The growth of wireless applications and the emerging technologies require more efficient management of frequency spectrum. Adaptive resource allocation (i.e. transmission power and transmission rate) and opportunistic spectrum sharing are two key techniques with enormous potential to enhance the efficient utilization of the precious bandwidth and optimizing the performance. Adaptive algorithms are a vital feature of radio resource management (RRM) in third generation (3G) communication systems, and cognitive radio (CR) is a valuable technique with tremendous potential for improving the utilization of the radio spectrum. On the other hand, transition from the current 3G cellular communications systems to the fourth generation (4G) would happen gradually and take several years. During this period, both 3G and 4G technologies are expected to co-exist, and the aforementioned CR technology and adaptation techniques are suitable candidates to be used by service providers for achieving higher data rates in the future heterogeneous networks. In this thesis, novel adaptive transmission algorithms will be developed for shared-spectrum CR networks in the context of direct-sequence code division multiple access (DS-CDMA). This thesis makes several contributions. First, adaptive transmission is integrated into spectrum sharing problem. The joint optimization of power and rate is investigated, where optimality is in the sense of maximizing the average spectral efficiency of the reference cognitive user (CU). A close-form solution for the optimal outer loop power control target signal-to-noise ratio (SNR-target) of the reference CU is derived. The optimization is conducted when the reference CU is using interference-limited opportunistic spectrum access (IL-OSA) technique for utilizing the primary spectrum simultaneously with the primary users. The proposed optimization algorithm's performance is analyzed for a case, wherein the reference CU exploits the licensed spectrum subject to non-violation of the average and the peak received-interference constraints. Moreover, the evaluation is extended for more reliable scenario with better performance, wherein the imposed peak-interference constraint is dynamically set as a function of the number of active primary and cognitive users that exploit the licensed frequency band. Second, a shared-spectrum CR system is considered and the reference CU's total average spectral efficiency is derived when it uses access-bounded opportunistic spectrum access (AB-OSA) for exploiting the primary spectrum. The gain that can be attained by using a CR technology is highlighted and the reference CU performance investigated subject to average transmit power constraints. Finally, a novel access technique is proposed to overcome the AB-OSA limitation in CDMA/CDMA, CR networks. This method is referred to as access-bounded-interference-limited opportunistic spectrum access (AB-IL-OSA). This novel access strategy incorporates a mixed access-and interference- limited spectrum access strategy through spectrum sensing, to maximize the achievable spectral efficiency of the CUs. Various OSA strategies were proposed for CDMA-based CR networks to achieve a balance between the full exploitation of the primary spectrum and the inflicted interference on the primary service.

The future of mobile and personal communications requires a system that can support a large number of users and has the ability to provide high-speed services. Meeting this demand is challenging since wireless communications are subject to these four major constraints: complex fading channels, a scarce usable radio spectrum, and limitations on the power and size of the mobile terminals. Space-time codes, such as the Alamouti/space-time block coding (STBC) and cyclic delay diversity (CDD) can provide diversity and coding gains in multipleinput multiple-output (MIMO) systems over fading channels. However in an ad-hoc network, nodes are often constrained in hardware complexity and size, which makes MIMO systems impractical for certain applications. Cooperative relay diversity schemes have been introduced in an eort to overcome these limitations. Cooperative techniques allow a collection of communication nodes to cooperate with to relay signals amongst each other, eectively creating a virtual antenna array, which combat multipath fading in wireless channels. This thesis investigates relay diversity techniques in cooperative communications, namely the distributed space-time block coding (DSTBC) and relay cyclic delay diversity (RCDD) for OFDM and MC-CDMA systems. The performance analysis of the wireless relay networks under dierent protocols and fading channels are investigated. We derived the formulas for the symbol error rate (SER) of the investigated schemes in fading channels. Finally, a novel hybrid relay diversity (HRD) is presented for OFDMbased system. This new relay structure addresses the issue of achieving a full spatial and multipath diversity by developing a hybrid for CDD and STBC. Simulation on the system performance of HRD shows the superiority of the technique compared to standalone DSTBC and RCDD.

During recent years there has been a continuous increase in traffic load over wireless networks. The next generation of mobile networks must grow to handle the increase in demand in IP traffic. Micro mobility management and provisioning of Quality of Service (QoS) have become two main factors for the successful deployment of the next generation of IP based mobile communication networks. Hierarchical Mobile IPv6 (HMIPv6) protocol has become prominent for providing micro mobility to these networks. Deployment of Mobility Anchor Points (MAPs) in conventional HMIPv6 introduces two main drawbacks in HMIPv6based access networks. Firstly, they are points of bandwidth contention within the access networks. Secondly, Mobile Nodes (MNs) generate excessive handover signalling overhead as a result of ping-pong movement between ARs of different MAP domains. In this thesis, a novel network architecture in HMIPv6 access networks is proposed to enhance the network performance in terms of both drawbacks mentioned above. In the proposed topology, each AR is assigned to more than one MAP by forming overlapped regions between MAP domains. Numerical and simulation analysis are carried out to quantify the impact of the new architecture on the handover signaling overhead, MAP congestion level, packet delay and network throughput. Linear programmes are formulated to optimise the throughput and congestion in access networks. The results illustrate that when MAP domains are overlapped, the lightly loaded MAPs will provide their residual capacities to the ARs located in the heavily loaded MAP domains. Overlapped regions are optimally configured between MAPs in order to minimise the handover signaling overhead. The results indicate that the gain can be considerably increased by deploying the proposed MAP selection mechanism. Three heuristic algorithms were proposed to dynamically adapt to network changes such as traffic and MN’s mobility characteristics. The simulation results show that all three proposed algorithms outperform Sanchis’ non-overlapping partitioning algorithm. A Dynamic QoS aware multi-MAP registration algorithm is proposed so that individual MNs can select a MAP according to their QoS requirement and level of handover support. In all this, the thesis provides a structured framework for the analysis and optimal configuration of overlapping MAP domains within a HMIPv6 based access network. This can enable the next generation of IP mobile networks to efficiently manage the huge volumes of IP traffic that are expected in the future.

Speech technology has had, and is having, a significant impact on society. The mobile phone has created a generation of users who rely upon it to maintain their social network. Mobile phones depend upon voice compression and the AdaptiveMulti- Rate (A.1V1R) Vocoder is the standard for GSM and 03. Automatic Speech Recognition (ASR) is waiting in the wings as the next paradigm. shift in the way users interact with technology; all that is needed is robust embedded applications. Speech processing algorithms are complex, resource intensive, and therefore difficult to implement upon embedded devices. The AMR Vocoder was designed specifically for a 16bit DSP, however implementing ASR upon an embedded device remains a significant challenge. The aim of this work was to research innovative ways to implement speech applications to either reduce the required resources or significantly increase application capabilities. The achievements are: Â? Â? A novel hardware accelerator was designed for the AMR Algebraic Codebook Search which reduced the power consumption by a factor of 5, thereby potentially increasing the battery life of a mobile device. A novel hardware accelerator was developed to speed up the calculation of Observation Probabilities during ASR, which leads to increased robustness and larger vocabularies in embedded devices. A novel hardware accelerator was developed to achieve Speaker Adaptation for a Hidden Markov Model in approximately 4.6 seconds on an embedded device. This is near workstation performance, and has the potential to revolutionise embedded ASR applications.

The antenna constitutes the air interface between wireless equipment and communication channel, thus, efficient operation is of great importance especially in compact user devices. An understanding of its behaviour within any communication system, and an accurate representation of its effects on received signals are vital. The nature of the multipath propagation channel and the local environment, the user's interference with its orientation and immediate vicinity are all factors that impact on antenna performance and need to be addressed. This thesis addresses issues directly arising from the nature of the interaction of impinging waves and antenna radiation patterns. Firstly, an important proportion of this research is devoted to the assessment of the application of the closed-form expression of the mean effective gain (MEG) of the antenna. Although the validity of this expression has been established in stochastic Rayleigh channels, its wider application is now questioned. Results show that the traditional MEG calculation method can result in inaccurate estimations with varying effect and significance especially in deterministic channels. Particularly, as phase is omitted from calculations on account of its random uniform distribution, it is found that phase information is in fact essential to quantify losses due to mismatches in polarisation. Here, a much more rigorous approach is taken in the form of the mean directivity formula which takes full complex information of impinging signals and radiation fields into account . Secondly, an experimental approach is taken in order to investigate the impact of scattering environments on full three dimensional, polarimetric antenna properties. A method for 3D directional channel characterisation built around standard antenna pattern measurement techniques is proposed and validated in a controlled multipath environment. The data from these measurements delivers spatial, phase and magnitude information of the channel around the antenna, within limitations imposed by the beamwidth of the measurement antenna. The results give an important insight into the phase characteristics of the incident field, namely, its continuity across clusters of incoming signal. Finally, the results obtained thus far are brought together in a performance evaluation tool, capable of predicting sensitivity to measured signal in 3D space. Indeed, future device antenna testing procedures must be truly representative of realistic usage scenarios. Tests can in no way be exhaustive in terms of the environment or usage situations they cover; they can however be made to target particularly sensitive scenarios and antenna properties that are known to have more influence on how the propagating signals are treated. The full antenna testing procedure proposed is therefore a complete tool able to predict antenna performance in the spatial and polarimetric domain

This thesis is primarily concerned with the development of adaptive algorithms for equaliser coefficient computation within the highly time variant radio communications environment. More specifically the problem of equaliser coefficient computation within the pan European digital mobile radio system is considered. The work encompasses both equaliser and channel estimator adaptive algorithms and techniques for the automatic synthesis of linear transversal equalisers (LTE), decision feedback equalisers (DFE) and adaptive maximum likelihood sequence estimators (MLSE) are developed. Within the thesis it is shown that equaliser performance can be significantly improved by adaptively updating the equaliser throughout unknown data transmission. Initially the performance of conventional techniques, such as gradient search (GS) and least squares (LS) algorithms, when employed in this respect is investigated. Although each is shown to yield performance improvement over the system in which no adaptive update is employed, it is shown that under highly time variant conditions the performance of the conventional algorithms is subject to several limitations. This conclusion provides the motivation for the development of a number of alternative adaptive algorithms which offer performance advantage under highly time variant conditions. Two classes of algorithm are proposed. Within each a priori knowledge of the time variant characteristics of the channel is used in order to partially cancel estimation error due to the channel time variation. Within the first this is achieved by augmenting the update equation of each of the conventional algorithms by inclusion of an additional parameter set representing an estimate of the rate of change (ROC) of the channel coefficients. The algorithms are thereby able to form a prediction of the instantaneous channel variation and, therefore, to compensate for the channel non-stationarity. In the second class of algorithm a predetermined model of channel time variability is incorporated directly into the algorithm structure.

The ability of higher-order statistics to preserve phase information makes them particularly useful in the study of non-Gaussian stationary linear processes. This thesis presents some new results in the estimation of the parameters of MA models, from the cumulants of the output processes. New general relationships between the output cumulants and the system parameters are presented. These relationships are used to develop new system identification methods which use only third-order or only fourth-order cumulants. Both least squares and recursive versions of the system identification algorithms are presented. The identifiability of the algorithms is formally proved and asymptotic performance expressions are derived. The issues of MA model order selection and ARMA parameter estimation are also discussed. In many applications the primary objective is the estimation of the inverse filter coefficients. New general relationships are presented which involve the output cumulants and the inverse filter coefficients. Based on these relationships, a unified description of existing deconvolution methods is presented and new deconvolution methods based on fourth-order cumulants or on a combination of second- and fourth-order cumulants are developed. Finally, this thesis investigates properties that characterise sets of numbers as being the cumulants of some MA model. This problem is easier to analyse if the numbers are organised in a matrix form and the properties are expressed using matrix theoretical notions such as the rank of a matrix and the features of linear structured matrices. Because of estimation errors sets of sample cumulants are not real cumulants of some MA model. Based on the characteristic properties of sets of cumulants, this thesis presents an iterative composite property mapping algorithm which maps the sample cumulants to a set of enhanced cumulants. If convergence is achieved the enhanced cumulants are true cumulants of some MA model. If convergence has not been achieved, the enhanced cumulants are "nearer" to a set of true MA cumulants than the original set of sample cumulants was. It is shown that when the enhanced cumulants are used for parameter estimation, they can improve the performance of parameter estimation algorithms.

This thesis presents high performance turbo decoder architecture for VLSI implementation in terms of area, power and critical path delay. A Max-Log-MAP (MLMAP) algorithm is used to implement the turbo decoder with sliding window (SW) method to reduce the latency. Low power and area efficient turbo decoder implementation is achieved by reducing memory blocks required to control the SW method and to store the branch metrics used for computing log-likelihood-ratio (LLR). Retiming and reordering method is applied to implementing the units needed to calculate the LLR and the state metrics, to save hardware costs. A novel method is proposed to achieve high speed turbo decoder implementation for high throughput without significant area and power overheads. The proposed method addresses the inherent critical path delay problem in the state metric computation process by normalising the branch metrics. A two-step soft-output Viterbi algorithm (TSOVA) based turbo decoder is implemented exploiting a novel concept for implementing a traceback algorithm (TBA) to achieve low area and power turbo decoder implementations as compared to the MLMAP turbo decoder without any significant BER performance degradation. Two reconfigurable application specific turbo decoders are implemented to support variable constraint length and binary and duo-binary turbo codes for targeting various wireless communication systems. The reconfigurable turbo decoder architectures are realised by a proposed mapping method applied to the process for computing the state metrics and the LLR values. It is found that radix-4 based turbo decoder architecture can be exploited to implement the reconfigurable turbo decoder for binary and duo-binary turbo codes.

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The work described in this thesis is concerned with the development of hardware efficient, image processing and machine vision algorithms for implementation, using recently developed low cost CMOS cameras. These allow the integration of processing on the same silicon substrate as the imaging sensor. The general approach differs from other image processing research in that algorithms are being developed for a target architecture, rather than hardware being developed for a particular image processing function. A particular application, namely intruder detection and tracking, has been chosen, to demonstrate this approach. The use of image processing in alarm systems has many advantages over active electronics: the main ones being installation costs and reliability. In particular, stereo vision has the potential of providing an invisible wall and estimates of intruder significance. However it is also desirable that alarm systems have wide angle lenses. Wide angle lenses create particular problems for stereo vision, in relation to pixel quantisation. Techniques to provide a low cost sub-pixel estimate of disparity are presented. Further, an original stereo matching algorithm is described which solves the stereo correspondence problem, in a computationally simple manner. Adaptations are also made to the low level segmentation stages which would allow an efficient implementation using CMOS sensors and processing. Other savings have been made by eliminating digital floating point calculations, multiplications and divisions at the lower levels of processing. Also, due to the reduced data rates required for global frame to frame computation, higher level calculations can be performed on an associated microprocessor. Thus, a Kalman tracking filter has been applied to integrate the three possible disparities from three cameras, with experimentally calculated error covariance matrices. A results chapter describes the extraction of these matrices, together with simulations of the algorithms applied to twelve different sequences. These show that the system could be effective as an alarm system. Also described, at various stages in the thesis, are possible hardware implementations of the algorithms and partitions between analogue and digital circuitry. The thesis finishes with some general conclusions.