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Title: Image improvement using dynamic optical low-pass filter.
Creator: Petljanski, Branko., College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: Professional imaging systems, particularly motion picture cameras, usually employ larger photosites and lower pixel counts than many amateur cameras. This results in the desirable characteristics of improved dynamic range, signal to noise and sensitivity. However, high performance optics often have frequency response characteristics that exceed the Nyquist limit of the sensor, which, if not properly addressed, results in aliasing artifacts in the captured image. Most contemporary still and...
Show moreProfessional imaging systems, particularly motion picture cameras, usually employ larger photosites and lower pixel counts than many amateur cameras. This results in the desirable characteristics of improved dynamic range, signal to noise and sensitivity. However, high performance optics often have frequency response characteristics that exceed the Nyquist limit of the sensor, which, if not properly addressed, results in aliasing artifacts in the captured image. Most contemporary still and video cameras employ various optically birefringent materials as optical low-pass filters (OLPF) in order to minimize aliasing artifacts in the image. Most OLPFs are designed as optical elements with a frequency response that does not change even if the frequency responses of the other elements of the capturing systems are altered. An extended evaluation of currently used birefringent-based OLPFs is provided. In this work, the author proposed and demonstrated the use of a parallel optical window p ositioned between a lens and a sensor as an OLPF. Controlled X- and Y-axes rotations of the optical window during the image exposure results in a manipulation of the system's point-spread function (PSF). Consequently, changing the PSF affects some portions of the frequency components contained in the image formed on the sensor. The system frequency response is evaluated when various window functions are used to shape the lens' PSF, such as rectangle, triangle, Tukey, Gaussian, Blackman-Harris etc. In addition to the ability to change the PSF, this work demonstrated that the PSF can be manipulated dynamically, which allowed us to modify the PSF to counteract any alteration of other optical elements of the capturing system. There are several instances presented in the dissertation in which it is desirable to change the characteristics of an OLPF in a controlled way., In these instances, an OLPF whose characteristics can be altered dynamically results in an improvement of the image quality.
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Date Issued: 2010
PURL: http://purl.flvc.org/FAU/1927613
Subject Headings: Image processing, Digital techniques, Signal processing, Digital techniques, Frequency response (Dynamics), Polymers and polymerization, Optical wave guides
Format: Document (PDF)

Title: Shamir's secret sharing scheme using floating point arithmetic.
Creator: Finamore, Timothy., Charles E. Schmidt College of Science, Department of Mathematical Sciences
Abstract/Description: Implementing Shamir's secret sharing scheme using floating point arithmetic would provide a faster and more efficient secret sharing scheme due to the speed in which GPUs perform floating point arithmetic. However, with the loss of a finite field, properties of a perfect secret sharing scheme are not immediately attainable. The goal is to analyze the plausibility of Shamir's secret sharing scheme using floating point arithmetic achieving the properties of a perfect secret sharing scheme and...
Show moreImplementing Shamir's secret sharing scheme using floating point arithmetic would provide a faster and more efficient secret sharing scheme due to the speed in which GPUs perform floating point arithmetic. However, with the loss of a finite field, properties of a perfect secret sharing scheme are not immediately attainable. The goal is to analyze the plausibility of Shamir's secret sharing scheme using floating point arithmetic achieving the properties of a perfect secret sharing scheme and propose improvements to attain these properties. Experiments indicate that property 2 of a perfect secret sharing scheme, "Any k-1 or fewer participants obtain no information regarding the shared secret", is compromised when Shamir's secret sharing scheme is implemented with floating point arithmetic. These experimental results also provide information regarding possible solutions and adjustments. One of which being, selecting randomly generated points from a smaller interval in one of the proposed schemes of this thesis. Further experimental results indicate improvement using the scheme outlined. Possible attacks are run to test the desirable properties of the different schemes and reinforce the improvements observed in prior experiments.
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Date Issued: 2012
PURL: http://purl.flvc.org/FAU/3342048
Subject Headings: Signal processing, Digital techniques, Mathematics, Data encryption (Computer science), Computer file sharing, Security measures, Computer algorithms, Numerical analysis, Data processing
Format: Document (PDF)

Title: Sensitivity analysis of blind separation of speech mixtures.
Creator: Bulek, Savaskan., College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: Blind source separation (BSS) refers to a class of methods by which multiple sensor signals are combined with the aim of estimating the original source signals. Independent component analysis (ICA) is one such method that effectively resolves static linear combinations of independent non-Gaussian distributions. We propose a method that can track variations in the mixing system by seeking a compromise between adaptive and block methods by using mini-batches. The resulting permutation...
Show moreBlind source separation (BSS) refers to a class of methods by which multiple sensor signals are combined with the aim of estimating the original source signals. Independent component analysis (ICA) is one such method that effectively resolves static linear combinations of independent non-Gaussian distributions. We propose a method that can track variations in the mixing system by seeking a compromise between adaptive and block methods by using mini-batches. The resulting permutation indeterminacy is resolved based on the correlation continuity principle. Methods employing higher order cumulants in the separation criterion are susceptible to outliers in the finite sample case. We propose a robust method based on low-order non-integer moments by exploiting the Laplacian model of speech signals. We study separation methods for even (over)-determined linear convolutive mixtures in the frequency domain based on joint diagonalization of matrices employing time-varying second order statistics. We investigate the sources affecting the sensitivity of the solution under the finite sample case such as the set size, overlap amount and cross-spectrum estimation methods.
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Date Issued: 2010
PURL: http://purl.flvc.org/FAU/2953201
Subject Headings: Blind source separation, Mathematical models, Signal processing, Digital techniques, Neural networks (Computer science), Automatic speech recognition, Speech processing systems
Format: Document (PDF)

Title: Spectral refinement to speech enhancement.
Creator: Charoenruengkit, Werayuth., College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: The goal of a speech enhancement algorithm is to remove noise and recover the original signal with as little distortion and residual noise as possible. Most successful real-time algorithms thereof have done in the frequency domain where the frequency amplitude of clean speech is estimated per short-time frame of the noisy signal. The state of-the-art short-time spectral amplitude estimator algorithms estimate the clean spectral amplitude in terms of the power spectral density (PSD) function...
Show moreThe goal of a speech enhancement algorithm is to remove noise and recover the original signal with as little distortion and residual noise as possible. Most successful real-time algorithms thereof have done in the frequency domain where the frequency amplitude of clean speech is estimated per short-time frame of the noisy signal. The state of-the-art short-time spectral amplitude estimator algorithms estimate the clean spectral amplitude in terms of the power spectral density (PSD) function of the noisy signal. The PSD has to be computed from a large ensemble of signal realizations. However, in practice, it may only be estimated from a finite-length sample of a single realization of the signal. Estimation errors introduced by these limitations deviate the solution from the optimal. Various spectral estimation techniques, many with added spectral smoothing, have been investigated for decades to reduce the estimation errors. These algorithms do not address significantly issue on quality of speech as perceived by a human. This dissertation presents analysis and techniques that offer spectral refinements toward speech enhancement. We present an analytical framework of the effect of spectral estimate variance on the performance of speech enhancement. We use the variance quality factor (VQF) as a quantitative measure of estimated spectra. We show that reducing the spectral estimator VQF reduces significantly the VQF of the enhanced speech. The Autoregressive Multitaper (ARMT) spectral estimate is proposed as a low VQF spectral estimator for use in speech enhancement algorithms. An innovative method of incorporating a speech production model using multiband excitation is also presented as a technique to emphasize the harmonic components of the glottal speech input., The preconditioning of the noisy estimates by exploiting other avenues of information, such as pitch estimation and the speech production model, effectively increases the localized narrow-band signal-to noise ratio (SNR) of the noisy signal, which is subsequently denoised by the amplitude gain. Combined with voicing structure enhancement, the ARMT spectral estimate delivers enhanced speech with sound clarity desirable to human listeners. The resulting improvements in enhanced speech are observed to be significant with both Objective and Subjective measurement.
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Date Issued: 2009
PURL: http://purl.flvc.org/FAU/186327
Subject Headings: Adaptive signal processing, Digital techniques, Spectral theory (Mathematics), Noise control, Fuzzy algorithms, Speech processing systems, Digital techniques
Format: Document (PDF)

Title: Discrete digital filter design for microelectromechanical systems (MEMS) accelerometers and gyroscopes.
Creator: Martin, Madison E., Harriet L. Wilkes Honors College
Abstract/Description: Microelectromechanical systems (MEMS) accelerometers and gyroscopes are small scale sensors that measure changes in linear acceleration and rotational velocity, respectively. They are fabricated using electronic circuit techniques such as etching and deposition. MEMS motion sensors can be used in an Inertial Measurement Unit (IMU) that can be integrated with the Global Positioning System (GPS) to make a navigation system that is more accurate than each system alone. However, since MEMS-based...
Show moreMicroelectromechanical systems (MEMS) accelerometers and gyroscopes are small scale sensors that measure changes in linear acceleration and rotational velocity, respectively. They are fabricated using electronic circuit techniques such as etching and deposition. MEMS motion sensors can be used in an Inertial Measurement Unit (IMU) that can be integrated with the Global Positioning System (GPS) to make a navigation system that is more accurate than each system alone. However, since MEMS-based IMUs are inherently noisy, we must overcome inaccuracies caused by the integration of random noise to find position. Accuracy can be increased by applying digital filters to the data before integration. Comparing the success of finite impulse response (FIR) filters and infinite impulse response (IIR) filters, we found that even though our highest order FIR filter yielded the most accurate position, it was limited by an offset bias in the accelerometer signal and a time delay in the determined position.
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Date Issued: 2010
PURL: http://purl.flvc.org/FAU/3335110
Subject Headings: Microelectromechanical systems, Design and construction, Signal processing, Digital techniques, Electric filters, Digital, Design and construction
Format: Document (PDF)

Title: Characterizing the Magnetic Signature of Internal Waves.
Creator: Nieves, Eric, Beaujean, Pierre-Philippe, Florida Atlantic University, College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: This study is performed in tandem with numerous experiments performed by the U.S. Navy to characterize the ocean environment in the South Florida region. The research performed in this study includes signal processing steps for isolating ocean phenomena, such as internal waves, in the magnetic field. Raw magnetometer signals, one on shore and one underwater, are processed and removed of common distortions. They are then run through a series of filtering techniques, including frequency domain...
Show moreThis study is performed in tandem with numerous experiments performed by the U.S. Navy to characterize the ocean environment in the South Florida region. The research performed in this study includes signal processing steps for isolating ocean phenomena, such as internal waves, in the magnetic field. Raw magnetometer signals, one on shore and one underwater, are processed and removed of common distortions. They are then run through a series of filtering techniques, including frequency domain cancellation (FDC). The results of the filtered magnetic residual are compared to similarly processed Acoustic Doppler Current Profiler (ADCP) data to correlate whether a magnetic signature is caused by ocean phenomena.
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Date Issued: 2017
PURL: http://purl.flvc.org/fau/fd/FA00004917, http://purl.flvc.org/fau/fd/FA00004917
Subject Headings: Ocean currents--Measurement., Adaptive signal processing., Wave-motion, Theory of., Wavelets (Mathematics)
Format: Document (PDF)

Title: Acoustic tracking of an unmanned underwater vehicle using a passive ultrashort baseline array and a single long baseline beacon.
Creator: Seaton, Kyle L., College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: This thesis discusses a new approach to tracking the REMUS 100 AUV using a modified version of the Florida Atlantic University (FAU) ultrashort baseline (USBL) acoustic positioning system (APS). The REMUS 100 is designed to utilize a long baseline (LBL) acoustic positioning system to obtain positioning data in mid-mission. If the placement of one of the transponders of the LBL field is known, then tracking the position of the REMUS 100 AUV using a passive USBL array is possible. As part of...
Show moreThis thesis discusses a new approach to tracking the REMUS 100 AUV using a modified version of the Florida Atlantic University (FAU) ultrashort baseline (USBL) acoustic positioning system (APS). The REMUS 100 is designed to utilize a long baseline (LBL) acoustic positioning system to obtain positioning data in mid-mission. If the placement of one of the transponders of the LBL field is known, then tracking the position of the REMUS 100 AUV using a passive USBL array is possible. As part of the research for this thesis, the FAU USBL system was used to find a relative range between the REMUS 100 ranger and a LBL transponder. This relative range was then combined with direction of arrival information and LBL field component position information to determine an absolute position of the REMUS 100 ranger. The outcome was the demonstration of a passive USBL based tracking system.
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Date Issued: 2013
PURL: http://purl.flvc.org/fcla/dt/3361057
Subject Headings: Underwater acoustic telemetry, Acoustic velocity meters, Array processors, Acoustical engineering, Adaptive signal processing
Format: Document (PDF)

Title: Nonlinear control of an unmanned amphibious vehicle.
Creator: Alvarez, Jose L., College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: The DUCKW-Ling is an 8.3 foot long, amphibious water plane area twin hull (SWATH) concept vehicle which is propelled by a pair of crawler tracks on land and dual propellers when water-borne. In its operational zone, the vehicle's dynamics change dramatically as it transitions from being completely water-borne and buoyancy supported to being completely land-borne and track supported. In the water environment, a cascaded, first-order sliding mode controller was used to control the surge and...
Show moreThe DUCKW-Ling is an 8.3 foot long, amphibious water plane area twin hull (SWATH) concept vehicle which is propelled by a pair of crawler tracks on land and dual propellers when water-borne. In its operational zone, the vehicle's dynamics change dramatically as it transitions from being completely water-borne and buoyancy supported to being completely land-borne and track supported. In the water environment, a cascaded, first-order sliding mode controller was used to control the surge and heading of the vehicle, and was capable of having a faster response when compared to using a proportional controller. Additionally, field trials of the DUKW-Ling show the capability of the vehicle to navigate and track predetermined waypoints in both terrestrial and aquatic terrains. In the transitional zone, the electric motor current from the tracks was used as the feedback mechanism to adequately actuate the propellers and tracks in the system as the dynamics of the vehicle change.
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Date Issued: 2013
PURL: http://purl.flvc.org/fcla/dt/3362334
Subject Headings: Nonlinear control theory, Feedback control systems, Motor vehicles, Amphibious, Design and construction, Adaptive signal processing
Format: Document (PDF)

Title: A modular guidance, navigation and control system for unmanned surface vehicles.
Creator: Furfaro, Thomas C., College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: The design and integration of an unmanned surface vehicle (USV) control system is described. A survey of related work in both USV control, and unmanned vehicle operating software is presented. The hardware subsystem comprising a modular Guidance, Navigation, and Control (GNC) package is explained. A multi-threaded software architecture is presented, utilizing a decentralized, mutex-protected shared memory inter-process communication subsystem to provide interoperability with additional...
Show moreThe design and integration of an unmanned surface vehicle (USV) control system is described. A survey of related work in both USV control, and unmanned vehicle operating software is presented. The hardware subsystem comprising a modular Guidance, Navigation, and Control (GNC) package is explained. A multi-threaded software architecture is presented, utilizing a decentralized, mutex-protected shared memory inter-process communication subsystem to provide interoperability with additional software modules. A generic GNC approach is presented, with particular elaboration on a virtual rudder abstraction of differential thrust platforms. A MATLAB Simulink simulation is presented as a tool for developing an appropriate controller structure, the result of which was implemented on the target platform. Software validation is presented via a series of sea trials. The USV was tested both in open- and closed-loop control configurations, the results of which are presented here. Lastly recommendations for future development of the GNC system are enumerated.
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Date Issued: 2012
PURL: http://purl.flvc.org/FAU/3342052
Subject Headings: Underwater navigation, Adaptive signal processing, Inertial navigation systems, Oceanographic instruments, Development
Format: Document (PDF)

Title: A low-cost, high rate motion measurement system for an unmanned surface vehicle with underwater navigation and oceanographic applications.
Creator: Gelin, Chrystel., College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: Standard GPS receivers are unable to provide the rate or precision required when used on a small vessel such as an Unmanned Surface Vehicles (USVs). To overcome this, the thesis presents a low cost high rate motion measurement system for an USV with underwater and oceanographic purposes. The work integrates an Inertial Measurement Unit (IMU), a GPS receiver, a flux-gate compass, a tilt sensor and develops a software package, using real time data fusion methods, for an USV to aid in the...
Show moreStandard GPS receivers are unable to provide the rate or precision required when used on a small vessel such as an Unmanned Surface Vehicles (USVs). To overcome this, the thesis presents a low cost high rate motion measurement system for an USV with underwater and oceanographic purposes. The work integrates an Inertial Measurement Unit (IMU), a GPS receiver, a flux-gate compass, a tilt sensor and develops a software package, using real time data fusion methods, for an USV to aid in the navigation and control as well as controlling an onboard Acoustic Doppler Current Profiler (ADCP).While ADCPs non-intrusively measure water flow, they suffer from the inability to discriminate between motions in the water column and self-motion. Thus, the vessel motion contamination needs to be removed to analyze the data and the system developed in this thesis provides the motion measurements and processing to accomplish this task.
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Date Issued: 2010
PURL: http://purl.flvc.org/FAU/2709690
Subject Headings: Oceanographic instruments, Evaluation, Underwater navigation, Adaptive signal processing, Inertial navigation systems, Wave motion, Theory of
Format: Document (PDF)

Title: A methodology to detect and classify underwater unexploded ordnance in DIDSON sonar images.
Creator: Brisson, Lisa Nicole., College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: High-resolution sonar systems are primarily used for ocean floor surveys and port security operations but produce images of limited resolution. In turn, a sonar-specific methodology is required to detect and classify underwater unexploded ordnance (UXO) using the low-resolution sonar data. After researching and reviewing numerous approaches the Multiple Aspect-Fixed Range Template Matching (MAFR-TM) algorithm was developed. The MAFR-TM algorithm is specifically designed to detect and classify...
Show moreHigh-resolution sonar systems are primarily used for ocean floor surveys and port security operations but produce images of limited resolution. In turn, a sonar-specific methodology is required to detect and classify underwater unexploded ordnance (UXO) using the low-resolution sonar data. After researching and reviewing numerous approaches the Multiple Aspect-Fixed Range Template Matching (MAFR-TM) algorithm was developed. The MAFR-TM algorithm is specifically designed to detect and classify a target of high characteristic impedance in an environment that contains similar shaped objects of low characteristic impedance. MAFR-TM is tested against a tank and field data set collected by the Sound Metrics Corp. DIDSON US300. This thesis document proves the MAFR-TM can detect, classify, orient, and locate a target in the sector-scan sonar images. This paper focuses on the MAFR-TM algorithm and its results.
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Date Issued: 2010
PURL: http://purl.flvc.org/FAU/2683533
Subject Headings: Ocean tomography, Unexploded ordnance, Detection, Methodology, Underwater acoustics, Signal processing, Digital techniques
Format: Document (PDF)

Title: Optimal chunk-based resource allocation for OFDMA systems with multiple BER requirements.
Creator: He, Tianzhou, Wang, Xin, Florida Atlantic University, College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: In wireless orthogonal frequency division multiple-access (OFDMA) standards, subcarriers are grouped into chunks and a chunk of subcarriers is made as the minimum allocation unit for subcarrier allocation. We investigate the chunk-based resource allocation for OFDMA downlink, where data streams contain packets with diverse bit-errorrate (BER) requirements. Supposing that adaptive transmissions are based on a number of discrete modulation and coding modes, we derive the optimal resource...
Show moreIn wireless orthogonal frequency division multiple-access (OFDMA) standards, subcarriers are grouped into chunks and a chunk of subcarriers is made as the minimum allocation unit for subcarrier allocation. We investigate the chunk-based resource allocation for OFDMA downlink, where data streams contain packets with diverse bit-errorrate (BER) requirements. Supposing that adaptive transmissions are based on a number of discrete modulation and coding modes, we derive the optimal resource allocation scheme that maximizes the weighted sum of average user rates under the multiple BER and total power constraints. With proper formulation, the relevant optimization problem is cast as an integer linear program (ILP). We can rigorously prove that the zero duality gap holds for the formulated ILP and its dual problem. Furthermore, it is shown that the optimal strategy for this problem can be obtained through Lagrange dual-based gradient iterations with fast convergence and low computational complexity per iteration. Relying on the stochastic optimization tools, we further develop a novel on-line algorithm capable of dynamically learning the underlying channel distribution and asymptotically approaching the optimal strategy without knowledge of intended wireless channels a priori. In addition, we extend the proposed approach to maximizing the a-fair utility functions of average user rates, and show that such a utility maximization can nicely balance the trade-off between the total throughput and fairness among users.
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Date Issued: 2014
PURL: http://purl.flvc.org/fau/fd/FA00004291, http://purl.flvc.org/fau/fd/FA00004291
Subject Headings: Adaptive signal processing, Mathematical optimization, Orthogonal frequency division multiplexing, Wireless communication systems
Format: Document (PDF)

Title: Source speed estimation using a pilot tone in a high-frequency acoustic modem.
Creator: Kathiroli, Poorani., College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: This thesis proposes to estimate the speed of a moving acoustic source by either linear or non linear processing of the resulting Doppler shift present in a high-frequency pilot tone. The source is an acoustic modem (Hermes) which currently uses moving average to estimate and compensate for Doppler shift. A new auto regressive approach to Doppler estimation (labeled IIR method in the text) promises to give a better estimate. The results for a simulated peak velocity of 2 m/s in the presence...
Show moreThis thesis proposes to estimate the speed of a moving acoustic source by either linear or non linear processing of the resulting Doppler shift present in a high-frequency pilot tone. The source is an acoustic modem (Hermes) which currently uses moving average to estimate and compensate for Doppler shift. A new auto regressive approach to Doppler estimation (labeled IIR method in the text) promises to give a better estimate. The results for a simulated peak velocity of 2 m/s in the presence of additive noise showed an RMSE of 0.23 m/s using moving average vs. 0.00018 m/s for the auto regressive approach. The SNR was 75 dB. The next objective was to compare the estimated Doppler velocity obtained using the two algorithms with the experimental values recorded in real time. The setup consisted of a receiver hydrophone attached to a towing carriage that moved with a known velocity with respect to a stationary acoustic source. The source transmitted 375 kHz pilot tone. The received pilot tone data were preprocessed using the two algorithms to estimate both Doppler shift and Doppler velocity. The accuracy of the algorithms was compared against the true velocity values of the carriage. The RMSE for a message from experiments conducted indoor for constant velocity of 0.4 m/s was 0.6055 m/s using moving average, 0.0780 m/s using auto regressive approach. The SNIR was 6.3 dB.
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Date Issued: 2011
PURL: http://purl.flvc.org/FAU/3171396
Subject Headings: Underwater acoustics, Measurement, SIgnal processing, Digital techniques, Digital filters (Mathematics), Radio frequency, Mathematical models
Format: Document (PDF)

Title: DSP implementation of turbo decoder using the Modified-Log-MAP algorithm.
Creator: Khan, Zeeshan Haneef., Florida Atlantic University, Zhuang, Hanqi, Sudhakar, Raghavan, College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: The design of any communication receiver needs to addresses the issues of operating under the lowest possible signal-to-noise ratio. Among various algorithms that facilitate this objective are those used for iterative decoding of two-dimensional systematic convolutional codes in applications such as spread spectrum communications and Code Division Multiple Access (CDMA) detection. A main theme of any decoding schemes is to approach the Shannon limit in signal-to-noise ratio. All these...
Show moreThe design of any communication receiver needs to addresses the issues of operating under the lowest possible signal-to-noise ratio. Among various algorithms that facilitate this objective are those used for iterative decoding of two-dimensional systematic convolutional codes in applications such as spread spectrum communications and Code Division Multiple Access (CDMA) detection. A main theme of any decoding schemes is to approach the Shannon limit in signal-to-noise ratio. All these decoding algorithms have various complexity levels and processing delay issues. Hence, the optimality depends on how they are used in the system. The technique used in various decoding algorithms is termed as iterative decoding. Iterative decoding was first developed as a practical means for decoding turbo codes. With the Log-Likelihood algebra, it is shown that a decoder can be developed that accepts soft inputs as a priori information and delivers soft outputs consisting of channel information, a posteriori information and extrinsic information to subsequent stages of iteration. Different algorithms such as Soft Output Viterbi Algorithm (SOVA), Maximum A Posteriori (MAP), and Log-MAP are compared and their complexities are analyzed in this thesis. A turbo decoder is implemented on the Digital Signal Processing (DSP) chip, TMS320C30 by Texas Instruments using a Modified-Log-MAP algorithm. For the Modified-Log-MAP-Algorithm, the optimal choice of the lookup table (LUT) is analyzed by experimenting with different LUT approximations. A low complexity decoder is proposed for a (7,5) code and implemented in the DSP chip. Performance of the decoder is verified under the Additive Wide Gaussian Noise (AWGN) environment. Hardware issues such as memory requirements and processing time are addressed for the chosen decoding scheme. Test results of the bit error rate (BER) performance are presented for a fixed number of frames and iterations.
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Date Issued: 2002
PURL: http://purl.flvc.org/fcla/dt/12948
Subject Headings: Error-correcting codes (Information theory), Signal processing--Digital techniques, Coding theory, Digital communications
Format: Document (PDF)

Title: Digital signal processing for a high-resolution three-dimensional sonar imaging system for autonomous underwater vehicles.
Creator: Cao, Ping., Florida Atlantic University, Cuschieri, Joseph M., Sudhakar, Raghavan, College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: In this dissertation, the digital signal processing techniques required for a 3-D sonar imaging system are examined. The achievable performance of the generated images is investigated by using a combination of theoretical analysis, computer simulation and field experiments. The system consists of a forward looking sonar, with separate projector and receiver. The projector is a line source with an 80 degrees by 1.2 degree beam pattern, which is electronically scanned within a 150 degree sector...
Show moreIn this dissertation, the digital signal processing techniques required for a 3-D sonar imaging system are examined. The achievable performance of the generated images is investigated by using a combination of theoretical analysis, computer simulation and field experiments. The system consists of a forward looking sonar, with separate projector and receiver. The projector is a line source with an 80 degrees by 1.2 degree beam pattern, which is electronically scanned within a 150 degree sector. The receiver is a multi element line array, where each transducer element has a directivity pattern that covers the full sector of view, that is 150 degrees by 80 degrees. The purpose of this sonar system is to produce three dimensional (3-D) images which display the underwater topography within the sector of view up to a range of 200 meters. The principle of operation of the proposed 3-D imaging system differs from other commonly used systems in that it is not based on the intensity of backscatter. The geometries of the targets are obtained from the delay and direction information that can be extracted from the signal backscatter. The acquired data is further processed using an approach based on sequential Fourier transforms to build the 3-D images. With careful selection of the system parameters, the generated images have sufficient quality to be used for AUV tasks such as obstacle avoidance, navigation and object classification. An approach based on a sophisticated two dimensional (2-D) autoregressive (AR) model is explored to further improve the resolution and generate images with higher quality. The real time processing requirements for image generation are evaluated, with the use of dedicated Digital Signal Processing (DSP) chips. A pipeline processing model is analyzed and developed on a selected system.
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Date Issued: 1992
PURL: http://purl.flvc.org/fcla/dt/12317
Subject Headings: Sonar, Signal processing--Digital techniques, Three-dimensional display systems, Submersibles
Format: Document (PDF)

Title: Feasibility of target tracking for high speed high seas cargo transfer.
Creator: Tucker, Glenn C., Florida Atlantic University, Driscoll, Frederick R., College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: At sea cargo transfer has historically been a logistical challenge for both the military and the offshore industry. Even in moderate seas, three to five foot wave heights, extreme pendulations of cargo and large relative motions between vessels can occur that halts cargo transfer activities. This work develops a six-degree-of-freedom rigid crane dynamics model that is used to investigate the feasibility of crane target tracking which could extend and enhance offshore crane operations. A...
Show moreAt sea cargo transfer has historically been a logistical challenge for both the military and the offshore industry. Even in moderate seas, three to five foot wave heights, extreme pendulations of cargo and large relative motions between vessels can occur that halts cargo transfer activities. This work develops a six-degree-of-freedom rigid crane dynamics model that is used to investigate the feasibility of crane target tracking which could extend and enhance offshore crane operations. A double girder crane system is developed that easily adapts to different configurations and efficiently supports long reach heavy lift applications. Target tracking is feasible in sea states up to 5 when using the double girder crane. When compared to a present crane system, the target tracking crane requires, on average, only 3.65% more absolute total system power and 13.4% less continuous power, indicating that the proposed system should be realizable.
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Date Issued: 2006
PURL: http://purl.flvc.org/fcla/dt/13388
Subject Headings: Cargo handling, Unitized cargo systems, Signal processing--Digital techniques, Ocean circulation--Mathematical models
Format: Document (PDF)

Title: Identification and approximation of one-dimensional and two-dimensional digital filters.
Creator: Wang, Dali., Florida Atlantic University, Zilouchian, Ali, College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: In this dissertation, identification and approximation of one-dimensional (1-D) and two-dimensional (2-D) recursive digital filters are addressed. In the identification phase, a novel Neural Network (NN) structure is proposed which provides the state-space model of 1-D filters based upon input-output data. The state space identification technique is also extended to 2-D digital filters and several comparison studies are performed. In the approximation phase, frequency-domain balanced...
Show moreIn this dissertation, identification and approximation of one-dimensional (1-D) and two-dimensional (2-D) recursive digital filters are addressed. In the identification phase, a novel Neural Network (NN) structure is proposed which provides the state-space model of 1-D filters based upon input-output data. The state space identification technique is also extended to 2-D digital filters and several comparison studies are performed. In the approximation phase, frequency-domain balanced structures for 1-D as well as 2-D digital filters are proposed. The model reduction technique is based on the conceptual view point of balancing the controllability and observability Grammians of a digital filter in an arbitrary frequency range of operation. Finally, the interrelations between these two phases are presented. Extensive simulation experiments are presented to demonstrate the effectiveness of proposed methods.
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Date Issued: 1998
PURL: http://purl.flvc.org/fcla/dt/12555
Subject Headings: Digital filters (Mathematics), Signal processing--Digital technique, Electric filters, Digital
Format: Document (PDF)

Title: Performance analysis of multitaper spectrum estimation.
Creator: Skoro Kaskarovska, Violeta, Florida Atlantic University, Erdol, Nurgun, College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: We characterize the Multitaper Spectral Estimation method as a tool for stationary signal analysis. We compare its performance to the conventional periodogram, the parametric autoregressive and multitaper autoregressive spectral estimates. We analyze single and two frequency sinusoids with additive Gaussian white noise and autoregressive processes of orders 2, 4 and 24. We extend its application to non-stationary signals and develop the multitaper spectrogram. We test the spectrograms with...
Show moreWe characterize the Multitaper Spectral Estimation method as a tool for stationary signal analysis. We compare its performance to the conventional periodogram, the parametric autoregressive and multitaper autoregressive spectral estimates. We analyze single and two frequency sinusoids with additive Gaussian white noise and autoregressive processes of orders 2, 4 and 24. We extend its application to non-stationary signals and develop the multitaper spectrogram. We test the spectrograms with simulated non-stationary autoregressive process of order 2 as the magnitude of its poles vary between 0 and 1 and the angle of the poles vary between 0 and pi. Our results show that the multitaper spectral estimate can be parameterized and is more accurate than others tested for non-sinusoidal signals. We also show applications to aero-acoustic data analysis.
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Date Issued: 2005
PURL: http://purl.flvc.org/fcla/dt/13235
Subject Headings: Spectral theory (Mathematics), Signal processing--Mathematics, System identification, Power spectra
Format: Document (PDF)

Title: The Comprehensive Evaluation of Performance and Environmental Influence on MPSK Modulated High-Speed Acoustic Communications in Shallow Water.
Creator: Proteau, Joshua C., Beaujean, Pierre-Philippe, Florida Atlantic University, College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: A daily study spanning a month of the shallow water acoustic channel was conducted to estimate the environmental influence on performance of an underwater acoustic communications system. An automated acoustic modem transmitted phase-coherent modulated sequences of identical data with 186 dB re IpPa source level, at coded rates from 4000 to 16000 bits/s with 4 or 8 kHz symbol bandwidth, three times daily for a month. A 64 channel Mills-Cross receiver array was used with horizontal and vertical...
Show moreA daily study spanning a month of the shallow water acoustic channel was conducted to estimate the environmental influence on performance of an underwater acoustic communications system. An automated acoustic modem transmitted phase-coherent modulated sequences of identical data with 186 dB re IpPa source level, at coded rates from 4000 to 16000 bits/s with 4 or 8 kHz symbol bandwidth, three times daily for a month. A 64 channel Mills-Cross receiver array was used with horizontal and vertical beams each containing 32 and 33 elements respectively, spaced 0.03 meters apart, with a sampling frequency of 72 kHz. Source and receiver were deployed at depths of 20 meters respectively, with a 720 meter separation range. Environmental measurements of wind velocity and direction, surface wave activity, current and sound velocity profiles, and tidal measurements were performed. Results demonstrate reliable achievement of high data-rate shallow water acoustic communications using phase-coherent modulation techniques.
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Date Issued: 2006
PURL: http://purl.flvc.org/fau/fd/FA00012541
Subject Headings: Underwater acoustics--Measurement, Artificial satellites in telecommunication, Underwater acoustic telemetry, Signal processing--Digital techniques
Format: Document (PDF)

Title: Matched filter bound for frequency-selective Nakagami fading channel.
Creator: Jiao, Jiemei., Florida Atlantic University, Aalo, Valentine A.
Abstract/Description: Based on the theoretical method developed by Clark and Greenstein for frequency-selective Rayleigh fading channel, we develop a general model for frequency-selective Nakagami fading channel. We derive analytical expressions of the average bit-error-rate in an ideal space diversity mobile radio receiver using the matched filter bound. Our simulation results show that the influences of the diversity order, the shape of the multipath delay profile, and delay spread of the multipath delay profile...
Show moreBased on the theoretical method developed by Clark and Greenstein for frequency-selective Rayleigh fading channel, we develop a general model for frequency-selective Nakagami fading channel. We derive analytical expressions of the average bit-error-rate in an ideal space diversity mobile radio receiver using the matched filter bound. Our simulation results show that the influences of the diversity order, the shape of the multipath delay profile, and delay spread of the multipath delay profile. Five shapes are considered in our simulation. Our simulation results highlight the importance of the ratio the normalized delay spread d. The results show that the delay profile is of no importance for $d<0.3,$ but can have a profound influence for $d\geq0.3.$
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Date Issued: 1995
PURL: http://purl.flvc.org/fcla/dt/15223
Subject Headings: Digital communications, Signal processing--Digital techniques, Radio--Transmitters and transmission--Fading
Format: Document (PDF)

FAU Theses and Dissertations (78)	+ -
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Underwater acoustic telemetry (10)	+ -

Underwater acoustics (9)	+ -
Wireless communication systems (7)	+ -
Wavelets (Mathematics) (6)	+ -
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Florida Atlantic University (62)	+ -
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Department of Ocean and Mechanical Engineering (29)	+ -
Erdol, Nurgun (12)	+ -

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