Showing papers by "Andreas Spanias published in 1998"

A Java signal analysis tool for signal processing experiments

Abstract: A program to simulate discrete time linear systems is presented. The software is written as a Java applet and can be accessed on the Internet. Object oriented programming allows the user to construct and simulate a variety of systems. The program uses a graphical user interface which is easy to learn and it provides a visualization of the system and signal flow. The software is currently being used at Arizona State University to support an online software laboratory for a senior level digital signal processing (DSP) course. This paper presents our experiences gained by using the program in a class setting and gives examples of possible laboratory problems.

A new phase model for sinusoidal transform coding of speech

Abstract: A phase modeling algorithm for sinusoidal analysis-synthesis of speech is presented, where short-time sinusoidal phases are approximated using a combination of linear prediction, spectral sampling, delay compensation, and phase correction techniques. The algorithm is different to phase compensation methods proposed for source-system LPC in that it has been tailored to sinusoidal representation of speech. Performance analysis on a large speech data base reveals an improvement in temporal and spectral signal matching, as well as in the subjective quality of reconstructed speech. The method can be applied to enhance phase matching in low bit rate sinusoidal coders, where underlying sine wave amplitudes are extracted from an all-pole model. Preliminary subjective results are presented for a 2.4 kb/s sinusoidal coder.

An adaptive system identification Java simulation for Internet based courseware

Abstract: In this paper, an educational software tool on adaptive filters is presented. The simulation software is written in Java and provides an interactive environment that allows students to investigate and understand adaptive filters. The program exposes students to the concepts and applications of adaptive filters with special emphasis on convergence properties. The different adaptive algorithms that are implemented include LMS, NLMS, BLMS, IIR-LMS and IIR-SHARF. A frequency response visualization module was also developed to observe the magnitude and phase response for the corresponding pole-zero placement in the z-domain. This software tool supports an online software laboratory in the digital signal processing class. The paper also provides examples of possible laboratory exercises.

An interactive GUI-based tool for signal and speech processing courses

Abstract: In this paper, the authors present interactive, graphical software for introducing speech coding in undergraduate and graduate courses on digital signal processing (DSP) and speech processing The software facilitates students in understanding and experimentation for practical applications of signal processing concepts introduced in DSP courses It consists of two complementary components The first is a MATLAB-based graphical simulation of a state-of-the-art speech coder This component of the software is designed to complement the theoretical aspects of speech coding with practical exposure to the algorithms using hands-on simulations The second component is an application that provides a framework for evaluation of different speech coding algorithms for a variety of applications With this tool, students can easily categorize the performance of different speech coding algorithms through objective measurements as well as subjective listening tests These tools could be used in entry-level graduate courses in DSP and speech coding