Parametric spatial audio processing utilising compact microphone arrays

This dissertation focuses on the development of novel parametric spatial audio techniques using compact microphone arrays. Compact arrays are of special interest since they can be adapted to fit in portable devices, opening the possibility of exploiting the potential of immersive spatial audio algorithms in our daily lives. The techniques developed in this thesis consider the use of signal processing algorithms adapted for human listeners, thus exploiting the capabilities and limitations of human spatial hearing. The findings of this research are in the following three areas of spatial audio processing: directional filtering, spatial audio reproduction, and direction of arrival estimation.  In directional filtering, two novel algorithms have been developed based on the cross-pattern coherence (CroPaC). The method essentially exploits the directional response of two different types of beamformers by using their cross-spectrum to estimate a soft masker. The soft masker provides a probability-like parameter that indicates whether there is sound present in specific directions. It is then used as a post-filter to provide further suppression of directionally distributed noise at the output of a beamformer. The performance of these algorithms represent a significant improvement over previous state-of-the-art methods.  In parametric spatial audio reproduction, an algorithm is developed for multi-channel loudspeaker and headphone rendering. Current limitations in spatial audio reproduction are related to high inter-channel coherence between the channels, which is common in signal-independent systems, or time-frequency artefacts in parametric systems. The developed algorithm focuses on solving these limitations by utilising two sets of beamformers. The first set of beamformers, namely analysis beamformers, is used to estimate a set of perceptually-relevant sound-field parameters, such as the separate channel energies, inter-channel time differences and inter-channel coherences of the target-output-setup signals. The directionality of the analysis beamformers is defined so that it follows that of typical loudspeaker panning functions and, for headphone reproduction, that of the head-related transfer functions (HRTFs). The directionality of the second set of high audio quality beamformers is then enhanced with the parametric information derived from the analysis beamformers. Listening tests confirm the perceptual benefit of such type of processing. In direction of arrival (DOA) estimation, histogram analysis of beamforming and active intensity based DOA estimators has been proposed. Numerical simulations and experiments with prototype and commercial microphone arrays show that the accuracy of DOA estimation is improved

Implementation and evaluation of a low complexity microphone array for speaker recognition

Includes bibliographical references (leaves 83-86).This thesis discusses the application of a microphone array employing a noise canceling beamforming technique for improving the robustness of speaker recognition systems in a diffuse noise field

MVDR broadband beamforming using polynomial matrix techniques

This thesis addresses the formulation of and solution to broadband minimum variance distortionless response (MVDR) beamforming. Two approaches to this problem are considered, namely, generalised sidelobe canceller (GSC) and Capon beamformers. These are examined based on a novel technique which relies on polynomial matrix formulations. The new scheme is based on the second order statistics of the array sensor measurements in order to estimate a space-time covariance matrix. The beamforming problem can be formulated based on this space-time covariance matrix. Akin to the narrowband problem, where an optimum solution can be derived from the eigenvalue decomposition (EVD) of a constant covariance matrix, this utility is here extended to the broadband case. The decoupling of the space-time covariance matrix in this case is provided by means of a polynomial matrix EVD. The proposed approach is initially exploited to design a GSC beamformer for a uniform linear array, and then extended to the constrained MVDR, or Capon, beamformer and also the GSC with an arbitrary array structure. The uniqueness of the designed GSC comes from utilising the polynomial matrix technique, and its ability to steer the array beam towards an off-broadside direction without the pre-steering stage that is associated with conventional approaches to broadband beamformers. To solve the broadband beamforming problem, this thesis addresses a number of additional tools. A first one is the accurate construction of both the steering vectors based on fractional delay filters, which are required for the broadband constraint formulation of a beamformer, as for the construction of the quiescent beamformer. In the GSC case, we also discuss how a block matrix can be obtained, and introduce a novel paraunitary matrix completion algorithm. For the Capon beamformer, the polynomial extension requires the inversion of a polynomial matrix, for which a residue-based method is proposed that offers better accuracy compared to previously utilised approaches. These proposed polynomial matrix techniques are evaluated in a number of simulations. The results show that the polynomial broadband beamformer (PBBF) steersthe main beam towards the direction of the signal of interest (SoI) and protects the signal over the specified bandwidth, and at the same time suppresses unwanted signals by placing nulls in their directions. In addition to that, the PBBF is compared to the standard time domain broadband beamformer in terms of their mean square error performance, beam-pattern, and computation complexity. This comparison shows that the PBBF can offer a significant reduction in computation complexity compared to its standard counterpart. Overall, the main benefits of this approach include beam steering towards an arbitrary look direction with no need for pre-steering step, and a potentially significant reduction in computational complexity due to the decoupling of dependencies of the quiescent beamformer, blocking matrix, and the adaptive filter compared to a standard broadband beamformer implementation.This thesis addresses the formulation of and solution to broadband minimum variance distortionless response (MVDR) beamforming. Two approaches to this problem are considered, namely, generalised sidelobe canceller (GSC) and Capon beamformers. These are examined based on a novel technique which relies on polynomial matrix formulations. The new scheme is based on the second order statistics of the array sensor measurements in order to estimate a space-time covariance matrix. The beamforming problem can be formulated based on this space-time covariance matrix. Akin to the narrowband problem, where an optimum solution can be derived from the eigenvalue decomposition (EVD) of a constant covariance matrix, this utility is here extended to the broadband case. The decoupling of the space-time covariance matrix in this case is provided by means of a polynomial matrix EVD. The proposed approach is initially exploited to design a GSC beamformer for a uniform linear array, and then extended to the constrained MVDR, or Capon, beamformer and also the GSC with an arbitrary array structure. The uniqueness of the designed GSC comes from utilising the polynomial matrix technique, and its ability to steer the array beam towards an off-broadside direction without the pre-steering stage that is associated with conventional approaches to broadband beamformers. To solve the broadband beamforming problem, this thesis addresses a number of additional tools. A first one is the accurate construction of both the steering vectors based on fractional delay filters, which are required for the broadband constraint formulation of a beamformer, as for the construction of the quiescent beamformer. In the GSC case, we also discuss how a block matrix can be obtained, and introduce a novel paraunitary matrix completion algorithm. For the Capon beamformer, the polynomial extension requires the inversion of a polynomial matrix, for which a residue-based method is proposed that offers better accuracy compared to previously utilised approaches. These proposed polynomial matrix techniques are evaluated in a number of simulations. The results show that the polynomial broadband beamformer (PBBF) steersthe main beam towards the direction of the signal of interest (SoI) and protects the signal over the specified bandwidth, and at the same time suppresses unwanted signals by placing nulls in their directions. In addition to that, the PBBF is compared to the standard time domain broadband beamformer in terms of their mean square error performance, beam-pattern, and computation complexity. This comparison shows that the PBBF can offer a significant reduction in computation complexity compared to its standard counterpart. Overall, the main benefits of this approach include beam steering towards an arbitrary look direction with no need for pre-steering step, and a potentially significant reduction in computational complexity due to the decoupling of dependencies of the quiescent beamformer, blocking matrix, and the adaptive filter compared to a standard broadband beamformer implementation

Study and applications of retrodirective and self adaptive electromagnetic-wave phase controls to a Mars probe

Computer analyses of retrodirective, and self adaptive antenna phase control techniques for Mars prob

A beam pattern design procedure for multidimensional sonar arrays empolying minimum variance beamforming

Submitted in partial fulfillment of the requirements for the degree of Ocean Engineer at the Massachusetts Institute of Technology and the Woods Hole Oceanographic Institution September 1990This paper develops a beam pattern design procedure for general multidimensional irregular sonar arrays that incorporates the not well understood effects of array geometry into the design process. The procedure is implemented by generating a "penalty function" in a spectral covariance function form. Processing the penalty function causes beam pattern high sidelobes to be penalized and the main lobe to be emphasized. This is accomplished by forming the penalty function in terms of an isotropic noise field of specified strength modified with a finite sector of low coherent energy and stabilized with incoherent sensor noise. By inputting the penalty function into a minimum variance beamformer, the beam pattern and aperture weights are calculated based on the given array geometry. The beamformer used is Capon's Maximum Likelihood Method. The array used to test the procedure is located on a sixty degree sector of a cylindrical surface. The procedure is implemented by two different methods, each with some desirable characteristics. One method suppresses sidelobes directly by the placement of nulls. The other method suppresses sidelobes indirectly by the enhancement of the main lobe with anti-nulls. Both methods are evaluated in terms of a sensitivity factor which constrains the maximum white noise array gain. Results show that both methods result in sidelobe levels that range from 20 to 35 dB lower compared to a conventional beam pattern with uniform aperture weighting and that the design procedure is applicable to beam patterns steered to both true broadside and to off-broadside directions

Wideband data-independent beamforming for subarrays

The desire to operate large antenna arrays for e.g. RADAR applications over a wider frequency range is currently limited by the hardware, which due to weight, cost and size only permits complex multipliers behind each element. In contrast, wideband processing would have to rely on tap delay lines enabling digital filters for every element.As an intermediate step, in this thesis we consider a design where elements are grouped into subarrays, within which elements are still individually controlled by narrowband complex weights, but where each subarray output is given a tap delay line or finite impulse response digital filter for further wideband processing. Firstly, this thesis explores how a tap delay line attached to every subarray can be designed as a delay-and-sum beamformer. This filter is set to realised a fractional delay design based on a windowed sinc function. At the element level, we show that designing a narrowband beam w.r.t. a centre frequency of wideband operation is suboptimal,and suggest an optimisation technique that can yield sufficiently accurate gain over a frequency band of interest for an arbitrary look direction, which however comes at the cost of reduced aperture efficiency, as well as significantly increased sidelobes. We also suggest an adaptive method to enhance the frequency characteristic of a partial wideband array design, by utilising subarrays pointing in different directions in different frequency bands - resolved by means of a filter bank - to adaptively suppress undesired components in the beam patterns of the subarrays. Finally, the thesis proposes a novel array design approach obtained by rotational tiling of subarrays such that the overall array aperture is densely constructed from the same geometric subarray by rotation and translation only. Since the grating lobes of differently oriented subarrays do not necessarily align, an effective grating lobe attenuation w.r.t. the main beam is achieved. Based on a review of findings from geometry,a number of designs are highlight and transformed into numerical examples, and the theoretically expected grating lobe suppression is compared to uniformly spaced arrays.Supported by a number of models and simulations, the thesis thus suggests various numerical and hardware design techniques, mainly the addition of tap-delay-line per subarray and some added processing overhead, that can help to construct a large partial wideband array close in wideband performance to currently existing hardware.The desire to operate large antenna arrays for e.g. RADAR applications over a wider frequency range is currently limited by the hardware, which due to weight, cost and size only permits complex multipliers behind each element. In contrast, wideband processing would have to rely on tap delay lines enabling digital filters for every element.As an intermediate step, in this thesis we consider a design where elements are grouped into subarrays, within which elements are still individually controlled by narrowband complex weights, but where each subarray output is given a tap delay line or finite impulse response digital filter for further wideband processing. Firstly, this thesis explores how a tap delay line attached to every subarray can be designed as a delay-and-sum beamformer. This filter is set to realised a fractional delay design based on a windowed sinc function. At the element level, we show that designing a narrowband beam w.r.t. a centre frequency of wideband operation is suboptimal,and suggest an optimisation technique that can yield sufficiently accurate gain over a frequency band of interest for an arbitrary look direction, which however comes at the cost of reduced aperture efficiency, as well as significantly increased sidelobes. We also suggest an adaptive method to enhance the frequency characteristic of a partial wideband array design, by utilising subarrays pointing in different directions in different frequency bands - resolved by means of a filter bank - to adaptively suppress undesired components in the beam patterns of the subarrays. Finally, the thesis proposes a novel array design approach obtained by rotational tiling of subarrays such that the overall array aperture is densely constructed from the same geometric subarray by rotation and translation only. Since the grating lobes of differently oriented subarrays do not necessarily align, an effective grating lobe attenuation w.r.t. the main beam is achieved. Based on a review of findings from geometry,a number of designs are highlight and transformed into numerical examples, and the theoretically expected grating lobe suppression is compared to uniformly spaced arrays.Supported by a number of models and simulations, the thesis thus suggests various numerical and hardware design techniques, mainly the addition of tap-delay-line per subarray and some added processing overhead, that can help to construct a large partial wideband array close in wideband performance to currently existing hardware

Microwave antenna system for passive discrimination

A novel passive antenna system, capable of discriminating specific electromagnetic signals is addressed. This antenna system will be able to detect signals of certain bandwidths, amplitudes and propagation directions. The philosophy behind this design was to maximise the signal discrimination at a stage prior to reception. The development of such systems could relieve the work involved in post detection discrimination, which may be time consuming and expensive. A major motivation of these studies lies in the difficulties inherent in signal detection for mobile radio communication systems operating at microwave frequencies. Such an antenna system consists of two components. They are the filter section and the detector array. The filter is designed in such a way that only the near normal signal to the locally flat area will be admitted and the rest reflected. The detector array will be at an appropriate position below the filter. Two types of filter structures have been studied for this angular filtering property. They are the Dielectric Multilayers (DML) and periodic arrays of slots as Frequency Selective Surfaces (FSS). DML are constructed by stacking layers of dielectric material whose permittivities vary in a near sinusoidal manner. Such a structure is known to have the ability to admit certain frequency bands of signals. The conventional transmission/reflection matrix method is used for its analysis. Also an optimisation procedure is carried out to minimise the loss of the signal in the DML. The characteristics of the DML as a beam-director and Beam-shaper have also been investigated. FSS exhibit the characteristics of band pass and band stop filters, depending upon the nature of the surface (periodic arrays of elements or slots) . Here the band pass nature is utilised by using arrays of slotted elements. These surfaces are tuned to admit narrow band signals. The well-known modal analysis method has been employed to study the FSS characteristics. The FSS have been studied in the context of frequency scanning, beam shaping, beam directing as well as angular scanning. A prototype has been constructed to simulate a multi signal environment in which the above structures have been experimentally assessed

Broadband adaptive beamforming with low complexity and frequency invariant response

This thesis proposes different methods to reduce the computational complexity as well as increasing the adaptation rate of adaptive broadband beamformers. This is performed exemplarily for the generalised sidelobe canceller (GSC) structure. The GSC is an alternative implementation of the linearly constrained minimum variance beamformer, which can utilise well-known adaptive filtering algorithms, such as the least mean square (LMS) or the recursive least squares (RLS) to perform unconstrained adaptive optimisation.A direct DFT implementation, by which broadband signals are decomposed into frequency bins and processed by independent narrowband beamforming algorithms, is thought to be computationally optimum. However, this setup fail to converge to the time domain minimum mean square error (MMSE) if signal components are not aligned to frequency bins, resulting in a large worst case error. To mitigate this problem of the so-called independent frequency bin (IFB) processor, overlap-save based GSC beamforming structures have been explored. This system address the minimisation of the time domain MMSE, with a significant reduction in computational complexity when compared to time-domain implementations, and show a better convergence behaviour than the IFB beamformer. By studying the effects that the blocking matrix has on the adaptive process for the overlap-save beamformer, several modifications are carried out to enhance both the simplicity of the algorithm as well as its convergence speed. These modifications result in the GSC beamformer utilising a significantly lower computational complexity compare to the time domain approach while offering similar convergence characteristics.In certain applications, especially in the areas of acoustics, there is a need to maintain constant resolution across a wide operating spectrum that may extend across several octaves. To attain constant beamwidth is difficult, particularly if uniformly spaced linear sensor array are employed for beamforming, since spatial resolution is reciprocally proportional to both the array aperture and the frequency. A scaled aperture arrangement is introduced for the subband based GSC beamformer to achieve near uniform resolution across a wide spectrum, whereby an octave-invariant design is achieved. This structure can also be operated in conjunction with adaptive beamforming algorithms. Frequency dependent tapering of the sensor signals is proposed in combination with the overlap-save GSC structure in order to achieve an overall frequency-invariant characteristic. An adaptive version is proposed for frequency-invariant overlap-save GSC beamformer. Broadband adaptive beamforming algorithms based on the family of least mean squares (LMS) algorithms are known to exhibit slow convergence if the input signal is correlated. To improve the convergence of the GSC when based on LMS-type algorithms, we propose the use of a broadband eigenvalue decomposition (BEVD) to decorrelate the input of the adaptive algorithm in the spatial dimension, for which an increase in convergence speed can be demonstrated over other decorrelating measures, such as the Karhunen-Loeve transform. In order to address the remaining temporal correlation after BEVD processing, this approach is combined with subband decomposition through the use of oversampled filter banks. The resulting spatially and temporally decorrelated GSC beamformer provides further enhanced convergence speed over spatial or temporal decorrelation methods on their own

On an acoustic field generated by subsonic jet at low Reynolds numbers

An acoustic field generated by subsonic jets at low Reynolds numbers was investigated. This work is motivated by the need to increase the fundamental understanding of the jet noise generation mechanism which is essential to the development of further advanced techniques of noise suppression. The scope of this study consists of two major investigation. One is a study of large scale coherent structure in the jet turbulence, and the other is a study of the Reynolds number dependence of jet noise. With this in mind, extensive flow and acoustic measurements in low Reynolds number turbulent jets (8,930 less than or equal to M less than or equal to 220,000) were undertaken using miniature nozzles of the same configuration but different diameters at various exist Mach numbers (0.2 less than or equal to M less than or equal to 0.9)

Static and reconfigurable devices for near-field and far-field terahertz applications

The terahertz frequency electromagnetic radiation has gathered a growing interest from the scientific and technological communities in the last 30 years, due to its capability to penetrate common materials, such as paper, fabrics, or some plastics and offer information on a length scale between 100 µm and 1 mm. Moreover, terahertz radiation can be employed for wireless communications, because it is able to sustain terabit-per-second wireless links, opening to the possibility of a new generation of data networks. However, the terahertz band is a challenging range of the electromagnetic spectrum in terms of technological development and it falls amidst the microwave and optical techniques. Even though this so-called “terahertz gap” is progressively narrowing, the demand of efficient terahertz sources and detectors, as well as passive components for the management of terahertz radiation, is still high. In fact, novel strategies are currently under investigation, aiming at improving the performance of terahertz devices and, at the same time, at reducing their structure complexity and fabrication costs. In this PhD work, two classes of devices are studied, one for near-field focusing and one for far-field radiation with high directivity. Some solutions for their practical implementation are presented. The first class encompasses several configurations of diffractive lenses for focusing terahertz radiation. A configuration for a terahertz diffractive lens is proposed, numerically optimized, and experimentally evaluated. It shows a better resolution than a standard configuration. Moreover, this lens is investigated with regard to the possibility to develop terahertz diffractive lenses with a tunable focal length by means of an electro-optical control. Preliminary numerical data present a dual-focus capability at terahertz frequencies. The second class encompasses advanced radiating systems for controlling the far-field radiating features at terahertz frequencies. These are designed by means of the formalism of leaky-wave theory. Specifically, the use of an electro-optical material is considered for the design of a leaky-wave antenna operating in the terahertz range, achieving very promising results in terms of reconfigurability, efficiency, and radiating capabilities. Furthermore, different metasurface topologies are studied. Their analytical and numerical investigation reveals a high directivity in radiating performance. Directions for the fabrication and experimental test at terahertz frequencies of the proposed radiating structures are addressed