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.     

High-resolution plane-wave decomposition in an auditorium using a dual-radius scanning spherical microphone array

The spatial and temporal distribution of early reflections in an auditorium is considered important for sound perception. Previous studies presented measurement and analysis methods based on spherical microphone arrays and plane-wave decomposition that could provide information on the direction and time of arrival of early reflections. This paper presents recent results of room acoustics analysis based on a spherical microphone array, which employs high spherical harmonics order for improved spatial resolution, and a dual-radius spherical measurement array to avoid ill-conditioning at the null frequencies of the spherical Bessel function. Spatial-temporal analysis is performed to produce directional impulse responses, while analysis based on the windowed Fourier transform is employed to detect direction of arrival of individual reflections at selected frequencies. Experimental results of sound-field analysis in a real auditorium are also presented.     

Field measurement of airborne sound insulation between rooms with non-diffuse sound fields at low frequencies

Procedures for the field measurement of airborne sound insulation between rooms with diffuse fields are described in International Standard ISO 140-4. However, many dwellings contain rooms with volumes less than 50 m³, where low frequency measurements are less reliable; hence there is a need for a measurement procedure to improve the reliability of field measurements in rooms with non-diffuse fields. Procedures are proposed for sound pressure level and reverberation time measurements for the 50, 63 and 80 Hz third octave bands. The sound pressure level measurement combines corner microphone positions with positions in the central region of each room. This provides a good estimate of the room average sound pressure level with significantly improved repeatability.     

A system for simulating room acoustical environments for one’s own voice

The real-time simulation of room acoustical environments for one’s own voice using generic software has been difficult until very recently due to the computational load involved: requiring real-time convolution of a person’s voice with a potentially large number of long room impulse responses. This paper describes a software-based solution that accomplishes real-time convolution with head-tracking to simulate the effect of room acoustical environments on the sound of one’s own voice, using binaural technology. Actual rooms are characterized by measuring the room impulse response from the mouth to ears of the same head (oral binaural room impulse response, OBRIR). By repeating this process at 2° yaw increments for a given head position, the rooms are binaurally scanned around a given position to obtain a collection of OBRIRs, which is then used by the software-based simulation system. In the simulated rooms, a person equipped with a near-mouth microphone and near-ear loudspeakers can speak or sing and hear their voice, as it would sound in the recorded rooms, while physically being in an anechoic room. By continually updating the person’s head orientation using head-tracking, the corresponding OBRIR is chosen for convolution with their voice. The system described in this paper achieves the low latency that is required to simulate nearby reflections, and it can perform convolution with long room impulse responses.     

Development of sound measurement systems for auditory functional magnetic resonance imaging

Auditory functional magnetic resonance imaging (fMRI) requires quantification of sound stimuli in the magnetic environment and adequate isolation of background noise. We report the development of two novel sound measurement systems that accurately measure the sound intensity inside the ear, which can simultaneously provide the similar or greater amount of scanner- noise protection than ear-muffs. First, we placed a 2.6 x 2.6-mm microphone in an insert phone that was connected to a headphone [microphone-integrated, foam-tipped insert-phone with a headphone (MIHP)]. This attenuated scanner noise by 37.8+/-4.6 dB, a level better than the reference amount obtained using earmuffs. The nonmetallic optical microphone was integrated with a headphone [optical microphone in a headphone (OMHP)] and it effectively detected the change of sound intensity caused by variable compression on the cushions of the headphone. Wearing the OMHP reduced the noise by 28.5+/-5.9 dB and did not affect echoplanar magnetic resonance images. We also performed an auditory fMRI study using the MIHP system and presented increase in the auditory cortical activation following 10-dB increment in the intensity of sound stimulation. These two newly developed sound measurement systems successfully achieved the accurate quantification of sound stimuli with maintaining the similar level of noise protection of wearing earmuffs in the auditory fMRI experiment.     

基于时间反转的复杂声场拾声传声器阵列性能研究

探讨时间反转技术在复杂声场传声器阵列拾声中应用的可行性及其机理,给出其一般规律和性能。研究表明:在自由空间中,其拾声性能与频率,阵列形状和半径有关,频率越高,半径越大,拾声效果越好。在普通房间中,在语音频段内,圆弧阵列在预定目标点处的阵列增益性能要比离预定目标点约25 cm远处的位置处大5 dB以上。在普通房间和混响室中的实验验证了上述结论。      

Spatial correlation and coherence in reverberant acoustic fields: Extension to microphones with arbitrary first-order directivity

Spatial correlation and coherence functions in reverberant sound fields are relevant to the acoustics of enclosed spaces and related areas. Theoretical expressions for the spatial correlation and coherence functions between signals representing the pressure and/or the components of the particle velocity vector in a reverberant sound field are established in the literature and most of these have also been corroborated with measurements [F. Jacobsen, J. Acoust. Soc. Am. 108, 204-210 (2000)]. In the present paper, these expressions are generalized to microphones of first-order directivity, whereby the directivity can be expressed in terms of pressure and pressure gradient. It is shown that the resulting spatial correlation and coherence functions can be expressed in terms of the established spatial correlation and coherence functions. The derived theoretical expression for the spatial coherence function is validated with a modeled diffuse sound field. Further, it is compared with the experimental coherence obtained from the reverberant tails of room impulse responses measured with two common surround sound microphone setups in a concert and a lecture hall.     

The effects of non-cardioid directivity on incidence angle estimation using the polar energy time curve

Assessment of desirable reflections and control of undesirable reflections in rooms are best accomplished if the reflecting surfaces are properly localized. Several measurement techniques exist to identify the incident direction of reflected sound, including the useful polar energy time curve (Polar ETC), which requires six cardioid impulse response measurements along the Cartesian axes. The purpose of this investigation is to quantify the incidence angle estimation error introduced into the Polar ETC by non-cardioid microphone directivities. The results demonstrate that errors may be minimized with a cardioid-family microphone possessing a certain range of directivities and by maximizing the measurement signal-to-noise ratio.     

光纤麦克风传感探头设计的理论分析

鉴于光纤传感器具有体积小、结构简单、灵敏度高、抗电磁干扰且光纤本身的低损耗、耐腐蚀和安全可靠等优良特性,将光纤传感器应用到麦克风中可使麦克风的体积显著降低而灵敏度和抗电磁干扰性显著提高。提出一种新型反射式光纤麦克风的设计方法,采用Y形单根多模反射式光纤传感探头结构形式,给出了光纤麦克风的系统框图。由光纤出射光场分布理论出发,给出弹性膜片在声场作用下,中心形变大小h与出射光光强调制函数I(Z″)的关系。根据实际应用中有关参数的取值,用Matlab软件模拟出膜片形变后变形量h与光强调制函数I(Z″)的关系曲线。     

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。      

光纤microphone的理论与实验研究

本文提出了一种新型的反射式光纤ｍｉｃｒｏｐｈｏｎｅ ,它把反射式光纤传感探头应用于传统的麦克风上,来实现对声波的调制．本文从理论和实验两方面给出了反射式光纤ｍｉｃｒｏｐｈｏｎｅ的光强调制函数,并对反射式光纤ｍｉｃｒｏｐｈｏｎｅ系统进行了研究．     

Uncertainty of array-based measurement of radiated and absorbed sound intensity

Patch near-field acoustic holography (NAH) coupled with an array of sound intensity probes allows separating the sound field incident on a surface from the one radiated by the surface itself. Although the measurement principle has been successfully used to separate the noise source contribution from disturbing sources and/or noise reflections, the method accuracy has not been investigated in the literature. We describe the results of experiments meant to evaluate the uncertainty in the identification of noise radiated by vibrating panels with different absorption characteristics in presence of an incident acoustic radiation using the statistically optimized near-field acoustic holography. Measurement errors were evaluated through tests performed in controlled acoustic conditions. Results evidenced that the measurement uncertainty depends on the accuracy of the microphone array positioning and on the incident sound field. These conclusions were in agreement with the results obtained by simulations in the phase of instrument optimization.     

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.     

A STUDY OF VARIOUS BARRIERS IN ENCLOSED SOUND FIELD BY USING THE COMPUTER PROGRAM—SOFIS

In the study of the behaviors of barriers in an enclosed field, one should take into account such phenomena as sound energy reflection, absorption, scattering and diffraction. Therefore, the study is much more difficult than that in free field. In this paper, sound barriers are classified into four kinds according to their size, number and shape. Each kind of barriers is modelled by a corresponding method based on a computer program—SOFIS. The program combines the ray-tracing technique and statistical method. The impulse response and some acoustical parameters such as sound pressure level at different positions can be calculated by the program, no matter there are a certain kind of barriers in the field or the field is empty. The ray-tracing program and the algorithms for various barriers are validated by the comparison between measurement and prediction of the reverberation room and the anechoic room of the Northwestern Polytechnic University.     

Regularization for improving the deconvolution in real-time near-field acoustic holography

Near-field acoustic holography is a measuring process for locating and characterizing stationary sound sources from measurements made by a microphone array in the near-field of the acoustic source plane. A technique called real-time near-field acoustic holography (RT-NAH) has been introduced to extend this method in the case of nonstationary sources. This technique is based on a formulation which describes the propagation of time-dependent sound pressure signals on a forward plane using a convolution product with an impulse response in the time-wavenumber domain. Thus the backward propagation of the pressure field is obtained by deconvolution. Taking the evanescent waves into account in RT-NAH improves the spatial resolution of the solution but makes the deconvolution problem "ill-posed" and often yields inappropriate solutions. The purpose of this paper is to focus on solving this deconvolution problem. Two deconvolution methods are compared: one uses a singular value decomposition and a standard Tikhonov regularization and the other one is based on optimum Wiener filtering. A simulation involving monopoles driven by nonstationary signals demonstrates, by means of objective indicators, the accuracy of the time-dependent reconstructed sound field. The results highlight the advantage of using regularization and particularly in the presence of measurement noise.     

Sound image method and sound energy vortex in rectangular room

1.TntroductionMostprob1emsofsolvingsoundficlds,suchassoundpressuredistributionsinroomsareclassiia1.lllIn10wfrequencyrange,wavemotionmethodisapplied.Inthemethod,thenorma1modesofroomaresolvedandthensummeduptocvaluatetheirunknownamplitudesbymatchingsoundsource.Inhighfrequencyrange,thenumberofnormalmodesneedcdisnumer0usandtogetfina1solutionwouldbeimpossiblebecauseofthetimeconsumptionincomputation.Hence,ther0omacousticsbasedongeometri-calacousticsandthestatisticswasdevcloped.Thewavemotionmethodcan…     

Feasibility of subjective speech intelligibility assessment based on auralization

Subjective speech intelligibility can be assessed by speech recorded in an anechoic chamber and then convolved with room impulse responses that can be created by acoustic simulation. The speech intelligibility (SI) assessment based on auralization was validated in three rooms. The articulation scores obtained from simulated sound field were compared with the ones from measured sound field and from direct listening in rooms. Results show that the speech intelligibility prediction based on auralization technique with simulated binaural room impulse responses (BRIRs) is in agreement with reality and results from measured BRIRs. When this technique is used with simulated and measured monaural room impulse responses (MRIRs), the predicted results underestimate the reality. It has been shown that auralization technique with simulated BRIRs is capable of assessing subjective speech intelligibility of listening positions in the room.     

Wavelet packet based analysis of sound fields in rooms using coincident microphone arrays

This paper presents a passive analysis method for determining the spatio-temporal characteristics of sound fields in small rooms. The analysis finds an approximate directional reflectogram (ADR) which reveals the approximate arrival directions, time delays and amplitudes of the direct sound and early reflections without using a special or known sound source. A coincident microphone array is used to obtain directional recordings. The recordings are analysed by wavelet packet decomposition to determine the direction of the sound source and select wavelet packet coefficients to reconstruct the estimate of the direct sound. ADR is then computed via deconvolution using this estimate. Experiments have been carried out using synthesized recordings that were obtained from actual room impulse responses measured in two rooms for various source locations. The method estimates the source direction with a mean absolute error of about 7°. Calculated ADRs provide a good estimate of the time delays and arrival directions of acoustical reflections, whereas the amplitudes differ slightly.     

基于声全息方法和双目视觉实现运动声源声场可视化

运动声源声场的可视化是一种重要的运动声源定位的技术手段,利用双目视觉测量技术实现运动声源声场空间的自动测量,自动确定运动声源表面的空间位置,针对声源表面,利用传声器阵列,基于声全息方法实现运动声源声场的重建,建立视频图像与声场的空间映射,并建立视频与声场之间的时序,实现实景视频图像与声场重建结果的融合,可以自动生成声源运动过程的视频。基于该方法所开发了一套试验测量系统,对运动声源的测量试验结果表明,该方法可以有效实现运动声源的视频可视化,使人可以直接从视频中看到声源及其变化过程,使声源的定位和识别变得更加简单。