Determination of the sound energy level of a gunshot and its applications in room acoustics

In some cases an impulsive noise source such as a gunshot can be a preferred alternative when investigating building acoustics, including sound insulation measurements, when compared to conventional steady state noise sources. A gun equipped with blank cartridges is an impulsive noise source that is lightweight and small enough to be easily transported. The differences in the noise characteristics between individual cartridges for the same gun are usually small, so the impulsive source can be replicated to a high degree. This paper is focused on the practical application of the sound exposure levels produced by a gunshot with a known sound energy level in the rooms under investigation. In this way, the equipment and methods required by the conventional method are simplified significantly. Furthermore, reverberation times need not be measured, since the equivalent absorption area can be directly obtained from the measured sound exposure levels. Using Green’s theorem, the roles of the sound source and measuring microphone were exchanged, which simplified the determination of sound insulation as it was easier to change the position of the gun than the microphone. The results obtained using the impulsive noise source were in good agreement with those obtained using the conventional method. Above 100 Hz, their difference in any frequency band of interest was less than 1 dB.     

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.     

厅堂声学测量中不同激励声源的比较

基於脉冲响应积分的音乐厅和剧院观众厅声学特性的测量目前有三种使用不同激励声源的测试方法:人工脉冲声源、伪随机噪声序列(MLS)、以及用正弦扫频信号。这些技术各有其优缺点,在实际应用中为了方便根据具体情况选择不同的激励声源,通过在一个音乐厅现场的实测数据比较丁三种声源的实测结果,发现对混响时间测量三种不同的激励声源给出的结果基本一致,但是对明晰度和一些其他的指标,脉冲声源给出的结果与用MLS和扫频信号给出的结果有较明显的差别。对实际中如何选择具体的技术也做了建议。     

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.     

窄脉冲声用于大样品的吸声测量

本文利用逆滤波器原理,在空间产生了波形可控、长度在毫秒量级的窄脉冲声信号,分别采用脉冲分离法和脉冲叠加法,对一种毛毡材料和三种不同厚度的海绵材料进行了吸声系数的测量。实验证明,基于窄脉冲声信号的吸声测量结果与ISO13472-1:2002中的MLS脉冲法及阻抗管的测量结果基本吻合。采用窄脉冲进行吸声测量,可以减少样品边缘和周围环境对测量信号的干扰,提高现场测量的准确性。     

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.     

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.     

Estimating the direct-to-reverberant energy ratio from the coherence between coincident pressure and particle velocity

An analytical expression for the relationship between the direct-to-reverberant energy ratio (DRR) and the coherence estimation function between coincident pressure and particle velocity component is derived. The analytical solution is first validated with simulated room impulse responses and then used to estimate the DRR in five octave bands for several receiver positions measured in a total of 11 rooms of vastly different sizes and acoustic characteristics. The accuracy is evaluated by comparison with the DRR estimated directly from the room impulse response. The difference is typically 5 dB. For two rooms, the variation of the DRR estimate with source-to-receiver position is also shown. The method is blind in the sense that it is virtually independent of the signal generated by a single sound source.     

Measuring the sound absorption properties of noise barriers with inverse filtered maximum length sequences

Loudspeaker characteristics can have an appreciable influence on the sound absorption properties obtained with CEN/TS 1793-5 [1]. Although sound sources with omnidirectional radiation properties are favored, these often hold the problem of long impulse responses due to their design potentially incorporating ports [2]. In this paper, the inverse filtering approach is applied to two different sound sources. It is shown that it is a valuable measure to reduce the influence of the loudspeaker characteristics on the obtained sound absorption values.Furthermore, an attempt is made to measure the absorption properties according to CEN/TS 1793-5 without the need for the subtraction procedure of a free-field measurement. Thereby, as long as only high frequencies are considered, reasonable values can be obtained.     

A model comparison of the absorption coefficient of a Microperforated Insertion Unit in the frequency and time domains

The absorption coefficient of acoustic materials can be measured either in the frequency or the time domain. At normal incidence, a sample of the material is fitted within an impedance tube and the absorption coefficient is calculated in the frequency domain from the measurement of the transfer function between two microphones [ISO 10534-2. Acoustics - determination of sound absorption coefficient and impedance in impedance tubes - Part 2: transfer function method. ISO, Geneva, Switzerland; 1996]. When the acoustic material must be characterized at oblique incidence or in situ (noise barriers, for instance) the absorption coefficient is calculated from measurements of the loudspeaker-microphone impulse response in the time domain, both in free field and in front of the sample [CEN/TS 1793-5. Road traffic noise reduction devices - test method for determining the acoustic performance - Part 5: intrinsic characteristics - in situ values of sound reflection and airborne sound insulation. CEN, Brussels, Belgium; 2003, ISO 13472-1. Acoustic measurement of sound absorption properties of road surfaces in situ - Part I: extended surface method. ISO, Geneva, Switzerland; 2002]. Since the absorption is an intrinsic property of the acoustic material, its measurement in either domain must provide the same result. However, this has not been formally demonstrated yet. The aim of this paper is to carry out a comparison between the absorption coefficient predicted by the impedance model of a Microperforated Insertion Unit and the absorption coefficient predicted from a simulated reflection trace taken into account the finite length of the time window.     

Experimental study of sound propagation in a street

This paper presents an experimental study of the sound propagation in a street, performed during July 2002, in order to measure impulse responses at numerous locations in a street canyon, and, for several positions of the sound source. A specific attention has been paid on the processing of the experimental data, in order to obtain accurate values of the reverberation time and the steady-state sound pressure level, including also the compensation of the atmospheric attenuation. In total, 11 592 useful data have been collected during this experiment, and are available by contacting the corresponding author. Reverberation times and sound levels, in the narrow street, are then analyzed and compared, for the 4 source positions in the street. This experimental study gives interesting results on the behavior of the sound field in the street.     

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.     

The prediction of airborne sound transmission between two rooms using first-order flanking paths

Airborne sound transmission between adjacent rooms can be predicted using the Standard EN 12354-1 (ISO 15712-1), which is equivalent to a first-order approximation of statistical energy analysis (SEA). This paper analyses airborne sound transmission between adjacent rooms in a masonry building, by comparing results obtained from EN 12354-1 to SEA predictions and measurements. It is shown that the restriction of the Standard to first-order flanking paths can lead to large errors in predictions when compared to measurements and SEA results taking into account all transmission paths. This is observed both for individual flanking paths and overall transmission between rooms, for which the Standard provides results similar to those obtained by the first-order approximation of SEA. The paper also looks at possible reasons why previous studies using the approach in EN 12354 have generally shown good agreement with measurements.     

Spatial sound resolution of an interpolated HRIR library

The paper evaluates the human directional resolution of virtual sound sources synthesised with the aid of a generalised head related impulse response (HRIR) library, i.e., an HRIR library measured using a dummy head and torso. The original HRIR set is first expanded using linear interpolation, and then directional resolution measurements are performed for playback through headphones. These results are compared to the results obtained using loudspeakers as sound sources in an anechoic chamber. Directional resolution is the ability of listeners to distinguish two closely-spaced sound sources alternately playing the same signal. Experiments show that two sound sources with insufficient spacing appear as a single source to the listener. Directional resolution for small azimuth changes is relatively high for both virtual and real sound sources. Most test subjects have no problem resolving two sound sources only 5° apart. Compared to real sound sources, detecting changes in elevation of virtual sound sources is much less accurate, which may be the main drawback of using a generalised HRIR library.     

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.     

Research on the procedure for analyzing the sound quality contribution of sound sources and its application

Transfer path analysis (TPA) plays an important role for identifying and quantifying the contribution by airborne and structure-borne in the automotive industry. The main bottleneck of the TPA method is the test time consumption and complex procedure. It becomes a key target in many applications to find out the source with dominant contribution to overall noise rather than to identify each source. In recent years the contribution pattern of sources to the vehicle overall interior noise has changed with the reduction of engine noise, which masks all other sources. The panel radiation noise of vehicle body could not be ignored. There is an increasing demand for analyzing the sound quality contribution of sound sources in simple ways. The procedure for analyzing sound quality contribution of panel radiation noise is suggested in this study, in which an operational path analysis (OPA) method combined with partial singular value decomposition (PSVD) analysis is applied and sound quality objective assessment is introduced. The experimental research for verifying the procedure is finished, from which the source with largest sound quality contribution is picked up from three sources. For engineering application, the sound quality contributions of panels to the interior noise of a micro commercial vehicle are analyzed by using the procedure. By investigating the contributions of sound sources to each sound quality attribute, the dominant sound source is determined.     

小尺度阻尼边界封闭空间声传递函数的有限元素法计算

如何求解阻尼边界封闭空间中声源点到接收点的低频声传递函数已成为目前小尺度封闭空间可听化技术研究的关键技术，能处理任意形状及复杂边界条件的有限元素法可作为求解该问题的适合方法，以室内声声有源Helmholtz方程及其相应边界方程为基础，本文推导出了用于小尺度阻尼边界封闭空间声传递函数的有限元素求解方法，并编制了相应的计算机程序，在算例中，首先通过与模态叠加法计算结果进行比较，验证了该方法的正确性。最后计算了某型车体内腔中任意两点间声传递函数。     

Characterisation of valves as sound sources: Structure-borne sound

Taps and other valves are major sound sources in piping systems and can cause unacceptable noise levels in buildings. The noise results from the fluid-, structure- and air-borne sound emission. At present the acoustic emission of water appliances is tested according to a European standard, the shortcomings of which are apparent as a result of a round robin test of different European laboratories. As a result, there are currently neither acceptable measurement methods for water appliances available nor input data for prediction models. This paper considers methods of characterizing water appliances as sources of structure-borne sound. The concepts of mobility and free velocity are employed for a source characterisation based on power. Taps are considered alone and also in combination with a basin, where again the mobility and free velocity are used. A reception plate method is assessed as an alternative. The two methods each provide an independent characterisation of a structure-borne sound source as a single value. The values are on a power basis and provide input data suitable for prediction of the installed structure-borne power and thence the resultant sound pressure in adjacent rooms. Measured and predicted values of sound pressure level, caused by a wash-basin installed in an adjacent room, are compared.     

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.