传声器阵列对高频弱声源的声成像识别方法

研究了传声器阵列对高频弱声源的识别定位方法。该方法根据高频声源的指向性和阵列探测特性等特点,提出了利用信噪比加权方法提高有效阵元对声成像的贡献,根据信噪比的大小对每个阵元添加不同的权值,可以显著提高传声器阵列对高频弱声源的声像清晰度。仿真分析了阵元加权和不加权两种方法对阵列声成像结果的影响,以某型号笔记本电脑电路板噪声为对象进行的实验表明,在阵列测量中充分利用有效阵元信号可以实现对声压级低达10~20 dB的微弱噪声源的精确测量。     

阵元随机均匀分布球面阵列联合噪声源定位方法

基于球面传声器阵列的噪声源定位方法,设计加工了阵元随机均匀分布64元球面传声器阵列,研究了球面近场声全息和球谐函数模态展开聚焦波束形成联合噪声源定位识别方法,对算法的性能进行了仿真分析,并利用球面传声器阵列进行了噪声源的定位识别试验.研究表明,阵元随机均匀分布球面阵列具有全空间稳定的目标定位性能,球面近场声全息对低频近距离声源具有较高的定位精度,球谐函数模态展开聚焦波束形成对高频远距离声源具有较高的定位精度,将两种方法联合进行声源的定位识别,可以在较小孔径的球面阵列和较少阵元的条件下,在宽频带范围内获得对目标声源良好的定位性能.     

大孔径声全息密集阵列的阵元数量去冗余方法

大孔径密集声全息阵列是实现对非平稳噪声成像的有效手段,通过现场快速成像可以掌握声场分布的动态特征。增加声阵列的孔径和阵元密度是提高近场声全息成像性能的主要措施,但随着声阵列阵元数量的增加,声压采集数据量增大,全息运算速度降低,全息成像动态性能下降,不利于现场即时成像分析。研究表明,在一定信噪比条件下,盲目增加阵元数量并不一定能提高近场声全息性能,为此,本文针对密集声全息阵列的阵元数量冗余问题进行研究,提出最优阵元数量估算方法,并采取去冗余处理。本文仿真和实验验证表明:针对一定信噪比条件下的密集阵列去冗余方法实际有效,可以在不降低近场声全息成像性能的前提下,降低运算量,提高全息成像速度,对于非平稳噪声的现场声全息成像测试具有实用意义。     

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。     

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...     

基于超指向性多极子矢量阵的水下低频声源方位估计方法研究

针对水下低频声源的方位估计问题,基于基元紧密排列的二维矢量阵,建立了一种超指向性波束形成方法.根据矢量基元差分运算构建各阶多极子模型,获得了几乎与频率无关的模态函数,并经加权计算可在低频条件下实现超指向性波束,以解决阵列孔径对波束性能的限制.同时,结合输出信噪比最大准则所得波束,分析了超指向性波束形成算法的稳定性与波导的影响程度,探索模态阶数与阵列参数的选取原则.通过阵列性能的仿真计算与实际阵列的测量数据表明,该算法可在小尺寸阵列孔径下获得良好的阵列波束,兼具了水下线型超指向性阵和环形超指向性阵的优点,可有效实现水下低频声源的水平方位估计;且波束性能可通过调节模态阶数与基元间距以达到最佳,并受水下声波导多途与频散效应影响有限.     

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...     

深海波导中基于采样简正波模态降维处理的广义似然比检测

针对深海波导中的水下声源检测,结合声传播与接收特性,提出了一种基于简正波模态信息降维处理的窄带声源检测方法,运用广义似然比(generalized likelihood ratio,GLR)方法导出了深海波导中的特征值检测器和恒虚警率特征值检测器(下统称为GLR检测器).理论分析表明,给定阵列输入信噪比下,GLR检测器的输出信噪比随接收数据空间维度的减小而增大.根据简正波理论,阵列接收信号声场位于由各阶采样简正波模态信息张成的空间(简称模态空间).由于阵列孔径尺寸的限制,在深海波导中常会出现"有效模态空间"维度小于阵元域接收数据空间维度的情况.基于此性质并根据阵列采样的简正波模态信息,提取"有效模态空间"以构造降维矩阵,分别导出了使用垂直线列阵和水平线列阵时的降维GLR检测统计量.数值仿真给出了GLR检测器的检测性能分析与对比,验证了降维GLR检测器的性能改善效果,同时表明水平线列阵接收声场位于更低维的"有效模态空间","有效模态空间"维度随阵元间距和声源频率的增大而减小.     

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。     

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.     

Uncertainty of array-based measurement of radiated and absorbed sound intensity

Patch near-field acoustic holography (NAH) coupled with an array of sound intensity probes allows separating the sound field incident on a surface from the one radiated by the surface itself. Although the measurement principle has been successfully used to separate the noise source contribution from disturbing sources and/or noise reflections, the method accuracy has not been investigated in the literature. We describe the results of experiments meant to evaluate the uncertainty in the identification of noise radiated by vibrating panels with different absorption characteristics in presence of an incident acoustic radiation using the statistically optimized near-field acoustic holography. Measurement errors were evaluated through tests performed in controlled acoustic conditions. Results evidenced that the measurement uncertainty depends on the accuracy of the microphone array positioning and on the incident sound field. These conclusions were in agreement with the results obtained by simulations in the phase of instrument optimization.     

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.     

Real-time near-field acoustic holography for continuously visualizing nonstationary acoustic fields

Near-field acoustic holography (NAH) is an effective tool for visualizing acoustic sources from pressure measurements made in the near-field of sources using a microphone array. The method involving the Fourier transform and some processing in the frequency-wavenumber domain is suitable for the study of stationary acoustic sources, providing an image of the spatial acoustic field for one frequency. When the behavior of acoustic sources fluctuates in time, NAH may not be used. Unlike time domain holography or transient method, the method proposed in the paper needs no transformation in the frequency domain or any assumption about local stationary properties. It is based on a time formulation of forward sound prediction or backward sound radiation in the time-wavenumber domain. The propagation is described by an analytic impulse response used to define a digital filter. The implementation of one filter in forward propagation and its inverse to recover the acoustic field on the source plane implies by simulations that real-time NAH is viable. Since a numerical filter is used rather than a Fourier transform of the time-signal, the emission on a point of the source may be rebuilt continuously and used for other post-processing applications.     

Active noise control with linear control source and sensor arrays for a noise barrier

A study is conducted on minimizing the sum of the squared acoustic pressures with a linear array of control sources and a perpendicular linear array of error sensors, placed above the top of a noise barrier. Particular angular orientations, with respect to the center of the barrier top, and spacings of the linear arrays of control sources and error sensors result in moderate to significant additional reduction of the acoustic pressure in the shadow zone. Visual inspection of the sound pressure field, with and without active noise control, found that uniform and significant additional insertion loss can be generated near the barrier. Numerical simulations were conducted to test the proposed method. For separations between control sources and error sensors much less than a quarter wavelength of the primary noise disturbance, results show that the angular orientation, of the combined linear control source and sensor arrays, is a weak factor for acoustic pressure reduction in the shadow zone. Weak angle dependence serves as an advantage to the proposed method, which yields uniform performance for any angular orientation. An angular orientation involving the alignment of the furthest error sensor with the first diffracting edge of the barrier and the primary source was observed to perform well for a variety of frequencies, since the spacing between error sensors and between control sources is of the order of a quarter-wavelength. Improved noise control in the shadow zone of a barrier is achieved by the use of two control sources and angular orientation as mentioned above. Further spatial extension of the area of reduced acoustic pressure is possible by utilizing an increased number of control sources.     

Direction-of-arrival estimation for extensible acoustic vector sensor array

Aiming at high-resolution estimation of the direction-of-arrival of closely-spaced sources at low signal-to-noise ratio regions, this paper proposes a DOA estimation algorithm that is suitable for an extensible acoustic vector sensor array. Taking the 3D array composed of the minimum number(four) of acoustic vector sensors as the acquisition module, a virtual array having the same structure as the original array structure is extended in the three-dimensional space based on the aperture expansion characteristic of higher-order cumulants. The virtual array and the real array can construct a matrix with rotational invariance, which contains the angular information for estimating DOA. The Cramer-Rao bound of the algorithm are derived. We analyze the influence of SNR, the number of snapshots and the elevation angle on the performance of the algorithm. Simulation results show that the proposed algorithm has better noise suppression ability and higher resolution in DOA estimation than the conventional ESPRIT algorithm using the acoustic vector array. Experiments are conducted to validate the proposed algorithm.     

Quantitative measurement of pass-by noise radiated by vehicles running at high speeds

It has been a challenge in the past to accurately locate and quantify the pass-by noise source radiated by the running vehicles. A system composed of a microphone array is developed in our current work to do this work. An acoustic-holography method for moving sound sources is designed to handle the Doppler effect effectively in the time domain. The effective sound pressure distribution is reconstructed on the surface of a running vehicle. The method has achieved a high calculation efficiency and is able to quantitatively measure the sound pressure at the sound source and identify the location of the main sound source. The method is also validated by the simulation experiments and the measurement tests with known moving speakers. Finally, the engine noise, tire noise, exhaust noise and wind noise of the vehicle running at different speeds are successfully identified by this method.     

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.     

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...