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.     

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。      

Monaural room acoustic parameters from music and speech

This paper compares two methods for extracting room acoustic parameters from reverberated speech and music. An approach which uses statistical machine learning, previously developed for speech, is extended to work with music. For speech, reverberation time estimations are within a perceptual difference limen of the true value. For music, virtually all early decay time estimations are within a difference limen of the true value. The estimation accuracy is not good enough in other cases due to differences between the simulated data set used to develop the empirical model and real rooms. The second method carries out a maximum likelihood estimation on decay phases at the end of notes or speech utterances. This paper extends the method to estimate parameters relating to the balance of early and late energies in the impulse response. For reverberation time and speech, the method provides estimations which are within the perceptual difference limen of the true value. For other parameters such as clarity, the estimations are not sufficiently accurate due to the natural reverberance of the excitation signals. Speech is a better test signal than music because of the greater periods of silence in the signal, although music is needed for low frequency measurement.     

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.     

Spectral loudness summation of nonsimultaneous tone pulses

The level of broadband signals is usually lower than that of equally loud narrow-band signals. This effect, referred to as spectral loudness summation, is commonly measured for broadband signals where all frequency components are presented simultaneously. The present study investigated to what extent spectral loudness summation also occurs for nonsimultaneously presented frequency components. Spectral loudness summation was measured in normal-hearing listeners with an adaptive forced-choice procedure for sequences of short tone pulses with varying frequencies, randomly chosen from a set of five frequencies. In addition, spectral loudness summation was measured for the simultaneous presentation of all five frequencies. The comparison stimulus consisted of tone pulses with the same frequency for all tone pulses of the sequence and the same repetition rate and overall duration as the test signal. The pulse duration was 10, 20, 50, or 100 ms and the inter-pulse interval ranged from 0 to 390 ms. In general, a considerable nonsimultaneous spectral loudness summation was found for short pulse durations and inter-pulse intervals, but a residual effect was also observed for the largest inter-pulse interval. The data are discussed in the light of repetition-rate dependent spectral loudness summation and effects of persistence of specific loudness after tone-pulse offset.     

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.     

汉语语境下混响感的语意调查与分析

探讨了汉语语境下混响和混响时间的定义问题。对两组具有声频工程和建筑声学专业背景的人员进行了主观混响感语意因素重要性的问卷调查,分析了听者个人背景对汉语语境下的混响感语意理解的影响。在实验室初步测量了混响时间与混响感之间的听觉心理关系。语意调查和听觉实验结果揭示出混响时间并不是唯一决定主观混响感的因素,混响感的构成有更为复杂和多元的机理,而且与听者的个人背景和经验有关。      

The acoustical characterization of orchestra platforms and uncertainty estimation of the results

Measurements have been carried out on furnished orchestra platforms in four concert halls in Italy in order to describe the sound field perceived by musicians. The heterogeneous nature of the orchestra suggested a procedure able to take into account the mutual hearing between instrumental sections. The measured parameters were the early, late and total support, the reverberation time, the early decay time and the clarity index. A part of the study has been devoted to the measurement uncertainty estimation. The source directivity and the small displacements of the microphone influence the early decay time to a great extent while the on-platform spatial variability affects both the early decay time and the clarity index. Per-section early support shows differences that render the overall spatial mean inappropriate to describe the stage as a whole. For the other parameters an overall mean platform value can instead be suitable, even though, for the case of clarity a more evident group variability is observed. The values of late support, reverberation time, early decay time and clarity index, proposed in literature as suitable measures of reverberance for musicians, are not all intercorrelated, indicating that not all these parameters can be associated to the same subjective impression.     

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.     

混响感知的听觉心理

综述了中国传媒大学传播声学研究所近年来在混响的主观感知机理上的研究工作和进展,涉及混响感的语意调查与分析,音乐听闻的混响偏爱度实验,混响感的差别阈限,混响感的因素分析,以及混响处理与音乐情感之间的相互影响等研究结果。对混响感知的研究对进一步探究在有界空间中主观听感的生理心理机制是十分有启发的。     

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。     

Fine structure of hearing threshold and loudness perception

Hearing thresholds measured with high-frequency resolution show a quasiperiodic change in level called threshold fine structure (or microstructure). The effect of this fine structure on loudness perception over a range of stimulus levels was investigated in 12 subjects. Three different approaches were used. Individual hearing thresholds and equal loudness contours were measured in eight subjects using loudness-matching paradigms. In addition, the loudness growth of sinusoids was observed at frequencies associated with individual minima or maxima in the hearing threshold from five subjects using a loudness-matching paradigm. At low levels, loudness growth depended on the position of the test- or reference-tone frequency within the threshold fine structure. The slope of loudness growth differs by 0.2 dB/dB when an identical test tone is compared with two different reference tones, i.e., a difference in loudness growth of 2 dB per 10-dB change in stimulus. Finally, loudness growth was measured for the same five subjects using categorical loudness scaling as a direct-scaling technique with no reference tone instead of the loudness-matching procedures. Overall, an influence of hearing-threshold fine structure on loudness perception of sinusoids was observable for stimulus levels up to 40 dB SPL--independent of the procedure used. Possible implications of fine structure for loudness measurements and other psychoacoustic experiments, such as different compression within threshold minima and maxima, are discussed.     

Modeling binaural loudness

A survey of data on the perception of binaurally presented sounds indicates that loudness summation across ears is less than perfect; a diotic sound is less than twice as loud as the same sound presented monaurally. The loudness model proposed by Moore et al. [J. Audio Eng. Soc. 45, 224-240 (1997)] determines the loudness of binaural stimuli by a simple summation of loudness across ears. It is described here how the model can be modified so as to give more accurate predictions of the loudness of binaurally presented sounds, including cases where the sounds at the two ears differ in level, frequency or both. The modification is based on the idea that there are inhibitory interactions between the internal representations of the signals at the two ears, such that a signal at the left ear inhibits (reduces) the loudness evoked by a signal at the right ear, and vice versa. The inhibition is assumed to spread across frequency channels. The modified model gives reasonably accurate predictions of a variety of data on the loudness of binaural stimuli, including data obtained using loudness scaling and loudness matching procedures.     

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.     

Influence of vehicle color on loudness judgments

This experiment investigates the effect of images of differently colored sports cars on the loudness of a simultaneously perceived car sound. Still images of a sports car, colored in red, light green, blue, and dark green, were displayed to subjects during a magnitude estimation task. The sound of an accelerating sports car was used as a stimulus. Statistical analysis suggests that the color of the visual stimulus may have a small influence on loudness judgments. The observed loudness differences are generally equivalent to a change in sound level of about 1 dB, with maximum individual differences of up to 3 dB.     

一种频域合成房间频率响应的人工混响方法

给出了一种频域合成房间频率响应的方法用于卷积法人工混响,基于频域内房间频率响应的后期部分为高斯随机过程的假设,用自回归滑动平均模型为其自协方差函数和功率谱密度进行参数化描述,在对自回归滑动平均模型中的参数求解后,通过逆滤波得到了房间频率响应后期部分,与房间频率响应前期部分组合后经过傅里叶反变换得到完整的房间脉冲响应。仿真结果表明该方法的混响效果与镜像源法接近,明显优于反馈延迟网络法,但其计算复杂度比镜像源法低,便于实时应用。     

Prediction of energy decay in room impulse responses simulated with an image-source model

A method is proposed that provides an approximation of the acoustic energy decay (energy-time curve) in room impulse responses generated using the image-source technique. A geometrical analysis of the image-source principle leads to a closed-form expression describing the energy decay curve, with the resulting formula being valid for a uniform as well as nonuniform definition of the enclosure's six absorption coefficients. The accuracy of the proposed approximation is demonstrated on the basis of impulse-response simulations involving various room sizes and reverberation levels, with uniform and nonuniform sound absorption coefficients. An application example for the proposed method is illustrated by considering the task of predicting an enclosure's reflection coefficients in order to achieve a specific reverberation level. The technique presented in this work enables designers to undertake a preliminary analysis of a simulated reverberant environment without the need for time-consuming image-method simulations.     

房间脉冲响应模拟法及混响时间盲估计应用

在构建混响语声数据集时,由于缺乏真实长混响房间脉冲响应且模拟的房间脉冲响应与真实不符,因而导致数据驱动的混响时间盲估计模型性能下降。提出了一种基于条件生成对抗网络的房间脉冲响应模拟法,该方法利用真实的房间脉冲响应训练条件生成对抗网络,可以根据指定的混响时间模拟更加真实的房间脉冲响应。使用不同方法模拟的房间脉冲响应构建训练集用于训练盲估计模型,通过声学实验评估模型性能。实验结果表明,由该方法模拟的房间脉冲响应训练的估计模型在不同信噪比下均具有最小的均方根误差且在长混响情况下显著优于其他模型。     

AN ACOUSTIC DECAY MEASUREMENT BASED ON TIME-FREQUENCY ANALYSIS USING WAVELET TRANSFORM

In acoustic decay measurement using the third-octave band pass filter, it is known that an inevitable experimental error is produced by “ringing” at the tail part of the impulse response of the third-octave band pass filter. This ringing gives rise to distortion of the decay curve. In order to reduce this error and to obtain an acceptable acoustic decay curve, it has been recommended that the product of the 3 dB bandwidth B of the third-octave band pass filter and the reverberation time T₆₀ of the room under test be at least 16. For a listening room having short reverberation time and at the low-frequency band with narrow bandwidth, the decay curve cannot, therefore, be measured reliably by using the third-octave band pass filter. In this paper, the continuous wavelet transform (CWT) has been proposed to determine accurately the decay curve with a low value of BT₆₀. The CWT decomposes an acoustic decay signal into time-frequency domain using the third-octave band wavelet filter bank. When the CWT is applied to the measurement of an acoustic decay curve, it is found that the requirement BT₆₀>16 can be replaced by the replacementBT₆₀ >4.     

Influence of pitch, loudness, and timbre on the perception of instrument dynamics

The effect of variations in pitch, loudness, and timbre on the perception of the dynamics of isolated instrumental tones is investigated. A full factorial design was used in a listening experiment. The subjects were asked to indicate the perceived dynamics of each stimulus on a scale from pianissimo to fortissimo. Statistical analysis showed that for the instruments included (i.e., clarinet, flute, piano, trumpet, and violin) timbre and loudness had equally large effects, while pitch was relevant mostly for the first three. The results confirmed our hypothesis that loudness alone is not a reliable estimate of the dynamics of musical tones.