Michael Buerger

Design of robust polynomial beamformers for symmetric arrays

Polynomial broadband beamforming designs enable an easy, smooth, and dynamic steering of the main beam. A number of design methods based on constrained optimization have been proposed recently which allow for the control of the robustness of these designs. Of course, the addition of robustness constraints reduces the number of degrees of freedom of the design. In this paper, we present a method to enhance the spatial selectivity of the robust polynomial beamformer design by exploiting the structure of symmetric arrays while still satisfying the robustness constraints. The effectiveness of this method is shown in design examples for symmetric linear and circular arrays.

Multizone sound field synthesis based on the joint optimization of the sound pressure and particle velocity vector on closed contours

In this paper, we propose a novel approach to multizone sound reproduction, which is motivated by the Kirchhoff-Helmholtz integral equation and aims at the simultaneous optimization of the sound pressure and particle velocity vector on contours around multiple local listening areas. The benefit of this approach is that the control points do not need to be distributed within the local listening areas, as is typically the case for previous multipoint approaches, but on surrounding contours only. This is especially desirable when it comes to practical realizations using real microphones as control points. A closed-form solution is presented for determining the loudspeaker weights in the frequency domain, and the effectiveness of the proposed method, which we refer to as Joint Pressure and Velocity Matching (JPVM), is validated by simulation results.

HRTF-based two-dimensional robust least-squares frequency-invariant beamformer design for robot audition

In this work, we propose a two-dimensional Head-Related Transfer Function (HRTF)-based robust beamformer design for robot audition, which allows for explicit control of the beamformer response for the entire three-dimensional sound field surrounding a humanoid robot. We evaluate the proposed method by means of both signal-independent and signal-dependent measures in a robot audition scenario. Our results confirm the effectiveness of the proposed two-dimensional HRTF-based beamformer design, compared to our previously published one-dimensional HRTF-based beamformer design, which was carried out for a fixed elevation angle only.

Broadband multizone sound rendering by jointly optimizing the sound pressure and particle velocity

In this paper, a recently proposed approach to multizone sound field synthesis, referred to as joint pressure and velocity matching (JPVM), is investigated analytically using a spherical harmonics representation of the sound field. The approach is motivated by the Kirchhoff-Helmholtz integral equation and aims at controlling the sound field inside the local listening zones by evoking the sound pressure and particle velocity on surrounding contours. Based on the findings of the modal analysis, an improved version of JPVM is proposed, which provides both better performance and lower complexity. In particular, it is shown analytically that the optimization of the tangential component of the particle velocity vector, as is done in the original JPVM approach, is very susceptible to errors and thus not pursued anymore. Furthermore, the analysis provides fundamental insights as to how the spherical harmonics used to describe three-dimensional sound fields translate into two-dimensional basis functions as observed on the contours surrounding the zones. By means of simulations, it is verified that discarding the tangential component of the particle velocity vector ultimately leads to an improved performance. Finally, the impact of sensor noise on the reproduction performance is assessed.

Robust data-independent object-related transfer function-based beamformer design for remote listening applications

Virtual reality offers users to experience remote acoustic scenes which are captured in the original environment by a microphone array carried by, e.g., a third person or a robot. To extract and preserve the spatiotemporal nature of specific sounds of interest in the original acoustic environments, data-independent acoustic beamforming appears to be a suitable technique. In this work, we present recent work on robust data-independent beamformer design which is applicable to unconstrained sensor topologies and allows for an intuitive control of the beamformer’s robustness. Combined with the concept of polynomial beamforming, flexible beam steering in real time is possible. Consequently, the beamformer design is well suited for remote listening applications. If the beamformer design is applied to a microphone array, which is integrated into a scatterer, object-related transfer functions (ORTFs) need to be incorporated into the beamformer design. We briefly discuss selected approaches to obtaining the requir...

Retrieval of individualized head-related transfer functions for hearing aid applications

The capability of modern hearing aids to provide hearing-impaired humans with enhanced signals, which ultimately leads to an increased speech intelligibility, may benefit from fitting the device for each subject individually. This ideally also involves the exploitation of Head-Related Impulse Responses (HRIRs). However, HRIRs vary from person to person and thus require tedious measurements for each individual. In this work, we investigate two approaches which aim at speeding up the HRIR acquisition procedure. These are continuous measurements and interpolation, where Dynamic Time Warping (DTW) as well as linear interpolation of the magnitude and phase responses are considered. In contrast to related publications, the continuous HRIR measurements are not performed in anechoic environments here. The quality of the obtained HRIRs is assessed by means of the system mismatch and the proposed error of relative transfer functions. Both measures reveal that continuous HRIR measurements are on average much more capable than the investigated interpolation approaches, and they furthermore provide a more uniform performance for different source directions.

The spatial coherence of noise fields evoked by continuous source distributions

In this work, analytic expressions for the spatial coherence of noise fields are derived in the modal domain with the aim of providing a sparse representation. For this purpose, the sound field in a region of interest is expressed in terms of a given pressure distribution on a virtual surrounding cylindrical or spherical surface. According to the Huygens-Fresnel principle, the sound pressure on this surface is represented by a continuous distribution of elementary line or point sources, where orthogonal basis functions characterize the spatial properties. To describe spatially windowed pressure distributions with arbitrary angular extensions, orthogonal basis functions of limited angular support are proposed. As special cases, circular and spherical pressure distributions with uncorrelated source modes of equal power are investigated. It is shown that these distributions result, respectively, in cylindrically isotropic and spherically isotropic, i.e., diffuse noise fields. The analytic expressions derived in this work allow for a prediction of the spatial coherence between arbitrary positions within the region of interest, such that no microphones need to be placed at the actual points of interest. Simulation results are presented to validate the derived relations.

Higher-order listening room compensation with additive compensation signals

The performance of sound reproduction systems for spatial audio is impaired by time-variant, reverberant listening environments. To tackle this issue, the Loudspeaker-Enclosure-Microphone System (LEMS) between the loudspeakers and reference microphones in the listening environment can be identified adaptively to allow an LEMS-specific pre-processing of the loudspeaker signals. This contribution introduces a broadband implementation of a narrowband Listening Room Compensation (LRC) method with additive compensation signals, recently proposed by Talagala et al. [1], it extends the concept to higher-order compensation, and compares LRC to Listening Room Equalization (LRE) analytically. Evaluations in an image-source environment confirm the efficacy of higher-order LRC and its suitability as a complexity-reduced alternative to LRE.