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Title: A VLSI NMOS IMPLEMENTATION OF A BUILDING BLOCK PROCESSOR USING CORDIC ALGORITHMS (ARRAY PROCESSOR).
Creator: GIVEN, RAYMOND E., Florida Atlantic University, Shankar, Ravi
Date Issued: 1985
PURL: http://purl.flvc.org/fcla/dt/14247
Subject Headings: Array processors, Signal processing
Format: Document (PDF)

Title: A data acquisition and signal processing system for analysis of electromyogram signals.
Creator: Moller, Hans Carl., Florida Atlantic University, Szabo, Bela, College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: This thesis focuses on the development of a data acquisition and signal processing system for analysis of electromyogram (EMG) signals. The data acquisition system was based on a personal computer and was set up for simultaneous recording of three analog channels. Two of these channels were used to record the EMG signals from the triceps and biceps muscles respectively, and the third channel was used to record the acceleration signals obtained from an accelerometer placed on the subject's arm...
Show moreThis thesis focuses on the development of a data acquisition and signal processing system for analysis of electromyogram (EMG) signals. The data acquisition system was based on a personal computer and was set up for simultaneous recording of three analog channels. Two of these channels were used to record the EMG signals from the triceps and biceps muscles respectively, and the third channel was used to record the acceleration signals obtained from an accelerometer placed on the subject's arm. The objective of the signal processing was to find some characteristic parameters for the EMG signals, so that these parameters could be used in a microprocessor based system for Functional Electrical Stimulation (FES). Such a system may be useful in the rehabilitation of patients with partial paralysis of limbs as a result of brain damage.
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Date Issued: 1988
PURL: http://purl.flvc.org/fcla/dt/14467
Subject Headings: Signal processing, Electric stimulation
Format: Document (PDF)

Title: The de-Dopplerization of acoustic signatures.
Creator: Mouches, Jean-Marc., Florida Atlantic University, Glegg, Stewart A. L., College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: When acoustic measurements of moving vehicles are made by a stationary observer, the Doppler shift has two detrimental effects on the interpretation of the data. The spectra are smeared by the change in Doppler factor during the vehicle pass by, and the motion induced phase shift in the signals causes errors. The measured signals can be corrected back to source time if a moving time delay correction is applied. However, when the signals are sampled digitally this time delay correction...
Show moreWhen acoustic measurements of moving vehicles are made by a stationary observer, the Doppler shift has two detrimental effects on the interpretation of the data. The spectra are smeared by the change in Doppler factor during the vehicle pass by, and the motion induced phase shift in the signals causes errors. The measured signals can be corrected back to source time if a moving time delay correction is applied. However, when the signals are sampled digitally this time delay correction requires an estimate to be made of the signal level between samples. This can be achieved by using a digital filter with time varying coefficients which estimates the signal from at least two adjacent samples. Results of both numerical simulations and real applications of this technique will be given.
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Date Issued: 1988
PURL: http://purl.flvc.org/fcla/dt/14489
Subject Headings: Doppler effect, Signal processing
Format: Document (PDF)

Title: Time-frequency estimation for cyclostationary signals.
Creator: Frederick, Thomas James., Florida Atlantic University, Erdol, Nurgun, College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: This thesis provides detailed analysis and design techniques for Wigner-Ville spectrum (WVS) estimators for use with cyclostationary signals. The resulting class of estimators represent a newly defined subset of Cohen's class characterized by a mixed discrete-time/continuous-frequency smoothing kernel. Although both time-variant and shift invariant versions of the estimator are developed, emphasis is placed on the shift-invariant version which is designed to estimate the WVS over an entire...
Show moreThis thesis provides detailed analysis and design techniques for Wigner-Ville spectrum (WVS) estimators for use with cyclostationary signals. The resulting class of estimators represent a newly defined subset of Cohen's class characterized by a mixed discrete-time/continuous-frequency smoothing kernel. Although both time-variant and shift invariant versions of the estimator are developed, emphasis is placed on the shift-invariant version which is designed to estimate the WVS over an entire period from a single observation. Bias and variance expressions are derived for the new estimator, and these are compared with the general estimator. For this development, we also derive mean and covariance expressions for the general quasi-stationary based estimators, both for the autocorrelation estimator and for the WVS estimator. The concept of quasi-stationarity is extended to cyclostationary models, and we develop a novel measure of kernel smoothing and variance reduction termed the time-bandwidth area. This is a generalization of time-bandwidth product to describe arbitrary kernel functions, even those which are not governed by the uncertainty principle (such as the newly proposed estimators). The properties of the estimator are examined in terms of constraints on the smoothing kernel. In sharp contrast to the conventional estimators based on the quasi-stationary assumption, the low bias and low variance constraints for the new class of estimators do not contradict one another. The relationship between time dependent spectral estimation for nonstationary processes and classical Blackman-Tukey type spectral estimation for stationary processes is developed next. Using examples the utility of the new estimator kernels are shown. It is seen that in random or noisy environments it may be difficult to achieve a reasonable trade-off between variance reduction and bias using conventional estimators. In the examples any assumption of quasi-stationarity sufficient to produce a low variance estimate would destroy many or all of the nonstationary features of the signal. However, since the signals are cyclostationary we can employ the new class of estimators to achieve an excellent balance between bias and variance reduction.
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Date Issued: 1997
PURL: http://purl.flvc.org/fcla/dt/12537
Subject Headings: Signal processing, Wigner distribution
Format: Document (PDF)

Title: APPLICATION OF SIGNAL DECOMPOSITION TO IMPROVE TIME DELAY ESTIMATES FOR SYNTHETIC APERTURE SONAR MOTION COMPENSATION.
Creator: Gazagnaire, Julia, Beaujean, Pierre-Philippe, Florida Atlantic University, Department of Ocean and Mechanical Engineering, College of Engineering and Computer Science
Abstract/Description: Synthetic Aperture Sonar (SAS) provides the best opportunity for side-looking sonar mounted on underwater platforms to achieve high-resolution images. However, SAS processing requires strict constraints on resolvable platform motion. The most common approach to estimate this motion is to use the Redundant Phase Center (RPC) technique. Here the ping interval is set, such that a portion of the sonar array overlaps as the sensor moves forward. The time delay between the pings received on these...
Show moreSynthetic Aperture Sonar (SAS) provides the best opportunity for side-looking sonar mounted on underwater platforms to achieve high-resolution images. However, SAS processing requires strict constraints on resolvable platform motion. The most common approach to estimate this motion is to use the Redundant Phase Center (RPC) technique. Here the ping interval is set, such that a portion of the sonar array overlaps as the sensor moves forward. The time delay between the pings received on these overlapping elements is estimated using cross-correlation. These time delays are then used to infer the pingto-ping vehicle motion. Given the stochastic nature of the operational environment, some level of decorrelation between these two signals is likely. In this research, two iterative signal decomposition methods well suited for nonlinear and non-stationary signals, are investigated for their potential to improve the Time Delay Estimation (TDE). The first of this type, the Empirical Mode Decomposition (EMD) was introduced by Huang in the seminal paper, The empirical mode decomposition and the Hilbert spectrum for nonlinear and non-stationary time series analysis and is the foundation for the algorithms used in this research. This method decomposes a signal into a finite sequence of simple components termed Intrinsic Mode Functions (IMFs). The Iterative Filter (IF) approach, developed by Lin, Wang and Zhou, builds on the EMD framework. The sonar signals considered in this research are complex baseband signals. Both the IF and EMD algorithms were designed to decompose real signals. However, the IF variant, the Multivariate Fast Iterative Filtering (MFIF) Algorithm, developed by Cicone, and the EMD variant, the Fast and Adaptive Multivariate Empirical Mode Decomposition (FAMVEMD) algorithm, developed by Thirumalaisamy and Ansell, preserve both the magnitude and phase in the decomposition and hence were chosen for this analysis.
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Date Issued: 2021
PURL: http://purl.flvc.org/fau/fd/FA00013795
Subject Headings: Sonar, Signal processing, Synthetic apertures
Format: Document (PDF)

Title: Discrete signal representation using triangular basis functions.
Creator: Nallur, Padmanabha., Florida Atlantic University, Hartt, William H., College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: This thesis deals with the representation of discrete signals using triangular basis functions. Signals are usually represented by Fourier series expansions where the basis functions are cosine and sine functions which are all mutually orthogonal. The triangular basis functions used here are called TRIC (triangular cosine) and TRIS (triangular sine) functions. The TRIC and TRIS functions are like their cosine and sine function counterparts except that they are linear. The TRIC and TRIS...
Show moreThis thesis deals with the representation of discrete signals using triangular basis functions. Signals are usually represented by Fourier series expansions where the basis functions are cosine and sine functions which are all mutually orthogonal. The triangular basis functions used here are called TRIC (triangular cosine) and TRIS (triangular sine) functions. The TRIC and TRIS functions are like their cosine and sine function counterparts except that they are linear. The TRIC and TRIS functions are not all mutually orthogonal, though most of them are. A matrix method of representing discrete signals using TRIC and TRIS functions is presented. A discrete triangular transform matrix is developed and a method of deriving this matrix is presented. A Fortran program is written to derive the discrete triangular transform matrix and to prove the reconstruction of several basic functions like impulse, step, pulse and sinusoidal waveforms.
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Date Issued: 1988
PURL: http://purl.flvc.org/fcla/dt/14451
Subject Headings: Signal processing--Mathematical models
Format: Document (PDF)

Title: Multiresolution analysis of glottal pulses.
Creator: Miguel, Agnieszka C., Florida Atlantic University, Erdol, Nurgun, College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: Glottal pulse models provide vocal tract excitation signals which are used in producing high quality speech. Most of the currently used glottal pulse models are obtained by concatenating a small number of parametric functions over the pitch period. In this thesis, a new glottal pulse model is proposed. It is an alternative approach, which is based on the projection of glottal volume velocity over multiresolution subspaces spanned by wavelets and scaling functions. A detailed multiresolution...
Show moreGlottal pulse models provide vocal tract excitation signals which are used in producing high quality speech. Most of the currently used glottal pulse models are obtained by concatenating a small number of parametric functions over the pitch period. In this thesis, a new glottal pulse model is proposed. It is an alternative approach, which is based on the projection of glottal volume velocity over multiresolution subspaces spanned by wavelets and scaling functions. A detailed multiresolution analysis of the glottal models is performed using the compactly supported orthogonal Daubechies wavelets. The wavelet representation has been tested for optimality in terms of the reconstruction error and the energy compactness of the coefficients. It is demonstrated that by choosing proper parameters of the wavelet representation, high compression ratios and low rms error can be achieved.
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Date Issued: 1996
PURL: http://purl.flvc.org/fcla/dt/15334
Subject Headings: Signal processing, Speech processing systems, Wavelets (Mathematics)--Data processing
Format: Document (PDF)

Title: Acoustic impulse response mapping for acoustic communications in shallow water.
Creator: Caimi, F. M., Tongta, R., Harbor Branch Oceanographic Institute
Date Issued: 1998
PURL: http://purl.flvc.org/FCLA/DT/3183706
Subject Headings: Electro-acoustics, Sound --Measurement, Acoustical engineering, Digital communications, Signal processing, Signals and signaling, Underwater acoustics, Signal processing --Digital techniques
Format: Document (PDF)

Title: Blind source separation using a spatial fourth-order cumulant matrix-pencil.
Creator: Dishman, John Fitzgerald., Florida Atlantic University, Aalo, Valentine A., College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: The research presented investigates the use of cumulants in conjunction with a spectral estimation technique of the signal subspace to perform the blind separation of statistically independent signals with low signal-to-noise ratios under a narrowband assumption. A new blind source separation (BSS) algorithm is developed that makes use of the generalized eigen analysis of a matrix pencil defined on two similar spatial fourth-order cumulant matrices. The algorithm works in the presence of...
Show moreThe research presented investigates the use of cumulants in conjunction with a spectral estimation technique of the signal subspace to perform the blind separation of statistically independent signals with low signal-to-noise ratios under a narrowband assumption. A new blind source separation (BSS) algorithm is developed that makes use of the generalized eigen analysis of a matrix pencil defined on two similar spatial fourth-order cumulant matrices. The algorithm works in the presence of spatially and/or temporally correlated noise and, unlike most existing higher-order BSS techniques, is based on a spectral estimation technique rather than a closed loop optimization of a contrast function, for which the convergence is often problematic. The dissertation makes several contributions to the area of blind source separation. These include: (1) Development of a robust blind source separation technique that is based on higher-order cumulant based principle component analysis that works at low signal-to-noise ratios in the presence of temporally and/or spatially correlated noise. (2) A novel definition of a spatial fourth-order cumulant matrix suited to blind source separation with non-equal gain and/or directional sensors. (3) The definition of a spatial fourth-order cumulant matrix-pencil using temporal information. (4) The concept of separation power efficiency (SPE) as a measure of the algorithm's performance. Two alternative definitions for the spatial fourth-order cumulant matrix that are found in the literature are also presented and used by the algorithm for comparison. Additionally, the research contributes the concept of wide sense equivalence between matrix-pencils to the field of matrix algebra. The algorithm's performance is verified by computer simulation using realistic digital communications signals in white noise. Random mixing matrices are generated to ensure the algorithm's performance is independent of array geometry. The computer results are promising and show that the algorithm works well down to input signal-to-noise ratios of -6 dB, and using as few as 250 x 103 samples.
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Date Issued: 2001
PURL: http://purl.flvc.org/fcla/dt/11963
Subject Headings: Matrix pencils, Adaptive signal processing, Matrices
Format: Document (PDF)

Title: Applications of logic programming to manipulator kinematics and problems in controls and signal processing.
Creator: Diaz-Robainas, Regino R., Florida Atlantic University, Huang, Ming Z.
Abstract/Description: A PC-based Expert System that uses symbolic manipulations and an inference engine rule-based system to solve direct and inverse kinematics of revolute-jointed manipulators of arbitrary configuration is presented and discussed. Similar applications in the areas of Discrete Signal Processing and Optimal Control are analyzed.
Date Issued: 1992
PURL: http://purl.flvc.org/fcla/dt/14794
Subject Headings: Signal processing, Kinematics, Computer-aided design
Format: Document (PDF)

Title: Array processing techniques for frequency hopping multiple frequency shift keying long-range communications.
Creator: Bernault, Emmanuel Pierre., Florida Atlantic University, Schock, Steven G.
Abstract/Description: Underwater communication is an important component of Autonomous Underwater Vehicle (AUV) operations. Communicating underwater is limited to very low communication rates without the use of processing techniques that mitigate the influence of the acoustic channel. This thesis develops array processing techniques for frequency hopping and multiple frequency shift keying to achieve long range, reliable high speed communications. The thesis makes the comparison between two techniques for...
Show moreUnderwater communication is an important component of Autonomous Underwater Vehicle (AUV) operations. Communicating underwater is limited to very low communication rates without the use of processing techniques that mitigate the influence of the acoustic channel. This thesis develops array processing techniques for frequency hopping and multiple frequency shift keying to achieve long range, reliable high speed communications. The thesis makes the comparison between two techniques for calculating beamforming coefficients: a coherent Least Mean Square (LMS) adaptive filter and a non-coherent LMS. An Equal Gain Combiner (EGC) and a Maximum Likelihood (ML) were used to determine the performance of the coherent and non-coherent LMS. The results show that by using the coherent LMS, the ML or the EGC, communications at rates of 493 bit per second (bps) and 370bps can be achieved with no frame error at 5km in 40 feet of water using 16.3kHz of bandwidth centered at 25kHz.
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Date Issued: 2002
PURL: http://purl.flvc.org/fcla/dt/12914
Subject Headings: Underwater acoustics, Signal processing--Digital techniques
Format: Document (PDF)

Title: Cross-correlation of phase centers to estimate direction-of-arrival for a 3-row bathymetric sidescan sonar.
Creator: Crenshaw, Edward T. G., Florida Atlantic University, LeBlanc, Lester R.
Abstract/Description: A new method for calculating the direction-of-arrival (DOA), and thus the bathymetry of the seafloor, is presented. This method will calculate the DOA directly from the phase difference between the phase centers of the array. In parallel, a bathymetric sidescan sonar system originally built at Woods Hole and now here at Florida Atlantic University's Department of Ocean Engineering, was completed. Once this system was working, the above mentioned signal analysis regime will be implemented on...
Show moreA new method for calculating the direction-of-arrival (DOA), and thus the bathymetry of the seafloor, is presented. This method will calculate the DOA directly from the phase difference between the phase centers of the array. In parallel, a bathymetric sidescan sonar system originally built at Woods Hole and now here at Florida Atlantic University's Department of Ocean Engineering, was completed. Once this system was working, the above mentioned signal analysis regime will be implemented on actual data to test its validity.
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Date Issued: 2001
PURL: http://purl.flvc.org/fcla/dt/12738
Subject Headings: Sidescan sonar, Bathymetric maps, Signal processing
Format: Document (PDF)

Title: Digital techniques for electronic countermeasures signal-processing.
Creator: Lopez, Juan J., Florida Atlantic University, Erdol, Nurgun, College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: The purpose of this thesis is to show the use of digital techniques for Electronic Countermeasures (ECM) signal processing. The main objective is the use of only digital circuitry for the processing of the ECM signals. A recent design of an ECM controller called the Oscillator Waveform Controller (OWC) follows this philosophy. The OWC digitally controls the generation of its nine jamming modes plus the modes generated by the other ECM modules within the ECM system. The use of advance...
Show moreThe purpose of this thesis is to show the use of digital techniques for Electronic Countermeasures (ECM) signal processing. The main objective is the use of only digital circuitry for the processing of the ECM signals. A recent design of an ECM controller called the Oscillator Waveform Controller (OWC) follows this philosophy. The OWC digitally controls the generation of its nine jamming modes plus the modes generated by the other ECM modules within the ECM system. The use of advance microcircuitry technology allows the OWC the capability of controlling all the parameters within an ECM system. The most desirable feature of the OWC is the use of high level communications for programming ECM mode parameters from an external computer or terminal and digitally storing this parameters upon removal of power.
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Date Issued: 1987
PURL: http://purl.flvc.org/fcla/dt/14427
Subject Headings: Electronic countermeasures, Signal processing--Digital techniques
Format: Document (PDF)

Title: A broadband signal processor for acoustic imaging using ambient noise.
Creator: Olivieri, Marc P., Florida Atlantic University, Glegg, Stewart A. L., College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: Buckingham et al. (Nature Vol. 356, p 327) first introduced the concept of acoustic imaging using ambient noise as a method for passively detecting objects in the ocean. Several analytical studies followed, and it was shown that a two dimensional acoustic image could be obtained based on this approach, and that at least 900 pixels are necessary to restitute the details of spherical objects placed in an underwater sound channel. The alternative approach described in this paper is based on a...
Show moreBuckingham et al. (Nature Vol. 356, p 327) first introduced the concept of acoustic imaging using ambient noise as a method for passively detecting objects in the ocean. Several analytical studies followed, and it was shown that a two dimensional acoustic image could be obtained based on this approach, and that at least 900 pixels are necessary to restitute the details of spherical objects placed in an underwater sound channel. The alternative approach described in this paper is based on a signal processing which uses the broadband nature of the ambient noise in the ocean, and therefore, optimizes the use of available sound energy scattered by the object. Images with thousands of pixels can be obtained using a relatively small number of transducers. This method has been validated using simple experiments in air, scaled to represent an ocean application, and results showing images of various objects will be presented.
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Date Issued: 1994
PURL: http://purl.flvc.org/fcla/dt/15065
Subject Headings: Acoustic imaging, Signal processing, Underwater acoustics
Format: Document (PDF)

Title: Wavelet transform-based digital signal processing.
Creator: Basbug, Filiz., Florida Atlantic University, Erdol, Nurgun, College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: This study deals with applying the wavelet transform to mainly two different areas of signal processing: adaptive signal processing, and signal detection. It starts with background information on the theory of wavelets with an emphasis on the multiresolution representation of signals by the wavelet transform in Chapter 1. Chapter 2 begins with an overview of adaptive filtering in general and extends it to transform domain adaptive filtering. Later in the chapter, a novel adaptive filtering...
Show moreThis study deals with applying the wavelet transform to mainly two different areas of signal processing: adaptive signal processing, and signal detection. It starts with background information on the theory of wavelets with an emphasis on the multiresolution representation of signals by the wavelet transform in Chapter 1. Chapter 2 begins with an overview of adaptive filtering in general and extends it to transform domain adaptive filtering. Later in the chapter, a novel adaptive filtering architecture using the wavelet transform is introduced. The performance of this new structure is evaluated by using the LMS algorithm with variations in step size. As a result of this study, the wavelet transform based adaptive filter is shown to reduce the eigenvalue ratio, or condition number, of the input signal. As a result, the new structure is shown to have faster convergence, implying an improvement in the ability to track rapidly changing signals. Chapter 3 deals with signal detection with the help of the wavelet transform. One scheme studies signal detection by projecting the input signal onto different scales. The relationship between this approach and that of matched filtering is established. Then the effect of different factors on signal detection with the wavelet transform is examined. It is found that the method is robust in the presence of white noise. Also, the wavelets are analyzed as eigenfunctions of a certain random process, and how this gives way to optimal receiver design is shown. It is further demonstrated that the design of an optimum receiver leads to the wavelet transform based adaptive filter structure described in Chapter 2.
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Date Issued: 1993
PURL: http://purl.flvc.org/fcla/dt/12354
Subject Headings: Wavelets (Mathematics), Signal processing--Digital techniques
Format: Document (PDF)

Title: High-speed acoustic communication in shallow water using spatio-temporal adaptive array processing.
Creator: Beaujean, Pierre-Philippe, Florida Atlantic University, LeBlanc, Lester R., College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: A novel method of achieving stable high-speed underwater acoustic communication with a fairly low-complexity of implementation is proposed. The proposed approach is to split the space and time processing into two separate sub-optimal processes. As a result, processing complexity is significantly reduced and the instabilities associated with large tap vectors at large time-frequency spread products are reduced. The proposed space-time signal processing method utilizes a different beamformer...
Show moreA novel method of achieving stable high-speed underwater acoustic communication with a fairly low-complexity of implementation is proposed. The proposed approach is to split the space and time processing into two separate sub-optimal processes. As a result, processing complexity is significantly reduced and the instabilities associated with large tap vectors at large time-frequency spread products are reduced. The proposed space-time signal processing method utilizes a different beamformer optimization strategy compared to the time domain optimization strategy. This allows to separately adjust the adaptation parameters for the spatial and temporal characteristics of the signal, which have vastly different requirements. The time domain signal is subject to variations in phase that require rapid filter updates whereas the directional characteristics of the signal do not vary appreciably over the message length and do not require a rapid adaptation response. The proposed method allows for high-speed underwater acoustic communication in very shallow water using coherent modulation techniques, and offers a series of unique features: significant reduction of the signal-to-noise and interference ratio (SNIR), improvement of the bandwidth efficiency by reduction of the forward-error coding redundancy requirements, real-time evaluation of the time-spread by Doppler spread product (BL) and channel stability estimate. Experimental results demonstrate that stable acoustic communication can be achieved at rates of 32000 bits per second at a distance of 3 km, in 40 feet of water and in sea-state 2 conditions. Fast and slow fading properties of the channel are measured, as the BL product can vary by a decade in 116 ms, and by two decades within minutes, from 0.001 to 0.1. The real-time analysis shows a strong correlation between time spread, Doppler spread, spatial coherence of the acoustic channel and communication performance. Overall, this research provides more scientific and experimental ground to understand the limitations of multi-channel adaptive receiver techniques in terms of stability, hardware requirements and channel tracking capability.
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Date Issued: 2001
PURL: http://purl.flvc.org/fcla/dt/11952
Subject Headings: Underwater acoustic telemetry, Adaptive signal processing
Format: Document (PDF)

Title: Voice activity detection over multiresolution subspaces.
Creator: Schultz, Robert Carl., Florida Atlantic University, Erdol, Nurgun, College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: Society's increased demand for communications requires searching for techniques that preserve bandwidth. It has been observed that much of the time spent during telephone communications is actually idle time with no voice activity present. Detecting these idle periods and preventing transmission during these idle periods can aid in reducing bandwidth requirements during high traffic periods. While techniques exist to perform this detection, certain types of noise can prove difficult at best...
Show moreSociety's increased demand for communications requires searching for techniques that preserve bandwidth. It has been observed that much of the time spent during telephone communications is actually idle time with no voice activity present. Detecting these idle periods and preventing transmission during these idle periods can aid in reducing bandwidth requirements during high traffic periods. While techniques exist to perform this detection, certain types of noise can prove difficult at best for signal detection. The use of wavelets with multi-resolution subspaces can aid detection by providing noise whitening and signal matching. This thesis explores its use and proposes a technique for detection.
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Date Issued: 1999
PURL: http://purl.flvc.org/fcla/dt/15740
Subject Headings: Speech processing systems, Signal processing--Digital techniques, Wavelets (Mathematics)
Format: Document (PDF)

Title: The directionality of noise created by turbulent flow over roughness.
Creator: Kaufman, Gerard P., College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: Flow over a rough surface is known to radiate sound as a dipole source that is directional. In order to better understand this source, measurements are being made in a wind tunnel using a microphone array. The measurements collected by a microphone array are beamformed to give a source image and can be deconvolved with an assumed point spread function in order to obtain the source levels. This thesis considers alternative analysis algorithms that can be used to analyze wind tunnel data. Only...
Show moreFlow over a rough surface is known to radiate sound as a dipole source that is directional. In order to better understand this source, measurements are being made in a wind tunnel using a microphone array. The measurements collected by a microphone array are beamformed to give a source image and can be deconvolved with an assumed point spread function in order to obtain the source levels. This thesis considers alternative analysis algorithms that can be used to analyze wind tunnel data. Only numerical examples of how these algorithms work will be presented and the analysis of real data will be considered in later studies. It will be shown how estimates can be made of the source directivity by comparing the measured data with a theoretical source model and minimizing the error between the model and the measurements.
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Date Issued: 2011
PURL: http://purl.flvc.org/FAU/3171394
Subject Headings: Electromagnetic fields, Signal processing, Digital techniques, Noise control, Adaptive signal processing, Acoustic emission, Measurement
Format: Document (PDF)

Title: Concurrent linear predictive coding.
Creator: McLean, William Gregory., Florida Atlantic University, Erdol, Nurgun, College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: This thesis describes a software model of a Linear Predictive Coding (L.P.C.) that is written in the Ada language. The novel feature of this program is that it attempts to execute the maximum possible number of concurrent arithmetic operations in the L.P.C. algorithm. Each arithmetic operation is implemented by an active process which is the "task" construct in the Ada language. The computational part of the algorithm is implemented as a wavefront array of computing tasks. These computational...
Show moreThis thesis describes a software model of a Linear Predictive Coding (L.P.C.) that is written in the Ada language. The novel feature of this program is that it attempts to execute the maximum possible number of concurrent arithmetic operations in the L.P.C. algorithm. Each arithmetic operation is implemented by an active process which is the "task" construct in the Ada language. The computational part of the algorithm is implemented as a wavefront array of computing tasks. These computational arrays are driven by a driver task which coordinates the flow of data into and out of the computing surfaces. If the inter process communications time between tasks is small, then this model shows a potential for speed-up. If this be the case, one may conclude that this model is an appropriate implementation for a linear predictive coding.
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Date Issued: 1989
PURL: http://purl.flvc.org/fcla/dt/14498
Subject Headings: Signal processing--Digital techniques, Signal processing, Programming languages (Electronic computers)
Format: Document (PDF)

Title: Maximum likelihood estimates of azimuth and elevation for a frequency-hopped active source using a tetrahedral ultra-short baseline.
Creator: Warin, Raphael., Florida Atlantic University, Beaujean, Pierre-Philippe, College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: Ultra-Short-BaseLine (USBL) is the most practical underwater acoustic positioning system for autonomous underwater vehicles because of its small space requirement. The objective of this research is to develop a USBL system capable of estimating a source location transmitting frequency-hopped tones sequences. Such sequences are characteristic of spread spectrum signaling used in underwater acoustic communication network. It must be able to provide azimuth and elevation of a modem-type source...
Show moreUltra-Short-BaseLine (USBL) is the most practical underwater acoustic positioning system for autonomous underwater vehicles because of its small space requirement. The objective of this research is to develop a USBL system capable of estimating a source location transmitting frequency-hopped tones sequences. Such sequences are characteristic of spread spectrum signaling used in underwater acoustic communication network. It must be able to provide azimuth and elevation of a modem-type source with an accuracy of 0.3 degrees; for both angles using the synchronization stage of the transmission. The acoustic antenna is composed of four transducers arranged as a tetrahedron. Using the model of Quazi and Lerro, which provides an expression for the variance of the bearing angle, azimuth and elevation of the transmitter are estimated employing maximum likelihood estimation. This system is simulated, tested and calibrated in a tank. Simulated results satisfy the requirement with a SNR of 32dB and 8 symbols. The latest experimental measurements present an accuracy of 3 degrees.
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Date Issued: 2004
PURL: http://purl.flvc.org/fcla/dt/13135
Subject Headings: Underwater acoustics--Instruments, Underwater acoustic telemetry, Signal processing--Technique, Adaptive signal processing
Format: Document (PDF)

FAU Theses and Dissertations (78)	+ -
HBOI Faculty Contributions (2)	+ -
Honors Student Theses (1)	+ -

Signal processing (29)	+ -
Adaptive signal processing (20)	+ -
Signal processing--Digital techniques (18)	+ -
Digital techniques (12)	+ -
Underwater acoustic telemetry (10)	+ -

Underwater acoustics (9)	+ -
Wireless communication systems (7)	+ -
Wavelets (Mathematics) (6)	+ -
Speech processing systems (5)	+ -
Digital communications (4)	+ -
Underwater navigation (4)	+ -
Acoustical engineering (3)	+ -
Adaptive signal processing. (3)	+ -
Code division multiple access (3)	+ -
Design and construction (3)	+ -
Inertial navigation systems (3)	+ -
MIMO systems (3)	+ -
Mathematical models (3)	+ -
Measurement (3)	+ -
Oceanographic instruments (3)	+ -
Array processors (2)	+ -
Coding theory (2)	+ -
Computer network protocols (2)	+ -
Data processing (2)	+ -
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