Sound pressure-level prediction in large rooms containing barriers

A simple-to-use graphical method for estimating the sound pressure level in the shadow zone of a rigid straight-edged barrier for sound radiated from a point source located in a large room is presented and discussed. Also presented is a means based on the graphs introduced for evaluating the noise reduction potential of a barrier in a room or factory work space.     

Localization of distinct reflections in rooms using spherical microphone array eigenbeam processing

This paper presents an experimental and comparative study of several spherical microphone array eigenbeam (EB) processing techniques for localization of early reflections in room acoustic environments, which is a relevant research topic in both audio signal processing and room acoustics. This paper focuses on steered beamformer-based and subspace-based localization techniques implemented in the spherical EB domain, including the plane-wave decomposition, eigenbeam delay and sum, eigenbeam minimum variance distortionless response, eigenbeam multiple signal classification (EB-MUSIC), and eigenbeam estimation of signal parameters via rotational invariance techniques (EB-ESPRIT) methods. The directions of arrival of the original sound source and the associated reflection signals in acoustic environments are estimated from acoustic maps of the rooms, which are obtained using a spherical microphone array. The EB-domain-based frequency smoothing and white noise gain control techniques are derived and employed to improve the performance and robustness of reflection localization. The applicability of the presented methods in practice is confirmed by experiments carried out in real rooms.     

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.     

Measurement of sound absorption coefficients of flat surfaces in a workshop

To improve the acoustic treatment of facings and provide appropriate solutions for noise control at workplace, it is necessary to develop methods of acoustic characterization of the walls in industrial halls. Sound absorption coefficient measurement in industrial rooms is however quite a difficult task because of the partially reverberant conditions. This work describes the measurement of the sound absorption coefficient of flat panels subject to small angle sound incidence, in an industrial hall using an experimental device equipped with an acoustic array. The directivity of this array has been optimized so that the major part of the received acoustic energy would come from one portion only of the investigated facing, this, in turn attenuating the reflected beams due to the reverberation. This new device includes an impulse sound source targeting the panels. The present article focuses mainly on the sound source design and implementation. It also describes some sound absorption measurements carried in a semi-anechoic chamber and in an industrial hall in order to examine the performance of the device. Sound absorption coefficients of several standard liners obtained through this device have been compared to those resulted from the two microphone technique.     

Measurements of the influence of fittings and roof pitch on the sound field in panel-roof factories

Measurements are presented of the sound absorption of two factory machines. Their third octave absorption, between 160 Hz and 5 kHz, is in the range 0.5–2.6 m².Measurements of reverberation time (RT) and/or sound propagation (SP) in four factories with panel roofs, when empty and/or fitted, are also presented. The general characteristics of RT and SP in such factories, and the factors that influence them, are discussed in the light of the results. Fittings reduce the RT and large-distance SP levels in proportion to their volume density and to the absorption of the panel roof. Levels near a source can increase due to backscattering. Preferentially orientated fitting arrangements do not significantly affect the variation of SP with direction in a factory.The effect of roof pitch on factory SP is complicated. Large distance levels are generally lowest in the direction across the roof pitches.     

厅堂声学计算机模型的应用(英)

此处所介绍的厅堂声学计算机模型计算了房间和多功能厅的声学响应。这个模型是以房厅的三维表示为基础的。每个墙面都赋予一个吸声系数,并且分成更小的单元。首先计算了单元间影响系数所组成的矩阵。对于给定的声源,可以计算每个单元收到的能量,产由此定出房厅内任一点的声级,这样又能够算出（ａ）随距离变化的衰减,（ｂ）在任一平面上的声照度,和（ｃ）任一点所收到声能的接收指向性。可以算出房厅内任一被选定点的回波图,作     

Sound strength and reverberation time in small concert halls

Sound strength and reverberation time measurements have been carried out in six small concert halls in Cambridge, UK. The sound strength G is a measure of the physical sound level in a concert hall and is closely related to the subjective sensation of loudness. It compares integrated impulse responses at a point in the measured room with that measured at ten metres distance in the free field.The aim of the measurements is to investigate the acoustic characteristics of the halls concerning sound strength and reverberation time. Furthermore the effect of the variable acoustics in the halls on these parameters is discussed in this paper. Especially for bigger ensembles it is often desirable to reduce the sound level in a small concert hall. The measurement results show that for a fixed hall volume, this can primarily be achieved by decreasing the reverberation time in the hall. However, with regard to the sound quality of a hall and the recommended reverberation times for chamber music, reverberation time cannot be reduced by an arbitrary extent. Therefore reverberation time and strength have to be balanced very carefully in order to obtain sufficient reverberation whilst at the same time avoiding excessive loudness. Finally the measured strength levels are compared to values derived from traditional and revised theory [Barron M, Lee L-J. Energy relations in concert auditoriums. J Acoust Soc Am 1988;84(2):618-28] on strength calculations in order to assess the accuracy of the theory for small chamber music halls. Possible reasons for the low measured strength levels observed are discussed with reference to related design features and objective acoustic parameters.     

Estimation of the location of a farfield acoustic source

A backward integration method for estimating the location of a source of sound waves in the atmosphere is presented. This geometric acoustics method is based upon the analysis of microphone array measurements to determine the incoming ray direction in three dimensions. The equations governing the propagation of the ray are then integrated backward in time. The sound source lies somewhere along the calculated ray path. The intersection of such loci from more than one array would provide an estimate of the source location. The method appears to be very rapid to implement and, assuming the time delays to be accurately measured, limited in accuracy only by the timeliness of the input sound speed and velocity profiles in the atmosphere.     

Acoustic characterization and prediction for fan-duct-plenum-room integrations

This paper investigates mutual influence of duct and room acoustics in the whole fan-duct-plenum-room integrations. Applying the parametric design language of finite element software ANSYS (APDL), dimensional and positional influence on system acoustics has been studied. Models with different room dimensions, duct lengths, duct cross-sections, duct locations, duct discharges and duct elbow were constructed, and their characteristics were compared qualitatively. Results show that small rooms, short ducts, large duct cross-sections and bell mouth duct discharges help to increase room sound pressure levels (SPLs); SPLs in ducts and plenums are sensitive to duct dimensions and duct discharge types but insensitive to duct locations and room dimensions; duct elbows have relatively indistinct acoustic influence in each component. Based on the calculation results, a semi-experimental method was proposed for simply and approximately evaluating indoor acoustic spectra of fan-duct-plenum-room integrations, then an example was used to demonstrate the prediction process. Finally, by adopting several ideal models, sound field constitutions, duct and room wall admittances and duct end reflection were explored quantitatively. This study may give a detailed understanding of fan-duct-plenum-room acoustics for researchers, also it might provide a new, simple and approximate prediction method for professionals to evaluate and improve fan-ducted acoustics.     

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.     

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。      

An analytical model to evaluate the cocktail party effect in restaurant dining rooms: A case study

This paper proposes the use of a simplified analytical model to evaluate acoustic conditions in restaurant dining rooms required for ensuring the intelligibility of conversations. The model is useful for design applications and is suitable for evaluating the maximum number of speakers present in a restaurant room in order to ensure intelligibility of conversations taking place at each table in the presence of background noise caused by conversations at other tables. The maximum number of speakers is studied in relation to the sound level difference between useful and disturbing sound, sound absorption of the room, and the average speaker–listener distance. The model is applied to the case of a dining hall in a multipurpose centre, which is currently in the planning stage.     

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.     

Sound reproduction systems using variable-directivity loudspeakers

Sound reproduction systems using omnidirectional loudspeakers produce reflections from room surfaces which interfere with the desired sound field within the array. While active compensation systems can reduce the reverberant level, they require calibration in each room and are processor-intensive. Directional loudspeakers allow the direct to reverberant level to be improved within the array, but still produce a finite exterior field which reflects from the room surfaces. The use of variable-directivity loudspeakers allows the exterior field to be eliminated at low frequencies by implementing the Kirchhoff-Helmholtz integral equation. This paper investigates the performance of variable-directivity arrays in reducing reverberant levels and compares the results with those derived in a previous paper for fixed-directivity arrays. The results presented may have some impact on the design of commercial multi-channel systems for sound reproduction.     

On holographic reconstruction without reference: Application to machinery noise and comparison with multi-reference holography approach

This article presents the implementation and application, to two complex machines, of two holography methods called CIBNAH (Complex Intensity Based Near-field Acoustic Holography) and MRNAH (Multi Reference Near-field Acoustic Holography) based on residual spectra. The first is a newly developed method that uses complex sound intensity as its basis and hence bypasses the need for reference signals. Here, several approaches are presented to validate both methods and to investigate each method’s advantages and limitations.     

Experimental implementation of a low-frequency global sound equalization method based on free field propagation

An experimental implementation of a global sound equalization method in a rectangular room using active control is described in this paper. The main purpose of the work has been to provide experimental evidence that sound can be equalized in a continuous three-dimensional region, the listening zone, which occupies a considerable part of the complete volume of the room. The equalization method, based on the simulation of a progressive plane wave, was implemented in a room with inner dimensions of 2.70 m × 2.74 m × 2.40 m. With this method, the sound was reproduced by a matrix of 4 × 5 loudspeakers in one of the walls. After traveling through the room, the sound wave was absorbed on the opposite wall, which had a similar arrangement of loudspeakers, by means of active control. A set of 40 digital FIR filters was used to modify the original input signal before it was fed to the loudspeakers, one filter for each transducer. The optimal arrangement of the loudspeakers and the maximum frequency that can be equalized is analyzed theoretically in this paper. The presented experimental results show that sound equalization was possible from 10 Hz to approximately 425 Hz in the listening zone. A flat frequency response with deviations within ±5 decibels from the desired value was achieved. A higher demanding performance with deviations within ±1.5 decibels from a flat frequency response was attained in the interval between 20 Hz and 280 Hz. At the same time, the impulse response was quite well approximated to a delayed delta function in the listening zone. Examples of the spatial distribution of the sound field are also shown.     

Decay rates and wall absorption at low frequencies

This paper is concerned with evaluating the error of conventional estimates of the boundary absorption of rectangular enclosures, with particular reference to reverberation room sound power measurements. The reverberation process is examined theoretically; the relative contributions to the decay rate from different modes in a rectangular room are calculated from an ensemble average over rooms with nearly the same dimensions. It is shown that the traditional method of determining the absorption of the walls of reverberation rooms systematically underestimates the absorption at low frequencies; the error is computed from the ensemble average. Finally, an unbiased estimate of the sound power radiated by a source in a reverberation room is derived. This estimate involves measurement of the initial decay rates of the room and is, unlike the usual reverberation room sound power estimate, neither based on statistical diffuse field considerations nor on the normal mode theory.     

Reception plate method for characterisation of structure-borne sound sources in buildings: Installed power and sound pressure from laboratory data

In a companion paper, a laboratory method is described to obtain the structure-borne sound power of machines before they are installed in heavy-weight buildings. The laboratory method is based on the concept of the reception plate. In this paper, the method is shown to provide appropriate input data for the prediction of the installed structure-borne power, and thence the resultant sound pressure level in rooms removed from the room containing the machine. Case studies of two common sources are described: a whirlpool bath and a water cistern. It is shown that the method can be incorporated into recently proposed standard prediction models and that sound pressure levels in buildings can be predicted.