The aim of this project is to address the challenges associated with extending the radio frequency capability of Electric Potential Sensors to greater than 10 MHz. This has culminated in a single broadband sensor, with a frequency range of 200 Hz to greater than 200 MHz. The use of Electric Potential Sensors for the measurement of electric field with minimal perturbation has already been demonstrated at Sussex. These high impedance sensors have been successfully employed in measuring signals with frequencies in the range 1 mHz to 2 MHz. Many different versions of these sensors have been produced to cater for specific measurement requirements in a wide variety of experimental situations. From the point of view of this project, the relevant prior work is the acquisition of a 2 MHz electric field nuclear magnetic resonance signal, and the non-destructive testing of composite materials at audio frequency. Two very distinct electric field measurement scenarios are described which illustrate the diverse capabilities of the broadband sensor. Firstly, an electric field readout system for nuclear quadrupole resonance is demonstrated for the first time, with a sodium chlorate sample at a frequency of 30 MHz. Nuclear quadrupole resonance is an important technique with applications in the detection of explosives and narcotics. Unlike nuclear magnetic resonance a large magnet is not required, opening up the possibility of portable equipment. The electric field readout system is shown to be simpler than the conventional magnetic readout and may therefore contribute to the development of portable devices. Secondly, a broadband, high spatial resolution microscope system for materials characterisation with four different imaging modes is described. This includes; the surface topography of a conducting sample; the dielectric constant variation in glass/epoxy composite; the conductivity variation in a carbon fibre composite; and the electrode pixels inside a solid state CMOS fingerprint sensor.

The thesis investigates new approaches to increase the user capacity and improve the error performance of Code Division Multiple Access (CDMA) by employing adaptive interference cancellation and collaborative spreading and space diversity techniques. Collaborative Coding Multiple Access (CCMA) is also investigated as a separate technique and combined with CDMA. The advantages and shortcomings of CDMA and CCMA are analysed and new techniques for both the uplink and downlink are proposed and evaluated. Multiple access interference (MAI) problem in the uplink of CDMA is investigated first. The practical issues of multiuser detection (MUD) techniques are reviewed and a novel blind adaptive approach to interference cancellation (IC) is proposed. It exploits the constant modulus (CM) property of digital signals to blindly suppress interference during the despreading process and obtain amplitude estimation with minimum mean squared error for use in cancellation stages. Two new blind adaptive receiver designs employing successive and parallel interference cancellation architectures using the CM algorithm (CMA) referred to as ‘CMA-SIC' and ‘BA-PIC', respectively, are presented. These techniques have shown to offer near single user performance for large number of users. It is shown to increase the user capacity by approximately two fold compared with conventional IC receivers. The spectral efficiency analysis of the techniques based on output signal-to interference-and-noise ratio (SINR) also shows significant gain in data rate. Furthermore, an effective and low complexity blind adaptive subcarrier combining (BASC) technique using a simple gradient descent based algorithm is proposed for Multicarrier-CDMA. It suppresses MAI without any knowledge of channel amplitudes and allows large number of users compared with equal gain and maximum ratio combining techniques normally used in practice. New user collaborative schemes are proposed and analysed theoretically and by simulations in different channel conditions to achieve spatial diversity for uplink of CCMA and CDMA. First, a simple transmitter diversity and its equivalent user collaborative diversity techniques for CCMA are designed and analysed. Next, a new user collaborative scheme with successive interference cancellation for uplink of CDMA referred to as collaborative SIC (C-SIC) is investigated to reduce MAI and achieve improved diversity. To further improve the performance of C-SIC under high system loading conditions, Collaborative Blind Adaptive SIC (C-BASIC) scheme is proposed. It is shown to minimize the residual MAI, leading to improved user capacity and a more robust system. It is known that collaborative diversity schemes incur loss in throughput due to the need of orthogonal time/frequency slots for relaying source's data. To address this problem, finally a novel near-unity-rate scheme also referred to as bandwidth efficient collaborative diversity (BECD) is proposed and evaluated for CDMA. Under this scheme, pairs of users share a single spreading sequence to exchange and forward their data employing a simple superposition or space-time encoding methods. At the receiver collaborative joint detection is performed to separate each paired users' data. It is shown that the scheme can achieve full diversity gain at no extra bandwidth as inter-user channel SNR becomes high. A novel approach of ‘User Collaboration' is introduced to increase the user capacity of CDMA for both the downlink and uplink. First, collaborative group spreading technique for the downlink of overloaded CDMA system is introduced. It allows the sharing of the same single spreading sequence for more than one user belonging to the same group. This technique is referred to as Collaborative Spreading CDMA downlink (CS-CDMA-DL). In this technique T-user collaborative coding is used for each group to form a composite codeword signal of the users and then a single orthogonal sequence is used for the group. At each user's receiver, decoding of composite codeword is carried out to extract the user's own information while maintaining a high SINR performance. To improve the bit error performance of CS-CDMA-DL in Rayleigh fading conditions, Collaborative Space-time Spreading (C-STS) technique is proposed by combining the collaborative coding multiple access and space-time coding principles. A new scheme for uplink of CDMA using the ‘User Collaboration' approach, referred to as CS-CDMA-UL is presented next. When users' channels are independent (uncorrelated), significantly higher user capacity can be achieved by grouping multiple users to share the same spreading sequence and performing MUD on per group basis followed by a low complexity ML decoding at the receiver. This approach has shown to support much higher number of users than the available sequences while also maintaining the low receiver complexity. For improved performance under highly correlated channel conditions, T-user collaborative coding is also investigated within the CS-CDMA-UL system.

One of the main issues involved in the development of future wireless communication systems is the multiple access technique used to efficiently share the available spectrum among users. In rich multipath environment, spatial dimension can be exploited to meet the increasing number of users and their demands without consuming extra bandwidth and power. Therefore, it is utilized in the multiple-input multiple-output (MIMO) technology to increase the spectral efficiency significantly. However, multiuser MIMO (MU-MIMO) systems are still challenging to be widely adopted in next generation standards. In this thesis, new techniques are proposed to increase the channel and user capacity and improve the error performance of MU-MIMO over Rayleigh fading channel environment. For realistic system design and performance evaluation, channel correlation is considered as one of the main channel impurities due its severe influence on capacity and reliability. Two simple methods called generalized successive coloring technique (GSCT) and generalized iterative coloring technique (GICT) are proposed for accurate generation of correlated Rayleigh fading channels (CRFC). They are designed to overcome the shortcomings of existing methods by avoiding factorization of desired covariance matrix of the Gaussian samples. The superiority of these techniques is demonstrated by extensive simulations of different practical system scenarios. To mitigate the effects of channel correlations, a novel constellation constrained MU-MIMO (CC-MU-MIMO) scheme is proposed using transmit signal design and maximum likelihood joint detection (MLJD) at the receiver. It is designed to maximize the channel capacity and error performance based on principles of maximizing the minimum Euclidean distance (dmin) of composite received signals. Two signal design methods named as unequal power allocation (UPA) and rotation constellation (RC) are utilized to resolve the detection ambiguity caused by correlation. Extensive analysis and simulations demonstrate the effectiveness of considered scheme compared with conventional MU-MIMO. Furthermore, significant gain in SNR is achieved particularly in moderate to high correlations which have direct impact to maintain high user capacity. A new efficient receive antenna selection (RAS) technique referred to as phase difference based selection (PDBS) is proposed for single and multiuser MIMO systems to maximize the capacity over CRFC. It utilizes the received signal constellation to select the subset of antennas with highest (dmin) constellations due to its direct impact on the capacity and BER performance. A low complexity algorithm is designed by employing the Euclidean norm of channel matrix rows with their corresponding phase differences. Capacity analysis and simulation results show that PDBS outperforms norm based selection (NBS) and near to optimal selection (OS) for all correlation and SNR values. This technique provides fast RAS to capture most of the gains promised by multiantenna systems over different channel conditions. Finally, novel group layered MU-MIMO (GL-MU-MIMO) scheme is introduced to exploit the available spectrum for higher user capacity with affordable complexity. It takes the advantages of spatial difference among users and power control at base station to increase the number of users beyond the available number of RF chains. It is achieved by dividing the users into two groups according to their received power, high power group (HPG) and low power group (LPG). Different configurations of low complexity group layered multiuser detection (GL-MUD) and group power allocation ratio (η) are utilized to provide a valuable tradeoff between complexity and overall system performance. Furthermore, RAS diversity is incorporated by using NBS and a new selection algorithm called HPG-PDBS to increase the channel capacity and enhance the error performance. Extensive analysis and simulations demonstrate the superiority of proposed scheme compared with conventional MU-MIMO. By using appropriate value of (η), it shows higher sum rate capacity and substantial increase in the user capacity up to two-fold at target BER and SNR values.

Distributed video coding (DVC) is a new video coding methodology that shifts the highly complex motion search components from the encoder to the decoder, such a video coder would have a great advantage in encoding speed and it is still able to achieve similar rate-distortion performance as the conventional coding solutions. Applications include wireless video sensor networks, mobile video cameras and wireless video surveillance, etc. Although many progresses have been made in DVC over the past ten years, there is still a gap in RD performance between conventional video coding solutions and DVC. The latest development of DVC is still far from standardization and practical use. The key problems remain in the areas such as accurate and efficient side information generation and refinement, quality control between Wyner-Ziv frames and key frames, correlation noise modelling and decoder complexity, etc. Under this context, this thesis proposes solutions to improve the state-of-the-art side information refinement schemes, enable consistent quality control over decoded frames during coding process and implement highly efficient DVC codec. This thesis investigates the impact of reference frames on side information generation and reveals that reference frames have the potential to be better side information than the extensively used interpolated frames. Based on this investigation, we also propose a motion range prediction (MRP) method to exploit reference frames and precisely guide the statistical motion learning process. Extensive simulation results show that choosing reference frames as SI performs competitively, and sometimes even better than interpolated frames. Furthermore, the proposed MRP method is shown to significantly reduce the decoding complexity without degrading any RD performance. To minimize the block artifacts and achieve consistent improvement in both subjective and objective quality of side information, we propose a novel side information synthesis framework working on pixel granularity. We synthesize the SI at pixel level to minimize the block artifacts and adaptively change the correlation noise model according to the new SI. Furthermore, we have fully implemented a state-of-the-art DVC decoder with the proposed framework using serial and parallel processing technologies to identify bottlenecks and areas to further reduce the decoding complexity, which is another major challenge for future practical DVC system deployments. The performance is evaluated based on the latest transform domain DVC codec and compared with different standard codecs. Extensive experimental results show substantial and consistent rate-distortion gains over standard video codecs and significant speedup over serial implementation. In order to bring the state-of-the-art DVC one step closer to practical use, we address the problem of distortion variation introduced by typical rate control algorithms, especially in a variable bit rate environment. Simulation results show that the proposed quality control algorithm is capable to meet user defined target distortion and maintain a rather small variation for sequence with slow motion and performs similar to fixed quantization for fast motion sequence at the cost of some RD performance. Finally, we propose the first implementation of a distributed video encoder on a Texas Instruments TMS320DM6437 digital signal processor. The WZ encoder is efficiently implemented, using rate adaptive low-density-parity-check accumulative (LDPCA) codes, exploiting the hardware features and optimization techniques to improve the overall performance. Implementation results show that the WZ encoder is able to encode at 134M instruction cycles per QCIF frame on a TMS320DM6437 DSP running at 700MHz. This results in encoder speed 29 times faster than non-optimized encoder implementation. We also implemented a highly efficient DVC decoder using both serial and parallel technology based on a PC-HPC (high performance cluster) architecture, where the encoder is running in a general purpose PC and the decoder is running in a multicore HPC. The experimental results show that the parallelized decoder can achieve about 10 times speedup under various bit-rates and GOP sizes compared to the serial implementation and significant RD gains with regards to the state-of-the-art DISCOVER codec.

The future of global telecommunications looks even more promising with the advent of Free Space Optics (FSO) to complement Fibre Optics technology. With the main impairments to Free Space Optics known to be diffraction and atmospheric turbulence, it is critical to adequately characterise the atmospheric medium for effective FSO system design. Most laser sources can be designed to produce Gaussian-like beam profiles, which suffer from diffraction issues. To address this, a non-diffracting beam called the Bessel beam is introduced; its central core has been proven to be resistant to diffractive spreading whilst propagating. However, both Gaussian and Bessel beams will experience distortion when propagating through atmospheric turbulence. The strength of atmospheric turbulence Cn2 is considered constant for ground-to-ground (terrestrial) applications, but proven variable and gradually-weakening for ground-to-space (satellite) applications. In this research, we investigate the propagation of the two beams both in the ground-to-ground scenario and in the ground-to-space scenario. For the ground-to-space scenario, we define a maximum height of 22 km above which the effect of atmospheric turbulence is considered negligible. We also investigate the propagation of the beams from the ground, beyond the 22 km limit, into deep space. We analyse and compare the performance of the beams for all the scenarios based on predefined performance measures. The Bessel beam offers enhanced performance and is shown to outperform the Gaussian on a number of the performance measures.

Leung Cheung-chi. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2001. / Includes bibliographical references (leaves 76-78). / Abstracts in English and Chinese. / Chapter 1 --- INTRODUCTION --- p.1 / Chapter 1.1 --- Background --- p.1 / Chapter 1.2 --- Objective --- p.2 / Chapter 1.3 --- Overview of the Thesis --- p.3 / Chapter 2 --- APPLICATION OF VOIP IN CALL CENTER --- p.6 / Chapter 2.1 --- An Intelligent IP-based Call Center Model --- p.6 / Chapter 2.1.1 --- Major Components --- p.7 / Chapter a) --- VoIP Gateways --- p.7 / Chapter b) --- Automatic Call Distributor (ACD) --- p.8 / Chapter c) --- Operators --- p.8 / Chapter d) --- Monitoring Tool --- p.9 / Chapter 2.1.2 --- Major Functions --- p.9 / Chapter 2.2 --- Experimental Study of an IP-to-IP Call Center - VoIP Application in Education --- p.10 / Chapter 2.2.1 --- Architecture --- p.11 / Chapter 2.2.2 --- Voice Connection Server --- p.12 / Chapter 2.2.3 --- Call Establishment --- p.14 / Chapter 2.2.4 --- A Preliminary Implementation --- p.14 / Chapter 3 --- THE ACD AND ITS SOFTWARE STRUCTURE --- p.17 / Chapter 3.1 --- Three-Layer Software Structure --- p.17 / Chapter 3.1.1 --- Network Infrastructure Layer --- p.18 / Chapter 3.1.2 --- Call Management Layer --- p.18 / Chapter 3.1.3 --- Application Layer --- p.19 / Chapter 3.1.4 --- Interoperation Between Layers --- p.19 / Chapter 3.2 --- Advantages of Adopting this Software Structure --- p.20 / Chapter 3.3 --- Functional Overview of the ACD --- p.21 / Chapter 3.3.1 --- Call Establishment --- p.21 / Chapter 3.3.2 --- Call Waiting --- p.23 / Chapter 3.3.3 --- Call Forwarding --- p.25 / Chapter 3.3.4 --- Routing Mechanism in the ACD --- p.26 / Chapter a) --- "Queues, Operator Groups and Operators" --- p.26 / Chapter b) --- Priority Based Call Routing --- p.28 / Chapter c) --- Routing of New Incoming Calls --- p.29 / Chapter d) --- Assigning Calls in Waiting Queues to Operators --- p.32 / Chapter 4 --- IMPLEMENTATION OF THE ACD --- p.34 / Chapter 4.1 --- Requirements in implementing the ACD --- p.34 / Chapter 4.1.1 --- Asynchronous Method Call --- p.34 / Chapter 4.1.2 --- Transaction Planning --- p.36 / Chapter 4.1.3 --- Failure Handling --- p.37 / Chapter 4.2 --- Available Technologies --- p.38 / Chapter 4.2.1 --- Enterprise JavaBean (EJB) --- p.38 / Chapter a) --- Entity Bean --- p.40 / Chapter b) --- Session Bean --- p.40 / Chapter c) --- Usage of Session Beans and Entity Beans --- p.41 / Chapter 4.2.2 --- COM+ --- p.42 / Chapter 4.2.3 --- EJB vs COM+ --- p.43 / Chapter 4.3 --- Implementation --- p.47 / Chapter 4.3.1 --- Mapping the EJB model to the Implementation of the ACD --- p.47 / Chapter 4.3.2 --- Design of Entity Beans --- p.49 / Chapter 4.3.3 --- Design of Session Beans --- p.51 / Chapter 4.3.4 --- Asynchronous Method Call --- p.53 / Chapter 4.3.5 --- Transaction Planning --- p.55 / Chapter 4.3.6 --- Failure Handling --- p.57 / Chapter a) --- Failure Handling for VoIP gateways --- p.58 / Chapter b) --- Failure Handling in the ACD --- p.60 / Chapter 5 --- AN EXPERIMENT --- p.64 / Chapter 5.1 --- Experiment on the Call Center Prototype --- p.64 / Chapter 5.1.1 --- Setup of the Experiment --- p.64 / Chapter 5.1.2 --- Experimental Results --- p.66 / Chapter a) --- Startup Time for Different Components --- p.66 / Chapter b) --- Possessing Time for Different Requests --- p.67 / Chapter 5.2 --- Observations --- p.69 / Chapter 5.2.1 --- Observations on Experimental Results --- p.69 / Chapter 5.2.2 --- Advantages and Disadvantages of Using EJB --- p.70 / Chapter 6 --- CONCLUSIONS --- p.72 / BIBLIOGRAPHY --- p.76

In recent years, millions of customers all over the world have been subscribing to mobile telephony services which are based on modern digital transmission. At the high transmission bit rates that these systems use, the mobile radio channel exhibits frequency-selective fading characteristics. Under such conditions, the received signal could experience significant intersymbol interference (ISI) from severe distortion to the waveform of the received baseband signal. Therefore, such techniques as adaptive waveform equalisation or adaptive maximum likelihood sequence estimation (MLSE) are used in modern digital mobile radio telephone systems to combat this undesirable ISI. These adaptive schemes have almost always been used in conjunction with coherent demodulation in the receivers.This study examines the application of noncoherent demodulation, in the form of frequency discriminator detection, as an alternative to coherent demodulation. The GSM (Global System for Mobile Communications) standard has been used as the basis for this investigation. It has been shown analytically that in the presence of frequency-selective fading, a propagation environment common to the GSM system, the use of frequency discriminator detection gives rise to nonlinear ISI in the demodulated signal. It has also been shown that frequency-selective fading could cause large unwanted "spikes" to appear in the demodulated signal, thus leading to a severe degradation in the bit-error-rate (BER) performance. Consequently, several waveform distortion cancellation schemes for combatting the nonlinear ISI have been formulated. The BER performances of these proposed schemes, under various propagation conditions, have been studied by computer simulation.Furthermore, it has been observed that the undesirable "spikes", that occur in the demodulated signal due to frequency-selective fading, could be ++ / suppressed by the use of inverse-limiting in conjunction with frequency discriminator detection. As a result, an effective adaptive detection scheme has been formulated, based on modelling the combination of the GMSK modulator, the mobile channel, the frequency discriminator, and any transmit and receive fitters, as a finite-state machine. The transmitted data is then detected using an MLSE. The BER performance of this proposed adaptive detection scheme has been extensively investigated by computer simulation. This has been carried out assuming various propagation conditions, including the two-ray fading channel model with equal path powers and relative delays of up to four bit periods, the maximum relative delay considered in the GSM system. Also, the effectiveness of the proposed adaptive detection scheme in combatting IS] has been investigated by computer simulation based on the six-ray GSM empirical propagation models for typical urban (TU), hilly terrain (HT) and rural area (RA) environments. The computer simulated results confirm that the voice grade performance required for the GSM system could be achieved by the proposed adaptive detection scheme in all the recommended GSM propagation models considered. Furthermore, the BER performance of the receiver remains unaffected by a carrier frequency offset of up to 2 kHz.

<p>Voice over IP technology (VoIP) has recently gained popularity among consumers. Many popular VoIP services exist only as software for PCs. The need of taking such services out of the PC, into a stand-alone device has been discovered, and this thesis work deals with the development of such a device. The thesis work is done for Häger Scandinavia AB, a Swedish telephone manufacturer. This thesis work covers the design of a complete prototype of a table-top VoIP telephone running an embedded Linux Operating system. Design areas include product development, hardware design and software design.The result is a working prototype with hardware and corresponding Linux device drivers. The prototype can host a Linux application adapted to it. Conclusions are that the first hardware version has worked well and that using an open-source operating system is very useful. Further work consists of implementing a complete telephony software application in the system, evaluation of system requirements and adapting the prototype for a commercial design.</p>

<p>Telefonrådgivning via en sjuksköterska har funnits ända sedan 1930-talet men det är först de senaste åren som denna kompetens har erkänts som en egen specialitet. Inom vården sker allt fler möten mellan vårdgivare och vårdsökande via telefonen vilket leder till att allt högre krav ställs på sjuksköterskans yrkeskunnighet. Studiens syfte var att beskriva vad sjuksköterskor upplever vid telefonrådgivning till personer med psykiska problem. Metoden som använts i denna litteraturstudie var kvalitativ innehållsanalys med manifest och latent ansats. Resultatet baseras på 10 vetenskapliga artiklar där fyra kategorier framkom: trygghet, utsatthet, ett känsligt öra samt meningsfullhet. I resultatet framkom att kunskap och klinisk erfarenhet är faktorer som har stor betydelse för yrkesrollen inom telefonrådgivning. Som rådgivningssjuksköterska inom psykiatrisk vård upplevde många en hög stressnivå vad gäller den komplexa yrkesrollen. Att inte se patienten med sina egna ögon upplevdes som att delvis arbeta i blindo. Detta ledde till att rådgivningssjuksköterskorna lyssnade aktivt för att försöka förstå vad som uttrycktes mellan raderna i samtalet med den vårdsökande. Ett etiskt dilemma som rådgivningssjuksköterskor beskriver är upplevelse av svårigheter att hantera andrahandsinformation vid telefonrådgivning. Dessa svårigheter uppkommer då rådgivningssjuksköterskan ej har möjlighet att tala direkt till den person som telefonråden gäller. För övrigt gavs uttryck för att arbetet som rådgivningssjuksköterska är stimulerande och utmanande</p>

<p> </p><p>The aim of this thesis is to find out the attitudes towards using the mobile telephone as a communication tool within the District Education Offices (DEO) in Iringa in Tanzania. This in order to collect an understanding of the use of Information and Communication Technology (ICT) in a work place in a developing country and to develop an understanding of how and why ICT is used.</p><p> </p><p>The theoretical framework which is the base for this study is made up by several theories within the field of development communication. Diffusion of innovations, ICT for developing countries, Technology Acceptance Model and a framework for sustainable ICT are the theories from which factors affecting attitudes have been collected to make up the theoretical framework for this study.</p><p> </p><p>The study was conducted with qualitative interviews with the employees of two out of seven DEOs in Iringa in Tanzania. The interview data was also complemented with data from questionnaires, filled out by employees at the DEOs. The questions in interview guide and the questionnaire were formed with the theoretical framework as the starting point and later also analysed through the same theoretical framework. Four individual interviews and one group interview with five interviewees were conducted and twelve questionnaires were collected.</p><p> </p><p>The compiled data showed that private usage, perceived ease of use, costs of usage, possible side effects of usage and the compatibility of the technology are the factors influencing the attitudes towards using the mobile phone at work and it can be concluded that the employees of the two District Education Offices in general had a positive attitude towards using the mobile phone, but that the technology and the work situation not yet were perfect for them to use the mobile phone without any restrains in their work. </p>