In a communication system, audio quality is one of the parameters by which the end user defines the value of a product. This thesis examines the term audio quality, breaking it down into two subsidiary components, speech quality and speech intelligibility. One key goal in assessing audio quality is quantifying it in an accurate and repeatable way. As a part of this project a system was developed that achieved this goal. The system was then used to evaluate a number of existing products based on speech quality and intelligibility. Using these results the relationship between the two parameters was investigated. Investigations were also conducted in order to determine and quantify the effect communication systems have on perceptual speech parameters, and examine the relationship between them and speech quality and intelligibility. Using the testing systems developed a possible method of audio quality optimization was investigated and tested. The analysis methods that were incorporated into the test suite included the Perceptual Evaluation of Speech Quality, the Speech Transmission Index, vowel space analysis and segmental, psychoacoustic based methods. The testing incorporated a number of different handheld portable radios as speakers.

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Thesis (Ed.D.)--Teachers College, Columbia University. / Typescript. Includes tables. Sponsor: Goodwin Watson. Dissertation Committee: Donald Tewksbury, Lyman Bryson, . Type B project. Includes bibliographical references (leaves 163-165); "Annotated audio-visual bibliography" (leaves 139-162).

The subject of this paper is old--so old that to some it is new again. In simpler days young people learned everything except the formal mechanics of education from the people and places surrounding them. The prevailing type of "book learning" could best be accomplished indoors, where an errant butterfly might less successfully distract attention from the sad and solemn business of learning to read, spell, and cipher. Unfortunately, as education broadened its scope, it did not move out of its imprisoning walls and children continued to read textbooks about methods of seed distribution, while, unnoticed, milkweed parachutes set sail outside. / Typescript. / "August, 1948." / "Submitted to the Graduate Council of Florida State University in partial fulfillment of the requirements for the degree of Master of Arts." / Advisor: R. L. Eyman, Professor Directing Paper. / Includes bibliographical references (leaves 74-76).

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The development and improvement of 3-D immersive audio is gaining momentum through the growing interest in virtual reality. Possible applications reach from recreating real world environments to immersive concerts and performances to exploiting big data acoustically. To improve the immersive experience several measures can be taken. The recording of the sound field, the spatialization and the development of the loudspeaker arrays are some of the greatest challenges. In this thesis, these challenges for improving immersive audio will be explored. First, there will be a short introduction about 3D audio and a review about the state of the art technology and research. Next, the thesis will provide an introduction to 3D loudspeaker arrays and describe the systems used during this research. Furthermore, the development of a new 16-element 3rd order sound field microphone will be described. Afterwards, different spatial audio algorithms such as higher order ambisonics, wave field synthesis and vector based amplitude panning will be described, analyzed and compared. For each spatialization algorithm, the quality of soundfield reproduction will be quantified using listener perception tests for clarity and sound source localization. / Master of Science

The position of a sound source can be accurately determined in both azimuth and elevation through the use of localisation cues extracted from the incident audio signals. Compared to lateral localisation, 2-D hardware localisation is novel and requires the extraction of spectral cues in addition to time delay cues. The objective of this work is to develop an analogue VLSI system which extracts these cues from audio signals arriving at the left and right channels of the system, and then map these cues to the source position. The use of analogue hardware, which is broadly adapted from the biological auditory system, enables fast and low power computation. To obtain accurate 2-D localisation from the hardware-extracted cues a novel algorithm for the mapping process has been developed. The performance of this algorithm is evaluated via simulation under different environmental conditions. The effects of hardware non-idealities on the localisation accuracy, including mismatches and noise are also assessed. The analogue hardware implementation is divided into three main sections: a front-end for splitting the input signal into different frequency bands and extraction of spectral cues, an onset detector for distinguishing between the incident portion and the echo portion of the acoustic signal, and a correlator for determination of time delay cues. Novel building blocks have been designed using standard CMOS in order to enable low voltage low power operation of the differential architecture essential for the accuracy of the extracted cues. A novel feedback technique enables accurately controlled Class AB operation of a low voltage switched-current memory cell. A novel cross-coupling technique ensures correct Class AB operation of a log-domain bandpass filter. The five chips developed here operate at ± 0.9 V supply. The system has been tested by applying audio signals convolved with a position-dependent transfer function at the input, and then processing the resulting hardware-generated cues. Measurement results show that 2-D localisation within 5° accuracy is achievable using hardware extracted cues. Key words: sound localisation, analogue VLSI, silicon cochlea, log domain, switched capacitor, switched current, current mode, analogue processing.