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。      

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。     

Calibration of the pressure sensitivity of microphones by a free-field method at frequencies up to 80 khz

A free-field (FF) substitution method for calibrating the pressure sensitivity of microphones at frequencies up to 80 kHz is demonstrated with both grazing and normal-incidence geometries. The substitution-based method, as opposed to a simultaneous method, avoids problems associated with the nonuniformity of the sound field and, as applied here, uses a 1/4-in. air-condenser pressure microphone as a known reference. Best results were obtained with a centrifugal fan, which is used as a random, broadband sound source. A broadband source minimizes reflection-related interferences that can plague FF measurements. Calibrations were performed on 1/4-in. FF air-condenser, electret, and microelectromechanical systems (MEMS) microphones in an anechoic chamber. The uncertainty of this FF method is estimated by comparing the pressure sensitivity of an air-condenser FF microphone, as derived from the FF measurement, with that of an electrostatic actuator calibration. The root-mean-square difference is found to be +/- 0.3 dB over the range 1-80 kHz, and the combined standard uncertainty of the FF method, including other significant contributions, is +/- 0.41 dB.     

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.     

A miniaturized adaptive microphone array under directional constraint utilizing aggregated microphones

This paper introduces a miniaturized microphone array using the Directionally Constrained Minimization of Power (DCMP) method, which utilizes the transfer functions of microphones located at the same place, namely aggregated microphones. The phased microphone array realizes a noise reduction and direction of arrival (DOA) estimation system according to differences in the arrival time, phase shift, and/or the level of the sound wave for each microphone. Hence it is difficult to miniaturize the microphone array. The objective of our research is to miniaturize the system size using aggregated microphones. In this paper, we first show that the phased microphone array system and the proposed aggregated microphone system can be described within the same framework. We then apply a microphone array under directional constraint to the aggregated microphones and compare the proposed method with the microphone array. We show the directional pattern of the aggregated microphones. We also show the experimental results regarding DOA estimation.     

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.     

Foil electret transducer for blood pressure monitoring

A foil electret microphone for use under the cuff of an automatic blood pressure monitoring system is described. The transducer is designed to operate with relatively flat sensitivity over a static pressure range of 40 to 250 mm Hg (5.33 X 10(4) to 3.33 X 10(5) dyn/cm2). The new electret microphone differs from conventional microphones used for airborne sound reception in two ways: (1) the diaphragm thickness is 50 micron rather than the typical 12.5 or 25 micron, and (2) the backplate contains a set of annular ridges spaced at 4 mm rather than the typical 7-10 mm. This microphone offers three advantages over the piezoelectric microphone now in use: (1) greater tolerance in positioning the microphone over the brachial artery, (2) nearly 20-dB higher sensitivity and signal-to-noise ratio, and (3) the ability to obtain measurements with the microphone placed midway between the elbow and shoulder. Tests of the new foil electret microphone in conjunction with the automatic blood pressure monitoring system indicate that the automatic and conventional measurements of systolic and diastolic blood pressure agree to within 5 mm Hg at least 90% of the time. In addition, the electret microphone is able to obtain automatic measurements on subjects with a wider range of ages and sizes.     

Source characterization of a subsonic jet by using near-field acoustical holography

In the present study, patch near-field acoustical holography was used in conjunction with a multireference, cross-spectral sound pressure measurement to visualize the sound field emitted by a subsonic jet and to predict its farfield radiation pattern. A strategy for microphone array design is described that accounts for the low spatial coherence of aeroacoustic sources and for microphone self-noise resulting from entrained flow near the jet. In the experiments, a 0.8-cm-diameter burner was used to produce a subsonic, turbulent jet with a Mach number of 0.26. Six fixed, linear arrays holding eight reference microphones apiece were disposed circumferentially around the jet, and a circular array holding sixteen, equally spaced field microphones was traversed along the jet axis to measure the sound field on a 30-cm-diameter cylindrical surface enclosing the jet. The results revealed that the jet could be modeled as a combination of eleven uncorrelated dipole-, quadrupole-, and octupole-like sources, and the contribution of each source type to the total radiated sound power could be identified. Both the total sound field reconstructed in a three-dimensional space and the farfield radiation directivity obtained by using the latter model were successfully validated by comparisons to directly measured results.     

Instrumentation for space averaging sound pressure levels

The measurement of machinery noise, sound-transmission loss, and reverberation decay inside reverberant enclosures requires multiple measurements obtained from several positions in the test enclosure. These measurements are used to establish a space-average sound pressure. The paper describes a multichannel microphone system that sequentially samples six microphones to provide a single microphone signal. This composite signal is then analyzed. No further space-averaging computation is required. The averaging time of the recorder is discussed with reference to swinging microphone arrangements and microphone multiplexing systems.     

Design and implementation of a MEMS microphone array system for real-time speech acquisition

Despite many attractive features and the potential for capturing sound in challenging acoustic environments, arrays with a large number of microphones have for a long time been discarded as a practical solution for speech acquisition. This is, among other reasons, due to the high production and computational costs. Only a few realizations of large microphone array systems have been documented, mainly for research and instrumentation use. The advent of MEMS microphones and computationally powerful off-the-shelf hardware has created new possibilities for microphone array development. We investigate a real life application, specifically the case of live sports broadcast, and the requirements that a such application imposes on a microphone array system. We present a system architecture of the first large (300 element circular array with a diameter of 2 m) MEMS microphone array system. In the proposed system, the latest technological advances are utilized to create a user-friendly array control interface. The array’s performance is examined in an anechoic chamber and on a crowded basketball field, and finally compared with existing solutions. The results illustrate the potential of a large MEMS microphone array as part of the technological development in sound acquisition for entertainment and security applications.     

Measurement of in-duct acoustic properties by using a single microphone with fixed position

Acoustic properties of sound absorption materials and other acoustic structures can be measured in an impedance tube using the well-established two-microphone method to resolve the two traveling wave components of a standing wave pattern. The accuracy of such measurements depends crucially on the calibration of the two microphones placed in close proximity. To eliminate such calibration, the one-microphone method [Chu, J. Acoust. Soc. Am. 80, 555-560 (1986)] uses the same microphone to probe at two positions sequentially using the voltage driving the loudspeaker as a reference signal. A variant of this method is introduced in this study in which the microphone is fixed at one position while a rigid end plate moves between two positions to resolve the standing wave. The sound source is installed as a side branch, and its driving signal is also used as a reference in the two-step measurement. Close agreement is found with the established two-microphone method, and factors which might affect the accuracy of the new technique are discussed. As a demonstration of the robustness of the method, a low-budget electret microphone is used and the result also matches well with those obtained by the two-microphone method with high-quality condenser type microphones.     

A directional acoustic array using silicon micromachined piezoresistive microphones

The need for noise source localization and characterization has driven the development of advanced sound field measurement techniques using microphone arrays. Unfortunately, the cost and complexity of these systems currently limit their widespread use. Directional acoustic arrays are commonly used in wind tunnel studies of aeroacoustic sources and may consist of hundreds of condenser microphones. A microelectromechanical system (MEMS)-based directional acoustic array system is presented to demonstrate key technologies to reduce the cost, increase the mobility, and improve the data processing efficiency versus conventional systems. The system uses 16 hybrid-packaged MEMS silicon piezoresistive microphones that are mounted to a printed circuit board. In addition, a high-speed signal processing system was employed to generate the array response in near real time. Dynamic calibrations of the microphone sensor modules indicate an average sensitivity of 831 microV/Pa with matched magnitude (+/-0.6 dB) and phase (+/-1 degree) responses between devices. The array system was characterized in an anechoic chamber using a monopole source as a function of frequency, sound pressure level, and source location. The performance of the MEMS-based array is comparable to conventional array systems and also benefits from significant cost savings.     

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.     

Binaural source location

It is shown how the coherence and phase spectra 0f signals from a closely spaced pair of microphones in the far-field can be used to compute the moments of a line distribution of arbitrarily correlated omni-directional sound radiators. This line source gives a far-field which is equivalent to that of model-scale and full-size turbojet engines in terms of measured power and cross-spectra 0f microphone signals. The necessary spectra can be computed rapidly on a small digital computer and the simplicity of the technique has meant that experiments could be performed in parallel with the usual far-field noise measurements. In this way, it has been possible t0 identify important properties of noise generation by turbojet engines at minimum cost and development and application of more sophisticated techniques has been accelerated. It is shown how the apparent properties of the source distribution may depend markedly on distance from the source to the microphones. Interpretation of results is guided by consideration of simple cases.     

Development of a multi-microphone calibrator

This paper presents the theory, design, and validation of a microphone calibrator used to simultaneously calibrate the amplitudes of multiple microphones on a single probe. The probe uses four 6 mm diameter electret microphones to acquire the data needed to compute acoustic energy density. This probe has prompted the need for simultaneous, multi-microphone amplitude calibration. The calibration process simultaneously subject each microphone on the probe to the same known acoustic pressure using four equal-length, small-diameter tubes connected to a single excitation source. A reference microphone connected to a fifth tube is used to calibrate the microphones. Test results show that the calibrator can calibrate each probe microphone within ±0.5 dB up to 2 kHz, and within ±1 dB up to 4.9 Hz with a confidence level of 95%.     

The effect of an air flow on a single side branch helmholtz resonator in a circular duct

A single side branch Helmholtz resonator has been tested in a circular duct in which both sound and an air flow could be passed in the same direction. The sound input was broad band and the transfer function across the resonator was determined by using two microphones and a cross-correlation analysis technique in which the broad band sound input was cross-correlated with each microphone signal in turn. The fundamental resonant frequency, as obtained from the transfer function of the resonator, was found to increase with increasing flow velocities in the duct. It was possible for the entire mass end correction of the orifice of the resonator to be eliminated by the flow. The higher natural frequencies of the resonator were not affected as much by the air flow.     

Developments in the use of Knowles miniature microphones to measure conditions inside the ear muffs

The Knowles series of miniature microphones have been used in the Flight Systems Department of the Royal Aircraft Establishment, Farnborough, Great Britain, for some years for the investigation of noise and speech levels at the ear inside the ear muff.This paper describes the types of microphone that have been used and the power units developed to enable them to feed a variety of measuring and recording equipment.This equipment has many other uses where a miniature microphone is required and the sound pressure level does not exceed 140 dB re 20 μPa.The equipment described has logged thousands of flying hours in RAF aircraft.     

A microphone multiplexer for the measurement of the space average of sound levels

Many of the acoustic measurements performed in rooms require that the space average sound pressure level be measured. In the system described in this paper, a fast scanning microphone multiplexer and a true RMS level detector are used for obtaining a direct reading of the SPL space average. The optimum scan rate (channels/sec) is dependent upon the number of microphones (between two and seven), the frequency, and the filter applied.     

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.