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.     

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.     

Blind single-channel suppression of late reverberation based on perceptual reverberation modeling

A blind method for suppressing late reverberation from speech and audio signals is presented. The proposed technique operates both on the spectral and on the sub-band domains employing a single input channel. At first, a preliminary rough clean signal estimation is required and for this, any standard technique may be applied; however here the estimate is obtained through spectral subtraction. Then, an auditory masking model is employed in sub-bands to extract the reverberation masking index (RMI) which identifies signal regions with perceived alterations due to late reverberation. Utilizing a selective signal processing technique only these regions are suppressed through sub-band temporal envelope filtering based on analytical expressions. Objective and subjective measures indicate that the proposed method achieves significant late reverberation suppression for both speech and music signals over a wide range of reverberation time (RT) scenarios.     

A blind algorithm for reverberation-time estimation using subband decomposition of speech signals

An algorithm for blind estimation of reverberation time (RT) in speech signals is proposed. Analysis is restricted to the free-decaying regions of the signal, where the reverberation effect dominates, yielding a more accurate RT estimate at a reduced computational cost. A spectral decomposition is performed on the reverberant signal and partial RT estimates are determined in all signal subbands, providing more data to the statistical-analysis stage of the algorithm, which yields the final RT estimate. Algorithm performance is assessed using two distinct speech databases, achieving 91% and 97% correlation with the RTs measured by a standard nonblind method, indicating that the proposed method blindly estimates the RT in a reliable and consistent manner.     

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.     

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.     

Numerical solutions of the acoustic eigenvalue equation in the rectangular room with arbitrary (uniform) wall impedances

Two numerical procedures for finding the acoustic eigenvalues in the rectangular room with arbitrary (uniform) wall impedances are developed. One numerical procedure applies Newton's method. Here, starting with soft walls, the eigenvalues are found by increasing the impedances of each wall pair in small increments up to the terminal impedances. Another procedure poses the eigenvalue problem as one of homotopic continuation from a non-physical reference configuration in which all eigenvalues are known and obvious. The continuation is performed by the numerical integration of two differential equations. The latter procedure was found to be faster and finds all possible solutions. The set of eigenvalues allowed the room modal natural frequencies and damping constants to be obtained. From sound decays measured in a hard-walled rectangular room, and from the collective-modal-decay curve, the impedances of the hard walls are estimated. These are then used to find the reverberation times of the modes in the room with the floor lined with sound absorbing material of known acoustic impedance. It was found that a single reverberation time, for all modes, is only supported in the rectangular room with hard walls and at the higher frequency bands, consistent with Sabine's theory, which assumes a diffuse sound field. In the rectangular room with hard walls and at the lower frequency bands, and in the rectangular room with the floor lined with sound absorbing material and for all frequency bands, modes with rather distinctive reverberation times may produce sound decays not always consistent with Sabine's prediction.     

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。     

早晚期混响划分对理想比值掩蔽在语音识别性能上的影响

真实环境中存在的噪声和混响会降低语音识别系统的性能。封闭空间中的混响包括直达声、早期反射和后期混响3部分,它们对语音识别系统具有不同的影响.我们研究了早期反射和后期混响的不同划分方法,以其中的早期反射为目标语音,计算出了不同的理想比值掩蔽并研究了它们对语音识别系统性能的影响;在此基础上,利用双向长短时记忆网络(BLSTM)估计理想比值掩蔽,测试它们对语音识别系统性能的影响.实验结果表明,基于Abel早期反射和后期混响的划分方法,理想比值掩蔽能够降低词错误率约2.8%;基于BLSTM的估计方法过低估计了理想比值掩蔽,未能有效提高语音识别系统的性能。      

Ceiling baffles and reflectors for controlling lecture-room sound for speech intelligibility

Reinforcing speech levels and controlling noise and reverberation are the ultimate acoustical goals of lecture-room design to achieve high speech intelligibility. The effects of sound absorption on these factors have opposite consequences for speech intelligibility. Here, novel ceiling baffles and reflectors were evaluated as a sound-control measure, using computer and 1/8-scale models of a lecture room with hard surfaces and excessive reverberation. Parallel ceiling baffles running front to back were investigated. They were expected to absorb reverberation incident on the ceiling from many angles, while leaving speech signals, reflecting from the ceiling to the back of the room, unaffected. Various baffle spacings and absorptions, central and side speaker positions, and receiver positions throughout the room, were considered. Reflective baffles controlled reverberation, with a minimum decrease of sound levels. Absorptive baffles reduced reverberation, but reduced speech levels significantly. Ceiling reflectors, in the form of obstacles of semicircular cross section, suspended below the ceiling, were also tested. These were either 7 m long and in parallel, front-to-back lines, or 0.8 m long and randomly distributed, with flat side up or down, and reflective or absorptive top surfaces. The long reflectors with flat side down and no absorption were somewhat effective; the other configurations were not.     

Ray-tracing prediction of optimal conditions for speech in realistic classrooms

Recent papers have discussed the optimal reverberation times in classrooms for speech intelligibility, based on the assumption of a diffuse sound field. Here this question was investigated for more ‘typical’ classrooms with non-diffuse sound fields. A ray-tracing model was modified to predict speech-intelligibility metric U₅₀. It was used to predict U₅₀ in various classroom configurations for various values of the room absorption, allowing the optimal absorption (that predicting the highest U₅₀)—and the corresponding optimal reverberation time—to be identified in each case. The range of absorptions and reverberation times corresponding to high speech intelligibility were also predicted in each case. Optimal reverberation times were also predicted from the optimal surface-absorption coefficients using Sabine and Eyring versions of diffuse-field theory, and using the diffuse-field expression of Hodgson and Nosal. In order to validate the ray-tracing model, predictions were made for three classrooms with highly diffuse sound fields; these were compared to values obtained by the diffuse-field models, with good agreement. The methods were then applied to three ‘typical’ classrooms with non-diffuse fields. Optimal reverberation times increased with room volume and noise level to over 1 s. The accuracy of the Hodgson and Nosal expression varied with classroom size and noise level. The optimal average surface-absorption coefficients varied from 0.19 to 0.83 in the different classroom configurations tested. High speech intelligibility was, in general, predicted for a wide range of coefficients, but could not be obtained in a large, noisy classroom.     

Investigation of Chinese word recognition scores of children in primary school classroom with different speech sound pressure levels

Chinese word recognition (CWR) test was conducted by grades 3 and 5 children under the different conditions of reverberation time (RT), background noise level (BNL) and speech sound pressure level (SSPL) in three primary-school classrooms. The CWR scores and signal to noise ratios (SNRs) have been obtained at listening positions. Results show that the CWR score for grades 3 and 5 children increases with increase of SSPL, decrease of RT or increase of age, but it decreases with increase of BNL under the same conditions. For a mixed noise of 56 dBA (speech-spectrum-like noise and ambient noise), the CWR scores in the classroom for grades 3 and 5 children reach a peak at SNR of 15–20 dBA under the same RT and age of children condition. For the natural ambient noise, the CWR score for grades 3 and 5 children gradually increases with increase of the SNR. The high SSPL could not guarantee good CWR for children in classroom, which also depends on RT and BNL in classroom. When the classroom has long RT or high BNL, the increase of SSPL would not be necessarily to achieve better CWR. The novelty of the present study is to further evaluate and confirm the results under environments of real classrooms (not simulated room in laboratory).     

Speech segregation in rooms: effects of reverberation on both target and interferer

Speech reception thresholds were measured to investigate the influence of a room on speech segregation between a spatially separated target and interferer. The listening tests were realized under headphones. A room simulation allowed selected positioning of the interferer and target, as well as varying the absorption coefficient of the room internal surfaces. The measurements involved target sentences and speech-shaped noise or 2-voice interferers. Four experiments revealed that speech segregation in rooms was not only dependent on the azimuth separation of sound sources, but also on their direct-to-reverberant energy ratio at the listening position. This parameter was varied for interferer and target independently. Speech intelligibility decreased as the direct-to-reverberant ratio of sources was degraded by sound reflections in the room. The influence of the direct-to-reverberant ratio of the interferer was in agreement with binaural unmasking theories, through its effect on interaural coherence. The effect on the target occurred at higher levels of reverberation and was explained by the intrinsic degradation of speech intelligibility in reverberation.     

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.     

An investigation of simple methods for assessing reverberation time

There is considerable interest in the development of a simple test for sound insulation between dwellings. The assessment of reverberation time is the most difficult part of the procedure to simplify. In this paper six alternative methods are described and evaluated. The first three require no electronic apparatus but are not accurate enough for general use. The fourth involves using sufficient absorbing material in the receiving room to effectively determine the RT. This approach appears worth further development. The fifth approach requires a source of known sound power and the final method employs a simple meter giving a direct reading of the decay time. The last two methods appear to be accurate enough for inclusion in a simplified test method but the simple meter seems to have some advantage.     

A study on the optimal English speech level for Chinese listeners in classrooms

Speech intelligibility in classrooms affects the learning efficiency of students directly, especially for the students who are using a second language. The speech intelligibility value is determined by many factors such as speech level, signal to noise ratio, and reverberation time in the rooms. This paper investigates the contributions of these factors with subjective tests, especially speech level, which is required for designing the optimal gain for sound amplification systems in classrooms. The test material was generated by mixing the convolution output of the English Coordinate Response Measure corpus and the room impulse responses with the background noise. The subjects are all Chinese students who use English as a second language. It is found that the speech intelligibility increases first and then decreases with the increase of speech level, and the optimal English speech level is about 71 dBA in classrooms for Chinese listeners when the signal to noise ratio and the reverberation time keep constant. Finally, a regression equation is proposed to predict the speech intelligibility based on speech level, signal to noise ratio, and reverberation time.     

Speech dereverberation method based on spectral subtraction and spectral line enhancement

Speech signals recorded with a distant microphone usually are interfered by the spatial reverberation in the room, which severely degrades the clarity and intelligibility of speech. A speech dereverberation method based on spectral subtraction and spectral line enhancement is proposed in this paper. Following the generalized statistical reverberation model, the power spectrum of late reverberation is estimated and removed from the reverberation speech by the spectral subtraction method. Then, according to the human auditory model, a spectral line enhancement technique based on adaptive post-filtering is adopted to further eliminate the reverberant components between adjacent speech formants. The proposed method can effectively suppress the spatial reverberation and improve the auditory perception of speech. The subjective and objective evaluation results reveal that the perceptual quality of speech is greatly improved by the proposed method.     

Fast estimation of speech transmission index using the reverberation time

The paper presents the function of STI in the domain of reverberation time. Through the application of the said function, we can quickly estimate the speech transmission index, knowing only the time of room reverberation. For that purpose we applied a known method which consists in physical estimation of speech intelligibility basing on the modulation transfer function (MTF) determined in a room. Then, the STI was described using a logarithmic function whereof argument was the room reverberation time. To verify the model, reverberation times of six rooms were measured. The selected rooms were very different deliberately. They had different cubature and shape. The selection included a small cuboid, lecture halls and a church. Then, the same rooms were modeled in the ODEON version 11.23 and their reverberation times were determined. Furthermore, the STI was determined in the ODEON and then compared with the reverberation time obtained in effect of fast estimation. The statistical verification with the use of correlation index and regression equation has demonstrated that the fast estimation yields results close to those obtained in the computer simulation in ODEON. We obtained the correlation index at the level close to 1. Furthermore, the test probability at the level lower than 0.05 bespeaks of a statistically significant linear relation for the confidence level of 95%.     

Pitch-based monaural segregation of reverberant speech

In everyday listening, both background noise and reverberation degrade the speech signal. Psychoacoustic evidence suggests that human speech perception under reverberant conditions relies mostly on monaural processing. While speech segregation based on periodicity has achieved considerable progress in handling additive noise, little research in monaural segregation has been devoted to reverberant scenarios. Reverberation smears the harmonic structure of speech signals, and our evaluations using a pitch-based segregation algorithm show that an increase in the room reverberation time causes degraded performance due to weakened periodicity in the target signal. We propose a two-stage monaural separation system that combines the inverse filtering of the room impulse response corresponding to target location and a pitch-based speech segregation method. As a result of the first stage, the harmonicity of a signal arriving from target direction is partially restored while signals arriving from other directions are further smeared, and this leads to improved segregation. A systematic evaluation of the system shows that the proposed system results in considerable signal-to-noise ratio gains across different conditions. Potential applications of this system include robust automatic speech recognition and hearing aid design.     

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.