Feasibility of subjective speech intelligibility assessment based on auralization

Subjective speech intelligibility can be assessed by speech recorded in an anechoic chamber and then convolved with room impulse responses that can be created by acoustic simulation. The speech intelligibility (SI) assessment based on auralization was validated in three rooms. The articulation scores obtained from simulated sound field were compared with the ones from measured sound field and from direct listening in rooms. Results show that the speech intelligibility prediction based on auralization technique with simulated binaural room impulse responses (BRIRs) is in agreement with reality and results from measured BRIRs. When this technique is used with simulated and measured monaural room impulse responses (MRIRs), the predicted results underestimate the reality. It has been shown that auralization technique with simulated BRIRs is capable of assessing subjective speech intelligibility of listening positions in the room.     

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.     

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.     

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.     

Evaluation of acoustical conditions for speech communication in working elementary school classrooms

Detailed acoustical measurements were made in 41 working elementary school classrooms near Ottawa, Canada to obtain more representative and more accurate indications of the acoustical quality of conditions for speech communication during actual teaching activities. This paper describes the room acoustics characteristics and noise environment of 27 traditional rectangular classrooms from the 41 measured rooms. The purpose of the work was to better understand how to improve speech communication between teachers and students. The study found, that on average, the students experienced: teacher speech levels of 60.4 dB A, noise levels of 49.1 dB A, and a mean speech-to-noise ratio of 11 dB A during teaching activities. The mean reverberation time in the occupied classrooms was 0.41 s, which was 10% less than in the unoccupied rooms. The reverberation time measurements were used to determine the average absorption added by each student. Detailed analyses of early and late-arriving speech sounds showed these sound levels could be predicted quite accurately and suggest improved approaches to room acoustics design.     

The influence of spectral characteristics of early reflections on speech intelligibility

The auditory system takes advantage of early reflections (ERs) in a room by integrating them with the direct sound (DS) and thereby increasing the effective speech level. In the present paper the benefit from realistic ERs on speech intelligibility in diffuse speech-shaped noise was investigated for normal-hearing and hearing-impaired listeners. Monaural and binaural speech intelligibility tests were performed in a virtual auditory environment where the spectral characteristics of ERs from a simulated room could be preserved. The useful ER energy was derived from the speech intelligibility results and the efficiency of the ERs was determined as the ratio of the useful ER energy to the total ER energy. Even though ER energy contributed to speech intelligibility, DS energy was always more efficient, leading to better speech intelligibility for both groups of listeners. The efficiency loss for the ERs was mainly ascribed to their altered spectrum compared to the DS and to the filtering by the torso, head, and pinna. No binaural processing other than a binaural summation effect could be observed.     

Predictors of speech intelligibility in rooms

Three different types of acoustical measures were compared as predictors of speech intelligibility in rooms of varied size and acoustical conditions. These included signal-to-noise measures, the speech transmission index derived from modulation transfer functions, and useful/detrimental sound ratios obtained from early/late sound ratios, speech, and background levels. The most successful forms of each type of measure were of similar prediction accuracy, but the useful/detrimental ratios based on a 0.08-s early time interval were most accurate. Several physical measures, although based on very different calculation procedures, were quite strongly related to each other.     

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.     

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.     

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.     

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.     

Equal autophonic level curves under different room acoustics conditions

The indirect auditory feedback from one's own voice arises from sound reflections at the room boundaries or from sound reinforcement systems. The relative variations of indirect auditory feedback are quantified through room acoustic parameters such as the room gain and the voice support, rather than the reverberation time. Fourteen subjects matched the loudness level of their own voice (the autophonic level) to that of a constant and external reference sound, under different synthesized room acoustics conditions. The matching voice levels are used to build a set of equal autophonic level curves. These curves give an indication of the amount of variation in voice level induced by the acoustic environment as a consequence of the sidetone compensation or Lombard effect. In the range of typical rooms for speech, the variations in overall voice level that result in a constant autophonic level are on the order of 2 dB, and more than 3 dB in the 4 kHz octave band. By comparison of these curves with previous studies, it is shown that talkers use acoustic cues other than loudness to adjust their voices when speaking in different rooms.     

Measures for assessing architectural speech security (privacy) of closed offices and meeting rooms

Objective measures were investigated as predictors of the speech security of closed offices and rooms. A new signal-to-noise type measure is shown to be a superior indicator for security than existing measures such as the Articulation Index, the Speech Intelligibility Index, the ratio of the loudness of speech to that of noise, and the A-weighted level difference of speech and noise. This new measure is a weighted sum of clipped one-third-octave-band signal-to-noise ratios; various weightings and clipping levels are explored. Listening tests had 19 subjects rate the audibility and intelligibility of 500 English sentences, filtered to simulate transmission through various wall constructions, and presented along with background noise. The results of the tests indicate that the new measure is highly correlated with sentence intelligibility scores and also with three security thresholds: the threshold of intelligibility (below which speech is unintelligible), the threshold of cadence (below which the cadence of speech is inaudible), and the threshold of audibility (below which speech is inaudible). The ratio of the loudness of speech to that of noise, and simple A-weighted level differences are both shown to be well correlated with these latter two thresholds (cadence and audibility), but not well correlated with intelligibility.     

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.     

Measurement of acoustical characteristics of mosques in Saudi Arabia

The study of mosque acoustics, with regard to acoustical characteristics, sound quality for speech intelligibility, and other applicable acoustic criteria, has been largely neglected. In this study a background as to why mosques are designed as they are and how mosque design is influenced by worship considerations is given. In the study the acoustical characteristics of typically constructed contemporary mosques in Saudi Arabia have been investigated, employing a well-known impulse response. Extensive field measurements were taken in 21 representative mosques of different sizes and architectural features in order to characterize their acoustical quality and to identify the impact of air conditioning, ceiling fans, and sound reinforcement systems on their acoustics. Objective room-acoustic indicators such as reverberation time (RT) and clarity (C50) were measured. Background noise (BN) was assessed with and without the operation of air conditioning and fans. The speech transmission index (STI) was also evaluated with and without the operation of existing sound reinforcement systems. The existence of acoustical deficiencies was confirmed and quantified. The study, in addition to describing mosque acoustics, compares design goals to results obtained in practice and suggests acoustical target values for mosque design. The results show that acoustical quality in the investigated mosques deviates from optimum conditions when unoccupied, but is much better in the occupied condition.     

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.     

Wavelet packet based analysis of sound fields in rooms using coincident microphone arrays

This paper presents a passive analysis method for determining the spatio-temporal characteristics of sound fields in small rooms. The analysis finds an approximate directional reflectogram (ADR) which reveals the approximate arrival directions, time delays and amplitudes of the direct sound and early reflections without using a special or known sound source. A coincident microphone array is used to obtain directional recordings. The recordings are analysed by wavelet packet decomposition to determine the direction of the sound source and select wavelet packet coefficients to reconstruct the estimate of the direct sound. ADR is then computed via deconvolution using this estimate. Experiments have been carried out using synthesized recordings that were obtained from actual room impulse responses measured in two rooms for various source locations. The method estimates the source direction with a mean absolute error of about 7°. Calculated ADRs provide a good estimate of the time delays and arrival directions of acoustical reflections, whereas the amplitudes differ slightly.     

The effects of non-cardioid directivity on incidence angle estimation using the polar energy time curve

Assessment of desirable reflections and control of undesirable reflections in rooms are best accomplished if the reflecting surfaces are properly localized. Several measurement techniques exist to identify the incident direction of reflected sound, including the useful polar energy time curve (Polar ETC), which requires six cardioid impulse response measurements along the Cartesian axes. The purpose of this investigation is to quantify the incidence angle estimation error introduced into the Polar ETC by non-cardioid microphone directivities. The results demonstrate that errors may be minimized with a cardioid-family microphone possessing a certain range of directivities and by maximizing the measurement signal-to-noise ratio.