M.Ing. (Electrical and Electronic Engineering) / This dissertation presents a comprehensive electrical model of the human auditory periphery. The model focuses on the generation and transmission of otoacoustic emissions (OAEs) under biometric conditions. The auditory system model was divided and studied in three sections, namely the outer, middle and inner ear sections. Existing models were used and improved for the study. The outer ear model was derived using electroacoustic analogies. The middle ear model was derived empirically. The inner ear model was derived by relating the mechanical properties of the inner ear to electrical principles. The outer ear model includes an analog diffraction circuit and a linear transmission line representation of the auditory canal and the concha. The variation of the radius of the auditory canal along its length was incorporated when computing the model of the outer ear. A pair of second order polynomials were used to create a new radius-length function which approximates the relationship between the radius of the auditory canal and its length. The frequency response of the outer ear model obtained using the radius-length function gave a wide frequency range representation of the outer ear characteristics. The middle ear is modelled using an analog network. Only the linear operation region of the middle ear was considered, thus excluding its reflex nonlinear mechanisms, namely; the stapedius muscle action and the stapes clipping displacement. The influence of the middle ear on the transmission of OAEs was evaluated by considering both the forward and reverse transmission characteristics/path of the middle ear. The middle ear response demonstrated great sensitivity to changes in the terminal loads connected to the middle ear as well as the transformer ratio. The inner ear behavior is represented by means of a nonlinear transmission line model. The nonlinear mechanism of the outer hair cells, which are taken as the primary sources of OAEs, are modelled using nonlinear voltage sources. The inner ear model was evaluated for conditions of both the active and inactive outer hair cells voltage sources. Due to limitations in the simulation software, a reduced active inner ear model was computed.The influence of the number of segments of the inner ear was explored. A reduced inner ear model having 40 segments was found to be sufficient in representing the frequency characteristics of the inner ear, whilst preserving the frequency-latency relationship of OAEs. The study not only improved the model of the auditory periphery, but also suggested several factors that can be incorporated in future research in order to better design signal acquisition and processing methods for OAE biometric applications.

Female mallard ducks (Anas platyrhynchos) exhibit diverse vocalizations. Duck hunters mimic these vocalizations using artificial calls made from hardwoods or plastics. Hardness of these calls and extent to which humans can mimic live mallards using an artificial call were unknown before this study. I compared hardness of 7 species of hardwoods and cast acrylic and found acrylic, cocobolo (Dalbergia retusa), bocote (Cordia alliodora), osage orange (Maclura pomifera), and pecan (Carya sp.) were the hardest materials tested. I also compared acoustic metrics of field recordings of vocalizing female mallards to those of experienced duck callers using calls of these materials equipped with single or double reeds. I found that cocobolo, osage orange, pecan, acrylic, and bocote calls with double reeds were acoustically most similar to female mallards. I recommend that duck call manufacturers use acrylics and harder wood species with single or double reeds, recognizing that double reed calls generally performed superior in this study.

No description available.

Measuring different acoustical properties have been the key in reducing noise and improving the sound quality from various sources. In this report, a broadband (200 Hz – 6.5 kHz) three-dimensional seven-microphone intensity probe system is developed to measure the sound intensity, and total energy density in different acoustical environments. Limitations of most commercial intensity probes in measuring the three-dimensional intensity for a broadband sound field was the main motivation in developing this probe. The finite-difference error and the phase mismatch error which are the two main errors associated with the intensity measurements are addressed in this report. As for the physical design, seven microphones were arranged in a two-concentric arrays with one microphone located at the center of the probe. The outer array is for low-frequencies (200 Hz – 1.0 kHz), and the inner one is for high-frequencies (1.0 kHz – 6.5 kHz). The screw adjustable center microphone is used for the microphone calibration, and as the reference microphone of the probe. The simultaneous calibrations of all the microphones in the probe were done in the anechoic room. Theories for the intensity and the energy densities calculations for the probe were derived from the existing four-microphone probe configuration. Reflection and diffraction effects on the intensity measurements due to the presence of the microphones, and the supporting structures were also investigated in this report. Directivity patterns of the calculated intensity showed the omnidirectional nature of the probe. The intensity, and total energy density were calculated and compared with the ideal values in the anechoic room environment. Characterization of sound fields in a reverberant enclosed space, and sound source identification are some applications that were investigated using this probe. Results of different measurements showed effectiveness of the probe as a tool to measure key acoustical properties in many practical environments. / text

Thesis (Ph. D.)--State University of New York at Binghamton, Mechanical Engineering Department, 2006. / Includes bibliographical references.

This master thesis examines the effects on the acoustical properties of a concert hall caused by an artificial reverberation system (ARS) and the possibility of simulating these properties. By examining the case of the Stockholm Concert Hall, which recently installed such a system, a greater understanding of the ARS will be gained and additional improvements of simulating such systems will be explored. This study comprises two parts: (1) objective data obtained through acoustical measurements are evaluated both internally and to other halls and (2) by computer simulation of the concert hall and its electronic reverberation system evaluate the acoustics of the hall. The study shows that the effect of the ARS on the acoustical properties of Stockholm Concert Hall is not excessive but noticeable. An 0.3 second increase in reverberation time is a desirable outcome but comes at the cost of clarity, which sees a reduction of 0.7 decibels. Moreover, it is possible to simulate a concert hall, having an ARS installed, with fairly realistic results. However, in order to compile the simulated impulse response, a script had to be created -combining the transfer functions related to each component of the reverberation chain from source to receiver, including all the microphones and loudspeakers of the ARS. / Det här examensarbetet undersöker påverkan på de akustiska egenskaperna hos en konsertlokal orsakad av ett artificiellt efterklangssystem. Likaså undersöks möjligheterna för att simulera dessa akustiska egenskaper. Genom att undersöka Stockholms konserthus, som nyligen installerade ett efterklangssystem, kommer en bättre förståelse för artificiella efterklangssystem skapas och ytterligare förbättringar för simulering kommer att möjliggöras. Den här studien genomförs i två delar: (1) objektiv data, inhämtad från akustiska mätningar, utvärderas både internt och mot andra konsertlokaler samt (2) genom datorsimulering av konsertlokalen och det elektroniska efterklangssystemet utvärderas de akustiska egenskaperna. Studien visar att inverkan på de akustiska egenskaperna hos Stockholms konserthus orsakade av det artificiella efterklangssystemet inte är överdrivna men noterbara. En önskad ökning av efterklangstiden med 0.3 sekunder uppnås men detta på bekostnad av att ljudets klarhet minskar med 0.7 decibel. Vidare är det möjligt att simulera ljudutbredningen i en konsertlokal som har ett efterklangssystem installerat med ett tämligen realistiskt resultat. För att uppnå detta simuleringsresultat skapas ett skript vilket väger samman alla överföringsfunktioner mellan ljudkällan och mottagaren, inklusive de mellan efterklangssystemets mikrofoner och högtalare.

This thesis generates certain sets of analytical and approximate solutions to a new class of partial differential equations stemming from a version of asymmetricstress elasticity theory appropriate for the study and prediction of the behaviour of fibre-reinforced materials containing fibres that resist bending. These new solutions are of theoretical and practical interest in the static and dynamic analysis of thinwalled, linearly elastic fibre-reinforced structures influenced by couple-stress and unsymmetric stress due to fibre bending stiffness. The static and free vibration solutions are constructed considering bending resistance fibres in a small deformation of beams and plates. Numerical results for displacements, stresses, couple-stress and natural frequencies of vibration are provided to investigate the influence of the fibres resistance in bending on the deformed beams and plates.

No description available.

The precise nature of the motion of the lips of the musician is critically important to the sound of the brass wind instrument. The player must match the oscillation of the lips to the acoustical properties of the instrument and it can take many years of practice to master the techniques involved. Visualisation techniques for capturing the motion of the lips during performance are described and the behaviour of the lips quantitatively analysed using digital image analysis. The concept of an artificial mouth for the sounding of brass wind instruments is discussed and the motion of the artificial lips is compared to that of human musicians. When a brass instrument is played loudly the energy of the higher harmonics increases, creating a distinctive ‘brassy’ timbre. It has been suggested that saturation or constraint of the lips of the musician during extremely loud playing is responsible for this change in sound. Measurements of the motion of the lips of a number of different musicians on different instruments suggest that the lips are not significantly constrained at any playing dynamic, and that it is the phenomena of nonlinear propagation and shockwave generation that is responsible for the increase in energy of the higher harmonics. It is widely accepted that the starting transient of a musical instrument is of great importance to both listener and musician. Previous studies of brass instruments have focused on the steady-state behaviour of the lip-instrument interaction. Measurements of the motion of the lips have been synchronised with the pressure in the mouthpiece of the instrument and the sound radiated from the bell in order to investigate the transient behaviour of the system during both the starting transient and slurs between notes. Thiswork has been extended to include measurements of the pressure in the mouth of the player during the starting transient, and this information used to recreate realistic transients using an artificial mouth. The transient behaviour of the system is clearly affected by the time delay between the start of the note and the acoustical feedback from the instrument beginning. The information obtained can be used to aid in the creation of accurate computational and physical models of brass wind instruments.

<div>Near-Field Acoustical Holography (NAH) is an inverse process in which sound pressure measurements made in the near-field of an unknown sound source can be used to reconstruct the sound field so that source locations can be identified. Usually a large number of measurements is required for the usual NAH methods since a large number of parameters in the source or field model need to be determined. However, a large-scale microphone measurement is costly and hard to perform, so the use of NAH is limited by practical experimental conditions. In the present work, with the motivation of decreasing the number of microphone measurements required, and thus facilitating the measurement process, two sparse Equivalent Source Method (ESM) algorithms were studied: i.e., Wideband Acoustical Holography (WBH) and l_1-norm minimization. Based on these two algorithms, a new hybrid NAH procedure was proposed and demonstrated.To study and verify the above mentioned algorithms, simulations of different sources were conducted and then experiments were conducted on different sources: i.e., a loudspeaker cabinet and a diesel engine.</div>