Effect of listening level and background noise on the subjective decay rate of room impulse responses: Using time-varying loudness to model reverberance

Room impulse responses (RIRs) are used very widely to characterize the acoustic conditions of rooms, such as in the derivation of reverberation time, early decay time and clarity index. This study investigates the subjective decay rate (or reverberance) of RIRs when directly listened to (rather than convolved with a dry signal such as speech or music). Through a subjective experiment, it investigates the effects of gain (or listening level) and background noise level on the reverberance of RIRs that had been measured in three concert auditoria. The task of the experiment was to match the decay rate of RIRs to that of a reference RIR by ear, by adjusting the RIRs’ exponential decay rate. Based on objective loudness modeling, gain should have a positive effect on reverberance, and background noise has a negative effect. This is confirmed in the results of the experiment. Furthermore, the objectively calculated loudness decay function provides an effective predictor of subjective decay rate, which performs better than conventional early decay time or reverberation time for the RIRs tested.     

Universal power-law decay of the impulse energy in granular protectors

Protecting a big impulse from outside is one of the important issues of our everyday life. A granular medium is often used as a protecting material. The impulse inside a granular medium is a solitary wave which may be confined temporarily to a particular region of the medium, which we call the granular container that plays the role of the protector. We find a universal power-law behavior in time for the leakage of the impulse energy confined inside various granular containers.     

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.     

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.     

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.     

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.     

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.     

Prediction of noise inside an acoustic cavity of elongated shape using statistical energy analyses with spatial decay consideration

In this paper, Statistical Energy Analysis (SEA) is used to predict the interior noise of an acoustic cavity of elongated shape. The disadvantage of the conventional SEA method, which quantifies the response in terms of the energy averaged over each subsystem, is overcome by introducing a one-dimensional spatial decay relation, through which information about the acoustic energy variation in the elongated direction is taken into account. The modified SEA is experimentally validated using a 1:5 scaled space station prototype, having the longitudinal dimension much larger than the cross-sectional dimension. It is also compared with a model reported in the literature. It is shown that, in the region where the acoustic pressure level decays at a constant rate, the two models agree well with each other and are capable of estimating the acoustic pressure variation along the space station cabin. However, near the end walls where the decay rate of the acoustic pressure level is not constant, the proposed model provides better accuracy.     

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.     

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.     

Sound decay in a rectangular room with impedance walls

The problem of sound decay in a rectangular room is considered for the case of a room with walls the acoustic properties of which are described by the impedance, which implies a dependence of the absorption coefficient on the angle of incidence of sound waves. The ray approximation is used to determine the sound decay laws for different distributions of wall absorption. It is shown that, in a room with impedance walls, the sound decay is slower than in the conventional reverberation model, in which the wall absorption coefficient is independent of the angle of incidence. The problem is also solved in the wave approximation to determine the decay law for a preset frequency band.     

基于声能密度模型的中高频复杂声场预报方法

提出并推导了一种基于声能密度分布方程的声场预报方法。在能量和功率流的本构关系基础上建立声能密度平衡方程。应用直达声场和反射混响声场的叠加原理和边界面散射模型,建立了面向中高频复杂声场细节预报的数值计算方法。通过有限元计算结果在一个简单声场模型上对此方法做了验证,对比结果显示了声能密度法预报有可靠的精度和准确度。     

A nonlinear energy sink with an energy harvester: Transient responses

This paper investigates energy harvesting using nonlinear energy sink. First a novel apparatus is described in detail outlining how the essential nonlinearity and energy harvesting are achieved. Then the system modeling is addressed, including the equations of motion for the mechanical system and the electromechanical system, and a formula for the transduction factor. The experimental identification is conducted to determine several key parameters and relationships. Using the established models, a computer simulation is carried out to investigate the apparatus?s performance under transient responses in terms of vibration absorption and energy harvesting. Finally experiments are conducted to validate the simulation results. It is shown that the system performs well, being capable of energy localization as well as broad band vibration absorption. The system is also shown to be capable of harvesting energy.     

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.     

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.     

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。     

Prediction of the reverberation time in high absorbent room using a modified-diffusion model

A modification of the diffusion model’s boundary condition, based on the Eyring absorption coefficient, to account for high walls absorption is proposed. Numerical comparisons are carried out for three geometrical configurations (a proportionate room, a corridor and a flat enclosure). Comparisons with the statistical theory and a ray-tracing software show that the modified boundary condition increases the accuracy of the diffusion model in term of reverberation time in all the simulated configurations. An experimental comparison in the case of a non-uniformly absorbent room (a reverberation chamber covered with patches of glass wool) is also carried out. The modified-diffusion model results match well with the ray-tracing ones. Both models are in agreement with the experimental data for most of third octave bands (discrepancy close to or below 10%). However, some discrepancies up to 40% can also be observed in a few octave bands, probably due to experimental considerations and to the modal behaviour of the room at low frequencies.     

Measuring short impulse responses with inverse filtered maximum-length sequences

Most loudspeakers have a non-flat frequency response which produces a long oscillating impulse response. An inverse filtering approach may be used to calculate the driving waveform necessary to equalize the response of the loudspeaker in order to radiate shorter acoustic pulses. When combined with the MLS technique, inverse filtering may be used to pre-emphasize the driving signal so that a shorter impulse response, with a prescribed waveform, is measured. This technique is described and illustrated by applying it to a distributed mode loudspeaker. Originally, this loudspeaker has a rather irregular response in a wide band. When the MLS signal is pre-emphasized with the proper inverse filter, a shorter impulse response is measured with a zero-phase cosine-magnitude spectrum.