Masters Degrees (Electronic Engineering)

Permanent URI for this collectionhttps://hdl.handle.net/10413/6868

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An intelligent multi-terminal interface.
(1987) Peplow, Roger Charles Samuel.; Nattrass, Henry Lee.
The document describes the development of a micro-processor based terminal multiplexer to connect four terminals to a standard Hewlett Packard series 1000 mini-computer. The project was required to fulfill the dual roll of both increasing the number of terminals that the HPI000 could support and of reducing the peripheral load on the host CPU. The final product occupied a standard 200mm square HP size interface card and used an 8085 micro-processor and several 8085 family peripheral chips to provide four full duplex serial channels and a high speed data link with the host. A multi-tasking executive was written to control the multiplexer software which was finally implemented as 15 independent tasks occupying 8 kilo-bytes of eprom. The software was written to perform all terminal interaction and editing in order to reduce the host CPU involvement to a single interrupt per record. The resultant interface proved capable of handling an aggregate throughput in excess of 4000 characters per second which was sufficient to cope with all four terminals running at 9600 bits per second, even when all four were transferring in burst mode. The interface also proved to be between five and eighteen times less demanding on the host than the two standard Hewlett Packard interfaces then available. When compared to the low cost HP12531 interface, the multiplexer increased the 9600b/s terminal handling capability of the host from 3 terminals to 52.
A comparative study of various speech recognition techniques.
(1990) Pitchers, Richard Charles.; Broadhurst, Anthony D.
Speech recognition systems fall into four categories, depending on whether they are speaker-dependent or independent of speaker population and on whether they are capable of recognizing continuous speech or only isolated words. A study was made of most methods used in speech recognition to date. Four speech recognition techniques for speaker-dependent isolated word applications were then implemented in software on an IBM PC with a minimum of interfacing hardware. These techniques made use of short-time energy and zero-crossing rates, autocorrelation coefficients, linear predictor coefficients and cepstral coefficients. A comparison of their relative performances was made using four test vocabularies that were 10, 30, 60 and 120 words in size. These consisted of 10 digits, 30 and 60 computer terms and lastly 120 airline reservation terms. The performance of any speech recognition system is affected by a number of parameters. The effects of frame length, pre-emphasis, window functions, dynamic time warping and the filter order were also studied experimentally.
A packet radio system for an industrial data network.
(1992) Sewnath, Gajadhar.; Levy, David C.
This project was undertaken for a commercial electronics company, CONTROL LOGIC (CONLOG) which is involved in the research, design, development and manufacture of data acquisition, control, energy management and automotive equipment. Currently CONLOG uses an inhouse token passing local area network CONET for industrial data communications.The need had arisen to provide a means of data communication amongst widely geographically distributed remote terminal units (RTUs) generating demands at a very low duty cycle. A need for communications between RTUs and a centralised controller was also required. In addition to this, multihop communications between the RTUs was required. Packet switching using a broadcast radio network provides an efficient means of achieving this. An investigation into to the various media access control protocols and contention techniques using packet radio was carried out. The various media access techniques were compared with respect to throughput and normalised delay. This led to the selection of a media access scheme for the packet radio network using RTUs. A protocol specification control is centralised or Interconnect Organisation.The switching protocol (OSI) for the packet radio network, in which distributed, was done. The architechure of specified adheres to the Open Systems model of the International Standards.An experimental packet switching radio network was implemented using the protocol specification defined above. The packet radio network (PACNET) uses existing off the shelf radios and purpose built hardware for the remote terminal units. The thesis describes methods of data communications suitable for widely dispersed industrial data communications, the selection of the packet switching media access methods and control protocols, and the design and implementation of the prototype system.
A reconfigurable distributed process control environment for a network of PC's using Ada and NetBIOS.
(1992) Randelhoff, Mark Charles.; Levy, David C.
No abstract
Towards automatic face recognition using discrete cosine transforms and neural networks.
(1998) Debipersad, Sanjeev Chundurduth.; Broadhurst, Anthony D.
Abstract available in PDF.
The design and construction of an experimental MgO cold cathode X-ray tube for use in XRF spectrometry.
(2000) Damjanovic, Daniel.; Thebock, P. A.; Broadhurst, Anthony D.
An introduction to the fundamental concepts of X-ray physics and X-ray tube design is given. This discussion also includes a brief description of various X-ray tube types available commercially for a number of different industrial applications. The design of a high-energy MgO cold cathode X-ray tube, which is to be used in an X-ray fluorescence (XRF) spectrometer, is described in detail with emphasis placed on the electron beam focusing mechanism and the theory of operation as well as the construction of the X-ray tube MgO cold cathode, which functioned as the electron emitter of the device. A detailed account is also given of the output characteristics of the X-ray tube power supply, which has a direct effect on the design requirements and consequently the performance of the X-ray tube. An investigation into the manufacture of the vacuum envelope with particular attention focused on the production of reliable metal-to-ceramic seals was performed. A number of tests were conducted especially with regard to the maximum temperature that such seals may withstand without becoming permanently damaged. These tests were essential, since high temperature gradients tend to develop in an X-ray tube during operation, which the metal-to-cerarnic seals of the tube must be capable of withstanding if damage to the device is to be avoided. The set-up of the XRF spectrometer in which the completed X-ray tube was tested is discussed, in which the X-ray current and voltage measuring techniques are described. Furthermore a detailed account of the operation of the X-ray detector system and the multichannel analyser is given, which was used to detect and record spectra of the sample elements excited by the primary radiation of the X-ray tube. Finally the measured X-ray tube performance characteristics are discussed and compared to the predicted results.
Parallel implementation of fractal image compression
(2000) Uys, Ryan F.; Prentice, J.; Broadhurst, Anthony D.
Fractal image compression exploits the piecewise self-similarity present in real images as a form of information redundancy that can be eliminated to achieve compression. This theory based on Partitioned Iterated Function Systems is presented. As an alternative to the established JPEG, it provides a similar compression-ratio to fidelity trade-off. Fractal techniques promise faster decoding and potentially higher fidelity, but the computationally intensive compression process has prevented commercial acceptance. This thesis presents an algorithm mapping the problem onto a parallel processor architecture, with the goal of reducing the encoding time. The experimental work involved implementation of this approach on the Texas Instruments TMS320C80 parallel processor system. Results indicate that the fractal compression process is unusually well suited to parallelism with speed gains approximately linearly related to the number of processors used. Parallel processing issues such as coherency, management and interfacing are discussed. The code designed incorporates pipelining and parallelism on all conceptual and practical levels ensuring that all resources are fully utilised, achieving close to optimal efficiency. The computational intensity was reduced by several means, including conventional classification of image sub-blocks by content with comparisons across class boundaries prohibited. A faster approach adopted was to perform estimate comparisons between blocks based on pixel value variance, identifying candidates for more time-consuming, accurate RMS inter-block comparisons. These techniques, combined with the parallelism, allow compression of 512x512 pixel x 8 bit images in under 20 seconds, while maintaining a 30dB PSNR. This is up to an order of magnitude faster than reported for conventional sequential processor implementations. Fractal based compression of colour images and video sequences is also considered. The work confirms the potential of fractal compression techniques, and demonstrates that a parallel implementation is appropriate for addressing the compression time problem. The processor system used in these investigations is faster than currently available PC platforms, but the relevance lies in the anticipation that future generations of affordable processors will exceed its performance. The advantages of fractal image compression may then be accessible to the average computer user, leading to commercial acceptance.
Design considerations and implementation of a RF front-end for CDMA adaptive array system.
(2000) Roopram, Kelesh D.; McDonald, Stephen A.
Recent studies have shown that considerable system capacity gains in mobile communication systems can be obtained by exploiting the use of antenna arrays at the base station. Unfortunately, these studies make little mention of practical issues concerning implementation. It is thus one of the objectives of the Centre of Excellence (CoE) in Radio Access Technologies at the University of Natal to investigate the development of a widehand CDMA adaptive array transceiver using Alcatel software radios as the transceiver platforms. Such a transceiver system can be subdivided into three major sections: RF front-end, signal digitization and baseband processing stages. Due to the enormity of such an undertaking, the research outlined in this thesis is focused on (but not isolated to) some aspects of the RF front-end implementation for the proposed system. The work in this thesis can be catergorized into two sections. The first section focuses on the theoretical and practical (or implementation) aspects of antenna arrays and beamforming. In particular, it is evident that digital (rather than analogue) beamforming in a multi user environment, is a more viable option from both a cost and implementation standpoint. The second section evaluates the impact of RF component noise and local oscillator generated phase noise in a DS-CDMA system. The implementation of a RP front-end for a BPSK transceiver also forms part of the work in this section. LO phase noise and Error Vector Magnitude (EVM) measurements are performed on this system to support relevant theory. By use of the HP89410A phase noise measurement utility and the phase noise theory developed in this thesis, a quantitative phase noise comparison between two frequency sources used in the system were made. EVM measurement results conclusively verified the importance of an LNA in the system. It has also been shown that the DS-CDMA simulated system exhibits superior performance to the implemented BPSK system. Furthermore, an EVM troubleshooting methodology is introduced to identify possible impairments within the BPSK receiver RF front-end. However, this thesis was written with the intention of bridging the gap between the theoretical and practical implementation aspects of RP wireless communication systems. It is the author's opinion that this has been achieved to a certain extent.
Multiuser demodulation for DS-CDMA systems in fading channels.
(2000) Singh, Navin Runjit.; Takawira, Fambirai.
The problems of optimal as well as suboptimal detection for CDMA transmissions over an additive white Gaussian noise (AWGN) channel, have been the focus of study in the recent past. However, CDMA transmissions are frequently made over channels which exhibit fading and/or dispersion; hence receivers need to be designed which take into account this behaviour. In spite of the major research effort invested in multiuser demodulation techniques, several practical as well as theoretical open problems still exist. Some of them are considered in more detail in this thesis. The aim of the thesis is to develop multiuser demodulation algorithms for mobile communication systems in frequency-selective fading channels, and to analyze their implementation complexity. The emphasis is restricted to the uplink of an asynchronous DS-CDMA system where the users transmit in an uncoordinated manner and are received by one centralized receiver. The original work that is undertaken for the MScEng study is the evaluation of a multiuser receiver structure for a frequency-selective fading channel, where there exists a steady specular path and two fading paths. Furthermore, the effect of using selection diversity is investigated by examining the bit error rate, asymptotic multi user efficiency and near-far resistance of the proposed detector structure. These results are confirmed both analytically and by simulation in the thesis. An investigation is also conducted into the application of neural networks to the problem of multiuser detection in code division multiple access systems. The neural network will be used as a classifier in an adaptive receiver which incorporates an extended Kalman filter for joint amplitude and delay estimation. Finally, some open problems for future research will be pointed out in the thesis. Keywords: AWGN channel , DS-CDMA system, frequency-selective, multi user demodulation, asymptotic multiuser efficiency, near-far resistance, neural network, Kalman filter.
The implementation of a CDMA system on a FPGA-based software radio.
(2000) Ellis, Timothy.; Peplow, Roger Charles Samuel.
This dissertation exammes two of the rlsing technologies in the world of wireless, cellular communications - CDMA and the software radio. This thesis covers many of the issues related to these two emerging field s of wireless communications, establish ing a theoretical framework for the broader issues of implementation. To this end, the thesis covers many of the basic issues of spread spectrum communications, in addition to establishing the need for, and defining the role of, the software radio. Amalgamation of these two key areas of interest is embellished in a presentation of many of the concerns of implementing a specific CDMA system on a particular type of software radio - the Alcatel Altech Telecomms Flexible Radio Platform. Of primary concern in the research methodology embraced in this thesis is the mastering of a variety of analysis and implementation tools. Once the theoretical background has been substantiated by current expositions, the thesis launches along a highly deterministic route. First, the research issues are tested in a mathematical environment for suitability to the given task. Second, an analysis of the appropriateness of the technique for the software radio environment is undertaken, culminating in the attempted deployment within the hardware environmenl. Rigorous testing of the input/output mapping characteristics of the hardware instantiations created in this manner complements the research methodology with a viability study. This procedure is repeated with many elements of the CDMA system design as they are examined, simu lated, deployed and tested.
Lattice-structure based adaptive MMSE detectors for DS-CDMA systems.
(2001) Thakadu, Batlhowahela C. D.; Takawira, Fambirai.
There has been significant interest in the research community on detectors for DS-CDMA systems. The conventional detector, which detects users ' data bits, by using a filter matched to the users' spreading codes, has two major drawbacks. These drawbacks are (1) its capacity is limited by multiple access interference (MAl) and (2) it suffers from the near-far problem. The remedy to these problems is to use a multiuser detector, which exploits knowledge of users ' transmission and channel parameters to mitigate MAl. Such detectors are called multi user detectors (MUD). A number of these detectors have been proposed in the literature. The first such detector is the optimal detector proposed by Verdu. Following Verdu's work a number of suboptimal detector were proposed. These detectors offer better computational complexity at the expense of the bit error rate performance. Examples of these detectors are the decorrelating detector, the minimum mean squared error detector (MMSE), the successive interference cancellation and parallel interference cancellation. In this thesis, we consider the adaptive DS-CDMA MMSE detector, where lattice-based filter algorithms are employed to suppress MAl. Most of the work in the literature has considered the implementation of this detector using the Least Mean Square (LMS) algorithm. The disadvantage of using the LMS algorithm to implement the MMSE detector is that the LMS algorithm converges very slowly. The main aims of this thesis are as follows. A review of the literature on MUD is presented. A lattice based MUD is then proposed and its performance evaluated using both simulation and analytical methods. The results obtained are compared with those of the LMSMMSE detector. From the results obtained the adaptive Lattice-MMSE detector is shown to offer good performance tradeoff between convergence results and BER results.
Performance of the transmission control protocol (TCP) over wireless with quality of service.
(2001) Walingo, Tom Mmbasu.; Takawira, Fambirai.
The Transmission Control Protocol (TCP) is the most widely used transport protocol in the Internet. TCP is a reliable transport protocol that is tuned to perform well in wired networks where packet losses are mainly due to congestion. Wireless channels are characterized by losses due to transmission errors and handoffs. TCP interprets these losses as congestion and invokes congestion control mechanisms resulting in degradation of performance. TCP is usually layered over the Internet protocol (lP) at the network layer. JP is not reliable and does not provide for any Quality of Service (QoS). The Internet Engineering Task Force (IETF) has provided two techniques for providing QoS in the Internet. These include Integrated Services (lntServ) and Differentiated Services (DiffServ). IntServ provides flow based quality of service and thus it is not scalable on connections with large flows. DiffServ has grown in popularity since it is scalable. A packet in a DiffServ domain is classified into a class of service according to its contract profile and treated differently by its class. To provide end-to-end QoS there is a strong interaction between the transport protocol and the network protocol. In this dissertation we consider the performance of the TCP over a wireless channel. We study whether the current TCP protocols can deliver the desired quality of service faced with the challenges they have on wireless channel. The dissertation discusses the methods of providing for QoS in the Internet. We derive an analytical model for TCP protocol. It is extended to cater for the wireless channel and then further differentiated services. The model is shown to be accurate when compared to simulation. We then conclude by deducing to what degree you can provide the desired QoS with TCP on a wireless channel.
Implementation of a WCDMA AAA receiver on an FPGA based software radio platform.
(2001) Kora, Saju P.; McDonald, Stephen A.
WCDMA promises to achieve high-speed internet, high quality image transmission and high-speed data services with larger system capacity. However, Multiple Access Interference is one of the major causes of transmission impairment, which reduces the link capacity in WCDMA systems. The Adaptive Antenna Array (AAA) technique reduces multiple access interference by directing antenna beam nulls towards the interfering signals by weighting the received signals from all antennas before combining the signals. With the very rapid advancement of wireless personal communications services, a new challenge to the cellular industry is the integration of multiple systems and applications on a single device. A software radio technique offers a possible solution to achieve this goal including international roaming and multiple standard operations within the same geographical area. The main attraction of a software radio is it's flexibility, in that it can be programmed for emerging cellular standards allowing it to be updated with new software without any changes in the hardware infrastructure. A software radio incorporating adaptive array beamforming at the receiver can increase the total carried traffic in a system and transmit power while the probability of call blocking and forced termination can also be decreased. This dissertation examines WCDMA, AAA, power control and software radio techniques in the world of wireless communication systems. Once the theoretical background of CDMA and AAA has been substantiated, the thesis establishes the need for power control in mobile systems by examining simulation results. An AAA receiver with six antenna elements is proposed and evaluated in different environments as a precursor to implementation. It can be inferred that when the link is interference limited, the link capacity can be increased and it has been shown that the AAA receiver with six antenna elements increases the link capacity to about 2.9 times that of the single antenna RAKE receiver. This thesis also examines the basic concepts of VHDL and considers this as the principle means to program reconfigurable core FPGA's in the software radio. A three-layered (PC/DSP/FPGA) software radio test bed is used to implement an AAA receiver. The architecture of the test bed is designed in such a way that it can be used to evaluate the performance of various FPGA based transceivers and coding schemes etc. Many of the desirable features and flexibilities inherent in the software radio concept are available on this test bed and the system has proved to be capable of high speed digital processing and is ideally suited to the development of time critical system components. The bit error rate achieved using the implemented receiver is assessed and compared to simulation results in an environment incorporating Rayleigh fading and AWGN.
Performance of turbo-coded DS-CDMA systems in fading and burst channels.
(2001) Nkouatchah, Telex Magloire Ngatched.; Takawira, Fambirai.
Turbo codes are a class of forward error correction (FEC) codes that offer energy efficiencies close to the limits predicted by information theory. The features of turbo codes include parallel code concatenation, recursive convolutional encoding, nonuniform interleaving, and an associated iterative decoding algorithm. The excellent performance of turbo codes explains why much of the current research is focused on applying turbo codes to different systems. This dissertation first outlines a new simple criterion for stopping the iterative process of the turbo decoder for each individual frame immediately after the bits are correctly estimated and thus prevents unnecessary computations and decoding delay. The dissertation then considers the performance of turbo coded DS-CDMA systems. The performance analysis begins with simulation results for turbo coded DS-CDMA over a multi-path Rayleigh fading channel. The channel is then modeled using the Gilbert-Elliott channel model and analytical expressions for the performance of the system are derived. The influence of various parameters such as the Doppler frequency, the signal-to-noise ratio threshold on the system performance are analyzed and investigated.
Concatenated space-time codes in Rayleigh fading channels.
(2002) Byers, Geoffrey James.; Takawira, Fambirai.
The rapid growth of wireless subscribers and services as well as the increased use of internet services, suggest that wireless internet access will increase rapidly over the next few years. This will require the provision of high data rate wireless communication services. However the problem of a limited and expensive radio spectrum coupled with the problem of the wireless fading channel makes it difficult to provide these services. For these reasons, the research area of high data rate, bandwidth efficient and reliable wireless communications is currently receiving much attention. Concatenated codes are a class of forward error correction codes which consist of two or more constituent codes. These codes achieve reliable communications very close to the Shannon limit provided that sufficient diversity, such as temporal or spatial diversity, is available. Space-time trellis codes (STTCs) merge channel coding and transmit antenna diversity to improve system capacity and performance. The main focus of this dissertation is on STTCs and concatenated STTCs in quasi-static and rapid Rayleigh fading channels. Analytical bounds are useful in determining the behaviour of a code at high SNRs where it becomes difficult to generate simulation results. A novel method is proposed to analyse the performance of STTCs and the accuracy of this analysis is compared to simulation results where it is shown to closely approximate system performance. The field of concatenated STTCs has already received much attention and has shown improved performance over conventional STTCs. It was recently shown that double concatenated convolutional codes in AWGN channels outperform simple concatenated codes. Motivated by this, two double concatenated STTC structures are proposed and their performance is compared to that of a simple concatenated STTCs. It is shown that double concatenated STTCs outperform simple concatenated STTCs in rapid Rayleigh fading channels. An analytical model for this system in rapid fading is developed which combines the proposed analytical method for STTCs with existing analytical techniques for concatenated convolutional codes. The final part of this dissertation considers a direct-sequencejslow-frequency-hopped (DSj SFH) code division multiple access (CDMA) system with turbo coding and multiple transmit antennas. The system model is modified to include a more realistic, time correlated Rayleigh fading channel and the use of side information is incorporated to improve the performance of the turbo decoder. Simulation results are presented for this system and it is shown that the use of transmit antenna diversity and side information can be used to improve system performance.
MMSE equalizers and precoders in turbo equalization.
(2003) Gaffar, Mohammed Yusuf Abdul.; Xu, Hongjun.
Transmission of digital information through a wireless channel with resolvable multipaths or a bandwidth limited channel results in intersymbol interference (1SI) among a number of adjacent symbols. The design of an equalizer is thus important to combat the ISI problem for these types of channels and hence provides reliable communication. Channel coding is used to provide reliable data transmission by adding controlled redundancy to the data. Turbo equalization (TE) is the joint design of channel coding and equalization to approach the achievable uniform input information rate of an ISI channel. The main focus of this dissertation is to investigate the different TE techniques used for a static frequency selective additive white Gaussian noise (AWGN) channel. The extrinsic information transfer (EXIT) chart is used to analyse the iterative equalization/decoding process and to determine the minimum signal to noise ratio (SNR) in order to achieve convergence. The use of the Minimum Mean Square Error (MMSE) Linear Equalizer (LE) using a priori information has been shown to achieve the same performance compared with the optimal trellis based Maximum A Posterior (MAP) equalizer for long block lengths. Motivated by improving the performance of the MMSE LE, two equalization schemes are initially proposed: the MMSE Linear Equalizer with Extrinsic information Feedback (LE-EF (1) and (U)). A general structure for the MMSE LE, MMSE Decision Feedback Equalizer (DFE) and two MMSE LE-EF receivers, using a priori information is also presented. The EXIT chart is used to analyse the two proposed equalizers and their characteristics are compared to the existing MAP equalizer, MMSE LE and MMSE DFE. It is shown that the proposed MMSE LE-EF (1) does have an improved performance compared with the existing MMSE LE and approaches the MMSE Linear Equalizer with Perfect Extrinsic information Feedback (LE-PEF) only after a large number of iterations. For this reason the MMSE LE-EF is shown to suffer from the error propagation problem during the early iterations. A novel way to reduce the error propagation problem is proposed to further improve the performance of the MMSE LE-EF (I). The MAP equalizer was shown to offer a much improved performance over the MMSE equalizers, especially during the initial iterations. Motivated by using the good quality of the MAP equalizer during the early iterations and the hybrid MAP/MMSE LE-EF (l) is proposed in order to suppress the error propagation problem inherent in the MMSE LE-EF (I). The EXIT chart analysis reveals that the hybrid MAP/MMSE LE-EF (l) requires fewer iterations in order to achieve convergence relative to the MMSE LE-EF (l). Simulation results demonstrate that the hybrid MAP/MMSE LE-EF (I) has a superior performance compared to the MMSE LE-EF (I) as well as approaches the performance of both the MAP equalizer and MMSE LE-PEF at high SNRs, at the cost of increased complexity relative to the MMSE LEEF (I) receiver. The final part of this dissertation considers the use of precoders in a TE system. It was shown in the literature that a precoder drastically improves the system performance. Motivated by this, the EXIT chart is used to analyse the characteristics of four different precoders for long block lengths. It was shown that using a precoder results in a loss in mutual information during the initial equalization stage. However" we show by analysis and simulations that this phenomenon is not observed in the equalization of all precoded channels. The slope of the transfer function, relating to the MAP equalization of a precoded ISI channel (MEP), during the high input mutual information values is shown to play an important role in determining the convergence of precoded TE systems. Simulation results are presented to show how the precoders' weight affects the convergence of TE systems. The design of the hybrid MAP/MEP equalizer is also proposed. We also show that the EXIT chart can be used to compute the trellis code capacity of a precoded ISI channel.
Power and performance trade-off in DS-CDMA receivers based on adaptive LMS-MMSE multi-user detector.
(2003) Wang, Qingsheng.; Takawira, Fambirai.
Third generation cellular communication systems based on CDMA techniques have shown great scope for improvement in system capacity. Over the last decade, there has been significant interest in DS-CDMA detectors. The conventional detector, the optimal detector and a number of sub-optimal multi-user detectors (MUD) have been extensively analyzed in the literature. Recently, the reduction of power consumption in DS-CDMA systems has also become another important consideration in both system design and in implementation. In order to support wireless multimedia services, all CDMA-based systems for third generation systems have a large bandwidth and a high data rate, therefore the power consumed by the digital signal processor (DSP) is high. This thesis focuses on power consumption in the adaptive Minimum Mean Square Error (MMSE) detector which is based on the Least Mean Square (LMS) algorithm. This thesis presents a literature survey on MUD and adaptive filter algorithms. A system model of the quantized LMS-MMSE MUD is proposed and its performance is analyzed. The quantization effects in the finite precision LMS-MMSE adaptive MUD including the steady-state weight covariance, mean square error (MSE) and bit error rate (BER) versus wordlength of data and coefficient are investigated when both the data and filter coefficients are quantized. The effects of wordlength size on power consumption are investigated and the tradeoff between the power consumption and performance degradation and the optimal allocation of bits to data and to LMS coefficients under power constraint is presented.
Subspace-based channel estimation for DS/CDMA systems exploiting pulse- shaping information.
(2003) Gaffar, Mohammed Yusuf Abdul.; Broadhurst, Anthony D.
Third generation wireless systems have adopted Direct-Sequence/Code-Division Multiple-Access (DS/CDMA) as the multiple access scheme of communication. This system would typically operate in a multipath fading channel. This dissertation only deals with the task of channel estimation at the base station where the multipath delays and attenuations for each user are estimated. This information is used to aid the recovery of data that was transmitted by each user. Subspace-based algorithms are popularly used to perform the task of channel estimation because they have the desirable property of perfectly estimating the channel in a noise-free environment. In this dissertation a new subspace-based channel estimation algorithm for DS/CDMA systems is presented. The proposed algorithm is based on the Parametric Subspace algorithm by Perros-Meilhac et al. for single-user systems. The main focus of this dissertation is to convert the Parametric Subspace algorithm from a single-user system to a multi-user DS/CDMA system. It has been shown in the literature that by using information of the pulse-shaping filter in the Channel Subspace algorithm, the variance of the channel estimates is decreased. However, this has only been applied to a single-user system. There are several subspace algorithms that have been proposed for DS/CDMA systems. Most of these algorithms sample the received signal at the chip rate, making it impossible to exploit knowledge of the pulse-shaping filter in the channel estimation algorithm. In this dissertation a new subspace-based channel estimation algorithm is derived for a DS/CDMA system with multiple receive antennas, where the output is oversampled with respect to the chip rate. By oversampling the received signal, knowledge of the pulse-shaping filter is used in the channel estimation algorithm. It is shown that the variance of the channel estimate for the proposed subspace algorithm is less than the Torlak/Xu subspace algorithm that does not exploit information of the pulse-shaping filter. A mathematical expression of the mean square error of estimation for the new algorithm is also derived. It was shown that the analytic expression provides a good approximation of the actual MSE for high SNR. The Parametric Subspace Delay Estimation (PSDE) algorithm was developed by Perros-Meilhac et al. to estimate the multipath delays introduced by the communications channel. The limitation of the PSDE algorithm is that the performance of the algorithm deteriorates as the power of the multipath signals decrease with increasing delay time. This dissertation proposes a modified version of the PSDE algorithm, called the Modified Parametric Subspace Delay Estimation (MPSDE) algorithm, which performs better than the PSDE algorithm in an environment where the power of the multipath signals varies. The final part of this dissertation discusses the Torlak/Xu channel estimation algorithm and the Bensley/Aazbang delay estimation algorithm. In order to compare the performance of these two subspace algorithms, the Torlak/Xu algorithm is converted to a delay estimation algorithm that is called the Parametric TX algorithm. The performance of the Bensley/Aazbang delay estimation algorithm and the proposed Parametric TX algorithm are compared and it is shown that the Parametric TX algorithm offers the better performance.
Development, implementation and quantification of an ad-hoc routing protocol for mobile handheld terminals.
(2003) Dearham, Nicholas Joseph.; McDonald, Stephen A.
An ad-hoc network is a collection of mobile nodes (wireless communication devices) that transmit data over systems that do not require any centralized control, such as that found in cellular networks. This makes ad-hoc networks suitable for military type applications, since there is no need for an established backbone infrastructure and hence no single-point-of-failure. However, other uses of ad-hoc systems include search and rescue missions, law enforcement operations, commercial and educational communication of laptop (and other handheld device) data, as well as in the transmission of environmental sensor information. The mobile ad-hoc concept brings many design challenges. The dynamic freedom of movement from mobile nodes causes random, sometimes rapidly time changing topologies, which are inappropriate for use through traditional wired protocols. In addition, wireless networks generally contain greater bandwidth, processing and power constraints than their wired counterparts, since they are implemented on embedded mobile, handheld devices. Thus, a different approach is needed in the wireless network domain. This has resulted in wireless routing protocols employing adaptive, multi-hop, distributed methodologies in which each node additionally acts as a router for each of its neighbouring nodes, in order to achieve a large degree of network connectivity. However, due to the broadcast nature of wireless transmissions, ad-hoc systems contain a point-to- multipoint communication architecture, making it well suited to multi-path traffic. One such application is in multicasting, which sends data from one source to two (or more) destinations. But, due to the shared characteristics of the communication channel, such traffic may cause multiple contentions and collisions to occur, which will degrade the efficiency and performance of a protocol. This dissertation examines these different design tradeoffs through the use of a freely available simulation package, known as NS-2 (Network Simulator - version 2). In addition, a novel routing protocol, known as LAMP (Location Aided Multicasting Protocol), is developed to handle time-bounded audio information, which is employed in a network that consists of sixteen commercial handheld devices. LAMP utilizes a destination-sequenced, next-hop routing table to forward multicast data. Since mobility causes neighbouring nodes to continually change, next-hop links need to be periodically updated. But, between each update period, a next-hop link may become broken. Thus, if a packet is required to be routed, for which its' next-hop link is unknown, LAMP reverts to a localized location aided flood to find a path to that destination. However, since flooding causes network congestion, it is only employed when its' table forwarding scheme fails. Results have shown that LAMP improves packet delivery ratios by up to 5% over exisiting flood-limiting schemes: Furthermore, LAMP has been shown to be comparable to leading schemes, even when employed to route data to a single source-destination pair.
Design and implementation of an on-demand ad-hoc routing algorithm for a positional communication system.
(2003) Quazi, Tahmid Al-Mumit.; McDonald, Stephen A.
A mobile ad-hoc network is an autonomous network of mobile devices that are connected via wireless links. In such networks there is no pre-existing infrastructure and nodes are free to move in a random fashion. Due to this mobility mobile ad-hoc networks have dynamic topologies. A host in the network typically has limited bandwidth and energy resources. Routing is a major challenge in the development of such systems and there have been many solutions proposed in the recent past. The aim of this work is to design and implement a routing scheme for a Positional Communication System (PCS). The PCS is a network of mobile handheld pocket PCs connected via wireless interfaces. The system allows voice and data communication between nodes in the network. This dissertation addresses the process of designing a routing protocol for an ad-hoc network. There have been many proposed algorithms that solve the routing problem in a mobile ad-hoc network. It is a difficult task to compare the performance of'these protocols qualitatively as there are many parameters that affect network performance. Various simulation packages for networks of this type exist. One such package is the Network Simulator (NS-2). It is a discrete time event simulator that can be used to model wired and wireless networks. This dissertation presents NS-2 simulations that compare four recently proposed routing algorithms. From this comparison study it is shown that on-demand algorithms perform best in a mobile ad-hoc environment. The dissertation then describes the design of a novel on-demand routing algorithm. The ondemand algorithms proposed thus far use a blind flooding technique during the route discovery process. This method is inefficient and creates excessive routing overhead. The routing protocol proposed in the dissertation implements a query localization technique that significantly reduces the network traffic. The protocol also introduces a load checking metric in addition to the metric used by most on-demand schemes, namely hop count. Simulation results show that such a scheme makes the on-demand routing algorithm more efficient and scalable than existing ones. It is widely believed that prior to implementing a routing protocol in real world systems it is essential that it is tested and validated on a test-bed. The dissertation presents the implementation of an on-demand routing algorithm in a Positional Communication System test-bed, where each handheld PC in the network runs an embedded Linux operating system.

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