阵列结构下的低频信号合成方法研究

基于高频天线产生低频电磁波信号,实现多波段信号对目标的照射,不仅有可能减小低频天线尺寸,而且可能成为提高雷达目标探测性能的一种途径.本文将多普勒效应与阵列天线结构相结合,基于对阵列中各辐射单元的信号时序、相位和间距等参数的控制,提出了一种在目标区产生低频信号的方法.本文给出了阵列参数的选择原则,介绍了目标位于阵列方向和45°角扫描时的低频信号合成情况,对存在辐射单元间距误差、相位误差、目标偏离预定位置以及等间隔稀疏条件下的合成信号性能进行了分析,并采用峰值旁瓣比和积分旁瓣比来评价合成信号的性能.将频率1 GHz载波信号合成为频率400 MHz信号的仿真分析结果,表明了本文方法的有效性.     

炉内管阵列中声源辐射特性数值研究

炉内换热器结构类似于声子晶体,基于声子晶体理论,采用数值计算与软件模拟得到管阵列中声源不同方向的声传播特性,并将两种方法所得结果进行对比分析,研究声源在管阵列中不同位置处的辐射特性.结果表明：声指向性特性与声源通过管排数量,声源频率是否处在声波禁带有关.     

识别运动声源随机阵列生成法研究

针对随机阵列结构设计方面欠缺高效的生成方法这一问题,在保证阵列对声源识别精度的前提下,提出一种在极坐标下用于识别运动声源的随机阵列生成方法。根据声阵列识别运动声源的原理,推导了非等间距基本同轴圆环的半径,再通过降列分区、极坐标下条件筛选和模拟评价三个步骤,生成适用于识别运动声源的目标随机阵列,最后进行数值模拟和运动汽车噪声源识别实验进行验证。研究结果表明,用该方法能够高效地生成目标随机阵列,与常用规则阵列相比具有更良好的声源识别特性,并且具有准确的运动声源识别性能。      

识别运动声源随机阵列生成法研究

针对随机阵列结构设计方面欠缺高效的生成方法这一问题,在保证阵列对声源识别精度的前提下,提出一种在极坐标下用于识别运动声源的随机阵列生成方法。根据声阵列识别运动声源的原理,推导了非等间距基本同轴圆环的半径,再通过降列分区、极坐标下条件筛选和模拟评价三个步骤,生成适用于识别运动声源的目标随机阵列,最后进行数值模拟和运动汽车噪声源识别实验进行验证。研究结果表明,用该方法能够高效地生成目标随机阵列,与常用规则阵列相比具有更良好的声源识别特性,并且具有准确的运动声源识别性能。     

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。     

面向目标的定向声辐射技术及其应用研究

面向目标的定向声辐射是指利用声源特性和阵列信号处理技术,将声波传送至目标方向或区域的声场控制方法。本文介绍了定向声辐射技术的原理和进展,重点阐述了普通扬声器阵列和参量阵扬声器这两种声源形式的理论、算法和应用研究。其中,针对扬声器阵列,主要分析了基于声能量对比度控制的声聚焦技术;针对参量阵扬声器,综述了利用非线性效应产生高指向性可听声的计算方法、信号预处理和应用等进展。面向目标的定向声辐射技术的研究对实现智能声场控制有着重要意义。     

环形稀疏声源阵列声涡旋的优化与操控

提出和优化了环形稀疏声源阵列涡旋声场的形成方法,进行了涡旋声压,相位和振动速度的理论推导和特性分析。通过改变声源数、声源频率和观测距离等参数,对声涡旋场的模拟结果证明,声涡旋半径主要由信号波长决定,声源频率越高,声涡旋半径越小,能量越集中;涡旋声压随着声源数的增加而线性提高,随着传播距离的增加而减小;声涡旋环形相位分布的线性度随着声源数、声源波长和传播距离的增加而提高。建立了3/4/6/8声源实验系统,径向声压和相位分布以及环形相位分布的测量结果和模拟结果具有较好的一致性,同时悬吊泡沫盘的旋转证明了声涡旋角动量的传输。本研究中声涡旋场的特性分析和参数优化为粒子操控及其在生物医学中的应用提供了依据。      

LED阵列模组化中的照度均匀性问题

为使LED阵列在目标面上产生均匀的照度分布,构建一个均匀度评价函数.通过优化阵列中各LED的位置,使得评价函数产生最小值,此时阵列在照明面上的均匀度最高.采用模拟退火算法对LED集成圆形阵列、4×4和4×5的矩形阵列进行优化,优化后的阵列在目标面上均产生了均匀照度分布,对应的均匀度分别为97%、96%和95%.对LED阵列中影响目标面均匀照度的参量进行研究,结果表明圆形阵列的最佳半径与LED的颗数无关,阵列面和目标面之间的距离与阵列的最佳半径呈线性关系;矩形阵列中,相邻LED之间的最佳距离与阵列面和目标面之间的距离呈线性关系.该方法与现有解析方法所得结果一致,且算法简单便捷,可用于计算机自动优化设计.     

基于相同参数抛物镜面旋转阵列的太阳能槽式聚光器聚焦特性研究

提出了一种基于相同抛物镜面单元旋转阵列的低成本新型槽式聚光器。采用光线跟踪方法建立了该聚光器的光学分析模型,并详细研究了镜面焦距（f）、镜面宽度、平面接收器位置、旋转阵列半径（R₁）和旋转阵列数量（N）等关键参数对其聚光性能的影响规律,为其设计与应用提供了重要依据。研究结果表明：新型槽式聚光器能很好地汇聚太阳光能,并且它不同于传统抛物槽式聚光器将平行光汇聚于某点,而是有所“分散”地进行汇聚,具备实现平面接收器上均匀聚焦能流分布的潜力;聚焦光斑的合理接收位置会随旋转阵列半径的改变而变化,通常接收器位于旋转阵列半径的一半位置是合适的,但要获得最小聚焦光斑则需将其下移150～200 mm;聚焦光斑宽度随镜面编号呈指数增大关系,特别是当镜面宽度较大且旋转阵列半径较小时;旋转阵列半径越大或镜面宽度越小,聚焦能流分布越集中且峰值聚光比越大（算例中已达到50）,此时聚焦能流基本呈高斯分布特征。此外,采用较小旋转阵列半径可降低接收器的安装高度并改善能流均匀性。在算例中,当f=8000 mm、R₁=4000 mm、N=5且接收器位于1830 mm位置时,聚焦光...     

用正四棱锥形阵对声目标定位研究

文章结合声阵列在智能雷系统中的应用，提出了用立体五元声阵列对声目标进行定位，将智能雷中用于预警的传感器也参与到定位阵列中来，这样就在不增加传感器的条件下实现了市体阵列对声目标的定位。文章推导了定位公式和误差公式，并对结果进行仿真分析，与平面阵相比取得了较好的效果。不但提高了精度还降低了声阵列的盲区。     

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.     

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。     

Dynamic measurement of room impulse responses using a moving microphone

A technique for the recording of large sets of room impulse responses or head-related transfer functions is presented. The technique uses a microphone moving with constant speed. Given a setup (e.g., length of the room impulse response), a careful choice of the recording parameters (excitation signal, speed of movement) leads to the reconstruction of all impulse responses along the trajectory. In the case of a moving microphone along a circle, the maximal angular speed is given as a function of the length of the impulse response, its maximal temporal frequency, the speed of sound propagation, and the radius of the circle. As a result of the presented algorithm, head-related transfer functions sampled at 44.1 kHz can be measured at all angular positions along the horizontal plane in less than 1 s. The presented theory is compared with a real system implementation using a precision moving microphone holder. The practical setup is discussed together with its limitations.     

Research on intensity measurement in room impulse field

A new system of sound intensity measurement for impulse field in the room was proposed. This measurement system consists of a repeatable inspiriting sound source and a microphone fixed on a slowly rotating platform, which is equivalent to a circle microphone array composed of many perfectly matched microphones. The test principle was presented and typical application was described. Based upon this system the sound intensity measurement for impulse field in the room was realized. Therefore, not only time but also spatial information of room impulse response can be obtained.     

多阶声模态分解的改进阻抗管法测量材料的高频吸声系数

为了测量高频材料吸声系数,采用声模态分解的方法,基于阻抗管构建测试设备,在阻抗管内测量超过平面波截止频率的的高频吸声系数.测量过程中,通过在阻抗管的周向和轴向分别布置传声器阵列,分离管道内前3阶周向声模态以及各阶声模态的轴向传播入射波和反射波,从而得到最高频率达10000 Hz的材料吸声系数,并通过对比常规阻抗管测试方...     

Field measurement of airborne sound insulation between rooms with non-diffuse sound fields at low frequencies

Procedures for the field measurement of airborne sound insulation between rooms with diffuse fields are described in International Standard ISO 140-4. However, many dwellings contain rooms with volumes less than 50 m³, where low frequency measurements are less reliable; hence there is a need for a measurement procedure to improve the reliability of field measurements in rooms with non-diffuse fields. Procedures are proposed for sound pressure level and reverberation time measurements for the 50, 63 and 80 Hz third octave bands. The sound pressure level measurement combines corner microphone positions with positions in the central region of each room. This provides a good estimate of the room average sound pressure level with significantly improved repeatability.     

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。      

Oblique incidence sound absorption of parallel arrangement of multiple micro-perforated panel absorbers in a periodic pattern

The sound absorption performance of a micro-perforated panel (MPP) absorber array at oblique incidence and in diffuse field is investigated both numerically and experimentally. The basic module of the MPP absorber array consists of four parallel-arranged MPP absorbers with different cavity depths, and the whole MPP absorber array is created by arranging the basic modules in a periodically repeating pattern. Results show that the influence of incidence angle mainly lies in two aspects. First, the parallel absorption mechanism breaks down at lower frequencies at oblique incidence than at normal incidence due to the non-compactness of the resonating MPP absorber, which becomes non-compact if the time delay of incident wave across it is comparable to or larger than π/2. Second, the equivalent acoustic impedance of the MPP varies with respect to incidence angle which in turn changes the sound absorption performance of the MPP absorber array. Influence of the azimuthal angle is insignificant. Because of mutual influence among the member MPP absorbers, the normal incidence sound absorption of the MPP absorber array can be noticeably different from that of the basic module tested in impedance tube. The measured sound absorption coefficients of a prototype specimen in reverberation room compare well with the numerical predictions. The extra sound absorption due to diffraction of sound at the free edges of test specimen is the most efficient around 500 Hz.     

An indirect hybrid sound transmission loss controller

The control of sound transmission through panels is an important noise control problem in the aerospace, aeronautical, and automotive industries. The trend towards using lightweight composite materials that have lower sound insulation performance is a negative factor regarding low frequency transmission loss. Double-panel partitions with the gap filled with sound absorption materials are often employed to improve the sound insulation performance with reduced added weight penalty. However, in the low frequency range, the strong coupling between the panels through the air cavity and mechanical paths may greatly reduce the sound transmission performance, making it even lower than the performance of a single panel in some frequency ranges. In this work, an experimental investigation of a new kind of hybrid (active/passive) acoustic actuator is presented. The idea consists of replacing the acoustic absorption material by a hybrid actuator aiming at improving the transmission loss at low frequencies without altering the passive attenuation. A prototype of the system is tested in a plane wave acoustic tube setup. Different kinds of SISO feedforward control implementations were used to attenuate the sound power transmitted through the hybrid active–passive panel using an error microphone or a particle velocity sensor placed downstream with respect to the sample panel. Measurement results of the transmission loss with active and hybrid attenuation are presented and discussed.     

经颅超声聚焦方法和声场特性研究

基于高分辨的CT数据建立了非均匀颅骨仿真模型,该模型引入了颅骨的声衰减系数,深入研究和分析了声波时间反转法和超声相控阵法在颅脑中的聚焦方法及效果。颅骨具有较强的声波衰减特性,使用时间反转聚焦时需要进行幅度补偿,对于0.7MHz的频率信号,幅度补偿后的时间反转聚焦声场主瓣宽度窄、旁瓣低,焦点处声场比无幅度补偿的时间反转法提高8.86dB,比超声相控阵聚焦法提高7.89dB,具有很好的空间聚焦精度和聚焦效率。研究了颅骨衰减系数、声场焦点位置、声波频率、换能器阵列位置和方位等参数对聚焦声场的影响,结果表明,幅度补偿时间反转法比相控阵法具有更低的旁瓣,且高频时的聚焦效果比相控阵好,相控阵聚焦对换能器阵列的位置和方位比较敏感,而时间反转经颅超声聚焦对声传播路径和入射角具有更高的鲁棒性。