Acoustic analysis by spherical microphone array processing of room impulse responses

Spherical microphone arrays have been recently used for room acoustics analysis, to detect the direction-of-arrival of early room reflections, and compute directional room impulse responses and other spatial room acoustics parameters. Previous works presented methods for room acoustics analysis using spherical arrays that are based on beamforming, e.g., delay-and-sum, regular beamforming, and Dolph-Chebyshev beamforming. Although beamforming methods provide useful directional selectivity, optimal array processing methods can provide enhanced performance. However, these algorithms require an array cross-spectrum matrix with a full rank, while array data based on room impulse responses may not satisfy this condition due to the single frame data. This paper presents a smoothing technique for the cross-spectrum matrix in the frequency domain, designed for spherical microphone arrays, that can solve the problem of low rank when using room impulse response data, therefore facilitating the use of optimal array processing methods. Frequency smoothing is shown to be performed effectively using spherical arrays, due to the decoupling of frequency and angular components in the spherical harmonics domain. Experimental study with data measured in a real auditorium illustrates the performance of optimal array processing methods such as MUSIC and MVDR compared to beamforming.     

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.     

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.     

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...     

Wavelet packet based analysis of sound fields in rooms using coincident microphone arrays

This paper presents a passive analysis method for determining the spatio-temporal characteristics of sound fields in small rooms. The analysis finds an approximate directional reflectogram (ADR) which reveals the approximate arrival directions, time delays and amplitudes of the direct sound and early reflections without using a special or known sound source. A coincident microphone array is used to obtain directional recordings. The recordings are analysed by wavelet packet decomposition to determine the direction of the sound source and select wavelet packet coefficients to reconstruct the estimate of the direct sound. ADR is then computed via deconvolution using this estimate. Experiments have been carried out using synthesized recordings that were obtained from actual room impulse responses measured in two rooms for various source locations. The method estimates the source direction with a mean absolute error of about 7°. Calculated ADRs provide a good estimate of the time delays and arrival directions of acoustical reflections, whereas the amplitudes differ slightly.     

A miniaturized adaptive microphone array under directional constraint utilizing aggregated microphones

This paper introduces a miniaturized microphone array using the Directionally Constrained Minimization of Power (DCMP) method, which utilizes the transfer functions of microphones located at the same place, namely aggregated microphones. The phased microphone array realizes a noise reduction and direction of arrival (DOA) estimation system according to differences in the arrival time, phase shift, and/or the level of the sound wave for each microphone. Hence it is difficult to miniaturize the microphone array. The objective of our research is to miniaturize the system size using aggregated microphones. In this paper, we first show that the phased microphone array system and the proposed aggregated microphone system can be described within the same framework. We then apply a microphone array under directional constraint to the aggregated microphones and compare the proposed method with the microphone array. We show the directional pattern of the aggregated microphones. We also show the experimental results regarding DOA estimation.     

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。     

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.     

基于蒙特卡罗法的实时空间三维声源定位

构建了一个基于四个声音传感器的信号时延采集系统,根据采集系统得到的三个时间差和传感器的响应顺序,提出了一种基于蒙特卡罗法实时空间的三维声源定位算法.该算法通过三维声音定位的非线性方程,构建一个三维模函数,通过寻找空间全局收敛点,并根据公差容限进行变步长搜索,准确快速地计算出声源的位置.     

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...     

单层传声器阵列重建相干声场的近场声全息方法研究

针对传声器阵列两侧存在相干声源的非自由声场重建问题,提出基于球面谐波函数扩展近场声全息理论的相干声场重建方法。该方法在已知测量面两侧声源几何位置时,使用单层传声器阵列获取测量面处的声压分布,通过最小二乘法获得与目标声源和干扰噪声源响应对应的最优球波函数扩展项数和最优系数向量,结合测点位置的空间坐标进行声波分解,并分别重建出各声源在测量面上的声压分布。为了验证方法的有效性,分别给出了相干噪声源为球形声源和非球形声源的仿真验证,并在全消声室内对双扬声器产生的相干声场的重建进行了实验验证。结果表明:该方法对球形声源和非球形声源干扰下的声场重建都具有较好的效果,球形声源干扰下的重建精度更高。      

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...     

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.     

Comparison of the measured and theoretical performance of a broadband circular microphone array

The design and construction of a circular microphone array (CMA) that has a wide frequency range suitable for human hearing is presented. The design of the CMA was achieved using a technique based on simulated directivity index (DI) curves. The simulated DI curves encapsulate the critical microphone array performance limitations: spatial aliasing, measurement noise, and microphone placement errors. This paper demonstrates how the non-regularized DI curves for a given beamforming order clearly define the bandwidth of operation, in other words, the frequency band for which the beamformer has relatively constant and maximum directivity. Detailed and comprehensive experimental data that characterizes the CMA beamformer are also presented.     

基于压缩感知估计的GIS击穿放电定位方法研究*

经典的空间谱估计方法,如多重信号分离(MUSIC)方法,对噪声敏感,难以在低信噪比环境中有效地进行波达方向估计。为提升在复杂电力环境中的气体绝缘金属封闭开关设备(GIS)击穿定位能力,该文提出了一种基于极化内插的压缩感知波达方向估计方法 CSP-DOA。该方法对传声器阵列接收到的数据进行建模,形成多测量矢量模型,结合压缩感知中稀疏重构技术进行波达方向估计;同时,该方法还采用极化内插技术解决了稀疏重构中的费网格问题,进一步提升了波达方向估计精度及计算效率。通过数值模拟对算法的定位效果进行了分析,仿真结果表明CSP-DOA方法对于击穿信号有更好的定位效果。结合可见光图像匹配实现了GIS击穿信源的二维可视化定位,在某高压大厅的GIS模型上进行耐压击穿定位试验研究,试验结果进一步验证了该文方法可较好的应用于GIS的击穿定位。     

Acoustic imaging using a volumetric array

A method is described for producing images of acoustic sources using a volumetric array. Theoretical results are presented and compared to those obtained experimentally from an acoustic 64 element random spherical volumetric array. Application areas for such a system include auditorium characterisation, recording studio optimisation and covert surveillance operations.     

使用同心多环阵提升声源定位鲁棒性*

房间混响带来的多径失真是影响声源波达方向估计精度的主要因素之一。应用于环形阵列的相干信号子空间方法可以降低相干反射声带来的不利影响。该方法对环形阵进行谐波展开,并利用环谐波域导向矢量的频率无关特性聚焦各频率下的空间相关矩阵。但单环阵列存在展开系数零点,这会导致严重的噪声放大而降低定位的鲁棒性。该文提出了一种环谐波域的最小模同心多环阵,来缓解系数零点处的噪声放大问题。设计了一套麦克风阵列系统,用于评估同心多环阵列定位的鲁棒性。仿真和实验结果均表明:与相同孔径和阵元数的单环阵相比,使用最小模准则设计的同心多环阵列可以显著提升混响场景下的声源定位的稳健性。     

The effects of non-cardioid directivity on incidence angle estimation using the polar energy time curve

Assessment of desirable reflections and control of undesirable reflections in rooms are best accomplished if the reflecting surfaces are properly localized. Several measurement techniques exist to identify the incident direction of reflected sound, including the useful polar energy time curve (Polar ETC), which requires six cardioid impulse response measurements along the Cartesian axes. The purpose of this investigation is to quantify the incidence angle estimation error introduced into the Polar ETC by non-cardioid microphone directivities. The results demonstrate that errors may be minimized with a cardioid-family microphone possessing a certain range of directivities and by maximizing the measurement signal-to-noise ratio.     

Robust tracking of moving sound source using multiple model Kalman filter

In this paper a novel method for tracking an active speaker in a noisy and reverberant environment by means of a spatially distributed microphone array is presented. Firstly, a sound source localization algorithm based on time delays of arrival (TDOA) in microphone pairs provides observed position estimates. Then these remarkably noisy estimates are filtered by a multiple model Kalman filter (MMKF) in order to obtain a smoothed trajectory of the speaker’s movement. Compared with the traditional Kalman filter (KF), simulated results prove the MMKF is more robust and effective in noisy environments.