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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: 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: Implementation and representation of the discrete wavelet transform.
Creator: Efthymoglou, George P., Florida Atlantic University, Erdol, Nurgun
Abstract/Description: This thesis presents a comprehensive analysis of a relatively new transform for discrete time signals, called the Discrete Wavelet Transform (DWT). We find how this transform is connected with the already existing theory of perfect reconstruction filter banks and the recently introduced theory of multiresolution analysis. We use the conditions obtained from these two theories in order to understand the construction of wavelet filters, which also generate continuous functions that prove to...
Show moreThis thesis presents a comprehensive analysis of a relatively new transform for discrete time signals, called the Discrete Wavelet Transform (DWT). We find how this transform is connected with the already existing theory of perfect reconstruction filter banks and the recently introduced theory of multiresolution analysis. We use the conditions obtained from these two theories in order to understand the construction of wavelet filters, which also generate continuous functions that prove to constitute an orthonormal basis for the L$\sp2$ space. We also investigate the connection of this transform to the sampled wavelet series of nonorthogonal functions with good time-frequency localization properties. Finally, we see the way that the DWT maps a discrete signal in the phase plane and the applications that such representations incorporate.
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Date Issued: 1993
PURL: http://purl.flvc.org/fcla/dt/14944
Subject Headings: Wavelets (Mathematics), Integrals, Singular, Signal processing--Digital techniques
Format: Document (PDF)

Title: Estimation of the scattering function of fading channels for acoustic communications in shallow waters.
Creator: Allemand, Vincent., Florida Atlantic University, Beaujean, Pierre-Philippe
Abstract/Description: The measurement of the Scattering function of time-variant fading channels is of strong interest in the field of underwater acoustic communications, as it indicates the limitations of the channel capacity. It also helps reducing the development time of acoustic communication systems and the need for on-site tests using so-called "fading simulators". The Scattering function is interpreted as the expected power received at a given time-delay and frequency shift for a given signal transmitted...
Show moreThe measurement of the Scattering function of time-variant fading channels is of strong interest in the field of underwater acoustic communications, as it indicates the limitations of the channel capacity. It also helps reducing the development time of acoustic communication systems and the need for on-site tests using so-called "fading simulators". The Scattering function is interpreted as the expected power received at a given time-delay and frequency shift for a given signal transmitted through the acoustic channel. It has been estimated using a fourth-moment method developed by Kailath from 18 to 30 kHz, 8-ms broad-band chirps and 20--28 kHz, 28-ms pseudo noise sequences. These signals were transmitted periodically in the shallow coastal waters of South Florida from a static source, and recorded from a 64-channel receiver array located at a distance of 1000 meters. Spatial beamforming has been applied to study the spatial sensitivity of the scattering function.
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Date Issued: 2005
PURL: http://purl.flvc.org/fcla/dt/13230
Subject Headings: Underwater acoustic telemetry, Signal processing--Digital techniques, Underwater acoustics--Mathematical models, Adaptive signal processing
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: 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: 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: 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: 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: Stochastic optimization of energy for multi-user wireless networks over fading channels.
Creator: Wang, Di, College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: Wireless devices in wireless networks are powered typically by small batteries that are not replaceable nor recharged in a convenient way. To prolong the operating lifetime of networks, energy efficiency is indicated as a critical issue and energy-efficient resource allocation designs have been extensively developed. We investigated energy-efficient schemes that prolong network operating lifetime in wireless sensor networks and in wireless relay networks. In Chapter 2, the energy-efficient...
Show moreWireless devices in wireless networks are powered typically by small batteries that are not replaceable nor recharged in a convenient way. To prolong the operating lifetime of networks, energy efficiency is indicated as a critical issue and energy-efficient resource allocation designs have been extensively developed. We investigated energy-efficient schemes that prolong network operating lifetime in wireless sensor networks and in wireless relay networks. In Chapter 2, the energy-efficient resource allocation that minimizes a general cost function of average user powers for small- or medium-scale wireless sensor networks, where the simple time-division multiple-access (TDMA) is adopted as the multiple access scheme. A class of Ç-fair cost-functions is derived to balance the tradeoff between efficiency and fairness in energy-efficient designs. Based on such cost functions, optimal channel-adaptive resource allocation schemes are developed for both single-hop and multi-hop TDMA sensor networks. In Chapter 3, optimal power control methods to balance the tradeoff between energy efficiency and fairness for wireless cooperative networks are developed. It is important to maximize power efficiency by minimizing power consumption for a given quality of service, such as the data rate; it is also equally important to evenly or fairly distribute power consumption to all nodes to maximize the network life. The optimal power control policy proposed is derived in a quasi-closed form by solving a convex optimization problem with a properly chosen cost-function. To further optimize a wireless relay network performance, an orthogonal frequency division multiplexing (OFDM) based multi-user wireless relay network is considered in Chapter 4., In the OFDM approach, each subcarrier is dynamically assigned to a source- destination link, and several relays assist communication between pairs of source-destination over their assigned subcarriers. Using a class of Ç-fair cost-functions to balance the tradeoff between energy efficiency and fairness, jointly with optimal subcarrier and power allocation schemes at the relays. Relevant algorithms are derived in quasi-closed form. Lastly, the proposed energy-efficient schemes are summarized and future work is discussed in Chapter 5.
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Date Issued: 2011
PURL: http://purl.flvc.org/FAU/3322519
Subject Headings: Stochastic processes, Data processing, Wireless communication systems, Mathematical models, Computer network protocols, Signal processing, Digital techniques, Code division multiple access, Waveless division multiplexing, Orthogonalization methods
Format: Document (PDF)

Digital techniques (5)	+ -
Signal processing--Digital techniques (5)	+ -
Mathematical models (3)	+ -
Signal processing (3)	+ -
Speech processing systems (3)	+ -

Wavelets (Mathematics) (3)	+ -
Adaptive signal processing (2)	+ -
Data processing (2)	+ -
Digital filters (Mathematics) (2)	+ -
Automatic speech recognition (1)	+ -
Blind source separation (1)	+ -
Cargo handling (1)	+ -
Code division multiple access (1)	+ -
Computer algorithms (1)	+ -
Computer file sharing (1)	+ -
Computer network protocols (1)	+ -
Data encryption (Computer science) (1)	+ -
Electric filters, Digital (1)	+ -
Fuzzy algorithms (1)	+ -
Integrals, Singular (1)	+ -
Mathematics (1)	+ -
Measurement (1)	+ -
Neural networks (Computer science) (1)	+ -
Noise control (1)	+ -
Numerical analysis (1)	+ -
Ocean circulation--Mathematical models (1)	+ -
Orthogonalization methods (1)	+ -
Radio frequency (1)	+ -
SIgnal processing (1)	+ -
Security measures (1)	+ -

College of Engineering and Computer Science (8)	+ -
Department of Computer and Electrical Engineering and Computer Science (6)	+ -
Florida Atlantic University (6)	+ -
Erdol, Nurgun (3)	+ -
Department of Ocean and Mechanical Engineering (2)	+ -

Allemand, Vincent. (1)	+ -
Basbug, Filiz. (1)	+ -
Beaujean, Pierre-Philippe (1)	+ -
Bulek, Savaskan. (1)	+ -
Charles E. Schmidt College of Science (1)	+ -
Charoenruengkit, Werayuth. (1)	+ -
Department of Mathematical Sciences (1)	+ -
Driscoll, Frederick R. (1)	+ -
Efthymoglou, George P. (1)	+ -
Finamore, Timothy. (1)	+ -
Kathiroli, Poorani. (1)	+ -
Schultz, Robert Carl. (1)	+ -
Tucker, Glenn C. (1)	+ -
Wang, Dali. (1)	+ -
Wang, Di (1)	+ -
Zilouchian, Ali (1)	+ -

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