Dissertationen: „Signal processing Digital techniques“

1

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.

2

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.

3

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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4

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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5

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'

6

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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7

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.
Ph.D.
Optics and Photonics
Optics and Photonics
Optics PhD

8

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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9

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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10

Lei, Chi-un, und 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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11

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.

12

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).

13

Keeton, Paul Ivan John. „Modern digital signal processing techniques applied to Doppler ultrasound“. Thesis, University of Leicester, 1997. http://hdl.handle.net/2381/30188.

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Doppler ultrasound is used clinically to detect stenosis in the carotid artery. The presence of stenosis may be identified by disturbed flow patterns distal to the stenosis which cause spectral broadening in the spectrum of the Doppler signal around peak systole. This thesis investigates the ability of the short-time Fourier transform (STFT) and the autoregressive (AR) spectral estimators to perform time-frequency analysis of the non-stationary Doppler signal. Quantitative analysis of the degree of spectral broadening was measured using the spectral broadening index (SBI). A real-time system was developed using a modern DSP board combined with an IBM PC-compatible computer to analyse the Doppler signal in real-time using the STFT and AR algorithms. The spectral estimators were compared using simulated Doppler spectra contaminated with noise over a range of signal-to-noise ratios (SNRs) and also real clinical Doppler signals recorded from both healthy subjects and patients with varying degrees of stenosis. The SBI was calculated using the mean and maximum frequency envelopes which were extracted from the STFT and AR sonograms using a threshold at -6 dB of the maximum component of each individual spectrum. The results of the analysis shows a strong correlation between the indices calculated using the FFT and AR algorithms. A qualitative improvement in both the appearance of the AR sonograms and the shape of the individual AR spectra was noticeable, however, the estimation of SBI for short data frames is not significantly improved using AR. The final section of this thesis describes the wavelet transform (WT) and illustrates its application to Doppler ultrasound with two examples. Firstly, it is shown how wavelets can be used as an alternative to the STFT for the extraction of the time-frequency distribution of Doppler ultrasound signals. Secondly, wavelet-based adaptive filtering is implemented for the extraction of maximum blood velocity envelopes in the post processing of Doppler signals.

14

Mirsalehi, Mir Mojtaba. „Optical digital parallel truth-table look-up processing“. Diss., Georgia Institute of Technology, 1985. http://hdl.handle.net/1853/15013.

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15

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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16

Zhu, Yong. „Digital signal and image processing techniques for ultrasonic nondestructive evaluation“. Thesis, City University London, 1996. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.336431.

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17

Dietl, Hubert. „Digital signal processing techniques for detection applied to biomedical data“. Thesis, University of Southampton, 2005. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.419141.

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18

Canagarajah, Cedric Nishanthan. „Digital signal processing techniques for speech enhancement in hearing aids“. Thesis, University of Cambridge, 1993. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.260433.

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19

Nagashima, Yoshihiro. „Digital signal processing techniques for the measurement of ocular counterrolling“. Thesis, Massachusetts Institute of Technology, 1985. http://hdl.handle.net/1721.1/83657.

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Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Aeronautics and Astronautics, 1985.
Microfiche copy available in Archives and Barker.
Vita.
Includes bibliographical references.
by Yoshihiro Nagashima.
M.S.

20

PILORI, DARIO. „Advanced Digital Signal Processing Techniques for High-Speed Optical Links“. Doctoral thesis, Politecnico di Torino, 2019. http://hdl.handle.net/11583/2729814.

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21

Hadi, Muhammad Usman <1992&gt. „Digital Signal Processing Techniques Applied to Radio over Fiber Systems“. Doctoral thesis, Alma Mater Studiorum - Università di Bologna, 2020. http://amsdottorato.unibo.it/9155/1/PhD_Thesis_ETIT_Feb_DRAFT.pdf.

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The dissertation aims to analyze different Radio over Fiber systems for the front-haul applications. Particularly, analog radio over fiber (A-RoF) are simplest and suffer from nonlinearities, therefore, mitigating such nonlinearities through digital predistortion are studied. In particular for the long haul A-RoF links, direct digital predistortion technique (DPDT) is proposed which can be applied to reduce the impairments of A-RoF systems due to the combined effects of frequency chirp of the laser source and chromatic dispersion of the optical channel. Then, indirect learning architecture (ILA) based structures namely memory polynomial (MP), generalized memory polynomial (GMP) and decomposed vector rotation (DVR) models are employed to perform adaptive digital predistortion with low complexities. Distributed feedback (DFB) laser and vertical capacity surface emitting lasers (VCSELs) in combination with single mode/multi-mode fibers have been linearized with different quadrature amplitude modulation (QAM) formats for single and multichannel cases. Finally, a feedback adaptive DPD compensation is proposed. Then, there is still a possibility to exploit the other realizations of RoF namely digital radio over fiber (D-RoF) system where signal is digitized and transmits the digitized bit streams via digital optical communication links. The proposed solution is robust and immune to nonlinearities up-to 70 km of link length. Lastly, in light of disadvantages coming from A-RoF and D-RoF, it is still possible to take only the advantages from both methods and implement a more recent form knows as Sigma Delta Radio over Fiber (S-DRoF) system. Second Order Sigma Delta Modulator and Multi-stAge-noise-SHaping (MASH) based Sigma Delta Modulator are proposed. The workbench has been evaluated for 20 MHz LTE signal with 256 QAM modulation. Finally, The 6x2 GSa/s sigma delta modulators are realized on FPGA to show a real time demonstration of S-DRoF system. The demonstration shows that S-DRoF is a competitive competitor for 5G sub-6GHz band applications.

22

Ng, Chiu-wa, und 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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23

Malassenet, Francois Jacques. „Self-Affine signals and weighted multiresolution processes“. Diss., Georgia Institute of Technology, 1991. http://hdl.handle.net/1853/14914.

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24

DeBardelaben, James Anthony. „An optimization-based approach for cost-effective embedded DSP system design“. Diss., Georgia Institute of Technology, 1998. http://hdl.handle.net/1853/15757.

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25

Armstrong, Richard Paul. „High-performance signal processing architectures for digital aperture array telescopes“. Thesis, University of Oxford, 2011. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.560917.

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An instrument with the ability to image neutral atomic hydrogen (HI) to cosmic redshift will allow the fundamental properties of the Universe to be more precisely determined; in particular the distribution, composition, and evolutionary history of its matter and energy. The Square Kilometre Array (SKA) is a radio survey telescope conceived with this aim. It will have the observational potential for much further fundamental science, including strong field tests of gravity and general relativity, revealing the origin and history of cosmological re-ionisation and magnetism, direct measures of gravitational radiation, and surveys of the unmapped Universe. And it is the advance of instrumentation that will enable it. This thesis makes three central contributions to radio instrumentation. Digital aperture arrays are a collector technology proposed for the key low- and mid- frequency ranges targeted by the SKA that have the potential to provide both the collecting area and field of view required for deep, efficient all-sky surveys of HI. The 2-Polarisations, All Digital (2-PAD) aperture array is an instrumental pathfinder for the SKA, novel in being a densely-spaced, wide-band aperture array that performs discrete signal filtering entirely digitally. The digital design of the 2-PAD radio receiver and the deployment of the aperture array and signal processing system at Jodrell Bank Radio Observatory is detailed in this thesis. The problem of element anisotropy in small arrays, the atomic unit of the SKA station array, ultimately affects beam quality. Addressing this issue, a metaheuristic digital beam-shape optimisation technique is applied to a small beamformed array, and is shown to outperform traditional analytic solutions. Digital processing for aperture arrays is challenging. A qualitative framework shows that energy, computational and communication requirements demand optimised processing architectures. A quantitative model reveals the physical limitations on architecture choice. An energy-optimised architecture, the IBM BIT integer array processor, is investigated in detail; a cycle-accurate architectural simulator and programming language are developed and used to build signal processing algorithms on the array architecture.

26

Amphlett, Robert W. „Multiprocessor techniques for high quality digital audio“. Thesis, University of Bristol, 1996. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.337273.

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27

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.

28

Rosenthal, Jordan. „Filters and filterbanks for hexagonally sampled signals“. Diss., Georgia Institute of Technology, 2001. http://hdl.handle.net/1853/13347.

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29

Colzi, Enrico. „Digital signal processing techniques for personal and broadcasting satellite communiction systems /“. Noordwijk : ESA, 1999. http://www.gbv.de/dms/goettingen/303785233.pdf.

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30

Frangakis, G. P. „Digital and microprocessor-based techniques in signal processing and system simulation“. Thesis, University of Southampton, 1985. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.370338.

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31

Ristic, Branko. „Some aspects of signal dependent and higher-order time-frequency and time-scale analysis of non-stationary signals“. Thesis, Queensland University of Technology, 1995.

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32

Morris, Robert W. „Enhancement and recognition of whispered speech“. Diss., Available online, Georgia Institute of Technology, 2004:, 2003. http://etd.gatech.edu/theses/available/etd-04082004-180338/unrestricted/morris%5frobert%5fw%5f200312%5fphd.pdf.

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33

De, Subrato Kumar. „Design of a retargetable compiler for digital signal processors“. Diss., Georgia Institute of Technology, 2002. http://hdl.handle.net/1853/15740.

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34

Natali, Francis D., und Gerard G. Socci. „DIGITAL RECEIVER PROCESSING TECHNIQUES FOR SPACE VEHICLE DOWNLINK SIGNALS“. International Foundation for Telemetering, 1985. http://hdl.handle.net/10150/615711.

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International Telemetering Conference Proceedings / October 28-31, 1985 / Riviera Hotel, Las Vegas, Nevada
Digital processing techniques and related algorithms for receiving and processing space vehicle downlink signals are discussed. The combination of low minimum signal to noise density (C/No), large signal dynamic range, unknown time of arrival, and high space vehicle dynamics that is characteristic of some of these downlink signals results in a difficult acquisition problem. A method for rapid acquisition is described which employs a Fast Fourier Transform (FFT). Also discussed are digital techniques for precise measurement of space vehicle range and range rate using a digitally synthesized number controlled oscillator (NCO).

35

Ng, Wing Chau. „Digital signal processing algorithms in single-carrier optical coherent communications“. Doctoral thesis, Université Laval, 2015. http://hdl.handle.net/20.500.11794/26102.

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Des systèmes de détection cohérente avec traitement numérique du signal (DSP) sont présentement déployés pour la transmission optique de longue portée. La modulation par déplacement de phase en quadrature à deux polarisations (DP-QPSK) est une forme de modulation appropriée pour la transmission optique sur de longues distances (1000 km ou plus). Une autre forme de modulation, le DP-16-QAM (modulation d’amplitude en quadrature) a été récemment utilisée pour les communications métropolitaines (entre 100 et 1000 km). L’extension de la distance maximum de transmission du DP-16-QAM est un domaine de recherche actif. Déterminer si l’utilisation de la détection cohérente pour les transmissions à courtes distances (moins de 100 km) en justifieraient les coûts demeure cependant une question ouverte. Dans cette thèse, nous nous intéresserons principalement au recouvrement de phase et au démultiplexage en polarisation dans les récepteurs numériques cohérents pour les applications à courte distance. La réalisation de systèmes optiques gigabauds cohérents en temps-réel utilisant des formats de modulation à monoporteuse plus complexes, comme le 64-QAM, dépend fortement du recouvrement de phase. Pour le traitement numérique hors-ligne, la récupération de phase utilisant les résultats de décisions (decision-directed phase recovery (DD-PR)) permet d’obtenir, au débit des symboles, les meilleures performances, et ce avec un effort computationnel moindre que celui des meilleurs algorithmes connus. L’implémentation en temps-réel de systèmes gigabauds requiert un haut degré de parallélisation qui dégrade de manière significative les performances de cet algorithme. La parallélisation matérielle et le délais de pipelinage sur la boucle de rétroaction imposent des contraintes strictes sur la largeur spectrale du laser, ainsi que sur le niveau de bruit spectral des sources laser. C’est pourquoi on retrouve peu de démonstrations de recouvrement de phase en temps-réel pour les modulations 64-QAM ou plus complexes. Nous avons analysé expérimentalement l’impact des lasers avec filtres optiques sur le recouvrement de phase realisé en pipeline sur un système cohérent à monoporteuse 64-QAM à 5 Gbaud. Pour les niveaux de parallélisation plus grands que 24, le laser avec filtres optiques a permis une amélioration de 2 dB du ratio signal-à-bruit optique, en comparaison avec le même laser sans filtre optique. La parallélisation du recouvrement de phase entraîne non seulement une plus grande sensibilité au bruit de phase du laser, mais aussi une plus grande sensibilité aux fréquences résiduelles induites par la présence de tonalités sinusoïdales dans la source. La modulation de fréquences sinusoïdales peut être intentionnelle, pour des raisons de contrôle, ou accidentelles, dues à l’électronique ou aux fluctuations environnementales. Nous avons étudié expérimentalement l’impact du bruit sinusoïdal de phase du laser sur le système parallèle de recouvrement de phase dans un système 64-QAM à 5 Gbauds, en tenant compte des effets de la compensation du décalage de fréquence et de l’égalisation. De nos jours, les filtres MIMO (multi-input multi-output) à réponse finie (FIR) sont couramment utilisés pour le démultiplexage en polarisation dans les systèmes cohérents. Cependant, ces filtres souffrent de divers problèmes durant l’acquisition, tels la singularité (les mêmes données apparaissent dans les deux canaux de polarisation) et la longue durée de convergence de certaines combinaisons d’états de polarisation (SOP). Pour réduire la consommation d’énergie exigée dans les systèmes cohérents pour les applications à courtes distances où le délais de groupe différentiel n’est pas important, nous proposons une architecture DSP originale. Notre approche introduit une pré-rotation de la polarisation, avant le MIMO, basée sur une estimation grossière de l’état de polarisation qui n’utilise qu’un seul paramètre Stokes (s1). Cette méthode élimine les problèmes de singularité et de convergence du MIMO classique, tout en réduisant le nombre de filtres MIMO croisés, responsables de l’annulation de la diaphonie de polarisation. Nous présentons expérimentalement un compromis entre la réduction de matériel et la dégradation des performances en présence de dispersion chromatique résiduelle, afin de permettre la réalisation d’applications à courtes distances. Finalement, nous améliorons notre méthode d’estimation à l’aveugle par un filtre Kalman étendu (EKF) à temps discret de faible complexité, afin de réduire la consommation de mémoire et les calculs redondants apparus dans la méthode précédante. Nous démontrons expérimentalement que la pré-rotation de polarisation basée sur le EKF operé au taux ASIC (Application-Specific Integrated Circuits) permet de récupérer la puissance de fréquence d’horloge du signal multiplexé en polarisation ainsi que d’améliorer la performance du taux d’erreur sur les bits (BER) en utilisant un MIMO de complexité réduite.
Coherent detection with digital signal processing (DSP) is currently being deployed in longhaul optical communications. Dual-polarization (DP) quadrature phase shift keying (QPSK) is a modulation format suitable for long-haul transmission (1000 km or above). Another modulation, DP-16-QAM (quadrature amplitude modulation) has been deployed recently in metro regions (between 100 and 1000 km). Extending the reach of DP-16QAM is an active research area. For short-reach transmission (shorter than 100 km), there is still an open question as to when the technology will be mature enough to meet cost pressures for this distance. In this dissertation, we address mainly on phase recovery and polarization demultiplexing in digital coherent receivers for short-reach applications. Implementation of real-time Gbaud (Gsymbol per second) optical coherent systems for singlecarrier higher-level modulation formats such as 64-QAM depends heavily on phase tracking. For offline DSP, decision-directed phase recovery is performed at the symbol rate with the best performance and the least computational effort compared to best-known algorithms. Real-time implementations at Gbaud requires significant parallelizing that greatly degrades performance of this algorithm. Hardware parallelization and pipelining delay on the feedback path impose stringent requirements on the laser linewidth, or the frequency noise spectral level of laser sources. This leads to the paucity of experiments demonstrating real-time phase tracking for 64- or higher QAM. We experimentally investigated the impact of opticallyfiltered lasers on parallel and pipelined phase tracking in a single-carrier 5 Gbaud 64-QAM back-to-back coherent system. For parallelization levels higher than 24, the optically-filtered laser shows more than 2 dB improvement in optical signal-to-noise ratio penalty compared to that of the same laser without optical filtering. In addition to laser phase noise, parallelized phase recovery also creates greater sensitivity to residual frequency offset induced by the presence of sinusoidal tones in the source. Sinusoidal frequency modulation may be intentional for control purposes, or incidental due to electronics and environmental fluctuations. We experimentally investigated the impact of sinusoidal laser phase noise on parallel decision-directed phase recovery in a 5 Gb 64-QAM system, including the effects of frequency offset compensation and equalization. MIMO (multi-input multi-output) FIR (finite-impulse response) filters are conventionally used for polarization demultiplexing in coherent communication systems. However, MIMO FIRs suffer from acquisition problems such as singularity and long convergence for a certain polarization rotations. To reduce the chip power consumption required in short-reach coherent systems where differential group delay is not prominent, we proposed a novel parallelizable DSP architecture. Our approach introduces a polarization pre-rotation before MIMO, based on a very-coarse blind SOP (state of polarization) estimation using only a single Stokes parameter (s1). This method eliminates the convergence and singularity problems of conventional MIMO, and reduces the number of MIMO cross taps responsible for cancelling the polarization crosstalk. We experimentally presented a tradeoff between hardware reduction and performance degradation in the presence of residual chromatic dispersion for short-reach applications. Finally, we extended the previous blind SOP estimation method by using a low-complexity discrete-time extended Kalman filter in order to reduce the memory depth and redundant computations of the previous design. We experimentally verified that our extended Kalman filter-based polarization prerotation at ASIC rates enhances the clock tone of polarization-multiplexed signals as well as the bit-error rate performance of using reduced-complexity MIMO for polarization demultiplexing.

36

Multanen, Eric W. „Characterization of quantization noise in oversampled analog to digital converters“. PDXScholar, 1992. https://pdxscholar.library.pdx.edu/open_access_etds/4424.

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The analog to digital converter (ADC) samples a continuous analog signal and produces a stream of digital words which approximate the analog signal. The conversion process introduces noise into the digital signal. In the case of an ideal ADC, where all noise sources are ignored, the noise due to the quantization process remains. The resolution of the ADC is defined by how many bits are in the digital output word. The amount of quantization noise is clearly related to the resolution of the ADC. Reducing the quantization noise results in higher effective resolution.

37

陳力 und Li Chen. „Design of linear phase paraunitary filter banks and finite length signal processing“. Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 1997. http://hub.hku.hk/bib/B31235608.

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38

Jen, Kwang-Suz. „Development of experiments for the digital signal processing teaching laboratory“. Thesis, Virginia Tech, 1988. http://hdl.handle.net/10919/45166.

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Digital Signal Processing (DSP) is a technology-driven field which develops as early as mid-1960 when computers and other digital circuitry became fast enough to process large amounts of data efficiently. Since then techniques and applications of DSP have been expanding at a tremendous rate. With the development of large-scale integration, the cost and size of digital components are reducing, and speed of digital components is increasing. Thus the range of applications of DSP techniques is growing. Almost all current discussions of speech bandwidth compression systems are directed toward digital implementation, because these are now the most practical. The importance of DSP appears to be increasing with no visible signs of saturation.

This thesis provides the description and results of designing laboratory experiments for the illustration of basic theory in the field of DSP. All experiments are written for the Texas Instruments TMS320I0 digital signal processing microcomputer and based on softwares provided by Atlanta Signal Process, Inc. (ASPI). The use of the 320/pc Algorithm Development Package (ADP) and Digital Filter Design Package (DFDP) developed by ASPI is introduced. The basic concepts, such as linear convolution, Finite Impulse Response (FIR) and Infinite Impulse Response (IIR) filter design, Fast Fourier Transform (FF1), are demonstrated. The IBM PC AT is interfaced with the TMS32010 processor. The experiments and their introductions in the thesis also serve as a manual for the DSP Laboratory; to complement the introductory signal processing course.
Master of Science

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Khokhar, Khawar Siddique. „Design and development of mobile channel simulators using digital signal processing techniques“. Thesis, Durham University, 2006. http://etheses.dur.ac.uk/2948/.

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A mobile channel simulator can be constructed either in the time domain using a tapped delay line filter or in the frequency domain using the time variant transfer function of the channel. Transfer function modelling has many advantages over impulse response modelling. Although the transfer function channel model has been envisaged by several researchers as an alternative to the commonly employed tapped delay line model, so far it has not been implemented. In this work, channel simulators for single carrier and multicarrier OFDM system based on time variant transfer function of the channel have been designed and implemented using DSP techniques in SIMULINK. For a single carrier system, the simulator was based on Bello's transfer function channel model. Bello speculated that about 10Βτ(_MAX) frequency domain branches might result in a very good approximation of the channel (where в is the signal bandwidth and τ(_MAX) is the maximum excess delay of the multi-path channel). The simulation results showed that 10Bτ(_MAX) branches gave close agreement with the tapped delay line model(where Be is the coherence bandwidth). This number is π times higher than the previously speculated 10Bτ(_MAX).For multicarrier OFDM system, the simulator was based on the physical (PHY) layer standard for IEEE 802.16-2004 Wireless Metropolitan Area Network (WirelessMAN) and employed measured channel transfer functions at the 2.5 GHz and 3.5 GHz bands in the simulations. The channel was implemented in the frequency domain by carrying out point wise multiplication of the spectrum of OFDM time The simulator was employed to study BER performance of rate 1/2 and rate 3/4 coded systems with QPSK and 16-QAM constellations under a variety of measured channel transfer functions. The performance over the frequency selective channel mainly depended upon the frequency domain fading and the channel coding rate.

40

Allay, Najib. „Application of nonuniform sampling techniques in digital signal processing and communication systems“. Thesis, University of Westminster, 2006. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.433854.

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41

Ziegler, Frank A. „Digital Signal Processing Techniques Used to Demodulate Multiple Types of Telemetry Data“. International Foundation for Telemetering, 1990. http://hdl.handle.net/10150/613766.

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International Telemetering Conference Proceedings / October 29-November 02, 1990 / Riviera Hotel and Convention Center, Las Vegas, Nevada
Telemetry systems today are required to receive a variety of modulation formats. Typically, to change the format required changing the demodulator unit or large switching systems. Using some common digital building blocks and multiplexers, the user can change demodulation mode by pressing a button. This paper describes a system that demodulates PM, FM, BPSK, QPSK and DSB AM.

42

Yamashita, Fumihiro. „Study on digital signal processing techniques for high-scalable mobile satellite communications“. 京都大学 (Kyoto University), 2006. http://hdl.handle.net/2433/143957.

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43

Pessoa, Lucio Flavio Cavalcanti. „Nonlinear systems and neural networks with hybrid morphological/rank/linear nodes : optimal design and applications to image processing and pattern recognition“. Diss., Georgia Institute of Technology, 1997. http://hdl.handle.net/1853/13519.

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44

Liu, Sam J. „Low bit-rate image and video compression using adaptive segmentation and quantization“. Diss., Georgia Institute of Technology, 1993. http://hdl.handle.net/1853/14850.

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45

Arrowood, Joseph Louis Jr. „Theory and application of adaptive filter banks“. Diss., Georgia Institute of Technology, 1999. http://hdl.handle.net/1853/15369.

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46

姚佩雯 und Pui-man Yiu. „Multiplier-less sinusoidal transformations and their applications to perfect reconstruction filter banks“. Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2002. http://hub.hku.hk/bib/B31228045.

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47

Hereford, James McCracken. „Optical implementation of morphological transformations“. Diss., Georgia Institute of Technology, 1990. http://hdl.handle.net/1853/14891.

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48

Lynch, Michael Richard. „Adaptive techniques in signal processing and connectionist models“. Thesis, University of Cambridge, 1990. https://www.repository.cam.ac.uk/handle/1810/244884.

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This thesis covers the development of a series of new methods and the application of adaptive filter theory which are combined to produce a generalised adaptive filter system which may be used to perform such tasks as pattern recognition. Firstly, the relevant background adaptive filter theory is discussed in Chapter 1 and methods and results which are important to the rest of the thesis are derived or referenced. Chapter 2 of this thesis covers the development of a new adaptive algorithm which is designed to give faster convergence than the LMS algorithm but unlike the Recursive Least Squares family of algorithms it does not require storage of a matrix with n2 elements, where n is the number of filter taps. In Chapter 3 a new extension of the LMS adaptive notch filter is derived and applied which gives an adaptive notch filter the ability to lock and track signals of varying pitch without sacrificing notch depth. This application of the LMS filter is of interest as it demonstrates a time varying filter solution to a stationary problem. The LMS filter is next extended to the multidimensional case which allows the application of LMS filters to image processing. The multidimensional filter is then applied to the problem of image registration and this new application of the LMS filter is shown to have significant advantages over current image registration methods. A consideration of the multidimensional LMS filter as a template matcher and pattern recogniser is given. In Chapter 5 a brief review of statistical pattern recognition is given, and in Chapter 6 a review of relevant connectionist models. In Chapter 7 the generalised adaptive filter is derived. This is an adaptive filter with the ability to model non-linear input-output relationships. The Volterra functional analysis of non-linear systems is given and this is combined with adaptive filter methods to give a generalised non-linear adaptive digital filter. This filter is then considered as a linear adaptive filter operating in a non-linearly extended vector space. This new filter is shown to have desirable properties as a pattern recognition system. The performance and properties of the new filter is compared with current connectionist models and results demonstrated in Chapter 8. In Chapter 9 further mathematical analysis of the networks leads to suggested methods to greatly reduce network complexity for a given problem by choosing suitable pattern classification indices and allowing it to define its own internal structure. In Chapter 10 robustness of the network to imperfections in its implementation is considered. Chapter 11 finishes the thesis with some conclusions and suggestions for future work.

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Chow, Wing Keung. „Applications of digital signal processing to real-time optical fibre holographic interferometry“. HKBU Institutional Repository, 1992. https://repository.hkbu.edu.hk/etd_ra/18.

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50

詹文達 und Man-tat Jimmy Tsim. „High speed realisation of digital filters“. Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 1989. http://hub.hku.hk/bib/B31208939.

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Dissertationen zum Thema „Signal processing Digital techniques“

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