Conventional pistonic loudspeakers, by employing whole-body vibration of the diaphragm, can reproduce good quality sound at the low end of the audio spectrum. Flat panel speakers, on the other hand, are better at high frequency operation as the reproduced sound at high frequency from a flat panel speaker is not omni-directional as in the case of a conventional loudspeaker. Although flat-panel speakers are compact, small and have a better high frequency response the poor reproduction of bass sound limits its performance severely. In addition, the flat panel speakers have a poor impulse response. The reason for such poor bass and impulse response is that, unlike the whole body movement of a conventional loudspeaker diaphragm, different parts of the panel in a flat panel loudspeaker vibrates independently. A novel loudspeaker has been successfully designed, developed and operated using miniature electromagnetic transducers in a matrix array configuration. In this device, the whole body vibration of the panel reduces the poor bass and impulse response associated with present flat panel speakers. The multi-actuator approach combines the advantages of conventional whole body motion with that of modern flat panel speakers. An innovative miniature electromagnetic transducer for the proposed loudspeaker has been designed, modelled and built for analysis. Frequency Responses show that this novel transducer is suitable for loudspeaker application because of its steady and consistent output over the whole audible frequency range and for various excitation currents. Measurements on various device configurations of this novel miniature electromagnetic transducer show that a moving coil transducer configuration having a magnetic diaphragm is best suited for loudspeaker applications. Finite element modeling has been used to examine single transducer operation and the magnetic interaction between neighbouring transducers in a matrix array format. Experimental results show the correct positioning of the transducers in a matrix configuration reduces the effects of interferences on the magnetic transducers. In addition, experimental results from the pressure response measurement show an improvement in bass response for the longer array speaker.

The Frequency Assignment Problem (FAP) is an important optimization problem that arises in operational cellular wireless networks. Solution techniques based on meta-heuristic algorithms have been shown to be successful for some test problems but they have not been usually demonstrated on large scale problems that occur in practice. This thesis applies a problem decomposition approach in order to solve FAP in stances with standard meta-heuristics. Three different formulations of the problem are considered in order of difficulty: Minimum Span (MS-FAP), Fixed Spectrum (MS-FAP), and Minimum Interference FAP (MI-FAP). We propose a decomposed assignment technique which aims to divide the initial problem into a number of subproblems and then solves them either independently or in sequence respecting the constraints between them. Finally, partial subproblem solutions are recomposed into a solution of the original problem. Standard implementations of meta-heuristics may require considerable run times to produce good quality results whenever a problem is very large or complex. Our results, obtained by applying the decomposed approach to a Simulated Annealing and a Genetic Algorithm with two different assignment representations (direct and order-based), show that the decomposed assignment approach proposed can improve their outcomes, both in terms of solution quality and runtime. A number of partitioning methods are presented and compared for each FAP, such as clique detection partitioning based on sequential orderings and novel applications of existing graph partitioning and clustering methods adapted for this problem.

The aim of this thesis is to investigate the integration of hardware description lamguaages (HDLs) and automated proof systems. Simulation of circuit designs written in an HDL is an important method of testing their correctness. However, due to the combinatorial explosion of possible inputs it is not feasible to verify designs using simulation alone. Formal hardware verification, using a proof system, has tried to address this issue. Whilst some medium-sized designs have been (partially) verified, industrial take-up of formal methods has been slow. This is partly due to the use of specialised, non-standard notations employed in various formalisms. By embedding a hardware description language in a proof system we hope to clarify the semantics of the particular HDL, and present a more standard interface to formal methodologies. We have given a new static structural operational semantics for a subset of the ELLA hardware language. The formal dynamic semantics of this subset is based on an existing informal model. We embedded the semantics of this HDL in the LAMBDA higher-order logic proof system. The embedding allows meta-theoretical results to be proved about this and other semantics. It has been proved that the semantics computes the least fixed point solution of the circuit description. Another semantics which computes a more defined output has also been embedded, and the relationship between both semantics has been proved formally. A number of paradigms such as operational semantics based formal symbolic simulation, formal interactive (top-down and bottom-up) synthesis, formal hardware generators, proved correct transformations and traditional hardware verification are presented as small case studies. However, scaling up of the examples turned out to be difficult and verification tended to be slow.

This thesis addresses the problem of adaptive channel equalisation in environments where the interfering noise exhibits non–Gaussian behaviour due to impulsive phenomena. The family of alpha-stable distributions has proved to be a suitable and flexible tool for the modelling of signals with impulsive nature. However,non–Gaussian alpha–stable signals have infinite variance, and signal processing techniques based on second order moments are meaningless in such environments. In order to exploit the flexibility of the stable family and still take advantage of the existing signal processing tools, a novel framework for the integration of the stable model in a communications context is proposed, based on a finite dynamic range receiver. The performance of traditional signal processing algorithms designed under the Gaussian assumption may degrade seriously in impulsive environments. When this degradation cannot be tolerated, the traditional signal processing methods must be revisited and redesigned taking into account the non–Gaussian noise statistics. In this direction, the optimum feed–forward and decision feedback Bayesian symbol–by–symbol equalisers for stable noise environments are derived. Then, new analytical tools for the evaluation of systems in infinite variance environments are presented. For the centers estimation of the proposed Bayesian equaliser, a unified framework for a family of robust recursive linear estimation techniques is presented and the underlying relationships between them are identified. Furthermore, the direct clustering technique is studied and robust variants of the existing algorithms are proposed. A novel clustering algorithm is also derived based on robust location estimation. The problem of estimating the stable parameters has been addressed in the literature and a variety of algorithms can be found. Some of these algorithms are assessed in terms of efficiency, simplicity and performance and the most suitable is chosen for the equalisation problem. All the building components of an adaptive Bayesian equaliser are then put together and the performance of the equaliser is evaluated experimentally. The simulation results suggest that the proposed adaptive equaliser offers a significant performance benefit compared with a traditional equaliser, designed under the Gaussian assumption. The implementation of the proposed Bayesian equaliser is simple but the computational complexity can be unaffordable. However, this thesis proposes certain approximations which enable the computationally efficient implementation of the optimum equaliser with negligible loss in performance.

This thesis considers the design of new adaptive equalisers for fading digital communication channels. The role of equalisation is discussed in the context of the functions of a digital radio communication system and both conventional and more recent novel equaliser designs are described. The application of recurrent neural networks to the problem of equalisation is developed from a theoretical study of a single node structure to the design of multinode structures. These neural networks are shown to cancel intersymbol interference in a manner mimicking conventional techniques and simulations demonstrate their sensitivity to symbol estimation errors. In addition the error mechanisms of conventional maximum likelihood equalisers operating on rapidly time-varying channels are investigated and highlight the problems of channel estimation using delayed and often incorrect symbol estimates. The relative sensitivity of Bayesian equalisation techniques to errors in the channel estimate is studied and demonstrates that the structure's equalisation capability is also susceptible to such errors. Applications of multiple channel estimator methods are developed, leading to reduced complexity structures which trade performance for a smaller computational load. These novel structures are shown to provide an improvement over the conventional techniques, especially for rapidly time-varying channels, by reducing the time delay in the channel estimation process. Finally, the use of confidence measures of the equaliser's symbol estimates in order to improve channel estimation is studied and isolates the critical areas in the development of the technique — the production of reliable confidence measures by the equalisers and the statistics of symbol estimation error bursts.

In any situation where private, proprietary or highly confidential material is being dealt with, the need to consider aspects of data security has grown ever more important. It is usual to secure such data from its source, over networks and on to the intended recipient. However, data security considerations typically stop at the recipient's processor, leaving connections to a display transmitting raw data which is increasingly in a digital format and of value to an adversary. With a progression to wireless display technologies the prominence of this vulnerability is set to rise, making the implementation of 'secure display' increasingly desirable. Secure display takes aspects of data security right to the display panel itself, potentially minimising the cost, component count and thickness of the final product. Recent developments in display technologies should help make this integration possible. However, the processing of large quantities of time-sensitive data presents a significant challenge in such resource constrained environments. Efficient high- throughput decryption is a crucial aspect of the implementation of secure display and one for which the widely used and well understood block cipher may not be best suited. Stream ciphers present a promising alternative and a number of strong candidate algorithms potentially offer the hardware speed and efficiency required. In the past, similar stream ciphers have suffered from algorithmic vulnerabilities. Although these new-generation designs have done much to respond to this concern, the relatively short 80-bit key lengths of some proposed hardware candidates, when combined with ever-advancing computational power, leads to the thesis identifying exhaustive search of key space as a potential attack vector. To determine the value of protection afforded by such short key lengths a unique hardware key search engine for stream ciphers is developed that makes use of an appropriate data element to improve search efficiency. The simulations from this system indicate that the proposed key lengths may be insufficient for applications where data is of long-term or high value. It is suggested that for the concept of secure display to be accepted, a longer key length should be used.

Modem telecommunication services require increasing data rates for both mobile and fixed applications. At frequencies in the range 2.5 GHz to 6 GHz physical constraints on the size of equipment result in antenna with moderate directivity typically with an antenna beam width of 20 degrees or greater. Thus building and ground clutter is present within the first Fresnel zones of the antenna system which gives rise to multi-path propagation. This multi-path propagation (average delay and RMS delay spread) has been investigated using a wideband FMCW channel sounder that is capable of operation at a number of frequencies. The channel sounder has been based upon a parallel architecture sounder operating within the 2 GHz band with a number of frequency conversion modules to translate operation to the new frequency bands under study. Two primary configurations have been explored. In the first of these, propagation has been measured simultaneously within the 2.5 GHz, 3.4 GHz and 5.7 GHz bands. This is believed to be novel and original. In the second configuration four parallel channels operating within the 5.7 GHz band may be operated simultaneously. This configuration supports multiple antennas at the receiver. To support the work in the bands from 2.5 GHz to 6 GHz wideband discone antenna have been designed and fabricated. A system to provide relative gain and phase calibration for up to four antennas has been developed and demonstrated. This is also believed to represent a novel method of performing antenna and array calibration. Finally, the frequency converters have been used in conjunction with additional components to provide an FMCพ sounder operating within the 60 GHz Oxygen absorption band. This work is novel in that up to 1 GHz of spectrum can be swept. To support this work a significant number of microwave components have been designed and developed. In particular a novel wide band balanced X3 multiplier and a novel impedance-matched amplitude-equaliser (to provide amplifier gain-slope equalisation) has been developed. Channel soundings have been performed at three frequencies simultaneously using band specific and common antenna. The average delay and RMS delay spread have been demonstrated to be essentially frequency independent for the environments evaluated.

Security is considered the most important requirement for the success of electronic commerce, which is built based on the security of hash functions, encryption algorithms and pseudorandom number generators. Chaotic systems and security algorithms have similar properties including sensitivity to any change or changes in the initial parameters, unpredictability, deterministic nature and random-like behaviour. Several security algorithms based on chaotic systems have been proposed; unfortunately some of them were found to be insecure and/or slow. In view of this, designing new secure and fast security algorithms based on chaotic systems which guarantee integrity, authentication and confidentiality is essential for electronic commerce development. In this thesis, we comprehensively explore the analysis and design of security primitives based on chaotic systems for electronic commerce: hash functions, encryption algorithms and pseudorandom number generators. Novel hash functions, encryption algorithms and pseudorandom number generators based on chaotic systems for electronic commerce are proposed. The securities of the proposed algorithms are analyzed based on some well-know statistical tests in this filed. In addition, a new one-dimensional triangle-chaotic map (TCM) with perfect chaotic behaviour is presented. We have compared the proposed chaos-based hash functions, block cipher and pseudorandom number generator with well-know algorithms. The comparison results show that the proposed algorithms are better than some other existing algorithms. Several analyses and computer simulations are performed on the proposed algorithms to verify their characteristics, confirming that these proposed algorithms satisfy the characteristics and conditions of security algorithms. The proposed algorithms in this thesis are high-potential for adoption in e-commerce applications and protocols.

A mobile channel simulator can be constructed either in the time domain using a tapped delay line filter or in the frequency domain using the time variant transfer function of the channel. Transfer function modelling has many advantages over impulse response modelling. Although the transfer function channel model has been envisaged by several researchers as an alternative to the commonly employed tapped delay line model, so far it has not been implemented. In this work, channel simulators for single carrier and multicarrier OFDM system based on time variant transfer function of the channel have been designed and implemented using DSP techniques in SIMULINK. For a single carrier system, the simulator was based on Bello's transfer function channel model. Bello speculated that about 10Βτ(_MAX) frequency domain branches might result in a very good approximation of the channel (where в is the signal bandwidth and τ(_MAX) is the maximum excess delay of the multi-path channel). The simulation results showed that 10Bτ(_MAX) branches gave close agreement with the tapped delay line model(where Be is the coherence bandwidth). This number is π times higher than the previously speculated 10Bτ(_MAX).For multicarrier OFDM system, the simulator was based on the physical (PHY) layer standard for IEEE 802.16-2004 Wireless Metropolitan Area Network (WirelessMAN) and employed measured channel transfer functions at the 2.5 GHz and 3.5 GHz bands in the simulations. The channel was implemented in the frequency domain by carrying out point wise multiplication of the spectrum of OFDM time The simulator was employed to study BER performance of rate 1/2 and rate 3/4 coded systems with QPSK and 16-QAM constellations under a variety of measured channel transfer functions. The performance over the frequency selective channel mainly depended upon the frequency domain fading and the channel coding rate.

Due to the incessant requirement for higher performance radio systems, wireless designers have been constantly seeking ways to improve spectrum efficiency, link reliability, service quality, and radio network coverage. During the past few years, space-time technology which employs multiple antennas along with suitable signalling schemes and receiver architectures has been seen as a powerful tool for the implementation of the aforementioned requirements. In particular, the concept of communications via Multiple-Input Multiple-Output (MIMO) links has emerged as one of the major contending ideas for next generation ad-hoc and cellular systems. This is inherently due to the capacities expected when multiple antennas are employed at both ends of the radio link. Such a mobile radio propagation channel constitutes a MIMO system. Multiple antenna technologies and in particular MIMO signalling are envisaged for a number of standards such as the next generation of Wireless Local Area Network (WLAN) technology known as 802.1 ln and the development of the Worldwide Interoperability for Microwave Access (WiMAX) project, such as the 802.16e. For the efficient design, performance evaluation and deployment of such multiple antenna (space-time) systems, it becomes increasingly important to understand the characteristics of the spatial radio channel. This criterion has led to the development of new sounding systems, which can measure both spatial and temporal channel information. In this thesis, a novel semi-sequential wideband MIMO sounder is presented, which is suitable for high-resolution radio channel measurements. The sounder produces a frequency modulated continuous wave (FMCW) or chirp signal with variable bandwidth, centre frequency and waveform repetition rate. It has programmable bandwidth up to 300 MHz and waveform repetition rates up to 300 Hz, and could be used to measure conventional high- resolution delay/Doppler information as well as spatial channel information such as Direction of Arrival (DOA) and Direction of Departure (DOD). Notably the knowledge of the angular information at the link ends could be used to properly design and develop systems such as smart antennas. This thesis examines the theory of multiple antenna propagation channels, the sounding architecture required for the measurement of such spatial channel information and the signal processing which is used to quantify and analyse such measurement data. Over 700 measurement files were collected corresponding to over 175,000 impulse responses with different sounder and antenna array configurations. These included measurements in the Universal Mobile Telecommunication Systems Frequency Division Duplex (UMTS-FDD) uplink band, the 2.25 GHz and 5.8 GHz bands allocated for studio broadcast MIMO video links, and the 2.4 GHz and 5.8 GHz ISM bands allocated for Wireless Local Area Network (WLAN) activity as well as for a wide range of future systems defined in the WiMAX project. The measurements were collected predominantly for indoor and some outdoor multiple antenna channels using sounding signals with 60 MHz, 96 MHz and 240 MHz bandwidth. A wide range of different MIMO antenna array configurations are examined in this thesis with varying space, time and frequency resolutions. Measurements can be generally subdivided into three main categories, namely measurements at different locations in the environment (static), measurements while moving at regular intervals step by step (spatial), and measurements while the receiver (or transmitter) is on the move (dynamic). High-scattering as well as time-varying MIMO channels are examined for different antenna array structures.