by Kingsley, King-chi, Kwan. / Thesis (M.Phil.)--Chinese University of Hong Kong, 1994. / Includes bibliographical references (leaves 85-88). / Tables of Contents / Lists of Figures & Tables / Acknowledgments / Abstract / Chapter Chapter 1 --- Introduction / Chapter 1.1 --- Digital Communication System --- p.P. 1 / Chapter 1.2 --- Channel Coding --- p.P. 1 / Chapter 1.3 --- Convolution Encoder --- p.P. 4 / Chapter 1.4 --- Additive White Gaussian Noise (AWGN) Channel --- p.P. 7 / Chapter 1.5 --- Trellis Diagram --- p.P. 8 / Chapter 1.6 --- Error Event and Free Distance --- p.P. 8 / Chapter 1.7 --- Euclidean Distance --- p.P. 10 / Chapter 1.8 --- Organization of the Thesis --- p.P. 11 / Chapter Chapter 2 --- QAM and MTCM / Chapter 2.1 --- Introduction --- p.P. 13 / Chapter 2.2 --- M-ary Quadrature Amplitude Modulation (QAM) / Chapter 2.2.1 --- M-ary Digital Modulation --- p.P. 13 / Chapter 2.2.2 --- Quadrature Amplitude Modulation (QAM) --- p.P. 14 / Chapter 2.2.3 --- Probability of Bit Error of M-ary QAM --- p.P. 16 / Chapter 2.3 --- Trellis Coded Modulation (TCM) --- p.P. 17 / Chapter 2.4 --- Multiple Trellis Coded Modulation (MTCM) --- p.P. 19 / Chapter Chapter 3 --- Set Partitioning of Signal Sets / Chapter 3.1 --- Introduction --- p.P. 21 / Chapter 3.2 --- Traditional Set Partitioning Methods / Chapter 3.2.1 --- Ungerboeck's Set Partitioning Method --- p.P. 21 / Chapter 3.22 --- Set Partitioning by M.K. Simon and D. Divsalvar --- p.P. 23 / Chapter 3.3 --- The new Set Partitioning Method / Chapter 3.3.1 --- Nomenclature of the Signal Points in the Signal Constellations --- p.P. 24 / Chapter 3.3.2 --- Generation of the Signal Sets --- p.P. 26 / Chapter 3.3.3 --- Partitioning of the Signal Sets / Chapter 3.3.3.1 --- Input Constraints of the Partitioning Method --- p.P. 30 / Chapter 3.3.3.2 --- The Set Partitioning Method --- p.P. 30 / Chapter 3.3.4 --- Distance Properties of the Partitioned Signal Sets --- p.P. 36 / Chapter 3.3.5 --- The Selection Scheme --- p.P. 39 / Chapter 3.3.6 --- Assignment of Signal Subsets into Trellis --- p.P. 42 / Chapter Chapter 4 --- Performance Evaluation / Chapter 4.1 --- Introduction --- p.P. 46 / Chapter 4.2 --- Upper Bound of Error Probability / Chapter 4.2.1 --- Probability of Symbol Error --- p.P. 46 / Chapter 4.2.1.1 --- Upper Bound on Probability of Symbol Error --- p.P. 48 / Chapter 4.2.1.2 --- Computation of the Transfer Function --- p.P. 49 / Chapter 4.2.2 --- Probability of Bit Error --- p.P. 51 / Chapter 4.3 --- Computation of the Free Distance --- p.P. 53 / Chapter Chapter 5 --- Results Presentation and Discussions / Chapter 5.1 --- Introduction --- p.P. 58 / Chapter 5.2 --- Results Presentations / Chapter 5.2.1 --- Normalized Square Free Euclidean Distance --- p.P. 58 / Chapter 5.2.2 --- Error Probability --- p.P. 71 / Chapter 5.3 --- Discussions --- p.P. 77 / Chapter Chapter 6 --- Conclusions --- p.P. 83 / Bibliography --- p.P. 85 / Chapter Appendix A - --- Flowchart of the Program --- p.P. 89 / Chapter Appendix B - --- Tabulated Results of d2free --- p.P. 104

In time encoded speech (TES), information is transmitted relating to the distances between zero crossings and the shape of the waveform between successive zero crossings. The quality of the reconstructed TES signal will therefore depend on the accuracy to which these original signal parameters are presented in the reconstructed signal. When transmitting the waveform parameter descriptors (symbols), the variable TES symbol generation rate has to be matched to constant rate transmission channels using first-in first-out storage buffers. Since there are large variations in generation rates, at modest transmission rates, these buffers overflow destroying some of the symbols. Therefore in practical TES systems, the description of the original signal parameters will also depend on the amount of buffer distortion introduced in the transmission process. In this thesis, two techniques of describing the waveshape more: accurately than existing TES methods, four methods of controlling buffer overflow, and the auditory effects of these waveshape describing the buffer overflow control techniques are presented. Using the two new waveshape describing techniques and a parabolic reconstruction techniques it is shown that to obtain a significant improvement in quality in high quality TES Systems, a substantial increase in precise original signal information is required. Ways of achieving this kind of increase in original signal information without significantly increasing the data rate, has been suggested and demonstrated. Using the four buffer control strategies it is shown that for the control strategies to operate satisfactorily, buffer overflow in the voiced regions should be avoided. It is then shown that this can be achieved without significantly increasing the transmission rate, by exploiting properties of speech perception. Further, various methods of quantising TES parameters and the tradeoffs between quantisation and buffer overflow distortion are also investigated.

The work reported in this thesis deals with aspects of synchronisation and tracking in direct sequence spread spectrum systems used in ranging and communications applications. This is regarded as a major design problem in such systems and several novel solutions are presented. Three main problem areas have been defined: i) reduction of the acquisition time of code sychronisation in the spread spectrum receiver; ii) reduction of the receiver complexity; iii) improvement of the signal to noise ratio performance of the system by better utilisation of the power spectrum in the main lobe of the transmitted signal. Greater tolerance to Doppler shift effects is also important. A general review of the spread spectrum concept and past work is first given in Chapter One, and common methods of synchronisation and tracking are reviewed in Chapter Two. There, current performance limitations are also included. In Chapter Three a novel method is given for increasing the speed of synchronisation between locally generated and received codes, using a technique of controlling the loop's error curve during acquisition. This method is applied to different width delay lock loops, and a significant increase in maximum search rate is obtained. The effect of the width of the discriminator characteristics and damping ratio on the maximum search rate are also examined. The technique is applied to data modulated spread spectrum systems which use either synchronous or asynchronous data communication systems. All methods have been tested experimentally and found to perform as predicted theoretically. Several novel spread sprectrum configurations are given in Chapter Four which employ multi-level sequences. Some configurations have reduced the complexity and cost of the spread spectrum receivers. Others show some improvement in the maximum search rate as well as the signal to noise ratio performance. Some of these configurations have been implemented experimentally. In Chapter Five, the generation and properties of the composite (Kronecker) sequences are explained. Several types of component sequences are examined. And the reception of these composite sequences are discussed. In particular, a technique is introduced for achieving a rapid acquisition of phase synchronisation using these codes. The effect of white Gaussian noise on the acquisition performance of the delay lock loop is given in Chapter Six. Experimental results are obtained for both digital and analogue correlators. Chapter Seven gives a final summary of the conclusions, and further work suggestions.

Time Encoded Speech (TES) transmits information concerning the duration between zero-crossings, shape and the amplitude of the signal between successive zero-crossings. This thesis examines a number of aspects of TES with the view of achieving data reductions to enable the transmission of speech, with acceptable quality and intelligibility, at low bit rates and a practical system delay. This thesis presents: (i) A study of techniques for signalling amplitude information in a TES coder. It was indicated that a minimum of the order of 1 bit per epoch is required. Diagnostic Rhyme Tests (DRT) yielded intelligibility scores of the order of 88% for algorithms employing 1 and 2 bits of amplitude information per epoch. (ii) Investigations into Median and Moving Average filtering for preprocessing the epoch duration sequences. It has been shown that such applications, which involve simple numerical smoothing, are of little value for they degrade the quality of the synthesised speech. (iii) Studies of Extremal Coding and Orthogonal Transformations for achieving data reductions in the signalling of epoch duration and, in some instances, the peak magnitude sequences. Each technique yielded a useful data reduction. The technique using Hadamard Transformations yielded the greatest data reduction, a ratio of 2:1 for the representation of the epoch duration sequences. The Hadamard Transformation also proved to be of low complexity in its implementation. (iv) A real-time simplex digital voice channel, developed during the course of this thesis, and a study of the implementation of TES and TES related coders. It is reported that speech of acceptable quality and intelligibility is achieved for a transmission rate of 10 or 15kb/s with a transmission delay of 300ms.

Our primary mode of communication is via speech. Therefore, any person who has difficulty in producing understandable speech, for whatever reason, is at a great disadvantage. It is the role of the speech therapist to help such people to improve their speech production wherever possible. The aim of this work was to develop a computer-based speech therapy tutor for vowel production. The Tutor would be able to analyse monosyllabic utterances in real-time, extract the vowel portion and match this to a pre-determined template, and display the result with no appreciable delay. A fully-working prototype has been developed which employs general principles of aircraft tracking in a novel way, to track the coefficients of the quadratic factors of the all-pole linear-prediction model for speech production. It is shown how tracking these parameters can be used to extract extended vowels from a monosyllabic utterance. It is also shown how the algorithm which is used to determine the optimum frame-to-frame tracks can be used to perform template matching. The real plane on which the parameters are tracked, the rs-plane, suffers from non-linear scaling of frequency measures. This leads to poor spectral resolution of the perceptually-important low frequency parameters. To overcome this problem, the rs-plane can be warped in order that distance measures taken between points on the plane are more meaningful perceptually. The Tutor is based on a personal computer (PC). In order that real-time operation can be achieved, the processing power of the PC is enhanced by the addition of a digital signal processor (TMS32020) board and a transputer (T800) board. The prototype Tutor was developed with help and advice from Dundee Speech Therapy Service, Tayside Health Board, who also conducted a short pilot study of the Tutor.

Ngai Chi Kin. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2004. / Includes bibliographical references (leaves 57-59). / Abstracts in English and Chinese. / Abstract --- p.i / Acknowledgement --- p.iii / Chapter 1 --- Introduction --- p.1 / Chapter 2 --- Linear Network Coding --- p.12 / Chapter 3 --- Combination Networks --- p.16 / Chapter 4 --- Multi-Source Multicast Networks --- p.31 / Chapter 5 --- Multi-source Network Coding with two sinks --- p.42 / Chapter 6 --- Conclusion --- p.55 / Bibliography --- p.59

多輸入多輸出傳輸在過去二十多年來無線通信研究中一直處於中心地位。人們對信息需求的爆炸性增長導致大規模多輸入多輸出系統的出現與發展。在大規模多輸入多輸出系統中有幾十甚至上百的天線與用戶。這種大規模天線能夠極大地提高系統容量及對噪聲的魯棒性。然而，大規模天線系統的物理實現卻是十分困難的。一方面，最優的信號處理算法通常需要指數增長的複雜度。另一方面，數目繁多的天線意味大量包括功率放大器和數模轉換器在內的硬件開銷。這篇論文的研究重點在於能夠降低信號處理複雜度和硬件開銷的信號檢測和預編碼算法。具體而言，本論文的研究包括三部分： / 在第一部分中，我們考慮多輸入多輸出系統中的一個基本問題信號檢測。格型解碼是信號檢測中的一種傳統方法。但是格型解碼（以及其快速近似算法格基規約輔助算法)放鬆了信號檢測中的符號邊界約束因而受到性能限制。我們提出一種自適應的正則化方法來避免格型解碼中邊界約束鬆弛帶來的負面影響。這種方法是基於最大似然解碼器的拉格朗日對偶鬆弛。我們發現了格型解碼和最大似然解碼的一個十分有趣的關係，而這個關係在現有的文獻中並沒有被提及。數值仿真結果顯示拉格朗日對偶鬆弛方法比現有的格型解碼更為優勝。 / 在第二部分中，我們考慮多用戶信號廣播中的矢量擾動方法。矢量擾動是一種能夠接近信道總容量以及簡化用戶數據處理方法。然而，傳統的矢量擾動會導致每根傳輸天線上都有相當大的功率， 導致天線模擬前端的硬件實現有相當大的難度。我們提出一種每天線功率受限的矢量擾動方法來解決這個問題。在這個方法中，我們需要解決一個整數規劃問題。然而，求解這個整數規劃問題需要用到複雜度十分高的枚舉算法。我們用拉格朗日對偶鬆弛方法把這個整數規劃轉化為標準的整數最小二乘問題，然後採用快速的近似算法來求解。數值仿真顯示提出的方法能夠顯著地降低高每天線功率造成的功率回饋。 / 在最後一部分，我們考慮單用戶通信中的恆定包絡預編碼。恆定包絡預編碼是一種最近被提出用於超大規模多輸入多輸出系統的方法。恆定包絡預編碼的優點在於能夠利用價格低廉但是功率效率高的功率放大器。但是恆定包絡預編碼中的一些信號處理問題在之前的文獻中只是得到了部分解答。我們為這些信號處理問題提供了一個完整的解決方案。更進一步地，我們用天線子集選擇來加強恆定包絡預編碼以優化天線傳輸信號及進一步降低天線成本。數值仿真結果顯示包絡預編碼的性能只稍遜於傳統的波束成型方法，但是能恆定包絡傳輸和降低活動的天線數目。 / Multiple-input multiple-output (MIMO) transmission has been at the core of wireless communication research for the past two decades. Driven by the explosive increase of data demand, the development of MIMO systems has entered a large-scale realm where there are dozens of or even more than a hundred antennas and users. The large number of antennas can significantly boost the system throughput and robustness against noise. However, the physical realization of such a large MIMO system can be very complicated and expensive. On the one hand, optimal signal processing algorithms usually have complexities that increase rapidly in the numbers of antennas and users. On the other hand, large number of antennas means increased hardware overheads, such as those of power amplifiers and D/A converters. This thesis considers efficient precoding and detection algorithms that can reduce implementation complexity and cost. Specifically, the thesis consists of the following three parts: / In the first part, we consider a fundamental problem in MIMO communication, namely MIMO detection. The traditional lattice decoding methods, as well as its efficient approximations by lattice reduction aided (LRA) methods, relax the symbol bounds in detection and thus suffer from performance loss. We propose a systematic adaptive regularization approach to lattice decoding to alleviate the adverse effect of symbol bound relaxation, which is based on the study of a Lagrangian dual relaxation (LDR) of the optimal maximum-likelihood (ML) detector. We find an intriguing relationship between lattice decoding and ML, which was not reported in the previous literature. Simulation results show that the proposed LDR approach can significantly outperform existing lattice decoding and LRA methods. / In the second part, we consider the vector perturbation approach which is a promising technique to achieve near-sum capacity and allows simple user processing in the multiuser multiple-input single-output (MISO) downlink scenario. However, the conventional vector perturbation designs can have very high perantenna powers, which causes significant difficulty to power amplifier implementations. To tackle this problem, we propose a vector perturbation design with per-antenna power constraints (VP-PAPC). The resulting optimization problem is an integer program which requires a computationally demanding enumeration process. Lagrangian dual relaxation is used to transform the VP-PAPC problem into standard integer least square problems which may have efficient approximations. Simulation results show that the proposed method can effectively reduce the power back-off caused by high per-antenna power in conventional vector perturbation. / In the last part, we consider constant envelope (CE) precoding in the singleuser MISO downlink scenario. CE precoding is recently proposed as a mean to utilize cheap but power-efficient power amplifiers in very large MIMO systems. We provide complete solutions to some fundamental signal processing issues in CE precoding which were only partially solved in the previous literature. In addition, we enhance CE precoding with antenna subset selection for transmit optimization and implementation cost reduction. Simulation results reveal that the proposed method only exhibits moderate power loss compared to non-CE beamforming but have the advantages of CE transmission and fewer active transmitting antennas. / Detailed summary in vernacular field only. / Detailed summary in vernacular field only. / Detailed summary in vernacular field only. / Detailed summary in vernacular field only. / Pan, Jiaxian. / Thesis (Ph.D.) Chinese University of Hong Kong, 2014. / Includes bibliographical references (leaves 135-147). / Abstracts also in Chinese.

Based on the weight properties of network codes, we present the refined versions of the Hamming bound, the Singleton bound and the Gilbert-Varshamov bound for linear network codes. We give two different algorithms to construct network codes with minimum distance constraints, both of which can achieve the refined Singleton bound. The first algorithm finds a codebook based on a given set of local encoding kernels defining a linear network code. The second algorithm finds a set of of local encoding kernels based on a given classical error-correcting code satisfying a certain minimum distance requirement. / First, the error correction/detection correction capabilities of a network code is completely characterized by a parameter which is equivalent to the minimum Hamming distance when the network code is linear and the weight measure on the error vectors is the Hamming weight. Our results imply that for a linear network code with the Hamming weight being the weight measure on the error vectors, the capability of the code is fully characterized by a single minimum distance. By contrast, for a nonlinear network code, two different minimum distances are needed for characterizing the capabilities of the code for error correction and for error detection. This leads to the surprising discovery that for a nonlinear network code, the number of correctable errors can be more than half of the number of detectable errors. (For classical algebraic codes, the number of correctable errors is always the largest integer not greater than half of the number of detectable errors.) / Network error correction provides a new method to correct errors in network communications by extending the strength of classical error-correcting codes from a point-to-point model to networks. This thesis considers a number of fundamental problems in coherent network error correction. / We further define equivalence classes of weight measures with respect to a general channel model. Specifically, for any given channel, the minimum weight decoders for two different weight measures are equivalent if the two weight measures belong to the same equivalence class. In the special case of network coding, we study four weight measures and show that all the four weight measures are in the same equivalent class for linear network codes. Hence they are all equivalent for error correction and detection when applying minimum weight decoding. / Yang, Shenghao. / Adviser: Raymond W.H. Yeung. / Source: Dissertation Abstracts International, Volume: 70-06, Section: B, page: 3708. / Thesis (Ph.D.)--Chinese University of Hong Kong, 2008. / Includes bibliographical references (leaves 89-93). / 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. / Abstracts in English and Chinese. / School code: 1307.

Song Lihua. / "September 2000." / Thesis (Ph.D.)--Chinese University of Hong Kong, 2000. / Includes bibliographical references (p. 93-[98]). / Electronic reproduction. Hong Kong : Chinese University of Hong Kong, [2012] System requirements: Adobe Acrobat Reader. Available via World Wide Web. / Mode of access: World Wide Web. / Abstracts in English and Chinese.

by Li Xia. / Thesis (Ph.D.)--Chinese University of Hong Kong, 1996. / Includes bibliographical references (p. 129-134). / Electronic reproduction. Hong Kong : Chinese University of Hong Kong, [2012] System requirements: Adobe Acrobat Reader. Available via World Wide Web. / Mode of access: World Wide Web.