A theoretical and experimental investigation of sound intensity distribution within a splitter silencer

Results are presented of an investigation into the sound intensity distribution in a ventilation duct splitter silencer subject to plane waves incident in the axial direction, under conditions of zero flow. A theory is developed which accounts for the generation of nonplane fields at the cross-sectional discontinuities of the duct-silencer-duct system. A local reaction model of the splitter material is employed, and analytical solutions and numerical results are presented for the sound intensity distribution within the silencer. Experiments were conducted to measure the distribution of sound intensity in a silencer subject to incident plane waves. The conventional cross-spectral method of measuring sound intensity could not be used because of the practical difficulties in manoeuvring the two-microphone assembly through the narrow air passage and obtaining measurements at positions close to the wall of the splitters. Instead, a modified cross-spectral method was devised which employs only one microphone for measurement. In this method, the transfer functions between the loudspeaker input signal and the microphone output signal were measured at an array of microphone positions within the air passages of the silencer. The divergence equation for sound intensity was employed to infer transverse intensity distributions from the computed axial distribution. Reasonable qualitative agreement is obtained between theoretical and observed sound intensity distribution, but the local reaction model of the splitter is found not to be satisfactory at the lower frequencies.     

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.     

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.     

A study of active tonal noise control for a small axial flow fan

Sound radiated by a computer cooling fan consists of tones which are phase locked with the rotation, and other less deterministic tones and broadband random noise. This paper demonstrates the feasibility of globally eliminating the rotation-locked tones by applying a very simple destructive interference to a modified cooling fan with the number of struts equal to the number of rotor blades. The rig consists of a miniature electret microphone used as a rotation sensor, an ordinary loudspeaker, and a bandpass filter with adjustable amplitude and phase delay. The microphone is located at the inlet bellmouth of the fan to pick up the fluctuating aerodynamic pressure caused by the passing rotor blades. The pressure spectrum is rich in the blade passing frequency (BPF) and its low-order harmonics. It provides much better performance than a pulse-generating tachometer. Analysis of the original fan noise shows that about 90% of the radiated tonal sound is phase locked with rotation, and this portion is almost completely eliminated in all directions. The reductions of the radiated sound power in the first two BPFs are 18.5 and 13.0 dB, respectively, and the overall sound power reduction is 11.0 dB.     

Using a thin actuator as secondary source for hybrid passive/active absorption in an impedance tube

This paper describes the practical implementation of a piezoelectric actuator as secondary source for hybrid passive/active broadband sound absorption in a standing-wave tube. This actuator consists of a thin circular aluminium plate driven by a piezoelectric patch and glued to a flexible rubber support. The resulting device has been mounted in a thin metallic ring that fits perfectly to the tube diameter. Passive absorption is afforded by either a porous layer or a microperforated panel, backed by an air gap. Active absorption is accomplished by releasing the sound pressure at a microphone behind the material, using either a loudspeaker or the actuator as secondary source. Results of broadband sound absorption reveal the feasibility of the piezoactuator. Compared to the loudspeaker, this actuator allows to greatly reduce the whole thickness of the hybrid absorber.     

Study of vertical sound image control with parametric loudspeakers

A parametric loudspeaker utilizes nonlinearity of a medium and is known as a super-directive loudspeaker. In this paper, the sound localization in the vertical direction using the upper and lower parametric loudspeakers is confirmed by listening tests and physical measurements. The differences in levels between the upper and lower parametric loudspeakers are varied as a parameter. The direction of sound localization in the vertical plane can be controlled not only when the acoustical axis is set to the right ear but also when it is set to at 5 deg to the right of the right ear. The effect of the level difference between the upper and lower loudspeakers is weaker than the differences observed when using ordinary loudspeakers. We obtained interesting characteristics of the left-right sound localization in the horizontal plane with the upper and lower parametric loudspeakers in the vertical plane. It is found that by setting the parametric loudspeaker at the right ear (that is, the horizontal angle of a listener to it is only 3 deg to the right), the direction of sound localization in the horizontal plane moved approximately 10 deg to the right. Moreover, by setting the parametric loudspeaker 5 deg to the right, the direction of sound localization moves approximately 20 deg to the right. The ILD (Interaural Level Difference) using a dummy head is calculated from the measured left and right sound signals. It is determined that ILDs of the parametric loudspeaker are larger than those of the ordinary loudspeaker. A simple geometrical acoustic model is introduced and analyzed. The analysis helps to explain the measured characteristics.     

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.     

基于蒙特卡罗法的实时空间三维声源定位

构建了一个基于四个声音传感器的信号时延采集系统,根据采集系统得到的三个时间差和传感器的响应顺序,提出了一种基于蒙特卡罗法实时空间的三维声源定位算法.该算法通过三维声音定位的非线性方程,构建一个三维模函数,通过寻找空间全局收敛点,并根据公差容限进行变步长搜索,准确快速地计算出声源的位置.     

The flow noise level at microphones in flow ducts

When sound measurements are made in flow ducts one has to insure that the signal-to-noise ratio of the sound signal to the flow noise, which is produced by the turbulent flow at the microphone, is sufficiently high. To accomplish this, knowledge of the flow noise level under the specific given flow conditions is required. In this paper two procedures for determining the flow noise level are described. For the first method, the mean flow velocity and the turbulence level have to be known to estimate the flow noise level as a function of duct diameter and frequency. For the second method knowledge of only the mean flow velocity is required. The procedure involves two measurements: one with a microphone fitted with a conventional nose cone and one with a microphone fitted with a slit-tube. The slit-tube is a special device for reducing the flow noise level. Finally, guidelines are given for sound measurements in the presence of flow and for the use of a microphone with a slit-tube under unusual temperature conditions or in gases other than air.     

An experimental study of swinbanks' method of active attenuation of sound in ducts

The concept of attenuating sound propagating down a duct by means of an antiphase copy of the sound was first put forward forty years ago, but to date no practical system has been produced. The principal problem is to introduce the antiphase signal in such a way that it propagates only in the direction of propagation of the original sound and to ensure that this property can be maintained over a useful frequency range.A suitable arrangement of secondary sound sources has recently been proposed by Swinbanks and this paper describes experiments carried out in an attempt to realise a practical system.A unidirectional array of secondary sources has been successfully constructed around a rectangular duct, using loudspeaker drive units and electronic delays. Sound propagating in the direction of these sources was sampled and a control signal applied to the sources which in turn acted to significantly reduce the amplitude of the sound. Pure tones at frequencies around 150 Hz have been attenuated by more than 50 dB but results with band-limited noise have been less successful. Further work is suggested which should result in a device having significant advantages over conventional splitter silencers, at low frequencies.     

气流管道有源降噪研究

如何消除气流的影响是管道有源降噪的难点之一，本文分析了湍流对有源降噪系统的影响，并开发了一种抗湍流传声器探管，最后在某种通风系统的进气管段进行了降噪试验，气流速度为２０ｍ／ｓ时，在６０－６３０Ｈｚ频带取得了１５ｄＢ（Ａ）的降噪效果。     

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。      

Active attenuation of noise: The Chelsea dipole

Methods of active attenuation of noise, that is to cancel the noise from a source by the addition of further noise, include the method of destructive interference. A number of configurations of active attenuators are possible and a new system which originated in work at Chelsea College has been developed further. This system employs two spaced secondary sources in a duct energized in antiphase, with the microphone situated centrally between them. The radiation from the secondary sources cancels at the microphone, which, ideally, responds only to the travelling wave in the duct. The microphone output is phase shifted by 90° and then amplitude compensated by a transfer function $\text{1}\text{2}$ sin ($\text{kl}\text{2}$) before being fed to the secondary sources, in order to bring its downstream radiation into antiphase with the travelling noise wave. There is radiation both upstream and downstream from the secondary sources, but the microphone is isolated from the resulting upstream standing waves and time delays are not required, unlike the systems employing the microphone remote from the attenuator. The action of the attenuator has been investigated on pure tones and bands of noise with a range of attenuator centre frequencies.     

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.     

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.     

Simple sound power measurements

A simple method of measuring sound power is described. Noise is added from a second calibrated source until the room sound pressure level is increased by 3 dB. The noise output of the device is then equal to the added power. A simple loudspeaker, modified to reduce the effect of the environment on the sound power output, is used as the calibrated source of sound power. Checks show that the sound power readings are substantially independent of the room in which they are made.     

The measurement of the free field impulse response of microphones in a laboratory environment

A method is presented of measuring the free field frequency and impulse response of microphones by using a pulse technique in an ordinary laboratory environment. The pressure transient generated by exciting a loudspeaker with a narrow pulse is detected at some point in the loudspeaker's far field by a “reference” microphone whose response is assumed flat over the frequency range of interest. The microphone to be tested is then substituted in exactly the same position and its response to the transient measured. The outputs of the two microphones are accurately sampled and deconvolved by using a discrete Fourier transform technique to give the magnitude and phase parts of the “test” microphone's frequency response, and hence, via the discrete Fourier transform, its response to a delta function of pressure propagating in the free field. The computed impulse responses are presented and the accuracy of the method discussed.To illustrate the use of the method, the free field frequency response and free field correction curves of a one inch instrumentation microphone are measured. The method is then used to measure the pressure which occurs at the centre of the flat end-face of a long a cylinder when excited by an impulse of acoustic pressure propagating in the free field from various angles of incidence.     

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。      

Hysteretic behavior induced by an electroacoustic feedback loop in a thermo-acousto-electric generator

An active control method of the spatial distribution of the acoustic field is applied in a thermo-acousto-electric generator. An auxiliary acoustic source is used to force the self-sustained thermoacoustic oscillation in order to control the thermoacoustic amplification. The auxiliary source consists of a loudspeaker, located inside the loop-tube close to the main ambient heat exchanger, and supplied with a delayed signal through an electric feedback loop, comprising a phase-shifter and an amplifier, connected to a reference microphone. Experiments are performed on a prototype engine working with air at a static gauge pressure of 5 bars. Experimental results demonstrate how it is possible to tune the acoustic oscillations in order to increase the global performance of the generator, compared to the case without control, as well as the existence of a hysteretic behavior induced by the electroacoustic feedback loop itself, which leads to a discrepancy between the onset heat input and the offset one.     

Adaptive fuzzy neural network control on the acoustic field in a duct

Base on the principle of the superposition of waves, active noise control is achieved by adaptively tuning a secondary source which produces an anti-noise of equal amplitude and opposite phase with primary source. This paper presents the study on the acoustic attenuation in a duct by using the combination of fuzzy neural network with error back propagation algorithm to control secondary source. The most important advantage of fuzzy inference system is that the structured knowledge is represented in the form of fuzzy IF-THEN rules. But it lacks the ability to accommodate the change of external environments. Combining neural network with fuzzy system can help in this tuning process by adapting fuzzy sets and creating fuzzy rules. The performance of attenuation and control error can be measured by the microphone placed in the downstream of duct. The results of this study, show that the acoustic attenuation by 40 dB for pure-tone noise and nearly 30 dB for dual-tones noise are obtained.