Sound rendering is increasingly being required to extend over only 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 multi-zone sound control in a room. Sound control over multiple zones is formulated as an optimisation 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 wave-domain sound field representations 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 greater control of sound fields over wide areas and reduced total number of loudspeaker units, thus making it particularly suitable for establishing personal sound zones.

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 th order microphones significantly reduces the number of microphone units by a factor of 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.

Abstract-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: (i) The parameterization is given in terms of a modal expansion of 3D basis functions. (ii) 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. (iii) 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 predefined 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. I. INTRODUCTIONThe room transfer function (RTF), demonstrates the collective effect of multipath propagation of sound between a source and a receiver within a given room enclosure. Accurate modeling of the RTF is useful in soundfield simulators as well as many other applications such as sound reproduction, soundfield equalization, echo cancellation, and speech dereverberation. These applications use appropriate RTF deconvolution methods to cancel the effects of room reflections (reverberation), and therefore, are highly dependent on the accuracy of the RTF model.The theoretical solution to the RTF based on the Green's function [1] was derived assuming a strict rectangular room geometry. It can only be applied to highly idealised cases with reasonable effort. The rooms with which we are concerned in our daily life however are more or less irregular in shape and the formulation of irregular boundary conditions will require extensive numerical calculations. For this reason, the immediate application of the classical model to practical problems in room acoustics is limited.In practice, RTFs are usually estimated as FIR filters, or as parametric equations based on the geometrical properties of the room. In the FIR filter approach, the RTF is assumed to behave as a linear time-invariant system, and then modeled as either an all-zero, all-pole, pole-zero [2] or a common pole-zero [3] system. The coefficients of these models are estimated as variable parameters of the RTF, and since the

Time-varying components are used in some multichannel sound systems designed for the enhancement of room acoustics. Time-variation can usefully reduce the risk of producing ringing tones and improve stability margins, provided that any modulation artefacts are inaudible. Frequency-shifting is one form of time-variation which provides the best case improvement in loop gain, and for which the single channel stability limit has been derived. This paper determines the stability limit for multiple channel systems with frequency-shifting by generalizing the previous single-channel analysis. It is shown that the improvement in stability due to frequency-shifting reduces with the number of channels. Simulations are presented to verify the theory. The stability limits are also compared with those of time-invariant systems, and preliminary subjective assessments are carried out to indicate useable loop gains with frequency-shifting.

Sound reproduction systems using open arrays of loudspeakers in rooms suffer from degradations due to room reflections. These reflections can be reduced using pre-compensation of the loudspeaker signals, but this requires calibration of the array in the room, and is processor-intensive. This paper examines 3D sound reproduction systems using spherical arrays of fixed-directivity loudspeakers which reduce the sound field radiated outside the array. A generalized form of the simple source formulation and a mode-matching solution are derived for the required loudspeaker weights. The exterior field is derived and expressions for the exterior power and direct to reverberant ratio are derived. The theoretical results and simulations confirm that minimum interference occurs for loudspeakers which have hyper-cardioid polar responses.