An investigation of the non-uniform sampling technique as applied to digital filter designs will be made. The objective of the design is to reduce the interference problems as one would· encounter in using uniform sampling technique in the synthesis. An analysis of the error function which measures the goodness in approximating a desired frequency response will also be undertaken. An algorithm which determines the optimal parameters.for a high pass filter will be developed and used to synthesize the particular high pass filter. The results of this design, the frequency response and its approximation error will be studied and evaluated. / Master of Science

A small-signal analysis of a two-stage distributed power system is performed. Although the distributed power system is composed of conventional PWM converters, the analysis methods can be extended to systems using other converter topologies. The analysis focuses on two important features of a two-stage distributed power system: parallel power modules and cascaded regulators. The small-signal characteristics of parallel module regulators are analyzed, and an expression for the loop-gain of a parallel-module system is obtained. It is found that a parallel module system can be configured so that the loop-gain of the system is independent of the number of modules. The effects of placing switching regulators in cascade are analyzed. Of primary concern is the effect of the second stage's dynamic impedance on the first stage's control loops. It is found that the negative dynamic resistance of the second stage gives rise to right-half-plane poles in the opened-loop gains of the first stage. Fundamental loop-gain analysis is used to determine that stability of the system. It is found that the first stage of a cascaded regulator system must satisfy additional design constraints in order to be stable while driving a negative dynamic resistance. / Master of Science

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No description available.

No description available.

This report introduces a novel architecture for a General Purpose Digital Signal Processing System and applies the system to implement a digital hearing aid. The theory and implementation of the general purpose digital signal processing system revolve around the architecture of the digital signal processor (DSP) and its use. The system consists of three subsystems: the Analog Interface Board, the DAAD Board, and the DSP Board. The general purpose digital signal processing system described takes into consideration both the basic needs of such a system as well as the many features which make it efficient in a wide range of applications. The system was used as a testbed for implementing various real-time DSP Algorithms. One of these algorithms is concerned with the problem of hearing loss. The final implementation of the digital hearing aid examines both the feasibility of the DHA as well as the usefulness of the general purpose digital signal processing system in a random application. Suggestions for future modification and expansion are discussed.

Indiana University-Purdue University Indianapolis (IUPUI) / The recording and processing of bioelectric signals over the decades has led to the development of many different types of analog filtering and amplification techniques. Meanwhile, there have also been many advancements in the realm of digital signal processing that allow for more powerful analysis of these collected signals. The issues with present acquisition schemes are that (1) they introduce irreversible distortion to the signals and may ultimately hinder analyses that rely on the unique morphological differences between bioelectric signal events and (2) they do not allow the collection of frequencies in the signal from direct-current (DC) to high-frequencies. The project put forth aims to overcome these two issues and present a new scheme for bioelectric signal acquisition and processing. In this thesis, a system has been developed, verified, and validated with experimental data to demonstrate the ability to build an invertible universal amplifier and digital restoration scheme. The thesis is primarily divided into four sections which focus on (1) the introduction and background information, (2) theory and development, (3) verification implementation and testing, and (4) validation implementation and testing. The introduction and background provides pertinent information regarding bioelectric signals and recording practices for bioelectric signals. It also begins to address some of the issues with the classical and present methods for data acquisition and make the case for why an invertible universal amplifier would be better. The universal amplifier transfer function and architecture are discussed and presented along with the development and optimization of the characterization and the inversion, or restoration, filter process. The developed universal amplifier, referred to as the invertible universal amplifier (IUA), while the universal amplifier and the digital restoration scheme together are referred to as the IUA system. The IUA system is then verified on the bench using typical square, sine, and triangle waveforms with varying offsets and the results are presented and discussed. The validation is done with in-vivo experiments showing that the IUA system may be used to acquire and process bioelectric signals with percent error less than to 6% when post-processed using estimated characteristics of and when compared to a standard flat bandwidth high-pass cutoff amplifier.

Thesis(M. Tech.) - Central University of Technology, Free State, 2004 / Growing markets leads to an increase in production. In these modern industries, weight measurement is of high priority. Weight measurement instrumentation is used for quality control, as well as for effective process control. Ineffective instrumentation with inaccurate data will influence the production process and profit margins negatively. Experimental data is gathered from an angled load cell, placed as a crossover between two conveyer belts. A weight measurement instrument with the ability to acquire accurate measurement of individual, moving parts is produced with the aid of DSP techniques. This was accomplished by analyzing the frequency spectrum for the undesirable signals with the use of Wavelets transformations (WT) and Fourier transformations (FT). After these undesired signals were identified a digital filter was designed to remove the undesired signals. Repetition of performance is achieved by the automatic zeroing of the instrument after every individual measurement. This weight measurement instrumentation also has the ability to store data consisting of the amount of objects and their individual weights. This instrument can also determine the material of which an object is made of. This is done by calculating the friction coefficient. This function has the ability to effectively identify between iron and rubber components irrespective of their mass or area.

Thesis (MScEng (Electrical and Electronic Engineering))--Stellenbosch University, 2008. / A singing transcription system transforming acoustic input into MIDI note sequences is presented. The transcription system is incorporated into a pronunciation-independent sight-singing tutor system, which provides note-level feedback on the accuracy with which each note in a sequence has been sung. Notes are individually modeled with hidden Markov models (HMMs) using untuned pitch and delta-pitch as feature vectors. A database consisting of annotated passages sung by 26 soprano subjects was compiled for the development of the system, since no existing data was available. Various techniques that allow efficient use of a limited dataset are proposed and evaluated. Several HMM topologies are also compared, in analogy with approaches often used in the field of automatic speech recognition. Context-independent note models are evaluated first, followed by the use of explicit transition models to better identify boundaries between notes. A non-repetitive grammar is used to reduce the number of insertions. Context-dependent note models are then introduced, followed by context-dependent transition models. The aim in introducing context-dependency is to improve transition region modeling, which in turn should increase note transcription accuracy, but also improve the time-alignment of the notes and the transition regions. The final system is found to be able to transcribe sung passages with around 86% accuracy. Finally, a note-level sight-singing tutor system based on the singing transcription system is presented and a number of note sequence scoring approaches are evaluated.

Blind source separation (BSS) refers to a class of methods by which multiple sensor signals are combined with the aim of estimating the original source signals. Independent component analysis (ICA) is one such method that effectively resolves static linear combinations of independent non-Gaussian distributions. We propose a method that can track variations in the mixing system by seeking a compromise between adaptive and block methods by using mini-batches. The resulting permutation indeterminacy is resolved based on the correlation continuity principle. Methods employing higher order cumulants in the separation criterion are susceptible to outliers in the finite sample case. We propose a robust method based on low-order non-integer moments by exploiting the Laplacian model of speech signals. We study separation methods for even (over)-determined linear convolutive mixtures in the frequency domain based on joint diagonalization of matrices employing time-varying second order statistics. We investigate the sources affecting the sensitivity of the solution under the finite sample case such as the set size, overlap amount and cross-spectrum estimation methods. / by Savaskan Bulek. / Thesis (Ph.D.)--Florida Atlantic University, 2010. / Includes bibliography. / Electronic reproduction. Boca Raton, Fla., 2010. Mode of access: World Wide Web.