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Dissertations / Theses on the topic 'Cepstrum'

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1

Bryan, Robert A. "Thin-bed resolution from cepstrum analysis." Thesis, Virginia Polytechnic Institute and State University, 1985. http://hdl.handle.net/10919/74514.

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A method of cepstrum analysis is developed for the purpose of resolving thin-beds. The method relies on the detection of periodic pulses of the cepstra of reflectivity functions, which are isolated by computing a sub-cepstrum and a sum-cepstrum, and highlighted with a discriminator, where the sub-cepstrum of the functions f₁(t) and f₂(t) is the difference between the cepstra of the two functions, the sum-cepstrum of f₁(t) is the sum of the sub-cepstra of f₁(t) and f<sub>k</sub>(t), k=2,3,4,... , and the discriminator is the product of the sum-cepstrum and the autocovariance of the sum-cepstrum
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2

Gudnason, Jon. "Voice source cepstrum processing for speaker identification." Thesis, Imperial College London, 2007. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.439448.

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3

Lelakis, Ioannis. "Speaker identification using the two-dimensional cepstrum transform." Thesis, Monterey, Calif. : Springfield, Va. : Naval Postgraduate School ; Available from National Technical Information Service, 1995. http://handle.dtic.mil/100.2/ADA294204.

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Thesis (M.S. in Electrical Engineering) Naval Postgraduate School, March 1995.<br>"March 1995." Thesis advisor(s): Monique Fargue, Ralph Hippenstiel. Includes bibliographical references. Also available online.
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4

Komadel, Michal. "Mutlimediální diff - audio dokumenty." Master's thesis, Vysoké učení technické v Brně. Fakulta informačních technologií, 2011. http://www.nusl.cz/ntk/nusl-237074.

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This work describes development of a diff tool working with audio files containing general sound such as music, speech and other sounds. There are presented facts from different domains of science related to sound, such as psychoacoustics, speech recognition and automatic music genre categorisation. This paper also contains description of some diff algorithms and external tools needed for development of the goal application. Moreover, there is introduced design and implementation of the application, settings used for sound features extraction and evaluation of attained results.
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5

Marvi, Hossein. "Efficient feature extraction based on two-dimensional cepstrum analysis for speech recognition." Thesis, University of Surrey, 2004. http://epubs.surrey.ac.uk/843940/.

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Solving speech recognition problems requires an adequate feature extraction technique to transform the raw speech signal to a set of feature vectors to preserve most of information corresponding to the speech signal. The features should ideally be compact, distinct and well representative of the speech signal. If the feature vectors do not represent the important content of the speech, the performance of the system will perform poorly regardless of the pattern recognition techniques applied. Many different feature extraction representations of the speech signal have been suggested and tried fo
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Hubbard, Stephen J. "A cepstrum-based acoustic echo cancellation technique for improving public address system performance." Diss., Georgia Institute of Technology, 1994. http://hdl.handle.net/1853/15617.

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7

Mahajan, Mayur. "Development of a speech recognition system using the Mel Frequency Cepstrum Coefficient method." Thesis, California State University, Long Beach, 2016. http://pqdtopen.proquest.com/#viewpdf?dispub=10141515.

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<p> Voice recognition systems have found widespread use in applications such as tele-shopping, tele-banking, information services, home automation, voice message security, and voice call dialing, which allows a driver to make calls safely while driving. </p><p> This project presents the development of a high performance speech recognition system using human voice models. Recognizing the behavior of the human ear, the Mel Frequency Cepstral Coefficient (MFCC) method is used to develop the system capability for feature extraction. Vector quantization optimized by the Linde Buzo Gray (LGB) algo
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Wei, Coach K. (Coach Kecheng) 1973. "Cepstrum-based deconvolution techniques for ultrasonic pulse-echo imaging of flaws in composite laminates." Thesis, Massachusetts Institute of Technology, 1998. http://hdl.handle.net/1721.1/79992.

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9

CASTELLANA, ANTONELLA. "Towards vocal-behaviour and vocal-health assessment using distributions of acoustic parameters." Doctoral thesis, Politecnico di Torino, 2018. http://hdl.handle.net/11583/2705908.

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Voice disorders at different levels are affecting those professional categories that make use of voice in a sustained way and for prolonged periods of time, the so-called occupational voice users. In-field voice monitoring is needed to investigate voice behaviour and vocal health status during everyday activities and to highlight work-related risk factors. The overall aim of this thesis is to contribute to the identification of tools, procedures and requirements related to the voice acoustic analysis as objective measure to prevent voice disorders, but also to assess them and furnish proof of
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10

Kupka, Petr. "Detekce alkoholu v řečovém signálu." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2021. http://www.nusl.cz/ntk/nusl-442414.

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The diploma thesis Detection of Alcohol in Speech Signal first describes the effect of alcohol on the human body. The second part deals with ways to obtain parameters that describe the speech signal. The third part provides a brief overview of previous case studies and patents focused on the detection of alcohol in the speech signal. The fourth part presents the collected own database of voice recordings and developed software application for the analysis of intoxicated speech. The final part describes the measured changes in speech signal parameters that indicate alcohol intoxication.
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11

Larsson, Alm Kevin. "Automatic Speech Quality Assessment in Unified Communication : A Case Study." Thesis, Linköpings universitet, Programvara och system, 2019. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-159794.

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Speech as a medium for communication has always been important in its ability to convey our ideas, personality and emotions. It is therefore not strange that Quality of Experience (QoE) becomes central to any business relying on voice communication. Using Unified Communication (UC) systems, users can communicate with each other in several ways using many different devices, making QoE an important aspect for such systems. For this thesis, automatic methods for assessing speech quality of the voice calls in Briteback’s UC application is studied, including a comparison of the researched methods.
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Šústek, Martin. "Ukázkový systém na rozpoznávání mluvčích." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2008. http://www.nusl.cz/ntk/nusl-217682.

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My diploma theses deals with the problem of the speaker recognition. The basic theory of this problem is described in the text as well as model and implementation of the system for speaker recognition. The scope of the system is to recognize up to three speakers. The theory is based on calculation parameters for speaker recognition and processing of voice. Program is made in Matlab as a independent application and it has got Czech and English interface.
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13

Jongens, Adrian Wynand Doride. "Application of cepstrum techniques and a guard tube to the measurement of the normal incident sound power absorption coefficient of road surfaces in-situ." Master's thesis, University of Cape Town, 1993. http://hdl.handle.net/11427/9639.

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Includes bibliographical references.<br>The work described in this thesis was directed towards studying signal processing techniques that could best be incorporated in an apparatus that was to measure the plane wave sound power absorption coefficient of road surfaces in-situ. Road traffic noise has been identified as the greatest noise pollutant in the industrialised world with the tyre/road interaction being the major source of noise for traffic speeds in excess of 50 km/hr. Open pore bitumen asphalt material has been found to present a sound absorbing surface that is able to contribute to th
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14

Hammoudeh, Ismail. "Qualitative nichtlineare Zeitreihenanalyse mit Anwendung auf das Problem der Polbewegung." Phd thesis, [S.l. : s.n.], 2002. http://pub.ub.uni-potsdam.de/2003/0003/hammoud.pdf.

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15

Spalt, Taylor B. "Background Noise Reduction in Wind Tunnels using Adaptive Noise Cancellation and Cepstral Echo Removal Techniques for Microphone Array Applications." Thesis, Virginia Tech, 2010. http://hdl.handle.net/10919/34247.

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Two experiments were conducted to investigate Adaptive Noise Cancelling and Cepstrum echo removal post-processing techniques on acoustic data from a linear microphone array in an anechoic chamber. A point source speaker driven with white noise was used as the primary signal. The first experiment included a background speaker to provide interference noise at three different Signal-to-Noise Ratios to simulate noise propagating down a wind tunnel circuit. The second experiment contained only the primary source and the wedges were removed from the floor to simulate reflections found in a wind tunn
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Vodička, Radek. "Rozpoznávání izolovaných slov." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2014. http://www.nusl.cz/ntk/nusl-220658.

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Main purpose of the thesis is to study the processes and methods of isolated words recognition. In the theoretical part a basic principals are explained. The practical part is about the program creating using these principles in practice. For isolated words recognition Hidden Markov Models (HMM) are used, for obtaining decision symptoms cepstral analysis is chosen.
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Sanchez, Fabrício Lopes. "Análise cepstral baseada em diferentes famílias transformada wavelet." Universidade de São Paulo, 2008. http://www.teses.usp.br/teses/disponiveis/82/82131/tde-01092010-113906/.

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Este trabalho apresenta um estudo comparativo entre diferentes famílias de transformada Wavelet aplicadas à análise cepstral de sinais digitais de fala humana, com o objetivo específico de determinar o período de pitch dos mesmos e, ao final, propõe um algoritmo diferencial para realizar tal operação, levando-se em consideração aspectos importantes do ponto de vista computacional, tais como: desempenho, complexidade do algoritmo, plataforma utilizada, dentre outros. São apresentados também, os resultados obtidos através da implementação da nova técnica (baseada na transformada wavelet) em comp
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18

Sannachi, Lakshmanan. "Investigation of anisotropic properties of musculoskeletal tissues by high frequency ultrasound." Doctoral thesis, Humboldt-Universität zu Berlin, Mathematisch-Naturwissenschaftliche Fakultät I, 2012. http://dx.doi.org/10.18452/16476.

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Knochen und Muskel sind die wichtigsten Gewebe im muskuloskelettalen System welche dem Körper die Bewegungen möglich machen. Beide Gewebetypen sind hochgradig strukturierter Extrazellulärmatrix zugrundegelegt, welche die mechanischen und biologischen Funktionen bestimmen. In dieser Studie wurden die räumliche Verteilung der anisotropen elastischen Eigenschaften und der Gewebemineralisation im humanen kortikalen Femur untersucht mit akustischer Mikroskopie und Synchrotron-µCT. Die homogenisierten elastischen Eigenschaften wurden aus einer Kombination der Porosität und der Gewebeelastizitätsma
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Freedman, Joseph Saul. "Using helicopter noise to prevent brownout crashes: an acoustic altimeter." Thesis, Georgia Institute of Technology, 2010. http://hdl.handle.net/1853/34833.

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This thesis explores one possible method of preventing helicopter crashes caused by brownout using the noise generated by the helicopter rotor as an altimeter. The hypothesis under consideration is that the helicopter's height, velocity, and obstacle locations with respect to the helicopter, can be determined by comparing incident and reflected rotor noise signals, provided adequate bandwidth and signal to noise ratio. Heights can be determined by measuring the cepstrum of the reflected helicopter noise. The velocity can be determined by measuring small amounts of Doppler distortion using the
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20

Danielson, Hugo, and Schmuck Benjamin von. "Robot Condition Monitoring : A first step in Condition Monitoring for robotic applications." Thesis, Luleå tekniska universitet, Institutionen för teknikvetenskap och matematik, 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:ltu:diva-66011.

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The industrial world is in constant demand for faster, cheaper and higher quality manufacturing. Robot utilisation and automation has evolved to become a necessary asset to master in order to stay competitive in the global market. With the growing dependency on robots, unexpected downtime and brakedowns can cause devastating loss of revenue. Consequently, this has lead to an increased importance for an accurate condition based way of performing robotic maintenance. As of writing, robots are predominantly maintained through time dependent maintenance. Part replacement is based on statistical mo
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21

Matuštík, Daniel. "Určování základního hlasového tónu." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2013. http://www.nusl.cz/ntk/nusl-219980.

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This diploma thesis with estimation of pitch period of the human voice. The paper listed some of the methods for Estimation of pitch period and method for preprocessing and final processing of the signal after application of functions to determine the frequency of the pitch period in graphical user interface.
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22

Adamec, Michal. "Moderní rozpoznávače řečové aktivity." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2008. http://www.nusl.cz/ntk/nusl-217322.

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This masters theses deals with standard detection methods of speech/pause - voice activity detectors are based on the principles of short-time energy, real spectrum, short-time intensity and on a combinations of these three detectors. In the next parts, there are mentioned other voice activity detectors based on hidden Markovov‘s models and a detector described in the ITU-T G.729 standard. All the detectors, mentioned above, were implemented in research environment MATLAB. Further there was created an user interface for testing functions of the implemented detectors. Finally, there was done an
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23

Smith, Paul Devon. "An Analog Architecture for Auditory Feature Extraction and Recognition." Diss., Georgia Institute of Technology, 2004. http://hdl.handle.net/1853/4839.

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Speech recognition systems have been implemented using a wide range of signal processing techniques including neuromorphic/biological inspired and Digital Signal Processing techniques. Neuromorphic/biologically inspired techniques, such as silicon cochlea models, are based on fairly simple yet highly parallel computation and/or computational units. While the area of digital signal processing (DSP) is based on block transforms and statistical or error minimization methods. Essential to each of these techniques is the first stage of extracting meaningful information from the speech signal, whi
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Pfeifer, Leon. "Automatické rozpoznávání emočních stavů člověka na základě analýzy řečového projevu." Master's thesis, Vysoké učení technické v Brně. Fakulta elektrotechniky a komunikačních technologií, 2008. http://www.nusl.cz/ntk/nusl-217520.

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The diploma thesis deals with the analysis of human emotional states. The thesis consists of three parts. The first part is charcterize, the process of speech generating, from phonetic and psychological poin of view. In the second part there are proccesed metods and contextual things.(preprocessing of signal, voice activity detector). For calculation fundamental Frequency it was used metod of central clipping, another used metod is formant frequency analyse and the last is metod of determinatin of nuber of thorns and planes. In the thirt part there are proccesesed results of measurements perfo
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Cole, David Ross. "Intelligibility enhancement of severely reverberant speech." Thesis, Queensland University of Technology, 1997.

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26

Hassanain, Elham. "Novel cepstral techniques applied to speech synthesis." Thesis, University of Surrey, 2006. http://epubs.surrey.ac.uk/842745/.

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The aim of this research was to develop an improved analysis and synthesis model for utilization in speech synthesis. Conventionally, linear prediction has been used in speech synthesis but is restricted by the requirement of an all-pole, minimum phase model. Here, cepstral homomorphic deconvolution techniques were used to approach the problem, since there are fewer constraints on the model and some evidence in the literature that shows that cepstral homomorphic deconvolution can give improved performance. Specifically the spectral root cepstrum was developed in an attempt to separate the magn
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Green, Richard C. "Walsh based cepstra for speech coding." Thesis, King's College London (University of London), 1991. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.392848.

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Riedißer, Wolfgang. "Phasenorientierte Signalanalyse unter besonderer Berücksichtigung des Cepstrums sowie minimalphasiger Systeme und Signale /." Düsseldorf : VDI-Verl, 1999. http://www.gbv.de/dms/bs/toc/270562567.pdf.

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29

Laporte, Marine. "Contribution à l'amélioration de l’estimation de la profondeur hypocentrale à partir de réseaux régionaux ou globaux." Electronic Thesis or Diss., Université Paris sciences et lettres, 2022. http://www.theses.fr/2022UPSLE055.

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Cette thèse s’intéresse aux différentes méthodes d’estimation de la profondeur hypocentrale ainsi qu’à la mise en évidence des incertitudes de localisation qui peuvent leur être associées. À distance régionale, on utilise ces méthodes pour cartographier les petites variations de profondeur hypocentrales liées à l’activité du grand chevauchement himalayen au Népal ou pour analyser des crises sismiques choisies selon l’actualité et les opportunités rencontrées durant ces trois ans de thèse. Les effets des différentes sources d’incertitudes sur l’estimation de la profondeur hypocentrale sont quan
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Cruz, Luís Manuel Monteiro de Sousa. "Aplicação de técnicas de classificação em séries temporais." Master's thesis, Universidade de Aveiro, 2015. http://hdl.handle.net/10773/15900.

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Mestrado em Engenharia Física<br>Hoje, há um interesse crescente pela aprendizagem de uma segunda língua, quer seja por razões profissionais ou pessoais. Esta é uma tendência que se vai afirmando num mundo cada vez mais interconectado. Por outro lado, a democratização das tecnologias computacionais torna possível pensar em desenvolver novas técnicas de ensino de línguas mais automatizadas e personalizadas. Esta dissertação teve como objetivo estudar e implementar um conjunto de técnicas de processamento de sinal e de classificação de séries temporais úteis para o desenvolvimento de meto
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Sarpal, Sanjeev. "Efficient system identification based on root cepstral deconvolution." Thesis, University of Surrey, 2003. http://epubs.surrey.ac.uk/843337/.

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This thesis summarizes approximately three years of research on signal modelling for the purposes of system identification. Improvements in signal modelling techniques have been encouraged over the years by society's demand for more efficient ways of accessing information. As a consequence, several modelling/compression techniques in both the time domain and the frequency domain have been developed as possible solutions to these problems. Cepstral deconvolution is a frequency domain modelling technique that has been successfully applied to many diverse fields, such as speech and seismic analys
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Bees, Duncan Charles. "Enhancement of acoustically reverberant speech using cepstral methods." Thesis, McGill University, 1990. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=59819.

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Acoustical reverberation has been shown to degrade the intelligibility and naturalness of speech. In this thesis, we discuss the application of cepstral methods to the enhancement of acoustically reverberant speech.<br>We first study previously described cepstral techniques for removal of simple echoes from signals. Our results show that these techniques are not directly applicable to the enhancement of speech of indefinite extent. We next recast these techniques specifically for speech. We propose new segmentation and windowing strategies, in combination with cepstral averaging, to accurately
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Chuang, Ping Derg. "Range estimation by cepstral techniques in image processing." Thesis, Imperial College London, 1986. http://hdl.handle.net/10044/1/37973.

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Busset, Julie. "Inversion acoustique articulatoire à partir de coefficients cepstraux." Phd thesis, Université de Lorraine, 2013. http://tel.archives-ouvertes.fr/tel-00838913.

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L'inversion acoustique-articulatoire de la parole consiste à récupérer la forme du conduit vocal à partir d'un signal de parole. Ce problème est abordé à l'aide d'une méthode d'analyse par synthèse reposant sur un modèle physique de production de la parole contrôlé par un petit nombre de paramètres décrivant la forme du conduit vocal : l'ouverture de la mâchoire, la forme et la position de la langue et la position des lèvres et du larynx. Afin de s'approcher de la géométrie de notre locuteur, le modèle articulatoire est construit à l'aide de contours articulatoires issus d'images cinéradiograp
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Busset, Julie. "Inversion acoustique articulatoire à partir de coefficients cepstraux." Electronic Thesis or Diss., Université de Lorraine, 2013. http://www.theses.fr/2013LORR0027.

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L'inversion acoustique-articulatoire de la parole consiste à récupérer la forme du conduit vocal à partir d'un signal de parole. Ce problème est abordé à l'aide d'une méthode d'analyse par synthèse reposant sur un modèle physique de production de la parole contrôlé par un petit nombre de paramètres décrivant la forme du conduit vocal : l'ouverture de la mâchoire, la forme et la position de la langue et la position des lèvres et du larynx. Afin de s'approcher de la géométrie de notre locuteur, le modèle articulatoire est construit à l'aide de contours articulatoires issus d'images cinéradiograp
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Bekli, Zeid, and William Ouda. "A performance measurement of a Speaker Verification system based on a variance in data collection for Gaussian Mixture Model and Universal Background Model." Thesis, Malmö universitet, Fakulteten för teknik och samhälle (TS), 2018. http://urn.kb.se/resolve?urn=urn:nbn:se:mau:diva-20122.

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Voice recognition has become a more focused and researched field in the last century,and new techniques to identify speech has been introduced. A part of voice recognition isspeaker verification which is divided into Front-end and Back-end. The first componentis the front-end or feature extraction where techniques such as Mel-Frequency CepstrumCoefficients (MFCC) is used to extract the speaker specific features of a speech signal,MFCC is mostly used because it is based on the known variations of the humans ear’scritical frequency bandwidth. The second component is the back-end and handles thes
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Alpan, Ali. "Objective assessment of disordered connected speech." Doctoral thesis, Universite Libre de Bruxelles, 2012. http://hdl.handle.net/2013/ULB-DIPOT:oai:dipot.ulb.ac.be:2013/209758.

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Within the context of the assessment of laryngeal function, acoustic analysis has an important place because the speech signal may be recorded non-invasively and it forms the base on which the perceptual assessment of voice is founded. Given the limitations of perceptual ratings, one has investigated vocal cues of disordered voices that are clinically relevant, summarize properties of speech signals and report on a speaker's phonation in general and voice in particular. Ideally, the acoustic descriptors should also be correlates of auditory-perceptual ratings of voice. Generally speaking, the
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Cédric, Peeters. "Advanced signal processing for the identification and diagnosis of the condition of rotating machinery." Thesis, Lyon, 2019. http://www.theses.fr/2019LYSEI107.

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Cette thèse porte sur des méthodes innovantes de contrôle de l'état de santé des machines tournantes par l’analyse des signaux vibratoires. En effet, la surveillance de l’état de santé des machines contribue à des améliorations substantielles des points de vue économique et de sureté. Afin d’y aboutir, l’une des manières les plus populaires est de recueillir les vibrations de la machine. La plupart de ces vibrations sont directement liées au comportement périodique des sous-systèmes de la machine tels que les arbres de rotation, engrenages, champs électriques rotationnels, etc. Cette connaissa
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Miyajima, C., Y. Nishiwaki, K. Ozawa, T. Wakita, K. Itou, and K. Takeda. "Cepstral Analysis of Driving Behavioral Signals for Driver Identification." IEEE, 2006. http://hdl.handle.net/2237/9596.

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Barreira, Ramiro Roque Antunes. "Modelo mel-cepstral generalizado para envoltória espectral de fala." [s.n.], 2010. http://repositorio.unicamp.br/jspui/handle/REPOSIP/259047.

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Orientadores: Fábio Violaro, Edmilson da Silva Morais<br>Dissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Elétrica e de Computação<br>Made available in DSpace on 2018-08-17T02:12:55Z (GMT). No. of bitstreams: 1 Barreira_RamiroRoqueAntunes_M.pdf: 2303475 bytes, checksum: 72e03fe8e41e9e440f2d4a266666763d (MD5) Previous issue date: 2010<br>Resumo: A análise Mel-Cepstral Generalizada (MGC) corresponde a uma abordagem para estimação de envoltória espectral de fala que unifica as análises LPC, Mel-LPC, Cepstral e Mel-Cepstral. A forma funcional do modelo MGC vari
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Lareau, Jonathan. "Application of shifted delta cepstral features for GMM language identification /." Electronic version of thesis, 2006. https://ritdml.rit.edu/dspace/handle/1850/2686.

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Darch, Jonathan J. A. "Robust acoustic speech feature prediction from Mel frequency cepstral coefficients." Thesis, University of East Anglia, 2008. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.445206.

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43

Nam, Jean Ok. "Signal recovery using cepstral smoothing in the shallow water environment." Thesis, Massachusetts Institute of Technology, 1994. http://hdl.handle.net/1721.1/34093.

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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1994.<br>Includes bibliographical references (leaves 47-48).<br>by Jean Ok Nam.<br>M.Eng.
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Gedeon, Ibrahim Joseph Carleton University Dissertation Engineering Electrical. "Cepstral analysis : a speech processing strategy for the cochlear implant." Ottawa, 1990.

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45

Sklar, Alexander Gabriel. "Channel Modeling Applied to Robust Automatic Speech Recognition." Scholarly Repository, 2007. http://scholarlyrepository.miami.edu/oa_theses/87.

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In automatic speech recognition systems (ASRs), training is a critical phase to the system?s success. Communication media, either analog (such as analog landline phones) or digital (VoIP) distort the speaker?s speech signal often in very complex ways: linear distortion occurs in all channels, either in the magnitude or phase spectrum. Non-linear but time-invariant distortion will always appear in all real systems. In digital systems we also have network effects which will produce packet losses and delays and repeated packets. Finally, one cannot really assert what path a signal will take, and
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46

Aslan, Gokhan. "Cepstral Deconvolution Method For Measurement Of Absorption And Scattering Coefficients Of Materials." Master's thesis, METU, 2007. http://etd.lib.metu.edu.tr/upload/3/12608021/index.pdf.

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Several methods are developed to measure absorption and scattering coefficients of materials. In this study, a new method based on cepstral deconvolution technique is proposed. A reverberation room method standardized recently by ISO (ISO 17497-1) is taken as the reference for measurements. Several measurements were conducted in a physically scaled reverberation room and results are evaluated according to these two methods, namely, the method given in the standard and cepstral deconvolution method. Two methods differ from each other in the estimation of specular parts of room impulse responses
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47

Chen, Jen Kwang, and 陳貞光. "A PRELIMINARY STUDY ON USING CEPSTRUM/MEL-CEPSTRUM FOR SPEECH RECOGNITION THROUGH TELEPHONE SYSTEM." Thesis, 1993. http://ndltd.ncl.edu.tw/handle/09335135352612888419.

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48

Liou, Sin-Jyun, and 劉信君. "A VLSI Design for LPC-Cepstrum." Thesis, 2005. http://ndltd.ncl.edu.tw/handle/43582368844753170445.

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碩士<br>國立暨南國際大學<br>電機工程學系<br>93<br>In digital speech signal processing, the feature coefficients extracted from speech signals plays an important role in either speech recognition or speech compression. Under a well-designed extraction algorithm for the feature coefficients, the speech recognition rate can be highly increased and the performance of the speech compression can also be improved. In this thesis, the implementation of the LPC-Cepstrum algorithm on one single VLSI chip is examined, which can be widely applied in the portable speech recognition systems. The whole LPC-Cepstrum circuit
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ZHUANG, DING-FENG, and 莊鼎豐. "Word recognition based on mel generalized cepstrum." Thesis, 1991. http://ndltd.ncl.edu.tw/handle/61545450640861380805.

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Huang, Chao-Cyun, and 黃超群. "A hybrid audio watermarking technique in cepstrum domain." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/04970192319920609086.

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碩士<br>國立東華大學<br>資訊工程學系<br>97<br>Along with the progress relating to computer hardware and software, the Internet has become the most popular medium for transmitting various forms of digital multimedia. Since the environment of the Internet is open and free browsed, the protection of digital information transmitted on the network has become an important research topic in recent years. Watermarking techniques, therefore, have received much attention. Embedding digital watermark in digital media is an important method for the protection of intellectual property right. In this thesis, we propose
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