无顶空间内混响感特性的研究

混响时间是反映时间域内声能衰变率的参量,它大体上可反映扩散空间内的主观混响感。但是,对于庭院等无顶空间而言是否适用,值得研究。因为在这些空间内,声能衰变过程中缺失了大量来自顶面的反射声,虽然混响时间可能变化不大,但会显著影响主观听觉上混响的感受。本文通过模拟声场的主观试听实验,发现反射声的方向性因素不可忽略,其对混响感有明显影响。由于混响时间是以单声道收录为界定的,因此,它不能充分反映无顶空间内的主观混响感。      

Prediction of reverberation time and speech transmission index in long enclosures

It is known that the sound field in a long space is not diffuse, and that the classic theory of room acoustics is not applicable. A theoretical model is developed for the prediction of reverberation time and speech transmission index in rectangular long enclosures, such as corridors and train stations, where the acoustic quality is important for speech. The model is based on an image-source method, and both acoustically hard and impedance boundaries are investigated. An approximate analytical solution is used to predict the frequency response of the sound field. The reverberation time is determined from the decay curve which is computed by a reverse-time integration of the squared impulse response. The angle-dependence of reflection coefficients of the boundaries and the change of phase upon reflection are incorporated in this model. Due to the relatively long distance of sound propagation, the effect of atmospheric absorption is also considered. Measurements of reverberation time and speech transmission index taken from a real tunnel, a corridor, and a model tunnel are presented. The theoretical predictions are found to agree well with the experimental data. An application of the proposed model has been suggested.     

On the steady-state and the transient decay methods for the estimation of reverberation time

The discrepancy between reverberation times of an enclosed sound field measured by the steady-state method and by the transient decay method is well-known. So far, no clear explanation has been obtained. In this paper, the steady-state bandlimited energy in an enclosure and bandlimited power flow into modally reactive boundaries are derived to describe the energy balance relationship and thus the reverberation time in a frequency band. This reverberation time is then compared to that obtained from the transient decay of the sound field based on the modal analysis. The comparison provides an understanding of the discrepancy mentioned above as well as the physical interpretations of the reverberation times estimated by both methods.     

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.     

Nonexponential sound decay in concert halls

The paper presents acoustic measurement results for two concert halls in which nonexponential sound decay is observed. Quantitative estimates are given for how the obtained decay laws differ from exponential. Problems are discussed that arise when using reverberation time to assess the quality of room acoustics with nonexponential sound decay.     

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.     

The effect of multiple branches on sound propagation in long enclosures

Since the classical room acoustics can not be used for long enclosures due to the inhomogeneous sound field, much work has been carried out recently to investigate the sound propagation in long enclosures, which are helpful to the acoustic design of practical long enclosures, such as the high-speed railway tunnels. However, most of these works focuses on the straight long enclosures without branches or with one branch. In this paper, the effects of the multiple branches on sound propagation of long enclosures are studied. The sound pressure level (SPL) attenuation, early decay time (EDT), and reverberation time (T30) of long enclosures with multiple branches have been investigated by physical scale models based acoustic experiments. Several interesting results have been obtained concerning the sound propagation of long enclosures with multiple branches. It shows that the sound field of long enclosures with multiple branches is more complex and inhomogeneous than that of the long enclosures without branches or with one branch.     

Walk away method versus reverberation time method in determining the room constant

The room constant is a quantity relating to the sound absorption in an enclosure. It is necessary to room acoustics engineering calculations based on the perfectly diffuse sound field theory, both in noise control work and sound system design. Normally, in the validation of the above-mentioned theory, in situ evaluations of the room constant are performed on the basis of reverberation time measurements in the relevant space. Such methods fail to give accurate results when uncertainties regarding the volumes and areas effectively involved in the reverberation process arise. In this case the walk away method is more advantageous in that it only needs steady state sound pressure level measurements at various distances from a small sound source whose directivity factor is known.In this paper both methods are discussed and experimental results relative to four different rooms are compared.     

A possible acoustic design approach for multi-purpose auditoria suitable for both speech and music

In several auditoria, it has been observed that the reverberation time is longer than expected and that the cause is a horizontal reverberant field established in the region near the ceiling, a field which is remote from the sound absorbing audience. This has been observed in the Boston Symphony Hall, Massachusetts, and the Stadthalle Göttingen, Germany. Subjective remarks on their acoustics suggest that there are no unfavourable comments linked to the secondary sound field. Two acoustic scale models are considered here. In a generic rectangular concert hall model, the walls and ceiling contained openings in which either plane or scattering panels could be placed. With plane panels, the model reverberation time (RT) was measured as 53% higher than the Sabine prediction (frequency 500/1000 Hz), compared with 8% higher with scattering panels. The second model of a 300 seat lecture theatre with a 6 m or 8 m high ceiling had raked seating. In this case, the amount of absorption in the model was increased until the point was reached where speech had acceptable intelligibility, with the early energy fraction, D ? 0.5. For this acceptable speech condition with the 6 m ceiling, the measured mid-frequency T₁₅ was 1.47 s, whereas the Sabine predicted RT was 1.06 s. The sound decay was basically non-linear with T₃₀ > T₁₅ > EDT. Exploiting a high-level horizontal reverberant field offers the possibility of acoustics that are better adapted as suitable for both speech and unamplified music, without any physical change in the auditorium. Using secondary reverberation in an auditorium for a wide variety of music might also be beneficial.     

Effect of diffusive and nondiffusive surfaces combinations on sound diffusion

One of room acoustic goals, especially in small to medium rooms, is sound diffusion in low frequencies, which have been the subject of lots of researches. Sound diffusion is a very important consideration in acoustics because it minimizes the coherent reflections that cause problems. It also tends to make an enclosed space sound larger than it is. Diffusion is an excellent alternative or complement to sound absorption in acoustic treatment because it doesn’t really remove much energy, which means it can be used to effectively reduce reflections while still leaving an ambient or live sounding space. Distribution of diffusive and nondiffusive surfaces on room walls affect sound diffusion in room, but the amount, combination, and location of these surfaces are still the matter of question. This paper investigates effects of these issues on room acoustic frequency response in different parts of the room with different source-receiver locations. Room acoustic model based on wave method is used (implemented) which is very accurate and convenient for low frequencies in such rooms. Different distributions of acoustic surfaces on room walls have been introduced to the model and room frequency response results are calculated. For the purpose of comparison, some measurements results are presented. Finally for more smooth frequency response in small and medium rooms, some suggestions are made.     

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.     

一种基于最大似然的混响时间盲估计方法*

混响是室内声学中的重要现象,在室内设计与音频信号处理中都需要测量或估计混响时间。本文改进了一种基于最大似然估计的混响时间盲估计方法,即采用说话人在房间中自然说话时发出的混响语音信号来估计混响时间的方法。该方法首先确定语音衰减段的最优边界,其次计算该衰减段的两个额外参数,据此筛选出符合条件的语音段,最后将满足条件的语音段采用最大似然估计得到混响时间估计值。在五个不同混响时间条件下的仿真表明,与已有方法相比,改进方法估计的混响时间同真实混响时间的偏差更小,方差更低,估计准确性较高。     

Point-to-point correlation of sound pressures in reverberation chambers

The reverberation chamber is such a fundamental tool in general acoustics and, in particular, architectural acoustics, that most of its properties have been under intensive study for some years. However, the point-to-point correlation of the sound pressures has received little attention except for an elegant paper by Cook et al. published in 1955.     

Computer models in room acoustics: The ray tracing method and the auralization algorithms

Computer algorithms are described for constructing virtual acoustic models of various rooms that should satisfy some specific sound quality criteria. The algorithms are based on the ray tracing method, which, in the general case, allows calculation of the amplitude of an acoustic ray that survived multiple reflections from arbitrary curved surfaces. As a result, calculations of room acoustics are reduced to tracing the trajectories of all the acoustic rays in the course of their propagation with multiple reflections from reflecting surfaces to the point of their complete decay. For this approach to be used, the following physical properties of a room should be known: the geometry of the reflecting surfaces, the absorption and diffusion coefficients on each of these surfaces, and the decay law for rays propagating in air. The proposed models allow for the solution of the important problem of architectural acoustics called the auralization problem, i.e., to predict how any given audio segment will sound in any given hall on the basis of computer simulation alone, without any full-scale testing in specific halls.     

Reverberation time and maximum background-noise level for classrooms from a comparative study of speech intelligibility metrics

Speech intelligibility metrics that take into account sound reflections in the room and the background noise have been compared, assuming diffuse sound field. Under this assumption, sound decays exponentially with a decay constant inversely proportional to reverberation time. Analytical formulas were obtained for each speech intelligibility metric providing a common basis for comparison. These formulas were applied to three sizes of rectangular classrooms. The sound source was the human voice without amplification, and background noise was taken into account by a noise-to-signal ratio. Correlations between the metrics and speech intelligibility are presented and applied to the classrooms under study. Relationships between some speech intelligibility metrics were also established. For each noise-to-signal ratio, the value of each speech intelligibility metric is maximized for a specific reverberation time. For quiet classrooms, the reverberation time that maximizes these speech intelligibility metrics is between 0.1 and 0.3 s. Speech intelligibility of 100% is possible with reverberation times up to 0.4-0.5 s and this is the recommended range. The study suggests "ideal" and "acceptable" maximum background-noise level for classrooms of 25 and 20 dB, respectively, below the voice level at 1 m in front of the talker.     

Sound focusing in rooms: the time-reversal approach

New perspectives in audible range acoustics, such as virtual sound space creation and active noise control, rely on the ability of the rendering system to recreate precisely a desired sound field. This ability to control sound in a given volume of a room is directly linked to the capacity to focus acoustical energy both in space and time. However, sound focusing in rooms remains a complicated problem, essentially because of the multiple reflections on obstacles and walls occurring during propagation. In this paper, the technique of time-reversal focusing, well known in ultrasound, is experimentally applied to audible range acoustics. Compared to classical focusing techniques such as delay law focusing, time reversal appears to considerably improve quality of both temporal and spatial focusing. This so-called super-resolution phenomenon is due to the ability of time reversal to take into account all of the different sound paths between the emitting antenna and the focal point, thus creating an adaptive spatial and temporal matched filter for the considered propagation medium. Experiments emphasize the strong robustness of time-reversal focusing towards small modifications in the medium, such as people in motion or temperature variations. Sound focusing through walls using the time-reversal approach is also experimentally demonstrated.     

An improved broad-spectrum room acoustics model including diffuse reflections

More and more literature has paid attention to the diffuse reflections in enclosed space during the past few years. In this paper, the current computer models including diffuse reflections have been reviewed briefly at first. Then, to realize the broad-spectrum simulation for enclosed sound fields including diffuse reflections, an improved ray-tracing algorithm, which combines the splitting coefficient diffusion model and a dynamic sound ray receiving method, has been given. The algorithm can deal with broad frequency bands simultaneously by using the frequency independent splitting coefficient. To test the algorithm and also to investigate the significance of the diffuse reflections in enclosed sound fields, experiments have been made in three spaces including a virtual room and two real rooms. The results and discussions have validated the applicability of the improved algorithm and they have also shown that diffuse reflections can improve room acoustic prediction, although it not always promote a sound field to be more diffused.     

Adaptive wave field synthesis for active sound field reproduction: experimental results

Sound field reproduction has applications in music reproduction, spatial audio, sound environment reproduction, and experimental acoustics. Sound field reproduction can be used to artificially reproduce the spatial character of natural hearing. The objective is then to reproduce a sound field in a real reproduction environment. Wave field synthesis (WFS) is a known open-loop technology which assumes that the reproduction environment is anechoic. The room response thus reduces the quality of the physical sound field reproduction by WFS. In recent research papers, adaptive wave field synthesis (AWFS) was defined as a potential solution to compensate for these quality reductions from which WFS objective performance suffers. In this paper, AWFS is experimentally investigated as an active sound field reproduction system with a limited number of reproduction error sensors to compensate for the response of the listening environment. Two digital signal processing algorithms for AWFS are used for comparison purposes, one of which is based on independent radiation mode control. AWFS performed propagating sound field reproduction better than WFS in three tested reproduction spaces (hemianechoic chamber, standard laboratory space, and reverberation chamber).     

基于声线跟踪法对闭空间声场预测的接收球半径计算

接收球半径是采用声线跟踪法进行声场预测的一个重要参数。以前所用到的接收球半径一般是通过经验估计得到的,缺乏详细的理论分析。本文从声波传播机理角度分析了根据声线密度来确定接收球半径的原因,并推导了矩形闭空间中声线密度和接收球半径的计算方法。声线密度可以通过原始声线数目,声场空间体积与形状,边界吸声系数来确定。在一给定闭空间里,声线密度可看作是均匀分布的,所以接收球半径与空间位置无关,可看作是一个常数。但对不同的声场空间来说,由于空间体积、形状和吸声系数的变化,声线密度是不同的,因此接收球半径也会不同。声线密度越大,接收球半径越小;声线密度越小,接收球半径越大。实验表明,所提出的接收球模型能用来较准确地预测闭空间里的声压级和混响时间等声学参数。