Acoustic centering of sources measured by surrounding spherical microphone arrays

The radiation patterns of acoustic sources have great significance in a wide range of applications, such as measuring the directivity of loudspeakers and investigating the radiation of musical instruments for auralization. Recently, surrounding spherical microphone arrays have been studied for sound field analysis, facilitating measurement of the pressure around a sphere and the computation of the spherical harmonics spectrum of the sound source. However, the sound radiation pattern may be affected by the location of the source inside the microphone array, which is an undesirable property when aiming to characterize source radiation in a unique manner. This paper presents a theoretical analysis of the spherical harmonics spectrum of spatially translated sources and defines four measures for the misalignment of the acoustic center of a radiating source. Optimization is used to promote optimal alignment based on the proposed measures and the errors caused by numerical and array-order limitations are investigated. This methodology is examined using both simulated and experimental data in order to investigate the performance and limitations of the different alignment methods.     

一种利用分布式传声器阵列的声源三维定位方法

为了提高噪声和混响条件下分布式传声器阵列进行声源定位的性能,提出一种利用空间稀疏性和压缩感知原理的声源三维定位方法。该方法首先通过两次离散余弦变换方式提取出声音信号特征,并用该特征来构建稀疏定位模型,以便能够综合利用语音信号的短时和长时特性,同时降低模型维数;然后利用在线字典学习技术动态调整字典,克服稀疏模型与实际信号之间的失配问题,增强稀疏定位模型的鲁棒性;进而提出一种改进的平滑l₀范数稀疏重构算法来进行声源位置解算,以提高低信噪比条件下的重构精度。仿真结果表明该方法不仅可以实现多目标定位,而且具有较强的抗噪声和抗混响能力.     

Three-dimensional sound source localization using distributed microphone arrays

To improve the performance of sound source localization based on distributed microphone arrays in noisy and reverberant environments,a sound source localization method was proposed.This method exploited the inherent spatial sparsity to convert the localization problem into a sparse recovery problem based on the compressive sensing(CS) theory.In this method two-step discrete cosine transform(DCT)-based feature extraction was utilized to cover both short-time and long-time properties of the signal and reduce the dimensions of the sparse model.Moreover,an online dictionary learning(DL) method was used to dynamically adjust the dictionary for matching the changes of audio signals,and then the sparse solution could better represent location estimations.In addition,we proposed an improved approximate l_0norm minimization algorithm to enhance reconstruction performance for sparse signals in low signal-noise ratio(SNR).The effectiveness of the proposed scheme is demonstrated by simulation results where the locations of multiple sources can be obtained in the noisy and reverberant conditions.     

一种分布式双麦克风线阵声源定位方法

分布式麦克风阵列由于比传统的单阵列具有更大的空间孔径,可以获得更好的声源定位性能,因此基于分布式麦克风阵列的声源定位方法成为当前麦克风阵列领域研究的热点之一。本文研究了一种基于分布式双麦克风线阵的声源定位方法,并进行了系统实现,从理论上剖析了该算法的定位精度与单阵列定向误差以及声源位置之间的关系,而且还揭示了声源高度扰动对该算法定位精度的影响。最后,分别通过仿真实验和实际定位系统的测试结果,验证了本文理论分析的正确性。     

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.     

差分传声器阵列期望最大化多声源方位估计方法

针对小尺寸传声器阵列多声源方位估计易受混响噪声影响的问题,提出了一种适用于差分传声器阵列的期望最大化多声源方位估计方法。首先,该方法利用期望最大化算法求解出各个时频点瞬时方位估计所应满足的高斯混合模型参数;然后,通过时频点分离技术估计出各声源的方位值。针对现有的硬、软时频点分离技术应用于差分传声器阵列所存在的缺陷,还提出了一种改进的时频点分离方法,该方法融合了软、硬分离方法所具有的优点,有效降低了时频点分离结果对混响噪声的敏感性。仿真和实测实验结果表明:相较于现有的差分传声器阵列多声源估计方法,所提方法在混响噪声环境下具有更高的估计精度和稳健性能。     

A localization approach for multiple sound sources via an expectation maximization algorithm using differential microphone arrays

Conventional sound localization approaches with small-sized microphone arrays are usually sensitive to noise and reverberation. To deal with the problem, an approach based on expectation maximization algorithm with differential microphone arrays(DMAs) is proposed.Firstly, the parameters of Gaussian mixture model for time-frequency instantaneous direction estimation are estimated through the EM algorithm, and then the direction of each sound source is estimated via time-frequency separation. In order to overcome the weakness of existing time-frequency separation techniques, an improved method, which combines the advantages of both the hard and soft separation methods, is also proposed. The improved time-frequency separation method is shown to be less sensitive to noise and reverberation. Simulation and experimental results demonstrate that the proposed localization approach is superior to its existing counterparts in terms of localization accuracy and robustness.     

Higher order differential-integral microphone arrays

This paper develops theory to design higher order directional microphone arrays. The proposed higher order designs have similar inter sensor spacings as traditional first and second order differential arrays. The Jacobi-Anger expansion is used to exploit the underlying structure of microphone signals from pairs of closely spaced sensors. Specifically, the difference and sum of these microphone signals are processed to design the novel directional array.     

Low-frequency wind noise correlation in microphone arrays

A three-axis orthogonal microphone array with ten sensors in each arm has been used to study wind noise in the frequency range from 0.05 to 50 Hz. Simultaneous measurements were made of the three components of the varying wind velocity. Measurements have been made for wind speeds from 4 to 7 m/s at three different sites. The frequency-dependent correlation of the wind noise over a range of wind velocities and atmospheric and environmental conditions in the downwind direction varies as exp(-3.2X)cos(27piX). For the crosswind and vertical directions, the correlation decays approximately as exp(-7Y), where X is the separation in wavelengths in the downwind direction and Y is this separation in the crosswind or vertical direction. Over a limited range of wave numbers, the power density spectra of the varying wind velocity varied as the wave number to the -(5/3) power and the pressure spectra as the -(7/3) power.     

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.     

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。     

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。     

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.     

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。     

基于双传声器对的多声源二维定位跟踪算法

提出一种房间混响声场环境下的多声源二维定位跟踪算法。研究了基于盲源分离的时延估计,以及联合空间分布的多个传声器对的定位算法。用高斯似然函数解决在多源、多维情况下声源定位的时延匹对模糊问题,使之能够用双传声器对实现对多个声源的二维定位,结合粒子滤波算法实现对多个运动声源的跟踪。仿真实验验证了提出算法的有效性。     

Acoustic field positioning for containerless processing

The non-contact positioning of materials in a space processing chamber is accomplished using a new type of acoustic levitator. Liquid and solid materials are positioned using a single source of sound. Fine control of position may be obtained by motion of an acoustical reflector. The electrical power required is usually less than 100 watts. The system operates satisfactorily at high and low temperatures and is adaptable as an “add-on” feature to existing space experiments. Containerless melting and solidification can be performed and a freely suspended liquid can be shaped to the contour of the sound field. Experiments are described in which aluminium, glass and plastic materials are melted and solidified in the containerless state. The system has applications to containerless crystal growth, melting and related processes.     

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.     

Location of multiple proximate flaws using perpendicular NDT ultrasonic arrays

An improved method for ultrasonic location of multiple flaws using perpendicular ultrasonic arrays, working in near-field conditions, is presented. The method uses the pulse-echo technique and is based on the processing of the information acquired by both the elemental emitting transducers and their adjacent ones. This type of processing improves the performance of a previous method that sometimes presents limitations for the detection of multiples flaws. This new method is applied to a set of ultrasonic traces, obtained with an ultrasonic prototype specifically designed for testing this method, showing the improvement obtained in imaging results.     

Front-to-back ratio analysis and optimal design of first-order steerable differential microphone arrays

First-order steerable differential microphone arrays offer many advantages,such as small size,steerable mainlobe orientation,frequency-invariant array response,...     

Acoustic scattering by multiple elliptical cylinders using collocation multipole method

This paper presents the collocation multipole method for the acoustic scattering induced by multiple elliptical cylinders subjected to an incident plane sound wave. To satisfy the Helmholtz equation in the elliptical coordinate system, the scattered acoustic field is formulated in terms of angular and radial Mathieu functions which also satisfy the radiation condition at infinity. The sound-soft or sound-hard boundary condition is satisfied by uniformly collocating points on the boundaries. For the sound-hard or Neumann conditions, the normal derivative of the acoustic pressure is determined by using the appropriate directional derivative without requiring the addition theorem of Mathieu functions. By truncating the multipole expansion, a finite linear algebraic system is derived and the scattered field can then be determined according to the given incident acoustic wave. Once the total field is calculated as the sum of the incident field and the scattered field, the near field acoustic pressure along the scatterers and the far field scattering pattern can be determined. For the acoustic scattering of one elliptical cylinder, the proposed results match well with the analytical solutions. The proposed scattered fields induced by two and three elliptical–cylindrical scatterers are critically compared with those provided by the boundary element method to validate the present method. Finally, the effects of the convexity of an elliptical scatterer, the separation between scatterers and the incident wave number and angle on the acoustic scattering are investigated.