The loudspeaker system and the room interface are the two main components in any listening environment. Research will be conducted focusing on the room component using Control Room 'A' of the McGill University Recording Studio in an attempt to optimize the monitoring situation. The sound field of the room will be broken down and analyzed in both time and frequency domains. The problem areas of the room will be identified and the surfaces altered by means of absorption, reflection and diffusion.

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.

Audio signal processing systems for converting two-channel (stereo) recordings to four or five channels are increasingly relevant. These audio upmixers can be used with conventional stereo sound recordings and reproduced with multichannel home theatre or automotive loudspeaker audio systems to create a more engaging and natural-sounding listening experience. This dissertation discusses existing approaches to audio upmixing for recordings of musical performances and presents specific design criteria for a system to enhance spatial sound quality. A new upmixing system is proposed and evaluated according to these criteria and a theoretical model for its behavior is validated using empirical measurements. / The new system removes short-term correlated components from two electronic audio signals using a pair of adaptive filters, updated according to a frequency domain implementation of the normalized-least-means-square algorithm. The major difference of the new system with all extant audio upmixers is that unsupervised time-alignment of the input signals (typically, by up to +/-10 ms) as a function of frequency (typically, using a 1024-band equalizer) is accomplished due to the non-minimum phase adaptive filter. Two new signals are created from the weighted difference of the inputs, and are then radiated with two loudspeakers behind the listener. According to the consensus in the literature on the effect of interaural correlation on auditory image formation, the self-orthogonalizing properties of the algorithm ensure minimal distortion of the frontal source imagery and natural-sounding, enveloping reverberance (ambiance) imagery. / Performance evaluation of the new upmix system was accomplished in two ways: Firstly, using empirical electroacoustic measurements which validate a theoretical model of the system; and secondly, with formal listening tests which investigated auditory spatial imagery with a graphical mapping tool and a preference experiment. Both electroacoustic and subjective methods investigated system performance with a variety of test stimuli for solo musical performances reproduced using a loudspeaker in an orchestral concert-hall and recorded using different microphone techniques. / The objective and subjective evaluations combined with a comparative study with two commercial systems demonstrate that the proposed system provides a new, computationally practical, high sound quality solution to upmixing.

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.

Thesis (MA--Communication Studies) -- Auckland University of Technology, 2005. / Chapter 5 not included in e-thesis. Also held in print (viii, 177 leaves, 30 cm.) in Wellesley Theses Collection. (T 338.4778149 HIL)

Thesis (M.A.)--Carleton University, 2003. / Includes bibliographical references (p. 244-254). Also available in electronic format on the Internet.

Thesis (Ed.D.)--Teachers College, Columbia University, 1970. / Typescript; issued also on microfilm. Sponsor: Gladys Tipton. Dissertation Committee: Charles W. Walton. Includes bibliographical references (leaves 219-227).

Thesis (Ph.D.)--Memorial University of Newfoundland, 2001. / Bibliography: leaves 181-208.