The purpose of this study was to present a comprehensive analysis of the acoustical properties of the trumpet mouthpiece. The first section will be concerned with the individual parts of the mouthpiece and their interrelationship. The second portion of the study will be the physical analysis of three commonly used trumpet mouthpieces. For this study, five examples of each size have been selected. The third section will present a tonal analysis of the selected mouthpieces. These investigative procedures will attempt to recognize any relationships between the individual mouthpieces and the resulting tones.

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The purpose of this work is to set forth the physical characteristics of the generator and resonator of the violin and to determine in what manner and degree they influence the technic of artistically manipulating the instrument.

[by] Adrianus J.M. Houtsma and Julius L. Goldstein. / Based on a Ph.D. thesis in the Dept. of Electrical Engineering, 1971, by A.J.M. Houtsma. / Bibliography: p. 71-75.

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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.

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.

The purpose of this study was to investigate the phenomenon of choral blend through acoustical analysis of individual vocal sounds. One aspect of the study involved identifying and comparing the acoustical qualities of sounds produced in the usual solo manner and sounds produced by the same singers attempting to blend with a unison ensemble. Another aspect of the study involved identifying and comparing the acoustical qualities of vocal sounds identified as blending well and poorly with a unison ensemble. Singers attempting to blend adjust their overall intensity not only to affect the perceived loudness of their tones, but also to facilitate other acoustical changes which are helpful for achieving blend. Vocal blend apparently may be achieved more readily on vowels having few upper partials than on vowels having numerous upper partials. Where vibrato is employed, certain vocal sounds can achieve a good blend even though their fundamental frequencies only approximate the theoretically correct frequency. There apparently is an interaction between the vibrato of a vocal tone and its spectral features, making it advantageous for the singer to adopt mutually beneficial approaches to both factors in order to blend. Vowel modification effective for achieving vocal blend- -at least for sopranos-- appears to consist primarily of changes in the intensity relationships between the vowel formants. Singers' variations in frequency extend beyond the boundaries of any system of tuning or temperament.