双扬声器近场声源重放实验研究

该文针对电子器件散热用的一款变速轴流风扇的气动噪声及其降噪方法进行实验研究。首先利用风扇旋转轴等高平面内圆周分布的传感器阵列测量风扇不同转速下远场噪声分布,总声压级与转速的对数关系验证散热风扇主要气动噪声属于偶极源噪声,频谱分析显示离散单音噪声为主要噪声影响因素。基于管道声学理论的管道模态截止方法,研究进出风口安装圆形短管对风扇气动噪声的影响,实验结果显示不同位置、不同长度的短管对风扇远场噪声影响不同。额定转速下,在进风口安装2 cm管道可以使远场1 m处平均总声压级下降4.1 dB(A),降噪效果显著。模态测量结果显示,此种情况下对应离散单音处的风扇主要模态幅值大大降低,风扇离散单音噪声降低从而噪声总声压级大幅减小。该方法为散热风扇降噪提供了一种新的途径。     

重放自由场虚拟源距离信息的动态双耳Ambisonics方法

双耳重放的目标之一是在耳机重放中产生不同方向和距离的虚拟源感知。本文研究了动态双耳Ambisonics重放自由场虚拟源方向和距离信息的简化信号处理方法。该信号处理方法包括两步:第1步是基于目标声场的球谐函数分解,合成采用扬声器的近场Ambisonics重放中逐级重构目标声场的信号;第2步是采用虚拟扬声器重放的方法,用动态头相关函数滤波处理将Ambisonics的扬声器重放信号转换为双耳重放信号并用耳机重放。进一步研究了动态双耳Ambisonics的阶数对定位效果的影响,为简化信号处理提供依据。对重放产生的双耳声压分析表明,5阶动态双耳Ambisonics重放足以提供听觉方向定位和距离感知的重要信息。同时心理声学的实验结果表明,结合声源距离相关的响度因素,5阶动态双耳Ambisonics重放可产生不同方向和1.0 m以下不同近场距离的自由场虚拟源的听觉感知。本文的方法仅需要固定距离的48个均匀空间方向的远场非个性化HRTF处理,实现了信号处理的简化。      

双声源位置和强度识别的三维声强方法

目前在远场识别声源空间位置和强度缺乏行之有效的方法。针对此问题,提出采用四传声器进行三维声强测量,从而构建出声强、声源坐标和声功率的非车线性方程组,求解方程得出声源空间坐标和强度的方法。以3个三维声强探头对两个同频率单极子声源的识别为例,分别利用数值仿真和半消声室内的实验进行方法验证,并对声源的识别空间分辨率做了测试,得出角度识别最大误差为3.83°,为真实值的8.5%,距离识别最大误差0.1 m,为真实距离的10%。结果表明采用该方法空间坐标和声功率识别均具有很高的准确度,双声源的空间位置分辨力也优于远场声全息方法。      

Assessing auditory distance perception using virtual acoustics

In most naturally occurring situations, multiple acoustic properties of the sound reaching a listener's ears change as sound source distance changes. Because many of these acoustic properties, or cues, can be confounded with variation in the acoustic properties of the source and the environment, the perceptual processes subserving distance localization likely combine and weight multiple cues in order to produce stable estimates of sound source distance. Here, this cue-weighting process is examined psychophysically, using a method of virtual acoustics that allows precise measurement and control of the acoustic cues thought to be salient for distance perception in a representative large-room environment. Though listeners' judgments of sound source distance are found to consistently and exponentially underestimate true distance, the perceptual weight assigned to two primary distance cues (intensity and direct-to-reverberant energy ratio) varies substantially as a function of both sound source type (noise and speech) and angular position (0 degrees and 90 degrees relative to the median plane). These results suggest that the cue-weighting process is flexible, and able to adapt to individual distance cues that vary as a result of source properties and environmental conditions.     

Mathematically trivial control of sound using a parametric beam focusing source

By exploiting a case regarded as trivial, this paper presents global active noise control using a parametric beam focusing source (PBFS). As with a dipole model, one is used for a primary sound source and the other for a control sound source, the control effect for minimizing a total acoustic power depends on the distance between the two. When the distance becomes zero, the total acoustic power becomes null, hence nothing less than a trivial case. Because of the constraints in practice, there exist difficulties in placing a control source close enough to a primary source. However, by projecting a sound beam of a parametric array loudspeaker onto the target sound source (primary source), a virtual sound source may be created on the target sound source, thereby enabling the collocation of the sources. In order to further ensure feasibility of the trivial case, a PBFS is then introduced in an effort to meet the size of the two sources. Reflected sound wave of the PBFS, which is tantamount to the virtual sound source output, aims to suppress the primary sound. Finally, a numerical analysis as well as an experiment is conducted, verifying the validity of the proposed methodology.     

Near-field characteristics of the parametric loudspeaker using ultrasonic transducers

A parametric speaker is a device for generating and focusing highly directional sound beams. It is essentially a by-product that comes with the nonlinearity of ultrasound. It is noteworthy that this directional beam was controlled and utilized mostly for far-field applications in the past. We empirically study the directivity and attenuation characteristics of the parametric loudspeaker in the near-field where we desire to use it. Physical parameters for experiments are imported from a theoretical model based on the far-field approximation. The findings are that increases in aperture size and modulation frequency cause higher directivity, but have more than twice the beamwidth of the far-field approximation. The attenuation also does not obey the inverse-square law which describes far-field spreading from acoustic sources. The results conclusively explain a series of formation and attenuation of the virtual sound sources and define limitations of use in the near-field.     

The dominant role of low-frequency interaural time differences in sound localization.

Two experiments are described in which listeners judge the apparent directions of virtual sound sources-headphone-presented sounds that are processed in order to simulate free-field sounds. Previous results suggest that when the cues to sound direction are preserved by the simulation, the apparent directions of virtual sources are nearly the same as the apparent directions of real free-field sources. In the experiments reported here, the interaural phase relations in the processing algorithms are manipulated in order to produce stimuli in which the interaural time difference cues signal one direction and interaural intensity and pinna cues signal another direction. The apparent directions of these conflicting cue stimuli almost always follow the interaural time cue, as long as the wideband stimuli include low frequencies. With low frequencies removed from the stimuli, the dominance of interaural time difference disappears, and apparent direction is determined primarily by interaural intensity difference and pinna cues.     

浅海环境中矢量水听器角度谱特性

为探明远场舰船噪声和近场平台噪声声能流相互作用的机制对矢量水听器角度谱特性的影响,基于简正波矢量场理论,构建舰船平台噪声声场模型,获得与海上实验结果相符的声能流角度谱特性,声能流方向均随频率变化。对远场舰船噪声声能流和近场平台噪声声能流相互作用进行仿真,两者声能流强度的此消彼长使角度谱出现4种条纹,与海上实验获得的角度谱特性一致,导致目标方位估计出现误差,且不同频带上的结果不一致。说明两个声能流相互作用时,影响合成能流角度谱特性的主要因素是两者的声压级之差,为浅海中同时存在多个声源时的目标方位估计提供参考。      

近场照度均匀的LED阵列设计

通过分析多颗LED组成的各种阵列在近场平面上的照度分布,来设计可实现近场照度均匀的LED排列方式和距离。基于LED的光强分布,构建单颗LED的光强分布模型（并非朗伯分布）,给出了方程和等式用于计算不同排列方式下的LED分布距离和封装密度,以实现近场平面上的照度均匀。     

内部体积源作用下的圆柱壳内外声场特性

圆柱壳内各型体积源辐射噪声特性研究是声场建模和声场预报的前提.为了研究具有指向性的大尺度体积源特性对水下航行器结构内外声场的影响,本文结合薄壳理论、等效源和柱腔Green函数构造了体积源激励下的壳体振动耦合方程,研究了体积源表面声散射作用和指向性强弱对圆柱壳内外声场的影响.数值计算结果表明,体积源构造的准确性与其等效源位置有关,等效源配置在体积源几何中心与其结构表面之间0.4—0.6时,可以提高声场计算结果的准确性;大尺度体积源表面的声散射作用会导致壳体内部声场结构发生改变,内声场声腔共振峰发生偏移,并且在部分频段引起较强的声透射现象;此外,体积源指向性变化对壳体内外声场强弱影响较小,其显著作用表现在改变了外辐射声场的远场指向性.该研究结果对噪声预报和控制有一定的参考价值.     

近场光学虚拟光探针的数值分析

虚拟光探针是基于近场光学隐失场干涉原理产生的一种非实体探针，可以应用于近场光学超高密度存储、纳米光刻、近场光学成像、光谱探测、纳米样品的近场光学操作等领域。本研究采用三维时间域有限差分（FDTD）方法对近场光学虚拟光探针的光场分布特性进行了数值模拟计算和比较，分析了孔的形状、大小及偏振态等因素对虚拟光探针光场分布的影响，研究结果表明虚拟光探针的通光效率较普通的纳米孔径光纤探针提高10^2-10^4倍；其光场分布的中间峰的半峰全宽（即虚拟光探针的尺寸）在一定距离范围内基本保持不变，从而可以解决近场光学系统中纳米间距控制的难题，避免光学头与介质的磁撞。优化虚拟光探针的设计参量能有效的抑制虚拟光探针中的旁瓣。文章还给出了应用虚拟探针实现高密度光存储的原理方案。     

基于格林函数法的奇型Mathieu-Gaussian光束

根据光束传播的独立性和叠加性原理,引入了一组能够产生第一类(2n+2阶)奇型Mathieu-Gaussian光束的虚光源点.利用虚源点技术和格林函数法,计算得到第一类奇型Mathieu-Gaussian光束的严格解析积分表达式.利用该表达式得到了轴上光场分布的积分解析解.以三阶非旁轴修正为例,得到了第一类奇型Mathieu-Gaussian光束保留到三阶非旁轴修正项的轴上光场分布精确解.     

Line source and site characterizations for defining the sound transmission loss of building facades

An analytical model is presented for defining the sound transmission loss of building facades exposed to noise from line sources. The model describes the non-diffuse sound field incident upon the facade in terms of both source and site parameters. The effects of facade orientation relative to the line source and the sound propagation with distance are introduced as a single term in the definition of the facade sound transmission loss. This term defines a mean angle of incidence for the exterior sound field that is equivalent to a point source location relative to a point on the facade. Numerical results are presented illustrating the magnitudes of these effects and it is shown that alternative methods for conducting field measurements of building facade sound transmission loss may be related by using this model.     

Design and validation on a multiple sound source fast-measurement system of near-field head-related transfer functions

Near-field head-related transfer functions(HRTFs) are essential to scientific researches of binaural hearing and practical applications of virtual auditory display. High efficiency, accuracy and repeatability are required in a near-field HRTF measurement. Hence,there is no reference which intents on solving the measuring difficulties of near-field HRTF for human subjects. In present work, an efficient near-field HRTF measurement system based on computer control is designed and implemented, and a fast calibration method for the system is proposed to first solve the measurement of near-field HRTF for human subjects. The efficiency of measurement is enhanced by a comprehensive design on the acoustic, electronic and mechanical parts of the system. And the accuracy and repeatability of the measurement are greatly improved by carefully calibrating the positions of sound source, subject and binaural microphones.This system is suitable for near-field HRTF measurement at various source distances within 1.0 m, for both human subject and artificial head. The time costs of HRTF measurement at a single sound source distance and full directions has been reduced to less than 20 minutes. The measurement results indicate that the accuracy of the system satisfies the actual requirements.The system is applicable to scientific research and can be used to establish an individualized near-field HRTF database for human subjects.     

Wave field synthesis of moving virtual sound sources with complex radiation properties

An approach to the synthesis of moving virtual sound sources with complex radiation properties in wave field synthesis is presented. The approach exploits the fact that any stationary sound source of finite spatial extent radiates spherical waves at sufficient distance. The angular dependency of the radiation properties of the source under consideration is reflected by the amplitude and phase distribution on the spherical wave fronts. The sound field emitted by a uniformly moving monopole source is derived and the far-field radiation properties of the complex virtual source under consideration are incorporated in order to derive a closed-form expression for the loudspeaker driving signal. The results are illustrated via numerical simulations of the synthesis of the sound field of a sample moving complex virtual source.     

Effect of Sound Source Scattering on Measurement of Near-Field Head-Related Transfer Functions

A simple spherical head and pulsating spherical sound source model are proposed to investigate the effect of multiple scattering between the head and the sound source on near-field head-related transfer function (HRTF) measurement. Multipole expansion method is used to calculate HRTFs of the model, then the relationships among the magnitude error of HRTF with frequency, source direction, source size, and the distance between the head centre and the sound source are analysed. The results show that to ensure the magnitude error of HRTF within 1.0dB up to 20kHz, for source distance not less than 0.15m or 0.20m, the radius of the sound source should not exceed 0.03m or 0.05m, respectively. The conclusion suggests an appropriate size of sound source in near-field HRTF measurement.     

人工头近场头相关传输函数及其特性

采用球形正十二面体声源及其空间定位系统,测量并建立了KEMAR人工头的近场头相关传输函数(HRTF)数据库。基于数据库分析了近场HRTF在频域和时域随声源距离变化的规律;讨论了用近场HRTF算得的双耳声级差(TLD)和双耳时间差(ITD)所包含的声源距离定位信息。结果表明,测量系统和所得数据具有较好的重复性和准确性,保留了1 kHz以下的低频定位信息。并且,近场HRTF幅度谱和ILD随声源距离的变化明显;用相关法算得2 kHz以下频段的ITD随声源距离略有变化。本文数据库及其分析结果将为声源距离定位的应用提供基础。      

近场头相关传输函数的多声源快速测量系统设计与验证

近场头相关传输函数(HRTF)是双耳听觉科学研究和虚拟听觉重放应用的重要基础数据。近场HRTF测量系统要求具有高的测量效率、精度和重复性,以至于目前未见文献解决真人受试者的近场HRTF测量困难。本文研究并实现了一种计算机控制的近场HRTF的高效测量系统,并提出系统的快速校准方法,首先实现了真人受试者的近场HRTF测量。通过声学、机械与电子硬件和软件的综合设计,提高了测量效率。通过准确校准声源、受试者和双耳传声器的位置,提高了测量精确度和重复性。系统可用于1.0 m范围内不同声源距离的真人受试者以及人工头的近场HRTF测量,单个声源距离的全空间近场HRTF测量时间减少至20 min以内。测量结果表明,系统测量精度满足实际需求,可用于科学研究和个性化近场HRTF测量及数据库建立。      

A study of sound intensity control for active noise barriers

Active noise control systems have been applied to increase the insertion loss of noise barriers where the squared sound pressure or the total acoustic energy density is used as the cost function in previous works. The absolute value of the mean active sound intensity is chosen as the cost function to obtain extra sound insertion loss in the dark area of a hybrid active noise barrier system in this note. The strategy of minimizing the near-field sound intensity at discrete locations along the edge of the passive barrier is shown to be able to provide better far-field noise reduction than that of minimizing the squared sound pressure control. Both numerical simulations and off-line experiments are carried out with a three-channel demonstration system, where the locations of the secondary sources and the error sensors are optimized and comparisons are made between the extra sound pressure attenuation of the sound intensity control and that of the squared sound pressure control.     

Implementation issues of the nearfield equivalent source imaging microphone array

This paper revisits a nearfield microphone array technique termed nearfield equivalent source imaging (NESI) proposed previously. In particular, various issues concerning the implementation of the NESI algorithm are examined. The NESI can be implemented in both the time domain and the frequency domain. Acoustical variables including sound pressure, particle velocity, active intensity and sound power are calculated by using multichannel inverse filters. Issues concerning sensor deployment are also investigated for the nearfield array. The uniform array outperformed a random array previously optimized for far-field imaging, which contradicts the conventional wisdom in far-field arrays. For applications in which only a patch array with scarce sensors is available, a virtual microphone approach is employed to ameliorate edge effects using extrapolation and to improve imaging resolution using interpolation. To enhance the processing efficiency of the time-domain NESI, an eigensystem realization algorithm (ERA) is developed. Several filtering methods are compared in terms of computational complexity. Significant saving on computations can be achieved using ERA and the frequency-domain NESI, as compared to the traditional method. The NESI technique was also experimentally validated using practical sources including a 125 cc scooter and a wooden box model with a loudspeaker fitted inside. The NESI technique proved effective in identifying broadband and non-stationary sources produced by the sources.