We propose a method for semi-blind estimation of direct-to-reverberant energy ratio (DRR) using a number of statistical features of the residual energy test (RENT), a recently proposed direct-path metric that evaluates the dominance of a direct or diffuse component in an ambisonics recording. Our method is based on the observations and correlations between these features of this test and various recording scenarios. This method requires a procedure resembling a calibration step to tune the parameters of a linear regression model. Recordings from various locations within a room are used to extract the statistics that connect the residual energy test values to direct-to-reverberant energy ratio value estimates. For rooms with simpler geometry and volume, our method gives a median mean absolute error of 0.5 dB. For rooms featuring various materials and coupled volumes, errors can grow with increasing distances, with a median mean absolute error of 2.5 dB. The performance remains the same for speech or music recordings and is found to improve when a wider variety of sounds are used in the calibration step.

Differential microphone arrays (DMAs) use multiple omnidirectional microphones for synthesising higher-order microphone directivity patterns. In their most basic form, they can be used to obtain fixed-directivity or horizontally steerable beamformers that can satisfy certain constraints. We propose a vector differential microphone array (VDMA) which is frequency- and direction-invariantly steerable in three dimensions. The proposed design comprises pressure and particle velocity sensors positioned on a circular constellation in a plane and allows extracting the third-order spherical harmonic decomposition of the sound field. This decomposition can then be used to obtain spherically direction-invariant steered beams. Synthesis of a maximum directivity factor (MaxDF) directivity pattern is demonstrated. A closed-form expression for the proposed array's white noise gain (WNG) is derived. The robustness of the proposed design to noise is analysed.

Perceptual soundfield reconstruction (PSR) is a multichannel audio recording and reproduction framework based on time-intensity panning in the horizontal plane. A practical limitation of PSR is that the optimal directivity patterns required by the system cannot be trivially and precisely obtained in practice, and it is limited to the horizontal plane. This paper extends the horizontal PSR to three dimensions and proposes a virtual microphone synthesis approach to obtain the PSR directivity pattern via sound field extrapolation. The proposed 3D extension and virtual microphone synthesis are evaluated using numerical simulations and a subjective localisation test. Comparisons with second-order Ambisonics rendering indicate that subjects localise sources rendered using 3D PSR more accurately and also with a higher certainty, particularly at an off-centre listening position for the low-channel count reproduction system employed.

Artificial reverberators provide a computationally viable alternative to full-scale room acoustics simulation methods for deployment in interactive, immersive systems. Scattering delay network (SDN) is an artificial reverberator that allows direct parametric control over the geometry of a simulated cuboid enclosure, as well as the directional characteristics of the simulated sound sources and microphones. This paper extends the concept of SDN reverberators to multiple enclosures coupled via an aperture. The extension allows independent control of the acoustical properties of the coupled enclosures and the size of the connecting aperture. Transfer functions of the coupled-volume SDN are derived. The effectiveness of the proposed method is evaluated in terms of rendered energy decay curves in comparison to full-scale ray-tracing models and scale model measurements.

Recording multiple sound sources in a reverberant environment results in convolutive mixtures. Sound sources can be extracted from microphone array recordings of such mixtures using acoustic source separation techniques. Acoustic source separation using recordings obtained from rigid spherical microphone arrays (RSMA) benefit from the representation of sound fields as series of spherical harmonics. More specifically, RSMAs afford increased flexibility in acoustic beamforming and spatial filtering. We propose a data-driven DOA estimation and acoustic source separation method based on a dictionary-based sparse decomposition of sound fields. The proposed method involves identifying the time-frequency bins with contributions from a single source only and those with sensor noise or diffuse sound field components. The former set of bins is used in DOA estimation and beamforming in the sparse decomposition domain. The latter set is used to calculate the diffuse field covariance matrix used in Wiener post-filtering to improve the source separation performance further. We demonstrate the utility of the proposed method via extensive objective and subjective evaluations.

This article studies the effects of inter-channel time and level differences in stereophonic reproduction on perceived localization uncertainty, which is defined as how difficult it is for a listener to tell where a sound source is located. Towards this end, a computational model of localization uncertainty is proposed first. The model calculates inter-aural time and level difference cues, and compares them to those associated to free-field point-like sources. The comparison is carried out using a particular distance functional that replicates the increased uncertainty observed experimentally with inconsistent inter-aural time and level difference cues. The model is validated by formal listening tests, achieving a Pearson correlation of 0.99. The model is then used to predict localization uncertainty for stereophonic setups and a listener in central and off-central positions. Results show that amplitude methods achieve a slightly lower localization uncertainty for a listener positioned exactly in the center of the sweet spot. As soon as the listener moves away from that position, the situation reverses, with time-amplitude methods achieving a lower localization uncertainty.

The estimation of the directions-of-arrival (DOAs) of multiple sound sources is a fundamental stage in acoustic scene analysis. Many application areas such as robot audition and object-based audio (OBA) broadcast require that DOA estimation is computationally efficient to allow real-time operation. We propose a new DOA estimation approach based on a sparse representation of recorded sound fields as a linear combination of spatially bandlimited impulses in this paper. The proposed algorithm operates on a time-frequency representation of the spherical harmonic components of the sound field. We describe a residual energy test that can identify time-frequency bins with a single active source. DOA estimation is carried out at each time-frequency bin by seeking a single-source dictionary atom which provides the best match to the steered response function calculated at the selected bins. We demonstrate the accuracy of the proposed method via a set of emulations using acoustic impulse responses measured in a highly reverberant room.

Perceptual sound field reconstruction (PSR) is a spatial audio recording and reproduction method based on the application of stereophonic panning laws in microphone array design. PSR allows rendering a perceptually veridical and stable auditory perspective in the horizontal plane of the listener, and involves recording using near-coincident microphone arrays. This paper extends the PSR concept to three dimensions using sound field extrapolation carried out in the spherical-harmonic domain. Sound field rendering is performed using a two-level loudspeaker rig. An active-intensity-based analysis of the rendered sound field shows that the proposed approach can render direction of monochromatic plane waves accurately.

Estimation of the direction-of-arrival (DOA) of sound sources is an important step in sound field analysis. Rigid spherical microphone arrays allow the calculation of a compact spherical harmonic representation of the sound field. The standard method for analyzing sound fields recorded using such arrays is steered response power (SRP) maps wherein the source DOA can be estimated as the steering direction that maximizes the output power of a maximally directive beam. This approach is computationally costly since it requires steering the beam in all possible directions. This paper presents an extension to SRP called steered response power density (SRPD) and an associated, signal-adaptive search method called hierarchical grid refinement for reducing the number of steering directions needed for DOA estimation. The proposed method can localize near-coherent as well as incoherent sources while jointly providing the number of prominent sources in the scene. It is shown to be robust to reverberation and additive white noise. An evaluation of the proposed method using simulations and real recordings under highly reverberant conditions as well as a comparison with the state-of-the-art methods are presented.

Quaternion Fourier transforms (QFT) provide a powerful tool for the analysis of signals obtained from vector probes. Acoustic particle velocity is one such signal which can be measured with specially designed microphone arrays. This paper presents a time-frequency source separation method based on the short-time quaternion Fourier transform of acoustic particle velocity signals and the k-plane clustering of the vector part of the resulting representation. Two example cases, one with a single and one with two interfering sources are presented.

An acoustic reverberator consisting of a network of delay lines connected via scattering junctions is proposed. All parameters of the reverberator are derived from physical properties of the enclosure it simulates. It allows for simulation of unequal and frequency-dependent wall absorption, as well as directional sources and microphones. The reverberator renders the first-order reflections exactly, while making progressively coarser approximations of higher-order reflections. The rate of energy decay is close to that obtained with the image method (IM) and consistent with the predictions of Sabine and Eyring equations. The time evolution of the normalized echo density, which was previously shown to be correlated with the perceived texture of reverberation, is also close to that of the IM. However, its computational complexity is one to two orders of magnitude lower, comparable to the computational complexity of a feedback delay network and its memory requirements are negligible.

Affective computing is a term for the design and development of algorithms that enable computers to recognize the emotions of their users and respond in a natural way. Speech, along with facial gestures, is one of the primary modalities with which humans express their emotions. While emotional cues in speech are available to an interlocutor in a dyadic conversation setting, their subjective recognition is far from accurate. This is due to the human auditory system which is primarily non-linear and adaptive. An automatic speech emotion recognition algorithm based on a computational model of the human auditory system is described in this paper. The devised system is tested on three emotional speech datasets. The results of a subjective recognition task is also reported. It is shown that the proposed algorithm provides recognition rates that are comparable to those of human raters.

Acoustic intensity is a vectorial measure of acoustic energy flow through a given region of interest. Three-dimensional measurement of acoustic intensity requires special microphone array configurations. This paper provides a theoretical analysis of open spherical microphone arrays for the 3-D measurement of acoustic intensity. The calculations of the pressure and the particle velocity components of the sound field inside a closed volume are expressed using the Kirchhoff-Helmholtz integral equation. The conditions which simplify the calculation are identified. This calculation is then constrained to a finite set of microphones positioned at prescribed points on an open sphere. Several open spherical array topologies are proposed. Their magnitude and directional errors and measurement bandwidths are investigated via numerical simulations. A comparison with conventional open-sphere 3-D intensity probes is presented.

This paper presents a systematic framework for the analysis and design of circular multichannel surround sound systems. Objective analysis based on the concept of active intensity fields shows that for stable rendition of monochromatic plane waves it is beneficial to render each such wave by no more than two channels. Based on that finding, we propose a methodology for the design of circular microphone arrays, in the same configuration as the corresponding loudspeaker system, which aims to capture inter-channel time and intensity differences that ensure accurate rendition of the auditory perspective. The methodology is applicable to regular and irregular microphone/speaker layouts, and a wide range of microphone array radii, including the special case of coincident arrays which corresponds to intensity-based systems. Several design examples, involving first and higher-order microphones are presented. Results of formal listening tests suggest that the proposed design methodology achieves a performance comparable to prior art in the center of the loudspeaker array and a more graceful degradation away from the center.

Multichannel dereverberation amounts to the inversion of a multiple-input/multiple-output linear time-invariant system. In this paper, necessary and sufficient conditions for perfect dereverberation using stable and finite impulse response (FIR) filters are established. It is then shown that the inverse system given by the pseudoinverse of the original transfer function matrix exhibits a noise reduction property. A necessary and sufficient condition under which this pseudoinverse system is FIR is also given. Further, an FIR approximation to the pseudoinverse system is considered and the effects of the length of this approximation on the dereverberation accuracy are investigated. Finally, an analytical and numerical assessment of the dependence of the dereverberation accuracy on the accuracy of the acquisition of room impulse responses is provided.

A novel systematic approach to the design of directivity patterns of higher order differential microphones is proposed. The directivity patterns are obtained by optimizing a cost function which is a convex combination of a front-back energy ratio and uniformity within a frontal sector of interest. Most of the standard directivity patterns—omnidirectional, cardioid, subcardioid, hypercardioid, supercardioid—are particular solutions of this optimization problem with specific values of two free parameters: the angular width of the frontal sector and the convex combination factor. More general solutions of practical use are obtained by varying these two parameters. Many of these optimal directivity patterns are trigonometric polynomials with complex roots. A new differential array structure that enables the implementation of general higher order directivity patterns, with complex or real roots, is then proposed. The effectiveness of the proposed design framework and the implementation structure are illustrated by design examples, simulations, and measurements.

Spherical microphone arrays provide a flexible solution to obtaining higher-order directivity patterns, which are useful in audio recording and reproduction. A general systematic approach to the design of directivity patterns for spherical microphone arrays is introduced in this paper. The directivity patterns are obtained by optimizing a cost function which is a convex combination of a front-back energy ratio and a smoothness term. Most of the standard directivity patterns - i.e. omnidirectional, cardioid, subcardioid, hypercardioid and supercardioid - are particular solutions of this optimization problem with specific values of two free parameters: the angle of the frontal sector, and the convex combination factor. By varying these two parameters, more general solutions of practical use are obtained.

Digital waveguide mesh (DWM) models are time-domain numerical methods providing computationally simple solutions for wave propagation problems. They have been used in various acoustical modeling and audio synthesis applications including synthesis of musical instrument sounds and speech, and modeling of room acoustics. A successful model of room acoustics should be able to account for source and receiver directivity. Methods for the simulation of directional sources in DWM models were previously proposed. This paper presents a method for the simulation of directional microphones in DWM-based models of room acoustics. The method is based on the directional weighting of the microphone response according to the instantaneous direction of incidence at a given point. The direction of incidence is obtained from instantaneous intensity that is calculated from local pressure values in the DWM model. The calculation of instantaneous intensity in DWM meshes and the directional accuracies of different mesh topologies are discussed. An intensity-based formulation for the response of a directional microphone is given. Simulation results for an actual microphone with frequency-dependent, non-ideal directivity function are presented.

In this paper, a new morphing algorithm for transient sounds is introduced. Input sounds are first projected onto orthogonal bases from which intermediate or novel sounds can be generated. The proposed algorithm uses a shift invariant version of discrete wavelet transform and the singular value decomposition (SVD) to represent the input sound signals over a set of orthogonal bases. Interpolation is carried out between the weight vectors from the SVD to produce a new weight vector used for synthesising a new set of wavelet coefficients. The morphed sound is generated by taking the inverse discrete wavelet transform of the combined weighted bases. The proposed algorithm not only generates a range of new sounds, but also represents the input sounds in a more compact fashion.

This article presents a technique that uses the intensity vector direction exploitation (IVDE) method for exhaustive separation of convolutive mixtures. While only a four-element compact sensor array is used, multiple channels for all possible source directions are produced by exhaustive separation. Singular value decomposition (SVD) is then applied to determine the signal subspace and the directions of the local maxima of the signal energy. This information is then used to select the channels containing the individual sources. While the original IVDE method requires the prior knowledge of the directions of sources for separation, the present method eliminates this need and achieves fully blind separation. Performing SVD at a post-processing stage also improves the sound quality. The method has been tested for convolutive mixtures of up to four sources and typical separation performances are given.

Digital waveguide mesh (DWM) models are numerical solvers for the wave equation in -dimensions. They are used for obtaining the traveling-wave solution in practical acoustical modeling applications. Although unstructured meshes can be used with DWMs, regular mesh topologies are traditionally used due to their implementation simplicity. This letter discusses the accuracy of first-order approximations to numerical derivatives on more general unstructured mesh topologies. The results are applied to structured, regular mesh topologies as used in DWM modeling. A comparison of 2-D and 3-D DWM topologies with respect to the accuracy of first-order approximations to numerical derivatives is presented.

Various techniques have previously been proposed for the separation of convolutive mixtures. These techniques can be classified as stochastic, adaptive, and deterministic. Stochastic methods are computationally expensive since they require an iterative process for the calculation of the demixing filters based on a separation criterion that usually assumes that the source signals are statistically independent. Adaptive methods, such as the adaptive beamformers, also exploit signal properties in order to optimize a multichannel filter structure. However, these algorithms need initialization and time to converge. Deterministic methods, on the other hand, provide a closed-form solution based on the deterministic aspects of the problem, such as the channel characteristics and the source directions. This paper presents a technique that exploits the intensity vector statistics to achieve a nearly closed-form solution for the separation of the convolutive mixtures as recorded with a coincident microphone array. No assumptions are made on the signals, but it is assumed that the source directions are known a priori. Directivity functions based on von Mises functions are designed for beamforming depending on the circular statistics of the calculated intensity vectors. Numerical evaluation results were presented for various speech and instrument sounds and source positions in two reverberant rooms.

Digital waveguide mesh (DWM) models offer a simple, accurate, time-domain, numerical solution of the wave equation. A specific case where such accurate and computationally simple solutions are needed is the acoustical modeling of open or closed volumes. It is possible to model 3-D propagation of waves in enclosures such as rooms using DWM models. Generally, idealized omnidirectional sources are used for obtaining the impulse response in the DWM. However, real-life sound sources are never completely isotropic, causing wavefronts with directional properties. This paper presents two methods to simulate analytical and empirical directivities in 3-D DWM models in the far-field. The first method is based on the direct excitation of the mesh with the velocity component of the directional source and is used to simulate analytical sources. The second method is based on the weighting of velocity components generated by an omnidirectional source at different octave-bands and is used to simulate sources with frequency-dependent empirical directivity functions. A simple interpolation method for obtaining a closed-form description of the directivity function from incomplete directivity data is also proposed. Simulation results are presented for two sources in an acoustical model of a rectangular room.

Fractional-delay filter is the general name given to filters modelling non-integer delays. Such filters have a flat phase delay for a wide frequency band, with the value of the phase delay approximating the fractional delay. A maximally-flat delay IIR fractional-delay filter can be obtained by the Thiran approximation. A simple and efficient method for obtaining filters modelling intermediate fractional delays from two Thiran fractional-delay filters is proposed. The proposed method allows continuously modifying the fractional delay. Computational complexity of the proposed method is discussed. A practical application of the method in model-based sound synthesis is given as an example