It can be challenging for opera singers to access their performance venues for rehearsal due to venue sched-ules or travel restrictions. Virtual reality (VR) or augmented reality (AR) technologies provide the possibility for rehearsal in a virtual venue. However, these technologies are mainly focused on visuals, rather than on the sonic plausibility of a virtual space. Moreover, existing self-auralization methods have one or more of the following limitations: a small choice of virtual venues, restricted user movement, generic rather than individualized configuration, and an expensive rehearsal space. This paper presents an Ambisonics method that addresses these limitations. The method simulates the acoustics of a chosen performance venue in real-time, by simulating oral-binaural room impulse responses (OBRIRs). This method allows changing the virtual venue and user-related data, and provides three-degrees-of-freedom (3DoF) for user head movement. The method is validated using quantitative and qualitative methods, and challenges of future real-time implementation are discussed. Despite the challenges, the method is capable of facilitating virtual rehearsal in real-time while providing for greater user flexibility.

This paper derives a superfast Toeplitz matrix inversion method using FFT techniques. The approach casts the non-periodic Toeplitz matrix solution into a periodic framework and uses the Woodbury matrix inverse identity to express the non-periodic solution as a correction to the time-windowed periodic solution. It is shown that the solution can be implemented using FFT techniques that avoid the calculation of a large matrix inverse. The method is implemented for both single- and multi-channel cases and typical computation times and accuracy are reported for the equalization of simulated and measured room impulse responses.

This paper describes how to encode into a spherical-harmonic representation of a 3-D sound field using a cylindrical microphone array. The standard way to represent a sound field is to use higher order Ambisonics specified over a spherical region. If the sound field is instead measured using a cylindrical array, it is more naturally expressed in terms of cylindrical harmonics. In this paper, the transforms for conversion between cylindrical and spherical-harmonic representations are derived. Simulations then compare the performance of cylindrical and spherical array processing in estimating the spherical-harmonic coefficients of a sound field.

In this paper, a pair of analytic expressions describing the correlation functions due to spherically symmetric radial Gaussian and uniform volume near-field source position distributions are presented. An approximate spatial correlation function solution for a radial Gaussian source location distribution is derived and compared with the existing numerical methods. An exact solution for a uniform volume source location distribution is also derived and compared with existing numerical methods. The approximate radial Gaussian solution produces a result consistent with numerical methods for compact source location distributions. The uniform volume solution matches the expected behavior. Finally, the spatial correlation functions were used to design spatially robust beamformers for compact microphone arrays. Both of the spatial correlation function solutions lead to improved spatial robustness for the applications of signal enhancement and suppression.

In this paper, a method of adaptive noise suppression combining spatially robust fixed beamforming and the TRINICON blind source separation algorithm is presented. A multichannel sensor array is first processed using complementary fixed beamformers into maximum and minimum SINR channels. The channels form the inputs to a single 2×2 second-order statistics TRINICON-BSS system which adaptively compensates for imperfections of the fixed beamformer design relative to the acoustic scenario. It is demonstrated that integrating the TRINICON-BSS algorithm leads to improved SINR performance over the initial imperfect beamformer design, and achieves a performance comparable to a perfect MVDR beamformer.

This paper presents spatially robust far-field microphone beamformers and nullformers derived using the von Mises and von Mises-Fisher distributions to model the expected direction of arrival. Simple analytic expressions are presented for 2D and 3D far-field correlation functions and used to design spatially robust beamformers and nullformers. It is demonstrated that the spatially robust beamformers show a modest improvement in tolerating uncertainty in the target direction of arrival without incurring a significant penalty in terms of SINR performance compared with the MVDR beamformer. In addition, the spatially robust formulation shows significantly improved numerical robustness, indicating improved ability in tolerating intrinsic array errors.

This paper proposes an efficient parameterization of the room transfer function (RTF). Typically, the RTF rapidly varies with varying source and receiver positions, hence requires an impractical number of point to point measurements to characterize a given room. Therefore, we derive a novel RTF parameterization that is robust to both receiver and source variations with the following salient features: 1) The parameterization is given in terms of a modal expansion of 3D basis functions. 2) The aforementioned modal expansion can be truncated at a finite number of modes given that the source and receiver locations are from two sizeable spatial regions, which are arbitrarily distributed. 3) The parameter weights/coefficients are independent of the source/receiver positions. Therefore, a finite set of coefficients is shown to be capable of accurately calculating the RTF between any two arbitrary points from a pre-defined spatial region where the source(s) lie and a pre-defined spatial region where the receiver(s) lie. A practical method to measure the RTF coefficients is also provided, which only requires a single microphone unit and a single loudspeaker unit, given that the room characteristics remain stationary over time. The accuracy of the above parameterization is verified using appropriate simulation examples.

Successful recording of large spatial soundfields is a prevailing challenge in acoustic signal processing due to the enormous numbers of microphones required. This paper presents the design and analysis of an array of higher order microphones that uses 2D wavefield translation to provide a mode matching solution to the height invariant recording problem. It is shown that the use of Mth order microphones significantly reduces the number of microphone units by a factor of 1/(2M + 1) at the expense of increased complexity at each microphone unit. Robustness of the proposed array is also analyzed based on the condition number of the translation matrix while discussing array configurations that result in low condition numbers. The white-noise gain (WNG) of the array is then derived to verify that improved WNG can be achieved when the translation matrix is well conditioned. Furthermore, the array's performance is studied for interior soundfield recording as well as exterior soundfield recording using appropriate simulation examples.

Three dimensional surround sound reproduction over large areas is a prevailing challenge due to the enormous numbers of loudspeakers required. In this paper, we propose an array of higher order loudspeakers which provide a mode matching solution to the problem based on 3D wavefield translation. It is shown that for a given bandwidth, the use of Lthorder sources significantly brings down the minimum loudspeaker requirement by a factor of 1=(L + 1)2. Furthermore, the array is shown to be capable of exterior field cancellation, increasing its performance in echoing environments. Design examples are given for interior field, exterior field and interior and exterior combined field reproduction.

The implementation of 3D sound reproduction is well founded theoretically, but the requirements for the number of loudspeakers and array geometry make it impractical using conventional technology. In practice, a 2D array of loudspeakers is commonly used which restricts the reproduction of sources to those in the horizontal plane and requires a restricted form for the free-field loudspeaker driving signals based on the sectorial spherical harmonics. However, reflections can impair reproduction quality when using the free-field solution. Since first-order loudspeakers are commercially available which increase the direct to reflected sound ratio, we extend the sectorial solution for the loudspeaker excitation signals to the first-order case. We also investigate the reproduction accuracy for both zeroth and first-order loudspeakers using numerical simulations.

Sound reproduction systems aim to produce a desired sound field over a region of space. At high frequencies, the number of loudspeakers required is prohibitive. This paper shows that the use of Nth order loudspeakers, in which each loudspeaker produces polar responses up to cos(Nφ) and sin (Nφ) , produces accurate reproduction over N times the area of a first order array and can largely eliminate any exterior field. This allows a significant reduction in the number of loudspeaker units, at the expense of increased complexity in each loudspeaker unit.

The reproduction of a sound field over a spatial region can require large output powers from the loudspeakers. In such a case, the sound field reproduction can be non-robust or cause loudspeaker failure. One method of limiting loudspeaker power is to regularize the solution. We instead assume each loudspeaker is capable of delivering a certain peak power, and find a solution minimizing the error between the created and desired field subject to this power constraint. We apply a power constraint at a single frequency and also constrain total output power summed across many frequencies. Rapidly converging interior point methods are derived and shown to possess similar robustness to the regularized solution.

We devise a method for sound field reproduction (SFR) around a solid object in a reverberant room. Until now, work have focussed on reproducing sound in an empty listening space and, for the most part, in non-reverberant environments. However, in a reverberant, room as soon as a listener steps into the space he alters his acoustic environment, generating a sound component which is body-scattered and successively reverberated throughout the room. Building on the model of the sound field around a solid sphere in free space, we extend to reproduction around a human head in a reverberant room. In doing so, we show the relationship between the pressure matching and mode matching approaches of SFR.

Previous work has indicated that a limitation on the low frequency performance of a circular microphone array for holographic sound field recording is phase mismatch between the microphones in the array. At low frequencies these variations become more significant than at mid-range and high frequencies because the high order phase mode responses are lower in amplitude. This paper investigates the possibility of performing a "self" calibration of a microphone array. The basis of the calibration is to estimate the location of one or more sources using mid-range frequencies and to use this source location information to perform correction to the array at low frequencies. This of course implies that the calibration is performed in a relatively anechoic environment, since multipath effects at widely differing frequencies are uncorrelated. Initial results confirm that a significant improvement in the array response is possible using this approach.

Assisted reverberation systems are used to electronically enhance the reverberation time of an auditorium. A limitation of such systems is that they produce enhanced room transfer functions that contain unnaturally large peaks. These peaks produce colouration of the sound decay. A stochastic simulation technique is presented that quantifies colouration by measuring the probability density function of the transfer functions and comparing it with the ideal Rayleigh distribution of an unassisted room. The technique is applied to existing reverberation systems, and to a new system which provides large increases in reverberation time without increasing the loop gain. It is shown that, for the same power gain, the new system produces a higher colouration, but that this problem may be eliminated by the use of an allpass reverberator.>