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This thesis is about query optimisation studies in a Prolog object-oriented database (P/FDM). Although many of the query optimisation techniques used in relational database systems can be adapted to solve problems in query optimisation for object-oriented databases (OODBs), there are also many problems that are unique to the OODBs, and new techniques need to be developed. Techniques developed for optimising queries in OODB systems can be categorised into three distinct approaches, namely, <i>algebraic transformation, type-based transformation</i>, and <i>method optimisation</i>. However, few OODB systems have used the combination of all three techniques in the design of their query optimisers. We have adopted an unified approach by combining them into the optimisation strategy of a query optimiser for P/FDM. The overall architecture of the P/FDM query optimiser is based on the <i>top-down</i> framework. The source language of the query optimiser is an extension of Shipman's original <i>DAPLEX</i> language. We choose <i>ZF notation</i>, based on set comprehensions in Zermelo-Frankel (ZF) set theory, to represent the intermediate forms of DAPLEX queries. The target language of optimiser is a set of P/FDM primitives which are implemented in Prolog. We have taken the approach of asking the database to reveal the definitions of methods to the optimiser as privileged code. Most OODB systems do not perform method optimisation because of the difficulties caused by defining methods in a procedural programming language. We have been greatly helped by using a functional programming approach which holds methods in a declarative form, and thus makes it much easier to combine and transform them. Heuristics are incorporated into the query optimisation strategy to avoid exhaustive search, so that the optimiser is able to produce efficient query evaluation plans within satisfactory response times.

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Discrete-time audio equalisers introduce a variety of undesirable artefacts into audio mixing systems, namely, distortions caused by finite wordlength constraints, frequency response distortion due to coefficient calculation and signal disturbances that arise from real-time coefficient update. An understanding of these artefacts is important in the design of computationally affordable, good quality equalisers. A detailed investigation into these artefacts using various forms of arithmetic, filter frequency response, input excitation and sampling frequencies is described in this thesis. Novel coefficient calculation techniques, based on the matched z-transform (MZT) were developed to minimise filter response distortion and computation for on-line implementation. It was found that MZT-based filter responses can approximate more closely to s-plane filters, than BZTbased filters, with an affordable increase in computation load. Frequency response distortions and prewarping/correction schemes at higher sampling frequencies (96 and 192 kHz) were also assessed. An environment for emulating fractional quantisation in fixed and floating point arithmetic was developed. Various key filter topologies were emulated in fixed and floating point arithmetic using various input stimuli and frequency responses. The work provides detailed objective information and an understanding of the behaviour of key topologies in fixed and floating point arithmetic and the effects of input excitation and sampling frequency. Signal disturbance behaviour in key filter topologies during coefficient update was investigated through the implementation of various coefficient update scenarios. Input stimuli and specific frequency response changes that produce worst-case disturbances were identified, providing an analytical understanding of disturbance behaviour in various topologies. Existing parameter and coefficient interpolation algorithms were implemented and assessed under fihite wordlength arithmetic. The disturbance behaviour of various topologies at higher sampling frequencies was examined. The work contributes to the understanding of artefacts in audio equaliser implementation. The study of artefacts at the sampling frequencies of 48,96 and 192 kHz has implications in the assessment of equaliser performance at higher sampling frequencies.

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A dissertation submitted in ful lment of the requirements for the degree of Masters in Science in the Information Engineering School of Electrical and Information Engineering August 2016 / The work presented extends and contributes to research in error-control coding and information theory. The work focuses on the construction of block codes using an IIR lter structure. Although previous works in this area uses FIR lter structures for error-detection, it was inherently used in conjunction with other error-control codes, there has not been an investigation into using IIR lter structures to create codewords, let alone to justify its validity. In the research presented, linear block codes are created using IIR lters, and the error-correcting capabilities are investigated. The construction of short codes that achieve the Griesmer bound are shown. The potential to construct long codes are discussed and how the construction is constrained due to high computational complexity is shown. The G-matrices for these codes are also obtained from a computer search, which is shown to not have a Quasi-Cyclic structure, and these codewords have been tested to show that they are not cyclic. Further analysis has shown that IIR lter structures implements truncated cyclic codes, which are shown to be implementable using an FIR lter. The research also shows that the codewords created from IIR lter structures are valid by decoding using an existing iterative soft-decision decoder. This represents a unique and valuable contribution to the eld of error-control coding and information theory. / MT2017

This work is concerned with encoding shape descriptors for a succession of the waveform segments to enable the transmission of speech signals at a low data rate. The segmentation was dependent on the identification of waveform features in speech signals thereby producing an irregular data rate from the time encoding process. The shape descriptors have been related to the real and complex zeros of a waveform through the theory of zero-based signal representation. A study of factors governing the data rates, the speech intelligibility and the buffer delay has been made for the above coding process based on waveform segmentation at zero-crossings. The redundancy in the average information conveyed by the zero-crossing data was investigated from conditional probability measurements resulting in the conclusion that a significant reduction in the data was available from coding procedures utilising the correlation in the data sequence. Signal pre-emphasis and dynamic range were found to control the segmentation rate, the variations in segmentation rate during an utterance determining the buffer size and delay. The transmission rate and the system delay necessary for time encoding were strongly influenced by the distortion arising from buffer management in matching the variable information rate to a constant transmission rate. A reduction by approximately a third in the transmission rate was observed to introduce data underflow distortion at a 200ms system delay setting into approximately 5% of the speech. Finally, a performance assessment of the time encoding process was made, subjectively by a reduced form of Diagnostic Rhyme Test (DRT) and objectively by spectral density plots comparisons. The results have indicated a data rate less than that for delta modulation and a processing complexity less than that for vocoders.

Helium speech is a term used to denote the speech produced by a deep sea diver breathing a Helium Oxygen mixture. The replacement of nitrogen in normal air by Helium solves some of the physiological problems associated with diving under pressure. However, it introduces severe distortion in diver's speech. The principal distortion is the nonlinear frequency expansion in the formant frequencies. A real time enhancement system has been constructed and partially tested. The design specification for this unscrambler has been generalised to enable the system to correct most of the Helium speech distortions. The system operates in the frequency domain and is based on the wide band analysis-synthesis technique. The system's algorithm for correcting the Helium speech distortion, is flexible and could be easily changed to satisfy different diving conditions. The possible use of the system to study Helium speech characteristics has also been considered.

This thesis is concerned primarily with the architecture of Digital Signal Computers. The work is supported by the design, development and application of a novel Digital Signal Computer system, the MAC68. The MAC68 is a Functional Multiprocessor, using two independent processors, one of which executes general-purpose tasks, and the other executes sequences of arithmetic. The particular MAC68 design was arrived at after careful evaluation of existing Digital Signal Computer architectures. MAC68 features are fully evaluated via its application to the Sub-Band Coding of speech, and in particular by the development of a 16Kb/s Sub-band Coder using six sub-bands. MAC68 performance was found to be comparable to that of current DSP micros for basic digital filter tasks, and superior for FFT tasks. The MAC68 architecture is a balance of high-speed arithmetic and general- purpose capabilities, and is likely to have a greater range of application than General-Purpose micros or DSP micros used alone. Suggestions are put forward for MAC68 enhancements utilising state-of-the-art hardware and software technologies. Because of the current widespread use of General-Purpose micros, and because of the possible performance gains to be had with the MAC68-type architecture, it is thought that MAC68 architectural concepts will be of value in the design of future high-performance Digital Signal Computer systems.

Another objective of this work is to study the perceptual optimized video object coding. Since MPEG-4 treats a scene as a composition of video objects that are separately encoded and decoded, such a flexible video coding framework makes it possible to code different video objects with different priorities. It is necessary to analyze the priorities of video objects according to their intrinsic properties and psycho-visual characteristics such that the bit budget can be distributed properly to video objects to improve the perceptual quality of the compressed video. An object-level visual attention model is developed to automatically obtain the visual attention information of video objects. The visual attention values of video objects are calculated and incorporated in the newly developed dynamic bit allocation mechanism to improve the objective quality of the high priority objects such that the perceptual quality of the overall picture can be maximized. / As strict rate control algorithms used in video coding sacrifice the quality consistency, the rate distortion tradeoff is important to achieve a balance between the bit rate and quality. A novel separable rate distortion modeling method is proposed to analyze the rate distortion characteristics of the color video signal. This method provides higher estimation accuracy when compared to the non-separable modeling method. To achieve rate distortion tradeoff in H.264/AVC, a new control strategy is presented. The feedback from the encoder buffer is analyzed by a control-theoretic adaptation approach to avoid buffer overflow and underflow. A novel rate distortion tradeoff controller is designed by considering both the quality variation and buffer fluctuation. Smooth video quality is achieved and the relevant constraints are satisfied. / Due to the unique features of the video object coding such as both texture and shape introducing distortions and video objects being of arbitrarily shapes, the rate distortion analysis and optimization strategies are different from the traditional rectangular frame-based techniques. Two new rate distortion modeling methods are proposed for the shape coding. The first one is a linear rate distortion modeling method. The computational complexity is low and the estimation is accurate. To further improve the modeling performance, a novel statistical learning based method is proposed by incorporating shape features to provide rate distortion analysis for the shape coding. Therefore, a joint texture-shape rate distortion modeling approach is derived by integrating the texture and shape rate distortion models. The new joint texture-shape distortion models provide the basis for optimal bit allocation for the video object coding to minimize the coding distortion with the bit rate constraint and stabilize the buffer fullness. The major contribution of this optimal bit allocation scheme is to provide a unified solution for the following two problems: how to allocate bits between the texture and shape and how to distribute the hit budget for multiple video objects, simultaneously. / This thesis addresses rate distortion analysis, optimization, and control problems in video coding. These rate distortion issues not only provide the theoretical background but also are concerned with the practical design for video coding systems. The main objective of this thesis is to consider the problems associated with analyzing the rate distortion characteristics of the video source and providing optimal solutions or tradeoffs for the rate and distortion in video coding systems. More specifically this thesis focuses on both the object-based video coding system, MPEG-4, and the rectangular frame-based video coding system, H.264/AVC. / Chen, Zhenzhong. / "July 2007." / Adviser: King Ngi Ngan. / Source: Dissertation Abstracts International, Volume: 69-02, Section: B, page: 1194. / Thesis (Ph.D.)--Chinese University of Hong Kong, 2007. / Includes bibliographical references (p. 225-247). / Electronic reproduction. Hong Kong : Chinese University of Hong Kong, [2012] System requirements: Adobe Acrobat Reader. Available via World Wide Web. / Electronic reproduction. [Ann Arbor, MI] : ProQuest Information and Learning, [200-] System requirements: Adobe Acrobat Reader. Available via World Wide Web. / Abstract in English and Chinese. / School code: 1307.