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。      

混响声场条件下水声互易传递函数的测量及应用

对不规则的混响声场情况的互易测量方法进行了理论研究,对相关测量误差提出了修正方法。通过测量结果的空间平均和混响环境对声能密度影响的评估,修正了声源体积速度的计算值,并在水中进行了试验验证。经修正后的互易测量结果与正向实测结果基本一致。表明混响声场不影响互易原理有效性的成立,但影响了声源体积速度的计算,进而影响了应用互易原理对传递函数的测量。提出的修正方法在不规则的混响声场情况下简单有效,结论为今后基于互易原理的工程应用提供了参考和依据。      

混响声场的有源控制

本文对室内声场的有源控制问题进行了探讨。发现直达声是指向性的,除非用同样指向性的反声源如对户外大型噪声源那样,是不能用有源控制的。但混响声场是驻波场,在屋角的次声源可以激发所有驻波(简正振动),因此可以作适当调制使其抑制整个混响声场。理论分析和初步实验证实此点,取得令人鼓舞的结果。对于须待解决的问题也进行了讨论。     

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。      

论室内声场

室内声场的经典理论中只有简正波(混响声)而缺乏直达声,这不符合实际。实际上声源具有二重作用,或说声源的作用有两项功能。一是声源按自由空间辐射声波,只要不遇到室的墙壁、边界,就不会改变。自由辐射达到任一边界后,就逐渐失去其球面波的特性,成为被反射为壁面上的一系列小波,时间、空间、方向都是无规分布,但是小波的来源都可推到声源上,这是第二功能。这些小波继续传播、反射,最后形成室内允许的驻波,这些就是简正波了。所以声源既是直达声波的声源,又是室内驻波的声源。据此可改正室内声波方程,解出室内声场,包括直达声波和简正波,并求得直达声和混响声的关系。声衰变过程的特性。声衰变中,稳态声中的直达声不会立刻停止,其后果值得注意。PACS数: 43.20,43.55     

线性和非线性单频混响声场空间起伏的初步研究

对于线性单频混响声场的统计特性的研究人们曾做了许多工作,建立了均方根声压和平均声能密度统计分布理论。本文在此基础上推导出声压级统计分布函数,并把处理方法推广到混响室内在强大声源激发下的非线性单频混响声场,求得各谐波声场和总声场的统计分布函数。本文实验采用数字测量技术,对混响声场进行了大量的测量,得到的实验结果与理论符合良好。     

一种简便的表达声波波动特性的演示方法

1引言 普通扬声器发出的声波不是强指向性的；当声源发出的直达声波到达接收传声器后，由房间内对应各个反射面向声源产生的声波也陆续到达，造成严重的干扰．由于室内反射面状况较复杂，对时域中具有一定持续时间的信号，其直达声与反射声完全混叠，无法将各个像声源所对应的声波一一辨别出，因此直接采用扬声器发出的窄带或单频信号作为波源的实验方法就无法应用于普通的课堂教学．     

室内稳态声场

经典简正波理论把一声源在室内产生的声场表达为一系列简正波之和．实验测量结果常不能用这个理论解释，因此猜想简正波公式不是室内波动方程的全解，而只是其混响声部分，还应有一代表直达声的部分．在本文中从简正波理论推导的严格审查中，从室内声场的理论分析和实验测量中，证实了这个猜想，证明室内声场应包括直达声和固边界影响而形成的驻波（简正波），并求得相应的表达式．     

厅堂音质中的响度评价

厅堂音质评价的各项指标中，响度是最重要和最基本的内容之一。但由于长期来缺乏合适的参量，因此迄今无法在完工后的厅堂中去测量这项指标，当然也难以在设计阶段对此参量进行估算．不少人常把仅仅适用于稳态声源和混响场的声场估算法（即以直达声加上混响声）作为厅堂内各处总声强的评价，无论从音质设计和现场测量来看，显然很不合适。近年Ｌｅｈｍａｎｎ提出以声强指数Ｇ（Ｓｔａｒｋｅｍａｓｓ）（ｄＢ）作为评价参量是一个好的建议。但根据我们的研究结果来看，鉴于早期反射声对响度起主导作用，因此厅堂内各处的声强指数应取５０ｍｓ（语言）和８０ｍｓ（音乐）的早期反射声积分值更符合实际，。以代替ｔ从０积分到∞的评价方法。因此Ｇ（５０ｍｓ）和Ｇ（８０ｍｓ）将分别用于评价厅堂内对语言和音乐的响度评价参量。     

Suppression of reverberation distortions of a receiver signal using the water tank transfer function

The paper considers a method for suppressing the reverberant distortions of an underwater sound receiver signal during receiver calibration in a laboratory water tank. The method is based on using the water tank transfer function, which is a complex frequency-dependent coefficient that establishes, for the point of signal reception, the relation between the sound pressures in the reverberant sound field of the water tank and in the free sound field. The procedure for experimentally obtaining the water tank transfer function is considered. Examples of suppressing reverberant distortions during noise and pulse sound reception are presented.     

Application and measurement of underwater acoustic reciprocity transfer functions in reverberant sound field

The reciprocity measurement theory in anomalous reverberant sound fields was investigated.An improved method Was proposed due to the interrelated errors.The source volume velocity Was corrected by spatial average of measurement results and evaluation of the reverberant sound field influence on acoustic energy density.The result was validated in underwater experiment,corrected reciprocity measurement results were almost the same as direct measurement results.It indicates that reverberant sound field does not affect the validitv of the principle,but influences the obtainment of source volume velocity,then influences the measurement of transfer functions with the principle.The proposed method is simple and effective in anomalous reverberant sound fields.The study mav be valuable for the applications which are based on the principle.     

Subjective and objective verifications of the inverse functions of binaural room impulse responses

The binaural room impulse responses (BRIRs) can be applied to 3-D sound field reconstruction, virtual reality, noise control, et al. Because the BRIRs are non-minimum phase functions, it is difficult to find the exact inverse functions of the BRIRs, especially when there are two or more sources in a reverberant space. In this work, a method was proposed to find the inverse functions of BRIRs with two sound sources in a reverberant space. The concept of time delays and the method of weighted least squares were used to find the causal, however, approximate inverse functions. The accuracy of the inverse functions was first evaluated objectively by a dummy head system. The result shows that the distortion due to crosstalk and room reverberation can be improved by 16∼18 dB. The inverse functions were also verified subjectively by 20 students. The result of subjective evaluation also shows that the inverse functions can be used successfully to reduce the crosstalk effect and the room reverberation.     

一种预测扩散声场在非规则形状刚性壁面附近干涉图样的方法

扩散声场会在反射边界附近形成干涉图样,研究方法包括平面波模型、简正模态分析、渐进模态分析等,但仅适用于尺度远大于声波波长的矩形声腔。提出一种预测扩散声场在非规则刚性壁面结构附近形成的干涉图样的数值方法,表明结构附近“受挡”声压的互谱矩阵取决于:(1)假定该结构在自由空间中振动辐射声音时其表面法向振速到表面及场点声压的边界元系数矩阵;(2)假定结构置于自由空间中且表面刚性时,点声源辐射声波入射到结构表面上产生的散射声场的边界元系数矩阵;(3)扩散声场均方声压。仿真表明,该途径预测的干涉图样与理论值完全吻合。该预测方法还可用于混响环境下声源附近直达声压均方值的空间分布估计,为混响环境下设备的声源定位提供帮助。      

Performance of an adaptive beamforming noise reduction scheme for hearing aid applications. I. Prediction of the signal-to-noise-ratio improvement

Adaptive beamformers have been proposed as noise reduction schemes for conventional hearing aids and cochlear implants. A method to predict the amount of noise reduction that can be achieved by a two-microphone adaptive beamformer is presented. The prediction is based on a model of the acoustic environment in which the presence of one acoustic target-signal source and one acoustic noise source in a reverberant enclosure is assumed. The acoustic field is sampled using two omnidirectional microphones mounted close to the ears of a user. The model takes eleven different parameters into account, including reverberation time and size of the room, directionality of the acoustic sources, and design parameters of the beamformer itself, including length of the adaptive filter and delay in the target signal path. An approximation to predict the achievable signal-to-noise improvement based on the model is presented. Potential applications as well as limitations of the proposed prediction method are discussed and a FORTRAN subroutine to predict the achievable signal-to-noise improvement is provided. Experimental verification of the predictions is provided in a companion paper [J. Acoust. Soc. Am. 109, 1134 (2001)].     

Simulating the response of multiple microphones to a single acoustic source in a reverberant room

A method is presented for simulating the impulse response between an acoustic source and multiple microphones in a reverberant room. The method is similar to the image method described by Allen and Berkley [J. Acoust. Soc. Am. 65, 943-950 (1979)] but includes modifications to simulate received echo arrival time accurately. The essential modification is to represent each received echo as a low-pass-filtered impulse at the correct arrival time. Using this "low-pass impulse" method, reverberant rooms can be simulated with sufficient accuracy to investigate multiple-microphone systems that are sensitive to interchannel phase.     

Corrected Statistical Energy Analysis Model in a Non-Reverberant Acoustic Space

Statistical Energy Analysis (SEA) is a well-known method to analyze the flow of acoustic and vibration energy in a complex structure. This study investigates the application of the corrected SEA model in a non-reverberant acoustic space where the direct field component from the sound source dominates the total sound field rather than a diffuse field in a reverberant space which the classical SEA model assumption is based on. A corrected SEA model is proposed where the direct field component in the energy is removed and the power injected in the subsystem considers only the remaining power after the loss at first reflection. Measurement was conducted in a box divided into two rooms separated by a partition with an opening where the condition of reverberant and non-reverberant can conveniently be controlled. In the case of a non-reverberant space where acoustic material was installed inside the wall of the experimental box, the signals are corrected by eliminating the direct field component in the measured impulse response. Using the corrected SEA model, comparison of the coupling loss factor (CLF) and damping loss factor (DLF) with the theory shows good agreement.     

Effect of source spectrum on sound localization in an everyday reverberant room

Two experiments explored how frequency content impacts sound localization for sounds containing reverberant energy. Virtual sound sources from thirteen lateral angles and four distances were simulated in the frontal horizontal plane using binaural room impulse responses measured in an everyday office. Experiment 1 compared localization judgments for one-octave-wide noise centered at either 750 Hz (low) or 6000 Hz (high). For both band-limited noises, perceived lateral angle varied monotonically with source angle. For frontal sources, perceived locations were similar for low- and high-frequency noise; however, for lateral sources, localization was less accurate for low-frequency noise than for high-frequency noise. With increasing source distance, judgments of both noises became more biased toward the median plane, an effect that was greater for low-frequency noise than for high-frequency noise. In Experiment 2, simultaneous presentation of low- and high-frequency noises yielded performance that was less accurate than that for high-frequency noise, but equal to or better than for low-frequency noise. Results suggest that listeners perceptually weight low-frequency information heavily, even in reverberant conditions where high-frequency stimuli are localized more accurately. These findings show that listeners do not always optimally adjust how localization cues are integrated over frequency in reverberant settings.