多阶声模态分解的改进阻抗管法测量材料的高频吸声系数

为了测量高频材料吸声系数,采用声模态分解的方法,基于阻抗管构建测试设备,在阻抗管内测量超过平面波截止频率的的高频吸声系数.测量过程中,通过在阻抗管的周向和轴向分别布置传声器阵列,分离管道内前3阶周向声模态以及各阶声模态的轴向传播入射波和反射波,从而得到最高频率达10000 Hz的材料吸声系数,并通过对比常规阻抗管测试方...     

The horizontal directivity of noise radiated by a rail and implications for the use of microphone arrays

Structural waves propagating along a railway rail form an extended source of sound radiation. Using an equivalent source model the distribution of this sound in a horizontal plane is investigated and shown to consist mainly of sound propagation at a particular angle to the normal. This direction is determined by the ratio of the wavenumbers in the rail and in air. Due to the extended nature of the rail as a source, the spatial distribution of the sound field in the direction along the track does not lend itself to the use of a simple directivity factor. The consequences for the measurement of noise from the rail using a microphone array are then explored. It is shown that a microphone array focussed normal to the rail does not detect most of the sound radiated by the rail. By turning the focus angle, the sound detected becomes a maximum when this angle corresponds to the angle of propagation of the sound radiation. Measurements on a test track using artificial excitation and measurements during the passage of a train confirm these conclusions.     

Transmission loss prediction of reactive silencers using 3-D time-domain CFD approach and plane wave decomposition technique

A predictive method is proposed to determine the transmission loss of reactive silencers using the three-dimensional (3-D) time-domain computational fluid dynamics (CFD) approach and the plane wave decomposition technique. Firstly, a steady flow computation is performed with a mass-flow-inlet boundary condition, which provides an initial condition for the following two unsteady flow computations. The first unsteady flow computation is conducted by imposing an impulse (acoustic excitation) superimposed on the constant mass flow at the inlet of the model and then adding the non-reflecting boundary condition (NRBC) when the impulse completely propagates into the silencer. The second unsteady flow computation is conducted for the case without acoustic excitation at the inlet. The time histories of pressure and velocity at the upstream monitoring point as well as history of pressure at the downstream monitoring point are recorded during the two transient computations. The differences between the two unsteady flow computational results are the corresponding acoustic quantities. Therefore, the incident sound pressure signal is obtained by using plane wave decomposition at upstream, while the transmitted sound pressure signal is just the sound pressure at downstream. Finally, those two sound pressure signals in the time-domain are transformed into the frequency-domain by Fast Fourier Transform (FFT) and then the transmission loss (TL) of silencer is determined. For the straight-through perforated tube silencers with and without flow, the numerical results agree well with the published measurements.     

Constraining the minute amount of audible energy radiated from binary collisions of light plastic spheres in conditions of incomplete angular coverage of the measured pressure

Usually, the energy released as air-coupled sound following a collision is dismissed as negligible. The goal of this Letter is to quantify the value of this small but measurable quantity, since it can be useful to impact studies. Measurements of sound radiation from binary collisions of polypropylene balls were performed in order to constrain the fraction of incident energy radiated as sound in air. In the experiments, one ball is released from rest, directly above a stationary target ball. The transient acoustic waveforms are detected by a microphone rotated about the impact point at a radius of 10 cm. The sound pressure was measured as a function of the polar angle θ (the azimuthal symmetry of the problem was verified by rotating the microphone in the horizontal plane). The angular pattern has two main lobes that are asymmetric with respect to the impact plane. This asymmetry is ascribable to interference and/or scattering effects. Gaps in the acoustic measurements at the "poles" (i.e., around 0° and 180°) pose a challenge similar to that of extrapolating the cosmic microwave background in the galactic "cut." The data was continued in the gaps by polynomial interpolation rather than least-squares fitting, a choice dictated by the accuracy of the reconstructed pattern. The acoustic energy radiated during the impact, estimated by multiplying the collision time by the sound intensity integrated over a spherical surface centered at the impact point, is calculated as four orders of magnitude smaller than the incident energy (0.23 μJ versus 1.6 mJ).     

基于四水听器的充水弹性管声速测量方法*

实验室中水声材料声学参数的测量主要在水声声管中进行。管内平面波声速是正确测量这些参数的基础。该文提出一种基于四水听器结合不同边界的测量充水弹性管中声速的新方法。该方法利用4个固定位置处的水听器,采用最小二乘的方法,使得两组水听器分别得到的声管末端入射波声压差值的平方最小的声速即为管内平面波声速。该方法利用单频信号,在每一频率点均可测得声速,可以在任一种声管末端边界下进行测量,同时无需知道各水听器到边界的精确距离,在文中的3种边界下声速测量结果具有很好的一致性,实验操作简单、误差很小。该方法的仿真结果与管内声速的理论值吻合得很好,同时实验测量结果与仿真值之间的误差很小,证明了方法的准确性以及鲁棒性,为声管声速测量提供一个很好的思路。     

数字化声学测量技术 Ⅱ.数字式声强及声功率测量系统

本文介绍在数字声学测量分析系统中，通过双传声器信号互谱密度的计算进行声强及声功率测量的基本原理。该数学分析系统由微计算机，数字信号处理卡和Ａ／Ｄ变换卡组成。在一个数字系统中，通过快速傅里叶变换（ＦＥＴ）进行互谱计算是十分有效的。本文着重介绍了，在声强的测量分析中对声强探头两传声器的固有相位差进行补偿的重要性和补偿方法，这是声强测量的重要环节。     

On the acoustic performance of rectangular splitter silencers in the presence of mean flow

Dissipative splitter silencers are often used to reduce the noise emitted in ventilation and gas turbine systems. It is well known that the acoustic performance of a splitter silencer changes under the influence of the convective effects of a mean gas flow and so in this article a theoretical model is developed to include the effects of mean flow. The theoretical model is based on a hybrid finite element method which enables the inclusion of bull nose fairings and a perforated screen separating the mean gas flow from a bulk reacting porous material. Predictions are compared against experimental measurements obtained both with and without mean flow. Good agreement between prediction and measurement is generally observed in the absence of mean flow, although it is seen that for silencers with a low percentage open area the silencer insertion loss is over predicted at higher frequencies. When mean flow is present, problems with the experimental methodology are observed at relatively modest mean flow velocities, and so comparison between prediction and experiment is limited to relatively low face velocities. However, experiment and theory both show that the insertion loss reduces at low frequencies when mean flow is in the direction of sound propagation, and at high frequencies the influence of mean flow is generally much smaller. Following additional theoretical investigations it is concluded that the influence of mean flow on splitter silencer performance should be accounted for at low frequencies when silencer airway velocities are greater than about 20 m/s; however, at higher frequencies one may generally neglect the effect of mean flow, even at higher velocities. Predictions obtained using the hybrid method are also compared to a simplified point collocation approach and it is demonstrated that the computationally efficient point collocation method may be used to investigate the effects of mean flow in a splitter silencer without loss of accuracy.     

Airflow noise in telephone handsets and methods for its reduction

The noise in a telephone handset produced by the bursts of airflow that accompany plosive sounds such as a "p" or "t" has been investigated. Measurements have been made on a series of modified handsets, using various arrangements of ports and tubes between microphone and the exterior of the telephone mouthpiece. A narrow stream of air was used to probe the mouthpiece and the signal picked up by the microphone recorded. The use of one, two, and multiple exterior holes was considered. Two-hole approaches alleviate the static pressure buildup that occurs when a single microphone hole is used and in many cases they result in reduced noise generation; for some directions of incident airflow, though, noise can be markedly increased. This was found to be due to interaction between intersecting streams of air. By using a large number of small (less than 0.5 mm) exterior holes, the airflow noise can be reduced by over 30 dB for all incident directions of airflow. It is also beneficial to ensure that none of the exterior holes is in direct line with the sound tube leading to the microphone.     

Attenuation of a resonant dissipative silencer in a rigid duct

Sound attenuation characteristics of a resonant-type dissipative silencer consisting of a reactive chamber with a porous facing have been considered. Such a silencer provides a high degree of attenuation within a narrow frequency range. Predicted attenuation values are compared with experiment for plane waves propagating in a rigid duct containing the dissipative silencer. The sound field is described by one-dimensional acoustical expressions taking into account the effect of boundary conditions and the presence of the silencer. The theoretical model incorporates the acoustical properties of porous materials and inertance of the sound field in the duct adjacent to the silencer. Good agreement was achieved between theoretical predictions and actual measurements. Results presented indicate the dependence of the attenuation spectrum upon flow resistivity and thickness of the porous material.     

Characteristics of surface sound pressure and absorption of a finite impedance strip for a grazing incident plane wave

Distributions of sound pressure and intensity on the surface of a flat impedance strip flush-mounted on a rigid baffle are studied for a grazing incident plane wave. The distributions are obtained by superimposing the unperturbed wave (the specularly reflected wave as if the strip is rigid plus the incident wave) with the radiated wave from the surface vibration of the strip excited by the unperturbed pressure. The radiated pressure interferes with the unperturbed pressure and distorts the propagating plane wave. When the plane wave propagates in the baffle-strip-baffle direction, it encounters discontinuities in acoustical impedance at the baffle-strip and strip-baffle interfaces. The radiated pressure is highest around the baffle-strip interface, but decreases toward the strip-baffle interface where the plane wave distortion reduces accordingly. As the unperturbed and radiated waves have different magnitudes and superimpose out of phase, the surface pressure and intensity increase across the strip in the plane wave propagation direction. Therefore, the surface absorption of the strip is nonzero and nonuniform. This paper provides an understanding of the surface pressure and intensity behaviors of a finite impedance strip for a grazing incident plane wave, and of how the distributed intensity determines the sound absorption coefficient of the strip.     

A technique for the in situ phase calibration of in-duct axial microphone arrays

In aeroengine noise experiments in-duct microphone arrays are often used to make detailed measurements of the sound field transmitted along the duct. The individual microphones in the array must be calibrated with respect to magnitude, and often more critically with respect to phase. Calibration is difficult to perform in situ due to the presence of the duct. This paper presents a technique to allow in situ phase calibration of axial microphone arrays. It relies on the observation that the measured cross-spectral pressure matrix at the array has a Hermitian Töplitz form in the case where the propagating duct modes are mutually incoherent. Using this property a system of equations can be written which, when solved, allows the phase calibration factors to be obtained. The technique is verified experimentally using a no-flow laboratory rig by comparing the phase calibration factors obtained with those measured in free-field conditions. The accuracy of the phase calibration factors obtained by the technique is limited by the degree of deviation of the measured cross-spectral matrix from Töplitz behaviour. In the experimental results shown this is less than 15° at duct frequencies below ka=25. The technique is a robust and rapid method for calibrating in-duct axial microphone arrays.     

Flat-response sound source technique for using the two-microphone method in an impedance tube

In this letter, to improve the acoustic impedance measurement in a tube using the two-microphone method, a sound source technique is developed that is capable of generating incident sound waves of uniform intensity over the entire frequency range of interest. The basic principle is to use a digitally synthesized compensation input signal to achieve a flat response of the sound source provided by one or more loudspeakers. The present sound source technique has been demonstrated to be very effective and easily implemented with the current impedance tube testing systems.     

Acoustic focusing through two layer annuluses in air

We report an acoustic focusing lens composed of two-layer annuluses made of metal cylinders in air. We find that the cylindrical waves can be focused into a perfect point without diffraction in the centre of the annuluses, which arises from the Mie-resonance modes in the annuluses. The focusing frequencies are related to the size of the inner annulus, and the focusing effect can be applied to the annuluses with different shapes and incident positions. Interesting applications of the focusing lens in the acoustic beam splitter and directional transmitter with energy enhancement are further discussed.     

DIFFRACTION STUDIES OF SOME CONVENTIONAL AND NOVEL MICROPHONE BAFFLES

A directional microphone system for field recording of sounds in the air often involves a parabolic reflector to focus the sound waves on the microphone (transducer) element. Some deficiencies of such a system are noted with respect to reproduction of spectra. The reflector system, involving as it does a structure comparable to a wavelength in linear dimension, is not susceptible to traditional high- or low-frequency approximate methods of computation. Modern numerical techniques now permit precise calculation of the directional responses of small reflectors of various shapes. One result is a proposal for a very economical and effective system involving a plane reflector. Other baffle shapes are also investigated, which may be of interest in special applications.     

Dynamic measurement of room impulse responses using a moving microphone

A technique for the recording of large sets of room impulse responses or head-related transfer functions is presented. The technique uses a microphone moving with constant speed. Given a setup (e.g., length of the room impulse response), a careful choice of the recording parameters (excitation signal, speed of movement) leads to the reconstruction of all impulse responses along the trajectory. In the case of a moving microphone along a circle, the maximal angular speed is given as a function of the length of the impulse response, its maximal temporal frequency, the speed of sound propagation, and the radius of the circle. As a result of the presented algorithm, head-related transfer functions sampled at 44.1 kHz can be measured at all angular positions along the horizontal plane in less than 1 s. The presented theory is compared with a real system implementation using a precision moving microphone holder. The practical setup is discussed together with its limitations.     

Wave field synthesis of a sound field described by spherical harmonics expansion coefficients

Near-field compensated higher order Ambisonics (NFC-HOA) and wave field synthesis (WFS) constitute the two best-known analytic sound field synthesis methods. While WFS is typically used for the synthesis of virtual sound scenes, NFC-HOA is typically employed in order to synthesize sound fields that have been captured with appropriate microphone arrays. Such recorded sound fields are essentially represented by the coefficients of the underlying surface spherical harmonics expansion. A sound field described by such coefficients cannot be straightforwardly synthesized in WFS. This is a consequence of the fact that, unlike in NFC-HOA, it is critical in WFS to carefully select those loudspeakers that contribute to the synthesis of a given sound source in a sound field under consideration. In order to enable such a secondary source selection, it is proposed to employ the well-known concept of decomposing the sound field under consideration into a continuum of plane waves, for which the secondary source selection is straightforward. The plane wave representation is projected onto the horizontal plane and a closed form expression of the secondary source driving signals for horizontal WFS systems of arbitrary convex shape is derived.     

Transmission of low-frequency internal sound through pipe walls

Transmission loss measurements are reported for long steel pipes of circular crosssection, with air inside and out, excited by internal sound. At low frequencies (wavelength greater than the pipe diameter), most of the radiated sound is accounted for by pipe bending waves. In order to approach the much higher transmission loss predicted for pure breathing motion of the pipe, bending waves must be suppressed; this has been achieved for a straight pipe by careful isolation. A sharp 90 bend in the pipe is shown to cause significant bending-wave excitation when plane waves are incident on the bend.     

Two-dimensional higher-diffraction-order optical beam splitter based on phenanthrenequinone-doped poly(methyl methacrylate) photopolymer

A two-dimensional optical beam splitter has been realized that uses the higher diffraction orders of a refractive-index grating. Gratings were recorded experimentally with light from a semiconductor laser incident at a small angle on phenanthrenequinone-doped poly(methyl methacrylate) photopolymer. The incident signal beam, which was made up of three different wavelengths (632.8, 532.0, and 488.0 nm), was split by the grating into multiple output beams with nearly equal size and separation. Results are given for when the sample grating was placed behind, in front of, and in the focal plane of a Fourier lens. The properties of higher-order-diffraction images have been discussed. The discussion shows that a two-dimensional higher-diffraction-order optical beam splitter provides a practical method for splitting a signal beam.     

Edge wave on axis behind an aperture or disk having a ragged edge

Diffraction by a circular aperture or disk having a ragged edge is investigated. Theory and measurements are reported. The ragged edge is modeled as N arcs, of differing radii a(i), each of which contributes a scattered signal to the edge wave on axis behind the aperture or disk. The amplitude of each scattered signal is proportional to the angle of the arc, and the corresponding time delay is square root of [(ai)2 + (s0)2/c0], where s0 is the axial distance from the aperture plane and c0 is the sound speed. Kirchhoff theory is used to make the calculation. A formula is derived for the rms pressure of the edge wave in terms of the rms pressure and autocorrelation function of the incident wave. The formula can be evaluated for incident waves that are sinusoidal, random (e.g., noise), or transient. Predictions agree reasonably well with underwater measurements made with a spark-generated pulse incident on various apertures. The main result is that making the edge ragged reduces the rms pressure of the edge wave. Indeed, an edge profile is presented that, for a given frequency and axial observation point, eliminates the edge wave completely.     

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.