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Tesi sul tema "Speech processing systems"

Autore: Grafiati

Pubblicato: 4 giugno 2021

Ultima modifica: 30 luglio 2024

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1

Coetzee, H. J. "The development of a new objective speech quality measure for speech coding applications." Diss., Georgia Institute of Technology, 1990. http://hdl.handle.net/1853/15474.

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2

Morris, Robert W. "Enhancement and recognition of whispered speech." Diss., Available online, Georgia Institute of Technology, 2004:, 2003. http://etd.gatech.edu/theses/available/etd-04082004-180338/unrestricted/morris%5frobert%5fw%5f200312%5fphd.pdf.

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3

Quackenbush, Schuyler Reynier. "Objective measures of speech quality." Diss., Georgia Institute of Technology, 1995. http://hdl.handle.net/1853/13376.

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4

Lucey, Simon. "Audio-visual speech processing." Thesis, Queensland University of Technology, 2002. https://eprints.qut.edu.au/36172/7/SimonLuceyPhDThesis.pdf.

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Speech is inherently bimodal, relying on cues from the acoustic and visual speech modalities for perception. The McGurk effect demonstrates that when humans are presented with conflicting acoustic and visual stimuli, the perceived sound may not exist in either modality. This effect has formed the basis for modelling the complementary nature of acoustic and visual speech by encapsulating them into the relatively new research field of audio-visual speech processing (AVSP). Traditional acoustic based speech processing systems have attained a high level of performance in recent years, but the p

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5

Chiou, Fred Y. "User-interactive speech enhancement using fuzzy logic." Diss., Georgia Institute of Technology, 1998. http://hdl.handle.net/1853/14916.

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6

陳我智 and Ngor-chi Chan. "Text-to-speech conversion for Putonghua." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 1990. http://hub.hku.hk/bib/B31209580.

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7

Barger, Peter James. "Speech processing for forensic applications." Thesis, Queensland University of Technology, 1998. https://eprints.qut.edu.au/36081/1/36081_Barger_1998.pdf.

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This thesis examines speech processing systems appropriate for use in forensic analysis. The need for automatic speech processing systems for forensic use is justified by the increasing use of electronically recorded speech for communication. An automatic speaker identification and verification system is described which was tested on data gathered by the Queensland Police Force. Speaker identification using Gaussian mixture models (GMMs) is shown to be useful as an indicator of identity, but not sufficiently accurate to be used as the sole means of identification. It is shown that training G

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8

Yatrou, Paul M. "Analysis of predictor mistracking in ADPCM speech coders." Thesis, McGill University, 1987. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=66242.

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9

Fang, Jie. "Design of secure speech encryption systems." Thesis, Queensland University of Technology, 1990. https://eprints.qut.edu.au/36471/1/36471_Fang_1990.pdf.

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This thesis investigates the design of digital speech encryption systems based on low bit rate vocoders. The speech quality and the cryptographic strength of the system are determined by vocoder and encryptor respectively. Three different low bit rate vocoders, 2400 BPS LPC ( Linear Prediction Coding) vocoder, 9600 BPS MELPC (Mul tipulse Excited Linear Prediction Coding) vocoder and 4800 BPS CELP (Codebook Excited Linear Prediction coding) vocoder, have been simulated. The performances of these vocoders are evaluated by using four objective measures. The thesis considers the follows

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10

Liu, Zhu Lin. "Speech synthesis via adaptive Fourier decomposition." Thesis, University of Macau, 2011. http://umaclib3.umac.mo/record=b2493215.

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11

Chan, Ngor-chi. "Text-to-speech conversion for Putonghua /." [Hong Kong : University of Hong Kong], 1990. http://sunzi.lib.hku.hk/hkuto/record.jsp?B12929475.

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12

Mazel, David S. "Sinusoidal modeling of speech." Thesis, Georgia Institute of Technology, 1986. http://hdl.handle.net/1853/13873.

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13

Alphonso, Issac John. "Network training for continuous speech recognition." Master's thesis, Mississippi State : Mississippi State University, 2003. http://library.msstate.edu/etd/show.asp?etd=etd-10252003-105104.

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14

Little, M. A. "Biomechanically informed nonlinear speech signal processing." Thesis, University of Oxford, 2007. http://ora.ox.ac.uk/objects/uuid:6f5b84fb-ab0b-42e1-9ac2-5f6acc9c5b80.

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Linear digital signal processing based around linear, time-invariant systems theory finds substantial application in speech processing. The linear acoustic source-filter theory of speech production provides ready biomechanical justification for using linear techniques. Nonetheless, biomechanical studies surveyed in this thesis display significant nonlinearity and non-Gaussinity, casting doubt on the linear model of speech production. In order therefore to test the appropriateness of linear systems assumptions for speech production, surrogate data techniques can be used. This study uncovers sys

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15

Wark, Timothy J. "Multi-modal speech processing for automatic speaker recognition." Thesis, Queensland University of Technology, 2001.

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16

Cao, Yuchang. "Speech enhancement with single and multiple microphones." Thesis, Queensland University of Technology, 1996.

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17

Chung, Jae H. "A new homomorphic vocoder framework using analysis-by-synthesis excitation analysis." Diss., Georgia Institute of Technology, 1991. http://hdl.handle.net/1853/15471.

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18

Crosmer, Joel R. "Very low bit rate speech coding using the line spectrum pair transformation of the LPC coefficients." Diss., Georgia Institute of Technology, 1985. http://hdl.handle.net/1853/15739.

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19

Rose, Richard C. "The design and performance of an analysis-by-synthesis class of predictive speech coders." Diss., Georgia Institute of Technology, 1988. http://hdl.handle.net/1853/16693.

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20

Wang, Raymond Jian-Wei. "Neurocomputing systems for auditory processing." Thesis, The University of Sydney, 1998. https://hdl.handle.net/2123/26278.

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This thesis studies neural computation models and neuromorphic implementations of the auditory pathway with applications to cochlear implants and artiﬁcial auditory sensory and processing systems. Very low power analogue computation is addressed through the design of micropower analogue building blocks and an auditory preprocessing module targeted at cochlear implants. The analogue building blocks have been fabricated and tested in a standard Complementary Metal Oxide Silicon (CMOS) process. The auditory pre-processing module design is based on the cochlea signal processing mechanisms

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21

Fisher, Andrew John. "Speech enhancement for forensic applications." Thesis, Queensland University of Technology, 1995. https://eprints.qut.edu.au/36243/1/36243_Fisher_1995.pdf.

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Abstract (sommario):

Law enforcement agencies often engage in surveillance operations which involve the recording of spoken conversations. As is often the case, these recordings are made with a single microphone under covert conditions. Under this non-ideal situation, the speech signal is highly susceptible to be severely corrupted by various forms of noise, the most common of which is broadband in nature. This thesis presents a study conducted to investigate the enhancement of speech recordings for forensic applications. A new speech enhancement scheme has been proposed here, to provide noise reduction without

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22

Hosom, John-Paul. "Automatic time alignment of phonemes using acoustic-phonetic information /." Full text open access at:, 2000. http://content.ohsu.edu/u?/etd,282.

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23

Ikram, Muhammad Zubair. "Multichannel blind separation of speech signals in a reverberant environment." Diss., Georgia Institute of Technology, 2001. http://hdl.handle.net/1853/15023.

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24

Wilson, Shawn C. "Voice recognition systems : assessment of implementation aboard U.S. naval ships." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 2003. http://library.nps.navy.mil/uhtbin/hyperion-image/03Mar%5FWilson.pdf.

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Abstract (sommario):

Thesis (M.S. in Information Systems and Operations)--Naval Postgraduate School, March 2003.<br>Thesis advisor(s): Michael T. McMaster, Kenneth J. Hagan. Includes bibliographical references (p. 47-49). Also available online.

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25

Müller, J. J. "USB telephony interface device for speech recognition applications /." Link to the online version, 2005. http://hdl.handle.net/10019/1127.

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26

Boulis, Constantinos. "Topic learning in text and conversational speech /." Thesis, Connect to this title online; UW restricted, 2005. http://hdl.handle.net/1773/5914.

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27

Farges, Eric P. "An analysis-synthesis hidden Markov model of speech." Diss., Georgia Institute of Technology, 1987. http://hdl.handle.net/1853/14775.

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28

LeBlanc, Wilfrid P. (Wilfrid Paul) Carleton University Dissertation Engineering Electrical. "Speech coding at low to medium bit rates." Ottawa, 1992.

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29

Anderson, David Verl. "Audio signal enhancement using multi-resolution sinusoidal modeling." Diss., Georgia Institute of Technology, 1999. http://hdl.handle.net/1853/15394.

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30

Kale, Kaustubh R. "Low complexity, narrow baseline beamformer for hand-held devices." [Gainesville, Fla.] : University of Florida, 2003. http://purl.fcla.edu/fcla/etd/UFE0001223.

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31

Hild, Kenneth E. "Blind separation of convolutive mixtures using Renyi's divergence." [Gainesville, Fla.] : University of Florida, 2003. http://purl.fcla.edu/fcla/etd/UFE0002387.

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32

Keenaghan, Kevin Michael. "A Novel Non-Acoustic Voiced Speech Sensor Experimental Results and Characterization." Link to electronic thesis, 2004. http://www.wpi.edu/Pubs/ETD/Available/etd-0114104-144946/.

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33

Iyengar, Vasu. "A low delay 16 kbit/sec coder for speech signals /." Thesis, McGill University, 1987. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=63799.

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34

Ng, H. N. Elaine. "Effects of noise type on speech understanding." Click to view the E-thesis via HKUTO, 2006. http://sunzi.lib.hku.hk/hkuto/record/B37990159.

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35

Ng, H. N. Elaine, and 吳凱寧. "Effects of noise type on speech understanding." Thesis, The University of Hong Kong (Pokfulam, Hong Kong), 2006. http://hub.hku.hk/bib/B37990159.

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36

Lai, Yiu Pong. "Maximum likelihood normalization for robust speech recognition /." View Abstract or Full-Text, 2003. http://library.ust.hk/cgi/db/thesis.pl?ELEC%202003%20LAI.

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Thesis (M. Phil.)--Hong Kong University of Science and Technology, 2003.<br>Includes bibliographical references (leaves 98-103). Also available in electronic version. Access restricted to campus users.

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37

Li, Chak Fai. "Improved polynomial segment model for speech recognition /." View abstract or full-text, 2004. http://library.ust.hk/cgi/db/thesis.pl?ELEC%202004%20LI.

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Thesis (M. Phil.)--Hong Kong University of Science and Technology, 2004.<br>Includes bibliographical references (leaves 80-84). Also available in electronic version. Access restricted to campus users.

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38

Ertan, Ali Erdem. "Pitch-synchronous processing of speech signal for improving the quality of low bit rate speech coders." Diss., Georgia Institute of Technology, 2004. http://hdl.handle.net/1853/36534.

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39

Ertan, Ali Erdem. "Pitch-synchronous processing of speech signal for improving the quality of low bit rate speech coders." Available online, Georgia Institute of Technology, 2004:, 2003. http://etd.gatech.edu/theses/available/etd-06072004-131138/unrestricted/ertan%5Fali%5Fe%5F200405%5Fphd.pdf.

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Thesis (Ph. D.)--School of Electrical and Computer Engineering, Georgia Institute of Technology, 2004. Directed by Thomas P. Barnwell, III.<br>Vita. Includes bibliographical references (leaves 221-226).

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40

Macon, Michael W. "Speech synthesis based on sinusoidal modeling." Diss., Georgia Institute of Technology, 1996. http://hdl.handle.net/1853/13904.

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41

Lam, Victor T. M. "The stability of pitch synthesis filters in speech coding /." Thesis, McGill University, 1985. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=63361.

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42

O'Rourke, William Thomas. "Real-world evaluation of mobile phone speech enhancement algorithms." [Gainesville, Fla.] : University of Florida, 2002. http://purl.fcla.edu/fcla/etd/UFE0000585.

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43

Ellis, Richard T. "Speech enhancement system implemented in CMOS." Thesis, Georgia Institute of Technology, 2002. http://hdl.handle.net/1853/14801.

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44

Chu, Kam Keung. "Feature extraction based on perceptual non-uniform spectral compression for noisy speech recognition /." access full-text access abstract and table of contents, 2005. http://libweb.cityu.edu.hk/cgi-bin/ezdb/thesis.pl?mphil-ee-b19887516a.pdf.

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Thesis (M.Phil.)--City University of Hong Kong, 2005.<br>"Submitted to Department of Electronic Engineering in partial fulfillment of the requirements for the degree of Master of Philosophy" Includes bibliographical references (leaves 143-147)

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45

Wasmeier, Hans. "Development of tests and preprocessing algorithms for evaluation and improvement of speech recognition units." Thesis, University of British Columbia, 1986. http://hdl.handle.net/2429/26750.

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This study considered the evaluation of commercially available isolated word, speaker dependent, speech recognition units, and preprocessing techniques that may be used for improving their performance. The problem was considered in three separate stages. A series of tests were designed to exercise an isolated word, speaker dependent, speech recognition unit. These tests provided a sound basis for determining a given unit's strengths and weaknesses. This knowledge permits a more informed decision on the best recognition device for a given price range. As well, this knowledge may be used in the

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46

Lee, Spencer Jaehoon Gilbert Juan E. "Post-speech-recognition processiing in domain-specific text-corpus-based distributed listening system analysis, interpretation and selection of speech recognition results /." Auburn, Ala., 2006. http://repo.lib.auburn.edu/2006%20Summer/Theses/LEE_SPENCER_7.pdf.

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47

Rao, Ram Raghavendra. "Audio-visual interaction in multimedia." Diss., Georgia Institute of Technology, 1998. http://hdl.handle.net/1853/13349.

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48

Necioğlu, Burhan F. "Objectively measured descriptors for perceptual characterization of speakers." Diss., Georgia Institute of Technology, 1999. http://hdl.handle.net/1853/15035.

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49

McCree, Alan V. "A new LPC vocoder model for low bit rate speech coding." Diss., Georgia Institute of Technology, 1992. http://hdl.handle.net/1853/15053.

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50

Mathan, Luc Stefan. "Speaker-independent access to a large lexicon." Thesis, McGill University, 1987. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=63773.

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