Thesis (M. Phil.)--University of Hong Kong, 1991.

Thesis (M.S.)--University of Florida, 2003. / Title from title page of source document. Includes vita. Includes bibliographical references.

Thesis (Ph. D.)--University of Florida, 2003. / Title from title page of source document. Includes vita. Includes bibliographical references.

Thesis (Ph. D.)--Carleton University, 1993. / Also available in electronic format on the Internet.

Thesis (M.S.)--Rowan University, 2008. / Typescript. Includes bibliographical references.

An adaptive optimal silence detector is designed and implemented in four speech coding schemes: N-bit PCM (N = 5 to 12), N-bit A-law PCM (N = 4 to 8), N-bit ADPCM (N = 3 to 8) and ADM (Adaptive Delta Modulation) for bit-rates of 16Kps, 24Kps and 32Kps. The amount of compression is approximately 35% for voice recordings such as radio newscasts, highly active conversations and readings from prepared texts. Subjective evaluation shows that the silence-edited versions (silence played back as absolute silence) have acceptability scores of 1.07 lower than the unedited versions with respect to a specific coding scheme for a score range of 1 to 5. With noise-edited versions (silence replaced by random noise during playback) the score degradation is 0.5. / Applied Science, Faculty of / Electrical and Computer Engineering, Department of / Graduate

No description available.

No description available.

The feasibility of representing human speech by serial digital codes was investigated by exercising specially constructed digital logic coupled with standard audio output equipment. The theories being tested represent a radical departure from previous efforts in the field of speech research. Therefore, this initial investigation was limited in scope to a study of unconnected English language speech sounds at the phenome level. The experiments were conducted in two parts, with the first being the development of serialized digital codes, for selected speech sounds, derived from actual human speech. The second part was to synthesize these sounds using the specially constructed digital synthesizer, and have human listeners analyze them for intelligibility. The results seem to indicate that this is a viable scheme for speech synthesis.

This paper studies the quality of synthetic speech produced by integrated circuit (IC) hardware using fixed-point arithmetic and Linear Predictive Coding (LPC). A theoretical model explaining the combined effects of finite wordlength and parametric model order is developed. This model is used to predict the results obtained in the experimental phase of this study. In the experimental phase, selected model utterances are synthesized under finite wordlength constraints using LPC parameters. The synthetic speech is evaluated in terms of the log area ratios which define objective speech quality as a parametric distance. A theoretical model is developed to predict the experimental results. Simulations of this model produce data that predict the experimental results. The same information is extracted from the model as that obtained from actually running the fixed-point synthesizer simulator. Since the predictions of the theoretical model agree quite well with the experimental measurements, it is concluded that fixed-point synthesizer performance can be predicted without actually running a complicated and expensive fixed-point synthesizer. Secondly, results obtained from either method clearly indicate that for 15 or 16 bits, ten is the best number of poles to use. Eight useable poles are indicated for 14 bits, while seven are indicated for 13 bits. Based on the results of this study, the use of less than 13 bits for fixed-point calculations is not recommended.