A technique for simulating the amplifier-to-eardrum transfer function of an in situ hearing aid

There are numerous articles wherein mathematical models of various parts of an in situ hearing aid have been reported. Such parts include, for example, the microphone, receiver, cylindrical tubes carrying sound to the eardrum and out through the earmold vent, and the external path from the vent back to the microphone. This article extends these earlier works to include the hearing-aid amplifier. In particular, a mathematical technique for characterizing the amplifier in combination with the receiver is reported. Cascade parameters of a two-port model of one particular amplifier/receiver combination are obtained by this method. The cascade-parameter data and the method of obtaining this data are verified by two different experimental procedures. One procedure involves both computing and measuring the input driving-point impedance of the amplifier/receiver combination. In the second procedure, the amplifier-to-eardrum transfer function of a hearing aid incorporating this same amplifier/receiver combination and mounted on an artificial ear is both computed and measured. Experimental and computed values of this transfer function for three different earmold geometries are in reasonably close agreement. The amplifier/receiver model reported herein will be used in future studies of acoustic feedback in hearing aids.     

Simulating the open-loop transfer function as a means for understanding acoustic feedback in hearing aids

Suppressing unstable acoustic feedback in hearing aids will first require knowledge of the open-loop transfer functions of such systems. Reported herein is a mathematical technique for simulating the open-loop transfer function of an in situ eyeglass-type hearing aid. In particular, a computer program was developed that characterized the hearing aid as a serial connection of two-port blocks, each representing one individual component of a hearing aid. Included, for example, were two-port blocks representing the microphone, amplifier, receiver, sound tubes leading to the eardrum (including the ear canal itself), earmold vent, and external pathway from the vent outlet back to the microphone. The computer program was validated by replicating laboratory data derived from an experiment involving a nonstandard manikin fitted with a nonstandard artificial ear. Next, the open-loop transfer function of an eyeglass-type hearing aid in situ on the manikin was simulated via the computer program. Unfortunately, those computer-generated data were not replicated in the laboratory due to the difficulty encountered in actually measuring the open-loop transfer function. Nevertheless, investigators were able to utilize those data to predict, within +/- 25 Hz, the "squeal" frequency of unstable acoustic feedback.     

Nonlinear model and system identification of a capacitive dual-backplate MEMS microphone

This paper presents the nonlinear identification of a capacitive dual-backplate microelectromechanical systems (MEMS) microphone. First, a nonlinear lumped element model of the coupled electromechanical microphone dynamics is developed. Nonlinear finite element analyses are performed to verify the accuracy of the lumped linear and cubic stiffnesses of the diaphragm. In order to experimentally extract the system parameters, an approximate solution using the second-order multiple scales method is synthesized for a nonlinear microphone model, subject to an electrical step input. A nonlinear least-squares technique is then implemented to extract system parameters from laser vibrometry data of the diaphragm motion. The results indicate that the theoretical fundamental resonant frequency, damping ratio and nonlinear stiffness parameter agree with the corresponding extracted experimental parameters with 95% confidence interval estimates.     

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。      

Modelling the pressure field in the vicinity of a microphone membrane using PIV

A new approach for estimating the acoustic pressure in the near field of a microphone based on non-intrusive direct measurement of acoustic particle velocity is proposed.This method enables the estimation of the acoustic pressure inside a domain located in front of the microphone membrane. The acoustic pressure is calculated using the acoustic particle velocity on the frontiers of this domain and a physical model based on the Green function of the system.Results are obtained using the acoustic velocity measured with Particle Image Velocimetry (PIV) in front of a microphone excited with a plane wave inside a rectangular waveguide. They show that the diffraction of the plane wave by the microphone leads to an increase of the acoustic pressure on the microphone edge in the order of magnitude of 0.1 dB.     

0.1~20Hz低失真度激光活塞发声器的设计与实现

为实现次声频段传声器的准确校准,研制了低失真、低泄漏的激光活塞发声器。激光活塞发声器采用具有空气轴承和位移反馈控制的超低频振动台驱动的活塞-腔体组合获得低失真度的声压信号;采用气浮自对中技术的小间隙缸塞配合以及大体积的腔体设计,获得了低泄漏的活塞发声器。在0.1~20 Hz,声压波形总谐波失真低于0.8%,实现高达50 s的腔体泄漏时间常数,使其适用于更低频段的传声器校准。在0.1~20 Hz频率范围内,激光活塞发声器的测量不确定度不超过0.58 dB (k=2)。与耦合腔互易法、关联传声器法的比较结果表明,激光活塞发声器法获得的传声器灵敏度与其他方法之间具有很好的一致性。      

Uncertainty analysis of the standard delay-and-sum beamformer and array calibration

Beamforming has become an ubiquitous task in aeroacoustic noise measurements for source localization and power estimation. The standard delay-and-sum (DAS) beamformer is the most commonly used beamforming algorithm due to its simplicity and robustness and also serves as the basis for more sophisticated algorithms, such as the deconvolution approach for the mapping of acoustic sources (DAMAS). The DAS data reduction equation is a function of many parameters including the microphone locations, microphone transfer functions, temperature and the cross-spectral matrix (CSM), where each one of these parameters has a unique uncertainty associated with it. This paper provides a systematic uncertainty analysis of the DAS beamformer and Dougherty's widely used calibration procedure under the assumption that the underlying mathematical model of incoherent, monopole sources is correct. An analytical multivariate method based on a first-order Taylor series expansion and a numerical Monte-Carlo method based on assumed uncertainty distributions for the input variables are considered. The uncertainty of calibration is analyzed using the Monte-Carlo method, whereas the uncertainty of the DAS beamformer is analyzed using both the complex multivariate and the Monte-Carlo methods. It is shown that the multivariate uncertainty analysis method fails when the perturbations are relatively large and/or the output distribution is non-Gaussian, and therefore the Monte-Carlo analysis should be used in the general case. The calibration procedure is shown to greatly reduce the uncertainties in the DAS power estimates. In particular, 95 percent confidence intervals for the DAS power estimates are presented with simulated data for various scenarios. Moreover, the 95 percent confidence intervals for the integrated DAS levels at different frequencies are computed using experimental data. It is shown that with experimental data, the 95 percent confidence intervals for the integrated power levels are within of the mean levels when the component uncertainties are set at low but achievable values.     

Evaluation of a noise reduction system for the assessment of click-evoked otoacoustic emissions

A recently proposed noise reduction system intended to facilitate the assessment of click-evoked otoacoustic emission (CEOAE) in noisy environments [Comput. Biol. Med. 30, 341 (2000)] is evaluated using 13 normally hearing ears and 9 ears with a sensorineural hearing loss. The noise reduction system is based on an adaptive noise canceller design using an additional noise-only reference microphone and intended to reduce externally generated noise. The system is tested in quiet and at different levels of white noise. The three main design parameters of the noise reduction system (adaptation time constant, length of the adaptive filter, and position of the noise reference microphone) are varied systematically in different experiments. With the noise reduction system active, CEOAE can be assessed correctly at noise levels which are 5 to 9 dB higher than without the noise reduction system. For the range of adaptation time constants considered (65.6 to 656 ms), no statistically significant effect on the amount of noise reduction is observed. Noise reduction is highest when the reference microphone is positioned close to the ear probe. Using this reference microphone position and adaptive filters of 6.56 ms in length, average noise reductions of 7.17 to 8.50 dB are achieved.     

A new microphone system for near whispering

A new microphone system was developed to monitor the human voice near the microphone in a noisy environment. The system is equipped with two special functions in addition to the usual microphone functions: reduction of air-blow effects by the mouth and focused reception to a sound source. A wind filter was developed to reduce the air-blow effects from the mouth during speaking. This filter is a plate perforated by an array of small holes; the method used to design the filter is also presented. To achieve focused reception, four microphones were used in conjunction with a new signal-processing method. The proposed signal-processing method effectively increases the directivity in the desired direction. Additionally, it provides the system with focusing on the source since the source is located adjacent to the system. A prototype of the proposed system was fabricated and subjected to performance tests. The results showed that air-blow effects can be reduced by up to 20 dB and the directional gain is more than 4 dB. The proposed microphone system shows such good performance that it can be used in mobile phones for whispering communication.     

基于BLT方程的双层腔体屏蔽效能计算方法

基于波导理论,将BLT方程进一步拓展,提出一种可快速、准确地计算双层腔体内任意点屏蔽效能的方法。首先将腔体壁孔所在面等效为二端口网络,腔体等效为两端短路的波导,建立信号流图。引入孔阻抗计算二端口网络散射参数,依据信号流图构建广义BLT方程,得到内层腔体中心线上点的屏蔽效能;然后根据波导理论的场分布特性,推导出腔体内任意点电场与腔体中心线上电场的关系,最终得到内层腔体内任意点的屏蔽效能。将计算结果与等效电路法及CST数值仿真结果对比,三者吻合良好,计算结果在较大频率范围内比等效电路法精度更高。该方法可以准确预测双层腔体在0~2.5GHz范围内所有谐振点,有助于分析腔体谐振现象,且计算效率较高,占用资源大幅减少。     

An alternative to the traveling-wave approach for use in two-port descriptions of acoustic bores

For more than a decade, the digital waveguide model for musical instruments has been improved through the simulation of cylindrical and conical bores. But several difficulties remain, such as instabilities due to growing exponentials which appear when two conical bores are connected with decreasing taper. In this paper, an alternative overcoming these difficulties is proposed and can be extended to shapes other than cylinders, cones, and hyperbolic horns. A two-port model with more general state variables than usual traveling waves works efficiently for any shape without discontinuities in cross section. The equations for connecting separate elements at discontinuities make this two-port model appropriate for use in time domain simulation of the physical behavior of the wind instrument and its interactions with the player. The potential of this new approach is illustrated by several detailed examples.     

用于自适应波束形成语音增强的球谐域掩蔽函数估计方法

提出一种用于球形阵列自适应波束形成的掩蔽函数估计方法.该方法利用包含空间信息的球谐系数提取低维空间向量,并采用复高斯混合模型和深度学习两种方案来估计掩蔽函数,最终利用估计的掩蔽函数设计最小方差无失真响应波束形成器,以达到空域滤波的效果.理论分析和仿真实验证明,对于相同时长的声信号,球谐域掩蔽函数估计方法的计算复杂度比传...     

Conversion of a head-mounted microphone signal intocalibrated SPL units

A method is reviewed for conversion of a microphone signal into calibrated Sound Pressure Level (SPL) units. The method follows American National Standards Institute (ANSI) standard for S1.4 SPL meters and requires an accurate SPL meter and an accurate calibration sound source for conversion. Accuracy and validation data from test signals and human phonation are provided. The results indicate that under typical speech conditions, an absolute accuracy of plus or minus 1.6 dB (type I SPL meter) can be obtained with a miniature head-mounted microphone.     

共光声池腔芯轴型空气衬底光纤麦克风

声传感器和光声池是激光光声光谱技术的核心组件。结合光纤迈克耳孙干涉仪、相位载波解调技术和纵向共振光声池,提出一种共光声池腔的芯轴型空气衬底光纤麦克风。光纤麦克风中的铜毛细管被用作光声池的共振腔,传感臂由10 m长的超细光纤缠绕在铜毛细管上构成,参考臂为5 cm长的短臂且已进行隔声隔振处理。基于结构共振频率稳定的特点,优化光纤麦克风的共振频率,使其略低于光声池的一阶纵向共振频率,以实现准双共振。实验结果表明,麦克风在共振频率为1443 Hz处的最小可检测声压为0.69μPa√Hz。在1 kHz处,声压电压响应线性度为99.98%(5 mPa~3 Pa),动态范围为112.52 dB。该光纤麦克风适用于高温、易爆和高电磁干扰等特殊环境下痕量气体的高精度检测。     

A laser pistonphone based on self-mixing interferometry for the absolute calibration of measurement microphones

A laser pistonphone for the absolute calibration of microphones at low frequencies has been developed at UME. The motion of an electro-dynamically driven piston in a small close cavity produces a sound pressure. Accurate measurement of the piston displacement, by self-mixing interferometry, enables this sound pressure to be calculated, and consequently the pressure sensitivity of a microphone, exposed to this sound pressure, to be determined. Absolute calibrations of type LS1P and WS1P microphones have been carried out with an uncertainty of less than 0.15 dB. The performance of the laser pistonphone has been validated by comparing the measured microphone sensitivities with those obtained by the closed coupler reciprocity method.     

Determination of energy density in ducts by a three-microphone phaseless method and estimation of measurement uncertainties

A new method to measure the total energy density of waves traveling in opposite directions in ducts is suggested in order to completely eliminate phase errors that lead to bias errors and are difficult to control in industrial tests. Only the auto-power spectral densities are measured by the three microphones. The inversion of a linear system based on a propagation model, where the two opposite waves are partially coherent, makes it possible to obtain the energy density. The sensitivity of this method to errors in the speed of sound, errors of microphone calibration and errors of microphone positions in the duct is analyzed. To complete the study on the robustness of the method, an evaluation of the statistical errors is carried out. The total uncertainty is used to make recommendations on the choice of the experimental parameters. The selection of the frequency limits permits to maintain the measurement uncertainty within a given confidence interval.     

基于遗传算法的传声器阵列位置有源校正研究

传声器阵列的位置误差会导致高精度定位应用场景中算法性能下降,为解决这一问题,该文提出了一种二维平面传声器阵列位置参数的有源校正方法.在校正声源位置已知的情况下,利用各阵元间的到达时间差均方误差之和最小化作为优化目标设计代价函数,根据遗传算法搜索阵列真实位置的全局最优解,从而估计得到传声器阵列的准确位置信息.仿真与半消声...     

传声器次声段灵敏度校准的误差机理研究*

次声波广泛存在于自然界与人类活动中,由于次声波具有穿透能力强、衰减小等特点,隔离和消除十分困难,对次声的监测防护与利用成为人们关注的焦点。次声波的定量研究依托传声器的测量实现,对传声器的灵敏度原级校准是保证测量精度的前提。该文基于活塞发生器校准原理下的压力泄漏与热传导耦合衰减模型,在COMSOL多物理场仿真软件中对校准腔和传声器后腔内的声场特征进行了数值仿真。创新性地提出了模型比较法,准确量化了次声段校准核心的泄漏、热传导独立与耦合修正量,验证并揭示了声压泄漏与热传导损失的幅值与相位变化机理。进一步地对传声器的次声段灵敏度校准过程进行了联合数值模拟,仿真结果表明,内外均压校准声场下传声器灵敏度的幅相频响应存在明显差异,但与校准腔内的泄漏与热传导效应无关。基于搭建的活塞发生器原级校准平台对传声器系统进行了原级校准,通过实验手段初步揭示了内外均压校准机制下传声器幅值灵敏度与相位灵敏度的显著差异。     

基于场路仿真的PCIe电磁干扰分析及优化设计

采用电磁仿真技术,提前评估PCB的电磁兼容设计是否合理,当对PCB进行电磁兼容测试时,减少其电磁干扰不满足GMW 3097标准的情况出现。首先对PCB进行3D电磁场仿真,再与高速串行计算机扩展总线标准模块内芯片的通用模拟电路仿真模型的电路仿真动态链接,进行场路协同仿真。实验验证表明,该仿真方法的精度在6 dBμV之内,满足PCB加工工艺的误差和实验测试的不确定度,符合仿真精度要求。通过该仿真方法评估PCB的电磁干扰强度以及优化PCB的设计,将高速串行计算机扩展总线标准模块上的33Ω电阻替换为磁珠后,该PCB在1.6 GHz处的电磁干扰强度降低了13.4 dB。根据CISPR 25标准规定的1-m法进行测试,PCB的电磁干扰变为-3.4 dBμV,低于GMW 3097标准要求,从而验证了该措施的有效性。