The objective of this dissertation research is to identify sounds that grab a listener's attention. These sounds that draw a person's attention are sounds that are considered salient. The focus here will be on investigating the role of saliency in the auditory attentional process. In order to identify these salient sounds, we have developed a computational auditory saliency model inspired by our understanding of the human auditory system and auditory perception. By identifying salient sounds we can obtain a better understanding of how sounds are processed by the auditory system, and in particular, the key features contributing to sound salience. Additionally, studying the salience of different auditory stimuli can lead to improvements in the performance of current computational models in several different areas, by making use of the information obtained about what stands out perceptually to observers in a particular scene. Auditory saliency also helps to rapidly sort the information present in a complex auditory scene. Since our resources are finite, not all information can be processed equally. We must, therefore, be able to quickly determine the importance of different objects in a scene. Additionally, an immediate response or decision may be required. In order to respond, the observer needs to know the key elements of the scene. The issue of saliency is closely related to many different areas, including scene analysis. The thesis provides a comprehensive look at auditory saliency. It explores the advantages and limitations of using auditory saliency models through different experiments and presents a general computational auditory saliency model that can be used for various applications.

Το σύστημα που σχεδιάστηκε βασίζεται στον soft-core επεξεργαστή Nios II της ALTERA. Εκτελεί βασικές ρουτίνες επεξεργασίας ήχου όπως echo, reverberation, volume leveling, fir filtering και ο έλεγχος του επιτυγχάνεται ασύρματα, μέσω ενός πληκτρολογίου υπερύθρων. Αναλύονται όλες οι βασικές έννοιες, και περιγράφεται τόσο ο επεξεργαστής όσο και οι περιφερειακές μονάδες που χρησιμοποιήθηκαν (VGA, LCD, PIO κτλ.). Το όλο σύστημα υλοποιήθηκε στο DE2 Board της ALTERA. / The system that has been developed, is based on the ALTERA‘s soft-core processor Nios II. It implements basic sound-processing routines such as echo, reverberation, volume leveling, fir filtering and the control of the system is achieved via an infra red keyboard (wireless). All fundamental concepts are analyzed, and both the processor and the peripherals used (VGA, LCD, PIO etc) are described in detail. The system was implemented on the ALTERAs DE2 Board.

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No description available.

The sustained increase in computing performance over the last decades has brought enough computing power to perform significant audio processing in affordable personal computers. Following this revolution, we have witnessed a series of improvements in sound transformation techniques and the introduction of numerous digital audio effects to modify effectively the time, pitch, and loudness dimensions of audio signals. Due to the complex and multi-dimensional nature of timbre however, it is significantly more difficult to achieve meaningful and convincing qualitative transformations. The tools currently available for timbre modifications (e.g. equalizers) do not operate along perceptually meaningful axes of singing voice timbre (e.g. breathiness, roughness, etc.) resulting in a transformation control problem. One of the goals of this work is to examine more intuitive procedures to achieve high-fidelity qualitative transformations explicitly controlling certain dimensions of singing voice timbre. Quantitative measurements (i.e. voice timbre descriptors) are introduced and used as high-level controls in an adaptive processing system dependent on the characteristics observed in the input signal. / The transformation methods use a harmonic plus noise representation from which voice timbre descriptors are derived. This higher-level representation, closer to our perception of voice timbre, offers more intuitive controls over timbre transformations. The topics of parametric voice modeling and timbre descriptor computation are first introduced, followed by a study of the acoustical impacts of voice breathiness variations. A timbre transformation system operating specifically on the singing voice quality is then introduced with accompanying software implementations, including an example digital audio effect for the control and modification of the breathiness quality on normal voices.

This thesis describes the construction, form, purpose and motivation for LogoRhythms, a sound synthesis and computer audition API intended to be used as a tool in the teaching of computer programming, computer science and associated skills. LogoRhythms is built into Berkeley Logo (UCB Logo), a contemporary open source Logo interpretter. In addition to serving as a user manual complete with program description and code examples, this work documents an exercise in experimental archaeology that traces the unfortunate shift in educational computing and personal computing in general from an emphasis of 'computer literacy' to one of 'user-friendly.' Arguments in defense of command-line and text based computing parallel those for computing as a tool for creative expression and are made in three ways: historical analysis, a new user-study and philosophical investigation. Programming is a widely learnable skill and debugging a useful skill transcending a utility limited to computer programs. Digital musical composition provides a perpetually renewable Opportunity for custom software, underscoring that programming is a creative endeavor.

This thesis describes the design and use of two novel digital musical instruments (DMIs) based on moving mechanical systems. The motivation behind using mechanical devices was threefold: to explore the effect of physical effort on DMIs, to make use of the device's inherent haptic and visual feedback, and to serve as a starting point for sound mappings. It was hoped that their mechanical nature would give the instruments a character that could emerge through each of the mappings. The first DMI built was the Gyrotyre, a hand-held DMI based around a small bicycle wheel outfitted with sensors that measure its speed of rotation and as well as its angle of orientation. The second DMI built was the Springwave, which consists of a loose metal spring stretched to one meter and fixed at both ends to a metal frame. The frame is in turn mounted horizontally on a hi-hat stand so that it can be raised and lowered with the pedal, thus inducing oscillation in the spring. Various mappings were designed to reflect and make use of the physical nature of both instruments. It was found that the nature of interaction with each instrument was very different depending on the mapping used. The use of mechanical devices was found to be a useful starting point for the development of mappings, and made playing the instruments engaging for the performer by the relationship between DMIs and musical contexts, a framework for characterizing DMIs that takes musical context into account is presented.

Thesis (Ph.D.)--University of Wollongong, 2006. / Typescript. Includes bibliographical references: leaf 297-321.

Thesis (Ph.D.)--Victoria University of Wellington, 2008. / Includes bibliographical references.

Thesis (M.C.A.-Res.)--University of Wollongong, 2005. / Typescript. Includes CD-ROM in back pocket. Includes bibliographical references: leaf 66-75.