Thesis (Ph.D.)--Massachusetts Institute of Technology, Dept. of Aeronautics and Astronautics, 1981. / MICROFICHE COPY AVAILABLE IN ARCHIVES AND AERONAUTICS. / Bibliography: p. 235-239. / by Jere Schenck Meserole, Jr. / Ph.D.

The control theory of nonlinear systems has been receiving increasing attention in recent years, both for its technical importance as well as for its impact in various fields of application. In several key areas, such as aerospace, chemical and petrochemical industries, bioengineering, and robotics, a new practical application for this tool appears every day. System nonlinearity is characterized when at least one component or subsystem is nonlinear. Classical methods used in the study of linear systems, particularly superposition, are not usually applied to the nonlinear systems. It is necessary to use other methods to study the control of these systems. For a wide class of nonlinear systems, a rather important structural feature comes from the strong nonlinearity appearing as coupling between spectrally decoupled parts of the system. Even in the case of low frequencies, where lumped models can still be employed the nonlinear coupling between parts of the system requires specific treatment, using advanced mathematical tools. In this context, an alternative, frequency domain approach is pursued here. In the rest of this work, a specific system form of linearly decoupled but nonlinearly coupled subsystems is examined. The mathematical toolbox of the Hilbert transform is appropriately introduced for obtaining two low-pass subsystems that form an equivalent description of the essential overall system dynamics. The nonlinear coupled dynamics is investigated systematically by partitioning the coupled system state vector in such a way as to fully exploit the low-pass and the band-pass intrinsic features of free dynamics. In particular, by employing the Hilbert Transform, a low-pass equivalent system is derived. Then, a typical case is investigated thoroughly by means of numerical simulation of the original coupled low and band-pass, real-state-variable system and the low-pass-equivalent, complex-state-variable derived one. The nonlinear model equations considered here pave the way for a systematic investigation of nonlinear feedback control options designed to operate mechatronic transducers in energy harvesting, sensing or actuation modes.

Active magnetic bearings (AMBs) have become a key technology in industrial applications with a continued drive for cost reduction and an increase in reliability. AMBs require position feedback to suspend the rotor. Conventional contactless position sensors are used to measure the rotor's position. The major disadvantages of conventional position sensors are their cost and that the sensors are viewed as a weak point in an AMB system. A self-sensing sensor is a type of sensor which is cost effective, reduces sensor wire-length and increases reliability, thus ideal for the industry. This type of sensor relies on the current and voltage signals of the AMB's to obtain the rotor position. Due to the rapid and advanced development of digital electronics, it has become more powerful and cheaper, thus self-sensing in mass production will be cost effective. Different self-sensing approaches were developed in the past and can be divided into two main categories: state estimation and amplitude modulation approaches. In this research the focus will be on the amplitude modulation approach. Amplitude modulation makes use of two signals, namely the modulation signal and the carrier signal. In a self-sensing AMB system the carrier can be a high frequency component injected into the system or the switching ripple of the switch mode power amplifier can be used. The modulation signal is the change in rotor position which results in changing inductances. The actuator material introduces nonlinear effects on the estimated position. Due to these nonlinear effects, it is rather difficult to obtain the rotor position. The first industrial application of a self-sensing turbomolecular pump system was implemented in 2005 by S2M. The aim of this thesis is to evaluate existing self-sensing schemes, devise improvements and investigate possible new schemes. Four different demodulation methods and two new self-sensing schemes are evaluated. An AMB transient simulation model which includes saturation, hysteresis, eddy currents and cross-coupling is used to evaluate the schemes in simulation. The self-sensing schemes are implemented in hardware and evaluated on a 7 A rms 500 N AMB. A comparative study was done on the different self-sensing schemes. From the comparative study it was determined that the gain- and phase effects have a direct effect on the sensitivity of the system. It was also proved that self-sensing can be implemented on a coupled AMB with a sensitivity of 10.3 dB. / Thesis (Ph.D. (Electrical and Electronic Engineering))--North-West University, Potchefstroom Campus, 2009.

In recent years, the state-of-the-art in field programmable gate array (FPGA) technology has been advancing rapidly. Consequently, the use of FPGAs is being considered in many applications which have traditionally relied upon application-specific integrated circuits (ASICs). FPGA-based designs have a number of advantages over ASIC-based designs, including lower up-front engineering design costs, shorter time-to-market, and the ability to reconfigure devices in the field. However, ASICs have a major advantage in terms of computational resources. As a result, expensive high performance ASIC algorithms must be redesigned to fit the limited resources available in an FPGA. <p> Concurrently, coaxial cable television and internet networks have been undergoing significant upgrades that have largely been driven by a sharp increase in the use of interactive applications. This has intensified demand for the so-called upstream channels, which allow customers to transmit data into the network. The format and protocol of the upstream channels are defined by a set of standards, known as DOCSIS 3.0, which govern the flow of data through the network. <p> Critical to DOCSIS 3.0 compliance is the upstream demodulator, which is responsible for the physical layer reception from all customers. Although upstream demodulators have typically been implemented as ASICs, the design of an FPGA-based upstream demodulator is an intriguing possibility, as FPGA-based demodulators could potentially be upgraded in the field to support future DOCSIS standards. Furthermore, the lower non-recurring engineering costs associated with FPGA-based designs could provide an opportunity for smaller companies to compete in this market. <p> The upstream demodulator must contain complicated synchronization circuitry to detect, measure, and correct for channel distortions. Unfortunately, many of the synchronization algorithms described in the open literature are not suitable for either upstream cable channels or FPGA implementation. In this thesis, computationally inexpensive and robust synchronization algorithms are explored. In particular, algorithms for frequency recovery and equalization are developed. <p> The many data-aided feedforward frequency offset estimators analyzed in the literature have not considered intersymbol interference (ISI) caused by micro-reflections in the channel. It is shown in this thesis that many prominent frequency offset estimation algorithms become biased in the presence of ISI. A novel high-performance frequency offset estimator which is suitable for implementation in an FPGA is derived from first principles. Additionally, a rule is developed for predicting whether a frequency offset estimator will become biased in the presence of ISI. This rule is used to establish a channel excitation sequence which ensures the proposed frequency offset estimator is unbiased. <p> Adaptive equalizers that compensate for the ISI take a relatively long time to converge, necessitating a lengthy training sequence. The convergence time is reduced using a two step technique to seed the equalizer. First, the ISI equivalent model of the channel is estimated in response to a specific short excitation sequence. Then, the estimated channel response is inverted with a novel algorithm to initialize the equalizer. It is shown that the proposed technique, while inexpensive to implement in an FPGA, can decrease the length of the required equalizer training sequence by up to 70 symbols. <p> It is shown that a preamble segment consisting of repeated 11-symbol Barker sequences which is well-suited to timing recovery can also be used effectively for frequency recovery and channel estimation. By performing these three functions sequentially using a single set of preamble symbols, the overall length of the preamble may be further reduced.

In recent years, the state-of-the-art in field programmable gate array (FPGA) technology has been advancing rapidly. Consequently, the use of FPGAs is being considered in many applications which have traditionally relied upon application-specific integrated circuits (ASICs). FPGA-based designs have a number of advantages over ASIC-based designs, including lower up-front engineering design costs, shorter time-to-market, and the ability to reconfigure devices in the field. However, ASICs have a major advantage in terms of computational resources. As a result, expensive high performance ASIC algorithms must be redesigned to fit the limited resources available in an FPGA. <p> Concurrently, coaxial cable television and internet networks have been undergoing significant upgrades that have largely been driven by a sharp increase in the use of interactive applications. This has intensified demand for the so-called upstream channels, which allow customers to transmit data into the network. The format and protocol of the upstream channels are defined by a set of standards, known as DOCSIS 3.0, which govern the flow of data through the network. <p> Critical to DOCSIS 3.0 compliance is the upstream demodulator, which is responsible for the physical layer reception from all customers. Although upstream demodulators have typically been implemented as ASICs, the design of an FPGA-based upstream demodulator is an intriguing possibility, as FPGA-based demodulators could potentially be upgraded in the field to support future DOCSIS standards. Furthermore, the lower non-recurring engineering costs associated with FPGA-based designs could provide an opportunity for smaller companies to compete in this market. <p> The upstream demodulator must contain complicated synchronization circuitry to detect, measure, and correct for channel distortions. Unfortunately, many of the synchronization algorithms described in the open literature are not suitable for either upstream cable channels or FPGA implementation. In this thesis, computationally inexpensive and robust synchronization algorithms are explored. In particular, algorithms for frequency recovery and equalization are developed. <p> The many data-aided feedforward frequency offset estimators analyzed in the literature have not considered intersymbol interference (ISI) caused by micro-reflections in the channel. It is shown in this thesis that many prominent frequency offset estimation algorithms become biased in the presence of ISI. A novel high-performance frequency offset estimator which is suitable for implementation in an FPGA is derived from first principles. Additionally, a rule is developed for predicting whether a frequency offset estimator will become biased in the presence of ISI. This rule is used to establish a channel excitation sequence which ensures the proposed frequency offset estimator is unbiased. <p> Adaptive equalizers that compensate for the ISI take a relatively long time to converge, necessitating a lengthy training sequence. The convergence time is reduced using a two step technique to seed the equalizer. First, the ISI equivalent model of the channel is estimated in response to a specific short excitation sequence. Then, the estimated channel response is inverted with a novel algorithm to initialize the equalizer. It is shown that the proposed technique, while inexpensive to implement in an FPGA, can decrease the length of the required equalizer training sequence by up to 70 symbols. <p> It is shown that a preamble segment consisting of repeated 11-symbol Barker sequences which is well-suited to timing recovery can also be used effectively for frequency recovery and channel estimation. By performing these three functions sequentially using a single set of preamble symbols, the overall length of the preamble may be further reduced.

The reason of the essay research find position of leakage point, and design a fiber optical leakage detection system. The research of fiber optical detection system in past year, because property of structure produce SNR smaller, quality of detection system is too bed. The essay brings to a hybrid Mach-Zehnder & Sagnac interferomtric can improve the SNR to 10dB. Therefore, it provides the better SNR. The experiment is added to signal process, such as PTL, PGC structure. To provide the systematic characteristic, such as dynamic range(60dB), percentage error(0.025%). The focal of the essay provide leak detection systematic characteristics how we make use of signal process.

The main frame of the distributed fiber optic leak detection system adopted the hybrid Mach-Zehnder & Sagnac interferomtric. We use the sensing fiber of In-Line frame to detect leak physical field. We can measure the position of the leak physical field through our sensing system and signal process system. In the cause of improving detective ability of leak detection system, we modify three elements of the system, including (1) the choice of the acoustic response of sensing fiber, (2) modification of the PZT phase modulator, and (3) modification of the PGC demodulator. The frame of our experiment is composed of the distributed fiber optic leak detection system and leak system of the fluid pipes. In which leak system of fluid pipes is designed the leaky frame of high-pressure fluid pipes. The main of experiment introduce the leak detection system to measure the leak acoustics of the fluid pipes. Then we can discuss the experimental result. The measurable minimum range of our distributed fiber optic leak detection system is3.3x10^-4(rad/¡ÔHz), and the dynamic range is above 75 dB. The dynamic range of this system can improve the original system to above 15 dB.

Optical coherence tomography (OCT) and optical coherence microscopy (OCM) are non-invasive optical coherence imaging techniques, which enable micron-scale resolution, depth resolved imaging capability. Both OCT and OCM are based on Michelson interferometer theory. They are widely used in ophthalmology, gastroenterology and dermatology, because of their high resolution, safety and low cost. OCT creates cross sectional images whereas OCM obtains en face images. In this dissertation, the design and development of three increasingly complicated analog signal processing (ASP) solutions for optical coherence imaging are presented.The first ASP solution was implemented for a time domain OCT system with a Rapid Scanning Optical Delay line (RSOD)-based optical signal modulation and logarithmic amplifier (Log amp) based demodulation. This OCT system can acquire up to 1600 A-scans per second. The measured dynamic range is 106dB at 200A-scan per second. This OCT signal processing electronics includes an off-the-shelf filter box with a Log amp circuit implemented on a PCB board.The second ASP solution was developed for an OCM system with synchronized modulation and demodulation and compensation for interferometer phase drift. This OCM acquired micron-scale resolution, high dynamic range images at acquisition speeds up to 45,000 pixels/second. This OCM ASP solution is fully custom designed on a perforated circuit board.The third ASP solution was implemented on a single 2.2 mm x 2.2 mm complementary metal oxide semiconductor (CMOS) chip. This design is expandable to a multiple channel OCT system. A single on-chip CMOS photodetector and ASP channel was used for coherent demodulation in a time domain OCT system. Cross-sectional images were acquired with a dynamic range of 76dB (limited by photodetector responsivity). When incorporated with a bump-bonded InGaAs photodiode with higher responsivity, the expected dynamic range is close to 100dB.

Continuous Phase Modulation is a popular digital modulation scheme for systems which have tight spectral efficiency and Peak-to-Average ratio (PAR) constraints. In this thesis we propose a method of estimating the capacity for a Continuous Phase Modulation (CPM) system and also describe techniques for design of codes for this system. We note that the CPM modulator can be decomposed into a trellis code followed by a memoryless modulator. This decomposition enables us to perform iterative demodulation of the signal and improve the performance of the system. Thus we have the option of either performing iterative demodulation, where the channel decoder and the demodulator are invoked in an iterative fashion, or a non-iterative demodulation, where the demodulation is performed only once followed by the decoding of the message. We highlight the recent results in the estimation of capacity for channels with memory and apply it to a CPM system. We estimate two different types of capacity of the CPM system over an Additive White Gaussian Noise (AWGN). The first capacity assumes that optimum demodulation and decoding is done, and the second one assumes that the demodulation is done only once. Having obtained the capacity of the system we try to approach this capacity by designing outer codes matched to the CPM system. We utilized LDPC codes, since they can be designed to perform very close to capacity limit of the system. The design complexity for LDPC codes can be reduced by assuming that the input to the decoder is Gaussian distributed. We explore three different ways of approximating the CPM demodulator output to a Gaussian distribution and use it to design LDPC codes for a Bit Interleaved Coded Modulation (BICM) system. Finally we describe the design of Multi Level Codes (MLC) for CPM systems using the capacity matching rule.

Most of the conventional communication systems use channel interleaving as well as hard decision decoding in their designs, which lead to discarding channel memory and soft-decision information. This simplification is usually done since the complexity of handling the memory or soft-decision information is rather high. In this work, we design two lossy joint source-channel coding (JSCC) schemes that do not use explicit algebraic channel coding for a recently introduced channel model, in order to take advantage of both channel memory and soft-decision information. The channel model, called the non-binary noise discrete channel with queue based noise (NBNDC-QB), obtains closed form expressions for the channel transition distribution, correlation coefficient, and many other channel properties. The channel has binary input and $2^q$-ary output and the noise is a $2^q$-ary Markovian stationary ergodic process, based on a finite queue, where $q$ is the output's soft-decision resolution. We also numerically show that the NBNDC-QB model can effectively approximate correlated Rayleigh fading channels without losing its analytical tractability. The first JSCC scheme is the so called channel optimized vector quantizer (COVQ) and the second scheme consists of a scalar quantizer, a proper index assignment, and a sequence maximum a posteriori (MAP) decoder, designed to harness the redundancy left in the quantizer's indices, the channel's soft-decision output, and noise time correlation. We also find necessary and sufficient condition when the sequence MAP decoder is reduced to an instantaneous symbol-by-symbol decoder, i.e., a simple instantaneous mapping. / Thesis (Master, Electrical & Computer Engineering) -- Queen's University, 2011-09-25 19:43:28.785