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。     

A coherent model for predicting noise reduction in long enclosures with impedance discontinuities

A theoretical model has been developed for the prediction of sound propagation in a rectangular long enclosure with impedance discontinuities. Based on the image-source method, the boundaries are assumed to be geometrically reflective. An infinite number of image sources are generated by multiple reflections. The sound pressure of each image is obtained by an approximate analytical solution, known as the Weyl-van der Pol formula. The total sound field is then calculated by summation of the contribution from all images. The phase information of each image and the phase change upon reflection are included in the model. A single change of impedance in a two-dimensional duct is focused on as the fundamental problem of the current study. The diffraction effect at the impedance discontinuity is proved to be insignificant, and it is ignored in the formulation. On the assumption that the diffraction effect is not important, the investigation is moved on to a rectangular long enclosure. Measurements are conducted in two model tunnels to validate the proposed prediction model. The predictions are found to give good approximations of the experimental results. The theoretical model serves as the first attempt to optimize the position and pattern of sound absorption materials in a long enclosure, such as an underground railway station or a building corridor, for the reduction of noise and improvement of sound quality.     

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.     

Auralization study of optimum reverberation times for speech intelligibility for normal and hearing-impaired listeners in classrooms with diffuse sound fields

Speech-intelligibility tests auralized in a virtual classroom were used to investigate the optimal reverberation times for verbal communication for normal-hearing and hearing-impaired adults. The idealized classroom had simple geometry, uniform surface absorption, and an approximately diffuse sound field. It contained a speech source, a listener at a receiver position, and a noise source located at one of two positions. The relative output levels of the speech and noise sources were varied, along with the surface absorption and the corresponding reverberation time. The binaural impulse responses of the speech and noise sources in each classroom configuration were convolved with Modified Rhyme Test (MRT) and babble-noise signals. The resulting signals were presented to normal-hearing and hearing-impaired adult subjects to identify the configurations that gave the highest speech intelligibilities for the two groups. For both subject groups, when the speech source was closer to the listener than the noise source, the optimal reverberation time was zero. When the noise source was closer to the listener than the speech source, the optimal reverberation time included both zero and nonzero values. The results generally support previous theoretical results.     

浅海海底反射系数幅值参数的反演

理论分析了一种通过混响强度衰减特性获取海底反射系数的幅值参数的方法.将海底反射系数的幅值参数和相位参数引入到全波动混响模型中,为海底反射系数的反演提供理论基础。理论分析和数值仿真表明,在小掠射角条件下,利用混响强度衰减特性反演海底反射系数幅值参数的可行性和准确性。该反演方法只需要输入4个变量:本地混响强度的衰减特性,反射系数的相位参数,海深以及海深处的声速,同时要求混响数据具有一定的混响噪声比(大于6 dB)才能够使反演结果准确可信。根据本地静态混响实验数据成功反演得到海底反射系数的幅值参数.      

Sound field characteristics of underground railway stations – Effect of interior materials and noise source positions

In the case of most underground railway stations, no acoustical solutions are used to reduce train noise. Because the reflecting features of train noise in an underground station are not known, appropriate methods for controlling these features have yet to be established. The aim of this study was to clarify the sound field characteristics of underground stations by putting a sound source and receiver on the railway track and platform, respectively. The impulse responses for two vacant underground stations were measured to clarify the effects of the interior materials of the station (Comparison I), and the sound source was put in each station and tunnel to clarify the effect of the noise source positions (Comparison II). Results showed that the sound fields were similar between the stations whose lateral walls were covered with either metallic or fire-resistant wooden panels (Comparison I), and that the sound field for the sound sources near or in the tunnel presented a higher strength (G) by 5.1 dB and longer reverberation time (EDT) by 0.7 s compared to the sound source in the station (Comparison II). The sound sources in the tunnel presented strong and long reverberations at around 500 Hz due to the convergence effect of the tunnel. Therefore, this study proposes a platform screen with doors to limit noise transmission into the platform.     

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.     

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.     

Improving sound diffusion in a reverberation tank using a randomly fluctuating surface

Underwater reverberation environments that satisfy the conditions of uniformity and isotropy of the diffuse field can be used to measure the acoustic characteristics of underwater targets. This study combines two practical indicators — the standard deviation of the absolute sound pressure field (to indicate uniformity) and the analysis of the wavenumber spectrum in the spherical harmonics domain (to indicate isotropy) — for an accurate evaluation of the diffusion of the sound field in a reverberation tank. A method is proposed that can improve the narrow-band diffusion of the sound field by employing a randomly fluctuating surface. An acoustic experiment was performed in a reverberation water tank (1.2 m×1 m×0.8 m), where a randomly fluctuating surface was generated by making waves. The experimental results show that as the wave motion contributes effectively to the random reflection of sound rays in all directions, the uniformity and isotropy are improved significantly when the surface is fluctuating randomly. This work helps to ensure accurate measurements of the characteristics of underwater targets in reverberation tanks.     

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.     

Coherent active methods for applications in room acoustics

An adjustment of reverberation time in rooms is often desired, even for low frequencies where passive absorbers fail. Among the active (electroacoustic) systems, incoherent ones permit lengthening of reverberation time only, whereas coherent active methods will allow sound absorption as well. A coherent-active wall lining consists of loudspeakers with microphones in front and adjustable control electronics. The microphones pick up the incident sound and drive the speakers in such a way that the reflection coefficient takes on prescribed values. An experimental device for the one-dimensional case allows reflection coefficients between almost zero and about 1.5 to be realized below 1000 Hz. The extension to three dimensions presents problems, especially by nearfield effects. Experiments with a 3 X 3 loudspeaker array and computer simulations proved that the amplitude reflection coefficient can be adjusted between 10% and 200% for sinusoidal waves at normal and oblique incidence. Future developments have to make the system work with broadband excitation and in more diffuse sound fields. It is also planned to combine the active reverberation control with active diffusion control.     

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。      

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.     

Numerical modeling of the sound fields in urban squares

This paper studies the basic characteristics of sound fields in urban squares surrounded by reflecting building fa?ades and the effectiveness of architectural changes and urban design options. A radiosity model and an image source model are developed, and a parametric study is carried out in hypothetical squares. The results show that the reverberation time (RT) is rather even in a square, whereas the early decay time (EDT) is very low in the near field, and then becomes close to RT after a rapid increase. Compared to diffuse boundaries, with geometrical boundaries the RT and EDT are significantly longer and the sound pressure level (SPL) attenuation with distance is generally smaller unless the height/side ratio is high. With a boundary diffusion coefficient of 0.2, the sound field is already close to that resulting from purely diffusely reflecting boundaries. The SPL in far field is typically 6-9 dB lower if the square side is doubled; 8 dB lower if the height of building fa?ades is decreased from 50 m to 6 m (diffuse boundaries); 5 dB (diffuse boundaries) or 2 dB (geometrical boundaries) lower if the length/width ratio is increased from 1 to 4; and 10-12 dB lower if the boundary absorption coefficient is increased from 0.1 to 0.9.     

The prediction of speech intelligibility in underground stations of rectangular cross section

Long enclosures are spaces with nondiffuse sound fields, for which the classical theory of acoustics is not appropriate. Thus, the modeling of the sound field in a long enclosure is very different from the prediction of the behavior of sound in a diffuse space. Ray-tracing computer models have been developed for the prediction of the sound field in long enclosures, with particular reference to spaces such as underground stations which are generally long spaces of rectangular or curved cross section. This paper describes the development of a model for use in underground stations of rectangular cross section. The model predicts the sound-pressure level, early decay time, clarity index, and definition at receiver points along the enclosure. The model also calculates the value of the speech transmission index at individual points. Measurements of all parameters have been made in a station of rectangular cross section, and compared with the predicted values. The predictions of all parameters show good agreement with measurements at all frequencies, particularly in the far field of the sound source, and the trends in the behavior of the parameters along the enclosure have been correctly predicted.     

Passive reverberation nulling for target enhancement

Echo-to-reverberation enhancement previously has been demonstrated using time reversal focusing when knowledge of the channel response between a target and the source array elements is available. In the absence of this knowledge, direct focusing is not possible. However, active reverberation nulling still is feasible given observations of reverberation from conventional source array transmissions. For a given range of interest, the response between the source array elements and the dominant sources of boundary reverberation is provided by the corresponding reverberation from this range. Thus, an active transmission can be projected from the source array which minimizes the energy interacting with the boundaries at a given range while still ensonifying the waveguide between the boundaries. As an alternative, here a passive reverberation nulling concept is proposed. In a similar fashion, the observed reverberation defines the response between the source array elements and the dominant sources of boundary reverberation at each range and this is used to drive a range-dependent sequence of projection operators. When these projection operators subsequently are applied to the received data vectors, reverberation can be diminished. The improvement in target detectability is demonstrated using experimental data with an echo repeater simulating the presence of a target.     

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.     

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.     

Effects of a single reflection with varied horizontal angle and time delay on speech intelligibility.

Previously, almost all physical measures for estimating speech intelligibility in a room have been derived from only temporal-monaural criteria. This paper shows that speech intelligibility for a sound field with a single reflection depends not only on the temporal-monaural factor but also on the spatial-binaural factor of the sound field. Articulation tests for sound fields simulated with a single reflection of delay time delta t1 after the direct sound were conducted changing the horizontal incident angle xi of the reflection. Remarkable findings are as followings: (1) speech intelligibility (SI) decreases with increasing delay time delta t1, (2) SI increases when xi approaches 90 degrees; the horizontal angle of the reflection causes a significant effect on SI, and (3) the analysis of variance for articulation test scores clearly demonstrated that the effects of both delta t1 and xi on SI are fully independent. Concerning result (2), if listeners get a spatial separation of signals at the two ears, then the listener's capability for speech perception is assumed to be improved due to "adding" further information to the temporal pattern recognition.