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Title: Complex-valued adaptive digital signal enhancement for applications in wireless communication systems.
Creator: Liu, Ying, Mikhael, Wasfy, Batarseh, Issa, Yang, Thomas, Hunter, Matthew, Haralambous, Michael, Myers, Brent, University of Central Florida
Abstract / Description: In recent decades, the wireless communication industry has attracted a great deal of research efforts to satisfy rigorous performance requirements and preserve high spectral efficiency. Along with this trend, I/Q modulation is frequently applied in modern wireless communications to develop high performance and high data rate systems. This has necessitated the need for applying efficient complex-valued signal processing techniques to highly-integrated, multi-standard receiver devices.In this...
Show moreIn recent decades, the wireless communication industry has attracted a great deal of research efforts to satisfy rigorous performance requirements and preserve high spectral efficiency. Along with this trend, I/Q modulation is frequently applied in modern wireless communications to develop high performance and high data rate systems. This has necessitated the need for applying efficient complex-valued signal processing techniques to highly-integrated, multi-standard receiver devices.In this dissertation, novel techniques for complex-valued digital signal enhancement are presented and analyzed for various applications in wireless communications. The first technique is a unified block processing approach to generate the complex-valued conjugate gradient Least Mean Square (LMS) techniques with optimal adaptations. The proposed algorithms exploit the concept of the complex conjugate gradients to find the orthogonal directions for updating the adaptive filter coefficients at each iteration. Along each orthogonal direction, the presented algorithms employ the complex Taylor series expansion to calculate time-varying convergence factors tailored for the adaptive filter coefficients. The performance of the developed technique is tested in the applications of channel estimation, channel equalization, and adaptive array beamforming. Comparing with the state of the art methods, the proposed techniques demonstrate improved performance and exhibit desirable characteristics for practical use.The second complex-valued signal processing technique is a novel Optimal Block Adaptive algorithm based on Circularity, OBA-C. The proposed OBA-C method compensates for a complex imbalanced signal by restoring its circularity. In addition, by utilizing the complex Taylor series expansion, the OBA-C method optimally updates the adaptive filter coefficients at each iteration. This algorithm can be applied to mitigate the frequency-dependent I/Q mismatch effects in analog front-end. Simulation results indicate that comparing with the existing methods, OBA-C exhibits superior convergence speed while maintaining excellent accuracy. The third technique is regarding interference rejection in communication systems. The research on both LMS and Independent Component Analysis (ICA) based techniques continues to receive significant attention in the area of interference cancellation. The performance of the LMS and ICA based approaches is studied for signals with different probabilistic distributions. Our research indicates that the ICA-based approach works better for super-Gaussian signals, while the LMS-based method is preferable for sub-Gaussian signals. Therefore, an appropriate choice of interference suppression algorithms can be made to satisfy the ever-increasing demand for better performance in modern receiver design.
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Date Issued: 2012
Identifier: CFE0004572, ucf:49192
Format: Document (PDF)
PURL: http://purl.flvc.org/ucf/fd/CFE0004572

Title: Detection of DDH in Infants and Children Using Audible Acoustics.
Creator: Hassan, Tanvir, Mansy, Hansen, Song, Sang-Eun, Kassab, Alain, University of Central Florida
Abstract / Description: Detection of developmental dysplasia of the hip (DDH) in infants and children is important as it leads to permanent hip instability. Current methods for detecting DDH, such as ultrasound and x-rays, are relatively expensive and need qualified medical personnel to administer the test. Furthermore, x-ray ionizing radiation can have potential harmful effects. In the current study, an acoustic non-invasive and simple approach was investigated for detection of DDH. Different benchtop simplified...
Show moreDetection of developmental dysplasia of the hip (DDH) in infants and children is important as it leads to permanent hip instability. Current methods for detecting DDH, such as ultrasound and x-rays, are relatively expensive and need qualified medical personnel to administer the test. Furthermore, x-ray ionizing radiation can have potential harmful effects. In the current study, an acoustic non-invasive and simple approach was investigated for detection of DDH. Different benchtop simplified models and pig models were constructed and tested. Models were stimulated with band-limited white acoustic noise (10-2500 Hz) and the response of the models was measured. The power spectrum density, transfer function, and coherence were determined for different hip dysplasia levels and for normal cases. Results showed that the power spectrum density, transfer function, and coherence were affected by dysplasia occurrence. Effects appear larger for more severe dysplastic hips. This suggests that the proposed approach may have potential for DDH detection.
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Date Issued: 2019
Identifier: CFE0007816, ucf:52350
Format: Document (PDF)
PURL: http://purl.flvc.org/ucf/fd/CFE0007816

Title: A microcomputer implementation of real time, continuously programmable digital filters.
Creator: Storma, William Edward, Simmons, Fred O., Engineering
Abstract / Description: University of Central Florida College of Engineering Thesis; When a filter transfer function in s is replaced with the bilinear transform in z, t he resulting discrete model represents the original continuous model within a second order accuracy of integration. A unique set of recently discovered minimum memory algorithms that perform the bilinear transform on a continuous transfer function are implemented on an INTEL 8080 microprocessor system. Scal1ng techniques are used to frequency scale...
Show moreUniversity of Central Florida College of Engineering Thesis; When a filter transfer function in s is replaced with the bilinear transform in z, t he resulting discrete model represents the original continuous model within a second order accuracy of integration. A unique set of recently discovered minimum memory algorithms that perform the bilinear transform on a continuous transfer function are implemented on an INTEL 8080 microprocessor system. Scal1ng techniques are used to frequency scale all transfer functions to a standardized frequency. All data words are represented in a signed binary double precision format to maintain higher calculation speed and accuracy. Three test case transfer functions of different order are implemented using the bilinear transform algorithms. First, the algorithms are used to generate the three discrete models. Second, the continuous time models are driven by a step input function, generating a continuous time output. Third, the step function input is discretized and used to drive the bilinear algorithm derived models. Finally, the discrete outputs are compared with the continuous time outputs to validate and evaluate the software techniques used to implement the bilinear algorithms, which imply that the techniques provide a basis for new hardware designs.
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Date Issued: 1979
Identifier: CFR0003497, ucf:53140
Format: Document (PDF)
PURL: http://purl.flvc.org/ucf/fd/CFR0003497

Title: DIGITAL SIGNAL PROCESSING TECHNIQUES FOR COHERENT OPTICAL COMMUNICATION.
Creator: Goldfarb, Gilad, Li, Guifang, University of Central Florida
Abstract / Description: 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...
Show moreCoherent 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.
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Date Issued: 2008
Identifier: CFE0002384, ucf:47763
Format: Document (PDF)
PURL: http://purl.flvc.org/ucf/fd/CFE0002384

Title: Method for Real-Time Signal Selection for Passive Coherent Location Systems.
Creator: Johnson, Nicholas, Jones, W Linwood, Gong, Xun, Mikhael, Wasfy, Rockway, John, Lichtenberg, Christopher, University of Central Florida
Abstract / Description: Passive coherent location (PCL) systems use signals of opportunity to perform traditional radar detection, targeting, and tracking functions. Traditionally these signals include FM radio, digital TV, GSM, and GPS because of their availability in most urban environments. A benefit of having an abundance of signals is the ability to choose which of those best meet the desired system intentions. For example, one may want to choose a digital TV signal over an FM radio signal due to its range...
Show morePassive coherent location (PCL) systems use signals of opportunity to perform traditional radar detection, targeting, and tracking functions. Traditionally these signals include FM radio, digital TV, GSM, and GPS because of their availability in most urban environments. A benefit of having an abundance of signals is the ability to choose which of those best meet the desired system intentions. For example, one may want to choose a digital TV signal over an FM radio signal due to its range resolution characteristics. This work presents a novel algorithm for characterizing commercial signals for use in a PCL system. By analyzing each signal's ambiguity function in terms of amplitude, transmitter geometry, range and Doppler resolution, and sidelobe levels, a comparative evaluation can be made to decide which signals are best suited for an intended radar function. In addition, this research shows that multiple signals can be combined in the detection process to increase the probability of detection over that of a single signal. Finally, this research investigates the geometric considerations for PCL systems in terms of bistatic radar geometry. The results show zones of linear and non-linear relationships between time delay, range, and Doppler frequency.
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Date Issued: 2017
Identifier: CFE0007123, ucf:51964
Format: Document (PDF)
PURL: http://purl.flvc.org/ucf/fd/CFE0007123

Title: Mode-Division Multiplexed Transmission in Few-mode Fibers.
Creator: Bai, Neng, Li, Guifang, Christodoulides, Demetrios, Schulzgen, Axel, Abouraddy, Ayman, Phillips, Ronald, Ip, Ezra, University of Central Florida
Abstract / Description: As a promising candidate to break the single-mode fiber capacity limit, mode-division multiplexing (MDM) explores the spatial dimension to increase transmission capacity in fiber-optic communication. Two linear impairments, namely loss and multimode interference, present fundamental challenges to implementing MDM. In this dissertation, techniques to resolve these two issues are presented.To de-multiplex signals subject to multimode interference in MDM, Multiple-Input-Multiple-Output (MIMO)...
Show moreAs a promising candidate to break the single-mode fiber capacity limit, mode-division multiplexing (MDM) explores the spatial dimension to increase transmission capacity in fiber-optic communication. Two linear impairments, namely loss and multimode interference, present fundamental challenges to implementing MDM. In this dissertation, techniques to resolve these two issues are presented.To de-multiplex signals subject to multimode interference in MDM, Multiple-Input-Multiple-Output (MIMO) processing using adaptive frequency-domain equalization (FDE) is proposed and investigated. Both simulations and experiments validate that FDE can reduce the algorithmic complexity significantly in comparison with the conventional time-domain equalization (TDE) while achieving similar performance as TDE. To further improve the performance of FDE, two modifications on traditional FDE algorithm are demonstrated. i) normalized adaptive FDE is applied to increase the convergence speed by 5 times; ii) master-slave carrier recovery is proposed to reduce the algorithmic complexity of phase estimation by number of modes.Although FDE can reduce the computational complexity of the MIMO processing, due to large mode group delay (MGD) of FMF link and block processing, the algorithm still requires enormous memory and high hardware complexity. In order to reduce the required tap length (RTL) of the equalizer, differential mode group delay compensated fiber (DMGDC) has been proposed. In this dissertation, the analytical expression for RTL is derived for DMGDC systems under the weak mode coupling assumption. Instead of depending on the overall MGD of the link in DMGD uncompensated (DMGDUC) systems, the RTL of DMGDC systems depend on the MGD of a single DMGDC fiber section. The theoretical and numerical results suggest that by using small compensation step-size, the RTL of DMGDC link can be reduced by 2 orders of magnitude compared to DMGDUC link. To compensate the loss of different modes, multimode EDFAs are presented with re-configurable multimode pumps. By tuning the mode content of the multimode pump, mode-dependent gain (MDG) can be controlled and equalized. A proto-type FM-EDFA which could support 2 LP modes was constructed. The experimental results show that by using high order mode pumps, the modal gain difference can be reduced. By applying both multimode EDFA and equalization techniques, 26.4Tb/s MDM-WDM transmission was successfully demonstrated.A brief summary and several possible future research directions conclude this dissertation.
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Date Issued: 2013
Identifier: CFE0004811, ucf:49751
Format: Document (PDF)
PURL: http://purl.flvc.org/ucf/fd/CFE0004811

Title: BIOSIGNAL PROCESSING CHALLENGES IN EMOTION RECOGNITIONFOR ADAPTIVE LEARNING.
Creator: Vartak, Aniket, Mikhael, Wasfy, University of Central Florida
Abstract / Description: User-centered computer based learning is an emerging field of interdisciplinary research. Research in diverse areas such as psychology, computer science, neuroscience and signal processing is making contributions the promise to take this field to the next level. Learning systems built using contributions from these fields could be used in actual training and education instead of just laboratory proof-of-concept. One of the important advances in this research is the detection and assessment of...
Show moreUser-centered computer based learning is an emerging field of interdisciplinary research. Research in diverse areas such as psychology, computer science, neuroscience and signal processing is making contributions the promise to take this field to the next level. Learning systems built using contributions from these fields could be used in actual training and education instead of just laboratory proof-of-concept. One of the important advances in this research is the detection and assessment of the cognitive and emotional state of the learner using such systems. This capability moves development beyond the use of traditional user performance metrics to include system intelligence measures that are based on current neuroscience theories. These advances are of paramount importance in the success and wide spread use of learning systems that are automated and intelligent. Emotion is considered an important aspect of how learning occurs, and yet estimating it and making adaptive adjustments are not part of most learning systems. In this research we focus on one specific aspect of constructing an adaptive and intelligent learning system, that is, estimation of the emotion of the learner as he/she is using the automated training system. The challenge starts with the definition of the emotion and the utility of it in human life. The next challenge is to measure the co-varying factors of the emotions in a non-invasive way, and find consistent features from these measures that are valid across wide population. In this research we use four physiological sensors that are non-invasive, and establish a methodology of utilizing the data from these sensors using different signal processing tools. A validated set of visual stimuli used worldwide in the research of emotion and attention, called International Affective Picture System (IAPS), is used. A dataset is collected from the sensors in an experiment designed to elicit emotions from these validated visual stimuli. We describe a novel wavelet method to calculate hemispheric asymmetry metric using electroencephalography data. This method is tested against typically used power spectral density method. We show overall improvement in accuracy in classifying specific emotions using the novel method. We also show distinctions between different discrete emotions from the autonomic nervous system activity using electrocardiography, electrodermal activity and pupil diameter changes. Findings from different features from these sensors are used to give guidelines to use each of the individual sensors in the adaptive learning environment.
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Date Issued: 2010
Identifier: CFE0003301, ucf:48503
Format: Document (PDF)
PURL: http://purl.flvc.org/ucf/fd/CFE0003301

Title: DISCUSSION ON EFFECTIVE RESTORATION OF ORAL SPEECH USING VOICE CONVERSION TECHNIQUES BASED ON GAUSSIAN MIXTURE MODELING.
Creator: Alverio, Gustavo, Mikhael, Wasfy, University of Central Florida
Abstract / Description: Today's world consists of many ways to communicate information. One of the most effective ways to communicate is through the use of speech. Unfortunately many lose the ability to converse. This in turn leads to a large negative psychological impact. In addition, skills such as lecturing and singing must now be restored via other methods. The usage of text-to-speech synthesis has been a popular resolution of restoring the capability to use oral speech. Text to speech synthesizers convert...
Show moreToday's world consists of many ways to communicate information. One of the most effective ways to communicate is through the use of speech. Unfortunately many lose the ability to converse. This in turn leads to a large negative psychological impact. In addition, skills such as lecturing and singing must now be restored via other methods. The usage of text-to-speech synthesis has been a popular resolution of restoring the capability to use oral speech. Text to speech synthesizers convert text into speech. Although text to speech systems are useful, they only allow for few default voice selections that do not represent that of the user. In order to achieve total restoration, voice conversion must be introduced. Voice conversion is a method that adjusts a source voice to sound like a target voice. Voice conversion consists of a training and converting process. The training process is conducted by composing a speech corpus to be spoken by both source and target voice. The speech corpus should encompass a variety of speech sounds. Once training is finished, the conversion function is employed to transform the source voice into the target voice. Effectively, voice conversion allows for a speaker to sound like any other person. Therefore, voice conversion can be applied to alter the voice output of a text to speech system to produce the target voice. The thesis investigates how one approach, specifically the usage of voice conversion using Gaussian mixture modeling, can be applied to alter the voice output of a text to speech synthesis system. Researchers found that acceptable results can be obtained from using these methods. Although voice conversion and text to speech synthesis are effective in restoring voice, a sample of the speaker before voice loss must be used during the training process. Therefore it is vital that voice samples are made to combat voice loss.
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Date Issued: 2007
Identifier: CFE0001793, ucf:47286
Format: Document (PDF)
PURL: http://purl.flvc.org/ucf/fd/CFE0001793

Title: HYBRID PHOTONIC SIGNAL PROCESSING.
Creator: Ghauri, Farzan, Riza, Nabeel, University of Central Florida
Abstract / Description: This thesis proposes research of novel hybrid photonic signal processing systems in the areas of optical communications, test and measurement, RF signal processing and extreme environment optical sensors. It will be shown that use of innovative hybrid techniques allows design of photonic signal processing systems with superior performance parameters and enhanced capabilities. These applications can be divided into domains of analog-digital hybrid signal processing applications and free-space-...
Show moreThis thesis proposes research of novel hybrid photonic signal processing systems in the areas of optical communications, test and measurement, RF signal processing and extreme environment optical sensors. It will be shown that use of innovative hybrid techniques allows design of photonic signal processing systems with superior performance parameters and enhanced capabilities. These applications can be divided into domains of analog-digital hybrid signal processing applications and free-space--fiber-coupled hybrid optical sensors. The analog-digital hybrid signal processing applications include a high-performance analog-digital hybrid MEMS variable optical attenuator that can simultaneously provide high dynamic range as well as high resolution attenuation controls; an analog-digital hybrid MEMS beam profiler that allows high-power watt-level laser beam profiling and also provides both submicron-level high resolution and wide area profiling coverage; and all optical transversal RF filters that operate on the principle of broadband optical spectral control using MEMS and/or Acousto-Optic tunable Filters (AOTF) devices which can provide continuous, digital or hybrid signal time delay and weight selection. The hybrid optical sensors presented in the thesis are extreme environment pressure sensors and dual temperature-pressure sensors. The sensors employ hybrid free-space and fiber-coupled techniques for remotely monitoring a system under simultaneous extremely high temperatures and pressures.
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Date Issued: 2007
Identifier: CFE0001983, ucf:47423
Format: Document (PDF)
PURL: http://purl.flvc.org/ucf/fd/CFE0001983

Title: LOW NOISE, HIGH REPETITION RATE SEMICONDUCTOR-BASED MODE-LOCKED LASERS FOR SIGNAL PROCESSING AND COHERENT COMMUNICATIONS.
Creator: Quinlan, Franklyn, Delfyett, Peter, University of Central Florida
Abstract / Description: This dissertation details work on high repetition rate semiconductor mode-locked lasers. The qualities of stable pulse trains and stable optical frequency content are the focus of the work performed. First, applications of such lasers are reviewed with particular attention to applications only realizable with laser performance such as presented in this dissertation. Sources of timing jitter are also reviewed, as are techniques by which the timing jitter of a 10 GHz optical pulse train may be...
Show moreThis dissertation details work on high repetition rate semiconductor mode-locked lasers. The qualities of stable pulse trains and stable optical frequency content are the focus of the work performed. First, applications of such lasers are reviewed with particular attention to applications only realizable with laser performance such as presented in this dissertation. Sources of timing jitter are also reviewed, as are techniques by which the timing jitter of a 10 GHz optical pulse train may be measured. Experimental results begin with an exploration of the consequences on the timing and amplitude jitter of the phase noise of an RF source used for mode-locking. These results lead to an ultralow timing jitter source, with 30 fs of timing jitter (1 Hz to 5 GHz, extrapolated). The focus of the work then shifts to generating a stabilized optical frequency comb. The first technique to generating the frequency comb is through optical injection. It is shown that not only can injection locking stabilize a mode-locked laser to the injection seed, but linewidth narrowing, timing jitter reduction and suppression of superfluous optical supermodes of a harmonically mode-locked laser also result. A scheme by which optical injection locking can be maintained long term is also proposed. Results on using an intracavity etalon for supermode suppression and optical frequency stabilization then follow. An etalon-based actively mode-locked laser is shown to have a timing jitter of only 20 fs (1Hz-5 GHz, extrapolated), optical linewidths below 10 kHz and optical frequency instabilities less than 400 kHz. By adding dispersion compensating fiber, the optical spectrum was broadened to 2 THz and 800 fs duration pulses were obtained. By using the etalon-based actively mode-locked laser as a basis, a completely self-contained frequency stabilized coupled optoelectronic oscillator was built and characterized. By simultaneously stabilizing the optical frequencies and the pulse repetition rate to the etalon, a 10 GHz comb source centered at 1550 nm was realized. This system maintains the high quality performance of the actively mode-locked laser while significantly reducing the size weight and power consumption of the system. This system also has the potential for outperforming the actively mode-locked laser by increasing the finesse and stability of the intracavity etalon. The final chapter of this dissertation outlines the future work on the etalon-based coupled optoelectronic oscillator, including the incorporation of a higher finesse, more stable etalon and active phase noise suppression of the RF signal. Two appendices give details on phase noise measurements that incorporate carrier suppression and the noise model for the coupled optoelectronic oscillator.
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Date Issued: 2008
Identifier: CFE0002252, ucf:47878
Format: Document (PDF)
PURL: http://purl.flvc.org/ucf/fd/CFE0002252

Title: Signal processing with Fourier analysis, novel algorithms and applications.
Creator: Syed, Alam, Foroosh, Hassan, Sun, Qiyu, Bagci, Ulas, Rahnavard, Nazanin, Atia, George, Katsevich, Alexander, University of Central Florida
Abstract / Description: Fourier analysis is the study of the way general functions may be represented or approximatedby sums of simpler trigonometric functions, also analogously known as sinusoidal modeling. Theoriginal idea of Fourier had a profound impact on mathematical analysis, physics, and engineeringbecause it diagonalizes time-invariant convolution operators. In the past signal processing was atopic that stayed almost exclusively in electrical engineering, where only the experts could cancelnoise, compress...
Show moreFourier analysis is the study of the way general functions may be represented or approximatedby sums of simpler trigonometric functions, also analogously known as sinusoidal modeling. Theoriginal idea of Fourier had a profound impact on mathematical analysis, physics, and engineeringbecause it diagonalizes time-invariant convolution operators. In the past signal processing was atopic that stayed almost exclusively in electrical engineering, where only the experts could cancelnoise, compress and reconstruct signals. Nowadays it is almost ubiquitous, as everyone now dealswith modern digital signals.Medical imaging, wireless communications and power systems of the future will experience moredata processing conditions and wider range of applications requirements than the systems of today.Such systems will require more powerful, efficient and flexible signal processing algorithms thatare well designed to handle such needs. No matter how advanced our hardware technology becomeswe will still need intelligent and efficient algorithms to address the growing demands in signalprocessing. In this thesis, we investigate novel techniques to solve a suite of four fundamentalproblems in signal processing that have a wide range of applications. The relevant equations, literatureof signal processing applications, analysis and final numerical algorithms/methods to solvethem using Fourier analysis are discussed for different applications in the electrical engineering /computer science. The first four chapters cover the following topics of central importance in thefield of signal processing: Fast Phasor Estimation using Adaptive Signal Processing (Chapter 2) Frequency Estimation from Nonuniform Samples (Chapter 3) 2D Polar and 3D Spherical Polar Nonuniform Discrete Fourier Transform (Chapter 4)iv Robust 3D registration using Spherical Polar Discrete Fourier Transform and Spherical Harmonics(Chapter 5)Even though each of these four methods discussed may seem completely disparate, the underlyingmotivation for more efficient processing by exploiting the Fourier domain signal structureremains the same. The main contribution of this thesis is the innovation in the analysis, synthesis, discretization of certain well-known problems like phasor estimation, frequency estimation, computations of a particular non-uniform Fourier transform and signal registration on the transformed domain. We conduct propositions and evaluations of certain applications relevant algorithms suchas, frequency estimation algorithm using non-uniform sampling, polar and spherical polar Fourier transform. The techniques proposed are also useful in the field of computer vision and medical imaging. From a practical perspective, the proposed algorithms are shown to improve the existing solutions in the respective fields where they are applied/evaluated. The formulation and final proposition is shown to have a variety of benefits. Future work with potentials in medical imaging, directional wavelets, volume rendering, video/3D object classifications, high dimensional registration are also discussed in the final chapter. Finally, in the spirit of reproducible research, we release the implementation of these algorithms to the public using Github.
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Date Issued: 2017
Identifier: CFE0006803, ucf:51775
Format: Document (PDF)
PURL: http://purl.flvc.org/ucf/fd/CFE0006803

Title: Brain stethoscope: A non-invasive method for monitoring intracranial pressure.
Creator: Azad, Md Khurshidul, Mansy, Hansen, Kassab, Alain, Bhattacharya, Samik, University of Central Florida
Abstract / Description: Monitoring intracranial pressure (ICP) is important for patients with increased intracranial pressure. Invasive methods of ICP monitoring include lumbar puncture manometry, which requires high precision, is costly, and can lead to complications. Non-invasive monitoring of ICP using tympanic membrane pulse (TMp) measurement can provide an alternative monitoring method that avoids such complications. In the current study, a piezo based sensor was designed, constructed and used to acquire TMp...
Show moreMonitoring intracranial pressure (ICP) is important for patients with increased intracranial pressure. Invasive methods of ICP monitoring include lumbar puncture manometry, which requires high precision, is costly, and can lead to complications. Non-invasive monitoring of ICP using tympanic membrane pulse (TMp) measurement can provide an alternative monitoring method that avoids such complications. In the current study, a piezo based sensor was designed, constructed and used to acquire TMp signals. The results showed that tympanic membrane waveform changed in morphology and amplitude with increased ICP, which was induced by changing subject position using a tilt table. In addition, the results suggest that TMp are affected by breathing, which has small effects on ICP. The newly developed piezo based brain stethoscope may be a way to monitor patients with increased intracranial pressure thus avoiding invasive ICP monitoring and reducing associated risk and cost.
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Date Issued: 2018
Identifier: CFE0006972, ucf:51643
Format: Document (PDF)
PURL: http://purl.flvc.org/ucf/fd/CFE0006972

Title: SIGNAL PROCESSING OF AN ECG SIGNALIN THE PRESENCE OF A STRONG STATIC MAGNETIC FIELD.
Creator: Gupta, Aditya, Weeks, Arthur, University of Central Florida
Abstract / Description: This dissertation addresses the problem of elevation of the T wave of an electrocardiogram (ECG) signal in the magnetic resonance imaging (MRI). In the MRI, due to the strong static magnetic field the interaction of the blood flow with this strong magnetic field induces a voltage in the body. This voltage appears as a superimposition at the locus of the T wave of the ECG signal. This looses important information required by the doctors to interpret the ST segment of the ECG and detect...
Show moreThis dissertation addresses the problem of elevation of the T wave of an electrocardiogram (ECG) signal in the magnetic resonance imaging (MRI). In the MRI, due to the strong static magnetic field the interaction of the blood flow with this strong magnetic field induces a voltage in the body. This voltage appears as a superimposition at the locus of the T wave of the ECG signal. This looses important information required by the doctors to interpret the ST segment of the ECG and detect diseases such as myocardial infarction. This dissertation aims at finding a solution to the problem of elevation of the T wave of an ECG signal in the MRI. The first step is to simulate the entire situation and obtain the magnetic field dependent T wave elevation. This is achieved by building a model of the aorta and simulating the blood flow in it. This model is then subjected to a static magnetic field and the surface potential on the thorax is measured to observe the T wave elevation. The various parameters on which the T wave elevation is dependent are then analyzed. Different approaches are used to reduce this T wave elevation problem. The direct approach aims at computing the magnitude of T wave elevation using magneto-hydro-dynamic equations. The indirect approach uses digital signal processing tools like the least mean square adaptive filter to remove the T wave elevation and obtain artifact free ECG signal in the MRI. Excellent results are obtained from the simulation model. The model perfectly simulates the ECG signal in the MRI at all the 12 leads of the ECG. These results are compared with ECG signals measured in the MRI. A simulation package is developed in MATLAB based on the simulation model. This package is a graphical user interface allowing the user to change the strength of magnetic field, the radius of the aorta and the orientation of the aorta with respect to the heart and observe the ECG signals with the elevation at the 12 leads of the ECG. Also the artifacts introduced due to the magnetic field can be removed by the least mean square adaptive filter. The filter adapts the ECG signal in the MRI to the ECG signal of the patient outside the MRI. Before the adaptation, the heart rate of the ECG outside the MRI is matched to the ECG in the MRI by interpolation or decimation. The adaptive filter works excellently to remove the T wave artifacts. When the cardiac output of the patient changes, the simulation model is used along with the adaptive filter to obtain the artifact free ECG signal.
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Date Issued: 2007
Identifier: CFE0001857, ucf:47389
Format: Document (PDF)
PURL: http://purl.flvc.org/ucf/fd/CFE0001857

Signal Processing (3)	+ -
Digital Signal Processing (2)	+ -
ECG (2)	+ -
Optical Fiber Communication (2)	+ -
biomedical signal processing (2)	+ -

signal processing (2)	+ -
Acoustic (1)	+ -
Adaptive Filter (1)	+ -
Coherent Detection (1)	+ -
DSP (1)	+ -
Developmental (1)	+ -
Digital filters (Mathematics) (1)	+ -
EDR (1)	+ -
EEG (1)	+ -
Eyetracking (1)	+ -
Fiber Amplifier (1)	+ -
Frequency Estimation (1)	+ -
GSR (1)	+ -
Intracranial pressure (1)	+ -
LMS (1)	+ -
Laser Beam Profiler (1)	+ -
MATLAB (1)	+ -
MFD (1)	+ -
MHD (1)	+ -
MRI (1)	+ -
Machine Learning (1)	+ -
Microwave Photonics (1)	+ -
Multiplexing (1)	+ -
Nonuniform Fourier Transforms (1)	+ -
Optical Sensors (1)	+ -

University of Central Florida (12)	+ -
Mikhael, Wasfy (4)	+ -
Kassab, Alain (2)	+ -
Li, Guifang (2)	+ -
Mansy, Hansen (2)	+ -

Abouraddy, Ayman (1)	+ -
Alverio, Gustavo (1)	+ -
Atia, George (1)	+ -
Azad, Md Khurshidul (1)	+ -
Bagci, Ulas (1)	+ -
Bai, Neng (1)	+ -
Batarseh, Issa (1)	+ -
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