Iterative seismic data interpolation using plane-wave shaping

textGeophysical applications often require finding an appropriate solution to an ill-posed inverse problem. An example application is interpolating irregular or sparse data to a regular grid. This data regularization problem must be addressed appropriately before many data processing techniques can begin. In this thesis, I investigate plane-wave shaping in two and three dimensions as a data regularization algorithm, which can be used for the interpolation of seismic data and images. I use plane-wave shaping to interpolate several synthetic and field datasets and test its accuracy in image reconstruction. Because plane-wave shaping adheres to the direction of the local slopes of an image, the image-guided interpolation scheme attempts to preserve information of geologic structures. I apply several alternative interpolation schemes - formulated as an inverse problem with a convolutional operator to constrain the model space - namely: plane-wave destruction, plane-wave construction, and prediction-error filters. Investigating their iterative convergence rates, I find that plane-wave shaping converges to a solution in fewer iterations than the alternative techniques. I find that the only required parameter for this method, the smoothing radius, is best chosen to be approximately the same size as the holes for missing-data problems. The optional parameter for edge padding is best selected as approximately half of the smoothing radius. Applications of this research project include potential applications in well-log interpolation, seismic tomography, and 5-D seismic data interpolation.Geological Science

Multichannel biomedical telemetry system using delta modulation

Telemetering of biomedical data from unrestrained subjects requires a system to be compact, reliable and efficient. A survey of the existing multi-channel biomedical telemetry showed that most of the systems employ analogue or uncoded (digital) techniques of encoding biomedical signals. These techniques are less reliable, employ wider bandwidth and are difficult to implement compared to the coded (digital) techniques of modulation. A theoretical study of the coded techniques of modulation for encoding biomedical signals showed-that pulse code modulation, though more efficient, calls for extensive circuitry and makes it expensive and difficult to implement. Delta modulation and delta sigma modulation were found to be simpler, easier to Implement and efficient. [Continues.

Hardward and algorithm architectures for real-time additive synthesis

Additive synthesis is a fundamental computer music synthesis paradigm tracing its origins to the work of Fourier and Helmholtz. Rudimentary implementation linearly combines harmonic sinusoids (or partials) to generate tones whose perceived timbral characteristics are a strong function of the partial amplitude spectrum. Having evolved over time, additive synthesis describes a collection of algorithms each characterised by the time-varying linear combination of basis components to generate temporal evolution of timbre. Basis components include exactly harmonic partials, inharmonic partials with time-varying frequency or non-sinusoidal waveforms each with distinct spectral characteristics. Additive synthesis of polyphonic musical instrument tones requires a large number of independently controlled partials incurring a large computational overhead whose investigation and reduction is a key motivator for this work. The thesis begins with a review of prevalent synthesis techniques setting additive synthesis in context and introducing the spectrum modelling paradigm which provides baseline spectral data to the additive synthesis process obtained from the analysis of natural sounds. We proceed to investigate recursive and phase accumulating digital sinusoidal oscillator algorithms, defining specific metrics to quantify relative performance. The concepts of phase accumulation, table lookup phase-amplitude mapping and interpolated fractional addressing are introduced and developed and shown to underpin an additive synthesis subclass - wavetable lookup synthesis (WLS). WLS performance is simulated against specific metrics and parameter conditions peculiar to computer music requirements. We conclude by presenting processing architectures which accelerate computational throughput of specific WLS operations and the sinusoidal additive synthesis model. In particular, we introduce and investigate the concept of phase domain processing and present several “pipeline friendly” arithmetic architectures using this technique which implement the additive synthesis of sinusoidal partials

A new user interface for musical timbre design

This thesis characterises and addresses problems and issues associated with the design of intuitive user interfaces for timbral control. The usability of a range of synthesis methods and representative implementations of these methods is assessed, and three interface architectures - fixed architecture, architecture specification and direct specification - are identified. The characteristics of each of these architectures, as well as problems of usability inherent to each of them are discussed; it is argued that none of them provide intuitive tools for the manipulation and control of timbre. The study examines the nature of timbre and the notion of timbre space; different kinds of timbre space are considered and criteria are proposed for the selection of suitable timbre spaces as vehicles for synthesis. A number of listening tests, designed to demonstrate the feasibility of subsequent work, were devised and carried out; the results of these tests provide evidence that, where Euclidean distances between sounds located in a given timbre space are reflected in perceptual distances, the ability of subjects to detect relative distances in different parts of the space varies with the perceptual granularity of the space. Three contrasting timbre spaces conforming to the proposed criteria for use in synthesis are constructed; the purpose of these spaces is to provide an environment for a novel user interaction approach for timbral design which incorporates a search strategy based on weighted centroid localization. Two prototypes which exemplify the proposed approach in alternative ways are designed, implemented and tested with potential users in order to validate the approach; a third contrasting prototype which represents a simple contrasting alternative is tested for purposes of comparison. The results of these tests are evaluated and discussed, and areas of further work identified

Algorithms and architectures for the multirate additive synthesis of musical tones

In classical Additive Synthesis (AS), the output signal is the sum of a large number of independently controllable sinusoidal partials. The advantages of AS for music synthesis are well known as is the high computational cost. This thesis is concerned with the computational optimisation of AS by multirate DSP techniques. In note-based music synthesis, the expected bounds of the frequency trajectory of each partial in a finite lifecycle tone determine critical time-invariant partial-specific sample rates which are lower than the conventional rate (in excess of 40kHz) resulting in computational savings. Scheduling and interpolation (to suppress quantisation noise) for many sample rates is required, leading to the concept of Multirate Additive Synthesis (MAS) where these overheads are minimised by synthesis filterbanks which quantise the set of available sample rates. Alternative AS optimisations are also appraised. It is shown that a hierarchical interpretation of the QMF filterbank preserves AS generality and permits efficient context-specific adaptation of computation to required note dynamics. Practical QMF implementation and the modifications necessary for MAS are discussed. QMF transition widths can be logically excluded from the MAS paradigm, at a cost. Therefore a novel filterbank is evaluated where transition widths are physically excluded. Benchmarking of a hypothetical orchestral synthesis application provides a tentative quantitative analysis of the performance improvement of MAS over AS. The mapping of MAS into VLSI is opened by a review of sine computation techniques. Then the functional specification and high-level design of a conceptual MAS Coprocessor (MASC) is developed which functions with high autonomy in a loosely-coupled master- slave configuration with a Host CPU which executes filterbanks in software. Standard hardware optimisation techniques are used, such as pipelining, based upon the principle of an application-specific memory hierarchy which maximises MASC throughput

Analysis and resynthesis of polyphonic music

This thesis examines applications of Digital Signal Processing to the analysis, transformation, and resynthesis of musical audio. First I give an overview of the human perception of music. I then examine in detail the requirements for a system that can analyse, transcribe, process, and resynthesise monaural polyphonic music. I then describe and compare the possible hardware and software platforms. After this I describe a prototype hybrid system that attempts to carry out these tasks using a method based on additive synthesis. Next I present results from its application to a variety of musical examples, and critically assess its performance and limitations. I then address these issues in the design of a second system based on Gabor wavelets. I conclude by summarising the research and outlining suggestions for future developments

Probabilistic characterization and synthesis of complex driven systems

Thesis (Ph.D.)--Massachusetts Institute of Technology, School of Architecture and Planning, Program in Media Arts and Sciences, 2000.Includes bibliographical references (leaves 194-204).Real-world systems that have characteristic input-output patterns but don't provide access to their internal states are as numerous as they are difficult to model. This dissertation introduces a modeling language for estimating and emulating the behavior of such systems given time series data. As a benchmark test, a digital violin is designed from observing the performance of an instrument. Cluster-weighted modeling (CWM), a mixture density estimator around local models, is presented as a framework for function approximation and for the prediction and characterization of nonlinear time series. The general model architecture and estimation algorithm are presented and extended to system characterization tools such as estimator uncertainty, predictor uncertainty and the correlation dimension of the data set. Furthermore a real-time implementation, a Hidden-Markov architecture, and function approximation under constraints are derived within the framework. CWM is then applied in the context of different problems and data sets, leading to architectures such as cluster-weighted classification, cluster-weighted estimation, and cluster-weighted sampling. Each application relies on a specific data representation, specific pre and post-processing algorithms, and a specific hybrid of CWM. The third part of this thesis introduces data-driven modeling of acoustic instruments, a novel technique for audio synthesis. CWM is applied along with new sensor technology and various audio representations to estimate models of violin-family instruments. The approach is demonstrated by synthesizing highly accurate violin sounds given off-line input data as well as cello sounds given real-time input data from a cello player.by Bernd Schoner.Ph.D

NASA Space Engineering Research Center Symposium on VLSI Design

The NASA Space Engineering Research Center (SERC) is proud to offer, at its second symposium on VLSI design, presentations by an outstanding set of individuals from national laboratories and the electronics industry. These featured speakers share insights into next generation advances that will serve as a basis for future VLSI design. Questions of reliability in the space environment along with new directions in CAD and design are addressed by the featured speakers