Measurement of the frequency response of a diaphragm-based pressure sensor by use of a pulsed excimer laser

We present a novel method for measuring the frequency response of a diaphragm-based optical fiber Fabry-Perot interferometric pressure sensor. The impulse response of the sensor to the radiation pressure generated by an excimer laser pulse is measured. The Fourier transform of the impulse response yields the frequency response of the pressure sensor. Experimental results show that it is a convenient and efficient method for measurement of the frequency response of diaphragm-based pressure sensors.     

Real-time near-field acoustic holography for continuously visualizing nonstationary acoustic fields

Near-field acoustic holography (NAH) is an effective tool for visualizing acoustic sources from pressure measurements made in the near-field of sources using a microphone array. The method involving the Fourier transform and some processing in the frequency-wavenumber domain is suitable for the study of stationary acoustic sources, providing an image of the spatial acoustic field for one frequency. When the behavior of acoustic sources fluctuates in time, NAH may not be used. Unlike time domain holography or transient method, the method proposed in the paper needs no transformation in the frequency domain or any assumption about local stationary properties. It is based on a time formulation of forward sound prediction or backward sound radiation in the time-wavenumber domain. The propagation is described by an analytic impulse response used to define a digital filter. The implementation of one filter in forward propagation and its inverse to recover the acoustic field on the source plane implies by simulations that real-time NAH is viable. Since a numerical filter is used rather than a Fourier transform of the time-signal, the emission on a point of the source may be rebuilt continuously and used for other post-processing applications.     

A time-selective technique for free-field reciprocity calibration of condenser microphones

In normal practice, microphones are calibrated in a closed coupler where the sound pressure is uniformly distributed over the diaphragm. Alternatively, microphones can be placed in a free field, although in that case the distribution of sound pressure over the diaphragm will change as a result of the diffraction of the body of the microphone, and thus, its sensitivity will change. In the two cases, a technique based on the reciprocity theorem can be applied for obtaining the absolute sensitivity either under uniform pressure or free-field conditions. In this paper, signal-processing techniques are considered that improve the accuracy of the free-field calibration method. In particular, a fast Fourier transform (FFT)-based time-selective technique for removing undesired reflections from the walls of the measurement chamber has been developed and applied to the electric transfer impedance function between two microphones. The acoustic centers of the microphones have been determined from the "cleaned" transfer impedance values. Then, the complex free-field sensitivities of the microphones have been calculated. The resulting complex sensitivities and acoustic centers have proved to be in good agreement with previously published data, and this confirms the reliability of the time-selective technique, even in nonanechoic environments. Furthermore, the obtained results give a new reference for further comparisons, because they cover a frequency range with an accuracy that has not been obtained by previously published data.     

Least squares approach in wavenumber domain for sound field recording and reproduction using multiple parallel linear arrays

A novel signal processing method is proposed for sound field recording and reproduction using multiple parallel linear microphone and loudspeaker arrays. In sound field recording and reproduction, the problem is how to calculate the transfer filters that transform the signals recorded by microphones into the driving signals of the loudspeakers. The proposed method is based on the spatial Fourier transform in the horizontal angle combined with the least squares (LS) approach in the elevation angle. In the proposed method, the signals recorded by each linear microphone array and those that drive each loudspeaker array are decomposed into the wavenumber domain by the spatial Fourier transform in the horizontal direction. The transfer filters are then calculated by the LS approach in the wavenumber domain. As a result, the size of the matrix of each transfer function in the wavenumber domain is much smaller than that of the conventional LS approach in the temporal frequency domain (LSTF), and well-conditioned stable transfer filters can be obtained with low computational cost without regularization. Computer simulation results show that the proposed method reconstructed a sound field around the control points as accurately as the conventional LSTF.     

Measurements of echoes with a stationary random sound source

The separation of the direct wave issued from a stationary random sound source and echoes reverberated by surfaces is made by computing the Fourier transform of the transfer function of two microphone signals. Two methods are described for different positions of the microphones and are illustrated by experimental results concerning (i) free field measurements of ground reflexion coefficients, and (ii) measurements of echoes which are reverberated by the linings of an anechoic chamber.     

The cepstrum: a viable method for the removal of ground reflections

Noise data from open air test facilities is contaminated by the effect of ground reflections causing cancellations and augmentations of the sound at certain frequencies. This problem is generally dealt with by placing microphones near the ground and subtracting 6 dB from the SPL spectrum. However the high frequency part of the spectrum obtained in this way suffers from problems due to temperature gradients near the ground on sunny days. The present empirical method used at Rolls-Royce for the estimation of one third octave band free field spectra involves the use of two microphones, one at 0·051 m above the ground (ground level) for frequencies up to 1 kHz and the other at 1·524 m above the ground for higher frequencies. Corrections are applied to the one third octave spectra from these low and high microphones to take account of intensity increases. These spectra are then “married” at 1 kHz to produce a single free field spectrum. At present there is no similar method to obtain narrow band free field spectra. In this paper cepstrum analysis is proposed as a satisfactory method to produce both narrow band and one third octave band free field spectra from high level microphones only. A series of tests has been carried out in an anechoic chamber facility in which ground reflections were simulated. The cepstrum technique was applied to this data to deduce the free field spectra. These compared very well with free field spectra obtained under anechoic conditions. Data is also included from open air tests on the Viper 11 engine and the spinning rig jet noise facility to show that the cepstrum technique is a viable way of removing ground reflections from high level microphone data.     

Measurement of resonance frequency and loss factor of a microphone diaphragm using a laser vibrometer

The pressure sensitivity of a laboratory standard microphone is determined using a reciprocity technique that measures the electrical transfer impedance of two microphones connected acoustically by a coupler. The electrical transfer impedance is a function of the coupler volume and the equivalent volumes of the microphones. The equivalent volume given as a function of the frequency can be determined in experiments or can be calculated if the equivalent volume at a low frequency as well as the resonance frequency and loss factor of the microphone diaphragm are known. Therefore, it is necessary to determine the resonance frequency and the loss factor accurately to obtain an accurate reading of the pressure sensitivity.In this paper, a new method to determine the resonance frequency and loss factor of a microphone diaphragm is proposed. The frequency response of the diaphragm displacement is measured by a laser vibrometer and the part of the response near the resonance frequency is used to determine the microphone parameters via least square fitting with the equation of a vibration model with one degree-of-freedom. Since the values measured by this method are close to the nominal values and the repeatability is highly feasible, the proposed method will be useful to determine the resonance frequency and loss factor of a microphone diaphragm.     

Calibration of ear canals for audiometry at high frequencies

A procedure is described for determining the absolute sound pressure at the inner end of the ear canal when a sound source is coupled to the ear, for frequencies in the range 8-20 kHz. The transducer that generates the sound is coupled to the ear canal through a lossy tube, yielding a source impedance that is approximately matched to the characteristic impedance of the ear canal. A small microphone is located in the coupling tube close to the entrance to the ear canal. Calibration is carried out by measuring the response at this microphone when an impulse is applied at the transducer. To estimate the sound pressure at the medial end of the ear canal, the Fourier transform of this impulse response is corrected by an all-pole function in which the poles are estimated from the minima in this Fourier transform. Data on individual ear canals are presented in terms of gain functions relating the sound pressure at the medial end of the ear canal to the sound pressure when the coupling tube is blocked. The average gain function for a group of adult ears increases from 2 to 12 dB over the frequency range 8-20 kHz, in rough agreement with data from ear-canal models. Possible sources of error in the calibration procedure are discussed.     

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.     

Modification of the discrete spectral parameters of optical solitons in fibers with variable dispersion

The spectral parameters of solitons result from the nonlinear Fourier transform for a pulse propagating in a fiber. A new approach for controlling the discrete component of spectral parameters is proposed. In the presence of a resonance between field oscillations and fiber dispersion, the real part of spectral parameters change, which leads to multisoliton pulse separation into fundamental solitons. It is proposed to use this effect for signal decoding in communication lines operating based on the nonlinear Fourier transform.     

Generation and reception of short acoustic pulses by multilayer piezoelectric transducers

By means of the Fourier transformation, a formula is obtained for calculating the pulse response of a transducer when a square pulse of a given duration is supplied to its input and the excited “sound” is reflected from the free flat end of the acoustic line and returns to the transducer. For a frequency of 100 MHz, real lithium niobate transducers that are connected through intermediate layers to the acoustic line made of fused quartz and to a rear load with a given acoustic impedance are considered. The pulse response is calculated for the transducers with different values of the sublayer thickness, rear acoustic load, transmission line impedance, and original pulse duration. The results of the calculations are compared with the experiment.     

Comparison of methods for processing acoustic intensity from orthogonal multimicrophone probes

One design for three-dimensional multimicrophone probes is the four-microphone orthogonal design consisting of one microphone at an origin position with the other three microphones equally spaced along the three coordinate axes. Several distinct processing methods have been suggested for the estimation of active acoustic intensity with the orthogonal probe; however, the relative merits of each method have not been thoroughly studied. This comparative study is an investigation of the errors associated with each method. Considered are orthogonal probes consisting of matched point sensor microphones both freely suspended and embedded on the surface of a rigid sphere. Results are given for propagating plane-wave fields for all angles of incidence. It is shown that the lowest error for intensity magnitude results from having the microphones in a sphere and using just one microphone for the pressure estimate. For intensity direction, the lowest error results from having the microphones in a sphere and using Taylor approximations to estimate the particle velocity and pressure.     

垂直极化平行板有界波电磁脉冲模拟器辐射近场的快速估算方法

为快速估算出垂直极化平行板有界波电磁脉冲(EMP)模拟器的时域近场,将散射传递函数法应用于该类型模拟器近场的时域计算中,即对于给定的脉冲源,先寻找有效频谱范围能覆盖该源的高斯脉冲源,并应用时域有限差分(FDTD)方法计算该高斯脉冲源激励时模拟器中测试点场的时域响应,再利用傅里叶变换、系统的传递函数及傅里叶逆变换计算得到给定脉冲源激励时各测试点场的瞬态响应。所得计算结果与直接使用给定脉冲源激励时FDTD方法的计算结果符合较好。所述方法可用于同一模拟器在不同脉冲源激励时辐射近场的快速估算,能大大减少FDTD模拟计算的次数,尤其对于中大型模拟器能有效减少计算时间和内存。     

Closed form solutions for the acoustical impulse response over a masslike or an absorbing plane

The transient sound field caused by a Dirac delta impulse function above an infinite locally reacting plane can be calculated by applying the inverse Fourier transform of the corresponding half-space Green's function in frequency domain. As a starting point, the representation given by Ochmann [J. Acoust. Soc. Am. 116(6), 3304-3311 (2004)] is used, which consists of discrete and continuous superposition of point sources. For a locally reacting plane with masslike character and also with pure absorbing behavior, it is possible to express the resulting impulse response in closed form. Such a result is surprising, because corresponding formulations in the frequency domain are not available yet. Hence, the first main result is the closed form solution Eq. (22) for an impulse response over an infinite plane with a pure imaginary impedance. The second main result is the closed form solution Eq. (53) for an impulse response over an infinite plane with a pure real impedance. As a particular application of both main results, a convolution technique is used for deriving formulas for point sources with a general time dependency. For special signals like an exponentially decaying time signal or a triangular shaped impulse, the resulting sound field can be presented in terms of elementary functions.     

基于Fournier Forand体积散射函数的水中激光脉冲后向散射特性分析模型

基于Fournier Forand体积散射函数,建立了一种水中激光脉冲后向散射仿真模型.运用该模型可用Monte Carlo方法模拟光子在水中的传播过程,并得到光波后向散射的冲击响应.将初始激光脉冲与冲击响应进行卷积并求其傅里叶谱,即可得到激光脉冲后向散射信号的时域和频域特征.利用该模型分析了入射为高斯型激光脉冲时,水中散射体的尺度分布、散射体与纯水的相对折射率以及水体衰减系数对激光脉冲后向散射特性的影响.结果表明：随着小尺度散射体相对数量的增多、散射体与纯水相对折射率的增加、水体衰减系数的增大,激光脉冲后向散射信号能量增强,宽度增加,低频分量显著增大.     

Secondary method for microphone calibration in normal rooms

In this paper a new secondary method for measuring the response curve of a microphone is proposed. It is based upon the linear anamorphical relationship that exists between the instantaneous levels of signals picked up by two microphones when they are placed in the same wave front. The main theoretical bases are presented along with experimental results for free and semi-reverberant fields for different kinds of test signals, including frequency swept signals. Accuracy under these conditions is analyzed and experimental procedures for maximizing it are proposed. When used in semi-reverberant fields and with continuous frequency sweep, this method has among its outstanding characteristics a great rapidity and reasonable precision. These features make the method very useful in the quality control of serial microphones in factories or laboratories in which anechoic chambers are not available.     

Measurements with an intensity meter of the acoustic power of a small machine in a room

A technique in which two closely spaced pressure microphones, a special purpose circuit, and a sound level meter are employed to measure acoustic intensity in octave bands, is used to estimate the intensity distribution around a small, 1200 electrical watt, machine situated in a room. The total acoustic power estimated therefrom is compared with that obtained by the conventional “direct field” method. The technique, which appears to be accurate over the range 250–400 Hz, produces values of intensity and power which are generally less than the “direct field” values. The difference tends to increase with frequency. A potential for source location application is indicated.     

The forward and backward projection of acoustic fields from axisymmetric ultrasonic radiators using impulse response and Hankel transform techniques

A generalized impulse response formulation to evaluate the harmonic pressure field of ultrasonic planar vibrators having axisymmetric nonuniform surface velocity distributions is presented. The harmonic pressure is expressed as a Fourier transform of a generalized impulse response which is a function of the spatially nonuniform velocity of the vibrator. A backward projection method is then developed to reconstruct the normal surface velocity of axisymmetric vibrators from harmonic field pressures using an angular spectrum or Hankel transform formulation. The numerical accuracy of the backward projection technique is evaluated using the impulse response formulation to evaluate the pressure fields for several velocity distributions on disk vibrators. Experiments were performed to reconstruct the velocity distributions over the surface of a uniformly driven piezoelectric ceramic disk and ceramic ring using farfield measurements of the complex pressure. The experimental results were in good agreement with theoretical results based on the electrode patterns of the transducers.     

FDTD结合DFT分析宽频高功率微波大气传播

联立麦克斯韦方程与电子流体方程,用时域有限差分法(FDTD)模拟高斯型和阻尼正弦型等宽频高功率微波(HPM)的大气传播.在每个时间网格上,根据窄带脉冲的电子速度,通过离散傅立叶变换(DFT)方法求解出宽频脉冲的等效电场,将等效电场和压强代入电离参数公式,使电离参数随空间网格不断更新,提高计算准确性.结果表明,宽频HPM脉冲幅值、脉宽以及海拔高度等参数对大气击穿有明显的影响;大气击穿导致尾蚀效应;随着传播距离的增加,宽频HPM脉冲的尾部衰减加剧,脉宽缩短,引起宽频脉冲的频谱出现展宽、分裂及中心频率移动等现象.     

Noise in miniature microphones

The internal noise spectrum in miniature electret microphones of the type used in the manufacture of hearing aids is measured. An analogous circuit model of the microphone is empirically fit to the measured data and used to determine the important sources of noise within the microphone. The dominant noise source is found to depend on the frequency. Below 40 Hz and above 9 kHz, the dominant source is electrical noise from the amplifier circuit needed to buffer the electrical signal from the microphone diaphragm. Between approximately 40 Hz and 1 kHz, the dominant source is thermal noise originating in the acoustic flow resistance of the small hole pierced in the diaphragm to equalize barometric pressure. Between approximately 1 kHz and 9 kHz, the noise originates in the acoustic flow resistances of sound entering the microphone and propagating to the diaphragm. To further reduce the microphone internal noise in the audio band requires attacking these sources. A prototype microphone having reduced acoustical noise is measured and discussed.