复杂度比传...     

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。      

基于自适应加权约束最小二乘法的麦克风阵列稳健频率不变波束形成算法

为了解决麦克风阵列通道失配时波束形成算法的稳健性问题, 提出一种基于自适应加权约束最小二乘法的麦克风阵列稳健频率不变波束形成算法. 该算法在分析无通道失配和通道失配时阵列模型特点基础上, 深入研究了通道失配时约束最小二乘频率不变波束形成算法存在的问题及其产生的原因; 将麦克风特性的概率密度函数作为稳健因子加入到约束最小二乘频率不变波束形成算法后, 其频率不变性的稳健性得到了一定的提高, 但稳健性仍较差. 为了进一步提高约束最小二乘法频率不变波束形成算法的稳键性, 通过定义代价函数中控制频率不变性的动态加权系数来调节旁瓣频谱能量, 大大提高了频率不变波束形成算法的稳键性, 将频率不变的频带范围内同一到达角度上不同频率所形成的阵列响应的最大值与最小值之比定义为波动误差, 并作为比较本文算法与约束最小二乘稳健波束形成算法和minmax稳健波束形成算法在通道失配时频率不变性稳键性的评价指标. 算法实例验证结果表明, 在麦克风阵列通道失配时, 本文算法的波动误差最小、频率不变波束形成稳健性最好, 而且适用于任意结构的阵列.