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Computer speech synthesis: to improve the naturalness of a formant-based speech synthesizer.January 1992 (has links)
by Leung Kai Hon. / Thesis (M.Phil.)--Chinese University of Hong Kong, 1992. / Includes bibliographical references (leaves 94-97). / INTRODUCTION / Chapter Part I. --- SPEECH SYNTHESIS THEORY / Chapter 1. --- Human Speech And Its Production Model --- p.4 / Chapter 1.1. --- Articulatory Organs / Chapter 1.2. --- Categories Of Sounds / Chapter 1.3. --- Speech As A Language / Chapter 1.4. --- Modeling Of Human Speech Production Process / Chapter 2. --- Conventional Speech synthesis --- p.8 / Chapter 2.1. --- Overview / Chapter 2.2. --- Concatenation Synthesis / Chapter 2.3. --- Synthesis-By-Rule / Chapter 2.4. --- Linear-Prediction Vocoder / Chapter 3. --- Speech Analysis In The Frequency domain --- p.16 / Chapter 3.1. --- Speech Analysis By Linear-Prediction / Chapter 3.2. --- Formants Calculation By LPC / Chapter 3.3. --- Formant Tracking / Chapter 3.4. --- Parametric Representation Of Formant Transition / Chapter 4. --- Formant-Based Speech Synthesis --- p.28 / Chapter 4.1. --- Synthesis Strategy / Chapter 4.2. --- Text-to-phonemes Processing / Chapter 4.3. --- Formant Transition Estimation / Chapter 4.4. --- Formant-to-speech Conversion / Chapter 5. --- Speaker Calibration Of A Formant Synthesizer --- p.45 / Chapter 5.1. --- Naturalness and Speaker Dependency / Chapter 5.2. --- A Feed-Back Speech Synthesis System To Enhance Speech-Dependent Characteristics / Chapter 5.3. --- Three experiments On Cantonese Synthesis / Chapter 5.4. --- More About Naturalness / Chapter Part II. --- SPEECH SYNTHESIZER IMPLEMENTATIONS / Chapter 6. --- ALAB -------A Speech Analyzing Tool --- p.65 / Chapter 6.1. --- Overview / Chapter 6.2. --- Speech Sampling And Playing / Chapter 6.3. --- Waveform Editing Facilities / Chapter 6.4. --- Energy Envelop Estimation / Chapter 6.5. --- FFT Analysis Of Speech / Chapter 6.6. --- LPC Analysis Of Speech / Chapter 6.7. --- TIMIT Database Accessing / Chapter 7. --- CALT ------- A Phoneme-To-Parameter Module --- p.71 / Chapter 7.1. --- Overview / Chapter 7.2. --- Input and Output Data / Chapter 7.3. --- Parameter Table Attributes / Chapter 7.4. --- Phonemes Classification / Chapter 7.5. --- Locating Formant Targets In Time Domain / Chapter 7.6. --- Synthesis Rule / Chapter 8. --- LSYNTH -------Formants-To-Speech Conversion --- p.79 / Chapter 8.1. --- Overview / Chapter 8.2. --- Input And Output Data / Chapter 8.3. --- Voicing Source / Chapter 8.4. --- Linear Smoothing Of The Gain Factor / Chapter 9. --- MATCHW ------Feed-Back Speech Synthesis system --- p.82 / Chapter 9.1. --- Overview / Chapter 9.2. --- Feed-Back Procedure / Chapter 9.3. --- Manual Parameter Prediction Process / Chapter 10. --- TPIT -------Six Tones Control Program in Cantonese --- p.84 / Chapter 10.1 --- Tonal Features In Cantonese / Chapter 10.2. --- Modeling Of Tone Trajectories In Cantonese / Chapter 10.3. --- Tone Patterns And Speaker Dependency / Chapter part III. --- CONCLUSION AND DISCUSSIONS / REFERENCE / Chapter APPENDIX A ----- --- Phoneme Symbols / Chapter APPENDIX B ----- --- Output File Format Of CALT / Chapter APPENDIX C----- --- Parameter Table Attributes / Chapter APPENDIX D----- --- TIMIT Database Accessing Scheme / Chapter APPENDIX E ----- --- LPC Analysis Modules / Chapter APPENDIX F------ --- Common Speech File Format
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Computer speech synthesis: a systematic method to extract synthesis parameters for formant synthesizers.January 1993 (has links)
by Yu Wai Leung. / Thesis (M.Phil.)--Chinese University of Hong Kong, 1993. / Includes bibliographical references (leaves 94-96). / Abstract --- p.1 / Introduction --- p.2 / Chapter 1. --- Human speech and its production model / Chapter 1.1 --- The human vocal system --- p.4 / Chapter 1.2 --- Speech production mechanism --- p.5 / Chapter 1.3 --- Acoustic properties of human speech --- p.5 / Chapter 1.4 --- Modeling the speech production process --- p.6 / Chapter 1.5 --- Speech as the spoken form of a language --- p.7 / Chapter 2. --- Speech analysis techniques / Chapter 2.1 --- Short time speech analysis and speech segmentation --- p.9 / Chapter 2.2 --- Pre-emphasis --- p.9 / Chapter 2.3 --- Linear predictive analysis --- p.10 / Chapter 2.4 --- Formant tracking --- p.13 / Chapter 2.5 --- Pitch determination --- p.20 / Chapter 3. --- Speech synthesis technology / Chapter 3.1 --- Overview --- p.24 / Chapter 3.2 --- Articulatory synthesis --- p.24 / Chapter 3.3 --- Concatenation synthesis --- p.24 / Chapter 3.4 --- LPC synthesis --- p.27 / Chapter 3.5 --- Formant speech synthesis --- p.28 / Chapter 3.6 --- Synthesis by rule --- p.29 / Chapter 4. --- LSYNTH: A parallel formant synthesizer / Chapter 4.1 --- Overview / Chapter 4.2 --- Synthesizer configuration: cascade and parallel --- p.32 / Chapter 4.3 --- Structure ofLSYNTH --- p.33 / Chapter 5. --- Automatic formant parameter extraction for parallel formant synthesizers / Chapter 5.1 --- Introduction --- p.47 / Chapter 5.2 --- The idea of a feedback analysis system --- p.48 / Chapter 5.3 --- Overview of the feedback analysis system --- p.49 / Chapter 5.4 --- Iterative spectral matching algorithm --- p.52 / Chapter 5.5 --- Results and discussions --- p.65 / Chapter 6. --- Generate formant trajectories in synthesis-by-rule systems / Chapter 6.1 --- Formant trajectories generation in synthesis-by-rule systems --- p.70 / Chapter 6.2 --- Modeling formant transitions --- p.71 / Chapter 6.3 --- Conventional formant transition calculation --- p.72 / Chapter 6.4 --- The 4-point Bezier curve model --- p.73 / Chapter 6.5 --- Modeling of formant transitions for Cantonese --- p.77 / Chapter 7. --- Some listening test results / Chapter 7.1 --- Introduction --- p.87 / Chapter 7.2 --- Tone recognition test --- p.87 / Chapter 7.3 --- Cantonese final recognition test --- p.89 / Chapter 7.4 --- Problems and discussions --- p.91 / Conclusion --- p.92 / References --- p.94 / Appendix A: The Cantonese phonetic system --- p.97 / "Appendix B: TPIT, A tone trajectory generator for Cantonese" --- p.103
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Low delay and low bit rate speech coding. / CUHK electronic theses & dissertations collectionJanuary 1996 (has links)
by Jian Zhang. / Thesis (Ph.D.)--Chinese University of Hong Kong, 1996. / Includes bibliographical references (p. 134-[144]). / Electronic reproduction. Hong Kong : Chinese University of Hong Kong, [2012] System requirements: Adobe Acrobat Reader. Available via World Wide Web. / Mode of access: World Wide Web.
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Short-time independent component analysis for blind separation of speech sources. / CUHK electronic theses & dissertations collectionJanuary 2007 (has links)
Among all the three LOD types, the Dominant LOD manifests to be with comparatively higher efficiency in yielding accurate separation performance. The production mechanism of the Dominant LOD indicates that higher energy ratio of sources helps to build this type of LOD. Considering the sparse energy distribution of speech signals in the time-frequency domain, the Dominant LOD may arise in some short time subbands even though it appears to be Non-dominant LOD in its fullband. Therefore the proposed LOD-based ICA is extended to the frequency subbands for more opportunities to attain such Dominant LOD type. / Based on the insight into the effect on the aforementioned problems by the input sources as well as the mixing channel, three basic short time Local Optima Distribution (LOD) types are investigated. Information is derived from the characteristics of these LOD types for: (1) choosing simultaneous or sequential ICA algorithm; (2) shrinking feasible search region; and (3) producing possible initial points in search of the de-mixing matrix. As a result, the technique of LOD-based ICA is developed in this thesis to assign different procedures according to the LOD type of the observed short time mixtures. The analytical and simulation results demonstrated that more accurate de-mixing matrix estimation could be obtained; thereby producing improved separation performance. / Independent Component Analysis (ICA) has long been regarded as a powerful technique for speech source separation. In practice, however, speaker moving or reverberant environments may necessitate ICA to be implemented in short time intervals, which makes the fundamental assumption of sources' independence collapse in ICA. This leads to two important but often overlooked problems, namely: (1) excursion of global optimum from the desired solution and (2) diffusion of local optima in search of the de-mixing matrix. These two problems occur in most practical situations and greatly degrade the performance of the existing ICA algorithms. / The effectiveness of the proposed short time LOD-based ICA is validated by applying it to a speaker-moving model and a mixing system with abrupt changes, which approaches the practical applications better since the mixing system is not always constant as in standard ICA model. We have also explored the separation task with noise-contaminated speech signals. This suggests us that: other than the long time analysis, the short time analysis may provide an alternative means with extra information for separation when the independence information is impaired and subsequently fails to yield the desirable separation performance. / Zhang, Jing. / "July 2007." / Adviser: Ching Pak Chung. / Source: Dissertation Abstracts International, Volume: 69-01, Section: B, page: 0579. / Thesis (Ph.D.)--Chinese University of Hong Kong, 2007. / Includes bibliographical references. / Electronic reproduction. Hong Kong : Chinese University of Hong Kong, [2012] System requirements: Adobe Acrobat Reader. Available via World Wide Web. / Electronic reproduction. [Ann Arbor, MI] : ProQuest Information and Learning, [200-] System requirements: Adobe Acrobat Reader. Available via World Wide Web. / Abstracts in English and Chinese. / School code: 1307.
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Speech synthesis via adaptive Fourier decompositionLiu, Zhu Lin January 2011 (has links)
University of Macau / Faculty of Science and Technology / Department of Mathematics
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Enhancement and recognition of whispered speechMorris, Robert W., January 2003 (has links) (PDF)
Thesis (Ph. D.)--School of Electrical and Computer Engineering, Georgia Institute of Technology, 2004. Directed by Mark A. Clements. / Vita. Includes bibliographical references (leaves 152-158).
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Towards a real-time implementation of loudness enhancement algorithms on a Motorola DSP 56600Sabuwala, Adnan H. January 2002 (has links)
Thesis (M.S.)--University of Florida, 2002. / Title from title page of source document. Includes vita. Includes bibliographical references.
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Real-world evaluation of mobile phone speech enhancement algorithmsO'Rourke, William Thomas. January 2002 (has links)
Thesis (M.S.)--University of Florida, 2002. / Title from title page of source document. Includes vita. Includes bibliographical references.
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Improved clipped speech systemsBoulay, Paul Frederick, 1936- January 1964 (has links)
No description available.
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The stability of pitch synthesis filters in speech coding /Lam, Victor T. M. January 1985 (has links)
No description available.
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