Relevant bibliographies by topics / Digital Signal Processing / Dissertations / Theses
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Dissertations / Theses on the topic 'Digital Signal Processing'

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Published: 4 June 2021
Last updated: 12 June 2025

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1

Hannaske, Roland. "Fast Digitizing and Digital Signal Processing of Detector Signals." Forschungszentrum Dresden, 2010. http://nbn-resolving.de/urn:nbn:de:bsz:d120-qucosa-27888.

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A fast-digitizer data acquisition system recently installed at the neutron time-of-flight experiment nELBE, which is located at the superconducting electron accelerator ELBE of Forschungszentrum Dresden-Rossendorf, is tested with two different detector types. Preamplifier signals from a high-purity germanium detector are digitized, stored and finally processed. For a precise determination of the energy of the detected radiation, the moving-window deconvolution algorithm is used to compensate the ballistic deficit and different shaping algorithms are applied. The energy resolution is determined in an experiment with γ-rays from a 22Na source and is compared to the energy resolution achieved with analogously processed signals. On the other hand, signals from the photomultipliers of barium fluoride and plastic scintillation detectors are digitized. These signals have risetimes of a few nanoseconds only. The moment of interaction of the radiation with the detector is determined by methods of digital signal processing. Therefore, different timing algorithms are implemented and tested with data from an experiment at nELBE. The time resolutions achieved with these algorithms are compared to each other as well as to reference values coming from analog signal processing. In addition to these experiments, some properties of the digitizing hardware are measured and a program for the analysis of stored, digitized data is developed. The analysis of the signals shows that the energy resolution achieved with the 10-bit digitizer system used here is not competitive to a 14-bit peak-sensing ADC, although the ballistic deficit can be fully corrected. However, digital methods give better result in sub-ns timing than analog signal processing.
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Hannaske, Roland. "Fast Digitizing and Digital Signal Processing of Detector Signals." Forschungszentrum Dresden-Rossendorf, 2009. https://hzdr.qucosa.de/id/qucosa%3A21615.

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A fast-digitizer data acquisition system recently installed at the neutron time-of-flight experiment nELBE, which is located at the superconducting electron accelerator ELBE of Forschungszentrum Dresden-Rossendorf, is tested with two different detector types. Preamplifier signals from a high-purity germanium detector are digitized, stored and finally processed. For a precise determination of the energy of the detected radiation, the moving-window deconvolution algorithm is used to compensate the ballistic deficit and different shaping algorithms are applied. The energy resolution is determined in an experiment with γ-rays from a 22Na source and is compared to the energy resolution achieved with analogously processed signals. On the other hand, signals from the photomultipliers of barium fluoride and plastic scintillation detectors are digitized. These signals have risetimes of a few nanoseconds only. The moment of interaction of the radiation with the detector is determined by methods of digital signal processing. Therefore, different timing algorithms are implemented and tested with data from an experiment at nELBE. The time resolutions achieved with these algorithms are compared to each other as well as to reference values coming from analog signal processing. In addition to these experiments, some properties of the digitizing hardware are measured and a program for the analysis of stored, digitized data is developed. The analysis of the signals shows that the energy resolution achieved with the 10-bit digitizer system used here is not competitive to a 14-bit peak-sensing ADC, although the ballistic deficit can be fully corrected. However, digital methods give better result in sub-ns timing than analog signal processing.
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3

Bland, Denise. "Alias-free signal processing of nonuniformly sampled signals." Thesis, University of Westminster, 1998. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.322992.

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4

Farag, Emad N. "VLSI low-power digital signal processing." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp04/nq22199.pdf.

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5

Ekstam, Ljusegren Hannes, and Hannes Jonsson. "Parallelizing Digital Signal Processing for GPU." Thesis, Linköpings universitet, Programvara och system, 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-167189.

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Because of the increasing importance of signal processing in today's society, there is a need to easily experiment with new ways to process signals. Usually, fast-performing digital signal processing is done with special-purpose hardware that are difficult to develop for. GPUs pose an alternative for fast performing digital signal processing. The work in this thesis is an analysis and implementation of a GPU version of a digital signal processing chain provided by SAAB. Through an iterative process of development and testing, a final implementation was achieved. Two benchmarks, both comprised of 4.2 M test samples, were made to compare the CPU implementation with the GPU implementation. The benchmark was run on three different platforms: a desktop computer, a NVIDIA Jetson AGX Xavier and a NVIDIA Jetson TX2. The results show that the parallelized version can reach several magnitudes higher throughput than the CPU implementation.
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Feiste, Kurt Alan. "Merged arithmetic for digital signal processing /." Digital version accessible at:, 1999. http://wwwlib.umi.com/cr/utexas/main.

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7

Kwan, Ching Chung. "Digital signal processing techniques for on-board processing satellites." Thesis, University of Surrey, 1990. http://epubs.surrey.ac.uk/754893/.

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In on-board processing satellite systems in which FDMA/SCPC access schemes are employed. transmultiplexers are required for the frequency demultiplexing of the SCPC signals. Digital techniques for the implementation of the transmultiplexer for such application were examined in this project. The signal processing in the transmultiplexer operations involved many parameters which could be optimized in order to reduce the hardware complexity whilst satisfying the level of performance required of the system. An approach for the assessment of the relationship between the various parameters and the system performance was devised. which allowed hardware requirement of practical system specifications to be estimated. For systems involving signals of different bandwidths a more flexible implementation of the trans multiplexer is required and two computationally efficient methods. the DFT convolution and analysis/synthesis filter bank. were investigated. These methods gave greater flexibility to the input frequency plan of the transmultiplexer. at the expense of increased computational requirements. Filters were then designed to exploit specific properties of the flexible transmultiplexer methods. resulting in considerable improvement in their efficiencies. Hardware implementation of the flexible transmultiplexer was considered and an efficient multi-processor architecture in combination with parallel processing software algorithms for the signal processing operations were designed. Finally. an experimental model of the payload for a land-mobile satellite system proposal. T -SAT. was constructed using general-purpose digital signal processors and the merits of the on-board processing architecture was demonstrated.
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8

DI, NUNZIO LUCA. "Reconfigurable digital architecture for high speed digital signal processing." Doctoral thesis, Università degli Studi di Roma "Tor Vergata", 2010. http://hdl.handle.net/2108/1295.

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Low cost microprocessors and DSPs are optimized to perform arithmetic and logic operations on data having a xed size, typically 16,32 or 64 bit. On the other hand, their e ciency decreases when data shorter respect than their native wordlength are processed (more clock cycles per operation are required). Recently di erent solutions have been proposed to overcome this problem. Among those, the ones based on a main processor with a Recon gurable Unit used as hardware accelerator are the most interesting in terms of performance and exibility. Typically those architectures are similar to very small FPGA; they consist in arrays of Look-Up Tables (LUTs) interconnected by pass transistors networks. This work proposes a new Recon gurable Accelerator called ADAPTO (Adderbased Dynamic Architecture for Processing Tailored Operators). The main di erent between ADAPTO and the others Recon gurable Units proposed in literature is the reduced hardware complexity in terms of silicon area. This feature give the possibility to integrate ADAPTO in embedded low cost microprocessors and DSPs (Digital Signal Processors), in fact, for these kind of processors, the area occupation and therefore the cost is a very critical aspect. The ADAPTO Unit supports both hardware recon guration and instruction execution in the same processor clock cycle. These goals have been obtained with the multicontext approach using a recon gurable unit based on full adders, instead LUTs. As discussed in this work this choice allows to the multicontext technique a reduced wasting of hardware resources.
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9

Wang, Limin. "The ECG signal processing by ADSP-21062 digital signal processor." Morgantown, W. Va. : [West Virginia University Libraries], 1999. http://etd.wvu.edu/templates/showETD.cfm?recnum=840.

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Thesis (M.S.)--West Virginia University, 1999.<br>Title from document title page. Document formatted into pages; contains vi, 110 p. : ill. (some col.) Includes abstract. Includes bibliographical references (p. 66-68).
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Nordström, Jesper. "Real time digital signal processing using Matlab." Thesis, Uppsala universitet, Signaler och System, 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:uu:diva-332075.

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Increased usage of electronic devices and the fast development of microprocessors has increased the usage of digital filters ahead of analog filters. Digital filters offer great benefits over analog filters in that they are inexpensive, they can be reprogrammed easily and they open up whole new range of possibilities when it comes to Internet of things. This thesis describes development of a program that can sample music from the computer's microphone input, filter it inside the program with user built filters and reconstruct the music to the computer's headphone output meaning that the music can be played from the speakers. All of this is to happen in real time. The program is developed for students studying at the department of ``Signals and Systems" and the program is supposed the be one of the educational tools to make sense of signals and filtering. The program works well and filters the sound with satisfying results. It is easy to create filters and filter the signal. Since it is music that is filtered constructing perfect filters with minimum ripple, minimum or linear phase is quite difficult to achieve. The program could be improved by improving the user interface, making the environment more interactive and less difficult to construct good filters. Some improvements could also be made to the implementation; as of now the program might run a bit slow on startup on slower computers.
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Yang, Shijun. "Smart receiver using baseband digital signal processing." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2000. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape4/PQDD_0017/MQ48478.pdf.

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12

Papaspiridis, Alexandros. "Digital signal processing techniques for gene prediction." Thesis, Imperial College London, 2012. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.590037.

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The purpose of this research is to apply existing Digital Signal Processing techniques to DNA sequences, with the objective of developing improved methods for gene prediction. Sections of DNA sequences are analyzed in the frequency domain and frequency components that distinguish intron regions are identified (21t/lOA). Novel detectors are created using digital filters and auto correlation, capable of identifying the location of intron regions in a sequence. The resulting signal from these detectors is used as a dynamic threshold in existing gene detectors, resulting in an improved accuracy of 12% and 25% respectively. Finally, DNA sequences are analyzed in terms of their amino acid composition, and new gene prediction algorithms are introduced.
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13

Roome, Stephen John. "The industrial application of digital signal processing." Thesis, City University London, 1989. http://openaccess.city.ac.uk/7405/.

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This thesis describes an investigation into the application of digital signal processing techniques to the solution of industrial signal processing problems. The investigation took the form of three case studies chosen to illustrate the variety of possible applications. The first was the computer simulation of a digital microwave communications link which utilised narrowband FM modulation and partial response techniques. In order to ensure that the behaviour of the simulation reliably matched that of the modelled system it was found necessary to have a sound theoretical background, implementation using good software engineering methodology together with methodical testing and validation. The second case study was a comprehensive investigation of adaptive noise cancelling systems concentrating on issues important for practical implementation of the technique: stability and convergence of the adaptation algorithm; misadjustment noise and effects due to realizability constraints. It was found that theoretical predictions of the systems behaviour were in good agreement with the results of computer simulation except for the level of output misadjustment noise. In order to make the mathematics of the LMS algorithm tractable it was assumed that the input data formed a series of uncorrelated vectors. It was found that this assumption is only appropriate for the prediction of misadjustment noise when the reference input is uncorrelated. The final case study concerned the automatic detection and assessment of pressing faults on gramophone records for quality assurance purposes. A pattern recognition technique for identifying the signals due to gramophone record defects and a numerical method for assessing the perceived severity of the defects were developed empirically. Prototype equipment was designed, built and tested in extended field trials. The equipment was shown to be superior to previous equipment developed using analogue signal processing techniques. These case studies demonstrate that digital signal processing is a powerful and widely applicable technique for the solution of industrial signal processing problems. Solutions may be theoretical or obtained by experiment or simulation. The strengths and weaknesses of each approach are illustrated.
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Esparcia, Alcázar Anna Isabel. "Genetic programming for adaptive digital signal processing." Thesis, University of Glasgow, 1998. http://theses.gla.ac.uk/4780/.

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Wells, Ian. "Digital signal processing architectures for speech recognition." Thesis, University of the West of England, Bristol, 1995. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.294705.

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Yang, Shijun Carleton University Dissertation Engineering Electronics. "Smart receiver using baseband digital signal processing." Ottawa, 1999.

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17

Vrcelj, Bojan Vaidyanathan P. P. "Multirate signal processing concepts in digital communications /." Diss., Pasadena, Calif. : California Institute of Technology, 2004. http://resolver.caltech.edu/CaltechETD:etd-06252003-115639.

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18

Musoke, David. "Digital image processing with the Motorola 56001 digital signal processor." Scholarly Commons, 1992. https://scholarlycommons.pacific.edu/uop_etds/2236.

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This report describes the design and testing of the Image56 system, an IBM-AT based system which consists of an analog video board and a digital board. The former contains all analog and video support circuitry to perform real-time image processing functions. The latter is responsible for performing non real-time, complex image processing tasks using a Motorola DSP56001 digital signal processor. It is supported by eight image data buffers and 512K words of DSP memory (see Appendix A for schematic diagram).
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Ali-Bakhshian, Mohammad. "Digital processing of analog information adopting time-mode signal processing." Thesis, McGill University, 2013. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=114237.

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As CMOS technologies advance to 22-nm dimensions and below, constructing analog circuits in such advanced processes suffers many limitations, such as reduced signal swings, sensitivity to thermal noise effects, loss of accurate switching functions, to name just a few. Time-Mode Signal Processing (TMSP) is a technique that is believed to be well suited for solving many of these challenges. It can be defined as the detection, storage, and manipulation of sampled analog information using time-mode variables. One of the important advantages of TMSP is the ability to realize analog functions using digital logic structures. This technique has a long history of application in electronics; however, due to lack of some fundamental functions, the use of TM variables has been mostly limited to intermediate stage processing and it has been always associated with voltage/current-to-time and time-to-voltage/current conversion. These conversions necessitate the inclusion of analog blocks that contradict the digital advantage of TMSP. In this thesis, an intensive research has been presented that provides an appropriate foundation for the development of TMSP as a general processing tool. By proposing the new concept of delay interruption, a completely new asynchronous approach for the manipulation of TM variables is suggested. As a direct result of this approach, practical techniques for storage, addition and subtraction of time-mode variables are presented. To Extend the digital implementation of TMSP to a wider range of applications, the comprehensive design of a unity gain dual-path time-to-time integrator (accumulator) is demonstrated. This integrator is then used to implement a digital second-order delta-sigma modulator. Finally, to demonstrate the advantage of TMSP, a very low power and compact tunable interface for capacitive sensors is presented that is composed of a number of delay blocks associated with typical logic gates. All the proposed theories are supported by experimental results and post-layout simulations.The emphasis on the digital construction of the proposed circuits has been the first priority of this thesis. Having the building blocks implemented with a digital structure, provides the feasibility of a simple, synthesizable, and reconfigurable design where affordable circuit calibrations can be adopted to remove the effects of process variations.<br>Les technologies CMOS progressant vers les procédés 22 nm et au delà, la abrication des circuits analogiques dans ces technologies se heurte a de nombreuses limitations. Entre autres limitations on peut citer la réduction d'amplitude des signaux, la sensibilité aux effets du bruit thermique et la perte de fonctions précises de commutation. Le traitement de signal en mode temps (TMSP pour Time-Mode Signal Processing) est une technique que l'on croit être bien adapté pour résoudre un grand nombre de problèmes relatifs a ces limitations. TMSP peut être défini comme la détection, le stockage et la manipulation de l'information analogique échantillonnée en utilisant des quantités de temps comme variables. L'un des avantages importants de TMSP est la capacité à réaliser des fonctions analogiques en utilisant des structures logiques digitales. Cette technique a une longue histoire en terme d'application en électronique. Cependant, en raison du manque de certaines fonctions fondamentales, l'utilisation de variables en mode temps a été limitée à une utilisation comme étape intermédiaire dans le traitement d'un signal et toujours dans le contexte d'une conversion tension/courant-temps et temps-tension/courant. Ces conversions nécessitent l'inclusion de blocs analogiques qui vont a l'encontre de l'avantage numérique des TMSP. Cette thèse fournit un fondement approprié pour le développement de TMSP comme outil général de traitement de signal. En proposant le concept nouveau d'interruption de retard, une toute nouvelle approche asynchrone pour la manipulation de variables en mode temps est suggéré. Comme conséquence directe de cette approche, des techniques pratiques pour le stockage, l'addition et la soustraction de variables en mode temps sont présentées. Pour étendre l'implémentation digitale de TMSP à une large gamme d'applications, la conception d'un intégrateur (accumulateur) à double voie temps- à -temps est démontrée. cet intégrateur est ensuite utilisé pour implémenter un modulateur delta-sigma de second ordre.Enfin, pour démontrer l'avantage de TMSP, une Interface de très basse puissance, compacte et réglable pour capteurs capacitifs est présenté. Cette interface est composé d'un certain nombre de blocs de retard associés à des portes logiques typiques. Toutes les théories proposées sont soutenues par des résultats expérimentaux et des simulations post-layout. L'implémentation digitale des circuits proposés a été la première priorité de cette thèse. En effet, une implémentation des bloc avec des structures digitales permet des conceptions simples, synthétisable et reconfigurables où des circuits de calibration très abordables peuvent être adoptées pour éliminer les effets des variations de process.
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Chan, Tsang Hung. "Digital signal processing in optical fibre digital speckle pattern interferometry." HKBU Institutional Repository, 1996. http://repository.hkbu.edu.hk/etd_ra/269.

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21

Orcutt, Edward Kerry 1964. "Correlation filters for time domain signal processing." Thesis, The University of Arizona, 1989. http://hdl.handle.net/10150/277215.

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This study proposes employing new filters in various configurations for use in digital communication systems. We believe that significant improvements in such performance areas as transmission rate and synchronization may be achieved by incorporating these filters into digital communications receivers. Recently reported in the literature, these filters may offer advantages over the matched filter which allow enhancements in data rates, ISI tolerance, and synchronization. To make full use of the benefits of these filters, we introduce the concept of parallel signal transmission over a single channel. We also examine the effects of signal set selection and noise on performance.
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Okullo-Oballa, Thomas Samuel. "Systolic realization of multirate digital filters." Thesis, [Hong Kong] : University of Hong Kong, 1988. http://sunzi.lib.hku.hk/hkuto/record.jsp?B12433998.

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23

Lei, Chi-un, and 李志遠. "VLSI macromodeling and signal integrity analysis via digital signal processing techniques." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2011. http://hub.hku.hk/bib/B45700588.

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24

Ng, Chiu-wa, and 吳潮華. "Bit-stream signal processing on FPGA." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2009. http://hub.hku.hk/bib/B41633842.

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Hamlett, Neil A. "Comparison of multiresolution techniques for digital signal processing." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from the National Technical Information Service, 1993. http://edocs.nps.edu/npspubs/scholarly/theses/1993/Mar/93Mar_Hamlett.pdf.

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Scraggs, David Peter Thomas. "Digital signal processing techniques for semiconductor Compton cameras." Thesis, University of Liverpool, 2007. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.491364.

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The work presented in this thesis has focused on the development of a low dose Compton camera for nuclear medicine. A Compton camera composed of two high-purity planar germanium orthogonal-strip detectors has been constructed. Fast digital data acquisition has been utilised for the application of pulse shape analysis techniques. A simple back projection imaging code has been developed and validated with a Geant4 radiation transport simulation of the Compton camera configuration. L A 137CS isotropic source and a 22Na anisotropic source have been experimentally reconstructed. Parametric pulse shape analysis was applied to both data sets and has been shown to increase the detector spatial resolution from a raw granularity of 5x5x20mm to a spatial resolution that can be represented by a Gaussian distribution with a standard deviation of 1.5mm < u < 2mm in all dimensions; this result was in-part derived from Geant4 simulations. Qualitatively poor images have been shown to result - based wholly on simulation - from Gaussian spatial-resolution distributions that have a standard deviation of greater than 4mm. A partial experimental basis-data-set has been developed and proved capable of providing 1.9mm FWHM average spatial resolution through the depth axis of a single detector crystal. A novel technique to identify gamma ray scattering within single detector c1osed-face-pixels - hitherto unrecognised - has also been introduced in this thesis. This technique, henceforth known as Digital Compton Suppression (DieS), is based on spectral analysis and has demonstrated the ability of identifying events in which the Compton scattering and photoelectric absorption sites are separated by 13mm in the direction ofthe electric field. Supplied by The British Library - 'The world's knowledge'
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Graumann, P. J. W. "Implementing digital signal processing algorithms using serial arithmetic." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1997. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp04/mq20869.pdf.

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Gilkerson, Paul. "Digital signal processing for time of flight sonar." Thesis, University of Oxford, 2000. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.343467.

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Al-Mbaideen, Amneh Ahmed. "Digital signal processing techniques fpr NIR spectroscopy analysis." Thesis, University of Sheffield, 2011. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.538095.

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Goldfarb, Gilad. "DIGITAL SIGNAL PROCESSING TECHNIQUES FOR COHERENT OPTICAL COMMUNICATION." Doctoral diss., University of Central Florida, 2008. http://digital.library.ucf.edu/cdm/ref/collection/ETD/id/2893.

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Coherent detection with subsequent digital signal processing (DSP) is developed, analyzed theoretically and numerically and experimentally demonstrated in various fiber‐optic transmission scenarios. The use of DSP in conjunction with coherent detection unleashes the benefits of coherent detection which rely on the preservation of full information of the incoming field. These benefits include high receiver sensitivity, the ability to achieve high spectral‐efficiency and the use of advanced modulation formats. With the immense advancements in DSP speeds, many of the problems hindering the use of coherent detection in optical transmission systems have been eliminated. Most notably, DSP alleviates the need for hardware phase‐locking and polarization tracking, which can now be achieved in the digital domain. The complexity previously associated with coherent detection is hence significantly diminished and coherent detection is once again considered a feasible detection alternative. In this thesis, several aspects of coherent detection (with or without subsequent DSP) are addressed. Coherent detection is presented as a means to extend the dispersion limit of a duobinary signal using an analog decision‐directed phase‐lock loop. Analytical bit‐error ratio estimation for quadrature phase‐shift keying signals is derived. To validate the promise for high spectral efficiency, the orthogonal‐wavelength‐division multiplexing scheme is suggested. In this scheme the WDM channels are spaced at the symbol rate, thus achieving the spectral efficiency limit. Theory, simulation and experimental results demonstrate the feasibility of this approach. Infinite impulse response filtering is shown to be an efficient alternative to finite impulse response filtering for chromatic dispersion compensation. Theory, design considerations, simulation and experimental results relating to this topic are presented. Interaction between fiber dispersion and nonlinearity remains the last major challenge deterministic effects pose for long‐haul optical data transmission. Experimental results which demonstrate the possibility to digitally mitigate both dispersion and nonlinearity are presented. Impairment compensation is achieved using backward propagation by implementing the split‐step method. Efficient realizations of the dispersion compensation operator used in this implementation are considered. Infinite‐impulse response and wavelet‐based filtering are both investigated as a means to reduce the required computational load associated with signal backward‐propagation. Possible future research directions conclude this dissertation.<br>Ph.D.<br>Optics and Photonics<br>Optics and Photonics<br>Optics PhD
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Jha, Sanjay Kumar. "Artificial neural networks for digital signal processing applications." Thesis, University of Strathclyde, 1990. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.284097.

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Orduna-Bustamante, Felipe. "Digital signal processing for multi-channel sound reproduction." Thesis, University of Southampton, 1995. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.261565.

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Sheet, Lenny. "Noise measurement to 40PPM using digital signal processing." Thesis, Massachusetts Institute of Technology, 1990. http://hdl.handle.net/1721.1/26832.

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Singer, Amy M. (Amy Michelle). "Top-down design of digital signal processing systems." Thesis, Massachusetts Institute of Technology, 1996. http://hdl.handle.net/1721.1/40000.

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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1996.<br>Includes bibliographical references (leaves 45-46).<br>by Amy M. Singer.<br>M.Eng.
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Erk, Patrick P. (Patrick Peter). "Digital signal processing techniques for laser-doppler anemometry." Thesis, Massachusetts Institute of Technology, 1990. http://hdl.handle.net/1721.1/43026.

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Millar, D. S. "Digital signal processing for coherent optical fibre communications." Thesis, University College London (University of London), 2011. http://discovery.ucl.ac.uk/1333248/.

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In this thesis investigations were performed into digital signal processing (DSP) algorithms for coherent optical fibre transmission systems, which provide improved performance with respect to conventional systems and algorithms. Firstly, an overview of coherent detection and coherent transmission systems is given. Experimental investigations were then performed into the performance of digital backpropagation for mitigating fibre nonlinearities in a dual-polarization quadrature phase shift keying (DP-QPSK) system over 7780 km and a dual-polarization 16- level quadrature amplitude modulation (DP-QAM16) system over 1600 km. It is noted that significant improvements in performance may be achieved for a nonlinear step-size greater than one span. An approximately exponential relationship was found between performance improvement in Q-factor and the number for required complex multipliers. DSP algorithms for polarization-switched quadrature phase shift keying (PS-QPSK) are then investigated. A novel two-part equalisation algorithm is proposed which provides singularity-free convergence and blind equalisation of PS-QPSK. This algorithm is characterised and its application to wavelength division multiplexed (WDM) transmission systems is discussed. The thesis concludes with an experimental comparison between a PS-QPSK transmission system and a conventional DP-QPSK system. For a 42.9 Gb/s WDM system, the use of PS-QPSK enabled an increase of reach of more than 30%. The resultant reach of 13,640 km was, at the time of publication, the longest transmission distance reported for 40 Gb/s transmission over an uncompensated link with standard fibre and optical amplification.
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Wahab, Lena. "Poisson relationships with applications to digital signal processing." Thesis, Imperial College London, 2007. http://hdl.handle.net/10044/1/11345.

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Eghbali, Amir. "Contributions to Flexible Multirate Digital Signal Processing Structures." Licentiate thesis, Linköping : Department of Electrical Engineering, Linköping University, 2009. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-17182.

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39

Tran, Merry Thi. "Applications of Digital Signal Processing with Cardiac Pacemakers." PDXScholar, 1992. https://pdxscholar.library.pdx.edu/open_access_etds/4582.

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Because the voltage amplitude of a heart beat is small compared to the amplitude of exponential noise, pacemakers have difficulty registering the responding heart beat immediately after a pacing pulse. This thesis investigates use of digital filters, an inverse filter and a lowpass filter, to eliminate the effects of exponential noise following a pace pulse. The goal was to create a filter which makes recognition of a haversine wave less dependent on natural subsidence of exponential noise. Research included the design of heart system, pacemaker, pulse generation, and D sensor system simulations. The simulation model includes the following components: \ • Signal source, A MA TLAB generated combination of a haversine signal, exponential noise, and myopotential noise. The haversine signal is a test signal used to simulate the QRS complex which is normally recorded on an ECG trace as a representa tion of heart function. The amplitude is approximately 10 mV. Simulated myopotential noise represents a uniformly distributed random noise which is generated by skeletal muscle tissue. The myopotential noise has a frequency spectrum extending from 70 to 1000Hz. The amplitude varies from 2 to 5 mV. Simulated exponential noise represents the depolarization effects of a pacing pulse as seen at the active cardiac lead. The amplitude is about -1 volt, large in comparison with the haversine signal. • AID converter, A combination of sample & hold and quantizer functions translate the analog signal into a digital signal. Additionally, random noise is created during quantization. • Digital filters, An inverse filter removes the exponential noise, and a lowpass filter removes myopotential noise. • Threshold level detector, A function which detects the strength and amplitude of the output signal was created for robustness and as a data sampling device. The simulation program is written for operation in a DOS environment. The program generates a haversine signal, myopotential noise (random noise), and exponential noise. The signals are amplified and sent to an AID converter stage. The resultant digital signal is sent to a series of digital filters, where exponential noise is removed by an inverse digital filter, and myopotential noise is removed by the Chebyshev type I lowpass digital filter. The output signal is "detected" if its waveform exceeds the noise threshold level. To determine what kind of digital filter would remove exponential noise, the spectrum of exponential noise relative to a haversine signal was examined. The spectrum of the exponential noise is continuous because the pace pulse is considered a non-periodic signal (assuming the haversine signal occurs immediately after a pace pulse). The spectrum of the haversine is also continuous, existing at every value of frequency co. The spectrum of the haversine is overlapped by the spectrum of and amplitude of the exponential, which is several orders of magnitude larger. The exponential cannot be removed by conventional filters. Therefore, an inverse filter approach is used to remove exponential noise. The transfer function of the inverse filter of the model has only zeros. This type of filter is called FIR, all-zero, non recursive, or moving average. Tests were run using the model to investigate the behavior of the inverse filter. It was found that the haversine signal could be clearly detected within a 5% change in the time constant of the exponential noise. Between 5% and 15% of change in the time constant, the filtered exponential amplitude swamps the haversine signal. The sensitivity of the inverse filter was also studied: when using a fixed exponential time constant but changing the location of the transfer function, the effect of the exponential noise on the haversine is minimal when zeros are located between 0.75 and 0.85 of the unit circle. After the source signal passes the inverse filter, the signal consists only of the haversine signal, myopotential noise, and some random noise introduced during quantization. To remove these noises, a Chebyshev type I lowpass filter is used.
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40

Saab, Sami James. "Design of digital filters for low-power digital signal processing applications." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1999. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape8/PQDD_0001/MQ41376.pdf.

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41

Hartley, David Andrew. "Image correlation using digital signal processors." Thesis, Liverpool John Moores University, 1991. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.304465.

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42

Peng, Dongming. "Exploiting parallelism within multidimensional multirate digital signal processing systems." Diss., Texas A&M University, 2003. http://hdl.handle.net/1969/141.

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43

Papenfuss, Frank. "Digital signal processing of nonuniform sampled signals contributions to algorithms & hardware architectures." Aachen Shaker, 2007. http://d-nb.info/987695959/04.

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44

Sherkat, Mohammad Reza. "Co-design of analog to digital interface and digital signal processing architecture /." The Ohio State University, 2000. http://rave.ohiolink.edu/etdc/view?acc_num=osu1488202678774616.

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45

Landqvist, Ronnie. "Signal processing techniques in mobile communication systems : signal separation, channel estimation and equalization /." Karlskrona : Blekinge Institute of Technology, 2005. http://www.bth.se/fou/Forskinfo.nsf/allfirst2/98bf8bfb44d67d86c1257099003e2fc1?OpenDocument.

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46

Ng, Chiu-wa. "Bit-stream signal processing on FPGA." Click to view the E-thesis via HKUTO, 2009. http://sunzi.lib.hku.hk/hkuto/record/B41633842.

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47

Wukitch, Matthew J. "Using Commercial Off the Shelf (COTS) Digital Signal Processors (DSP) for reliable space based Digital Signal Processing." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2001. http://handle.dtic.mil/100.2/ADA391766.

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Thesis (M.S. in Electrical Engineering) Naval Postgraduate School, March 2001.<br>Thesis advisors, Herschel H. Loomis, Alan A. Ross. Includes bibliographical references (p.97-98). Also available online.
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48

Myjak, Mitchell John. "A medium-grain reconfigurable architecture for digital signal processing." Online access for everyone, 2006. http://www.dissertations.wsu.edu/Dissertations/Spring2006/m%5Fmyjak%5F042706.pdf.

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49

Nader, Charles. "Enhancing Radio Frequency System Performance by Digital Signal Processing." Licentiate thesis, University of Gävle, Department of Electronics, Mathematics and Natural Sciences, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:hig:diva-7312.

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<p>In this thesis measurement systems for the purpose of characterization of radio frequency power amplifiers are studied. Methods to increase the speed, accuracy, bandwidth, as well as to reduce the sampling requirements and testing cost are presented. A method intended for signal shaping with respect to peak to-average ratio reduction and its effects-improvements on the radio frequency front-end performance is investigated.</p><p>A time domain measurement system intended for fast and accurate measurements and characterization of radio frequency power amplifiers is discussed. An automated, fast and accurate technique for power and frequency sweep measurements is presented. Multidimensional representation of measured figure of merits is evaluated for its importance on the production-testing phase of power amplifiers.</p><p>A technique to extend the digital bandwidth of a measurement system is discussed. It is based on the Zhu-Frank generalized sampling theorem which decreases the requirements on the sampling rate of the measurement system. Its application for power amplifiers behavioral modeling is discussed and evaluated experimentally.</p><p>A general method for designing multitone for the purpose of out-of-band characterization of nonlinear radio frequency modules using harmonic sampling is presented. It has an application with the validation of power amplifiers behavioral models in their out-of-band frequency spectral support when extracted from undersampled data.</p><p>A method for unfolding the frequency spectrum of undersampled wideband signals is presented. It is of high relevance to state-of-the-art radio frequency measurement systems which capture repetitive waveform based on a sampling rate that violates the Nyquist constraint. The method is presented in a compact form, it eliminates ambiguities caused by folded frequency spectra standing outside the Nyquist band, and is relevant for calibration matters.</p><p>A convex optimization reduction-based method of peaks-to-average ratio of orthogonal frequency division multiplexing signals is presented and experimentally validated for a wireless local area network system. Improvements on the radio frequency power amplifier level are investigated with respect to power added efficiency, output power, in-band and out-of-band errors. The influence of the power distribution in the excitation signal on power amplifier performance was evaluated.</p>
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50

Le, Son Thai. "Advanced digital signal processing for coherent optical OFDM transmissions." Thesis, Aston University, 2016. http://publications.aston.ac.uk/28889/.

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Coherent optical orthogonal frequency division multiplexing (CO-OFDM) has been actively considered as a potential candidate for long-haul transmission and 400 Gb/s to 1 Tb/s Ethernet transport because of its high spectral efficiency, efficient implementation, flexibility and robustness against linear impairments such as chromatic dispersion and polarization mode dispersion. However, due to the long symbol duration and narrow subcarrier spacing, CO-OFDM systems are sensitive to laser phase noise and fibre nonlinearity induced penalties. As a result, the development of CO-OFDM transmission technology crucially relies on efficient techniques to compensate for the laser phase noise and fibre nonlinearity impairments. In this thesis, high performance and low complexity digital signal processing techniques for laser phase noise and fibre nonlinearity compensation in CO-OFDM transmissions are demonstrated. For laser phase noise compensation, three novel techniques, namely quasipilot-aided, decision-directed-free blind and multiplier-free blind are introduced. For fibre nonlinear compensation, two novel techniques which are referred to as phase conjugated pilots and phase conjugated subcarrier coding, are proposed. All these abovementioned digital signal processing techniques offer high performances and flexibilities while requiring relatively low complexities in comparison with other existing phase noise and nonlinear compensation techniques. As a result of the developments of these digital signal processing techniques, CO-OFDM technology is expected to play a significant role in future ultra-high capacity optical network. In addition, this thesis also presents preliminary study on nonlinear Fourier transform based transmission schemes in which OFDM is a highly suitable modulation format. The obtained result paves the way towards a truly flexible nonlinear wave-division multiplexing system that allows the current nonlinear transmission limitations to be exceeded.
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