选择声强法用于强背景噪声下声源识别

选择声强法是一种新的测试方法,它集中了传统声强法和偏相干方法的优点,可以在声源间存在相干和高背景噪声的情况下,进行声源辐射声强的识别.本文首先论述了选择声强法的理论,并在多输入、双输出的频域模型基础上,讨论了如何利用选择声强法将声强矢量分解为与各噪声源有关的分量,然后结合在强背景噪声下扬声器辐射声强的区分实验,给出了此方法在实际应用中可能遇到的一些问题以及相应的解决办法.     

基于能量强度测量的多声源目标位置线性估计方法

基于能量强度的多声源定位模型,本文提出了一种声源发射能量强度未知下的多声源目标位置线性估计方法。将多声源定位模型转化为线性最小二乘估计问题,估计方法以代数解形式表示多声源目标位置初始值。对初始估计值进一步优化,得到了精确的多声源目标位置估计值。该定位计算方法将定位结果以代数解形式表示,避免了数值计算过程中因初始解选择不当而导致的局部最优问题。仿真测试了所设计算法的定位精度,并由此分析了噪声及声音能量强度增益对定位误差的影响。结果表明优化后的计算结果较初始估计值有较大改进,在一定噪声范围内其定位精度可以接近于克拉美罗（CRLB）下界值。     

确定声源频率和位置的超分辨阵列处理方法

基于均匀线阵接收一定距离的声源得到的时间和空间维的样本数据提出分离处理:用时间协方差矩阵确定声源频率;通过对近场空间协方差矩阵进行修正,用重构在相同方位上的远场近似空间协方差矩阵确定声源方位角。并在此基础上,通过改变阵列中心位置确定声源位置。计算机模拟和声学实验证实了上述方法的可行性.     

基于声强差效应的声源定位系统演示仪

介绍一种基于声强差效应进行设计的声源定位系统,以STM8单片机为数据处理核心,有源蜂鸣器组成近场声源模块。通过演示对近场声源的实时定位,解决了原有定位技术的对近场声源跟踪不连续的问题,使声源定位这一技术更加实用化、可视化。     

线阵波束形成声强缩放方法估算声源的辐射声功率

为了解决波束形成声源识别过程中声源辐射声功率定量计算的问题,给出了阵型简洁、便于组合的线阵声强缩放模型。通过推导线阵的声强缩放系数,建立起线阵波束输出结果与声源辐射声功率之间的换算关系。无论是线阵还是平面阵的声强缩放方法,对于偏离阵列中心位置较远处的声源进行辐射声功率估算时都存在较为明显的误差。通过理论推导和仿真模拟计算,研究了同一单极子点声源在不同位置处的声功率估算偏差随频率、幅度的变化规律,发现该估算偏差只与声源偏离位置有关,而与声源自身的强度信息无关的结论,据此给出了相应的声功率估算修正方法。半消声室实验结果和声压法测量结果对比表明:修正后的线阵声强缩放方法用于中高频声源的辐射声功率计算时,单频声源的估算误差不超过1.0 dB,宽带声源的估算误差不超过1.8 dB。     

改进的非负声强和有用声强识别表面声辐射*

识别结构表面对远场声辐射有实质贡献的区域,对控制远场声辐射有重要意义。该文以非负声强法为基础得到改进的声强计算方法,这种方法比非负声强法计算量小,能更好地识别出表面对远场有贡献的区域。该文对有用声强法从特征值和截断准则的角度进行了一些改进。最后,以四边简支薄板为例,分别采用这三种方法计算几种不同频率下的噪声辐射区域。数值计算结果表明,采用该文提出的方法可以较好地识别出结构表面对远场声辐射有实质贡献的区域,而且计算量相对较小。     

混响室内声源的声功率输出及声强分布

本文讨论了混响室内的声强分布,指出混响室内声强分布与自由场一样,对点声源服从平方反比律。对混响室及消声室的声压及声强随时间的起伏作了初步摸索,得到了几条实验规律,指出声强起伏比声压起伏更大。采用声强测量方法对同-声源在消声室及混响室内的声功率输出作了测量,说明声源的声功率输出是随环境变化的声学量,在混响室内声源的低频发射要比消声室内的发射要低。     

线阵声强缩放方法在水下声源辐射功率估算中的应用

为了解决水下声源辐射声功率难以计算的问题,利用线阵声强缩放方法在波束形成声源识别的基础上,根据波束输出结果与声源辐射声功率之间的换算关系来获得相应的声功率。为了提高线阵声强缩放方法的水下声功率估算精度,给出了一定动态范围限制的主瓣区域积分方法,并通过仿真验证了该方法的有效性。在消声水池中开展了水下声功率估算的实验研究。在不同的测试距离下,对双声源条件下的单频以及宽带声源在阵列侧的辐射声功率进行了估算,以混响法的测量结果为参考值,研究了估算误差随声源频率、测试距离等影响因素的变化规律。实验结果表明,无论是单频还是宽带声源,声功率的最大估算误差不超过2.6 dB,在高频时不超过1.6 dB。验证了线阵声强缩放方法应用于水下声源辐射声功率估算的正确性与可行性。     

基于波束形成缩放声强的声源局部声功率计算

基于波束形成法识别噪声源时,为计算主要噪声源的辐射声功率,给出了基于平面波模型的声强缩放方法,模拟计算了单极子点声源局部声功率的计算误差,结果显示:当阵列平面与声源计算平面间距离等于阵列直径时,基于波束形成缩放声强计算的声功率误差仅略高于0.1 dB。为克服旁瓣干扰,给出了具有一定动态范围的声源计算平面积分法,模拟计算了单极子点声源的局部声功率,结果表明:该积分法的计算值与主瓣区域积分法的计算值近似相等,均约等于理论声功率。进一步,波束形成法与声强法的对比算例试验验证了基于波束形成缩放声强计算声源局部声功率方法的有效性。     

声强测量技术及其在声源鉴别上的应用

在阅读了近百篇文献的基础上,本文综述了近些年来声强测量技术发展及其应用在鉴别机械声源中的一些问题,试图为今后的研究提供一些有用的线索。     

Identification of noise sources on a residential split-system air-conditioner using sound intensity measurements

This paper presents the results of experimental studies of the noise of a residential split-system air-conditioner unit. The compressor and condenser and associated fans were removed from the unit and did not form part of the studies. Care was taken with the unit to separate the inlet and exhaust noise from the noise radiated from the cabinet. The measurements were made with a two-microphone sound intensity probe and these resulted in sound power level data. The sound power levels produced by radiation from the inlet, exhaust and cabinet were obtained for five different volume flow rates. The effect on the sound power generated by removing the coil was investigated. Measurements and subjective studies show that the low frequency sound is predominantly radiated from the exhaust and inlet. At high frequency, the cabinet noise dominates.     

Enhancing maximum measurable sound reduction index using sound intensity method and strong receiving room absorption

The sound intensity method is usually recommended instead of the pressure method in the presence of strong flanking transmission. Especially when small and/or heavy specimens are tested, the flanking often causes problems in laboratories practicing only the pressure method. The purpose of this study was to determine experimentally the difference between the maximum sound reduction indices obtained by the intensity method, RI,max, and by the pressure method, Rmax. In addition, the influence of adding room absorption to the receiving room was studied. The experiments were carried out in an ordinary two-room test laboratory. The exact value of RI,max was estimated by applying a fitting equation to the measured data points. The fitting equation involved the dependence of the pressure-intensity indicator on measured acoustical parameters. In an empty receiving room, the difference between RI,max and Rmax was 4-15 dB, depending on frequency. When the average reverberation time was reduced from 3.5 to 0.6 s, the values of RI,max increased by 2-10 dB compared to the results in the empty room. Thus, it is possible to measure wall structures having 9-22 dB better sound reduction index using the intensity method than with the pressure method. This facilitates the measurements of small and/or heavy specimens in the presence of flanking. Moreover, when new laboratories are designed, the intensity method is an alternative to the pressure method which presupposes expensive isolation structures between the rooms.     

Interior and exterior sound field control using two dimensional higher-order variable-directivity sources

Spatial sound reproduction systems aim to produce a desired sound field over a volume of space. At high frequencies, the number of loudspeakers required is prohibitive. This paper shows that the use of loudspeakers with up to Nth order directivity allows reproduction over N times the bandwidth and produces a significantly attenuated exterior sound field. If the constraint on exterior cancellation of the field is removed, reproduction is possible over approximately 2N times the bandwidth. The use of higher order loudspeakers thus allows a significant reduction in the number of loudspeaker units, at the expense of increased complexity in each unit. For completeness, results are included for the generation of an exterior field with or without cancellation of the interior field.     

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。     

Modelling of sound propagation to three-dimensional urban courtyards using the extended fourier PSTD method

Noise from road traffic propagates to acoustically shielded areas as roadside courtyard by multiple reflection and diffraction paths in a complex three-dimensional (3D) environment. The computation of noise levels and assessment of candidate noise mitigation measures for these areas has up to now been based upon two-dimensional (2D) geometrical assumptions. Here, a recently developed efficient wave-based method, the extended Fourier pseudospectral time-domain (PSTD) method, is used to investigate the necessity of a 3D model. For frequencies up to 500 Hz and low traffic velocities of 30 km/h and 50 km/h, a road traffic noise configuration of an urban street canyon with or without cross streets and a closed roadside courtyard is compared to the 2D configuration as studied previously. It can be concluded that the contribution of distant sources is overpredicted by the 2D configuration. As noise mitigation measures, additional façade absorption, façade screens and roof screens have been studied. Results show that the 2D configuration underpredicts the effect of façade mitigation measures, by maximum 1.5 dB(A) for the absorption case and 4.4 dB(A) for the screens case. The effect of roof screens is overpredicted up to 1.7 dB(A). Given these deviations and the found deviations between the 3D configurations of street canyon with and without cross streets, the need for a 3D model can be concluded to be strongly configuration dependent. The 3D model is finally used to investigate the effect of a façade opening to the courtyard, which could lead to up to 10 dB(A) higher noise levels as compared to the noise propagating over the roof level and may prohibit the use of these courtyards as quiet areas. Absorption in the façade opening can significantly limit this negative effect.     

Localization of multiple sound sources with two microphones

This paper presents a two-microphone technique for localization of multiple sound sources. Its fundamental structure is adopted from a binaural signal-processing scheme employed in biological systems for the localization of sources using interaural time differences (ITD). The two input signals are transformed to the frequency domain and analyzed for coincidences along left/right-channel delay-line pairs. The coincidence information is enhanced by a nonlinear operation followed by a temporal integration. The azimuths of the sound sources are estimated by integrating the coincidence locations across the broadband of frequencies in speech signals (the "direct" method). Further improvement is achieved by using a novel "stencil" filter pattern recognition procedure. This includes coincidences due to phase delays of greater than 2pi, which are generally regarded as ambiguous information. It is demonstrated that the stencil method can greatly enhance localization of lateral sources over the direct method. Also discussed and analyzed are two limitations involved in both methods, namely missed and artifactual sound sources. Anechoic chamber tests as well as computer simulation experiments showed that the signal-processing system generally worked well in detecting the spatial azimuths of four or six simultaneously competing sound sources.     

Data completion method for the characterization of sound sources

This paper presents a different approach to solve the inverse acoustic problem. This problem is an "ill-posed" problem since the solution is very sensitive to measurement precision. A classical way to solve this problem consists in inversing a propagation operator which relates structure quantities (acoustic pressures or gradients) to near-field quantities (acoustic pressures or gradients). This can be achieved by using near-field acoustical holography (NAH) in separable coordinate systems. In order to overcome this limitation, the inverse boundary element method (IBEM) can be implemented to recover all acoustic quantities in a three-dimensional space and on an arbitrary three-dimensional source surface. In this paper, the data completion method (DCM) is developed: the acoustic gradients and pressures are known on a surface surrounding the source, but are unknown on its structure. The solution is given by the resolution of the Helmholtz formulation applied on the empty domain between the two boundaries made by the measurements quantities and the structure of the source. The conventional method applies directly the integral formulation for the empty domain. Another way of solving this Helmholtz formulation can be achieved by splitting it in two well-posed subproblems in a Steklov-Poincare?'s formulation. The data completion method allows one to solve the problem with acoustic perturbations due to sources on the exterior domain, or due to a confined domain, without altering the results.     

Identification of noise sources in factory’s sound field by using genetic algorithm

Noise control is important and essential in factory, where the noise level is restricted by the Occupational Safety and Health Act. Before noise abatement being performed, the identification work in searching for the location and sound power level (SWL) of noisy sound sources is absolutely prerequisite. Several researches on new techniques of single noise control have been well addressed and developed; however, the research work on sound identification for the existing multi-noise plant is hardly found sufficient. Under the circumstance of unrecognized noises, the noise control work will expectedly be extravagant and fruitless. Therefore, the numerical approach in distinguishing noises from a multi-noise plant becomes crucial and obligatory.In this paper, the novel technique of genetic algorithm (GA) in conjunction with the method of minimized variation square will be adopted and used in the following numerical optimization. In addition, various sound monitoring systems in detecting the noise condition within the plant area will also be introduced. Before noises identification, the accuracy of mathematical model has then been proved to be in good agreements comparing to the simulated data of SoundPlan, a commercialized simulation package in sound field. Moreover, three kinds of multi-noise plants have been fully discussed and acknowledged by GA optimization. The results reveal that the relevant locations and sound power levels (SWLs) of noises can be precisely recognized. This paper surely provides a rapid methodology in the noise identification work for a multi-noise plant.     

封闭空间声场重构的多层等效源法

对于封闭空间内的多途反射声,传统的等效声源法将其等效为距离边界一定距离的单层等效声源体进行声场重构,然而等效源与边界的距离选取依据不确定。因此,为获得等效声源配置的最优距离,在等效声源法(ESM)的基础上构建多层等效声源,提出一种适用于封闭环境声场重构的多层等效声源法(MESM),并依据等效声源的空间分布的稀疏性来获得等效声源强度信息。首先给出多层等效源法的理论依据,其次通过数值计算以及实验测试两种方式对比验证了所提方法。数值结果表明:MESM相比于ESM可在600 Hz以上频段获得低5~10 dB左右的重构误差,但是200 Hz以下的低频重构误差会增加5 dB左右。实验结果表明:MESM可比ESM获得更低的重构误差。文章最后基于数值计算研究了所提方法的主要影响因素。研究表明:虽然MESM会比ESM耗费2倍的计算时间,但在整体频率范围内,MESM可在ESM基础上提升600 Hz以上的重构性能。另外,等效声源的层数和层内数目的改变不会影响声场重构性能,而当传声器数目较多、阵列位置随机、空间边界的吸声系数不是很大时,MESM可获得比ESM更低重构误差,特别是600 Hz以上的中频段区间。