A novel Direction-of-Arrival (DOA) estimation method based on a plane wave sound field model was recently proposed for circular microphone arrays [1]. This article presents a detailed theoretical analysis of the method that relies on a Finite Rate of Innovation (FRI) assumption, investigating the impact of different theoretical acoustic source models. The method's estimation accuracy, and robustness against noisy measurement data and deviations from the model are investigated. The estimation performance is validated and assessed on the basis of results obtained from both simulations and experimental data.

In this paper, we propose a system for private sound which is based on the weighted pressure matching method (WPMM). The aim is to design the input signals to an array of loudspeakers that allow for the synthesis of a target field defined with large amplitude variations between the so-called dark points and the listener's position. The system enables listeners to control the trade-off between directivity performance and the accuracy of reproduction of the target field at the listening position when the input energy to the array is limited. This is achieved by calculating the WPMM weight in the dark zone based on a constraint on the characteristics of the sound field in the listening zone. The system is validated for a number of predefined use-case scenarios. The results of the experiments in an anechoic environment with a circular array prototype show that listeners can control the performance trade-off in a wide frequency range. In the second part of the paper, algorithms are presented for the fast update of the input signals when the user selects a new value of the performance constraint.

A very fast deconvolution method, which is based on the fast Fourier transform (FFT), can be used to control the outputs from a multichannel plant comprising any number of control sources and error sensors. The result is a matrix of causal finite impulse response filters whose performance is optimized at a large number of discrete frequencies. The paper is particularly aimed at multichannel sound reproduction and more specifically reproducing the sound field from a set of loudspeakers.

A discussion is given of two techniques for designing inverse filters for use in multichannel sound reproduction systems. The first is the multiple-input/output inverse filtering theorem, which uses direct inversion of a matrix containing the coefficients of filters used to specify the electroacoustic transmission paths. The second is an adaptive technique based on the multiple error LMS algorithm. The theory presented reconciles the two approaches and furthermore, derives explicit conditions which must be fulfilled if an exact inverse is to exist. A formula is derived which gives the number of coefficients required in the inverse filters in terms of the number of coefficients used to represent the transmission paths. Some numerical examples are also presented which illustrate the dependence of the mean square error on both the choice of modeling delay and the number of coefficients in the inverse filters. Finally, the results of some simulations are given which demonstrate the acoustical possibilities associated with these filtering techniques.>

The active control of harmonic sound using multiple secondary sources and multiple error microphones is formulated in an adaptive digital filtering framework. A stochastic gradient algorithm is presented to minimize the sum of the squares of the error microphone outputs by adapting the coefficients of an array of FIR (finite-impulse response) filters driving the secondary sources. The properties and implementation of the algorithm are discussed and some results are presented which demonstrate that the noise at the engine firing frequency inside an automobile can be reduced by 10-15 dB using a practical implementation of such an active sound control system, which had a response time of 0.1 s in this application.>

The problem is addressed of evaluating the maximum net reduction in acoustic power output produced by introducing a single point monopole secondary source to control the sound field radiated by a single point monopole primary source in a free field. The secondary source is constrained to act causally with respect to the primary source. The problem reduces to solving a Wiener-Hopf integral equation for the optimal realisable impulse response relating the secondary and primary source outputs. A solution is given which has a well-defined physical significance and demonstrates the need for the secondary source to act as a predictor in order that destructive interference of the primary field is possible at field points closer to the primary source than the secondary source.