Terence Betlehem

Personal Sound Zones: Delivering interface-free audio to multiple listeners

Sound rendering is increasingly being required to extend over certain regions of space for multiple listeners, known as personal sound zones, with minimum interference to listeners in other regions. In this article, we present a systematic overview of the major challenges that have to be dealt with for multizone sound control in a room. Sound control over multiple zones is formulated as an optimization problem, and a unified framework is presented to compare two state-of-the-art sound control techniques. While conventional techniques have been focusing on point-to-point audio processing, we introduce a wave-domain sound field representation and active room compensation for sound pressure control over a region of space. The design of directional loudspeakers is presented and the advantages of using arrays of directional sources are illustrated for sound reproduction, such as better control of sound fields over wide areas and reduced total number of loudspeaker units, thus making it particularly suitable for establishing personal sound zones.

A constrained optimization approach for multi-zone surround sound

A recent approach to surround sound is to perform exact control of the sound field over a region of space. Here, the driving signals for an array of loudspeakers are chosen to create a desired sound field over an extended area. An interesting subtopic is multi-zone surround sound, where two or more listeners can experience totally independent sound fields. However, multi-zone surround sound is a challenge because implementation can be very non-robust. We formulate multi-zone sound reproduction as a convex optimization problem, where the sound energy leakage into other listener zones is limited to fixed levels, and a constraint is placed on the loudspeaker weights to improve the robustness. An interior point algorithm is devised for computing the loudspeaker weights, and its performance is compared with least squares approaches of multi-zone reproduction in typical two-zone cases.

An efficient parameterization of the room transfer function

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.

Two dimensional sound field reproduction using higher order sources to exploit room reflections

In this paper, sound field reproduction is performed in a reverberant room using higher order sources (HOSs) and a calibrating microphone array. Previously a sound field was reproduced with fixed directivity sources and the reverberation compensated for using digital filters. However by virtue of their directive properties, HOSs may be driven to not only avoid the creation of excess reverberation but also to use room reflection to contribute constructively to the desired sound field. The manner by which the loudspeakers steer the sound around the room is determined by measuring the acoustic transfer functions. The requirements on the number and order N of HOSs for accurate reproduction in a reverberant room are derived, showing a 2N + 1-fold decrease in the number of loudspeakers in comparison to using monopole sources. HOSs are shown applicable to rooms with a rich variety of wall reflections while in an anechoic room their advantages may be lost. Performance is investigated in a room using extensions of both the diffuse field model and a more rigorous image-source simulation method, which account for the properties of the HOSs. The robustness of the proposed method is validated by introducing measurement errors.

Efficient crosstalk canceler design with impulse response shortening filters

An impulse response shortening approach is used to perform acoustic crosstalk cancellation. Crosstalk canceler filters are traditionally designed using least squares, with an approach that equalizes all room reverberation. However, depending upon end application, some reverberation may be permissible in the delivered signals. This idea is used to create more efficient crosstalk cancellation filters. The filter design is formulated as a minimax problem solvable with linear programming methods. Penalty functions on crosstalk levels and detrimental reverberation are introduced, which allow control of the reverberant tails and crosstalk levels. Shorter crosstalk cancellation filters are designed, by leaving in early echoes and/or allowing a slower decay of the late reverberant tail.

Analysis and control of multi-zone sound field reproduction using modal-domain approach

Multi-zone sound control aims to reproduce multiple sound fields independently and simultaneously over different spatial regions within the same space. This paper investigates the multi-zone sound control problem formulated in the modal domain using the Lagrange cost function and provides a modal-domain analysis of the problem. The Lagrange cost function is formulated to represent a quadratic objective of reproducing a desired sound field within the bright zone and with constraints on sound energy in the dark zone and global region. A fundamental problem in multi-zone reproduction is interzone sound interference, where based on the geometry of the sound zones and the desired sound field within the bright zone the achievable reproduction performance is limited. The modal-domain Lagrangian solution demonstrates the intrinsic ill-posedness of the problem, based on which a parameter, the coefficient of realisability, is developed to evaluate the reproduction limitation. The proposed reproduction method is based on controlling the interference between sound zones and sound leakage outside the sound zones, resulting in a suitable compromise between good bright zone performance and satisfactory dark zone performance. The performance of the proposed design is demonstrated through numerical simulations of two-zone reproduction in free-field and in reverberant environments.

A robust sparse approach to acoustic impulse response shaping

Impulse response shaping is a technique for partly equalizing impulse responses. In acoustics, it can be used for the reproduction of audio signals mitigated by distortions in a room. The most significant phenomenon among the distortions is reverberation, a straightforward characterization of which is the room impulse response. Room responses can be characterized but could contain measurement errors or noise. In addition, room responses vary with changes in atmospheric conditions such as temperature and humidity and also due to change in positions inside a room. The design of a shaping filter robust to at least some of these variations is likely to be very useful, which is considered in this work. The method uses a computationally efficient approach based on Basis Pursuit DeNoising (BPDN).

On room impulse response between arbitrary points: An efficient parameterization

Spatial sound field recording and reproduction in reverberant rooms require the measurement of room transfer functions (RTF) followed by room equalization to correct undesired reflections. Typically, the RTF rapidly varies over the room and hence requires an infinite number of point to point measurements to characterize the room. This paper provides a modal based parameterization of the two dimensional acoustic transfer function over a sizeable spatial region in which the source(s) lie and a sizeable spatial region in which the receiver(s) are located. As a result, a finite set of modal coefficients is shown to be capable of predicting the RTF between any two arbitrary points from the aforementioned source and receiver regions.

A sparsity based approach for acoustic room impulse response shortening.

Acoustic room impulse response shortening is a method for reducing the effect of unwanted reverberations in a surround sound system such as reflections from the walls, floor and corners and cross-talk from the loudspeakers. Impulse response shortening is a softer requirement than inversion, that reduces the effect of room impulse responses rather than removing them completely as in inversion. Thus, shortening allows for some reverberation to be present in the shortened responses. Such an approach results in shorter filters and less cross-talk. This paper proposes a sparse formulation approach for impulse response shortening. We assume that the resulting room impulse responses can be estimated by microphones located at or near the listening positions. Through simulations, it is demonstrated that cross-talk as low as -40dB can be achieved, with fast convergence.

Under-determined source separation based on power spectral density estimated using cylindrical mode beamforming

Sound source signals can be separated using Wiener post-filters calculated by estimating the power spectral densities (PSDs) of sources from the outputs of a set of beamformers. This approach has been shown effective in the under-determined case where the number of sources to be separated exceeds the number of microphones. In this paper, a limit on the maximum number of separable sources is derived beyond which the problem becomes rank deficient. This study reveals the number of sources that can be separated simultaneously is related to the order of the beam patterns. Further, using the principles of cylindrical mode beamforming, the performance can be predicted as a function of frequency. The result is consistent with simulations in which the performance of separating music and speech sound sources was quantified.