Acoustic analysis by spherical microphone array processing of room impulse responses

Spherical microphone arrays have been recently used for room acoustics analysis, to detect the direction-of-arrival of early room reflections, and compute directional room impulse responses and other spatial room acoustics parameters. Previous works presented methods for room acoustics analysis using spherical arrays that are based on beamforming, e.g., delay-and-sum, regular beamforming, and Dolph-Chebyshev beamforming. Although beamforming methods provide useful directional selectivity, optimal array processing methods can provide enhanced performance. However, these algorithms require an array cross-spectrum matrix with a full rank, while array data based on room impulse responses may not satisfy this condition due to the single frame data. This paper presents a smoothing technique for the cross-spectrum matrix in the frequency domain, designed for spherical microphone arrays, that can solve the problem of low rank when using room impulse response data, therefore facilitating the use of optimal array processing methods. Frequency smoothing is shown to be performed effectively using spherical arrays, due to the decoupling of frequency and angular components in the spherical harmonics domain. Experimental study with data measured in a real auditorium illustrates the performance of optimal array processing methods such as MUSIC and MVDR compared to beamforming.     

Reverberation time measurement by the product of two room impulse responses

The decay function for the evaluation of the reverberation time is often obtained by the method of the backward integration of a squared room impulse response as suggested by M.R. Schroeder more than four decades ago. Since then much work has been published about its implementation. However, soon after the initial exploitation of the method, it was realized that the effects of the background noise contaminating the room impulse response required a careful consideration for accomplishing better results.This paper describes an alternative method dealing with the problem of the backward integration of noisy room impulse responses. This method is based on the processing of two impulse responses sequentially recorded for a fixed source and receiver arrangement in a room. Statistical criteria are proposed to identify how the effect of the noise corrupts the level decay curve using a noise-free synthesized room impulse response as well as measurements performed in a real room.     

Measurement of sound intensity using a single moving microphone

A method for measuring sound intensity by using a single moving microphone is proposed. Experiments that confirm the validity of this method are reported and the advantages and disadvantages of the method are discussed.     

室内脉冲声场中声强的测量研究

采用多次重复发射的声源配合装置在缓慢转动平台上的接收传声器所组成的测量系统(简称为RRS),它等效于多个完全匹配的传声器组成的圆型阵列。文中着重阐述了借助此技术测量房间内脉冲声场中声强的时间与空间分布特性。阐明了以RRS测量入射脉冲声强的理论基础,分别对离散的脉冲声场及扩散声场提出了测量声强的定量关系式,对实验结果及应用作了分析与讨论。     

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.     

Localizing nearby sound sources in a classroom: binaural room impulse responses

Binaural room impulse responses (BRIRs) were measured in a classroom for sources at different azimuths and distances (up to 1 m) relative to a manikin located in four positions in a classroom. When the listener is far from all walls, reverberant energy distorts signal magnitude and phase independently at each frequency, altering monaural spectral cues, interaural phase differences, and interaural level differences. For the tested conditions, systematic distortion (comb-filtering) from an early intense reflection is only evident when a listener is very close to a wall, and then only in the ear facing the wall. Especially for a nearby source, interaural cues grow less reliable with increasing source laterality and monaural spectral cues are less reliable in the ear farther from the sound source. Reverberation reduces the magnitude of interaural level differences at all frequencies; however, the direct-sound interaural time difference can still be recovered from the BRIRs measured in these experiments. Results suggest that bias and variability in sound localization behavior may vary systematically with listener location in a room as well as source location relative to the listener, even for nearby sources where there is relatively little reverberant energy.     

Relationship between impulse response and other types of room acoustical responses

This paper presents a method of calculating sound build up, steady state and sound reduction phenomena from the impulse response of rooms. The noise components of both the testing signal and the room response are omitted and wave phenomena occurring in the room are also neglected. A situation corresponding to the geometrical propagation of sound is thus simulated. The resulting formulae are an extension of corresponding methods for the numerical modelling of acoustical fields in rooms. In this way, as well as the impulse response, sound build up and reverberation curves may also be obtained. An example using the ray tracing technique is presented.     

The fast measurement of head related impulse responses for all azimuthal directions using the continuous measurement method with a servo-swiveled chair

This paper proposes a method for the fast measurement of head related impulse responses (HRIRs) for all azimuthal directions using the continuous measurement method with a subject sitting on a servo-swiveled chair (SSC). The subject faces all azimuthal directions with a constant angular speed as the SSC rotates and hears corresponding spatial sounds. In our continuous measurement method, we use a maximum length sequence (MLS) as the excitation signal, record binaural signals using tiny microphones placed in the subject’s ear canal around 5 mm from the entrance, and feed them, along with the MLS signal, into a PC so that HRIRs can be extracted for all azimuthal directions and discrete elevations. In this paper, the underlying principles behind the method are described and conditions for successful implementation are examined through numerical simulations. Further, the validity of the proposed method is verified using a physical realization and empirical measurements.     

Beamforming correction for dipole measurement using two-dimensional microphone arrays

In this paper, a beamforming correction for identifying dipole sources by means of phased microphone array measurements is presented and implemented numerically and experimentally. Conventional beamforming techniques, which are developed for monopole sources, can lead to significant errors when applied to reconstruct dipole sources. A previous correction technique to microphone signals is extended to account for both source location and source power for two-dimensional microphone arrays. The new dipole-beamforming algorithm is developed by modifying the basic source definition used for beamforming. This technique improves the previous signal correction method and yields a beamformer applicable to sources which are suspected to be dipole in nature. Numerical simulations are performed, which validate the capability of this beamformer to recover ideal dipole sources. The beamforming correction is applied to the identification of realistic aeolian-tone dipoles and shows an improvement of array performance on estimating dipole source powers.     

A comparison of measured room acoustics metrics using a spherical microphone array and conventional methods

The traditional microphone configuration used to measure room impulse responses (IRs) according to ISO 3382:2009 is an omnidirectional and figure-8 microphone pair. IRs measurements were taken in a 2500-seat auditorium to determine how the results from a spherical microphone array (an mh acoustics Eigenmike-em32) compare to those from the traditional microphone setup (a Brüel & Kjær Type-4192 omnidirectional microphone and a Sennheiser MKH30 figure-8 microphone). Measurements were obtained at six receiver locations, with three repetitions each in order to first evaluate repeatability. The metrics considered in this study were: reverberation time (T30), early decay time (EDT), clarity index (C80), strength (G), lateral energy fraction (J_LF) and late lateral energy level (L_J). Before calculating these quantities, the IRs were filtered to equalize the frequency response of the microphones and sound source. For the spherical array measurements, the omnidirectional (monopole) and figure-8 (dipole) patterns were extracted using beamforming. In terms of repeatability, the average standard deviation of the three measurements at each receiver location averaged across all metrics, receivers, and octave bands was found to be 0.01 just noticeable differences (JNDs). The analysis comparing the measurements from the two microphone configurations yielded differences which were less than 1 JND for the majority of metrics, with a few exceptions of EDT and C80 slightly above 1 JND. Based on this case study, these results indicate that spherical microphone arrays can be used to obtain valid room IR measurements, which will allow for the development of new metrics utilizing the higher spatial resolution made possible with spherical arrays.     

基于三元伪随机序列集的多声源房间脉冲响应测量方法*

在多输入多输出系统中,诸多情况下需并行测量任意输入输出之间的脉冲响应。虽然利用以Kasami序列集为代表的二元伪随机序列集良好的相关特性可以实现多声源并行测量,但是囿于其编解码信号相同且为二元信号的特点,使得其相关特性仍存在较大的可优化空间。该文提供了一种基于三元伪随机序列集的多声源房间脉冲响应测量方法,具有更好的相关特性,提高了测量准确度。仿真和实验验证了该测量方法的可行性和准确性。在封闭空间几何模型构建等领域中,可使用该方法提高测量效率并降低由于各通道不同步导致的误差。     

Wind turbine noise measurement using a compact microphone array with advanced deconvolution algorithms

This paper experimentally investigates the noise from a large wind turbine (GE 1.5 MW) with a compact microphone array (OptiNav 24) using advanced deconvolution based beamforming methods, such as DAMAS and CLEAN-SC beamforming algorithms, for data reduction. Our study focuses on the ability of a compact microphone array to successfully locate both mechanical and aerodynamic noise sources on the wind turbine. Several interesting results have emerged from this study: (i) A compact microphone array is sufficient to perform a detailed study on wind turbine noise if advanced deconvolution methods are applied. (ii) Noise sources on the blade and on the nacelle can clearly be separated. (iii) Noise of the blades is dominated by trailing edge noise which is frequency dependent and is distributed along the length of the blade with the dominant noise source closer to the tip of the blade. (iv) The LP and DAMAS algorithms represent the distributed trailing edge noise source better than CLEAN-SC and classical beamforming. (v) Additional tonal noise produced during yawing operation is believed to be radiating from the tower of the wind turbine that acts like a resonator. (vi) Ground reflection is not believed to have a significant effect on noise source location estimates in this study.     

Subjective and objective verifications of the inverse functions of binaural room impulse responses

The binaural room impulse responses (BRIRs) can be applied to 3-D sound field reconstruction, virtual reality, noise control, et al. Because the BRIRs are non-minimum phase functions, it is difficult to find the exact inverse functions of the BRIRs, especially when there are two or more sources in a reverberant space. In this work, a method was proposed to find the inverse functions of BRIRs with two sound sources in a reverberant space. The concept of time delays and the method of weighted least squares were used to find the causal, however, approximate inverse functions. The accuracy of the inverse functions was first evaluated objectively by a dummy head system. The result shows that the distortion due to crosstalk and room reverberation can be improved by 16∼18 dB. The inverse functions were also verified subjectively by 20 students. The result of subjective evaluation also shows that the inverse functions can be used successfully to reduce the crosstalk effect and the room reverberation.     

Reliability of estimating the room volume from a single room impulse response

The methods investigated for the room volume estimation are based on geometrical acoustics, eigenmode, and diffuse field models and no data other than the room impulse response are available. The measurements include several receiver positions in a total of 12 rooms of vastly different sizes and acoustic characteristics. The limitations in identifying the pivotal specular reflections of the geometrical acoustics model in measured room impulse responses are examined both theoretically and experimentally. The eigenmode method uses the theoretical expression for the Schroeder frequency and the difficulty of accurately estimating this frequency from the varying statistics of the room transfer function is highlighted. Reliable results are only obtained with the diffuse field model and a part of the observed variance in the experimental results is explained by theoretical expressions for the standard deviation of the reverberant sound pressure and the reverberation time. The limitations due to source and receiver directivity are discussed and a simple volume estimation method based on an approximate relationship with the reverberation time is also presented.     

Prediction of energy decay in room impulse responses simulated with an image-source model

A method is proposed that provides an approximation of the acoustic energy decay (energy-time curve) in room impulse responses generated using the image-source technique. A geometrical analysis of the image-source principle leads to a closed-form expression describing the energy decay curve, with the resulting formula being valid for a uniform as well as nonuniform definition of the enclosure's six absorption coefficients. The accuracy of the proposed approximation is demonstrated on the basis of impulse-response simulations involving various room sizes and reverberation levels, with uniform and nonuniform sound absorption coefficients. An application example for the proposed method is illustrated by considering the task of predicting an enclosure's reflection coefficients in order to achieve a specific reverberation level. The technique presented in this work enables designers to undertake a preliminary analysis of a simulated reverberant environment without the need for time-consuming image-method simulations.     

Dynamic responses of axially moving viscoelastic beam under a randomly disordered periodic excitation

We investigate dynamic responses of axially moving viscoelastic beam subject to a randomly disordered periodic excitation. The method of multiple scales is used to derive the analytical expression of first-order uniform expansion of the solution. Based on the largest Lyapunov exponent, the almost sure stability of the trivial steady-state solution is examined. Meanwhile, we obtain the first-order and the second-order steady-state moments for the non-trivial steady-state solutions. Specially, we discuss the first mode theoretically and numerically. Results show that under the same conditions of the parameters, as the intensity of the random excitation increases, non-trivial steady-state solution fluctuation will become strenuous, which will result in the non-trivial steady-state solution lose stability and the trivial steady-state solution can be a possible. In the case of parametric principal resonance, the stochastic jump is observed for the first mode, which indicates that the stationary joint probability density concentrates at the non-trivial solution branch when the random excitation is small, but with the increase of intensity of the random excitation, the probability of the trivial steady-state solution will become larger. This phenomenon of stochastic jump can be defined as a stochastic bifurcation.     

Source excitation strategies for obtaining impulse responses in finite difference time domain room acoustics simulation

This paper considers source excitation strategies in finite difference time domain room acoustics simulations for auralization purposes. We demonstrate that FDTD simulations can be conducted to obtain impulse responses based on unit impulse excitation, this being the shortest, simplest and most efficiently implemented signal that might be applied. Single, rather than double, precision accuracy simulations might be implemented where memory use is critical but the consequence is a remarkably increased noise floor. Hard source excitation introduces a discontinuity in the simulated acoustic field resulting in a shift of resonant modes from expected values. Additive sources do not introduce such discontinuities, but instead result in a broadband offset across the frequency spectrum. Transparent sources address both of these issues and with unit impulse excitation the calculation of the compensation filters required to implement transparency is also simplified. However, both transparent and additive source excitation demonstrate solution growth problems for a bounded space. Any of these approaches might be used if the consequences are understood and compensated for, however, for room acoustics simulation the hard source is the least favorable due to the fundamental changes it imparts on the underlying geometry. These methods are further tested through the implementation of a directional sound source based on multiple omnidirectional point sources.     

A new definition of boundary point between early reflections and late reverberation in room impulse responses

The early reflections in the room impulse response are usually defined as those observed within the initial 80 ms after the arrival of the direct sound, after which time the sound field is called reverberant. This number was chosen from measurements of other functions in a limited number of halls. In order to give an objective foundation to this time separation and to establish a physical indicator for it, a new method is proposed that defines a "transition time t(L)," which is the time at which the energy correlation between the direct plus initial sound and the subsequent decaying sound first achieves a specified low value. For various halls this number is shown and its relevance as a new parameter is discussed.     

Profile measurement of a moving object using an improved projection grating phase-shifting profilometry

3-D profile measurement of a moving object using a novel phase-shifting technique is introduced. Digital gratings with two steps phase-shifting are projected periodically onto a measured object surface. The deformed fringe patterns are captured by a frame CCD camera within a short exposure time. By synchronizing the projector and the CCD camera accurately, there is an overlapping part which is the same part of the object among three neighbouring frames. The length of an overlapping part can be controlled as one third of a frame length. Hence the intensity values at the same surface point modulated by three neighbouring gratings can be obtained, and its phase value can be computed by an improved phase-extracting algorithm. The profile of a specimen is detected by the proposed method. Experimental results demonstrate that this method is effective for the profile measurement of a moving object.