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.     

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.     

Chinese speech intelligibility at different speech sound pressure levels and signal-to-noise ratios in simulated classrooms

The speech intelligibility in classroom can be influenced by background-noise levels, speech sound pressure level (SSPL), reverberation time and signal-to-noise ratio (SNR). The relationship between SSPL and subjective Chinese Mandarin speech intelligibility and the effect of different SNRs on Chinese Mandarin speech intelligibility in the simulated classroom were investigated through room acoustical simulation, auralisation technique and subjective evaluation. Chinese speech intelligibility test signals recorded in anechoic chamber were convolved with the simulated binaural room impulse responses, and then reproduced through the headphone by different SSPLs and SNRs. The results show that Chinese Mandarin speech intelligibility scores increase with increasing of SSPLs and SNRs within a certain range in simulated classrooms. Chinese Mandarin speech intelligibility scores have no significant difference with SNRs of no less than 15 dBA under the same reverberation time condition.     

Verifying two commercial software implementations of impulse-response-based speech intelligibility measurements

A number of objective evaluation methods are currently used to quantify the speech intelligibility in a built environment, including the speech transmission index (STI), rapid speech transmission index (RASTI), articulation index (AI), and the percent articulation loss of consonants (%ALCons). Certain software programs can quickly evaluate STI, RASTI, and %ALCons from a measured room impulse response. In this project, two impulse-response-based software packages (WinMLS and SIA-Smaart Acoustic Tools) were evaluated for their ability to determine intelligibility accurately. In four different spaces with background noise levels less than NC 45, speech intelligibility was measured via three methods: (1) with WinMLS 2000; (2) with SIA-Smaart Acoustic Tools (v4.0.2); and (3) from listening tests with humans. The study found that WinMLS measurements of speech intelligibility based on STI, RASTI, and %ALCons corresponded well with performance on the listening tests. SIA-Smaart results were correlated to human responses, but tended to under-predict intelligibility based on STI and RASTI, and over-predict intelligibility based on %ALCons.     

Psychoacoustic evaluation of multichannel reproduced sounds using binaural synthesis and spherical beamforming

The binaural auralization of a 3D sound field using spherical-harmonics beamforming (SHB) techniques was investigated and compared with the traditional method using a head-and-torso simulator (HATS). The new procedure was verified by comparing simulated room impulse responses with measured ones binaurally. The objective comparisons show that there is good agreement in the frequency range between 0.1 and 6.4 kHz. A listening experiment was performed to validate the SHB method subjectively and to compare it to the HATS method. Two musical excerpts, pop and classical, were used. Subjective responses were collected in two head rotation conditions (fixed and rotating) and six spatial reproduction modes, including phantom mono, stereo, and surround sound. The results show that subjective scales of width, spaciousness, and preference based on the SHB method were similar to those obtained for the HATS method, although the width and spaciousness of the stimuli processed by the SHB method were judged slightly higher than the ones using the HATS method in general. Thus, binaural synthesis using SHB may be a useful tool to reproduce a 3D sound field binaurally, while saving considerably on measurement time because head rotation can be simulated based on a single recording.     

Binaural prediction of speech intelligibility in reverberant rooms with multiple noise sources

When speech is in competition with interfering sources in rooms, monaural indicators of intelligibility fail to take account of the listener's abilities to separate target speech from interfering sounds using the binaural system. In order to incorporate these segregation abilities and their susceptibility to reverberation, Lavandier and Culling [J. Acoust. Soc. Am. 127, 387-399 (2010)] proposed a model which combines effects of better-ear listening and binaural unmasking. A computationally efficient version of this model is evaluated here under more realistic conditions that include head shadow, multiple stationary noise sources, and real-room acoustics. Three experiments are presented in which speech reception thresholds were measured in the presence of one to three interferers using real-room listening over headphones, simulated by convolving anechoic stimuli with binaural room impulse-responses measured with dummy-head transducers in five rooms. Without fitting any parameter of the model, there was close correspondence between measured and predicted differences in threshold across all tested conditions. The model's components of better-ear listening and binaural unmasking were validated both in isolation and in combination. The computational efficiency of this prediction method allows the generation of complex "intelligibility maps" from room designs.     

Subjective and objective verifications of the inverse functions of binaural room impulse responses

The binaural room impulse responses (BRIRs) can be applied to 3-D sound field reconstruction, virtual reality, noise control, et al. Because the BRIRs are non-minimum phase functions, it is difficult to find the exact inverse functions of the BRIRs, especially when there are two or more sources in a reverberant space. In this work, a method was proposed to find the inverse functions of BRIRs with two sound sources in a reverberant space. The concept of time delays and the method of weighted least squares were used to find the causal, however, approximate inverse functions. The accuracy of the inverse functions was first evaluated objectively by a dummy head system. The result shows that the distortion due to crosstalk and room reverberation can be improved by 16∼18 dB. The inverse functions were also verified subjectively by 20 students. The result of subjective evaluation also shows that the inverse functions can be used successfully to reduce the crosstalk effect and the room reverberation.     

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.     

Computation of edge diffraction for more accurate room acoustics auralization

Inaccuracies in computation and auralization of room impulse responses are related in part to inadequate modeling of edge diffraction, i.e., the scattering from edges of finite surfaces. A validated time-domain model (based on analytical extensions to the Biot-Tolstoy-Medwin technique) is thus employed here to compute early room impulse responses with edge diffraction. Furthermore, the computations are extended to include combinations of specular and diffracted paths in the example problem of a stage-house. These combinations constitute a significant component of the total nonspecular scattering and also help to identify edge diffraction in measured impulse responses. The computed impulse responses are then convolved with anechoic signals with a variety of time-frequency characteristics. Initial listening tests with varying orders and combinations of diffraction suggest that (1) depending on the input signal, the diffraction contributions can be clearly audible even in nonshadow zones for this conservative open geometry and (2) second-order diffraction to nonshadowed receivers can often be neglected. Finally, a practical implementation for binaural simulation is proposed, based on the singular behavior of edge diffraction along the least-time path for a given source-edge-receiver orientation. This study thus provides a first major step toward computing edge diffraction for more accurate room acoustics auralization.     

Relationship between listening difficulty and acoustical objective measures in reverberant sound fields

The previous work [Morimoto et al., J. Acoust. Soc. Am. 116, 1607-1613] showed that listening difficulty ratings can be used to evaluate speech transmission performance more exactly and sensitively than intelligibility. Meanwhile, speech transmission performance is usually evaluated using acoustical objective measures, which are directly associated with physical parameters of room acoustic design. However, the relationship between listening difficulty ratings and acoustical objective measures was not minutely investigated. In the present study, a total of 96 impulse responses were used to investigate the relationship between listening difficulty ratings and several objective measures in unidirectional sound fields. The result of the listening test showed that (1) the correlation between listening difficulty ratings and speech transmission index (STI) is the strongest of all tested objective measures, and (2) A-weighted D(50), C(50), and center time, which are obtained from the impulse responses passed through an A-weighted filter, also strongly correlate with listening difficulty ratings, and their correlations with listening difficulty ratings are not statistically different from the correlation between listening difficulty ratings and STI.     

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.     

Phonemic restoration effect reversed in a reverberant room

Classic demonstrations of the phonemic restoration effect show increased intelligibility of interrupted speech when the interruptions are caused by a plausible masking sound rather than by silent periods. Previous studies of this effect have been conducted exclusively under anechoic or nearly anechoic listening conditions. This study demonstrates that the effect is reversed when sounds are presented in a realistically simulated reverberant room (broadband T(60) = 1.1 s): intelligibility is greater for silent interruptions than for interruptions by unmodulated noise. Additional results suggest that the reversal is primarily due to filling silent intervals with reverberant energy from the speech signal.     

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.     

Localizing nearby sound sources in a classroom: binaural room impulse responses

Binaural room impulse responses (BRIRs) were measured in a classroom for sources at different azimuths and distances (up to 1 m) relative to a manikin located in four positions in a classroom. When the listener is far from all walls, reverberant energy distorts signal magnitude and phase independently at each frequency, altering monaural spectral cues, interaural phase differences, and interaural level differences. For the tested conditions, systematic distortion (comb-filtering) from an early intense reflection is only evident when a listener is very close to a wall, and then only in the ear facing the wall. Especially for a nearby source, interaural cues grow less reliable with increasing source laterality and monaural spectral cues are less reliable in the ear farther from the sound source. Reverberation reduces the magnitude of interaural level differences at all frequencies; however, the direct-sound interaural time difference can still be recovered from the BRIRs measured in these experiments. Results suggest that bias and variability in sound localization behavior may vary systematically with listener location in a room as well as source location relative to the listener, even for nearby sources where there is relatively little reverberant energy.     

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.     

Speech intelligibility and localization in a multi-source environment.

Natural environments typically contain sound sources other than the source of interest that may interfere with the ability of listeners to extract information about the primary source. Studies of speech intelligibility and localization by normal-hearing listeners in the presence of competing speech are reported on in this work. One, two or three competing sentences [IEEE Trans. Audio Electroacoust. 17(3), 225-246 (1969)] were presented from various locations in the horizontal plane in several spatial configurations relative to a target sentence. Target and competing sentences were spoken by the same male talker and at the same level. All experiments were conducted both in an actual sound field and in a virtual sound field. In the virtual sound field, both binaural and monaural conditions were tested. In the speech intelligibility experiment, there were significant improvements in performance when the target and competing sentences were spatially separated. Performance was similar in the actual sound-field and virtual sound-field binaural listening conditions for speech intelligibility. Although most of these improvements are evident monaurally when using the better ear, binaural listening was necessary for large improvements in some situations. In the localization experiment, target source identification was measured in a seven-alternative absolute identification paradigm with the same competing sentence configurations as for the speech study. Performance in the localization experiment was significantly better in the actual sound-field than in the virtual sound-field binaural listening conditions. Under binaural conditions, localization performance was very good, even in the presence of three competing sentences. Under monaural conditions, performance was much worse. For the localization experiment, there was no significant effect of the number or configuration of the competing sentences tested. For these experiments, the performance in the speech intelligibility experiment was not limited by localization ability.     

Source excitation strategies for obtaining impulse responses in finite difference time domain room acoustics simulation

This paper considers source excitation strategies in finite difference time domain room acoustics simulations for auralization purposes. We demonstrate that FDTD simulations can be conducted to obtain impulse responses based on unit impulse excitation, this being the shortest, simplest and most efficiently implemented signal that might be applied. Single, rather than double, precision accuracy simulations might be implemented where memory use is critical but the consequence is a remarkably increased noise floor. Hard source excitation introduces a discontinuity in the simulated acoustic field resulting in a shift of resonant modes from expected values. Additive sources do not introduce such discontinuities, but instead result in a broadband offset across the frequency spectrum. Transparent sources address both of these issues and with unit impulse excitation the calculation of the compensation filters required to implement transparency is also simplified. However, both transparent and additive source excitation demonstrate solution growth problems for a bounded space. Any of these approaches might be used if the consequences are understood and compensated for, however, for room acoustics simulation the hard source is the least favorable due to the fundamental changes it imparts on the underlying geometry. These methods are further tested through the implementation of a directional sound source based on multiple omnidirectional point sources.     

考虑界面声散射的室内声脉冲响应计算机仿真新算法

本文提出一种考虑界面声散射的室内声脉冲响应的计算机仿真新算法，该算法通过应用动态堆栈和虚拟内存，解决了模拟了中计算可能失运控制的问题，作为例子，文中对二个矩形房间的声脉冲响应进行了仿真。     

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.     

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.