Mark A. Poletti

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.

An approach to generating two zones of silence with application to personal sound systems.

An application of current interest in sound reproduction systems is the creation of multizone sound fields which produce multiple independent sound fields for multiple listeners. The challenge in producing such sound fields is the avoidance of interference between sound zones, which is dependent on the geometry of the zone and the direction of arrival of the desired sound fields. This paper provides a theoretical basis for the generation of two zones based on the creation of sound fields with nulls and the positioning of those nulls at arbitrary positions. The nulls are created by suppressing low-order mode terms in the sound field expansion. Simulations are presented for the two-dimensional case which shows that suppression of interference is possible across a broad frequency audio range.

Wavefield Analysis Over Large Areas Using Distributed Higher Order Microphones

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 arrays performance is studied for interior soundfield recording as well as exterior soundfield recording using appropriate simulation examples.

Comparison of methods for calculating the sound field due to a rotating monopole

A spherical harmonic expansion for the sound field due to a rotating oscillating point source has recently been derived. This paper provides further confirmation of the expansion results by comparing it with two known numerical approaches to determining the sound field. In the advanced time approach-applicable for Mach numbers below 1-the sound at transmission time determines the field at an observation point from the distance from source to observation point at the transmission time. In the retarded time approach the field at the observation point at the observation time is determined by solving for the retarded transmission times. The results from all three approaches are shown to be in good agreement. Expressions for the far-field instantaneous frequency are also derived and shown to agree with previous work.

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.

Spatially robust far-field beamforming using the von Mises(-Fisher) distribution

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.

3D soundfield reproduction using higher order loudspeakers

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 Lth order 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.

A GPU-accelerated real-time implementation of TRINICON-BSS for multiple separation units

In this demonstration, a GPU-accelerated implementation of the second-order statistics version of the TRINICON (Triple-N independent component analysis for convolutive mixtures) blind source separation (BSS) algorithm is presented. The single-precision arithmetic separation performance is compared with a double-precision CPU (Central Processing Unit) reference implementation to establish the effect of parallel algorithms and the reduction of floating-point precision. Computational performance results are presented for a single 2-channel separation unit and multiple simultaneous 2-channel separation units to study the potential future use in a sensor array network with multiple 2-channel separation units.

Trinicon-BSS system incorporating robust dual beamformers for noise reduction

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.

Simultaneous channel estimation and joint time-frequency domain crosstalk cancellation in multichannel personal audio systems

In this paper, we present two important contributions. The first is demonstration of the use of subliminal levels of pseudo-random noise to enhance channel estimation, and the second is a joint time and frequency domain algorithm for multichannel inversion. An adaptive system is presented where the acoustic channel is accurately estimated and utilized. In this implementation, maximum length sequences in the form of pseudo-random noise are superimposed on the source signals to aid the estimation quality of the acoustic channels. Upon estimation, crosstalk cancellation filters are designed using a time and frequency domain technique which uses a window to achieve more efficient and effective cancellation of crosstalk. For a 3×2 crosstalk system, the presented results show the improvement in channel estimation quality when low levels of maximum length sequences are superimposed on the source signals. On average -25dB of crosstalk cancellation is achieved.