This study introduces a Quality of Experience (QoE) model of loudspeaker-based spatial speech reproduction, which specifies quality elements and quality features relevant to overall listening experience and Quality of Service (QoS), respectively. Assumptions about the relations between selected quality elements and quality features were validated in a listening-only test. Participants had the task to behaviorally identify the voices of two talkers and provided subjective judgments on category rating scales. Sentences uttered by the two talkers were alternately presented through only a central loudspeaker (non-spatial mode) versus through either the central or one talker-specific lateral loudspeaker (spatial mode). The transmitted speech signals were clean, superimposed with background noise or bandpass-filtered. Results showed that transmission quality, but not reproduction mode significantly affected evaluative (speech quality, speech intelligibility), immersion (voice naturalness, spatial presence, social presence) and task-related (voice similarity, talker identification effort) aspects of overall listening experience. Contrary to previous findings, the spatial mode did not reduce the effort of talker identification. This suggested that noticeable advantages from spatial reproduction may not be evident in relatively simple listening situations, but could manifest if complexity is increased. Besides their task-related influence, noise and spectral distortion should be mitigated when targeting a higher level of immersion.

In this paper a new methodology for the evaluation of perceived quality for long duration audiovisual content is presented. This method allows the investigation of unexplored dependencies between perceived quality and impairment variations over extended periods of time. Instead of providing quality scores on predefined rating scales, assessors are asked to adjust the quality level of the displayed content themselves, whenever a degradation in perceived quality occurs. This method approaches the problem of quality evaluation from a different perspective than the existing methods, which might be valuable for future research and application. Preliminary results of an experiment conducted using this methodology are presented.

In this paper, we propose a design method for 3-D higher order ambisonics (3-D HOA) encoding matrices which offers the possibility to impose spatial stop-bands in the directivity patterns of all the spherical-harmonic audio channels while keeping the transformed audio channels still compatible with the 3-D HOA reproduction sound format. This might be useful as an encoding technique which suppresses interfering signals from specific directions in a 3-D HOA recording, or in other situations where certain spatial areas should be suppressed. The design method is adapted from recent work on the optimization of spherical microphone array beamforming. Using the proposed optimization method and the spherical harmonics mathematics framework, the relationship between several design factors, e.g., distortions in the desired response, the dynamic range of matrix coefficients, can be analyzed and illustrated as function of frequency. Based on the proposed optimization formulation, additional constraints can also be easily included and solved. In some of the formulations, the processing can be applied as a matrix multiplication to recorded spherical harmonics coefficients, that is, already encoded 3-D HOA format signals. The modified signals can be of the same or a lower spherical harmonics order. For a full optimization that gives a globally optimal solution, on the other hand, the processing must be applied to the microphone signals themselves. Numerical and experimental results validate the proposed method.

A robust minimum sidelobe beamforming approach based on the spherical harmonics framework for spherical microphone arrays is proposed. It minimizes the peaks of sidelobes while keeping the distortionless response in the look direction and maintaining the mainlobe width. A white noise gain constraint is also derived and employed to improve the robustness against array errors. The resulting beamformer can provide optimal tradeoff between the sidelobe level, the beamwidth and robustness, so it could be more practical than the existing spherical array Dolph-Chebyshev modal beamformer in the presence of array errors. The optimal modal beamforming problem is formulated as a tractable convex second-order cone programming program, which is more efficient than conventional element-space based approaches, since the dimension of array weight vectors can be significantly decreased by using the properties of spherical harmonics and Legendre polynomials. For the purpose of performance comparison, we also formulate current robust modal beamformers as equivalent optimization problems based on the proposed array model. Numerical results show the high flexibility and efficiency of the proposed beamforming approach.

Most of the existing spherical array modal beamformers are implemented in the frequency domain, where a block of snapshots is required to perform the discrete Fourier transform. In this paper, an approach to real-valued time-domain implementation of the modal beamformer for broadband spherical microphone arrays is presented. The microphone array data are converted to the spherical harmonics domain by using the discrete spherical Fourier transform, and then steered to the look direction followed by the pattern generation unit implemented using the filter-and-sum structure. We derive the expression for the array response, the beamformer output power against both isotropic noise and spatially white noise, and the mainlobe spatial response variation in terms of the finite impulse response (FIR) filters' tap weights. A multiple-constraint problem is then formulated to find the filters' tap weights with the aim of providing a suitable tradeoff among multiple conflicting performance measures such as directivity index, robustness, sidelobe level, and mainlobe response variation. Simulation and experimental results show good performance of the proposed time-domain broadband modal beamforming approach.

An approach to optimal array pattern synthesis based on spherical harmonics is presented. The array processing problem in the spherical harmonics domain is expressed with a matrix formulation. The beamformer weight vector design problem is written as a multiply constrained problem, so that the resulting beamformer can provide a suitable trade-off among multiple conflicting performance measures such as directivity index, robustness, array gain, sidelobe level, mainlobe width, and so on. The multiply constrained problem is formulated as a convex form of second-order cone programming which is computationally tractable. We show that the pure phase-mode spherical microphone array can be viewed as a minimum variance distortionless response (MVDR) beamformer in the spherical harmonics domain for the case of spherically isotropic noise. It is shown that our approach includes the delay-and-sum beamformer and a pure phase-mode beamformer as special cases, which leads to very flexible designs. Results of simulations and experimental data processing show good performance of the proposed array pattern synthesis approach. To simplify the analysis, the assumption of equidistant spatial sampling of the wavefield by microphones on a spherical surface is used and the aliasing effects due to noncontinuous spatial sampling are neglected.

A space domain optimal beamforming approach for spherical microphone arrays is presented, which is based on original microphone signals rather than spherical harmonics. A simple correlated multipath signal model is employed. The beamforming optimization criteria, including adaptive beamforming, multi-beam steering, sidelobe control, correlated interference cancellation and robustness control, are proposed and reformulated in a convex form of second order cone programming, which is computationally tractable. Using the spherical harmonics framework, we also prove that, given a perfect spatial sampling scheme, the conventional plan-wave decomposition (PWD) beamformer can be interpreted as a space domain minimum variance distortionless response (MVDR) beamformer under the condition of a spherically isotropic sound field.

When Higher Order Ambisonics (HOA) is used to represent a sound field, the channels might contain a lot of redundancy in some cases. This redundancy can be exploited in order to provide more efficient network transmission and storage. In this work the amount of inter-channel redundancy for Higher Order Ambisonics is investigated. Furthermore, lossless compression techniques that build on this redundancy are studied, with a focus on low-delay algorithms for real-time, or two-way, applications. The presented encoding scheme results in a delay of 256 samples, but with a rather high computational complexity both for encoding and decoding. The system also preserves the desired features of the HOA format, such as the scalability and the ability to reproduce over arbitrary loudspeaker layouts.