The Intermediate Frequency Acoustic Modem (IFAM), developed by Dr. Beaujean, is designed to transmit the command-and-control messages from the top-side to the wet-side unit in ports and very shallow waters. This research presents the design of the turbo coding scheme and its implementation in the IFAM modem with the purpose of meeting a strict requirement for the IFAM error rate performance. To simulate the coded IFAM, a channel simulator is developed. It is basically a multi-tap filter whose parameters are set depending on the channel geometry and system specifics. The simulation results show that the turbo code is able to correct 89% of the messages received with errors in the hostile channel conditions. The Bose-Chadhuri-Hocquenghem (BCH) coding scheme corrects less that 15% of these messages. The other simulation results obtained for the system operation in different shallow water settings are presented. / by Milutin Pajovic. / Thesis (M.S.C.S.)--Florida Atlantic University, 2009. / Includes bibliography. / Electronic reproduction. Boca Raton, Fla., 2009. Mode of access: World Wide Web.

Electronic equalization has recently attracted considerable interest for impairment compensation for its significant cost saving and adaptive compensation capability. In this thesis, we propose novel maximum-likelihood sequence estimation (MLSE) structures for various advanced modulation formats. Electronic equalization of advanced modulation formats further extends the transmission reach and relaxes the speed limitation of electronic devices. We also propose novel application of MLSE for mitigation of timing misalignment between the pulse carver and data modulator in return-to-zero (RZ) systems. / In access networks, we focus on the achievement of centralized light source (CLS) wavelength-division-multiplexing passive optical networks (WDM-PON) with data rate of 10 Gbit/s for both downstream and upstream signals. The previous CLS WDM-PON schemes at 10 Gbit/s suffer from chromatic dispersion (CD) and/or asynchronous upstream modulation. We propose two solutions to mitigate these impairments. By eliminating the modulation synchronization module and all-optical CD compensation module, the proposed methods greatly reduce the cost and operation complexity of high-speed WDM-PON. / In the monitoring for impairment compensation, we propose a polarization-insensitive monitoring scheme for synchronized phase re-modulation by using a narrowband optical-passband filter (OBPF). With the optimal central wavelength of the OBPF, high monitoring sensitivity is achieved. / The increasing bandwidth demands have aroused a myriad of industry and academic activities to develop cost-effective optical communication systems with data rates of 10 Gbit/s and beyond. However, as the capacity grows, many signal degradation effects become prominent and seriously limit the data rate and the transmission distance. The mitigation of the impairments inevitably increases the operation complexity and implementation cost. The focus of this thesis is to develop new impairment mitigation approaches to improve the impairment compensation performance and/or to reduce the operation complexity and cost. As a result, cost-effective high-speed optical communication systems are enabled. / To freely enable the employment of advanced modulation formats for optical communications, we propose all-optical conversion from 40-Gbit/s RZ signal to 40-Gbit/s inverse-RZ/10-Gbit/s differential-phase-shift-keying orthogonal modulation signal to interface high-speed transmission systems using RZ format with networks using orthogonal modulation format. We also propose a novel all-optical coding and decoding scheme for 20-Gbit/s four-amplitude-shift-keying signal. / Zhao Jian. / "July 2007." / Adviser: Lian-kuan Chen. / Source: Dissertation Abstracts International, Volume: 69-01, Section: B, page: 0579. / Thesis (Ph.D.)--Chinese University of Hong Kong, 2007. / Includes bibliographical references (p. 152-173). / 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.

Chok-ki Chan. / Bibliography: leaves 117-122 / Thesis (Ph.D.) - Chinese University of Hong Kong, 1984

by Ge Gao. / Thesis (M.Phil.)--Chinese University of Hong Kong, 1999. / Includes bibliographical references (leaves 101-107). / Text in English; abstracts in English and Chinese. / by Ge Gao. / Chapter 1 --- Introduction --- p.1 / Chapter 1.1 --- Attributes of speech coders --- p.1 / Chapter 1.1.1 --- Bit rate --- p.2 / Chapter 1.1.2 --- Speech quality --- p.3 / Chapter 1.1.3 --- Complexity --- p.3 / Chapter 1.1.4 --- Delay --- p.4 / Chapter 1.1.5 --- Channel-error sensitivity --- p.4 / Chapter 1.2 --- Development of speech coding techniques --- p.5 / Chapter 1.3 --- Motivations and objectives --- p.7 / Chapter 2 --- Waveform interpolation speech model --- p.9 / Chapter 2.1 --- Overview of speech production model --- p.9 / Chapter 2.2 --- Linear prediction(LP) --- p.11 / Chapter 2.3 --- Linear-prediction based analysis-by-synthesis coding(LPAS) --- p.14 / Chapter 2.4 --- Sinusoidal model --- p.15 / Chapter 2.5 --- Mixed Excitation Linear Prediction(MELP) model --- p.16 / Chapter 2.6 --- Waveform interpolation model --- p.16 / Chapter 2.6.1 --- Principles of waveform interpolation model --- p.18 / Chapter 2.6.2 --- Outline of a WI coding system --- p.25 / Chapter 3 --- Pitch detection --- p.31 / Chapter 3.1 --- Overview of existing pitch detection methods --- p.31 / Chapter 3.2 --- Robust Algorithm for Pitch Tracking(RAPT) --- p.33 / Chapter 3.3 --- Modifications of RAPT --- p.37 / Chapter 4 --- Development of a 1.7kbps speech coder --- p.44 / Chapter 4.1 --- Architecture of the coder --- p.44 / Chapter 4.2 --- Encoding of unvoiced speech --- p.46 / Chapter 4.3 --- Encoding of voiced speech --- p.46 / Chapter 4.3.1 --- Generation of PCW --- p.48 / Chapter 4.3.2 --- Variable Dimensional Vector Quantization(VDVQ) --- p.53 / Chapter 4.3.3 --- Sparse frequency representation(SFR) of speech --- p.56 / Chapter 4.3.4 --- Sample selective linear prediction (SSLP) --- p.58 / Chapter 4.4 --- Practical implementation issues --- p.60 / Chapter 5 --- Development of a 2.0kbps speech coder --- p.67 / Chapter 5.1 --- Features of the coder --- p.67 / Chapter 5.2 --- Postfiltering --- p.75 / Chapter 5.3 --- Voice activity detection(VAD) --- p.76 / Chapter 5.4 --- Performance evaluation --- p.79 / Chapter 6 --- Conclusion --- p.85 / Chapter A --- Subroutine for pitch detection algorithm --- p.88 / Chapter B --- Subroutines for Pitch Cycle Waveform(PCW) generation --- p.96 / Chapter B.1 --- The main subroutine --- p.96 / Chapter B.2 --- Subroutine for peak picking algorithm --- p.98 / Chapter B.3 --- Subroutine for encoding the residue (using VDVQ) --- p.99 / Chapter B.4 --- Subroutine for synthesizing PCW from its residue --- p.100 / Bibliography --- p.101

Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Mechanical Engineering, 1978. / Includes bibliographical references. / by Chun-Chieh Chen. / M.S.

by Hu, Chen. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2004. / Includes bibliographical references (leaves 36-39). / Abstracts in English and Chinese. / Chapter 1 --- Introduction --- p.1 / Chapter 1.1 --- Noise Model --- p.1 / Chapter 1.1.1 --- Impulse Noise --- p.1 / Chapter 1.2 --- Removing Impulse Noise --- p.2 / Chapter 1.2.1 --- Nonlinear Filter --- p.3 / Chapter 1.2.2 --- Variational Method --- p.4 / Chapter 1.3 --- Organization of the Dissertation --- p.5 / Chapter 2 --- Review of ACWMF and DPVM --- p.7 / Chapter 2.1 --- Review of ACWMF --- p.7 / Chapter 2.2 --- Review of DPVM --- p.9 / Chapter 2.2.1 --- Minimization Scheme --- p.9 / Chapter 3 --- Two-Phase Iterative Method --- p.12 / Chapter 3.1 --- Introduction --- p.12 / Chapter 3.2 --- Two-Phase Scheme --- p.13 / Chapter 3.2.1 --- Detection Phase --- p.13 / Chapter 3.2.2 --- Restoration Phase --- p.13 / Chapter 3.2.3 --- Summary of the Algorithm --- p.14 / Chapter 4 --- Nonlinear Equation Solver --- p.16 / Chapter 4.1 --- Introduction --- p.16 / Chapter 4.2 --- Newton's Method --- p.17 / Chapter 4.2.1 --- Newton's Method --- p.17 / Chapter 4.2.2 --- Order of Convergence --- p.17 / Chapter 4.3 --- Secant Method --- p.19 / Chapter 4.3.1 --- Secant Method --- p.19 / Chapter 4.3.2 --- Order of Convergence --- p.19 / Chapter 4.4 --- Secant-like Method --- p.21 / Chapter 4.4.1 --- Secant-like Method --- p.21 / Chapter 4.4.2 --- Order of Convergence --- p.24 / Chapter 5 --- Numerical Experiments --- p.27 / Chapter 5.1 --- Removing Noise --- p.27 / Chapter 5.2 --- Complexity of Algorithm --- p.33 / Chapter 6 --- Concluding Remarks --- p.35 / Bibliography --- p.36

Ho Chung Wa. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2004. / Includes bibliographical references. / Abstracts in English and Chinese. / Introduction --- p.6 / Paper I --- p.13 / Paper II --- p.34 / Concluding Remark --- p.51

Cheung Man Ting. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2005. / Includes bibliographical references (leaves 69-71). / Abstracts in English and Chinese. / Chapter 1 --- Introduction --- p.1 / Chapter 1.1 --- The Emergence of ASIP --- p.1 / Chapter 1.1.1 --- Related Work --- p.3 / Chapter 1.2 --- Motivation --- p.6 / Chapter 1.3 --- ASIP Design Methodologies --- p.7 / Chapter 1.4 --- Fundamentals of Speech Recognition --- p.8 / Chapter 1.5 --- Thesis outline --- p.10 / Chapter 2 --- Automatic Speech Recognition --- p.11 / Chapter 2.1 --- Overview of ASR system --- p.11 / Chapter 2.2 --- Theory of Front-end Feature Extraction --- p.12 / Chapter 2.3 --- Theory of HMM-based Speech Recognition --- p.14 / Chapter 2.3.1 --- Hidden Markov Model (HMM) --- p.14 / Chapter 2.3.2 --- The Typical Structure of the HMM --- p.14 / Chapter 2.3.3 --- Discrete HMMs and Continuous HMMs --- p.15 / Chapter 2.3.4 --- The Three Basic Problems for HMMs --- p.17 / Chapter 2.3.5 --- Probability Evaluation --- p.18 / Chapter 2.4 --- The Viterbi Search Engine --- p.19 / Chapter 2.5 --- Isolated Word Recognition (IWR) --- p.22 / Chapter 3 --- Design of ASIP Platform --- p.24 / Chapter 3.1 --- Instruction Fetch --- p.25 / Chapter 3.2 --- Instruction Decode --- p.26 / Chapter 3.3 --- Datapath --- p.29 / Chapter 3.4 --- Register File Systems --- p.30 / Chapter 3.4.1 --- Memory Hierarchy --- p.30 / Chapter 3.4.2 --- Register File Organization --- p.31 / Chapter 3.4.3 --- Special Registers --- p.34 / Chapter 3.4.4 --- Address Generation --- p.34 / Chapter 3.4.5 --- Load and Store --- p.36 / Chapter 4 --- Implementation of Speech Recognition on ASIP --- p.37 / Chapter 4.1 --- Hardware Architecture Exploration --- p.37 / Chapter 4.1.1 --- Floating Point and Fixed Point --- p.37 / Chapter 4.1.2 --- Multiplication and Accumulation --- p.38 / Chapter 4.1.3 --- Pipelining --- p.41 / Chapter 4.1.4 --- Memory Architecture --- p.43 / Chapter 4.1.5 --- Saturation Logic --- p.44 / Chapter 4.1.6 --- Specialized Addressing Modes --- p.44 / Chapter 4.1.7 --- Repetitive Operation --- p.47 / Chapter 4.2 --- Software Algorithm Implementation --- p.49 / Chapter 4.2.1 --- Implementation Using Base Instruction Set --- p.49 / Chapter 4.2.2 --- Implementation Using Refined Instruction Set --- p.54 / Chapter 5 --- Simulation Results --- p.56 / Chapter 6 --- Conclusions and Future Work --- p.60 / Appendices --- p.62 / Chapter A --- Base Instruction Set --- p.62 / Chapter B --- Special Registers --- p.65 / Chapter C --- Chip Microphotograph of ASIP --- p.67 / Chapter D --- The Testing Board of ASIP --- p.68 / Bibliography --- p.69

Chan Kwok-po. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2003. / Includes bibliographical references. / Abstracts in English and Chinese. / Abstract --- p.i / Acknowledgement --- p.iii / Chapter Chapter 1 --- Introduction --- p.1 / Chapter Chapter 2 --- Characterization of Nonlinearity in Power Amplifier --- p.6 / Chapter 2.1. --- Ideal Amplifier Representation --- p.6 / Chapter 2.2. --- Nonlinear Amplifier Representation --- p.7 / Chapter 2.2.1 --- Series Representation --- p.7 / Chapter 2.2.2 --- AM-AM and AM-PM Distortion --- p.7 / Chapter 2.2.3 --- Two-tone Intermodulation --- p.9 / Chapter 2.2.4 --- Nonlinearity on Digital Modulation Formats --- p.11 / Chapter Chapter 3 --- Linearization Techniques --- p.13 / Chapter 3.1. --- Power Back-off --- p.14 / Chapter 3.2. --- Feedback Technique --- p.15 / Chapter 3.3. --- Pre-distortion Technique --- p.16 / Chapter 3.4. --- Feed-forward Technique --- p.18 / Chapter 3.5. --- Linearization Systems with Signal Processing --- p.19 / Chapter 3.5.1 --- Envelope Elimination and Restoration (EER) --- p.19 / Chapter 3.5.2 --- Linear Amplification Using Nonlinear Components (LINC) --- p.20 / Chapter 3.5.3 --- Combined Analogue-locked Loop Universal Modulator (CALLUM) --- p.21 / Chapter 3.5.4 --- Linear Amplification Employing Sampling Techniques (LIST) --- p.21 / Chapter 3.6. --- Other Linearization Techniques --- p.22 / Chapter Chapter 4 --- Feed-forward Power Amplifier System --- p.23 / Chapter 4.1. --- General Description --- p.23 / Chapter 4.2. --- Adaptive Feed-forward Power Amplifier System --- p.25 / Chapter 4.2.1 --- Power Minimization --- p.28 / Chapter 4.2.2 --- Pilot Injection Technique --- p.29 / Chapter 4.2.3 --- Look-up-table Technique (Temperature Compensation) --- p.31 / Chapter 4.2.4 --- Correlation Based Feedback Control (Dual-loop) --- p.32 / Chapter 4.2.5 --- Correlation Based Feedback Control (Triple-loop) --- p.34 / Chapter 4.2.6 --- Digital Implementation on Adaptive FFPA --- p.35 / Chapter Chapter 5 --- DSP-based Adaptive FFPA Analysis --- p.37 / Chapter 5.1. --- System Architecture --- p.37 / Chapter 5.2. --- System Modeling --- p.39 / Chapter 5.3. --- Principle of Adaptation --- p.40 / Chapter 5.3.1 --- Adaptation in Error Extraction Loop --- p.40 / Chapter 5.3.2 --- Adaptation in Main-tone Suppression Loop --- p.43 / Chapter 5.3.3 --- Adaptation in Distortion Cancellation Loop --- p.44 / Chapter 5.3.4 --- Complex Adaptation --- p.46 / Chapter 5.4. --- Adaptation Performance Analysis --- p.47 / Chapter 5.4.1 --- Condition for Convergence --- p.47 / Chapter 5.4.2 --- Rate of Convergence --- p.48 / Chapter 5.4.3 --- Misadjustment --- p.49 / Chapter 5.4.4 --- Summary of the System Performance --- p.51 / Chapter 5.5. --- System Design Consideration --- p.51 / Chapter 5.5.1 --- Quadrature Sampling --- p.51 / Chapter 5.5.2 --- Data Processing --- p.52 / Chapter 5.6. --- Sensitivity Analysis --- p.55 / Chapter 5.6.1 --- Vector Representation --- p.55 / Chapter 5.6.2 --- Amplitude and Phase Matching --- p.56 / Chapter 5.6.3 --- Time-delay Matching --- p.58 / Chapter 5.7. --- Analog-to-digital Interface: Design Consideration --- p.60 / Chapter 5.7.1 --- Sampling Rate Consideration --- p.60 / Chapter 5.7.2 --- Finite Word-length --- p.61 / Chapter 5.8. --- Digital-to-analog Interface: Design Consideration --- p.63 / Chapter Chapter 6 --- New DSP Algorithms for High Performance Adaptive FFPA --- p.67 / Chapter 6.1. --- Variable Loop-gain Algorithm --- p.67 / Chapter 6.2. --- Variable Step-size Algorithm --- p.71 / Chapter 6.3. --- Least-mean-fourth Algorithm --- p.74 / Chapter Chapter 7 --- Implementation of DSP-based Adaptive FFPA --- p.79 / Chapter 7.1. --- Hardware Construction --- p.79 / Chapter 7.2. --- Experimental Results: LMS Algorithm --- p.82 / Chapter 7.3. --- Experimental Results: Variable Loop-gain Algorithm --- p.86 / Chapter 7.4. --- Experimental Results: Variable Step-size Algorithm --- p.88 / Chapter 7.5. --- Experimental Results: Lesat-mean-fourth Algorithm --- p.90 / Chapter Chapter 8 --- Conclusion --- p.92 / Appendix I Matlab Program for Computer Simulation of Adaptive FFPA --- p.A-l / Appendix II DSP Program for Experimental Adaptive FFPA --- p.A-5 / References --- p.R-1 / Author's Publications --- p.AP-1

Photocopy of typescript. / Digitized by Kansas Correctional Industries