Thesis (D.M.A.)--University of Texas at Austin, 2001. / Vita. Includes bibliographical references. Available also from UMI/Dissertation Abstracts International.

This thesis draws from music technology to create novel sonifications of heart rate information that may be of clinical utility to physicians. Current visually-based methods of analysis involve filtering the data, so that by definition some aspects are illuminated at the expense of others, which are decimated. However, earlier research has demonstrated the suitability of the auditory system for following multiple streams of information. With this in mind, sonification may offer a means to display a potentially unlimited number of signal processing operations simultaneously, allowing correlations among various analytical techniques to be observed. This study proposes a flexible listening environment in which a cardiologist or researcher may adjust the rate of playback and relative levels of several parallel sonifications that represent different processing operations. Each sonification "track" is meant to remain perceptually segregated so that the listener may create an optimal audio mix. A distinction is made between parameters that are suited for illustrating information and parameters that carry less perceptual weight, which are employed as stream separators. The proposed sonification model is assessed with a perception test in which participants are asked to identify four different cardiological conditions by auditory and visual displays. The results show a higher degree of accuracy in the identification of obstructive sleep apnea by the auditory displays than by visual displays. The sonification model is then fine-tuned to reflect unambiguously the oscillatory characteristics of sleep apnea that may not be evident from a visual representation. Since the identification of sleep apnea through the heart rate is a current priority in cardiology, it is thus feasible that sonification could become a valuable component in apnea diagnosis.

The crux of the research problem for this study pertains to the fact that, as the world moves towards a digital age, it is imperative that we gain insight into the current copyright laws which govern the transferring of music from the old analogue form to the new digital formats. In terms of the research problem, this study explores the South African Copyright Act No. 98 of 1978 as it pertains to the transferring of sound recordings from analogue to digital format. The study also examined digital copyright laws for sound recordings in the United States, the United Kingdom and Australia as well as major copyright conventions and treaties as these international copyright laws impact on the South African situation, especially in terms of reciprocity. Furthermore, the study addressed the issue of balancing the rights between copyright holders and the public good in the preservation and dissemination of knowledge in the digital age. The study employed methodological triangulation which included a literature search, a questionnaire and informal interviews. The population constituted 16 music librarians and two legal librarians who were surveyed. Quantitative and qualitative techniques were employed. Considering the size of the population (18) the results of a self-administered questionnaire were analysed using a calculator. Data collected for the informal interviews was analysed qualitatively. The study revealed that music librarians in South Africa are not well-versed in South African copyright law, especially as it applies to sound recordings. Guidelines, based on the South African Copyright Act No. 98 of 1978, and specifically for South African music librarians, have been formulated concerning both print and the actual sound recordings. It is important for the music librarian to take note that the composition of songs in a sound recording has an individual copyright that is separate from the copyright of the sound recording. Further copyrights can also exist in, for example, the sleeve of an album. It is anticipated that the guidelines will give clarity to music librarians on South African digital copyright legislation with regard to sound recordings. / Thesis (M.I.S.)-University of KwaZulu-Natal, Pietermaritzburg, 2009.

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.

The development of computer algorithms for music instrument identification and parameter extraction in digital audio signals is an active research field. A musician can listen to music and instantly identify different instruments and the timbres produced by various playing techniques. Creating software to allow computers to do the same is much more challenging. This thesis will use digital signal processing and machine learning techniques to differentiate snare drum timbres produced by different stroke positions and stroke techniques.

This study had two purposes: first, to collect information relating to harmonic and intermodulation distortion from several sources into one comprehensive report and second, to describe and develop an interactive CD-ROM as a new method of learning harmonic and intermodulation distortion. The CD-ROM allows students to interact and control the percentage amount of harmonic and intermodulation distortion in order to "calibrate" their ears. In addition, the practical testing environment changes on a random basis to continually challenge the users. These situations allow students to interact at their own speed and therefore customize their learning development. / School of Music

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.

We develop a theoretical analysis of generalization performances of regularized least-squares on reproducing kernel Hilbert spaces for supervised learning. We show that the concept of effective dimension of an integral operator plays a central role in the definition of a criterion for the choice of the regularization parameter as a function of the number of samples. In fact, a minimax analysis is performed which shows asymptotic optimality of the above-mentioned criterion.

This thesis investigates the feasibility of using existing IP-based control and monitoring protocols within professional audio installations utilising IEEE 1394 technology. Current control and monitoring technologies are examined, and the characteristics common to all are extracted and compiled into an object model. This model forms the foundation for a set of evaluation criteria against which current and future control and monitoring protocols may be measured. Protocols considered include AV/C, MIDI, QSC-24, and those utilised within the UPnP architecture. As QSC-24 and the UPnP architecture are IP-based, the facilities required to transport IP datagrams over the IEEE 1394 bus are investigated and implemented. Example QSC-24 and UPnP architecture implementations are described, which permit the control and monitoring of audio devices over the IEEE 1394 network using these IP-based technologies. The way forward for the control and monitoring of professional audio devices within installations is considered, and recommendations are provided. / KMBT_363 / Adobe Acrobat 9.54 Paper Capture Plug-in

This thesis investigates the feasibility of using a new digital interconnection technology, the IEEE-1394 High Performance Serial Bus, for audio and control data distribution in local and remote music recording studio environments. Current methods for connecting studio devices are described, and the need for a new digital interconnection technology explained. It is shown how this new interconnection technology and developing protocol standards make provision for multi-channel audio and control data distribution, routing, copyright protection, and device synchronisation. Feasibility is demonstrated by the implementation of a custom hardware and software solution. Remote music studio connectivity is considered, and the emerging standards and technologies for connecting future music studio utilising this new technology are discussed. / Microsoft Word / Adobe Acrobat 9.46 Paper Capture Plug-in