This thesis aims to produce a similarity metric for tracks of the genre: Electronic Dance Music, by taking a high-dimensional data representation of each track and then project it to a low-dimensional embedded space (2D and 3D) by applying two Dimensionality Reduction (DR) techniques called t-distributed stochastic neighbor embedding (t-SNE) and Pairwise Controlled Manifold Approximation (PaCMAP). A content-based approach is taken to identify similarity, which is defined as the distances between points in the embedded space. This work strives to explore the connection between the extractable content and the feel of a track. Features are extracted from every track over a 30 second window with Digital Signal Processing tools. Three evaluation methods were conducted with the purpose of establishing ground truth in the data. The first evaluation method established expected similarity sub clusters and tuned the DR techniques until the expected clusters appeared in the visualisations of the embedded space. The second evaluation method attempted to generate new tracks with a controlled level of separation by applying various distortion techniques with increasing magnitude to copies of a track. The third evaluation method introduces a data set with annotated scores on valence and arousal values of music snippets which was used to train estimators that was used to estimate the feeling of tracks and to perform classification. Lastly, a similarity metric was computed based on distances in the embedded space. Findings suggest that certain contextual groups such as remixes and tracks by the same artist, can be identified with this metric and that tracks with small distortions (similar tracks) are located more closely in the embedded space than tracks with large distortions.

<p>Fundamentally the amplitude, sign, and frequency of a magnetic signal are inherently linked to curvature. This thesis employs curvature analysis as a semi-automated tool for source signal extraction from a magnetic field surface represented by a grid. The first step is to compute the full, profile, and plan curvatures from the magnetic grid. These values are used in two approaches to curvature analysis: statistical and lineament. The descriptive statistics mean, standard deviation, kurtosis, and skew are computed for quantitative analysis. Mean is used in conjunction with kurtosis and skew to assess frequency content of the signal, magnetization and source dip. Standard deviation characterizes low, moderate, and extreme curvatures. A rapid technique to statistical analysis is applied using a graphical approach with histograms and scatterplots. Histograms display frequency distribution and scatterplots display the relationship between different curvatures. Curvature in the maximum dip direction is used to systematically identify surficial lineaments characterized as continuous troughs or ridges. These lineaments may represent geological sources or remanent acquisition artefacts. Lineaments representing faults and dykes are used in conjunction with <em>a priori</em> knowledge to determine mineralization vectors since many ore deposits are structurally controlled. Quality control of the aeromagnetic grid levelling application may be assessed using spatial correlation of flight lines and magnetic lineaments. In this work curvature analysis is applied to simple synthetic models and two Canadian aeromagnetic data sets. Curvature analysis was applied to magnetic data from the Wopmay Orogen to identify bedrock contacts, fault configurations, and dyke swarms. The data was also used to show lineaments displayed as rose diagrams may be used as an alternative to standard Fourier power spectrums for assessment of levelling. Magnetic survey data from Southern Ontario was used to show a statistical approach to identify regional dip, dominant magnetization, and interference in anomalies.</p> / Doctor of Philosophy (PhD)

Presented here is an examination of the issues surrounding the analysis of lung sounds and their display. The project is aimed at providing a visual representation of the information that a physician gleans from auscultation of the lungs. Such a tool would be of benefit to those who are hearing impaired and also in teaching auscultation. A second goal is to provide a tool that will allow the examination and quantification of lung sounds thus permitting linkage between the acoustic events and their physical causes. The project is divided into two tasks. The first is the isolation of the wheezes and crackles; the second is their display. The isolation problem is difficult due to the variance in the frequency characteristics of the sounds; wheezes may appear anywhere in a two thousand hertz band and crackles also display a varying spectrum. The difficulty in separation is further compounded by the spectral overlap of the two. These problems preclude any 'simple' filter solution. In order to separate the sounds, filtering methods based on exploiting the statistical differences namely the stationarity of the wheeze and non-stationarity of the crackle are utilized. Of the several methods attempted, the most promising was the Adaptive Line Enhancement process when driven by the Least Mean Squares adaptive algorithm. An important criteria for being able to display the sounds was to access their temporal information. Accomplishing this with the standard short time Fourier transform precludes adequate resolution to identify the frequency characteristics of crackles. Display of the crackle information was facilitated by the use of high resolution time-frequency methods based on Cohen's Class of time-frequency representations. These methods are able to simultaneously provide high time and frequency resolution. A method for automatic adjustment of the parameters involved in the process was developed in order to yield the best display possible. / Thesis / Master of Engineering (ME)

A procedure is presented that is capable of extracting various speaker features, and is of particular value for analyzing records containing single words and shorter segments of speech. By taking advantage of the fast convergence properties of adaptive filtering, the approach is capable of modeling the nonstationarities due to both the vocal tract and vocal cord dynamics. Specifically, the procedure extracts the vocal tract estimate from within the closed glottis interval and uses it to obtain a time-domain glottal signal. This procedure is quite simple, requires minimal manual intervention (in cases of inadequate pitch detection), and is particularly unique because it derives both the vocal tract and glottal signal estimates directly from the time-varying filter coefficients rather than from the prediction error signal. Using this procedure, several glottal signals are derived from human and synthesized speech and are analyzed to demonstrate the glottal waveform modeling performance and kind of glottal characteristics obtained therewith. Finally, the procedure is evaluated using automatic speaker identity verification. / Ph. D.

Multipath on far-field antenna ranges causes distortion of antenna pattern measurements: The multi path components have differing path lengths and hence can be separated by illuminating the antenna under test with short-duration pulses. Alternatively, antenna measurements can be made in the frequency domain, and the Fourier transform can be used to relate the frequency-domain measurements to the antenna's time-domain response. The interference then can be removed with a time-domain gate, and transformed back into the frequency domain to yield improved CW antenna patterns. Virginia Tech has recently completed a major upgrade of their far-field antenna range and implemented a system to perform this data collection and data processing. This thesis describes the principles and implementation of the time-domain processing part of the system. Further, it demonstrates the validity of the method by showing the improvements in pattern measurement that have been achieved with the new system. / Master of Science

In this dissertation novel methods are proposed for communicating in interference limited environments as well as detecting such interference. The methods include introducing redundancies into multicarrier signals to make them more robust, applying a novel filtering structure for mitigating radar interference to orthogonal frequency division multiplexing (OFDM) signals and for exploiting the cyclostationary nature of signals to whiten the spectrum in blind signal detection. Data symbols are repeated in both time and frequency across orthogonal frequency division multiplexing (OFDM) symbols, creating a cyclostationary nature in the signal. A Frequency Shift (FRESH) filter can then be applied to the cyclostationary signal, which is the optimal filter and is able to reject interference much better than a time-invariant filter such as the Wiener filter. A novel time-varying FRESH filter (TV-FRESH) filter is developed and its Minimum Mean Squared Error (MMSE) filter weights are found. The repetition of data symbols and their optimal combining with the TV-FRESH filter creates an effect of improving the Bit Error Rate (BER) at the receiver, similar to an error correcting code. The important distinction for the paramorphic method is that it is designed to operate within cyclostationary interference, and simulation results show that the symbol repetition can outperform other error correcting codes. Simulated annealing is used to optimize the signaling parameters, and results show that a balance between the symbol repetition and error correcting codes produces a better BER for the same spectral efficiency than what either method could have achieved alone. The TV-FRESH filter is applied to a pulsed chirp radar signal, demonstrating a new tool to use in radar and OFDM co-existence. The TV-FRESH filter applies a set of filter weights in a periodically time-varying fashion. The traditional FRESH filter is periodically time-varying due to the periodicities of the frequency shifters, but applies time-invariant filters after optimally combine any spectral redundancies in the signal. The time segmentation of the TV-FRESH filter allows spectral redundancies of the radar signal to be exploited across time due to its deterministic nature. The TV-FRESH filter improves the rejection of the radar signal as compared to the traditional FRESH filter under the simulation scenarios, improving the SINR and BER at the output of the filter. The improvement in performance comes at the cost of additional filtering complexity. A time-varying whitening filter is applied to blindly detect interference which overlaps with the desired signal in frequency. Where a time-invariant whitening filter shapes the output spectrum based on the power levels, the proposed time-varying whitener whitens the output spectrum based on the spectral redundancy in the desired signal. This allows signals which do not share the same cyclostationary properties to pass through the filter, improving the sensitivity of the algorithm and producing higher detection rates for the same probability of false alarm as compared to the time-invariant whitener. / Ph. D. / This dissertation proposes novel methods for building robust wireless communication links which can be used to improve their reliability and resilience while under interference. Wireless interference comes from many sources, including other wireless transmitters in the area or devices which emit electromagnetic waves such as microwaves. Interference reduces the quality of a wireless link and depending on the type and severity may make it impossible to reliably receive information. The contributions are both for communicating under interference and being able to detect interference. A novel method for increasing the redundancy in a wireless link is proposed which improves the resiliency of a wireless link. By transmitting additional copies of the desired information the wireless receiver is able to better estimate the original transmitted signal. The digital receiver structure is proposed to optimally combine the redundant information, and simulation results are used to show its improvement over other analogous methods. The second contribution applies a novel digital filter for mitigating interference from a radar signal to an Orthogonal Frequency Division Multiplexing (OFDM) signal, similar to the one which is being used in Long Term Evolution (LTE) mobile phones. Simulation results show that the proposed method out performs other digital filters at the most of additional complexity. The third contribution applies a digital filter and trains it such that the output of the filter can be used to detect the presence of interference. An algorithm which detects interference can tip off an appropriate response, and as such is important to reliable wireless communications. Simulation results are used to show that the proposed method produces a higher probability of detection while reducing the false alarm rate as compared to a similar digital filter trained to produce the same effect.

The Global System for Mobile (GSM) cellular standard, having been in existence for over two decades, is the most widely deployed wireless technology in the world. While third generation networks and beyond, such as Universal Mobile Telecommunications System (UMTS) and Long Term Evolution (LTE), are undergoing extraordinary growth and driving a large share of current cellular development, technologies and deployments based on GSM are still dominant on a global scale and, like more recent standards, continue to evolve very rapidly. The software-defined radio (SDR) base station is one technology that is driving rapid change in cellular infrastructure. While commercial vendors have now embraced SDR, there is another movement that has recently gained prominence. That movement is the convergence of open source software and hardware with cellular implementation. OpenBTS, a deployable implementation of the GSM radio air interface, and the Universal Software Radio Peripheral (USRP), a RF hardware platform, are two primary examples of such open source software and hardware products. OpenBTS and the USRP underlie three GSM features that are implemented and presented in this thesis. This thesis describes the extension of the OpenBTS software-defined radio transceiver in the three critical areas of user capacity, transmit signal integrity, and the embedded small form factor. First, an optimized wideband multicarrier implementation is presented that substantially increases the capacity beyond that of a single carrier system. Second, the GSM modulator is examined in depth and extended to provide performance that exceeds standards compliance by a significant margin. Third, operation of the GSM transceiver on an E100 embedded platform with ARM and fixed point DSP processors will be explored, optimized, and tested. / Master of Science

In cellular and personal communications services (PCS) systems based on code division multiple access (CDMA), a pilot signal is used on the forward link for synchronization, coherent detection, soft handoff, maintaining orthogonality between base stations, and, in the future, position location. It is critical that the percentage of power allocated to the pilot signal transmitted by each base station be fixed properly to ensure the ability of the CDMA network to support subscriber demand. This thesis reports on the design and implementation of a prototype receiver for measuring pilot signals in CDMA PCS systems. Since the pseudonoise (PN) signal of the pilot channel is a priori information, the receiver searches for pilot signals by digitally correlating the received signal with this known, locally generated pilot signal. By systematically changing the phase of this locally generated pilot signal, the receiver scans the received signal to identify all possible signs of pilot signal activity. Large values of correlation indicate the presence of a pilot signal at the particular phase of the locally generated pilot signal. The receiver can also detect multipath components of the pilot signal transmitted from a given base station. One issue associated with this receiver is its ability to keep the signal power within the dynamic range of the analog-to-digital (A/D) converter at its input. This necessitated the design of an automatic gain control (AGC) mechanism, which is digitally implemented in this receiver. Simulation studies were undertaken to assist in the design and implementation of the pilot signal scanning receiver. These simulations were used to quantify how various non-idealities related to the radio frequency (RF) front-end and A/D converter adversely affect the ability of the digital signal processing algorithms to detect and measure pilot signals. Because the period of the pilot signal is relatively long, methods were developed to keep the receiver's update period as small as possible without compromising its detection ability. Furthermore, the high sampling rate required strains the ability of the digital logic to produce outputs at a rate commensurate with real-time operation. This thesis presents techniques that allow the pilot signal scanning receiver to achieve real-time operation. These techniques involve the judicious use of partial correlations and windowing the received signal to decrease the transfer rate from the A/D converter to the digital signal processor. This thesis provides a comprehensive discussion of these and other issues associated with the actual hardware implementation of the pilot signal scanning receiver. / Master of Science

Το αντικείμενο της διδακτορικής διατριβής αφορά στην ανάπτυξη μιας ευέλικτης και επεκτάσιμης αρχιτεκτονικής που θα αξιοποιηθεί στον σχεδιασμό συστημάτων για την διεξαγωγή πειραμάτων από απόσταση. Τα συστήματα αυτά αναφέρονται ως RLs (Remote Laboratories) και επιτρέπουν στους χρήστες να χειρίζονται απομακρυσμένα τον διαθέσιμο εργαστηριακό εξοπλισμό με σκοπό την διεξαγωγή πειραμάτων. Στην διεθνή βιβλιογραφία έχουν καταγραφεί σημαντικές ερευνητικές προσπάθειες που σχετίζονται με την ανάπτυξη RLs σε διάφορα γνωστικά αντικείμενα. Όμως ακόμη και σήμερα δεν έχει υιοθετηθεί από την επιστημονική κοινότητα κάποια κοινά αποδεκτή αρχιτεκτονική για την ανάπτυξη RLs. Αρχικά προτείνεται μια αρχιτεκτονική για την ανάπτυξη RLs η οποία ονομάζεται ARIAL (Architecture of Internet Accessible Laboratories) η οποία είναι ανεξάρτητη από το γνωστικό αντικείμενο των υποστηριζόμενων από απόσταση πειραμάτων. Η συγκεκριμένη αρχιτεκτονική είναι επίσης ανεξάρτητη τόσο από το υλικό (hardware) όσο και από το λογισμικό (software) που θα αξιοποιηθεί για την ανάπτυξη ενός RL. Η ARIAL αποτελείται από δύο δομικά στοιχεία, τον MWS (Main Web Server) και το WS (WorkStation). Ο MWS αναλαμβάνει κυρίως την διαχείριση των χρηστών και των διαθέσιμων WSs. Ενώ τα WSs που συνήθως βρίσκονται σε πολλαπλότητα, αναλαμβάνουν αποκλειστικά την διεξαγωγή των υποστηριζόμενων από απόσταση πειραμάτων. Η επικοινωνία μεταξύ του MWS και των WSs επιτυγχάνεται μέσω μιας βάσης δεδομένων που επιτρέπει την πρόσβαση μέσω διαδικτύου. Επομένως, τα WSs μπορούν να εγκατασταθούν σε οποιαδήποτε γεωγραφική τοποθεσία επιτρέποντας την ανάπτυξη ομοσπονδιακών RLs. Όμως το σημαντικότερο χαρακτηριστικό της προτεινόμενης αρχιτεκτονικής το οποίο συμβάλει αποφασιστικά στην βιωσιμότητα ενός RL, είναι η υποστήριξη από απόσταση πειραμάτων που έχουν σχεδιαστεί και υλοποιηθεί από τους χρήστες. Με στόχο την επιβεβαίωση της ARIAL προτείνεται ένα RL στην ψηφιακή επεξεργασία σήματος με DSPs που ονομάζεται R-DSP Lab (Remote Digital Signal Processors Laboratory). Το R-DSP Lab παρέχει στους χρήστες την δυνατότητα είτε να διεξάγουν ένα από τα προκαθορισμένα από απόσταση πειράματα είτε να επιβεβαιώσουν την ορθή λειτουργία μιας DSP εφαρμογής που ανέπτυξαν οι ίδιοι. Το συγκεκριμένο RL επιτρέπει επίσης την ανάπτυξη από απόσταση πειραμάτων από τους χρήστες. Στην περίπτωση αυτή οι χρήστες εκτός από την DSP εφαρμογή που επιθυμούν, θα πρέπει να υλοποιήσουν και το GUI (Graphical User Interface) που αναλαμβάνει τον απομακρυσμένο έλεγχο της παραπάνω DSP εφαρμογής. Κατά την διεξαγωγή οποιουδήποτε από τα παραπάνω απόσταση πειράματα οι χρήστες μέσω μιας κατάλληλα σχεδιασμένης ιστοσελίδας έχουν την δυνατότητα να ελέγχουν απομακρυσμένα τα διαθέσιμα εργαστηριακά όργανα. Στην συνέχεια προτείνεται ένα RL στην ψηφιακή επεξεργασία εικόνας με DSPs που ονομάζεται R-DImPr Lab (Remote Digital Image Processing Laboratory). Το συγκεκριμένο RL επιτρέπει την επιβεβαίωση μιας DSP εφαρμογής που αναπτύχθηκε από τον χρήστη αξιοποιώντας το API (Application Program Interface) του R-DImPr Lab. Η DSP εφαρμογή αναλαμβάνει την ψηφιακή επεξεργασία εικόνων που λαμβάνονται από τον διαθέσιμο αισθητήρα εικόνας. Κατά την διεξαγωγή του από απόσταση πειράματος ο χρήστης μέσω της ιστοσελίδας του RL αφού επιλέξει τις ρυθμίσεις του αισθητήρα εικόνας, έχει την δυνατότητα να παρατηρήσει τόσο στην αρχική όσο και στην επεξεργασμένη εικόνα. Με σκοπό την διεύρυνση των δυνατοτήτων του R-DimPr Lab σχεδιάστηκε και αναπτύχθηκε ένα σύστημα επεξεργασίας εικόνας με DSPs το οποίο παρέχει στους χρήστες την δυνατότητα να διεξάγουν από απόσταση πειράματα ελέγχοντας απομακρυσμένα, τόσο την λειτουργία της αντίστοιχης DSP εφαρμογής όσο και την θέση του αισθητήρα εικόνας. Ο έλεγχος της θέσης του αισθητήρα εικόνας επιτυγχάνεται μέσω ενός μηχανισμού κίνησης που βασίζεται σε δύο βηματικούς κινητήρες και επιτρέπει την περιστροφή του αισθητήρα εικόνας σε δύο άξονες. Επιπρόσθετα, διερευνείται η δυνατότητα ανάπτυξης από απόσταση πειραμάτων στην ψηφιακή επεξεργασία εικόνας με DSPs από τους χρήστες αξιοποιώντας το R-DSP Lab. Τέλος, προτείνεται ένα RL στην αρχιτεκτονική των υπολογιστών που επιτρέπει στους χρήστες να προγραμματίσουν σε assembly μια από τις δύο διαθέσιμες CPUs (Central Processing Units). Κατά την διαδικασία επιβεβαίωσης, αρχικά φορτώνεται στο FPGA (Field Programmable Gate Array) της διαθέσιμης αναπτυξιακής πλατφόρμας η υλοποίηση του συστήματος που βασίζεται στην επιλεγμένη CPU. Στην συνέχεια μέσω του GUI της ιστοσελίδας του προτεινόμενου RL, οι χρήστες έχουν την δυνατότητα να παρατηρήσουν βήμα προς βήμα τις μικρο-λειτουργίες που λαμβάνουν χώρα στην επιλεγμένη CPU κατά την εκτέλεση του προγράμματος. / The subject of this Ph.D. dissertation deals with the development of a flexible and expandable architecture which will be exploited in the design of systems for the conduction of remote experiments. These systems are referred as RLs (Remote Laboratories) and allow the users to handle remotely the available laboratory equipment in order to perform remote experiments. Significant scientific efforts which deal with the development of RLs in several cognitive fields, have been documented in the international literature. However, even today a commonly accepted architecture for the development of RLs has not been adopted by the scientific community. At the beginning, an architecture for the development of RLs which is called ARIAL (ARchitecture of Internet Accessible Laboratories) and is independent of the cognitive field of the supported remote experiments, is proposed. This architecture is also independent of both the hardware and the software which will be utilized for the development of the corresponding RL. The ARIAL consists of two structural elements, the MWS (Main Web Server) and the WS (WorkStation). The MWS undertakes the management of the users and the available WSs. Each one of the multiple WSs is exclusively responsible for the conduction of the supported remote experiments. The communication between the MWS and the WSs is achieved through an internet accessible database. Therefore, the WSs can be installed in any geographic location allowing the development of federal RLs. However, the most important feature of the proposed architecture which contributes decisively to the sustainability of a RL, is the support of remote experiments designed and implemented by the users. In order to confirm the ARIAL, this Ph.D. dissertation also proposes a RL in digital signal processing with DSPs which is called R-DSP Lab (Remote Digital Signal Processors Laboratory). The R-DSP Lab provides the users with the ability either to perform one of the predefined remote experiments or to confirm the operation of a DSP application which is developed by them. In addition, the proposed RL allows the development of remote experiments by the users. In this case, the users implement offline both the desired DSP application and the GUI (Graphical User Interface) which undertakes the remote control of the above DSP application. During the conduction of the above remote experiments, the users are able to remote control the available laboratory instruments through a carefully designed web page. Subsequently, a RL in digital image processing with DSPs which is called R-DImPr Lab (Remote Digital Image Processing Laboratory), is also proposed. This RL allows the verification of a DSP application developed by the user utilizing the API (Application Program Interface) of R-DImPr Lab. The DSP application undertakes the digital process of images which are captured by the available image sensor. During the conduction of the remote experiment, the user through the web page of the proposed RL, selects the parameters of the image sensor and observes both the original and the processed image. In order to expand the features of the R-DImPr Lab, a digital image processing system based on DSPs was designed and developed. This system allows the users to perform remote experiments by controlling remotely both the DSP application and the position of the image sensor. The control of the image sensor’s position is achieved through a motion actuator which is based on two stepper motors and allows the rotation of the image sensor in two axes. In addition, this Ph.D. dissertation explores the possibility of the development of remote experiments in digital image processing with DSPs by the users utilizing the features of the R-DSP Lab. Finally, a RL in computer architecture which allows the users to program in assembly language one of the two available CPUs (Central Processing Units), is proposed. During the verification process, the implementation of the system which is based on the selected CPU, is loaded into the FPGA (Field Programmable Gate Array) of the available development platform. The users through the GUI of the proposed RL’s web page, are able to observe the micro-operations which take place in the selected CPU during the step by step program execution.

The discrete Fourier transform has become an essential tool in the analysis of digital signals. Applications have become widespread since the discovery of the Fast Fourier Transform and the rise of personal computers. The field of digital signal processing is an exciting intersection of mathematics, statistics, and electrical engineering. In this study we aim to gain understanding of the mathematics behind algorithms that can extract chord information from recorded music. We investigate basic music theory, introduce and derive the discrete Fourier transform, and apply Fourier analysis to audio files to extract spectral data.