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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: 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: 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: Model-based classification of speech audio.
Creator: Thoman, Chris., College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: This work explores the process of model-based classification of speech audio signals using low-level feature vectors. The process of extracting low-level features from audio signals is described along with a discussion of established techniques for training and testing mixture model-based classifiers and using these models in conjunction with feature selection algorithms to select optimal feature subsets. The results of a number of classification experiments using a publicly available speech...
Show moreThis work explores the process of model-based classification of speech audio signals using low-level feature vectors. The process of extracting low-level features from audio signals is described along with a discussion of established techniques for training and testing mixture model-based classifiers and using these models in conjunction with feature selection algorithms to select optimal feature subsets. The results of a number of classification experiments using a publicly available speech database, the Berlin Database of Emotional Speech, are presented. This includes experiments in optimizing feature extraction parameters and comparing different feature selection results from over 700 candidate feature vectors for the tasks of classifying speaker gender, identity, and emotion. In the experiments, final classification accuracies of 99.5%, 98.0% and 79% were achieved for the gender, identity and emotion tasks respectively.
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Date Issued: 2009
PURL: http://purl.flvc.org/FAU/210518
Subject Headings: Signal processing, Digital techniques, Speech processing systems, Sound, Recording and reproducing, Digital techniques, Pattern recognition systems
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: Turbo-coded frequency division multiplexing for underwater acoustic communications between 60 kHz and 90 kHz.
Creator: Pajovic, Milutin., College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: The Intermediate Frequency Acoustic Modem (IFAM), developed by Dr. Beaujean, is designed to transmit the command-and-control messages from the top-side to the wet-side unit in ports and very shallow waters. This research presents the design of the turbo coding scheme and its implementation in the IFAM modem with the purpose of meeting a strict requirement for the IFAM error rate performance. To simulate the coded IFAM, a channel simulator is developed. It is basically a multi-tap filter whose...
Show moreThe Intermediate Frequency Acoustic Modem (IFAM), developed by Dr. Beaujean, is designed to transmit the command-and-control messages from the top-side to the wet-side unit in ports and very shallow waters. This research presents the design of the turbo coding scheme and its implementation in the IFAM modem with the purpose of meeting a strict requirement for the IFAM error rate performance. To simulate the coded IFAM, a channel simulator is developed. It is basically a multi-tap filter whose parameters are set depending on the channel geometry and system specifics. The simulation results show that the turbo code is able to correct 89% of the messages received with errors in the hostile channel conditions. The Bose-Chadhuri-Hocquenghem (BCH) coding scheme corrects less that 15% of these messages. The other simulation results obtained for the system operation in different shallow water settings are presented.
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Date Issued: 2009
PURL: http://purl.flvc.org/FAU/215291
Subject Headings: Underwater acoustics, Measurement, Coding theory, Signal processing, Digital techniques
Format: Document (PDF)

Title: Design of a frequency shift keying array receiver for the acoustic modem.
Creator: Boubli, Cecile., Florida Atlantic University, LeBlanc, Lester R.
Abstract/Description: Currently, our acoustic modems are used to communicate underwater to Autonomous Underwater Vehicles AUVs. These modems have only one sensor and can transmit at low data rates (from 200 to 1200 bits per second) using Frequency Shift Keying (FSK) modulation. A two-dimensional array receiver (MillsCross) has been developed to receive underwater signals with more reliability, at a higher data rate (about 30,000 bits per second). This array has been designed to operate with Phase Shift Keying...
Show moreCurrently, our acoustic modems are used to communicate underwater to Autonomous Underwater Vehicles AUVs. These modems have only one sensor and can transmit at low data rates (from 200 to 1200 bits per second) using Frequency Shift Keying (FSK) modulation. A two-dimensional array receiver (MillsCross) has been developed to receive underwater signals with more reliability, at a higher data rate (about 30,000 bits per second). This array has been designed to operate with Phase Shift Keying modulated signals. The purpose of this thesis is to design and implement a signal processing software to demodulate and decode FSK signals acquired by the MillsCross. By taking advantage of the spatial gain of the MillsCross receiver array, higher reliability and longer ranges are expected using FSK, in addition to achieving compatibility between the two systems. This software includes a robust synchronization scheme, a spatial and an equalizing filter, a time-window self-adjusting process and the error control decoding.
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Date Issued: 2000
PURL: http://purl.flvc.org/fcla/dt/15788
Subject Headings: Underwater acoustics, Modems, Signal processing--Digital techniques
Format: Document (PDF)

Title: Multi-pulse excited linear prediction for synthesizing the guitar.
Creator: Leeds, David Scott., Florida Atlantic University, Erdol, Nurgun, College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: In this paper we propose using parametric modeling by employing a Multi-Pulse Excited Linear Predictive Coded (MPE-LPC) filter to synthesize the guitar. First we introduce different methods for sound synthesis. A detailed discussion including the derivation of LPC and MPE presented. Then we study the impulse and steady state response of the guitar signal. An implementation of the MPE-LPC method to model the guitar is covered in detail and opportunities to improve the compression ratio are...
Show moreIn this paper we propose using parametric modeling by employing a Multi-Pulse Excited Linear Predictive Coded (MPE-LPC) filter to synthesize the guitar. First we introduce different methods for sound synthesis. A detailed discussion including the derivation of LPC and MPE presented. Then we study the impulse and steady state response of the guitar signal. An implementation of the MPE-LPC method to model the guitar is covered in detail and opportunities to improve the compression ratio are discussed. We then present simulation results with a set of fixed parameters, which are used as a benchmark to observe performance trade-offs by varying the model parameters to improve the compression ratio. Finally, we discuss limitations of the modeling algorithm for use with wide-band transient musical sounds and possible applications of the MPE-LPC model as a method to dynamically calculate samples for use with wavetable synthesis of steady state sounds.
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Date Issued: 2001
PURL: http://purl.flvc.org/fcla/dt/12759
Subject Headings: Computer music, Electric guitar, Signal processing--Digital techniques
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: Synchronous spatial array processing for underwater vehicle tracking.
Creator: Normand, Olivier., Florida Atlantic University, Schock, Steven G., College of Engineering and Computer Science, Department of Ocean and Mechanical Engineering
Abstract/Description: Navigation of Unmanned Underwater Vehicles (UUVs) is commonly assisted in confined areas by acoustic positioning systems. The Department of Ocean Engineenng at Florida Atlantic University is developing an altemative system based on submerged modems. This thesis describes an optimal target location estimation technique using a multi-channel spatial receiver array (Millscross) used as a development tool combined with a synchronous modem and transponder network mounted on buoys and UUVs. The...
Show moreNavigation of Unmanned Underwater Vehicles (UUVs) is commonly assisted in confined areas by acoustic positioning systems. The Department of Ocean Engineenng at Florida Atlantic University is developing an altemative system based on submerged modems. This thesis describes an optimal target location estimation technique using a multi-channel spatial receiver array (Millscross) used as a development tool combined with a synchronous modem and transponder network mounted on buoys and UUVs. The Millscross provides a reference to evaluate the performance of the navigation estimator. Spatial array principles are used to develop decoding and beamforming techniques to process modem messages, enabling the end user (the UUV) to estimate in real-time its own position and navigate in space. A simulation was used to compare actual results with theory and determine the processing and decoding algorithms. These algorithms were then applied to real data to estimate the target position (direction of arrival and geodetic coordinates).
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Date Issued: 2002
PURL: http://purl.flvc.org/fcla/dt/12917
Subject Headings: Underwater navigation, Underwater acoustics, Signal processing--Digital techniques
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: 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: Image improvement using dynamic optical low-pass filter.
Creator: Petljanski, Branko., College of Engineering and Computer Science, Department of Computer and Electrical Engineering and Computer Science
Abstract/Description: Professional imaging systems, particularly motion picture cameras, usually employ larger photosites and lower pixel counts than many amateur cameras. This results in the desirable characteristics of improved dynamic range, signal to noise and sensitivity. However, high performance optics often have frequency response characteristics that exceed the Nyquist limit of the sensor, which, if not properly addressed, results in aliasing artifacts in the captured image. Most contemporary still and...
Show moreProfessional imaging systems, particularly motion picture cameras, usually employ larger photosites and lower pixel counts than many amateur cameras. This results in the desirable characteristics of improved dynamic range, signal to noise and sensitivity. However, high performance optics often have frequency response characteristics that exceed the Nyquist limit of the sensor, which, if not properly addressed, results in aliasing artifacts in the captured image. Most contemporary still and video cameras employ various optically birefringent materials as optical low-pass filters (OLPF) in order to minimize aliasing artifacts in the image. Most OLPFs are designed as optical elements with a frequency response that does not change even if the frequency responses of the other elements of the capturing systems are altered. An extended evaluation of currently used birefringent-based OLPFs is provided. In this work, the author proposed and demonstrated the use of a parallel optical window p ositioned between a lens and a sensor as an OLPF. Controlled X- and Y-axes rotations of the optical window during the image exposure results in a manipulation of the system's point-spread function (PSF). Consequently, changing the PSF affects some portions of the frequency components contained in the image formed on the sensor. The system frequency response is evaluated when various window functions are used to shape the lens' PSF, such as rectangle, triangle, Tukey, Gaussian, Blackman-Harris etc. In addition to the ability to change the PSF, this work demonstrated that the PSF can be manipulated dynamically, which allowed us to modify the PSF to counteract any alteration of other optical elements of the capturing system. There are several instances presented in the dissertation in which it is desirable to change the characteristics of an OLPF in a controlled way., In these instances, an OLPF whose characteristics can be altered dynamically results in an improvement of the image quality.
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Date Issued: 2010
PURL: http://purl.flvc.org/FAU/1927613
Subject Headings: Image processing, Digital techniques, Signal processing, Digital techniques, Frequency response (Dynamics), Polymers and polymerization, Optical wave guides
Format: Document (PDF)

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

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

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

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

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