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.     

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%.     

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.     

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.     

Sound pressure distribution and power flow within the gerbil ear canal from 100 Hz to 80 kHz

Sound pressure was mapped in the bony ear canal of gerbils during closed-field sound stimulation at frequencies from 0.1 to 80 kHz. A 1.27-mm-diam probe-tube microphone or a 0.17-mm-diam fiber-optic miniature microphone was positioned along approximately longitudinal trajectories within the 2.3-mm-diam ear canal. Substantial spatial variations in sound pressure, sharp minima in magnitude, and half-cycle phase changes occurred at frequencies >30 kHz. The sound frequencies of these transitions increased with decreasing distance from the tympanic membrane (TM). Sound pressure measured orthogonally across the surface of the TM showed only small variations at frequencies below 60 kHz. Hence, the ear canal sound field can be described fairly well as a one-dimensional standing wave pattern. Ear-canal power reflectance estimated from longitudinal spatial variations was roughly constant at 0.2-0.5 at frequencies between 30 and 45 kHz. In contrast, reflectance increased at higher frequencies to at least 0.8 above 60 kHz. Sound pressure was also mapped in a microphone-terminated uniform tube-an "artificial ear." Comparison with ear canal sound fields suggests that an artificial ear or "artificial cavity calibration" technique may underestimate the in situ sound pressure by 5-15 dB between 40 and 60 kHz.     

Micromachined optical microphone structures with low thermal-mechanical noise levels

Micromachined microphones with diffraction-based optical displacement detection have been introduced previously [Hall et al., J. Acoust. Soc. Am. 118, 3000-3009 (2005)]. The approach has the advantage of providing high displacement detection resolution of the microphone diaphragm independent of device size and capacitance-creating an unconstrained design space for the mechanical structure itself. Micromachined microphone structures with 1.5-mm-diam polysilicon diaphragms and monolithically integrated diffraction grating electrodes are presented in this work with backplate architectures that deviate substantially from traditional perforated plate designs. These structures have been designed for broadband frequency response and low thermal mechanical noise levels. Rigorous experimental characterization indicates a diaphragm displacement detection resolution of 20 fm radicalHz and a thermal mechanical induced diaphragm displacement noise density of 60 fm radicalHz, corresponding to an A-weighted sound pressure level detection limit of 24 dB(A) for these structures. Measured thermal mechanical displacement noise spectra are in excellent agreement with simulations based on system parameters derived from dynamic frequency response characterization measurements, which show a diaphragm resonance limited bandwidth of approximately 20 kHz. These designs are substantial improvements over initial prototypes presented previously. The high performance-to-size ratio achievable with this technology is expected to have an impact on a variety of instrumentation and hearing applications.     

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.     

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.     

Behavioral avoidance threshold level of a harbor porpoise (Phocoena phocoena) for a continuous 50 kHz pure tone

The use of ultrasonic sounds in alarms for gillnets may be advantageous, but the deterring effects of ultrasound on porpoises are not well understood. Therefore a harbor porpoise in a large floating pen was subjected to a continuous 50 kHz pure tone with a source level of 122+/-3 dB (re 1 microPa, rms). When the test signal was switched on during test periods, the animal moved away from the sound source. Its respiration rate was similar to that during baseline periods, when the sound was switched off. The behavior of the porpoise was related to the sound pressure level distribution in the pen. The sound level at the animal's average swimming location during the test periods was approximately 107+/-3 dB (re 1 microPa, rms). The avoidance threshold sound pressure level for a continuous 50 kHz pure tone for this porpoise, in the context of this study, is estimated to be 108+/-3 dB (re 1 microPa, rms). This study demonstrates that porpoises may be deterred from an area by high frequency sounds that are not typically audible to fish and pinnipeds and would be less likely masked by ambient noise.     

Specification of absorbed-sound power in the ear canal: application to suppression of stimulus frequency otoacoustic emissions

An insert ear-canal probe including sound source and microphone can deliver a calibrated sound power level to the ear. The aural power absorbed is proportional to the product of mean-squared forward pressure, ear-canal area, and absorbance, in which the sound field is represented using forward (reverse) waves traveling toward (away from) the eardrum. Forward pressure is composed of incident pressure and its multiple internal reflections between eardrum and probe. Based on a database of measurements in normal-hearing adults from 0.22 to 8 kHz, the transfer-function level of forward relative to incident pressure is boosted below 0.7 kHz and within 4 dB above. The level of forward relative to total pressure is maximal close to 4 kHz with wide variability across ears. A spectrally flat incident-pressure level across frequency produces a nearly flat absorbed power level, in contrast to 19 dB changes in pressure level. Calibrating an ear-canal sound source based on absorbed power may be useful in audiological and research applications. Specifying the tip-to-tail level difference of the suppression tuning curve of stimulus frequency otoacoustic emissions in terms of absorbed power reveals increased cochlear gain at 8 kHz relative to the level difference measured using total pressure.     

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.     

Development of a hand-held sensor probe for detection of sound components radiated from a specific device using surface intensity measurements

In this paper, a hand-held sensor probe is developed for surface intensity measurements. The sensor probe is composed of a 1/2-in. condenser microphone and a lightweight accelerometer of 1 g (=10⁻³ kg) which are connected with a vibration damper made of silicon rubber. The reliable measurement range of the sensor probe is examined and shown to be 100 Hz to7 kHz for sound and vibration. The precision of intensity measurements is confirmed by experiments in noisy environment. The precision is shown to be less than 3 dB for a random noise environment when the S/N is greater than −10 dB and for pure tone environment when the S/N is greater than −5 dB. The sensor probe is applied to determine the sound power level of a hard disc drive unit of a personal computer in an office setting. Good agreement is obtained for A-weighted sound power levels determined by the ISO method.     

Non-line-of-sight sound source localization using matched-field processing

Acoustic diffraction allows sound to travel around opaque objects and therefore may allow beyond-line-of-sight sensing of remote sound sources. This paper reports simulated and experimental results for localizing sound sources based on fully shadowed microphone array measurements. The generic geometry includes a point source, a solid 90° wedge, and a receiving array that lies entirely in the shadow defined by the source location and the wedge. Source localization performance is assessed via matched-field (MF) ambiguity surfaces as a function of receiving array configuration, and received signal-to-noise ratio for the Bartlett and minimum variance distortionless (MVD) MF processors. Here, the sound propagation model is developed from a Green's function integral treatment. A simple 16 element line array of microphones is tested in three mutually orthogonal orientations. The experiments were conducted using an approximate 50-to-1-scaled tabletop model of a blind city-street intersection and produced ambiguity surfaces from source frequencies between 17.5 and 19 kHz that were incoherently summed. The experimental results suggest that a sound source may be localized by the MVD processor when using fully shadowed arrays that have significant aperture parallel to the edge of the wedge. However, this performance is reduced significantly for signal-to-noise ratios below 40 dB.     

Hearing thresholds of a harbor porpoise (Phocoena phocoena) for sweeps (1-2 kHz and 6-7 kHz bands) mimicking naval sonar signals

The distance at which active naval sonar signals can be heard by harbor porpoises depends, among other factors, on the hearing thresholds of the species for those signals. Therefore the hearing sensitivity of a harbor porpoise was determined for 1 s up-sweep and down-sweep signals, mimicking mid-frequency and low-frequency active sonar sweeps (MFAS, 6-7 kHz band; LFAS, 1-2 kHz band). The 1-2 kHz sweeps were also tested with harmonics, as sonars sometimes produce these as byproducts of the fundamental signal. The hearing thresholds for up-sweeps and down-sweeps within each sweep pair were similar. The 50% detection threshold sound pressure levels (broadband, averaged over the signal duration) of the 1-2 kHz and 6-7 kHz sweeps were 75 and 67 dB re 1 μPa(2), respectively. Harmonic deformation of the 1-2 kHz sweeps reduced the threshold to 59 dB re 1 μPa(2). This study shows that the presence of harmonics in sonar signals can increase the detectability of a signal by harbor porpoises, and that tonal audiograms may not accurately predict the audibility of sweeps. LFAS systems, when designed to produce signals without harmonics, can operate at higher source levels than MFAS systems, at similar audibility distances for porpoises.     

A method for estimating the orientation of a directional sound source from source directivity and multi-microphone recordings: principles and application

Taking into account directivity of real sound sources makes it possible to try solving an interesting and biologically relevant problem: estimating the orientation in three-dimensional space of a directional sound source. The source, of known directivity, produces a broadband signal (in the ultrasonic range, in this application) that is recorded by microphones whose position with respect to source is known. An analytical method to process the recorded signals and estimate source orientation is developed in this paper. Experiments testing method performance in estimating source orientation were performed both in a laboratory environment with a Polaroid transducer as source and in a flight room with a Myotis daubentonii bat. In the first case, results showed the estimation method to be accurate and pointed out its limitations. The latter case is significant as an example biological application of the method for extracting behavioral features from bats; results are compared with alternative calculations based on microphone root-mean-square (rms)-pressure values.     

基于传声器阵列过顶测量结果的飞机起落架噪声研究

应用由111个传声器组成的平面传声器阵列对当前流行的民用客机进场着陆过程中的机体噪声源进行了实验测量,本对七架窄体客机和七架宽体客机的起落架噪声进行了分析,得到了起落架噪声的频谱特性、指向特性和声级变化。研究发现,起落架噪声的频谱是由宽频随机噪声与一些较为明显的单噪声源组成,起落架噪声的指向性类似于一个水平放置的偶极子。不同飞机起落架噪声的声级相差较大,这说明可以通过重新结构设计降低起落架噪声。     

Blind extraction and localization of sound sources using point sources based approaches

This paper presents theoretical models for blind sound source localization and separation of the signals emitted by arbitrary point sources in free space. Source localizations are achieved by a model based approach that accounts for the spherical spreading of an acoustic wave and utilizes an iterative triangulation, based on the signals measured by a three-dimensional microphone array. Once source locations are determined, the source signals are separated by using the point source separation (PSS) method, which is valid for all types of signals, including harmonic, continuous, transient, random, narrowband and broadband. General solutions for signals separation are presented. Theoretically, PSS can reconstruct the individual source signals exactly. This is because it employs the free-space Green's function, which defines the exact correlation among individual sources and measurement microphones. To validate PSS, numerical simulations are carried out and results are compared with those obtained by FastICA (Independent Component Analysis) code. The impacts of various parameters such as the microphone configuration, type of source signals, signal to noise ratio, number of microphones and source localization errors on the quality of signals separation by using PSS and FastICA are examined. The advantages and disadvantages of PSS and FastICA are compared and discussed.     

传声器阵列特征值滤波去噪方法

作为二阶统计量的互谱矩阵(CSM)是声学成像算法的核心输入量.为增强传声器阵列的去噪表现,研究了互谱矩阵特征值滤波的机理,并提出了两种新型的特征值滤波方法的设计准则:(1)声源互谱矩阵的Stein无偏风险估计(SURE收缩),即基于SURE准则的特征值软阈值收缩;(2)进一步提高声源互谱矩阵EYM(Eckart-You...     

Measurements of human middle ear forward and reverse acoustics: implications for otoacoustic emissions

Middle and inner ears from human cadaver temporal bones were stimulated in the forward direction by an ear-canal sound source, and in the reverse direction by an inner-ear sound source. For each stimulus type, three variables were measured: (a) Pec--ear-canal pressure with a probe-tube microphone within 3 mm of the eardrum, (b) Vst--stapes velocity with a laser interferometer, and (c) Pv--vestibule pressure with a hydrophone. From these variables, the forward middle-ear pressure gain (M1), the cochlear input impedance (Zc), the reverse middle-ear pressure gain (M2), and the reverse middle-ear impedance (M3) are directly obtained for the first time from the same preparation. These measurements can be used to fully characterize the middle ear as a two-port system. Presently, the effect of the middle ear on otoacoustic emissions (OAEs) is quantified by calculating the roundtrip middle-ear pressure gain Gme(RT) as the product of M1 and M2. In the 2-6.8 kHz region, absolute value(Gme(RT)) decreases with a slope of -22 dB/oct, while OAEs (both click evoked and distortion products) tend to be independent of frequency; this suggests a steep slope in vestibule pressure from 2 kHz to at least 4 kHz for click evoked OAEs and to at least 6.8 kHz for distortion product OAEs. Contrary to common assumptions, measurements indicate that the emission generator mechanism is frequency dependent. Measurements are also used to estimate the reflectance of basally traveling waves at the stapes, and apically generated nonlinear reflections within the vestibule.     

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.