Darren B. Ward

Particle filtering algorithms for tracking an acoustic source in a reverberant environment

Traditional acoustic source localization algorithms attempt to find the current location of the acoustic source using data collected at an array of sensors at the current time only. In the presence of strong multipath, these traditional algorithms often erroneously locate a multipath reflection rather than the true source location. A recently proposed approach that appears promising in overcoming this drawback of traditional algorithms, is a state-space approach using particle filtering. In this paper we formulate a general framework for tracking an acoustic source using particle filters. We discuss four specific algorithms that fit within this framework, and demonstrate their performance using both simulated reverberant data and data recorded in a moderately reverberant office room (with a measured reverberation time of 0.39 s). The results indicate that the proposed family of algorithms are able to accurately track a moving source in a moderately reverberant room.

Reproduction of a plane-wave sound field using an array of loudspeakers

Reproduction of a sound field is a fundamental problem in acoustic signal processing. In this paper, we use a spherical harmonics analysis to derive performance bounds on how well an array of loudspeakers can recreate a three-dimensional (3-D) plane-wave sound field within a spherical region of space. Specifically, we develop a relationship between the number of loudspeakers, the size of the reproduction sphere, the frequency range, and the desired accuracy. We also provide analogous results for the special case of reproduction of a two-dimensional (2-D) sound field. Results are verified through computer simulations.

Theory and design of broadband sensor arrays with frequency invariant far‐field beam patterns

The theory and design of a broadband array of sensors with a frequency invariant far‐field beam pattern over an arbitrarily wide design bandwidth is presented. The frequency invariant beam pattern property is defined in terms of a continuously distributed sensor, and the problem of designing a practical sensor array is then treated as an approximation to this continuous sensor using a discrete set of filtered broadband omnidirectional array elements. The design methodology is suitable for one‐, two‐, and three‐dimensional sensor arrays; it imposes no restrictions on the desired aperture distribution (beam shape), and can cope with arbitrarily wide bandwidths. An important consequence of the results is that the frequency response of the filter applied to the output of each sensor can be factored into two components: One component is related to a slice of the desired aperture distribution, and the other is sensor independent. The results also indicate that the locations of the sensors are not a crucial desi...

Theory and design of high order sound field microphones using spherical microphone array

A major problem in sound field reconstruction systems is how to record the higher order (> 1) harmonic components of a given sound field. Spherical harmonics analysis is used to establish theory and design of a higher order recording system, which comprises an array of small microphones arranged in a spherical configuration and associated signal processing. This result has implications to the advancement of future sound field reconstruction systems. An example of a third order system for operation over a 10∶1 frequency range of 340 Hz to 3.4 kHz is given.

FIR filter design for frequency invariant beamformers

Two methods of implementing FIR filters for a frequency invariant beamformer are presented. Each of these methods uses a single underlying set of filter coefficients obtained directly from the desired beamformer response. One method uses multirate processing, and the other is based on a single sampling rate.

Broadband DOA estimation using frequency invariant beamforming

A new method of direction-of-arrival (DOA) estimation for multiple broadband farfield signals is presented. The technique uses a beamspace preprocessing structure based on frequency invariant beamforming. Specifically, a set of beam-shaping filters focus the received array data in the time domain, thereby avoiding the need for frequency decomposition. Hence, the proposed method is conceptually different from most other broadband DOA estimators, which require frequency decomposition. Numerical results are presented to demonstrate the use of the new method and compare it with conventional coherent signal subspace methods.

Broadband nearfield beamforming using a radial beampattern transformation

This paper presents a new method of designing a beamformer having a desired nearfield broadband beampattern. The methodology uses the spherical harmonic solution to the wave equation to transform the desired nearfield beampattern to an equivalent farfield beampattern. A farfield beamformer is then designed for a transformed farfield beampattern that, if achieved, gives the desired nearfield pattern exactly. Salient features of the new method are as follows. (i) The nearfield patterns can be achieved for all angles, not just the primary look direction. (ii) There is no theoretical restriction on the bandwidth. (iii) General array geometries may be used. As an illustration, we apply the method to the problem of producing a practical array design that achieves a nearfield beampattern that is frequency invariant over an octave bandwidth, where at the lowest frequency, the array-source separation is three wavelengths.

Particle filter beamforming for acoustic source localization in a reverberant environment

Traditional acoustic source localization uses a two-step procedure requiring intermediate time-delay estimates from pairs of microphones. An alternative single-step approach is proposed in this paper in which particle filtering is used to estimate the source location through steered beamforming. This scheme is especially attractive in speech enhancement applications, where the localization estimates are typically used to steer a beamformer at a later stage. Simulation results show that the algorithm is robust to reverberation, and is able to accurately follow the source trajectory.

Statistical analysis of the autoregressive modeling of reverberant speech.

Hands-free speech input is required in many modern telecommunication applications that employ autoregressive (AR) techniques such as linear predictive coding. When the hands-free input is obtained in enclosed reverberant spaces such as typical office rooms, the speech signal is distorted by the room transfer function. This paper utilizes theoretical results from statistical room acoustics to analyze the AR modeling of speech under these reverberant conditions. Three cases are considered: (i) AR coefficients calculated from a single observation; (ii) AR coefficients calculated jointly from an M-channel observation (M > 1); and (iii) AR coefficients calculated from the output of a delay-and sum beamformer. The statistical analysis, with supporting simulations, shows that the spatial expectation of the AR coefficients for cases (i) and (ii) are approximately equal to those from the original speech, while for case (iii) there is a discrepancy due to spatial correlation between the microphones which can be significant. It is subsequently demonstrated that at each individual source-microphone position (without spatial expectation), the M-channel AR coefficients from case (ii) provide the best approximation to the clean speech coefficients when microphones are closely spaced (<0.3m).

Subspace approach to blind and semi-blind channel estimation for space-time block codes

A new approach to blind and semi-blind channel estimation for space-time block codes (STBCs) is presented. By exploiting the recently proposed generalized space-time block code framework, an STBC system is reduced from multiple-input multiple-output to a single-input multiple-output system. A conventional blind algorithm based on the subspace algorithm is then used to identify the channel impulse responses and achieve equalization. Simulation results demonstrate the performance of the proposed algorithm and a comparable scheme.