采用声场成像的声源自适应降噪方法

针对现有声音定位技术动态适应能力弱的问题,基于由21个MSM261S4030H0组成的麦克风阵列群和声波接收时间差算法实现的声源定位技术,提出声源自适应降噪方法.该方法在声场成像的基础上,通过能根据声源实际情况调整参量的异常值剔除和迭代修正复合式降噪过程,兼顾定位的准确性和实时性,实现8 m内的声源自适应定位.     

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.     

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...     

基于倾斜阵的距离与方位联合估计*

针对阵列倾斜引起的阵不变量方法定位精度下降的问题,该文结合倾斜阵声源定位的三维模型提出一种基于倾斜阵的距离与方位联合估计算法。该算法通过联合自校正方位距离联合估计方法对阵列倾角在声源-接收平面的投影量进行估计和补偿,改善目标距离估计误差的同时,利用匹配出的阵列倾角投影量反向估计目标方位,获得对声源二维位置的估计。通过仿真验证了该算法能够在阵列倾角较小时实现目标方位与距离的联合估计。     

分布式压缩感知麦克风阵列多声源方位估计*

麦克风阵列已被广泛应用于音/视频会议等人机交互领域中时,多声源应用场景对声源方位估计性能提出了更高的要求。压缩感知(CS)声源定位算法将声源定位问题转化为信号的稀疏重构问题,相比传统的定位算法如相位变换加权（SRP-PHAT）和时延累加定位（DS）能够获得较高的定位性能,但多声源的存在一定程度上降低了稀疏程度,影响了CS重构性能。考虑到传统的CS定位算法并未利用多个连续语音帧之间声源空间向量的共同稀疏性,提出采用分布式压缩感知（DCS）理论以改善多声源的稀疏恢复估计的性能。仿真和实验结果表明,相比于传统定位算法和CS-OMP算法,DCS-SOMP算法在不同信噪比和不同声源强度的环境中,对多声源的方位估计都具有更好的定位性能和定位稳健性。     

正则化方法在相位麦克风阵列技术中的应用研究

相位麦克风阵列技术是最近十年快速发展起来的一种声学实验技术,旨在精确地识别和定位出噪声源,反演出每个声源的频率、位置和幅值信息。然而对于复杂分布声源,已有的麦克风阵列后处理技术难以获得令人满意的结果,声源频率通常可以精确识别,而位置和幅值重构误差较大。为此,本文引用了一种正则化技术来寻求获得精确解,进行了相关的数值验证和实验验证。结果表明,采用正则化方法能够很好地解决反问题算法中存在的幅值误差放大现象,从而比较准确地识别和定位出声源信息。     

风场环境中声速修正的分布式声源定位算法

为减小声速误差对定位精度的影响,提出了一种基于声速修正的分布式声源定位方法。首先,将声速表示为未知声源位置的函数,逼近风场中的声速场分布,然后将其代入TDOA (Time Differences of Arrival)算法中,构建非线性超定方程组,最后采用粒子群优化算法求解声源位置。对不同风速、不同声源位置及不同测试区域进行仿真,结果表明:修正后的定位精度比修正前有明显提高,尤其对于大范围并且声源靠近测试区域边缘位置的定位系统,改善更加明显;4个节点的定位系统实验结果表明,修正后的定位误差可降至修正前的4l%,该方法能更好的应用于风场中的定位系统。      

基于音视频信息融合的目标检测与跟踪算法

针对单一视觉跟踪算法易受遮挡影响的缺陷,提出一种基于音视频信息融合的目标检测与跟踪算法。整个算法框架包括视频检测与跟踪、声源定位、音视频信息融合跟踪3个模块。视频检测与跟踪模块采用YOLOv5m算法作为视觉检测的框架,使用无迹卡尔曼滤波和匈牙利算法实现多目标的跟踪与匹配;声源定位模块采用十字型麦克风阵列获取音频信息,结合各麦克风接收信号的时延计算声源方位;音视频信息融合跟踪模块构建音视频似然函数和音视频重要性采样函数,采用重要性粒子滤波作为音视频融合跟踪的算法,实现对目标的跟踪。在室内复杂环境下对算法性能进行测试,结果表明该算法跟踪准确率达到90.68%,相较于单一模态算法具有更好的性能。     

基于误差修正的传感器阵列水声定位算法

本文分析了海洋环境中定位精度下降的根源,提出了一种基于误差修正的传感器阵列自主定位算法,并进行了计算机仿真和数据分析,结果表明该算法对于提高传感器阵列自主定位精度是有效的。     

球形阵列自适应网格分层定位算法

刚性球形传声器阵列可以在无空间模糊的条件下进行球谐域数据处理,基于球形阵列的可控波束形成器导向响应功率(SRP)算法定位精度高,但是计算量大,计算效率低。通过将球面致密的全局网格搜索替换为分层搜索策略可以有效减小SRP算法的计算量。提出MRE-SRP算法保持球形阵列SRP定位精度的同时降低计算量,首先通过球谐域MUSIC(SH-MUSIC)算法判断入射声源的数量减小搜索区域;其次将相对熵模型引入球谐域SRP(SH-SRP)定位算法中,提取网格分层前后的信息增益,设计自适应网格选择判据,实现分层多分辨率网格的精准再细分,从而降低计算量。实验验证了所提出算法的性能,结果显示在单双声源定位中,该算法可以实现较高的定位精度,精准选择分层网格,计算量减少75%以上。     

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.     

The influence of environmental conditions on estimation of source distance and height using a single vertical array

The performance of microphone arrays outdoors is influenced by the environmental conditions. Numerical simulations indicate that, while horizontal arrays are hardly affected, direction-of-arrival (DOA) estimation with vertical arrays becomes biased in presence of ground reflections and sound speed gradients. Turbulence leads to a huge variability in the estimates by reducing the ground effect. Ground effect can be exploited by combining classical source localization with an appropriate propagation model (ground effect inversion). Not only does this allow the source elevation and range to be determined with a single vertical array but also it allows separation of sources which can no longer be distinguished by far field localization methods. Furthermore, simulations provide detail of the achievable spatial resolution depending on frequency range, array size and localization algorithm and show a clear advantage of broadband processing. Outdoor measurements with one or two sources confirm the results of the numerical simulations.     

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.      

Acoustic positioning using multiple microphone arrays

Passive acoustic techniques are presented to solve the localization problem of a sound source in three-dimensional space using off-the-shelf hardware. Multiple microphone arrays are employed, which operate independently, in estimating the direction of arrival of sound, or, equivalently, a direction vector from the array's geometric center towards the source. Direction vectors and array centers are communicated to a central processor, where the source is localized by finding the intersection of the direction lines defined by the direction vectors and the associated array centers. The performance of the method in the air is demonstrated experimentally and compared with a state-of-the-art method that requires centralized digitization of the signals from the microphones of all the arrays.     

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。      

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...     

A novel acoustic emission beamforming method with two uniform linear arrays on plate-like structures

A novel acoustic emission (AE) source localization approach based on beamforming with two uniform linear arrays is proposed, which can localize acoustic sources without accurate velocity, and is particularly suited for plate-like structures. Two uniform line arrays are distributed in the x-axis direction and y-axis direction. The accurate x and y coordinates of AE source are determined by the two arrays respectively. To verify the location accuracy and effectiveness of the proposed approach, the simulation of AE wave propagation in a steel plate based on the finite element method and the pencil-lead-broken experiment are conducted, and the AE signals obtained from the simulations and experiments are analyzed using the proposed method. Moreover, to study the ability of the proposed method more comprehensive, a plate of carbon fiber reinforced plastics is taken for the pencil-lead-broken test, and the AE source localization is also realized. The results indicate that the two uniform linear arrays can localize different sources accurately in two directions even though the localizing velocity is deviated from the real velocity, which demonstrates the effectiveness of the proposed method in AE source localization for plate-like structures.     

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.     

Non-line-of-sight sound source localization using matched-field processing

Acoustic diffraction allows sound to travel around opaque objects and therefore may allow beyond-line-of-sight sensing of remote sound sources. This paper reports simulated and experimental results for localizing sound sources based on fully shadowed microphone array measurements. The generic geometry includes a point source, a solid 90° wedge, and a receiving array that lies entirely in the shadow defined by the source location and the wedge. Source localization performance is assessed via matched-field (MF) ambiguity surfaces as a function of receiving array configuration, and received signal-to-noise ratio for the Bartlett and minimum variance distortionless (MVD) MF processors. Here, the sound propagation model is developed from a Green's function integral treatment. A simple 16 element line array of microphones is tested in three mutually orthogonal orientations. The experiments were conducted using an approximate 50-to-1-scaled tabletop model of a blind city-street intersection and produced ambiguity surfaces from source frequencies between 17.5 and 19 kHz that were incoherently summed. The experimental results suggest that a sound source may be localized by the MVD processor when using fully shadowed arrays that have significant aperture parallel to the edge of the wedge. However, this performance is reduced significantly for signal-to-noise ratios below 40 dB.