DOA Convergence of Unstructured Distributed Arrays with Time-varying and Space-varying Morphologies

This thesis mainly focuses on the research of the factors that influence the accuracy and efficiency of a UAV-based radio frequency (RF) and microwave data collection system. Swarming UAVs can be utilized to create the unstructured morphing antenna arrays that reduce aliasing and improve convergence in sub-space direction of arrival techniques. This thesis first reports on the ramifications of using unstructured antenna arrays based on sub-space techniques. This work evaluates the classical MUSIC algorithm and root-MUSIC algorithm, and Fourier domain root-MUSIC algorithm (FD Root-MUSIC). Compared to the MUSIC algorithm, the root-MUSIC algorithm avoids the search of spatial spectrum, reduces the computational complexity and improves the ability of real world applications. Then, this thesis comes up with the data model for the UAV swarming system. Based on the data model, this work examines the impact of UAV swarm density and heterogeneity on synthetic aperture DOA convergence. The synthetic aperture is derived from the displacement of distributed UAVs operating in a sparse volumetric swarm. Heterogeneity arises from the changing orientation of a UAV’s antenna and receiving pattern function as it swarms in the distributed cluster of UAVs. This alters the UAVs’ antenna pattern functions over time and alters the convergence and overall performance properties of vector-space direction of arrival techniques. This work evaluates the impact of the swarm density and orientation in this framework and studies the convergence and error using MUSIC algorithm. This work also discusses the impact of different type of errors introduced from UAV swarming. Furthermore, this thesis examines the DOA convergence performance of location-varying volumetric random array using MUSIC algorithm. Simulation and measurements for up to sixteen elements on a thirty-two-location test platform are provided and comparisons are made to benchmark their performance with theoretical expectations. MATLAB simulation indicates that the volumetric random arrays can be applied in a very noisy condition by increasing the iterations and multiplying the MUSIC spectrum and experimental observations demonstrate that the system accurately capture the azimuthal and elevation angles of the source. At last, this thesis investigates and designs the tunable FM band monopole and loop antennas to locate the FM broadcasting stations. The wavelength of the FM band is around three meters. This work uses lumped elements and meandering antenna structure technologies to reduce the antenna size and match the antenna. This work also uses the varactor diodes to tune the antenna. However, the antenna becomes electrically small and the antenna gain is so low that it cannot detect the FM signal from the local FM broadcasting stations

Modelling Aspects of Planar Multi-Mode Antennas for Direction-of-Arrival Estimation

Multi-mode antennas are an alternative to classical antenna arrays, and hence a promising emerging sensor technology for a vast variety of applications in the areas of array signal processing and digital communications. An unsolved problem is to describe the radiation pattern of multi-mode antennas in closed analytic form based on calibration measurements or on electromagnetic field (EMF) simulation data. As a solution, we investigate two modeling methods: One is based on the array interpolation technique (AIT), the other one on wavefield modeling (WM). Both methods are able to accurately interpolate quantized EMF data of a given multi-mode antenna, in our case a planar four-port antenna developed for the 6-8.5 GHz range. Since the modeling methods inherently depend on parameter sets, we investigate the influence of the parameter choice on the accuracy of both models. Furthermore, we evaluate the impact of modeling errors for coherent maximum-likelihood direction-of-arrival (DoA) estimation given different model parameters. Numerical results are presented for a single polarization component. Simulations reveal that the estimation bias introduced by model errors is subject to the chosen model parameters. Finally, we provide optimized sets of AIT and WM parameters for the multi-mode antenna under investigation. With these parameter sets, EMF data samples can be reproduced in interpolated form with high angular resolution

Äänikentän tila-analyysi parametrista tilaäänentoistoa varten käyttäen harvoja mikrofoniasetelmia

In spatial audio capturing the aim is to store information about the sound field so that the sound field can be reproduced without a perceptual difference to the original. The need for this is in applications like virtual reality and teleconferencing. Traditionally the sound field has been captured with a B-format microphone, but it is not always a feasible solution due to size and cost constraints. Alternatively, also arrays of omnidirectional microphones can be utilized and they are often used in devices like mobile phones. If the microphone array is sparse, i.e., the microphone spacings are relatively large, the analysis of the sound Direction of Arrival (DoA) becomes ambiguous in higher frequencies. This is due to spatial aliasing, which is a common problem in narrowband DoA estimation. In this thesis the spatial aliasing problem was examined and its effect on DoA estimation and spatial sound synthesis with Directional Audio Coding (DirAC) was studied. The aim was to find methods for unambiguous narrowband DoA estimation. The current State of the Art methods can remove aliased estimates but are not capable of estimating the DoA with the optimal Time-Frequency resolution. In this thesis similar results were obtained with parameter extrapolation when only a single broadband source exists. The main contribution of this thesis was the development of a correlation-based method. The developed method utilizes pre-known, array-specific information on aliasing in each DoA and frequency. The correlation-based method was tested and found to be the best option to overcome the problem of spatial aliasing. This method was able to resolve spatial aliasing even with multiple sources or when the source’s frequency content is completely above the spatial aliasing frequency. In a listening test it was found that the correlation-based method could provide a major improvement to the DirAC synthesized spatial image quality when compared to an aliased estimator.Tilaäänen tallentamisessa tavoitteena on tallentaa äänikentän ominaisuudet siten, että äänikenttä pystytään jälkikäteen syntetisoimaan ilman kuuloaistilla havaittavaa eroa alkuperäiseen. Tarve tälle löytyy erilaisista sovelluksista, kuten virtuaalitodellisuudesta ja telekonferensseista. Perinteisesti äänikentän ominaisuuksia on tallennettu B-formaatti mikrofonilla, jonka käyttö ei kuitenkaan aina ole koko- ja kustannussyistä mahdollista. Vaihtoehtoisesti voidaan käyttää myös pallokuvioisista mikrofoneista koostuvia mikrofoniasetelmia. Mikäli mikrofonien väliset etäisyydet ovat liian suuria, eli asetelma on harva, tulee äänen saapumissuunnan selvittämisestä epäselvää korkeammilla taajuuksilla. Tämä johtuu ilmiöstä nimeltä tilallinen laskostuminen. Tämän diplomityön tarkoituksena oli tutkia tilallisen laskostumisen ilmiötä, sen vaikutusta saapumissuunnan arviointiin sekä tilaäänisynteesiin Directional Audio Coding (DirAC) -menetelmällä. Lisäksi tutkittiin menetelmiä, joiden avulla äänen saapumissuunta voitaisiin selvittää oikein myös tilallisen laskostumisen läsnä ollessa. Työssä havaittiin, että nykyiset ratkaisut laskostumisongelmaan eivät kykene tuottamaan oikeita suunta-arvioita optimaalisella aikataajuusresoluutiolla. Tässä työssä samantapaisia tuloksia saatiin laajakaistaisen äänilähteen tapauksessa ekstrapoloimalla suunta-arvioita laskostumisen rajataajuuden alapuolelta. Työn pääosuus oli kehittää korrelaatioon perustuva saapumissuunnan arviointimenetelmä, joka kykenee tuottamaan luotettavia arvioita rajataajuuden yläpuolella ja useamman äänilähteen ympäristöissä. Kyseinen menetelmä hyödyntää mikrofoniasetelmalle ominaista, saapumissuunnasta ja taajuudesta riippuvaista laskostumiskuviota. Kuuntelukokeessa havaittiin, että korrelaatioon perustuva menetelmä voi tuoda huomattavan parannuksen syntetisoidun tilaäänikuvan laatuun verrattuna synteesiin laskostuneilla suunta-arvioilla

A room acoustics measurement system using non-invasive microphone arrays

This thesis summarises research into adaptive room correction for small rooms and pre-recorded material, for example music of films. A measurement system to predict the sound at a remote location within a room, without a microphone at that location was investigated. This would allow the sound within a room to be adaptively manipulated to ensure that all listeners received optimum sound, therefore increasing their enjoyment. The solution presented used small microphone arrays, mounted on the room's walls. A unique geometry and processing system was designed, incorporating three processing stages, temporal, spatial and spectral. The temporal processing identifies individual reflection arrival times from the recorded data. Spatial processing estimates the angles of arrival of the reflections so that the three-dimensional coordinates of the reflections' origin can be calculated. The spectral processing then estimates the frequency response of the reflection. These estimates allow a mathematical model of the room to be calculated, based on the acoustic measurements made in the actual room. The model can then be used to predict the sound at different locations within the room. A simulated model of a room was produced to allow fast development of algorithms. Measurements in real rooms were then conducted and analysed to verify the theoretical models developed and to aid further development of the system. Results from these measurements and simulations, for each processing stage are presented

Real-time Microphone Array Processing for Sound-field Analysis and Perceptually Motivated Reproduction

This thesis details real-time implementations of sound-field analysis and perceptually motivated reproduction methods for visualisation and auralisation purposes. For the former, various methods for visualising the relative distribution of sound energy from one point in space are investigated and contrasted; including a novel reformulation of the cross-pattern coherence (CroPaC) algorithm, which integrates a new side-lobe suppression technique. Whereas for auralisation applications, listening tests were conducted to compare ambisonics reproduction with a novel headphone formulation of the directional audio coding (DirAC) method. The results indicate that the side-lobe suppressed CroPaC method offers greater spatial selectivity in reverberant conditions compared with other popular approaches, and that the new DirAC formulation yields higher perceived spatial accuracy when compared to the ambisonics method

Augmented Intensity Vectors for Direction of Arrival Estimation in the Spherical Harmonic Domain

Pseudointensity vectors (PIVs) provide a means of direction of arrival (DOA) estimation for spherical microphone arrays using only the zeroth and the first-order spherical harmonics. An augmented intensity vector (AIV) is proposed which improves the accuracy of PIVs by exploiting higher order spherical harmonics. We compared DOA estimation using our proposed AIVs against PIVs, steered response power (SRP) and subspace methods where the number of sources, their angular separation, the reverberation time of the room and the sensor noise level are varied. The results show that the proposed approach outperforms the baseline methods and performs at least as accurately as the state-of-the-art method with strong robustness to reverberation, sensor noise, and number of sources. In the single and multiple source scenarios tested, which include realistic levels of reverberation and noise, the proposed method had average error of 1.5∘ and 2∘, respectively