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Acoustic measurements are susceptible to various sources of measurement uncertainty. One significant factor is loudspeaker directivity, which introduces temporal smearing and spectral coloration into room impulse responses (RIRs), predominantly influencing early reflections. Such an artifact affects parametric processing and perceptual evaluation of RIRs and lowers the measurement reproducibility. This study evaluates the impact of loudspeaker directivity on measured RIRs. We acquire directivity filters via measurements in an anechoic chamber, utilizing a custom-made microphone arc. Subsequently, we both capture a series of RIRs in a typical reverberant room and simulate corresponding RIRs with the image-source method (ISM). By convolving the simulations with the correct directivity filters, we match the early reflections of measured and simulated RIRs. Examining the cross-correlation between the simulated and measured RIRs reveals a pronounced likeness for first-order reflections, indicating a substantial influence of the loudspeaker directivity on recorded RIRs. This study is a step towards accounting for the influence of the sound source type and position on RIRs, resulting in better-informed acoustic measurements and higher fidelity of acoustic simulations.
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The objective evaluation of speech intelligibility of sound systems using the Speech Transmission Index (STI) is nowadays common practice. This paper discusses various aspects of optimizing intelligibility of sound systems for public spaces as driven by regulatory standards and client requirements. New workflows, utilizing acoustic modelling software and targeted towards the optimization of the STI, are showcased. Parameters such as predicted loudspeaker coverage, reflections, signal and noise levels, and corresponding spectra are considered. Special emphasis is put on the investigation of the effects of signal and noise masking. Results from real-world examples reveal a strong correlation between STI predictions and measurements.
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In active acoustics, signals from microphones within a room are processed and fed to loudspeakers in the same room, creating an extended reverberation time and modified room perception. The system’s performance is limited by the audibility and acceptability of colouration at gains close to instability. Some listening tests have been presented in the literature to assess perceptual colouration, but thresholds for when the colouration becomes annoying or unacceptable have not previously been established. In this paper, we revisit the prediction of the gain before instability and show how this can be used to equalize an active acoustics system. Then, we present new listening tests where listeners were asked to rate the audibility and annoyance of changes introduced by 8 channel active acoustics systems in two rooms at various simulated gains. We show that the annoyance depends on the initial room acoustics as well as the loop gain; perceptual thresholds for slightly annoying degradation varied from?5.4 dB to ?8.5 dB, relative to instability. These thresholds are discussed in the context of objective measurements calculated from the impulse responses. The resonance perception is linked to the gain where the reverberation time starts to grow much more quickly in some frequency bands than others. It is also shown to be well predicted by the standard deviation of the magnitude response, with a value of 0.62 corresponding to slightly annoying degradation.
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The recently launched WHO Global Standard for Safe Listening Venues and Events aims to make listening safer and more enjoyable for audiences around the world. Some key questions remain on how to practically monitor sound exposure as well as on how patrons’ hearing may be affected after significant exposure. This paper presents a case study where various sound exposure monitoring systems and methods were trialed in an indoor music venue. The aim of the work was to develop and validate a practical, accurate and repeatable technique to track sound exposure across music venues that can be presented in real-time. Results indicate that this can be achieved with no more than four, and as few as two, sound level monitoring locations alongside fixed calibration measurements and a small number of spot measurements at the mix position during a performance.
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Progressively curved line-source arrays became state of the art in large-scale sound reinforcement as they are flexibly adapted to the listening area by well-chosen splay angles between individual elements in the chain of line-source loudspeakers. Recent perceptual studies suggest using a dual-target design to meet contradictory goals in immersive sound reinforcement: 0 dB per doubling of the distance to preserve the direct sound mix, and -3 dB per doubling of the distance to preserve the envelopment at off-center listening positions. A practical implementation has been proposed to achieve both objectives simultaneously by driving the transducers of a curved array either in phase or with individual delays. Its feasibility was verified using measurements on a miniature line array that works for small audiences, but not specifically for large arrays and audiences that would be typically found in live events. To check the applicability of the dual-target approach to large-scale sound reinforcement systems, this contribution presents a simulation study with various professional line-source arrays and a sample measurement.
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Listener envelopment and listener engulfment refer to the sensations of being ’surrounded by sound’ and ’being covered by sound’, respectively. In multichannel loudspeaker arrangements, listeners at off-center seats typically experience a reduced sensation of envelopment and engulfment due to a directional imbalance towards nearby loudspeakers. The experiment presented in this study investigates the effect of different loudspeaker sound pressure level (SPL) decay profiles on the off-center distance limit, at which envelopment or engulfment break down. Three different profiles are considered: 0, -3, and -6 dB SPL decay per doubling of distance, simulated by controlling the levels of point-source loudspeakers based on the listener position. The experiment results indicate a significant expansion of the off-center limit of envelopment when horizontally surrounding loudspeakers exhibit a -3 dB SPL decay. Regarding engulfment, the experiment shows that the off-center limit is expanded by a wide distribution of height loudspeakers that covers the entire audience area. A computational model confirms that the optimal loudspeaker SPL decay for envelopment is the one that minimizes the interaural level difference (ILD) and interaural coherence (IC) over an extended area. An interesting finding from simulations is that purely lateral multichannel arrangements can benefit from a 0 dB rather than -3 dB SPL decay per doubling of distance.
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Spatial audio is employed more and more often in large-scale live music events. In events of this kind, loudspeakers can be widely spaced apart, which may result in large time differences of arrival between certain sources. These timing differences may in turn affect the perceived rhythmic quality of music, or groove, as the synchronization between instruments is modified. This paper presents the results of a perceptual experiment that investigated how different factors, such as the nature of the instrument or the musical genre, impact the perceived groove modification resulting from sound propagation time differences. The results indicate that different instruments can show more or less sensitivity to time shifts, even in the same musical excerpt. Based on these findings, we derive mixing and sound system design guidelines that aim at preserving an optimal musical quality for the majority of the audience.
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Most sound reinforcement systems employ a combination of full-range speakers and subwoofers to deliver a consistent sound pressure level over the audience, while maximizing the frequency bandwidth. A time alignment between the main (full-range) and sub (subwoofers) systems is generally required to ensure an efficient summation at low frequencies. This study investigates how time misalignments between the main and sub systems affect the perceived sound quality. We conducted a listening test whereby the listeners were asked to rate the sound quality as a function of the relative delay between main and sub systems. In addition, test participants were requested to qualify the nature of the perceived artifacts using spectral or temporal attributes. Our results suggest that the overall perceived quality does not decrease linearly with increasing delays, and that it reflects the presence of both spectral and temporal degradations. Lastly, temporal degradations are perceived more often when the sub system is delayed with respect to the main system, unlike spectral degradations for which the direction of the delay has very little influence.
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In this study, the directivity index, which is commonly used to characterize the radiation pattern of individual sound sources, is applied to assess an entire stage configuration concerning sound reinforcement system design, sound exposure, and noise control in the low-frequency range. In a parametric study, the influence on the value is investigated for two different datasets derived from measurements and simulations. For each dataset, the dependencies of input parameters, namely the angular resolution for uniform and non-uniform angle vectors and the distance, are evaluated. The results indicate that the directivity index is a metric that can be used to quickly evaluate the radiation pattern of a sound reinforcement system. Regardless of the dataset, the results show that the angular resolution can be significantly reduced compared to high-resolution loudspeaker models without having a large impact on the calculation. To use the directivity index as a criterion for noise control and sound system design, it must be integrated with other considerations, such as the energy radiated toward neighborhoods in need of protection and thus the absolute sound pressure levels in these areas as well as for the audience and staff.
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This study compares the efficiency and power consumption of typical designs for various acoustic loads such as ported, band-pass, horn, and tapped horn boxes for low-frequency reproduction. The study is based on Boundary Elements Method simulations. Several metrics are used as benchmarks: sensitivity, efficiency and power for a target level. Results show that sensitivity and enclosure volume are not necessarily related to efficiency. The importance of weighting and equalization is also explained.
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