Microcomputer-based music synthesis hardware is being developed at North Texas State University (NTSU). The work described in this paper continues this effort to develop hardware designs for inexpensive, but good quality, sound synthesizers. In order to pursue their activities, researchers in computer assisted instruction in music theory, psychoacoustics, and music composition need quality sound sources. The ultimate goal of my research is to develop good quality sound synthesis hardware which can fill these needs economically. This paper explores three topics: 1) how a computer makes music--a short nontechnical description; 2) what has been done previously--a review of the literature; and 3) what factors bear on the quality of microcomputer-based systems, including encoding of musical passages, software development, and hardware design. These topics lead to the discussion of a particular sound synthesizer which the author has designed.

This project focuses on a prototype system architecture which integrates features of an event-driven gate-level simulator and features of the multiple expert system architecture, HEARSAY-II. Combining artificial intelligence and simulation techniques, a knowledge-based simulator was designed and constructed to model non-standard circuit behavior. This non-standard circuit behavior is amplified by advances in integrated circuit technology. Currently available digital circuit simulators can not simulate this behavior. Circuit designer expertise on behavioral phenomena is used in the expert system to guide the base simulator by manipulating its events to achieve the desired behavior.

The first chapter discusses the concept of macros: its definition, structure, usage, design goals, and the related prior work. This thesis principally concerns my work on OLMP (an On-Line Macro Processor for the Motorola 6800 Microprocessor), which is a macro processor which interacts with the user. It takes Motorola assembler source code and macro definitions as its input; after the appropriate editing and expansions, it outputs the expanded assembler source statements. The functional objectives, the design for implementation of OLMP, the basic macro format, and the macro definition construction are specified in Chapter Two. The software and the hardware environment of OLMP are discussed in the third chapter. The six modules of OLMP are the main spine of the fourth chapter. The comments on future improvement and how to link OLMP with the Motorola 6800 assembler are the major concern of the final chapter.

The problem addressed by this thesis is the need for a software measurement tool that enforces a uniform measurement algorithm on several programming languages. The introductory chapter discusses the concern for software measurement and provides background for the specific models and metrics that are studied. A multilingual software measurement tool is then introduced, that analyzes programs written in Ada, C, Pascal, or PL/I, and quantifies over thirty different program attributes. Metrics computed by the program include McCabe's measure of cyclomatic complexity and Halstead's software science metrics. Some results and conclusions of preliminary data analysis, using the tool, are also given. The appendices contain exhaustive counting algorithms for obtaining the metrics in each language.

This paper describes multitasking facilities of Ada in concurrent and real-time programming. Synchronization and process communication mechanisms are discussed in detail, also, a new mechanism to solve the scheduling problem is developed. In the concurrent programming aspect, a comparison is made between Ada's rendezvous and Pascal's Monitor concept. In the real-time programming aspect, the differences between the Ada multitasking and the traditional "cyclic executive approaches are contrasted and their associated costs/benefits analyzed.

Speech sounds generated by a simple waveform synthesizer were used to create a vector quantization codebook for use in speech recognition. Recognition was tested over the TI-20 isolated word data base using a conventional DTW matching algorithm. Input speech was band limited to 300 - 3300 Hz, then passed through the Scott Instruments Corp. Coretechs process, implemented on a VET3 speech terminal, to create the speech representation for matching. Synthesized sounds were processed in software by a VET3 signal processing emulation program. Emulation and recognition were performed on a DEC VAX 11/750. The experiments were organized in 2 series. A preliminary experiment, using no vector quantization, provided a baseline for comparison. The original codebook contained 109 vectors, all derived from 2 formant synthesized sounds. This codebook was decimated through the course of the first series of experiments, based on the number of times each vector was used in quantizing the training data for the previous experiment, in order to determine the smallest subset of vectors suitable for coding the speech data base. The second series of experiments altered several test conditions in order to evaluate the applicability of the minimal synthesized codebook to conventional codebook training. The baseline recognition rate …