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  • About
  • The Global ETD Search service is a free service for researchers to find electronic theses and dissertations. This service is provided by the Networked Digital Library of Theses and Dissertations.
    Our metadata is collected from universities around the world. If you manage a university/consortium/country archive and want to be added, details can be found on the NDLTD website.
331

Advancements in Radio Astronomical Array Processing: Digital Back End Development and Interferometric Array Interference Mitigation

Burnett, Mitchell Costus 01 December 2017 (has links)
The Brigham Young University (BYU) Radio Astronomy Systems group, in collaboration with the National Radio Astronomy Observatory (NRAO), the Center for Astrophysics at West Virginia University (WVU), and the Green Bank Observatory (GBO) have developed, and commissioned, a broadband real-time digital back end processing system for a 38-element phased array feed (PAF) with 150 MHz of instantaneous bandwidth. This system is capable of producing coarse and fine channel correlations, and implements a real-time beamformer that forms 7 simultaneous dual-polarized beams. This thesis outlines the hardware and software development for the digital back end and presents on-telescope commissioning results. This system has been measured to provide an unprecedented low Tsys/η noise level of 28 K and can perform maps of galactic hydrogen observations in a fraction of the time of a conventional single horn feed. The National Radio Astronomy Observatory (NRAO) has recently announced the concept and development of the next generation Very Large Array (ngVLA), a large interferometric array consisting of 300 radio telescopes and longest baseline (distance between a pair of antennas) of 300 km. Large interferometric arrays have been shown to attenuate radio frequency interference (RFI) because it is decorrelated as it propagates across long baselines. This is not always sufficient, especially with dense core array geometries and with the ever-increasing amount of strong RFI sources. Conventional RFI projection-based mitigation techniques have performed poorly on large interferometers because of covariance matrix estimation error due to decorrelation when identifying interference subspace parameters. This thesis presents an algorithm that overcomes the challenge of decorrelation by applying subspace projection via subarray processing (SP-SAP). Each subarray is designed to have a set of elements with high mutual correlation in the interferer for better estimation of subspace parameters. In simulation, compared to the former approach of applying subspace projection on the full array, SP-SAP improves mitigation of the RFI on the order of 9 dB. A signal of interest is shown then to be observable through the RFI in a full synthetic image.
332

Analysis and Mitigation of SEU-induced Noise in FPGA-based DSP Systems

Pratt, Brian Hogan 11 February 2011 (has links)
This dissertation studies the effects of radiation-induced single-event upsets (SEUs) on digital signal processing (DSP) systems designed for field-programmable gate arrays (FPGAs). It presents a novel method for evaluating the effects of radiation on DSP and digital communication systems. By using an application-specific measurement of performance in the presence of SEUs, this dissertation demonstrates that only 5-15% of SEUs affecting a communications receiver (i.e. 5-15% of sensitive SEUs) cause critical performance loss. It also reports that the most critical SEUs are those that affect the clock, global reset, and most significant bits (MSBs) of computation. This dissertation also demonstrates reduced-precision redundancy (RPR) as an effective and efficient alternative to the popular triple modular redundancy (TMR) for FPGA-based communications systems. Fault injection experiments show that RPR can improve the failure rate of a communications system by over 20 times over the unmitigated system at a cost less than half that of TMR by focusing on the critical SEUs. This dissertation contrasts the cost and performance of three different variations of RPR, one of which is a novel variation developed here, and concludes that the variation referred to as "Threshold RPR" is superior to the others for FPGA systems. Finally, this dissertation presents several methods for applying Threshold RPR to a system with the goal of reducing mitigation cost and increasing the system performance in the presence of SEUs. Additional fault injection experiments show that optimizing the application of RPR can result in a decrease in critical SEUs by as much 65% at no additional hardware cost.
333

A Continuous-Time ADC and DSP for Smart Dust

Chhetri, Dhurv, Manyam, Venkata Narasimha January 2011 (has links)
Recently, smart dust or wireless sensor networks are gaining more attention.These autonomous, ultra-low power sensor-based electronic devices sense and process burst-type environmental variations and pass the data from one node (mote) to another in an ad-hoc network. Subsystems for smart dust are typically the analog interface (AI), analog-to-digital converter (ADC), digital signal processor (DSP), digital-to-analog converter (DAC), power management, and transceiver for communication. This thesis project describes an event-driven (ED) digital signal processing system (ADC, DSP and DAC) operating in continuous-time (CT) with smart dust as the target application. The benefits of the CT system compared to its conventional counterpart are lower in-band quantization noise and no requirement of a clock generator and anti-aliasing filter, which makes it suitable for processing burst-type data signals. A clockless EDADC system based on a CT delta modulation (DM) technique is presented. The ADC output is digital data, continuous in time, known as “data token”. The ADC employs an unbuffered, area efficient, segmented resistor-string (R-string) feedback DAC. A study of different segmented R-string DAC architectures is presented. A comparison in component reduction with prior art shows nearly 87.5% reduction of resistors and switches in the DAC and the D flip-flops in the bidirectional shift registers for an 8-bit ADC, utilizing the proposed segmented DAC architecture. The obtained SNDR for the 3-bit, 4-bit and 8-bit ADC system is 22.696 dB, 30.435 dB and 55.73 dB, respectively, with the band of interest as 220.5 kHz. The CTDSP operates asynchronously and process the data token obtained from the EDADC. A clockless transversal direct-form finite impulse response (FIR) low-pass filter (LPF) is designed. Systematic top-down test-driven methodology is employed through out the project. Initially, MATLAB models are used to compare the CT systems with the sampled systems. The complete CTDSP system is implemented in Cadence design environment. The thesis has resulted in two conference contributions. One for the 20th European Conference on Circuit Theory and Design, ECCTD’11 and the other for the 19th IFIP/IEEE International Conference on Very Large Scale Integration, VLSI-SoC’11. We obtained the second-best student paper award at the ECCTD.
334

Testing and evaluation of the integratability of the Senior processor / Testning och evaluering av Senior processorns integrerbarhet

Hedin, Alexander January 2011 (has links)
The first version of the Senior processor was created as part of a thesis projectin 2007. This processor was completed and used for educational purposes atLinköpings University. In 2008 several parts of the processor were optimized andthe processor expanded with additional functionality as part of another thesisproject. In 2009 an EU funded project called MULTI-BASE started, in which theComputer Division at the Department of Electrical Engineering participated in.For their part of the MULTI-BASE project, the Senior processor was selected tobe used. After continuous revision and development, this processor was sent formanufacturing. The assignment of this thesis project was to test and verify the different func-tions implemted in the Senior processor. To do this a PCB was developed fortesting the Senior processor together with a Virtex-4 FPGA. Extensive testingwas done on the most important functions of the Senior processor. These testsshowed that the manufactured Senior processor works as designed and that it alonecan perform larger calculations and use external hardware accelerators with thehelp of its various interfaces. / Den första versionen av Senior processorn skapades som en del i ett examensarbe-te under 2007, denna processor färdigställdes och användes i utbildningssyfte påLinköping Universitet. 2008 optimerades flera delar av processorn och utökadesmed extra funktionalitet som del av ytterligare ett examensarbete. 2009 startadeett EU finansierat projekt vid namn MULTI-BASE, som ISYs Datortekniks avdel-ning deltar i. Till deras del av MULTI-BASE projektet valdes Senior processorn attanvändas, efter ytterligare utveckling skickades denna processor för tillverkning. Detta examensarbete hade i uppgift att testa och verifiera de olika funktionernasom Senior processorn har implementerats med. För att göra detta tillverkades ettkretskort som ska användas för att testa Senior processorn tillsammans med enVirtex-4 FPGA. Utförliga tester gjordes på de viktigaste funktionerna hos Seniorprocessorn, dessa tester visade att den tillverkade Senior processorn fungerar somplanerat. Den kan på egen hand utföra större beräkningar och använda sig avexterna hårdvare acceleratorer med hjälp av sina olika gränssnitt.
335

Implementation Of A Digital Signal Synthesizer With High Spurious Free Dynamic Range

Kilic, Argun 01 July 2006 (has links) (PDF)
Today&amp / #8217 / s analog modulators and upconverters are inadequate to synthesize and modulate signals with high &amp / #8216 / Spurious Free Dynamic Range&amp / #8217 / (SFDR). Thus, the main objective of this thesis is to design and implement a &amp / #8216 / Digital Signal Synthesizer&amp / #8217 / (DSS) that is capable of synthesizing signals between 50-100 MHz with 60dB SFDR and to modulate them variable symbol rates and modulation techniques with very high phase/frequency resolution and switching speed while keeping the amplitude modulation occurring during a modulated symbol duration as small as possible. In this thesis, digital words of the desired signals are first synthesized in a &amp / #8216 / Field Programmable Gate Array&amp / #8217 / (FPGA) using &amp / #8216 / Direct Digital Synthesizer&amp / #8217 / (DDS) fundamentals and then converted to analog signals with a high speed &amp / #8216 / Digital to Analog Converter&amp / #8217 / (DAC). In order to attain the analog requirements, the system variables such as DAC analog performance, nonlinearities, sample and hold affects, DDS parameters, system clock, bandwidth requirements of analog filters and how they effect the output performance are studied. FPGA blocks that are capable of modulating and synthesizing desired signals are designed and programmed on a FPGA. Finally, single tone and modulated signals are synthesized with this DSS implementation and measured in order to verify this system&amp / #8217 / s performance and capabilities.
336

Magnetotransport measurement system and investigations of different materials in pulsed magnetic fields up to 60 T / Beschreibung der Magnetotransport-Meßanlage und Untersuchungen an verschiedenen Materialien in gepulsten Magnetfeldern bis 60 T

Kozlova, Nadezda 08 October 2005 (has links) (PDF)
In the present work, the magnetotransport measurement technique was developed and various materials, exhibiting resistances from 1 mOhm up to several tens of kOhm, were investigated in pulsed magnetic fields of up to 60 T. Phase diagrams of irreversibility and upper critical fields for pure and Zn-doped YBa2Cu3O_7-x high-temperature superconductors were measured. A high-field study of the electronic properties of the two semimetals LaBiPt and CeBiPt were presented. Magnetoresistance of La0.7Sr0.3MnO3 and La0.7Ca0.3MnO3 thin films were investigated.
337

Models and Methods for Development of DSP Applications on Manycore Processors

Bengtsson, Jerker January 2009 (has links)
Advanced digital signal processing systems require specialized high-performance embedded computer architectures. The term high-performance translates to large amounts of data and computations per time unit. The term embedded further implies requirements on physical size and power efficiency. Thus the requirements are of both functional and non-functional nature. This thesis addresses the development of high-performance digital signal processing systems relying on manycore technology. We propose building two-level hierarchical computer architectures for this domain of applications. Further, we outline a tool flow based on methods and analysis techniques for automated, multi-objective mapping of such applications on distributed memory manycore processors. In particular, the focus is put on how to provide a means for tunable strategies for mapping of task graphs on array structured distributed memory manycores, with respect to given application constraints. We argue for code mapping strategies based on predicted execution performance, which can be used in an auto-tuning feedback loop or to guide manual tuning directed by the programmer. Automated parallelization, optimisation and mapping to a manycore processor benefits from the use of a concurrent programming model as the starting point. Such a model allows the programmer to express different types and granularities of parallelism as well as computation characteristics of importance in the addressed class of applications. The programming model should also abstract away machine dependent hardware details. The analytical study of WCDMA baseband processing in radio base stations, presented in this thesis, suggests dataflow models as a good match to the characteristics of the application and as execution model abstracting computations on a manycore. Construction of portable tools further requires a manycore machine model and an intermediate representation. The models are needed in order to decouple algorithms, used to transform and map application software, from hardware. We propose a manycore machine model that captures common hardware resources, as well as resource dependent performance metrics for parallel computation and communication. Further, we have developed a multifunctional intermediate representation, which can be used as source for code generation and for dynamic execution analysis. Finally, we demonstrate how we can dynamically analyse execution using abstract interpretation on the intermediate representation. It is shown that the performance predictions can be used to accurately rank different mappings by best throughput or shortest end-to-end computation latency.
338

Respiratory sound analysis for flow estimation during wakefulness and sleep, and its applications for sleep apnea detection and monitoring

Yadollahi, Azadeh 15 April 2011 (has links)
Tracheal respiratory sounds analysis has been investigated as a non-invasive method to estimate respiratory flow and upper airway obstruction. However, the flow-sound relationship is highly variable among subjects which makes it challenging to estimate flow in general applications. Therefore, a robust model for acoustical flow estimation in a large group of individuals did not exist before. On the other hand, a major application of acoustical flow estimation is to detect flow limitations in patients with obstructive sleep apnea (OSA) during sleep. However, previously the flow--sound relationship was only investigated during wakefulness among healthy individuals. Therefore, it was necessary to examine the flow-sound relationship during sleep in OSA patients. This thesis takes the above challenges and offers innovative solutions. First, a modified linear flow-sound model was proposed to estimate respiratory flow from tracheal sounds. To remove the individual based calibration process, the statistical correlation between the model parameters and anthropometric features of 93 healthy volunteers was investigated. The results show that gender, height and smoking are the most significant factors that affect the model parameters. Hence, a general acoustical flow estimation model was proposed for people with similar height and gender. Second, flow-sound relationship during sleep and wakefulness was studied among 13 OSA patients. The results show that during sleep and wakefulness, flow-sound relationship follows a power law, but with different parameters. Therefore, for acoustical flow estimation during sleep, the model parameters should be extracted from sleep data to have small errors. The results confirm reliability of the acoustical flow estimation for investigating flow variations during both sleep and wakefulness. Finally, a new method for sleep apnea detection and monitoring was developed, which only requires recording the tracheal sounds and the blood's oxygen saturation level (SaO2) data. It automatically classifies the sound segments into breath, snore and noise. A weighted average of features extracted from sound segments and SaO2 signal was used to detect apnea and hypopnea events. The performance of the proposed approach was evaluated on the data of 66 patients. The results show high correlation (0.96,p < 0.0001) between the outcomes of our system and those of the polysomnography. Also, sensitivity and specificity of the proposed method in differentiating simple snorers from OSA patients were found to be more than 91%. These results are superior or comparable with the existing commercialized sleep apnea portable monitors.
339

Development of an integrated co-processor based power electronic drive / by Robert D. Hudson

Hudson, Robert Dearn January 2008 (has links)
The McTronX research group at the North-West University is currently researching self-sensing techniques for Active Magnetic Bearings (AMB). The research is part of an ongoing effort to expand the knowledge base on AMBs in the School of Electrical, Electronic and Computer Engineering to support industries that make use of the technology. The aim of this project is to develop an integrated co-processor based power electronic drive with the emphasis placed on the ability of the co-processor to execute AMB self-sensing algorithms. The two primary techniques for implementing self-sensing in AMBs are state estimation and modulation. This research focuses on hardware development to facilitate the implementation of the modulation method. Self-sensing algorithms require concurrent processing power and speed that are well suited to an architecture that combines a digital signal processor (DSP) and a field programmable gate array (FPGA). A comprehensive review of various power amplifier topologies shows that the pulse width modulation (PWM) switching amplifier is best suited for controlling the voltage and current required to drive the AMB coils. Combining DSPs and power electronics to form an integrated co-processor based power electronic drive requires detail attention to aspects of PCB design, including signal integrity and grounding. A conceptual design is conducted and forms part of the process of compiling a subsystem development specification for the integrated drive, in conjunction with the McTronX Research Group. Component selection criteria, trade-off studies and various circuit simulations serve as the basis for this essential phase of the project. The conceptual design and development specification determines the architecture, functionality and interfaces of the integrated drive. Conceptual designs for the power amplifier, digital controller, electronic supply and mechanical layout of the integrated drive is provided. A detail design is performed for the power amplifier, digital controller and electronic supply. Issues such as component selection, power supply requirements, thermal design, interfacing of the various circuit elements and PCB design are covered in detail. The output of the detail design is a complete set of circuit diagrams for the integrated controller. The integrated drive is interfaced with existing AMB hardware and facilitates the successful implementation of two self-sensing techniques. The hardware performance of the integrated coprocessor based power electronic drive is evaluated by means of measurements taken from this experimental self-sensing setup. The co-processor performance is evaluated in terms of resource usage and execution time and performs satisfactorily in this regard. The integrated co-processor based power electronic drive provided sufficient resources, processing speed and flexibility to accommodate a variety of self-sensing algorithms thus contributing to the research currently underway in the field of AMBs by the McTronX research group at the North-West University. / Thesis (M.Ing. (Electrical Engineering))--North-West University, Potchefstroom Campus, 2009.
340

Probabilistic modeling of neural data for analysis and synthesis of speech

Matthews, Brett Alexander 13 August 2012 (has links)
This research consists of probabilistic modeling of speech audio signals and deep-brain neurological signals in brain-computer interfaces. A significant portion of this research consists of a collaborative effort with Neural Signals Inc., Duluth, GA, and Boston University to develop an intracortical neural prosthetic system for speech restoration in a human subject living with Locked-In Syndrome, i.e., he is paralyzed and unable to speak. The work is carried out in three major phases. We first use kernel-based classifiers to detect evidence of articulation gestures and phonological attributes speech audio signals. We demonstrate that articulatory information can be used to decode speech content in speech audio signals. In the second phase of the research, we use neurological signals collected from a human subject with Locked-In Syndrome to predict intended speech content. The neural data were collected with a microwire electrode surgically implanted in speech motor cortex of the subject's brain, with the implant location chosen to capture extracellular electric potentials related to speech motor activity. The data include extracellular traces, and firing occurrence times for neural clusters in the vicinity of the electrode identified by an expert. We compute continuous firing rate estimates for the ensemble of neural clusters using several rate estimation methods and apply statistical classifiers to the rate estimates to predict intended speech content. We use Gaussian mixture models to classify short frames of data into 5 vowel classes and to discriminate intended speech activity in the data from non-speech. We then perform a series of data collection experiments with the subject designed to test explicitly for several speech articulation gestures, and decode the data offline. Finally, in the third phase of the research we develop an original probabilistic method for the task of spike-sorting in intracortical brain-computer interfaces, i.e., identifying and distinguishing action potential waveforms in extracellular traces. Our method uses both action potential waveforms and their occurrence times to cluster the data. We apply the method to semi-artificial data and partially labeled real data. We then classify neural spike waveforms, modeled with single multivariate Gaussians, using the method of minimum classification error for parameter estimation. Finally, we apply our joint waveforms and occurrence times spike-sorting method to neurological data in the context of a neural prosthesis for speech.

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