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Audio compression and speech enhancement using temporal masking modelsGunawan, Teddy Surya, Electrical Engineering & Telecommunications, Faculty of Engineering, UNSW January 2007 (has links)
Of the few existing models of temporal masking applicable to problems such as compression and enhancement, none are based on empirical data from the psychoacoustic literature, presumably because the multidimensional nature of the data makes the derivation of tractable functional models difficult. This thesis presents two new functional models of the temporal masking effect of the human auditory system, and their exploitation in audio compression and speech enhancement applications. Traditional audio compression algorithms do not completely utilise the temporal masking properties of the human auditory system, relying solely on simultaneous masking models. A perceptual wavelet packet-based audio coder has been devised that incorporates the first developed temporal masking model and combined with simultaneous masking models in a novel manner. An evaluation of the coder using both objective (PEAQ, ITU-R BS.1387) and extensive subjective tests (ITU-R BS.1116) revealed a bitrate reduction of more than 17% compared with existing simultaneous masking-based audio coders, while preserving transparent quality. In addition, the oversampled wavelet packet transform (ODWT) has been newly applied to obtain alias-free coefficients for more accurate masking threshold calculation. Finally, a low-complexity scalable audio coding algorithm using the ODWT-based thresholds and temporal masking has been investigated. Currently, there is a strong need for innovative speech enhancement algorithms exploiting the auditory masking effects of human auditory system that perform well at very low signal-to-noise ratio. Existing competitive noise suppression algorithms and those that incorporate simultaneous masking were examined and evaluated for their suitability as baseline algorithms. Objective measures using PESQ (ITU-T P.862) and subjective measures (ITU-T P.835) demonstrate that the proposed enhancement scheme, based on a second new masking model, outperformed the seven baseline speech enhancement methods by at least 6- 20% depending on the SNR. Hence, the proposed speech enhancement scheme exploiting temporal masking effects has good potential across many types and intensities of environmental noise. Keywords: human auditory system; temporal masking; simultaneous masking; audio compression; speech enhancement; subjective test; objective test.
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A trainable hearing aidZakis, Justin Andrew Unknown Date (has links) (PDF)
The main findings of this research project were that under controlled acoustic conditions, a hearing aid can be trained to provide amplification settings that are closer to hypothetical preferred settings than were the initial untrained settings, and in everyday acoustic environments, hearing aid users can train an aid to provide an amplification settings that they prefer to the untrained settings on a significant majority of occasions.
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Improvement of decoding engine & phonetic decision tree in acoustic modeling for online large vocabulary conversational speech recognitionXue, Jian, January 2007 (has links)
Thesis (Ph. D.)--University of Missouri-Columbia, 2007. / The entire dissertation/thesis text is included in the research.pdf file; the official abstract appears in the short.pdf file (which also appears in the research.pdf); a non-technical general description, or public abstract, appears in the public.pdf file. Title from title screen of research.pdf file (viewed on March 4, 2008) Vita. Includes bibliographical references.
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A statistical approach to formant tracking /Gayvert, Robert T. January 1988 (has links)
Thesis (M.S.)--Rochester Institute of Technology, 1989. / Includes bibliographical references (leaves 20-21).
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Simple and efficient solutions to the problems associated with acoustic echo cancellationMohammad, Asif Iqbal, January 2007 (has links) (PDF)
Thesis (Ph. D.)--University of Missouri--Rolla, 2007. / Vita. The entire thesis text is included in file. Title from title screen of thesis/dissertation PDF file (viewed November 29, 2007) Includes bibliographical references (p. 66-67).
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Phase-based speech processing /Shi, Guangji. January 2006 (has links)
Thesis (Ph. D.)--University of Toronto, 2006. / Source: Dissertation Abstracts International, Volume: 67-06, Section: B, page: 3354. Advisor: Parham Aarabi. Includes bibliographical references.
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Stochastic resonance in a neuron model with application to the auditory pathway /Hohn, Nicolas. January 2000 (has links)
Thesis (M.Sc.)--University of Melbourne, Dept. of Otolaryngology, 2000. / Typescript (photocopy). Includes bibliographical references (leaves 99-109).
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Wideband extension of narrowband speech for enhancement and coding /Epps, Julien. January 2000 (has links)
Thesis (Ph. D.)--University of New South Wales, 2000. / Online copy varies slightly. Also available online.
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Automatic formant labeling in continuous speech /Richards, Elizabeth A. January 1989 (has links)
Thesis (M.S.)--Rochester Institute of Technology, 1989. / Includes bibliographical references (leaves 81-85).
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Vowel recognition in continuous speech /Stam, Darrell C. January 1900 (has links)
Thesis (M.S.)--Rochester Institute of Technology, 1989. / Includes bibliographical references (leaves 73-75).
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