34th Midwest Symposium on Circuits and Systems-Final Program

Organized by the Naval Postgraduate School Monterey California. Cosponsored by the IEEE Circuits and Systems Society. Symposium Organizing Committee: General Chairman-Sherif Michael, Technical Program-Roberto Cristi, Publications-Michael Soderstrand, Special Sessions- Charles W. Therrien, Publicity: Jeffrey Burl, Finance: Ralph Hippenstiel, and Local Arrangements: Barbara Cristi

Optimizing Continued Fraction Expansion Based IIR Realization of Fractional Order Differ-Integrators with Genetic Algorithm

This is the author accepted manuscript. The final version is available from IEEE via the DOI in this record.Rational approximation of fractional order (FO) differ-integrators via Continued Fraction Expansion (CFE) is a well known technique. In this paper, the nominal structures of various generating functions are optimized using Genetic Algorithm (GA) to minimize the deviation in magnitude and phase response between the original FO element and the rationalized discrete time filter in Infinite Impulse Response (IIR) structure. The optimized filter based realizations show better approximation of the FO elements in comparison with the existing methods and is demonstrated by the frequency response of the IIR filters.This work has been supported by the Department of Science & Technology (DST), Govt. of India under the PURSE programme

Signal representation for symbolic and numerical processing

Originally presented as author's thesis (Ph. D.--Massachusetts Institute of Technology), 1986.Bibliography: leaves 232-235.Supported in part by the Advanced Research Projects Agency monitored by ONR under contract no. N00014-81-K-0742 Supported in part by the National Science Foundation under grant ECS-8407285 Supported in part by Sanders Associates, Inc and an Amoco Foundation Fellowship.Cory S. Myers

The design and implementation of a microprocessor controlled adaptive filter

This thesis describes the construction and implementation of a microprocessor controlled recursive adaptive filter applied as a noise canceller. It describes the concept of the adaptive noise canceller, a method of estimating the received signal corrupted with additive interference (noise). This canceller has two inputs, the primary input containing the corrupted signal and the reference input consisting of the additive noise correlated in some unknown way to the primary noise. The reference input is filtered and subtracted from the primary input without degrading the desired components of the signal. This filtering process is adaptive and based on Widrow-Hoff Least-Mean-Square algorithm. Adaptive filters are programmable and have the capability to adjust their own parameters in situations where minimum piori knowledge is available about the inputs. For recursive filters, these parameters include feed-forward (non-recursive) as well as feedback (recursive) coefficients. A new design and implementation of the adaptive filter is suggested which uses a high speed 68000 microprocessor to accomplish the coefficients updating operation. Many practical problems arising in the hardware implementation are investigated. Simulation results illustrate the ability of the adaptive noise canceller to have an acceptable performance when the coefficients updating operation is carried out once every N sampling periods. Both simulation and hardware experimental results are in agreement

Aerospace Applications of Microprocessors

An assessment of the state of microprocessor applications is presented. Current and future requirements and associated technological advances which allow effective exploitation in aerospace applications are discussed

A computer-aided design for digital filter implementation

Imperial Users onl

Automatic annotation of musical audio for interactive applications

PhDAs machines become more and more portable, and part of our everyday life, it becomes apparent that developing interactive and ubiquitous systems is an important aspect of new music applications created by the research community. We are interested in developing a robust layer for the automatic annotation of audio signals, to be used in various applications, from music search engines to interactive installations, and in various contexts, from embedded devices to audio content servers. We propose adaptations of existing signal processing techniques to a real time context. Amongst these annotation techniques, we concentrate on low and mid-level tasks such as onset detection, pitch tracking, tempo extraction and note modelling. We present a framework to extract these annotations and evaluate the performances of different algorithms. The first task is to detect onsets and offsets in audio streams within short latencies. The segmentation of audio streams into temporal objects enables various manipulation and analysis of metrical structure. Evaluation of different algorithms and their adaptation to real time are described. We then tackle the problem of fundamental frequency estimation, again trying to reduce both the delay and the computational cost. Different algorithms are implemented for real time and experimented on monophonic recordings and complex signals. Spectral analysis can be used to label the temporal segments; the estimation of higher level descriptions is approached. Techniques for modelling of note objects and localisation of beats are implemented and discussed. Applications of our framework include live and interactive music installations, and more generally tools for the composers and sound engineers. Speed optimisations may bring a significant improvement to various automated tasks, such as automatic classification and recommendation systems. We describe the design of our software solution, for our research purposes and in view of its integration within other systems.EU-FP6-IST-507142 project SIMAC (Semantic Interaction with Music Audio Contents); EPSRC grants GR/R54620; GR/S75802/01

THE APPLICATION OF REAL-TIME SOFTWARE IN THE IMPLEMENTATION OF LOW-COST SATELLITE RETURN LINKS

Digital Signal Processors (DSPs) have evolved to a level where it is feasible for digital modems with relatively low data rates to be implemented entirely with software algorithms. With current technology it is still necessary for analogue processing between the RF input and a low frequency IF but, as DSP technology advances, it will become possible to shift the interface between analogue and digital domains ever closer towards the RF input. The software radio concept is a long-term goal which aims to realise software-based digital modems which are completely flexible in terms of operating frequency, bandwidth, modulation format and source coding. The ideal software radio cannot be realised until DSP, Analogue to Digital (A/D) and Digital to Analogue (D/A) technology has advanced sufficiently. Until these advances have been made, it is often necessary to sacrifice optimum performance in order to achieve real-time operation. This Thesis investigates practical real-time algorithms for carrier frequency synchronisation, symbol timing synchronisation, modulation, demodulation and FEC. Included in this work are novel software-based transceivers for continuous-mode transmission, burst-mode transmission, frequency modulation, phase modulation and orthogonal frequency division multiplexing (OFDM). Ideal applications for this work combine the requirement for flexible baseband signal processing and a relatively low data rate. Suitable applications for this work were identified in low-cost satellite return links, and specifically in asymmetric satellite Internet delivery systems. These systems employ a high-speed (>>2Mbps) DVB channel from service provider to customer and a low-cost, low-speed (32-128 kbps) return channel. This Thesis also discusses asymmetric satellite Internet delivery systems, practical considerations for their implementation and the techniques that are required to map TCP/IP traffic to low-cost satellite return links