The problem of channel estimation and symbol detection in fast fading channels using multiple antennas is dealt with. This is one of the many challenges faced by extending spatial multiplexing to mobile terminals. A commonly used method of resolving this problem is pilot‐aided channel interpolation. An iterative algorithm that substantially improves the performance of pilot-aided MIMO channel interpolation systems is proposed. The algorithm involves alternating projection of parameters onto different sub-spaces so as to exploit the time selectivity of the channel to improve the quality of symbol detection. Simulations have shown that the algorithm, when applied to any interpolation scheme, improves the bit error performance of the system significantly.

In this paper, we propose a probabilistic technique to achieve joint channel estimation and symbol detection for orthogonal space time block coding (OSTBC) systems. The key result is that a multiple-input single-output system using OSTBC can be translated to an equivalent single-input multiple-output system using the implicit redundancy in OSTBC. This allows some probabilistic methods to be applicable at each output branch. Since each output branch is excited by the same input, redundant information can be exchanged between the multiple branches to improve the detection performance at individual branches. The proposed technique has the potential to be concatenated with a channel decoder and form a turbo receiver. The performance of the proposed algorithm is demonstrated through simulation results and compared with an existing scheme.

We study the effect of reverberation and source movement on the performance of blind source separation and deconvolution (BSSD) algorithms. Using the model of statistical room acoustics we derive theoretical performance measures for a class of unmixing algorithms when these are used in a reverberant room. We specifically investigate the cases: 1) where separation of only direct paths is performed and 2) the case where unmixing of the full reverberant paths is attempted. We develop closed-form performance measures that are dependent on the geometry used and the chosen unmixing system. Using these measures allows us to draw general conclusions on the robustness to source movement of typical BSSD algorithms. Results indicate that performance of systems that show very good separation in static reverberant environments is significantly reduced when sources move, with performance degrading to that of simple direct-path separation

This paper is concerned with the problem of channel estimation and symbol detection in fast fading channels, which is one of the challenges faced by extending spatial multiplexing to mobile terminals. A method of resolving this problem is the use of pilot aided channel interpolation techniques which are common in single channel systems. We propose an iterative algorithm that substantially improves the performance of pilot aided MIMO channel interpolation systems by exploiting the time selectivity of the channel.

A semi-blind method for minimum mean square square error (MMSE) equalization of space-time block coding (STBC) systems is proposed. By exploiting information from the redundancy in the structure of STBC as well as from the training symbols, MMSE equalization can be achieved without channel estimation. Simulation results demonstrate the performance of the proposed algorithm compared to existing methods.

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.

Using statistical room acoustics, we investigate the performance of blind source separation and deconvolution (BSSD) algorithms when used in a reverberant room. We focus on the case where one of the sources moves, and examine the relative impact of source movement and room reverberation on the expected performance. We derive theoretical expressions, and verify these through image model simulations.

We propose a new method for range and bearing estimation of wideband sources based on a novel beamspace structure. The proposed structure consists of an orthogonal set of beamspace processors that have a frequency-invariant beampattern property, and can be steered easily to any nearfield range while maintaining the frequency-invariance of the spatial response. These properties mean that the proposed structure can perform coherent signal subspace processing for nearfield wideband sources, making it ideal for wideband range and bearing estimation. Performance of the estimator is demonstrated through Monte Carlo simulations.

The key difference between space time block codes (STBCs) and a general multiple-input multiple-output system is that the transmitted signals of STBC systems are not mutually independent. In this paper we develop an algorithm that exploits the implicit structure of STBCs to achieve direct equalization without knowledge of the channel impulse responses. We analyze the identifiability of the equalizers, and develop explicit conditions for the existence of semi-blind equalizers. The performance of the algorithm is demonstrated through simulation results and compared with an existing scheme.

We propose novel blind separation techniques for multiple-input multiple-output systems based on a least-squares constant modulus algorithm (LSCMA). To ensure that each output signal is extracted from different input signals, the proposed algorithms have been derived by using successive interference cancellation and the Gram-Schmidt orthogonalization procedure. The performance is observed through computer simulations and compared with the multitarget LSCMA (MT-LSCMA) and LS multiuser CMA (LS-MU-CMA). Simulation results have shown that the proposed algorithms exhibit better performance in terms of both bit error rate (BER) and convergence speed.

Traditional acoustic source localization techniques attempt to determine the current location of an acoustic source from data obtained at an array of sensors during the current time only. Recently, state-space methods have been proposed that use particle filters to perform recursive estimation of the current source location using all previous data. In this paper we present an overview of these particle filter algorithms, and formulate performance measures for determining their ability to track a moving source. We present results of experiments using reverberant data recorded in a real room, and show that steered beamforming methods have improved performance over GCC-based approaches.

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.

In multiple antenna communications, the impulse responses of the channels are frequently assumed to be fixed. However, if multiple antennas are used in mobile terminals, this assumption is weakened. In this paper, we propose using a sequential Monte Carlo method, which is robust even in non-Gaussian environments, in conjunction with current detection algorithms for tracking the channel matrix to improve the performance of multiple antenna systems under time varying channels.

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.

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.

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.

It is well known that the effectiveness of loudspeaker-based 3D audio systems is critically dependent on the listener being in a known position, the so-called "sweet spot". In this paper we propose a new system that provides increased robustness to perturbations such as head movement, reverberation, and different head shapes. The system is based on using a particular geometry that offers a combination of symmetric and asymmetric loudspeaker positions. This combination increases the bandwidth over which the proposed system is robust, when compared with existing 3D audio systems.

This paper introduces a microphone array processing method that possesses the robustness of fixed beamforming along with the ability to be dynamically reconfigured to limit interference and reverberation. The basic approach is to partition the environment into two regions: an interior region (containing sources that are physically present within the room enclosure), and an exterior region (containing virtual sources of reverberation). The interior region is further divided into cells, and standard source localization techniques are used to identify those cells containing the desired source as well as sources of interference (e.g., competing talkers). Beamforming weights are then found to pass the desired signal, while simultaneously minimizing a weighted combination of interior interference and exterior reverberation. Simulation results are presented to demonstrate the effectiveness of the proposed technique when compared with conventional beamforming methods.

The objective of acoustic crosstalk cancellation is to use loudspeakers to deliver prescribed binaural signals (that reproduce a particular auditory scene) to a listener's ears. Such systems are very sensitive to the position of the listener's head. In this paper we perform a joint least squares optimization of both the filter coefficients and the modeling delay in order to obtain a robust solution. Simulation results demonstrate the effectiveness of the proposed technique.

Acoustic crosstalk cancellation is a signal processing technique whereby two (or possibly more) loudspeakers are used to deliver desired signals exactly at the listener's ears. Such a system is useful for three-dimensional (3-D) audio applications, and removes the requirement for the listener to wear headphones. However, crosstalk cancelers are notoriously nonrobust to slight movements in head position. Recently, a system consisting of two closely spaced loudspeakers (referred to as the "stereo-dipole") has been proposed to improve the robustness of the crosstalk canceler. In this letter we present a robustness analysis of the two-channel crosstalk canceler, and derive optimum loudspeaker positions.

Crosstalk cancellation is a signal processing technique whereby two (or more) loudspeakers are used to deliver desired signals exactly at the listener's ears. Such a system is useful for 3D audio applications, and removes the requirement for the listener to wear headphones. However, crosstalk cancelers are notoriously non-robust to slight movements in head position, and there currently exists no clear method for determining the best loudspeaker placement in a given situation. We propose a robustness measure to evaluate the performance of crosstalk cancelers as a function of loudspeaker spacing. Based on this analysis we conclude that certain loudspeaker spacings give far better robustness performance, and provide a simple empirically-derived equation for determining the optimum loudspeaker spacing in a given situation.

A new technique for broadband minimum variance beam-forming is presented that overcomes the signal cancellation problem of conventional adaptive beamformers in the presence of correlated interference. Specifically, frequency-invariant beam-forming is used to perform frequency domain averaging, thereby reducing the correlation between the desired signal and the interference. Such a technique is useful for hands-free speech acquisition using a microphone array, since correlated interference will be present due to room reflections.

We establish the asymptotic equivalence, up to complex conjugation, of two problems: (i) determining the nearfield performance of a farfield beampattern specification, and (ii) determining the equivalent farfield beampattern corresponding to a nearfield beampattern specification. Using this reciprocity relationship we develop a computationally simple procedure to design a beamforming array to achieve a desired nearfield beampattern response. The superiority of this approach to existing methods, both in ease of design implementation and performance obtained, is illustrated by a design example.

Frequency invariant beamforming is array processing in which the spatial response remains constant (with respect to frequency) within a wide frequency band of interest. We present a new algorithm for adaptive broadband beamforming which solves a minimum variance beamforming problem, with a structural frequency invariant beampattern constraint. This constraint allows us to reduce the dimension of the adaptation problem. The proposed algorithm is a block adaptive LMS algorithm which uses only a fraction of the parameters of a conventional fully adaptive array. Hence, the computational complexity is reduced and the convergence speed is increased. A simulation example is presented to demonstrate the new algorithm.

This paper presents a method for nearfield frequency invariant broadband array beamforming. This means that we can design an array with associated signal processing to achieve a desired frequency invariant beam pattern (as a function of direction) at any nominal finite distance over an arbitrarily wide range of frequencies. The challenge here is to compensate accurately for the nearfield distortions (both magnitude and phase) which are more acute at the lower frequencies of the operating bandwidth than the higher frequencies. The methodology uses spherical harmonics to transform the nearfield broadband frequency invariant beampattern specification to an equivalent farfield frequency varying beampattern specification. A simulation example is presented to demonstrate the effectiveness of this method in producing a nearfield beam pattern which is frequency invariant over an octave bandwidth.

This paper presents a new method of beam-space direction of arrival (DOA) estimation for multiple far-field broadband signals. A novel multi-rate beamforming structure having a frequency invariant property is applied to the array outputs. Using this multi-rate structure, only a single vector of parameters is required to implement a desired frequency invariant beam pattern regardless of the bandwidth or number of sensors in the array. This frequency-invariant beamformer performs focusing over a very wide bandwidth without resorting to the frequency bin design approach of other methods in which the solution of a multiparameter optimisation problem is required to design the beam-space focusing matrices.

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.