The motion of tissues within the human body can be investigated by a number of techniques, including ultrasonic echography. The ultrasonic echo patterns from regions of relatively homogeneous tissue bear an ambiguous relationship to their spatial structure, preventing the use of an lmaging system to study the motion of such tissue. This work describes the development and use of a method of analysing the ultrasonic echoes from tissue to investigate their motion. The properties of the echoes from tissue are studied using a theoretical model, and a number of methods of analysing the echoes are suggested. The theoretical model can also be used to simulate the ultrasonic imaging of such tissues. The apparatus for obtaining and analysing the echoes from tissues is described, with an investigation of the errors involved in the digitisation of analogue signals. The methods of echo analysis proposed for the study of tissue motion are evaluated experimentally, using specimens of fixed liver, and the results show that a correlation method of analysis provides the most consistent measure of tissue motion. This method is applied to the analysis of the in vivo motion of soft tissue. The majority of measurements are taken on the livers of normal subjects, with a simultaneous recording of the electrocardiograph. The amplitude, frequency and phase of the motion are measured and the results correlated with the electrocardiograph. The conclusions from this work are that the stochastic properties of the ultrasonic echoes from soft tissues require the use of statistical methods of analysis to study successfully tissue motion. Estimates are made of the extent to which the phenomenon of tissue motion can be used in the characterisation of the pathological state of tissue.

A microelectromechanical systems (MEMS) optical microphone that measures the interference of light resulting from its passage through a diffraction grating and reflection from a vibrating diaphragm (JASA, v. 122, no. 4, 2007) is described. In the present embodiment, both the diffractive optical element and the sensing diaphragm are micromachined on silicon. Additional system components include a semiconductor laser, photodiodes, and required readout electronics. Advantages of this optical detection technique have been demonstrated with both omni-directional microphones and biologically inspired directional microphones. In efforts to commercialize this technology for hearing-aids and other applications, a goal has been set to achieve a microphone contained in a small surface mount package (occupying 2mm x 2mm x 1mm volume), with ultra-low noise (20 dBA), and broad frequency response (20Hz–20kHz). Such a microphone would be consistent in size with the smallest MEMS microphones available today, but would have noise performance characteristic of professional-audio microphones significantly larger in size and more expensive to produce. This paper will present several unique challenges in our effort to develop the first surface mount packaged optical MEMS microphone. The package must accommodate both optical and acoustical design considerations. Dynamic models used for simulating frequency response and noise spectra of fully packaged microphones are presented and compared with measurements performed on prototypes. / text

No description available.

A digital loudspeaker converts digital signals into quasi-analogue sound pressure without an embedded digital to analogue converter. This work reports the findings for two types of digital loudspeaker; a transducer array and a multiple voice coil loudspeaker. The analogue to digital conversion process produces bit streams that have a much wider frequency range than the original analogue signal. During the digital to analogue re-conversion the amplitudes and phases of the additional harmonics cancel to reproduce the original analogue signal plus a small amount of digitally generated noise. A digital transducer array loudspeaker was proposed in which each transducer carries the same weight as one least significant bit and the number of transducers switched on during each sample interval is equal to the binary representation of the analogue input. Computer simulations show that correct analogue acoustic reconstruction can only occur at one listening position. At all other positions the different propagation paths for sound from each transducer to the listener disturb the relative amplitudes and phases of the additional digital harmonics so that they do not cancel, resulting in significant distortion. Although some methods are predicted to minimise this distortion, it is too large for high fidelity applications. A prototype seven bit multiple voice coil digital loudspeaker has been constructed and evaluated. The acceleration of the diaphragm is directly proportional to the radiated sound pressure, in common with conventional loudspeakers. In order for the force on the diaphragm to correctly mirror the digital input, the current in each coil should correspond to the binary significance of each bit in the digital representation, requiring the use of a switched constant current driver. Moreover, since each coil is wound on the same motor unit, motion of the diaphragm will induce an electro-motive force in each coil, and the close proximity of the coils to each other means that pairs of coils have a high mutually inductive coupling. It was found that the effective input impedance of a coil depends on the binary significance of the current it is carrying. The current in the low significant coils is most likely to be corrupted unless the output impedance of the current driver is very high indeed. It was found that with the current state of transducer technology a crossover is needed for a digital loudspeaker in order to reproduce the entire audio frequency range. Although the natural choice for implementing a crossover is undoubtedly with digital signal processing, it was found that analogue filters placed between the drivers and the transducers could perform the same function. This is a direct consequence of the associativity of the Fourier transform and the digital to analogue operation. The signals fed to each frequency selective loudspeaker driver unit are no longer truly digital, however their sum over the entire audio frequency spectrum does remain digital. The construction of a high resolution digital loudspeaker presents some problems for manufacturing. The use of bit reduction techniques e.g. oversampling and noise shaping may reduce the number of transducers required in an array or the number of coils needed for a multiple voice coil loudspeaker. The latter approach appears to be promising for the implementation of a fully digital loudspeaker system.

The aim of this study is to develop a ray tracing computer model for predicting speech intelligibility in underground stations. There are four parts to the study: correctly modelling the sound field in underground stations; developing a mathematical model for predicting speech intelligibility using a ray tracing computer model; using the model to investigate ways of improving speech intelligibility in underground stations; and using the model to analyze the sound field in long enclosures with multiple source systems. Four computer models have been developed for investigating acoustic parameters in different conditions or different types of space. The models have been validated with scale model measurements, and the predictions have also been compared with classical room acoustics. Three new contributions to the ray tracing computer model have been developed in this project: the reverberation time tail compensation, the exact representation of curved surfaces, and diffraction effects. A mathematical model for predicting speech transmission index in long enclosures using the ray tracing method has been developed. The model has been shown to be more accurate and efficient by comparison with scale model measurements and measurementsm ade in a real underground station. The model has been used to investigate ways of improving speech intelligibility in different noise levels and with different source spacing. Finally, the quasi-diffuse sound field theory for long enclosures with multiple source systems has been developed and justified as an approximation method for a quick investigation of speech intelligibility in underground stations.

No description available.

Typically, the art song recital is a performance event of primary importance within the university environment; however, the career Heldentenor entering academia may find that the aggressiveness of timbre and the scale of loudness he has developed to sing Wagner's heavily orchestrated, very long roles are inappropriate for the performance of most art song repertoire. If the Heldentenor possesses an appropriate pitch structure, he can adapt his technique to sing as a baritone and develop the warmth of timbre and nuance of expression appropriate to the intimate qualities of recital performance. Art song recital performance conditions and those of some Wagnerian tenor roles are described and contrasted, emphasizing the acoustical challenges and demands upon stamina that drive the Heldentenor toward specialization. The acoustics of audibility and voice type are described. A method for adapting the Heldentenor to recital baritone is presented, incorporating exercises developed from the acoustical principles examined within this document. Examples are presented of power spectra and the application of these techniques to repertoire, documenting the efficacy of the author's method in adapting the Heldentenor from the least intimate of repertoires to the most refined.

Experimental investigations of two ultrasonic lenses are presented, a graded acoustic index lens and a cylindrical lens. The first lens is a short rod which has a graded acoustic velocity profile in the radial direction giving it the ability to focus acoustic waves. Electron probe microanalyses, optical refractive index measurements and acoustic velocity measurements are used to characterize the lens. The ray acoustics approach is used for the theoretical interpretation and the focussing behavior is visualized with a Schlieren system. Potential applications of the lens are discussed. / The cylindrical lens consists of a lead zirconate titanate (PZT) coated wire or optical fibre. The fabrication and evaluation of the lens is described. Ultrasonic measurements of miniature transducers and electronically controllable deflection are demonstrated. A novel laser ablation patterning technique is used to make an interdigital transducer with a 20$ mu$m resolution on the outer electrode of the lens for special applications. Finally, possible applications of the cylindrical lens are outlined.

Computational Fluid Dynamics (CFD) are used to investigate two speech phenomena. The first phenomenon is the English bilabial plosive /pa/. Simulations are compared with microphone recordings and high speed video recordings to study the penetration rate and strength of the jet associated with the plosive /pa/. It is found that the dynamics in the first 10ms of the plosive are critical to penetration rate, and the static simulation was not able to capture this effect. However, the simulation is able to replicate the penetration rate after the initial 10ms. The second speech phenomenon is the English fricative /sh/. Here, the goal is to simulate the sound created during /sh/ to understand the flow mechanisms involved with the creation of this sound and to investigate the simulation design required to predict the sound adequately. A variety of simulation methods are tested, and the results are compared with previously published experimental results. It is found that all Reynolds-Averaged Navier-Stokes (RANS) simulations give bad results, and 2D Large Eddy Simulations (LES) also have poor results. The 3D LES simulations show the most promise, but still do not produce a closely matching spectra. It is found that the acoustic analogy matches the direct measurements fairly well in 3D simulations. The studies of /pa/ and /sh/ are compared and contrasted with each other. From the findings of the studies, and using theoretical considerations, arguments are made concerning which CFD methods are appropriate for speech research. The two studies are also considered for their direct applications to the field and future research directions which might be followed.

In this paper, we present a software tool for the computation of eigenvalues of starlike domains defined by polar boundary functions. We also offer and numerically test a conjecture on the monotonicity of the fundamental eigenvalues of the members of a family of starlike domains.