Directional distribution of the pseudo intensity vector in anisotropic late reverberation.

The pseudo intensity vector (PIV) is often used to analyze the directional properties of spatial room impulse responses. In the early part of the response, it is capable of estimating the directions of individual reflections. However, thus far, its behaviour in the late field is unclear. Specifically, it is unknown whether anisotropy, i.e., a direction-dependent energy distribution, is captured by the directional estimates. In this study, a closed-form expression of the directional distribution of the pressure-normalized pseudo intensity vector contingent on a general stochastical model of anisotropic fields was analytically derived. This paper shows that the probability density function of this PIV is a multivariate Cauchy distribution, which does indeed depend on the energy distribution of the field, yet the directional distribution has very limited degrees of freedom. The derived distribution is compared to the results of Monte Carlo simulations and fields captured with a microphone array in a real room. These results facilitate better understanding of the behaviour of parametric spatial room impulse response methods and may enable improved directional estimators for anisotropic fields.

The auditory perceived aperture position of the transition between rooms.

This exploratory study investigates the phenomenon of the auditory perceived aperture position (APAP): the point at which one feels they are in the boundary between two adjoined spaces, judged only using auditory senses. The APAP is likely the combined perception of multiple simultaneous auditory cue changes, such as energy, reverberation time, envelopment, decay slope shape, and the direction, amplitude, and colouration of direct and reverberant sound arrivals. A framework for a rendering-free listening test is presented and conducted in situ, avoiding possible inaccuracies from acoustic simulations, impulse response measurements, and auralisation to assess how close the APAP is to the physical aperture position under blindfold conditions, for multiple source positions and two room pairs. Results indicate that the APAP is generally within ± 1 m of the physical aperture position, though reverberation amount, listener orientation, and source position affect precision. Comparison to objective metrics suggests that the APAP generally falls within the period of greatest acoustical change. This study illustrates the non-trivial nature of acoustical room transitions and the detail required for their plausible reproduction in dynamic rendering and game audio engines.

Neural network for multi-exponential sound energy decay analysis.

An established model for sound energy decay functions (EDFs) is the superposition of multiple exponentials and a noise term. This work proposes a neural-network-based approach for estimating the model parameters from EDFs. The network is trained on synthetic EDFs and evaluated on two large datasets of over 20 000 EDF measurements conducted in various acoustic environments. The evaluation shows that the proposed neural network architecture robustly estimates the model parameters from large datasets of measured EDFs while being lightweight and computationally efficient. An implementation of the proposed neural network is publicly available.

Clearly audible room acoustical differences may not reveal where you are in a room.

A common aim in virtual reality room acoustics simulation is accurate listener position dependent rendering. However, it is unclear whether a mismatch between the acoustics and visual representation of a room influences the experience or is even noticeable. Here, we ask if listeners without any special experience in echolocation are able to identify their position in a room based on the acoustics alone. In a first test, direct comparison between acoustic recordings from the different positions in the room revealed clearly audible differences, which subjects described with various acoustic attributes. The design of the subsequent experiment allows participants to move around and explore the sound within different zones in this room while switching between visual renderings of the zones in a head-mounted display. The results show that identification was only possible in some special cases. In about 74% of all trials, listeners were not able to determine where they were in the room. The results imply that audible position dependent room acoustic rendering in virtual reality may not be noticeable under certain conditions, which highlights the importance of evaluation paradigm choice when assessing virtual acoustics.

Calibrating the Sabine and Eyring formulas.

Of the many available reverberation time prediction formulas, Sabine's and Eyring's equations are still widely used. The assumptions of homogeneity and isotropy of sound energy during the decay associated with those models are usually recognized as a reason for lack of agreement between predictions and measurements. At the same time, the inaccuracy in the estimation of the sound-absorption coefficient adds to the uncertainty of calculations. This paper shows that the error of incorrectly assumed sound absorption is more detrimental to the prediction precision than the inherent error in the formulas themselves. The proposed absorption calibration procedure reduces the differences between the measured and predicted reverberation time values, showing that an accuracy within ±10% from the target reverberation time values can be achieved regardless of the absorption distribution in a room. The paper also discusses the oft neglected air absorption of sound, which may introduce considerable bias to the measurement results. The need for an air-absorption compensation procedure is highlighted, and a method for the estimation of its parameters in octave bands is proposed and compared with other approaches. The results of this study provide justification for the use of the Sabine and Eyring formulas for reverberation time predictions.

Robust selection of clean swept-sine measurements in non-stationary noise.

The exponential sine sweep is a commonly used excitation signal in acoustic measurements, which, however, is susceptible to non-stationary noise. This paper shows how to detect contaminated sweep signals and select clean ones based on a procedure called the rule of two, which analyzes repeated sweep measurements. A high correlation between a pair of signals indicates that they are devoid of non-stationary noise. The detection threshold for the correlation is determined based on the energy of background noise and time variance. Not being disturbed by non-stationary events, a median-based method is suggested for reliable background noise energy estimation. The proposed method is shown to detect reliably 95% of impulsive noises and 75% of dropouts in the synthesized sweeps. Tested on a large set of measurements and compared with a previous method, the proposed method is shown to be more robust in detecting various non-stationary disturbances, improving the detection rate by 30 percentage points. The rule-of-two procedure increases the robustness of practical acoustic and audio measurements.

Perceptual roughness of spatially assigned sparse noise for rendering reverberation.

Multichannel auralizations based on spatial room impulse responses often employ sample-wise assignment of an omnidirectional response to form loudspeaker responses. This leads to sparse impulse responses in each reproduction loudspeaker and the auralization of transient signals can sound rough. Based on this observation, we conducted a listening test to examine the general phenomenon of roughness due to spatial assignment. First, participants assessed the roughness of both Gaussian noise and velvet noise, assigned sample-wise to up to 36 loudspeakers by two algorithms. The first algorithm assigns channels merely by selecting random indices, while the second one constrains the time between two peaks on each channel. The results show that roughness already occurs when few channels are used and that the assignment algorithm influences it. In a second experiment, virtualizations of the test were used to examine the factors contributing to increased roughness. We systematically show the effect of spatial assignment on noise and conclude that besides time-differences, level-differences caused by head-shadowing are the principal cause for the perceived roughness. The results have significance in spatial room impulse response rendering and spatial reverberator design.

Perceptual analysis of directional late reverberation.

The late reverberation characteristics of a sound field are often assumed to be perceptually isotropic, meaning that the decay of energy is perceived as equivalent in every direction. In this paper, we employ Ambisonics reproduction methods to reassess how a decaying sound field is analyzed and characterized and our capacity to hear directional characteristics within late reverberation. We propose the use of objective measures to assess the anisotropy characteristics of a decaying sound field. The energy-decay deviation is defined as the difference of the direction-dependent decay from the average decay. A perceptual study demonstrates a positive link between the range of these energy deviations and their audibility. These results suggest that accurate sound reproduction should account for directional properties throughout the decay.

Generating coherence-constrained multisensor signals using balanced mixing and spectrally smooth filters.

The spatial properties of a noise field can be described by a spatial coherence function. Synthetic multichannel noise signals exhibiting a specific spatial coherence can be generated by properly mixing a set of uncorrelated, possibly non-stationary, signals. The mixing matrix can be obtained by decomposing the spatial coherence matrix. As proposed in a widely used method, the factorization can be performed by means of a Choleski or eigenvalue decomposition. In this work, the limitations of these two methods are discussed and addressed. In particular, specific properties of the mixing matrix are analyzed, namely, the spectral smoothness and the mix balance. The first quantifies the mixing matrix-filters variation across frequency and the second quantifies the number of input signals that contribute to each output signal. Three methods based on the unitary Procrustes solution are proposed to enhance the spectral smoothness, the mix balance, and both properties jointly. A performance evaluation confirms the improvements of the mixing matrix in terms of objective measures. Furthermore, the evaluation results show that the error between the target and the generated coherence is lowered by increasing the spectral smoothness of the mixing matrix.

No dynamic visual capture for self-translation minimum audible angle.

Auditory localization is affected by visual cues. The study at hand focuses on a scenario where dynamic sound localization cues are induced by lateral listener self-translation in relation to a stationary sound source with matching or mismatching dynamic visual cues. The audio-only self-translation minimum audible angle (ST-MAA) is previously shown to be 3.3° in the horizontal plane in front of the listener. The present study found that the addition of visual cues has no significant effect on the ST-MAA.

Self-translation induced minimum audible angle.

The minimum audible angle has been studied with a stationary listener and a stationary or a moving sound source. The study at hand focuses on a scenario where the angle is induced by listener self-translation in relation to a stationary sound source. First, the classic stationary listener minimum audible angle experiment is replicated using a headphone-based reproduction system. This experiment confirms that the reproduction system is able to produce a localization cue resolution comparable to loudspeaker reproduction. Next, the self-translation minimum audible angle is shown to be 3.3° in the horizontal plane in front of the listener.

The stability of multichannel sound systems with time-varying mixing matrices.

Various time-varying algorithms have been applied in multichannel sound systems to improve the system's stability and, among these, frequency shifting has been demonstrated to reach the maximum stability improvement achievable by time-variation in general. However, the modulation artifacts have been found to diminish the gain improvement unusable for a higher number of channels and high-quality applications such as music reproduction. This paper proposes alternatively time-varying mixing matrices, which is an efficient algorithm corresponding to symmetric up and down frequency shifting. It is shown with a statistical approach that time-varying mixing matrices can as well achieve maximum stability improvement for a higher number of channels. A listening test demonstrates the improved quality of time-varying mixing matrices over frequency shifting.

Time-varying feedback matrices in feedback delay networks and their application in artificial reverberation.

This paper introduces a time-variant reverberation algorithm as an extension of the feedback delay network (FDN). By modulating the feedback matrix nearly continuously over time, a complex pattern of concurrent amplitude modulations of the feedback paths evolves. Due to its complexity, the modulation produces less likely perceivable artifacts and the time-variation helps to increase the liveliness of the reverberation tail. A listening test, which has been conducted, confirms that the perceived quality of the reverberation tail can be enhanced by the feedback matrix modulation. In contrast to the prior art time-varying allpass FDNs, it is shown that unitary feedback matrix modulation is guaranteed to be stable. Analytical constraints on the pole locations of the FDN help to describe the modulation effect in depth. Further, techniques and conditions for continuous feedback matrix modulation are presented.