The object of the work reported in this thesis is to construct and examine the performance of a direct-sequence spread-spectrum system and to investigate the feasibility of introducing the technique into the land mobile radio services, by means of bandsharing with the existing television services, as a means of improving spectrum utilisation. For comparative purposes, conventional modulation systems are also considered to be contenders for the scheme. Chapter 1 summarises the concepts of spread-spectrum systems. It introduces the modulation techniques that fall under this heading and describes the fundamental parameters associated with their operation. Chapter 2 indicates the reason behind the need to investigate spread-spectrum systems for adoption into the land mobile radio service and gives a general survey of the work previously carried out towards realising this aim. Chapter 3 examines the choice of initial design parameters and base modulation technique for an experimental direct-sequence system and describes the component sections used. Performance figures are presented for operation under both Gaussian and 'other-user' interference. Chapter 4 is devoted to the synchronisation aspect of a direct-sequence spread-spectrum system. It gives a survey of the techniques available for obtaining initial synchronisation and experimentally compares the operation of various delay lock loop implementations with respect to their ability to maintain accurate synchronisation. Chapter 5 covers the subjective assessment of television picture quality, as dictated by untrained observers, when subjected to various forms of modulation interference. Chapter 6 presents a feasibility study into the use of direct-sequence spread-spectrum or conventional modulation methods as possible contenders for a mobile service operating on a bandsharing basis with the television broadcast services in London and the South-East. Chapter 7 gives a final summary of all the conclusions.

This thesis presents the development of a new approach to EHV transmission line protection based on fundamental principles. The method is based on a travelling wave approach and utilises analysis of spectral information derived for the transient phenomena caused by fault inception. By establishing practical means to utilise a frequency bandwidth of about 70-400kHz, a new protective-scheme could be developed to operate on detecting HF current components due to travelling waves set up by the inception of a fault. Thus, the scheme operates at ultra-high-speed and can detect a fault within a fraction of a millisecond. It possesses its essential discriminative feature by virtue of applying line wave traps (LWTs) installed at each end of the protected line. The locations of these traps define the zone of protection that is formed due to the combined effect of both the trap and the station bus-bar capacitance which in turn provides correct termination of the transmission line when over the requisite frequency-bandwidth. Thus, in-band current components will ideally not be able to pass through LWTs when these are set up due to external faults. The performance of the scheme was successfully demonstrated by digitally simulating many different test conditions. It is a non-unit scheme, that does not require communication channels, but it nevertheless possesses the features of unit schemes. It can cater for the protection of whole length of line when faults occur near voltage-maximum point-on-wave and also covers relatively high resistive fault conditions. The scheme may also provide a solution to lonstanding problems encountered in the protection of series-compensated transmission lines. In this respect, further investigations may now be performed in order to examine the possibility of achieving this goal.

The main theme of this thesis is the control of acoustic echoes for modem voice communication systems by means of echo cancellation. Two important issues in acoustic echo cancellation, namely the efficient adaptation of the echo cancellation filter and the reliable adaptation of the echo cancellation filter in double talk environment, are investigated. The delayless subband adaptive filter architecture is studied. Efficient implementation of the analysis filter bank and the time domain filtering are derived. The transforming of the subband filter weights to a fullhand counterpart is examined. It is shown that the weight transform is a synthesis filtering procedure. Two new weight transform schemes that deliver substantial performance improvements are proposed. The open-loop optimal subband filter impulse responses are shown to be non-causal and several anti-causal laps in the subband filters are required to model this non-causality. Because of the inevitable double talk detection errors, adaptive filtering algorithms with built-in double talk robustness measures are needed for the reliable operation of the echo canceller. The basic idea of robust adaptive filtering is examined. A comparison of different existing time domain robust adaptive filtering algorithms demonstrates that excellent trade-off between the convergence and the tracking properties and the double talk robustness of the adaptive filtering algorithm can he achieved by using Huher’s method for both the update of the echo cancellation filter and the estimation of scale. A delayless closed-loop robust sub- hand adaptive filter is proposed. / By independently adapting the scale estimates and normalizing the adaptation in each subband, significant improvement in terms of the convergence and tracking speed over the time domain robust NLMS algorithm can be obtained without sacrificing the double talk robustness. Moreover, it is demonstrated that by using different thresholds in the update of the echo cancellation filter and the scales, the robust algorithms converge and track echo path variation as fast as their non-robust counter part while still maintaining a sufficiently low sensitivity to double talk detection errors. The application of two path adaptive filters to acoustic echo cancellation is examined. An analysis of the original two path adaptive filtering algorithm shows that it suffers from two kinds of performance degradation due to the divergence of the background filter during double talk, namely the slow tracking of echo path variation and the false filter coefficient copying after double talk. A robust two path adaptive filter is proposed to mitigate these problems.

Master's thesis. / Format: PDF. Bibl.

This dissertation assesses the external environment of the Telecommunications market in South Africa. It accomplishes this goal by explaining the political economy, market structure, key driving forces and conducting an Industry analysis. The study considers the literature on regulation and liberalisation, convergence and the various techniques of industry and competitive analysis. A case study on the historic and current Telecommunications environment is presented. The analytical framework developed for this study adopted from Naidoo (2002), consists of five components: the external environment, remote environment, industry environment, operating environment, key driving forces and an industry and competitive analysis. It also includes a background overview and recommendations. Data for the study was largely collected using publications and journals, which have been published in electronic format on various Internet hosted databases including: verbatim submissions to the White Paper on telecoms policy (Government Gazette no. 16995 Notice No 291 of 13 March 1996), submissions on the licensing of the Second Network Operator, participant-observation, government documents, statistical databases, published literature and unpublished papers. The perspective of the study is to learn and apply the knowledge elsewhere. The main purpose of this study is to describe the industry for the various Telecommunications operators (fixed, cellular and VANS) in the South African market and to conduct a strategic analysis. Another objective is to provide the researcher with further knowledge of this particular industry and to ascertain from a business environment viewpoint whether to invest in the telecommunication sector in South Africa. / Thesis (MBA)-University of Natal, 2003.

The choice of the modulation format is the principle factor in realising a highperformance bandwidth efficient communication system at an acceptable cost and complexity. Pulse time modulation technique represents an attractive alternative to purely digital or purely analogue modulation schemes which has received considerable attention over the years. But very little work has been reported on pulse slope modulation. In this work a pulse slope modulation introduced and full wavefonn characteristics is given. A simple novel receiver has been proposed, which is based on converting the PSM waveform into a PAM waveform by sampling the received PSM signal at the rise time interval. This design eliminates the use of a differentiator and a voltage slicer adopted in the classical demodulation technique, and it offers simplicity and improved noise performance. A new signal-t-noise formula has been presented for the first time thus enabling users to predict the system noise performance. Experimental results have shown excellent agreement to within + 1 dB with theoretical predictions using the new formula. Results obtained show the potential of PSM in tenns of simplicity and better noise perfonnance compared to its counter part pulse amplitude modulation. Although PTM has many advantages over analogue and digital schemes when employed as a single channel system, it becomes a challenge when multiplexing is involved. Isochronous PTM schemes are suitable while the anisochronous schemes are not. One solution to overcome this problem is to adopt a hybrid or compound modulation technique, where both isochronous and anisochronous schemes can be combined. Compound frequency and width modulation (CPFWM) is one such a scheme which offers bandwidth efficiency, simplicity and low cost over the more commonly used multiplexed techniques. Detailed investigation of CPFWM has been given and in an expression for its spectrum has been developed. The results obtained have been clarified practically and by means of computer simulation to within + 1 dB. In CPFWM, cross talk in the PFM and pWM channels are due to width modulation and frequency modulation, respectively. The main cause of cross talk is the carryover energy from one edge to the next edge. This effect is studied and he results obtained for the cross talk are compared with the predicted data showing an agreement to within + 1 dB. Results were also compared with time division multiplexed pulse position modulation, showing comparable performance and under certain conditions CPFWM offering better performance. Noise performance of PFM and PWM channels has been theoretically and experimentally investigated. It has been shown that PFM is superior to PWM over a wide range of modulation indices and pulse characteristics. When both channels are identical in bandwidth and modulation conditions, PFM offers a 7 dB improvement compared to PWM channel. The noise perfonnance of CPFWM system is also compared with digital and analogue modulation schemes. For 25 dB CNR CPFWM offers a 40 dB SNR which is 20 dB higher than amplitude modulation, 1 7 dB lower than PCM system. At CNR > 35 dB its performance approaches that of PCM system. Thus showing the potential of the scheme.

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