Relationships among various room acoustic indices

In 1965, the Catholic Church liturgy changed to allow priests to face the congregation. Whereas Church tradition, teaching, and participation have been much discussed with respect to priest orientation at Mass, the acoustical changes in this regard have not yet been examined scientifically. To discuss acoustic desired within churches, it is necessary to know the acoustical characteristics appropriate for each phase of the liturgy. In this study, acoustic measurements were taken at various source locations and directions using both old and new liturgies performed in Japanese churches. A directional loudspeaker was used as the source to provide vocal and organ acoustic fields, and impulse responses were measured. Various acoustical parameters such as reverberation time and early decay time were analyzed. The speech transmission index was higher for the new Catholic liturgy, suggesting that the change in liturgy has improved speech intelligibility. Moreover, the interaural cross-correlation coefficient and early lateral energy fraction were higher and lower, respectively, suggesting that the change in liturgy has made the apparent source width smaller.

Does the location of the source affect the results of an acoustical measurement? This is the question that sparked the author’s Master’s Project which explores the differences between measurements taken with a source located on the stage of a performance space and a source located higher above the stage using the space’s existing sound system. Impulse responses were gathered from four different performance halls with respect to source location, microphone location, and measurement devices used. Comparisons were made between trends in reverberation time, early decay time, and interaural cross-correlation coefficient. The results are not only interesting, but they also question the typical measurement practices of acousticians and confirm assumptions made regarding important acoustical characteristics of performance spaces.

This article aims to review the development of acoustic computer simulation for performance spaces. The databases of Web of Science and Scopus were searched for peer-reviewed journal articles published in English between 1960 and 2021, using the keywords for “simulation”, “acoustic”, “performance space”, “measure”, and their synonyms. The inclusion criteria were as follows: (1) the searched article should be focused on the field of room acoustics (reviews were excluded); (2) a computer simulation algorithm should be used; (3) it should be clearly stated that the simulated object is a performance space; and (4) acoustic measurements should be used for comparison with the simulation. Finally, twenty studies were included. A standardised data extraction form was used to collect the modelling information, software/algorithm, indicators for comparison, and other information. The results revealed that the most used acoustic indicators were early decay time (EDT), reverberation time (T30), strength (G), and definition (D50). The accuracy of these indicators differed greatly. For non-iterative simulation, the simulation accuracies of most indicators were outside their respective just noticeable differences. Although a larger sample size was required for further validation, simulations of T30, EDT, and D50 all showed an increase in accuracy with increasing time from 1979 to 2020, except for G. In terms of frequency, the simulation was generally less accurate at lower frequencies, which occurred at T30, G, D50 and T20. However, EDT accuracy did not exhibit significant frequency sensitivity. The prediction accuracy of inter-aural cross-correlation coefficients (IACC) was even higher at low frequencies than it was at high frequencies. The average value of most indicators showed a clear systematic deviation from zero, providing hints for future algorithm improvements. Limitations and the risks of bias in this review were discussed. Finally, various types of benchmark tests were suggested for various comparison goals.

Acoustic metrics are commonly expressed as single numbers in classroom acoustical designs, often neglecting the physical quantity's uncertainty due to the non-diffuse sound field in the seating area. A database of measured monaural and binaural room impulse responses (RIR) was previously gathered from a fully furnished mock-up classroom. Different wall and ceiling absorption configurations were used to alter the mid-frequency reverberation times (RT) in five scenarios between 0.4 and 1.1 s. The middle three RT scenarios were additionally created from two different material configurations. For each material configuration (eight in total), two furniture orientations were utilized. RIRs were measured at 9 to 10 receiver positions for each material/furniture configuration to document the spatial variation in the resulting sound field. Diffuseness has been calculated for each receiver position utilizing the measured RIRs by following Hanyu's (2013) method using normalized decay-canceled impulse responses. Variations in diffuseness and in the assorted acoustic metrics calculated from the measured RIRs are investigated across different receiver positions. These acoustic metrics, pertinent to classroom acoustical designs, include RT, speech transmission index, clarity, and interaural cross-correlation. Means to quantify uncertainty in these metrics due to spatial variation in the non-diffuse sound field will be discussed.

Does the location of the source affect the results of an acoustical measurement? This is the question that sparked the author's Master's Project which explores the differences between measurements taken with a source located on the stage of a performance space and a source located higher above the stage using the space's existing sound system. Impulse responses were gathered from four different performance halls with respect to source location, microphone location, and measurement devices used. Comparisons were made between trends in reverberation time, early decay time, and interaural cross-correlation coefficient. The results are not only interesting, but they also question the typical measurement practices of acousticians and confirm assumptions made regarding important acoustical characteristics of performance spaces.

Reverberation time (RT) is an important parameter for room acoustics characterization, intelligibility and quality assessment of reverberant speech, and for dereverberation. Commonly, RT is estimated from the room impulse response (RIR). In practice, however, RIRs are often unavailable or continuously changing. As such, blind estimation of RT based only on the recorded reverberant signals is of great interest. To date, blind RT estimation has focused on reverberant speech signals. Here, we propose to blindly estimate RT from non-speech signals, such as solo instrument recordings and music ensembles. To estimate the RT of non-speech signals, we propose a blind estimator based on an auditory-inspired modulation spectrum signal representation, which measures the modulation frequency of temporal envelopes computed from a 23-channel gammatone filterbank. We show that the higher modulation frequency bands are more sensitive to reverberation than the modulation bands below 20 Hz. When tested on a database of non-speech sounds under 23 different reverberation conditions with early decay time (EDT) ranging from 0.26 to 7.6 s, a blind estimator based on the ratio of high-to-low modulation frequencies outperformed two state-of-the-art methods and achieved correlations with EDT as high as 0.80 for solo instruments and 0.75 for ensembles.

Large acoustically coupled volumes can be found in a variety of places such as churches, factories, train stations, and concert halls, where sound reverberation occurs. In recent decades, coupled volume concert halls have garnered increasing attention (Beranek, 2004) as the use of reverberation chambers can provide both sound clarity and reverberance due to the nonexponential temporal sound energy decay. Furthermore, variations of clarity and reverberance are essentially controlled by modifying the coupling area between the adjacent volumes. The present study focuses on the appreciation of changes in reverberation in terms of perceived suitability for different musical instruments or ensembles. Reverberation is often described in terms of reverberation time (reaction time [RT]), defined as the time required for the energy of a stopped sound to decay by 60 dB. However this metric does not provide a unique quantification of the decay profile and is therefore not sufficient to characterize nonexponential decays. To focus the study on physically possible variations of a given architectural space, the present study relies on variations of an architectural element, the coupling area, which embeds acoustical parameter variations, estimated from room impulse responses. Participants were asked to rate the suitability of reverberation settings for different instruments and ensembles. The sound stimuli were synthesized with an auralization method, providing control over the coupling area.Previous ResearchSingle volume concert halls have been widely studied while in comparison little research has been conducted on perceptual aspects of coupled volume concert halls. Perceptual thresholds, or just noticeable differences (JND), based on reverberation time have been estimated in large single volume rooms (Billon & Embrechts, 2012; Katz, 2004; Seraphim, 1958) and were found to lie between 5% and 10% of a given RT value (for RT 1.0 sec). In coupled volume concert halls, a recent study (Luizard et al., 2013; Luizard et al., 2015) based on an architectural parameter whose variations affect the room reverberance found a perceptual threshold of changes of 10% of a given open coupling area.Previous research has determined that although discrimination performance does not seem to depend on sound material (Frissen et al., 2010), reverberation preference, or evaluation, depends on the sound material (Frissen et al., 2010; Kuhl, 1954) proposed to the listeners. Preference for reverberation in the context of concert hall design has been previously studied (Ando et al., 1997) with the definition of four subjective criteria: sound level, initial time delay gap, reverberance, and interaural cross correlation; these acoustical parameters being defined in the ISO standard (ISO 3382-1, 2009). Another study (Ando et al., 1982) in single volume concert halls has shown that long reverberation times (around 2.7 sec.) were preferred for baroque music played with an orchestra, slow classical music required lower RTs (around 1.2 sec.) and preferred RTs for speech were 0.3 sec. However, another research group (Schroeder et al., 1974) has tested one of these classical music pieces using the characteristics of 11 concert halls and listeners preferred the stimuli corresponding to an RT 2.0 sec. This discrepancy can be explained by variations in stimuli parameters other than RT, such as initial time delay gap or interaural cross correlation, which are linked to spatial aspects and depend on the room geometry. Preference of reverberation in coupled volumes as compared to single-slope reverberation has been studied (Ermann, 2007) by means of listening tests with sound stimuli generated by ray-tracing software with no significant differences being observed between single-slope and double-slope reverberation.Preference of reverberation within coupled spaces has also been studied (Bradley & Wang, 2010) by means of geometrical acoustics simulations with participants preferring low to medium levels of decay curvature, as compared to high levels of curvature and classical linear slope exponential reverberation. …

This paper presents a historical and architectural review of the Usina del Arte Symphony Hall, the main concert hall of Buenos Aires, Argentina, and the definitive host of the Symphonic and Philharmonic National orchestras. A complete acoustic study of the hall is carried out. ISO 3382 is complied in order to obtain multiple impulse responses, including monoaural, binaural and 3D sound field recordings. With the purpose of characterizing the hall, several acoustical parameters are evaluated: reverberation time (RT30), early decay time (EDT), definition (D50), clarity (C80), speech intelligibility (STI, RaSTI), inter-aural cross correlation (IACC) and background noise (NC curve), including 3D impulse response mapping and the sound field diffusion coefficient (SFDC).

We introduce a Virtual Studio Technology (VST) 2 audio effect plugin that performs convolution reverb using synthetic Room Impulse Responses (RIRs) generated via a Genetic Algorithm (GA). The parameters of the plugin include some of those defined under the ISO 3382-1 standard (e.g., reverberation time, early decay time, and clarity), which are used to determine the fitness values of potential RIRs so that the user has some control over the shape of the resulting RIRs. In the GA, these RIRs are initially generated via a custom Gaussian noise method, and then evolve via truncation selection, random weighted average crossover, and mutation via Gaussian multiplication in order to produce RIRs that resemble real-world, recorded ones. Binaural Room Impulse Responses (BRIRs) can also be generated by assigning two different RIRs to the left and right stereo channels. With the proposed audio effect, new RIRs that represent virtual rooms, some of which may even be impossible to replicate in the physical world, can be generated and stored. Objective evaluation of the GA shows that contradictory combinations of parameter values will produce RIRs with low fitness. Additionally, through subjective evaluation, it was determined that RIRs generated by the GA were still perceptually distinguishable from similar real-world RIRs, but the perceptual differences were reduced when longer execution times were used for generating the RIRs or the unprocessed audio signals were comprised of only speech.

Previous studies have found that the speech intelligibility of Mandarin monosyllables is in direct correlation to the effective duration of the autocorrelation function (e) of the syllable itself at the same reverberation levels (Chen and Chan, 2006; Chen, 2012). In addition, it has been further found (Chen, 2004) that the evaluation of a sound field is the opposite between the speech transmission index (STI, proposed by Steeneken and Houtgast, 1973) and magnitude of the interaural cross-correlation coefficient (IACC) where the slope of the ceiling was changed in a hall in a constricted range of STI (0.5~0.7). Takaoka et al. (2007) once used noises and utterances in Japanese to examine the influence of a sound field's reverberation time and the IACC on speech articulation. They found that under an IACC condition where the SNR (signal-to-noise ratio) was between 0 dB ~10 dB and reverberation time varied between 0.5s~4.0s, no obvious changes were noticed in speech articulation, and that only when SNR was lower than-10dB, did the IACC affect speech articulation within a range of IACC limited to between 0.5~1.0. Accordingly, this study focuses on a broadened IACC range (0.34~0.87), and adopts paired comparison to identify the relationship between word intelligibility and the IACC with or without subsequent reverberation energy in detail. The results show that the word intelligibility of a monosyllable could be affected by reverberation time (p＜0.001), but not by IACC. The effect of the IACC was significant only in an environment with reverberation energy (p＜0.05).

Performance spaces are characterized by a complex sound field, due to the presence of absorptive and diffusive surfaces. In situ evaluations of the acoustic effects that these surfaces have on the objective acoustic parameters and on sound perception have not yet been fully understood. To this aim, acoustic measurements have been performed in a variable-acoustic concert hall, the Espace de Projection, at the Institut de Recherche et Coordination Acoustique/Musique. These measurements have allowed the effects of one single wall to be determined. A diffusive and a reflective condition of one of the long lateral walls of the shoebox-like hall have been considered, while the other surfaces have been fixed in absorptive mode. Measurements have been carried out at different distances from the test wall, using an artificial head and an array of omnidirectional microphones. Objective acoustic parameters, such as early decay time, reverberation time (T30), clarity (C80), definition (D50), and interaural cross correlation, have been compared between both conditions. In addition to the objective indexes, a perceptual evaluation has been performed using listening tests that had the purpose of determining the maximum distance from a diffusive surface at which acoustic scattering effects are still audible.

The generalized spectral subtraction algorithm (GBSS), which has extraordinary ability in background noise reduction, is historically one of the first approaches used for speech enhancement and dereverberation. However, the algorithm has not been applied to de-noise the room impulse response (RIR) to extend the reverberation decay range. The application of the GBSS algorithm in this study is stated as an optimization problem, that is, subtracting the noise level from the RIR while maintaining the signal quality. The optimization process conducted in the measurements of the RIRs with artificial noise and natural ambient noise aims to determine the optimal sets of factors to achieve the best noise reduction results regarding the largest dynamic range improvement. The optimal factors are set variables determined by the estimated SNRs of the RIRs filtered in the octave band. The acoustic parameters, the reverberation time (RT), and early decay time (EDT), and the dynamic range improvement of the energy decay curve were used as control measures and evaluation criteria to ensure the reliability of the algorithm. The de-noising results were compared with noise compensation methods. With the achieved optimal factors, the GBSS contributes to a significant effect in terms of dynamic range improvement and decreases the estimation errors in the RTs caused by noise levels.

The room acoustical parameters, the reverberation time (RT), the early decay time (EDT), the bass ratio (BR), the clarity factor (C80), the initial time delay gap (ITDG), and the interaural cross-correlation coefficient (IACC) were measured for acoustical evaluation both in the audience seats and on the stage, using six different stage configurations in a 450-seat unoccupied hall. The stage configurations consisted of combinations of varying the position of the drapes with and without the reflector at the back of the stage. The listening test was conducted in a laboratory using high fidelity headphones to verify the subjective preference for the seats and the presence of the reflector. Analysis showed that the draped area of the stage was related to the RT, EDT, C80, BR, while the ITDG was strongly dependent on the presence of the reflector. There was not a strong impact observed in the IACC for the varying configurations. In general, the reflectors improved most of the acoustical parameters. The loudness of the musical sound was the primary parameter used to decide the subjective preference, which correlated with the C80 and the ITDG with the reflector.

Effect of listening level and background noise on the subjective decay rate of room impulse responses: Using time-varying loudness to model reverberance

A new method, employing machine learning techniques and a modified low frequency envelope spectrum estimator, for estimating important room acoustic parameters including Reverberation Time (RT) and Early Decay Time (EDT) from received music signals has been developed. It overcomes drawbacks found in applying music signals directly to the envelope spectrum detector developed for the estimation of RT from speech signals. The octave band music signal is first separated into sub bands corresponding to notes on the equal temperament scale and the level of each note normalised before applying an envelope spectrum detector. A typical artificial neural network is then trained to map these envelope spectra onto RT or EDT. Significant improvements in estimation accuracy were found and further investigations confirmed that the non-stationary nature of music envelopes is a major technical challenge hindering accurate parameter extraction from music and the proposed method to some extent circumvents the difficulty.