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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.
Show less  Date Issued
 1988
 PURL
 http://purl.flvc.org/fcla/dt/14467
 Subject Headings
 Signal processing, Electric stimulation
 Format
 Document (PDF)
 Title
 The deDopplerization of acoustic signatures.
 Creator
 Mouches, JeanMarc., 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.
Show less  Date Issued
 1988
 PURL
 http://purl.flvc.org/fcla/dt/14489
 Subject Headings
 Doppler effect, Signal processing
 Format
 Document (PDF)
 Title
 Timefrequency 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 WignerVille 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 discretetime/continuousfrequency smoothing kernel. Although both timevariant and shift invariant versions of the estimator are developed, emphasis is placed on the shiftinvariant version which is designed to estimate the WVS over an entire...
Show moreThis thesis provides detailed analysis and design techniques for WignerVille 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 discretetime/continuousfrequency smoothing kernel. Although both timevariant and shift invariant versions of the estimator are developed, emphasis is placed on the shiftinvariant 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 quasistationary based estimators, both for the autocorrelation estimator and for the WVS estimator. The concept of quasistationarity is extended to cyclostationary models, and we develop a novel measure of kernel smoothing and variance reduction termed the timebandwidth area. This is a generalization of timebandwidth 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 quasistationary 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 BlackmanTukey 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 tradeoff between variance reduction and bias using conventional estimators. In the examples any assumption of quasistationarity 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.
Show less  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, PierrePhilippe, 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 sidelooking sonar mounted on underwater platforms to achieve highresolution 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 sidelooking sonar mounted on underwater platforms to achieve highresolution 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 crosscorrelation. These time delays are then used to infer the pingtoping 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 nonstationary 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 nonstationary 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.
Show less  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.
Show less  Date Issued
 1988
 PURL
 http://purl.flvc.org/fcla/dt/14451
 Subject Headings
 Signal processingMathematical 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.
Show less  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
 Electroacoustics, 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 fourthorder cumulant matrixpencil.
 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 signaltonoise 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 fourthorder 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 signaltonoise 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 fourthorder cumulant matrices. The algorithm works in the presence of spatially and/or temporally correlated noise and, unlike most existing higherorder 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 higherorder cumulant based principle component analysis that works at low signaltonoise ratios in the presence of temporally and/or spatially correlated noise. (2) A novel definition of a spatial fourthorder cumulant matrix suited to blind source separation with nonequal gain and/or directional sensors. (3) The definition of a spatial fourthorder cumulant matrixpencil 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 fourthorder 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 matrixpencils 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 signaltonoise ratios of 6 dB, and using as few as 250 x 103 samples.
Show less  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
 DiazRobainas, Regino R., Florida Atlantic University, Huang, Ming Z.
 Abstract/Description

A PCbased Expert System that uses symbolic manipulations and an inference engine rulebased system to solve direct and inverse kinematics of revolutejointed 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, Computeraided design
 Format
 Document (PDF)
 Title
 Array processing techniques for frequency hopping multiple frequency shift keying longrange 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 noncoherent LMS. An Equal Gain Combiner (EGC) and a Maximum Likelihood (ML) were used to determine the performance of the coherent and noncoherent 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.
Show less  Date Issued
 2002
 PURL
 http://purl.flvc.org/fcla/dt/12914
 Subject Headings
 Underwater acoustics, Signal processingDigital techniques
 Format
 Document (PDF)
 Title
 Crosscorrelation of phase centers to estimate directionofarrival for a 3row bathymetric sidescan sonar.
 Creator
 Crenshaw, Edward T. G., Florida Atlantic University, LeBlanc, Lester R.
 Abstract/Description

A new method for calculating the directionofarrival (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 directionofarrival (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.
Show less  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 signalprocessing.
 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.
Show less  Date Issued
 1987
 PURL
 http://purl.flvc.org/fcla/dt/14427
 Subject Headings
 Electronic countermeasures, Signal processingDigital 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.
Show less  Date Issued
 1994
 PURL
 http://purl.flvc.org/fcla/dt/15065
 Subject Headings
 Acoustic imaging, Signal processing, Underwater acoustics
 Format
 Document (PDF)
 Title
 Wavelet transformbased 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.
Show less  Date Issued
 1993
 PURL
 http://purl.flvc.org/fcla/dt/12354
 Subject Headings
 Wavelets (Mathematics), Signal processingDigital techniques
 Format
 Document (PDF)
 Title
 Highspeed acoustic communication in shallow water using spatiotemporal adaptive array processing.
 Creator
 Beaujean, PierrePhilippe, 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 highspeed underwater acoustic communication with a fairly lowcomplexity of implementation is proposed. The proposed approach is to split the space and time processing into two separate suboptimal processes. As a result, processing complexity is significantly reduced and the instabilities associated with large tap vectors at large timefrequency spread products are reduced. The proposed spacetime signal processing method utilizes a different beamformer...
Show moreA novel method of achieving stable highspeed underwater acoustic communication with a fairly lowcomplexity of implementation is proposed. The proposed approach is to split the space and time processing into two separate suboptimal processes. As a result, processing complexity is significantly reduced and the instabilities associated with large tap vectors at large timefrequency spread products are reduced. The proposed spacetime 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 highspeed underwater acoustic communication in very shallow water using coherent modulation techniques, and offers a series of unique features: significant reduction of the signaltonoise and interference ratio (SNIR), improvement of the bandwidth efficiency by reduction of the forwarderror coding redundancy requirements, realtime evaluation of the timespread 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 seastate 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 realtime 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 multichannel adaptive receiver techniques in terms of stability, hardware requirements and channel tracking capability.
Show less  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 multiresolution subspaces can aid detection by providing noise whitening and signal matching. This thesis explores its use and proposes a technique for detection.
Show less  Date Issued
 1999
 PURL
 http://purl.flvc.org/fcla/dt/15740
 Subject Headings
 Speech processing systems, Signal processingDigital 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.
Show less  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 speedup. If this be the case, one may conclude that this model is an appropriate implementation for a linear predictive coding.
Show less  Date Issued
 1989
 PURL
 http://purl.flvc.org/fcla/dt/14498
 Subject Headings
 Signal processingDigital techniques, Signal processing, Programming languages (Electronic computers)
 Format
 Document (PDF)
 Title
 Maximum likelihood estimates of azimuth and elevation for a frequencyhopped active source using a tetrahedral ultrashort baseline.
 Creator
 Warin, Raphael., Florida Atlantic University, Beaujean, PierrePhilippe, College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
 Abstract/Description

UltraShortBaseLine (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 frequencyhopped 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 modemtype source...
Show moreUltraShortBaseLine (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 frequencyhopped 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 modemtype 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.
Show less  Date Issued
 2004
 PURL
 http://purl.flvc.org/fcla/dt/13135
 Subject Headings
 Underwater acousticsInstruments, Underwater acoustic telemetry, Signal processingTechnique, Adaptive signal processing
 Format
 Document (PDF)