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Dissertations / Theses on the topic 'Speech waveforms'

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Published: 4 June 2021

Last updated: 8 February 2022

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1

Lowry, Andrew. "Efficient structures for vector quantisation of speech waveforms." Thesis, Queen's University Belfast, 1988. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.292596.

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2

Carandang, Alfonso B., and n/a. "Recognition of phonemes using shapes of speech waveforms in WAL." University of Canberra. Information Sciences & Engineering, 1994. http://erl.canberra.edu.au./public/adt-AUC20060626.144432.

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Generating a phonetic transcription of the speech waveform is one method which can be applied to continuous speech recognition. Current methods of labelling a speech wave involve the use of techniques based on spectrographic analysis. This paper presents a computationally simple method by which some phonemes can be identified primarily by their shapes. Three shapes which are regularly manifested by three phonemes were examined in utterances made by a number of speakers. Features were then devised to recognise their patterns using finite state automata combined with a checking mechanism. These

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3

Deivard, Johannes. "How accuracy of estimated glottal flow waveforms affects spoofed speech detection performance." Thesis, Mälardalens högskola, Akademin för innovation, design och teknik, 2020. http://urn.kb.se/resolve?urn=urn:nbn:se:mdh:diva-48414.

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In the domain of automatic speaker verification, one of the challenges is to keep the malevolent people out of the system. One way to do this is to create algorithms that are supposed to detect spoofed speech. There are several types of spoofed speech and several ways to detect them, one of which is to look at the glottal flow waveform (GFW) of a speech signal. This waveform is often estimated using glottal inverse filtering (GIF), since, in order to create the ground truth GFW, special invasive equipment is required. To the author’s knowledge, no research has been done where the correl

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4

Bamini, Praveen Kumar. "FPGA-based Implementation of Concatenative Speech Synthesis Algorithm." [Tampa, Fla.] : University of South Florida, 2003. http://purl.fcla.edu/fcla/etd/SFE0000187.

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5

Choy, Eddie L. T. "Waveform interpolation speech coder at 4 kbs." Thesis, McGill University, 1998. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=20901.

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Speech coding at bit rates near 4 kbps is expected to be widely deployed in applications such as visual telephony, mobile and personal communications. This research focuses on developing a speech coder based on the waveform interpolation (WI) scheme, with an attempt to deliver near toll-quality speech at rates around 4 kbps. A WI coder has been simulated in floating-point using the C programming language. The high performance of the WI model has been confirmed by subjective listening tests in which the unquantized coder outperforms the 32 kbps G.726 standard (ADPCM) 98% of the time under clean

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6

Leong, Michael. "Representing voiced speech using prototype waveform interpolation for low-rate speech coding." Thesis, McGill University, 1992. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=56796.

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In recent years, research in narrow-band digital speech coding has achieved good quality speech coders at low rates of 4.8 to 8.0 kb/s. This thesis examines the method proposed by W. B. Kleijn called prototype waveform interpolation (PWI) for coding the voiced sections of speech efficiently to achieve a coder below 4.8 kb/s while maintaining, even improving, the perceptual quality of current coders.<br>In examining the PWI method, it was found that although the method generally works very well there are occasional sections of the reconstructed voiced speech where audible distortion can be hear

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7

Choy, Eddie L. T. "Waveform interpolation speech coder at 4 kb/s." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1998. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape11/PQDD_0028/MQ50596.pdf.

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8

Davis, Andrew J. "Waveform coding of speech and voiceband data signals." Thesis, University of Liverpool, 1988. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.232946.

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9

Eide, Ellen Marie. "A linguistic feature representation of the speech waveform." Thesis, Massachusetts Institute of Technology, 1993. http://hdl.handle.net/1721.1/12510.

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Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1993.<br>Includes bibliographical references (leaves 95-97).<br>by Ellen Marie Eide.<br>Ph.D.

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10

Zeghidour, Neil. "Learning representations of speech from the raw waveform." Thesis, Paris Sciences et Lettres (ComUE), 2019. http://www.theses.fr/2019PSLEE004/document.

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Bien que les réseaux de neurones soient à présent utilisés dans la quasi-totalité des composants d’un système de reconnaissance de la parole, du modèle acoustique au modèle de langue, l’entrée de ces systèmes reste une représentation analytique et fixée de la parole dans le domaine temps-fréquence, telle que les mel-filterbanks. Cela se distingue de la vision par ordinateur, un domaine où les réseaux de neurones prennent en entrée les pixels bruts. Les mel-filterbanks sont le produit d’une connaissance précieuse et documentée du système auditif humain, ainsi que du traitement du signal, et son

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11

ARAUJO, ANTONIO MARCOS DE LIMA. "ANALYSIS OF WAVEFORM CODERS FOR SPEECH AND DATA SIGNALS." PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 1986. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=9246@1.

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O trabalho examina o comportamento de Codificadores de forma de onda operando a 32,56 e 64kbit/s para transmissão digital de sinais de voz e de sinais de dados PSK-8 a 4800 bit/s e QAM-16 a 9600 bit/s. A partir de uma análise detalhada dos diversos sistemas, tanto em canal ideal como um canal ruidoso, é verificada a necessidade de se fazer uma identificação do tipo de sinal. De modo a permitir sua codificação de forma mais eficiente. É, então, proposta e avaliada a utilização de uma técnica de identificação estatística de sinais de voz e dados, em codificadores de forma de onda. A in

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12

Yaghmaie, Khashayar. "Prototype waveform interpolation based low bit rate speech coding." Thesis, University of Surrey, 1997. http://epubs.surrey.ac.uk/843152/.

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Advances in digital technology in the last decade have motivated the development of very efficient and high quality speech compression algorithms. While in the early low bit rate coding systems, the main target was production of intelligible speech at low information rates, expansion of new applications such as mobile satellite systems increased the demand for high quality speech at lowest possible bit rates. This resulted in the development of efficient parametric models for speech production system. These models were the basis of powerful speech compression algorithms such as CELP and Multib

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13

Khan, Mohammad M. A. "Coding of excitation signals in a waveform interpolation speech coder." Thesis, McGill University, 2001. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=32961.

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The goal of this thesis is to improve the quality of the Waveform Interpolation (WI) coded speech at 4.25 kbps. The quality improvement is focused on the efficient coding scheme of voiced speech segments, while keeping the basic coding format intact. In the WI paradigm voiced speech is modelled as a concatenation of the Slowly Evolving pitch-cycle Waveforms (SEW). Vector quantization is the optimal approach to encode the SEW magnitude at low bit rates, but its complexity imposes a formidable barrier.<br>Product code vector quantizers (PC-VQ) are a family of structured VQs that circumvent the c

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14

Choi, Hung Bun. "Pitch synchronous waveform interpolation for very low bit rate speech coding." Thesis, University of Liverpool, 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.243264.

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15

Pollard, Matthew Peter. "Waveform interpolation methods for pitch and time-scale modification of speech." Thesis, University of Liverpool, 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.263905.

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16

Smith, Lloyd A. (Lloyd Allen). "Speech Recognition Using a Synthesized Codebook." Thesis, University of North Texas, 1988. https://digital.library.unt.edu/ark:/67531/metadc332203/.

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Speech sounds generated by a simple waveform synthesizer were used to create a vector quantization codebook for use in speech recognition. Recognition was tested over the TI-20 isolated word data base using a conventional DTW matching algorithm. Input speech was band limited to 300 - 3300 Hz, then passed through the Scott Instruments Corp. Coretechs process, implemented on a VET3 speech terminal, to create the speech representation for matching. Synthesized sounds were processed in software by a VET3 signal processing emulation program. Emulation and recognition were performed on a DEC VAX 11/

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17

Ghaidan, Khaldoon A. "A study of the application of modern techniques to speech waveform analysis." Thesis, Loughborough University, 1986. https://dspace.lboro.ac.uk/2134/28015.

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Spectrograms are perhaps the most commonly used method for studying the characteristics of speech waveforms. Producing a spectrogram can conveniently be divided into two parts, the analysis and the display, and this thesis describes a study of both these aspects.

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18

Bliūdžius, Mindaugas. "Skaitmeninių kalbos įrašų glaudinimo metodai." Master's thesis, Lithuanian Academic Libraries Network (LABT), 2004. http://vddb.library.lt/obj/LT-eLABa-0001:E.02~2004~D_20040529_122424-17577.

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The past three decades has witnessed substantial progress towards the application of low-rate speech coders to civilian and military communications as well as computer-related voice applications. Central to this progress has been the development of new speech coders capable of producing high-quality speech at low data rates. Most of these coders incorporate mechanisms to: represent the spectral properties of speech, provide for speech waveform matching, and "optimize" the coder's performance for the human ear. A number of these coders have already been adopted in national and international cel

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19

Beall, Jeffery C. "Stored waveform adaptive motor control." Thesis, Virginia Tech, 1986. http://hdl.handle.net/10919/45746.

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This study investigates an adaptive control scheme designed to maintain accurate motor speed control in spite of high-frequency periodic variations in load torque, load inertia, and motor parameters. The controller adapts, stores and replays a schedule of torques to be delivered at discrete points throughout the periodic load cycle. The controller also adapts to non-periodic changes in load conditions which occur over several load cycles and contains inherent integrator control action to drive speed error to zero. Using computer simulations, the control method was successfully applied to a 3

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20

Pelteku, Altin E. "Development of an electromagnetic glottal waveform sensor for applications in high acoustic noise environments." Link to electronic thesis, 2004. http://www.wpi.edu/Pubs/ETD/Available/etd-0114104-142855/.

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Thesis (M.S.)--Worcester Polytechnic Institute.<br>Keywords: basis functions; perfectly matched layers; PML; neck model; parallel plate resonator; finite element; circulator; glottal waveform; multi-transmission line; dielectric properties of human tissues; radiation currents; weighted residuals; non-acoustic sensor. Includes bibliographical references (p. 104-107).

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21

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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22

Nehl, Albert Henry. "Investigation of techniques for high speed CMOS arbitrary waveform generation." PDXScholar, 1990. https://pdxscholar.library.pdx.edu/open_access_etds/4109.

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Today a growing number of applications in design engineering, production and environmental testing, and system service require specific analog waveforms and digital patterns. Such requirements are neither satisfactorily nor easily met by the use of standard function or single purpose, custom generators. Traditional methods of waveform generation suffer from undesirable complexity or mediocre performance and are otherwise limited. For the majority of arbitrary waveform generation applications, including medical engineering, modal analysis and electronic engineering, direct digital synthesis tec

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23

Joseph, Andrew Paul. "Assessing the effects of GMAW-Pulse parameters on arc power and weld heat input /." Connect to resource, 2001. http://rave.ohiolink.edu/etdc/view?acc%5Fnum=osu1130521651.

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24

Elangovan, Saravanan, Jerry L. Cranford, Letitia Walker, and Andrew Stuart. "A Comparison of the Mismatch Negativity and a Differential Waveform Response." Digital Commons @ East Tennessee State University, 2005. https://dc.etsu.edu/etsu-works/1556.

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A mismatch negativity response (MMN) and a new differential waveform were derived in an effort to evaluate a neural refractory or recovery effect in adult listeners. The MMN was elicited using oddball test runs in which the standard and deviant stimuli differed in frequency. To derive the differential waveform, the same standard and deviant stimuli were presented alone. MMN responses were obtained by subtracting the averaged responses to standards from the deviants. The differential waveforms were obtained by subtracting the averaged responses to standards presented alone from deviants present

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25

Torres, Juan Félix. "Estimation of glottal source features from the spectral envelope of the acoustic speech signal." Diss., Georgia Institute of Technology, 2010. http://hdl.handle.net/1853/34736.

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Speech communication encompasses diverse types of information, including phonetics, affective state, voice quality, and speaker identity. From a speech production standpoint, the acoustic speech signal can be mainly divided into glottal source and vocal tract components, which play distinct roles in rendering the various types of information it contains. Most deployed speech analysis systems, however, do not explicitly represent these two components as distinct entities, as their joint estimation from the acoustic speech signal becomes an ill-defined blind deconvolution problem. Nevertheless,

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26

Liu, Lup Shun Nelson Carleton University Dissertation Engineering Electronics. "Sensitivity analysis and optimization of high-speed VLSI interconnects using asymptotic waveform evaluation." Ottawa, 1993.

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27

Bhatta, Debesh. "Algorithms and methodology for incoherent undersampling based acquisition of high speed signal waveforms using low cost test instrumentation." Diss., Georgia Institute of Technology, 2014. http://hdl.handle.net/1853/54262.

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The objective of this research is to develop and demonstrate low-complexity, robust, frequency-scalable, wide-band waveform acquisition techniques for testing high speed com- munication systems. High resolution waveform capture is a versatile testing tool that enables flexible test strategies. However, waveform capture at high data rates requires costly hardware because the increased bandwidth of the signal waveform leads to an increase in the sampling rate requirement, cost of front-end components, and sensitivity to phase errors in traditional (source) synchronous Nyquist-rate tester archite

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28

Tsutsumi, Monike. "Avaliação da videolaringoscopia de alta velocidade de sujeitos normais." Universidade de São Paulo, 2015. http://www.teses.usp.br/teses/disponiveis/82/82131/tde-28032016-142207/.

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Estudos utilizando imagens laríngeas de sujeitos normais captadas por videolaringoscopia de alta velocidade revelam o uso de diversas ferramentas e métricas em diferentes populações. No entanto, é evidente a escassez de normatizações operacionais e de parâmetros vocais de referência. Os objetivos desse estudo foram obter parâmetros da dinâmica vocal utilizando ferramentas computacionais de uso corrente pelo Grupo de Pesquisa em Engenharia Médica (GPEM - CNPq) e caracterizar o padrão vibratório das pregas vocais de sujeitos normais por meio das ondas da área glótica e quimografia de alta veloci

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29

Schulz, Yvonne Katrin [Verfasser], Stefan [Akademischer Betreuer] Kniesburges, and Stefan [Gutachter] Kniesburges. "Parameter analysis of the Glottal Area Waveform based on high-speed recordings within a synthetic larynx model / Yvonne Katrin Schulz ; Gutachter: Stefan Kniesburges ; Betreuer: Stefan Kniesburges." Erlangen : Friedrich-Alexander-Universität Erlangen-Nürnberg (FAU), 2020. http://d-nb.info/120337769X/34.

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30

Baker, Katherine Louise. "Cognitive Evoked Auditory Potentials and Neuropsychological Measures Following Concussion in College Athletes." Miami University / OhioLINK, 2008. http://rave.ohiolink.edu/etdc/view?acc_num=miami1209744334.

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31

Narayanan, G. "Synchronised Pulsewidth Modulation Strategies Based On Space Vector Approach For Induction Motor Drives." Thesis, Indian Institute of Science, 1999. http://hdl.handle.net/2005/139.

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In high power induction motor drives, the switching frequency of the inverter is quite low due to the high losses in the power devices. Real-time PWM strategies, which result in reduced harmonic distortion under low switching frequencies and have maximum possible DC bus utilisation, are developed for such drives in the present work. The space vector approach is taken up for the generation of synchronised PWM waveforms with 3-Phase Symmetry, Half Wave Symmetry and Quarter Wave Symmetry, required for high-power drives. Rules for synchronisation and the waveform symmetries are brought out. Thes

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32

LIN, YAN-JUN, and 林妍君. "Speech waveform coding based on vector quantization." Thesis, 1986. http://ndltd.ncl.edu.tw/handle/88277127723707099722.

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33

Biskup, John Fredrick. "Applications of the speedy delivery waveform." Thesis, 2003. http://hdl.handle.net/2152/29809.

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The Speedy Delivery (SD) waveform was introduced in patent US 6,441,695 B1 issued August 27, 2002 to the inventor Dr. Robert Flake. In the most basic form, the SD boundary condition is an exponential, D⋅e [superscript α⋅t] . The propagating waveform is described by an analytic, closed form solution of the wave equation in lossy media and has several very special properties. The most surprising property is that the leading edge of the waveform propagates with attenuation but without distortion. The lack of distortion occurs even in lossy transmission media with frequency dependent parameters. T

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34

"Unit selection and waveform concatenation strategies in Cantonese text-to-speech." 2005. http://library.cuhk.edu.hk/record=b5892349.

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Oey Sai Lok.<br>Thesis (M.Phil.)--Chinese University of Hong Kong, 2005.<br>Includes bibliographical references.<br>Abstracts in English and Chinese.<br>Chapter 1. --- Introduction --- p.1<br>Chapter 1.1 --- An overview of Text-to-Speech technology --- p.2<br>Chapter 1.1.1 --- Text processing --- p.2<br>Chapter 1.1.2 --- Acoustic synthesis --- p.3<br>Chapter 1.1.3 --- Prosody modification --- p.4<br>Chapter 1.2 --- Trends in Text-to-Speech technologies --- p.5<br>Chapter 1.3 --- Objectives of this thesis --- p.7<br>Chapter 1.4 --- Outline of the thesis --- p.9<br>References --- p.11<b

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35

Xu, Wen-Long, and 許文龍. "A Mandarin Speech Synthesizer Using Time Proportionated Interpolation of Pitch Waveform." Thesis, 1996. http://ndltd.ncl.edu.tw/handle/70269225450402799343.

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Hsu, Wen-Lung, and 許文龍. "A Mandarin Speech Synthesizer Using Time Proportionated Interpolation of Pitch Waveform." Thesis, 1996. http://ndltd.ncl.edu.tw/handle/64478207895251783493.

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碩士<br>國立臺灣科技大學<br>電機工程研究所<br>84<br>In this thesis, a text-to-speech system is designed and implemented on MS-Windows operating system. The 408 first-tone Mandarin syllables are adopted as the synthesis units. For the synthesis of syllable-signal, a time-domain processing method called "Time Proportionated Interpolation of Pitch Waveform (TPIPW)" is proposed. About the prosodic processing unit, a rule-based method proposed by other researchers is adopted and slightly modified here. In our me

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37

Yan, Jyh-Horng, and 顏誌宏. "The Research of Voice Waveform Reconstruction in CELP Speech Coding due to the Speech Frame''s Codewords Loss." Thesis, 1999. http://ndltd.ncl.edu.tw/handle/77877098696811301575.

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碩士<br>國立臺北科技大學<br>電腦通訊與控制研究所<br>87<br>This thesis is mainly to explore the best ways to do the voice reconstruction when the transmitted speech signal in speech coding lost some frame’s codewords. In this thesis, we focus on speech coding method based on code excited linear prediction(CELP), and use the high quality low bit-rate speech vocoder -- FS1016 4.8 kbps CELP as the speech coding methods to compare the performance in the voice reconstruction under loss of frame’s codewords Considering the factors for speeding and simplicity, we use the lost frame’s codewords reconstruction as the major

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38

Huang, Tzu-Yun, and 黃姿云. "A Dual Complementary Acoustic Embedding Network: Mining Discriminative Characteristics from Raw-waveform for Speech Emotion Recognition." Thesis, 2019. http://ndltd.ncl.edu.tw/handle/y8zcm7.

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Chandravadan, Vora Santoshkumar. "Novel Methods For Estimation Of Static Nonlinearity Of High-Speed High-Resolution Waveform Digitizers." Thesis, 2009. http://hdl.handle.net/2005/1019.

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Analog-to-digital converter (ADC) is the main workhorse in a digital waveform recorder. Strictly speaking, an ADC is supposed to perform uniformly, irrespective of the characteristics of the signal to be acquired. However, because of certain hardware related inconsistencies, its performance declines, particularly, when acquiring non-repetitive, fast-rising, high frequency signals. The error and distortion contributed due to its declining performance, for the entire range of signals, can be comprehensively characterized by the static and dynamic nonlinearities. Actual testing of ADCs is the onl

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40

Chang, Lung-Yin, and 張龍吟. "Research on speed and quality of anodic bonding using applied voltage with various waveforms." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/s656s6.

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碩士<br>國立臺灣師範大學<br>工業教育學系<br>96<br>Anodic bonding technique is important and is often used in package of MEMS components. It uses ionic bond to obtain bonding results. Surface level of silicon and glass need to be very serious, and the technique belong to non-media bonding. There are important factors of bonding ratio and quality in bonding process, such as applied voltage, temperature, and type of electrode etc. In different forms of applied voltage, it causes different bonding results. The reason is described as follow: the maximum of bonding current by using constant voltage waveform would b

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Zheng, Yu-Xiang, and 鄭宇翔. "Analysis of Time-domain Waveform for Stub Effect in High-Speed Digital Circuits." Thesis, 2018. http://ndltd.ncl.edu.tw/handle/p7u4d6.

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碩士<br>中原大學<br>電子工程研究所<br>106<br>This paper investigates how via stubs effect the time-domain transmission (TDT) waveform, Time-Domain Reflectometry (TDR) waveform, S21, S11, and eye diagram in high speed circuit. Due to the limitations of the laboratory process, the via stub was changed to open stub. In order to understand the effect on the waveform, various parameters of the open stub are analysed. The manner in which the time-domain reflection noise affects the TDR waveform and the step voltage on the TDT waveform is investigated by lattice diagram. And in special cases, waveform overlays ar

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"low bit rate speech coder based on waveform interpolation =: 基於波形預測方法的低比特率語音編碼". 1999. http://library.cuhk.edu.hk/record=b5889941.

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by Ge Gao.<br>Thesis (M.Phil.)--Chinese University of Hong Kong, 1999.<br>Includes bibliographical references (leaves 101-107).<br>Text in English; abstracts in English and Chinese.<br>by Ge Gao.<br>Chapter 1 --- Introduction --- p.1<br>Chapter 1.1 --- Attributes of speech coders --- p.1<br>Chapter 1.1.1 --- Bit rate --- p.2<br>Chapter 1.1.2 --- Speech quality --- p.3<br>Chapter 1.1.3 --- Complexity --- p.3<br>Chapter 1.1.4 --- Delay --- p.4<br>Chapter 1.1.5 --- Channel-error sensitivity --- p.4<br>Chapter 1.2 --- Development of speech coding techniques --- p.5<br>Chapter 1.3 --- Motiv

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