An image source computer model for room acoustics analysis and electroacoustic simulation

A computer program for the determination of the impulse response of rooms, using the mirror image method, has been developed. It is intended for acoustical planning of auditoria and the simulation of sound fields. The computer also generates the necessary information for the automatic mixer, delay unit and reverberation unit, which makes it possible to change sound field simulations in a short time.The program allows arbitrary room shapes to be analyzed by representing curved surfaces with plane approximations. The program also calculates energy time gaps, radiation angles and angles of incidence for the investigated source and receiver positions. The absorption coefficients of the reflecting surfaces are also taken into account. The source and the receiver may be positioned anywhere in the room. Compared with other programs described in the literature, this program yields more information, necessary particularly for electroacoustic room acoustics simulation and acoustical planning.     

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.     

Perceptual validation of virtual room acoustics: Sound localisation and speech understanding

The reliability of algorithms for room acoustic simulations has often been confirmed on the basis of the verification of predicted room acoustical parameters. This paper presents a complementary perceptual validation procedure consisting of two experiments, respectively dealing with speech intelligibility, and with sound source front–back localisation.The evaluated simulation algorithm, implemented in software ODEON®, is a hybrid method that is based on an image source algorithm for the prediction of early sound reflection and on ray-tracing for the later part, using a stochastic scattering process with secondary sources. The binaural room impulse response (BRIR) is calculated from a simulated room impulse response where information about the arriving time, intensity and spatial direction of each sound reflection is collected and convolved with a measured Head Related Transfer Function (HRTF). The listening stimuli for the speech intelligibility and localisation tests are auralised convolutions of anechoic sound samples with measured and simulated BRIRs.Perception tests were performed with human subjects in two acoustical environments, i.e. an anechoic and reverberant room, by presenting the stimuli to subjects in a natural way, and via headphones by using two non-individualized HRTFs (artificial head and hearing aids placed on the ears of the artificial head) of both a simulated and a real room.Very good correspondence is found between the results obtained with simulated and measured BRIRs, both for speech intelligibility in the presence of noise and for sound source localisation tests. In the anechoic room an increase in speech intelligibility is observed when noise and signal are presented from sources located at different angles. This improvement is not so evident in the reverberant room, with the sound sources at 1-m distance from the listener. Interestingly, the performance of people for front–back localisation is better in the reverberant room than in the anechoic room.The correlation between people’s ability for sound source localisation on one hand, and their ability for recognition of binaurally received speech in reverberation on the other hand, is found to be weak.     

On the importance of early reflections for speech in rooms

This paper presents the results of new studies based on speech intelligibility tests in simulated sound fields and analyses of impulse response measurements in rooms used for speech communication. The speech intelligibility test results confirm the importance of early reflections for achieving good conditions for speech in rooms. The addition of early reflections increased the effective signal-to-noise ratio and related speech intelligibility scores for both impaired and nonimpaired listeners. The new results also show that for common conditions where the direct sound is reduced, it is only possible to understand speech because of the presence of early reflections. Analyses of measured impulse responses in rooms intended for speech show that early reflections can increase the effective signal-to-noise ratio by up to 9 dB. A room acoustics computer model is used to demonstrate that the relative importance of early reflections can be influenced by the room acoustics design.     

The evaluation of binaural playback systems for virtual sound fields

By means of virtual acoustics technologies the impulse response in a location of a real enclosure can be used to create a virtual room. Also in this virtual space, typically implemented in a dead room, an impulse response can be measured. From both “real” and respective “virtual” impulse response a group of room-acoustical parameters can be calculated. The match of the two groups of indicators mostly depends on the layout of the playback system and on the implemented processing. In this paper a procedure of virtual measurement in a dead room is developed and some typical layouts of playback systems employing two loudspeakers are compared. Based on the consistency of the room-acoustical indicators derived from “real” and “virtual” impulse responses can be investigated. It is shown how the performance depends on the angular separation of loudspeakers and the efficiency of the cross-talk is also tested and discussed. Finally, a set-up which is suitable for psychoacoustics tests is indicated.     

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.     

An analytical derivation of the optimum source patterns for the pseudospectral time-domain method

In the computational electromagnetics and acoustics, spatially smoothed sources are often utilized to alleviate the aliasing errors in the pseudospectral time-domain (PSTD) algorithms. In our work, an analytical derivation of the optimum source patterns is presented according to the accurately derived expressions of the dominant source-introduced aliasing errors according to the circular discrete convolution and Tailor series expansion method. We quantitatively demonstrate, for the first time in literature, that the aliasing errors can be optimally suppressed and rapidly reduced to the negligible levels by these optimum patterns and with the increment of source cells. We also provide the different implementation schemes of the optimal patterns both for the soft and hard source cases. The numerical calculation and 1D PSTD transient simulations are conducted to verify the excellent performance of these optimum sources.     

室内断续噪声的有源控制

在室内一角（或声压极大处）装一传声器-放大器-扬声器系统,用简正波抑制的原理降低全室混响声的有源噪声控制系统已证明是有效方法,其作用理论和效果也已明确.但过去只考虑了稳定噪声。噪声完全稳定的比较少,即使是稳定噪声也有起始和终止的时候,噪声的变化要激发有源噪声控制系统的瞬态效应,这对于“系统”的使用可能有不利影响.为了充分发挥“系统”的作用,对它的瞬态特性有确切了解的必要。本文目的就是阐明室内有源噪声控制系统的瞬态特性及其对于室内音质的影响。     

Comparison of different experimental methods for the assessment of the room’s acoustics

The paper presents the acoustics analysis of three different enclosed spaces. These spaces (rooms) have different geometrical shapes and sizes and serve for different purposes. The early decay time, reverberation time, clarity and center time are evaluated with Dirac, WinMLS, Aurora and Caracad software using simple, low-cost equipment. The listed acoustic parameters were determined using linear sine sweep and impulsive sources. Comparisons between experimental measurements, simulations and analytic results were done. The room impulse response measurement proved to be the most reliable method to evaluate the properties of different rooms even if the measurements are perturbed by non-idealities of the low-cost equipment.     

An explicit time-domain finite element method for room acoustics simulations: Comparison of the performance with implicit methods

This paper presents the applicability of an explicit time-domain finite element method (TD-FEM) using a dispersion reduction technique called modified integration rules (MIR) on room acoustics simulations with a frequency-independent finite impedance boundary. First, a dispersion error analysis and a stability analysis are performed to derive the dispersion relation and the stability condition of the present explicit TD-FEM for three-dimensional room acoustics simulations with an infinite impedance boundary. Secondly, the accuracy and efficiency of the explicit TD-FEM are presented by comparing with implicit TD-FEM using MIR through room acoustics simulations in a rectangular room with infinite impedance boundaries. Thirdly, the stability condition of the explicit TD-FEM is investigated numerically in the case with finite impedance boundaries. Finally, the performance of the explicit TD-FEM in room acoustics simulations with finite impedance boundaries is demonstrated in a comparison with the implicit TD-FEM. Although the stability of the present explicit TD-FEM is dependent on the impedance values given at boundaries, the explicit TD-FEM is computationally more efficient than the implicit method from the perspective of computational time for acoustics simulations of a room with larger impedance values at boundaries.     

Robustness of multichannel equalization in an acoustic reverberant environment

A theoretical framework is established, for the robustness of multichannel sound equalization in reverberant environments. Using results from statistical room acoustics, a closed-form expression is derived that predicts the degradation in performance of an equalization system as the sound source moves from its nominal position inside the enclosure. The presented analysis also provides means of identifying the performance bounds that can be expected when using such a system in an actual room. Using extensive computer simulations, the effect of physical parameters such as the relative positions of the source and the receivers, as well as effects of different design parameters are investigated. Based on the conditions imposed by these parameters, it is shown that, depending on the array geometry and the exact form of the equalizers, slight performance gains can be expected as the number of receivers is increased.     

Source implementation to eliminate low-frequency artifacts in finite difference time domain room acoustic simulation

The finite difference time domain (FDTD) method is a numerical technique that is straight forward to implement for the simulation of acoustic propagation. For room acoustics applications, the implementation of efficient source excitation and frequency dependent boundary conditions on arbitrary geometry can be seen as two of the most significant problems. This paper deals with the source implementation problem. Among existing source implementation methods, the hard source implementation is the simplest and computationally most efficient. Unfortunately, it generates a large low-frequency modulation in the measured time response. This paper presents a detailed investigation into these side effects. Surprisingly, some of these side effects are found to exist even if a transparent source implementation is used. By combing a time limited approach with a class of more natural source pulse function, this paper develops a source implementation method in FDTD that is as simple and computationally as efficient as a hard source implementation and yet capable of producing results that are virtually the same as a true transparent source. It is believed that the source implementation method developed in this paper will provide an improvement to the practical usability of the FDTD method for room acoustic simulation.     

Computation of edge diffraction for more accurate room acoustics auralization

Inaccuracies in computation and auralization of room impulse responses are related in part to inadequate modeling of edge diffraction, i.e., the scattering from edges of finite surfaces. A validated time-domain model (based on analytical extensions to the Biot-Tolstoy-Medwin technique) is thus employed here to compute early room impulse responses with edge diffraction. Furthermore, the computations are extended to include combinations of specular and diffracted paths in the example problem of a stage-house. These combinations constitute a significant component of the total nonspecular scattering and also help to identify edge diffraction in measured impulse responses. The computed impulse responses are then convolved with anechoic signals with a variety of time-frequency characteristics. Initial listening tests with varying orders and combinations of diffraction suggest that (1) depending on the input signal, the diffraction contributions can be clearly audible even in nonshadow zones for this conservative open geometry and (2) second-order diffraction to nonshadowed receivers can often be neglected. Finally, a practical implementation for binaural simulation is proposed, based on the singular behavior of edge diffraction along the least-time path for a given source-edge-receiver orientation. This study thus provides a first major step toward computing edge diffraction for more accurate room acoustics auralization.     

Estimating the mixing time of concert halls using the eXtensible Fourier Transform

In room acoustics, we measure room impulse responses (RIRs) in order to fully describe the hall. RIRs are composed of a succession of arrivals (i.e., some sound rays which have undergone one or more reflections on their way from the source to the receiver). We propose the eXtensible Fourier Transform (XFT) in order to investigate the time evolution of spectral components of RIRs. The phase evolution versus time allows to estimate the mixing time, which is defined as the time it takes for initially adjacent sound rays to spread uniformly across the room. After presenting some properties of the XFT, we show why one must compensate the natural energy decay of the RIR in order to obtain stationary signals. We estimate mixing times for a set of experimental RIRs and several that are synthesized using a stochastic model. Then, we estimate the dependance of mixing time upon the source/receiver distance in all these RIRs. Results are consistent up to the lack of reproducibility of the sound sources, but are strongly dependent on some parameters used for computing the XFT. We finally discuss the relevance of the name mixing time with respect to the theory and in regard to the time we estimate, that we propose to call cross-over time.     

Active control of intermittent noise in a room

I.IntroductionThereverberantnoisesmaybereducedwithanactivenoisecontrol(ANC)systcmformedbyacornermicrophonc-amplifier-1oudspcakersystemina.oom.lllThismeth-odbascdonthenormalmodccancelingprincip1chasalreadybeenclarifiedbothinthco-ryandexperimcnts,andthclimitationandpotcntialityofitsapp1icationalsodiscussedinapreviouspaper.Thediscussionsinthepastlimitedmost1ytothesteadyconditionofnoiscfield,andthisisextendedtothetransientcondihoninthcprcsentwork,viz.thein-llucnceofthecornerANCsystemwhenthenois…     

A simple method to detect audible echoes in room acoustical design

A simple method to detect audible echoes is proposed as an objective criterion for room acoustics. This method evaluates the perceptibility of sound reflections that are generated by an impulsive sound source and identifies from reflectograms harmful reflections perceived as echoes. Particularly with this method, the masking effect of reverberation is taken into consideration, which cannot be treated sufficiently by the existing objective criteria. The applicability to room acoustical design is verified by evaluating the impulse responses measured in real halls where audible echoes occurred. It is shown that the proposed method detects audible echoes at an accuracy of more than 90% and would be suitable for practical use.     

Impedance boundary conditions for pseudo-spectral time-domain methods in room acoustics

The pseudo-spectral time-domain (PSTD) method is an alternative time-marching method to classical leapfrog finite difference schemes in the simulation of wave-like propagating phenomena. It is based on the fundamentals of the Fourier transform to compute the spatial derivatives of hyperbolic differential equations. Therefore, it results in an isotropic operator that can be implemented in an efficient way for room acoustics simulations. However, one of the first issues to be solved consists on modeling wall absorption. Unfortunately, there are no references in the technical literature concerning to that problem. In this paper, assuming real and constant locally reacting impedances, several proposals to overcome this problem are presented, validated and compared to analytical solutions in different scenarios.     

双随机相位加密系统的选择明文攻击

在光学信息安全领域,双随机相位加密方法最引人注目并得到广泛研究,但由于双随机相位加密系统是基于傅里叶变换的系统,其本质上是一种线性变换系统,明文、密文之间的函数依赖关系比较简单,这就为其安全性留下了很大的隐患。双随机相位加密方法可以用光学和数字的方式实现,提出了一种选择明文攻击的方法,利用多个冲击函数作为选择的明文,成功破解了基于数字方法实现的双随机相位加密系统,并给出了恢复密钥的解析式,此方法最大的优点在于解密图像的无损性,并从理论上加以证明,给出了实验结果。     

Using cross correlations of turbulent flow-induced ambient vibrations to estimate the structural impulse response. Application to structural health monitoring

It has been demonstrated theoretically and experimentally that an estimate of the impulse response (or Green's function) between two receivers can be obtained from the cross correlation of diffuse wave fields at these two receivers in various environments and frequency ranges: ultrasonics, civil engineering, underwater acoustics, and seismology. This result provides a means for structural monitoring using ambient structure-borne noise only, without the use of active sources. This paper presents experimental results obtained from flow-induced random vibration data recorded by pairs of accelerometers mounted within a flat plate or hydrofoil in the test section of the U.S. Navy's William B. Morgan Large Cavitation Channel. The experiments were conducted at high Reynolds number (Re > 50 million) with the primary excitation source being turbulent boundary layer pressure fluctuations on the upper and lower surfaces of the plate or foil. Identical deterministic time signatures emerge from the noise cross-correlation function computed via robust and simple processing of noise measured on different days by a pair of passive sensors. These time signatures are used to determine and/or monitor the structural response of the test models from a few hundred to a few thousand Hertz.     

Room temperature stable single-photon source

We report on the realization of a stable solid state room temperature source for single photons. It is based on the fluorescence of a single nitrogen-vacancy (NV) color center in a diamond nanocrystal. Antibunching has been observed in the fluorescence light under both continuous and pulsed excitation. Our source delivers 2×10⁴ s^-1 single-photon pulses at an excitation repetition rate of 10 MHz. The number of two-photon pulses is reduced by a factor of five compared to strongly attenuated coherent sources. Received 1st August 2001 and Received in final form 2 October 2001