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Dissertations / Theses on the topic 'Temporal Representation in speech'

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Published: 4 June 2021

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1

Davies, David Richard Llewellyn, and dave davies@canberra edu au. "Representing Time in Automated Speech Recognition." The Australian National University. Research School of Information Sciences and Engineering, 2003. http://thesis.anu.edu.au./public/adt-ANU20040602.163031.

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This thesis explores the treatment of temporal information in Automated Speech Recognition. It reviews the study of time in speech perception and concludes that while some temporal information in the speech signal is of crucial value in the speech decoding process not all temporal information is relevant to decoding. We then review the representation of temporal information in the main automated recognition techniques: Hidden Markov Models and Artificial Neural Networks. We find that both techniques have difficulty representing the type of temporal information that is phonetically or phonologically significant in the speech signal. In an attempt to improve this situation we explore the problem of representation of temporal information in the acoustic vectors commonly used to encode the speech acoustic signal in the front-ends of speech recognition systems. We attempt, where possible, to let the signal provide the temporal structure rather than imposing a fixed, clock-based timing framework. We develop a novel acoustic temporal parameter (the Parameter Similarity Length), a measure of temporal stability, that is tested against the time derivatives of acoustic parameters conventionally used in acoustic vectors.

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2

Leach, Corinne. "MANIPULATING TEMPORAL COMPONENTS DURING SINGLE-WORD PROCESSING TO FACILITATE ACCESS TO STORED ORTHOGRAPHIC REPRESENTATIONS IN LETTER-BY-LETTER READERS." Master's thesis, Temple University Libraries, 2019. http://cdm16002.contentdm.oclc.org/cdm/ref/collection/p245801coll10/id/574233.

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Public Health
M.A.
This study investigated the benefits of rapid presentation of written words as a treatment strategy to enhance reading speed and accuracy in two participants with acquired alexia who are letter-by-letter readers. Previous studies of pure alexia have shown that when words are rapidly presented, participants can accurately perform lexical decision and category judgment tasks, yet they are unable to read words aloud. These studies suggest that rapid presentation of words could be used as a treatment technique to promote whole-word reading. It was predicted that treatment utilizing rapid presentation (250/500 ms) will increase reading speed and accuracy of both trained and untrained words compared to the words trained in standard presentation (5000 ms). A single-subject ABACA/ACABA multiple baseline treatment design was used. Treatment was provided twice per week for four weeks for both rapid and standard presentation treatment. Each session comprised a spoken-to-written word decision task and semantic category judgment task. Stimuli included 80 trained words divided between the two treatments and 20 untrained controls. Weekly probes to assess reading accuracy were administered after every two treatment sessions. Based on effect sizes, results showed no consistent unambiguous benefit for rapid or standard presentation treatment. However, possible generalization to untrained words due to rapid presentation treatment was observed. Future research is warranted to investigate the effectiveness of rapid presentation treatment in letter-by-letter readers.
Temple University--Theses

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3

Hernandez, Sierra Gabriel. "Métodos de representación y verificación del locutor con independencia del texto." Thesis, Avignon, 2014. http://www.theses.fr/2014AVIG0203/document.

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La reconnaissance automatique du locuteur indépendante du texte est une méthode récente dans le domaine des systèmes biométriques. Le développement de la reconnaissance du locuteur se reflète tout autant dans la participation croissante aux compétitions internationales et dans les progrès en termes de performance relevés dans ces campagnes. Cependant la précision des méthodes reste limitée par la quantité d'information discriminante du locuteur présente dans les représentations informatiques des énoncés vocaux. Cette thèse présente une étude sur ces représentations. Elle identifie deux faiblesses principales. Tout d’abord, les représentations usuelles ignorent les paramètres temporels de la voix pourtant connus pour leur pouvoir discriminant. Par ailleurs, ces représentations reposent sur le paradigme de l’apprentissage statistique et diminuent l’importance d’événements rares dans une population de locuteurs, mais fréquents dans un locuteur donné.Pour répondre à ces verrous, cette thèse propose une nouvelle représentation des énoncés. Celle-ci projette chaque vecteur acoustique dans un large espace binaire intrinsèquement discriminant du locuteur. Une mesure de similitude associée à une représentation globale (vecteurs cumulatifs) est également proposée. L’approche proposée permet ainsi à la fois de représenter des événements rares mais pertinents et de travailler sur des informations temporelles. Cette approche permet de tirer parti des solutions de compensation de la variabilité « session », qui provient de l’ensemble des facteurs indésirables, exploitées dans les approches de type « iVector ». Dans ce domaine, des améliorations aux algorithmes de l’état de l’art ont été proposées.Une solution originale permettant d’exploiter l’information temporelle à l’intérieur de cette représentation binaire a été proposée. La complémentarité des sources d’information a été attestée par un gain en performance relevé grâce à une fusion linéaire des deux types d’information, indépendant et dépendant de la séquence temporelle
Text-independent automatic speaker recognition is a recent method in biometric area. Its increasing interest is reflected both in the increasing participation in international competitions and in the performance progresses. Moreover, the accuracy of the methods is still limited by the quantity of speaker discriminant information contained in the representations of speech utterances. This thesis presents a study on speech representation for speaker recognition systems. It shows firstly two main weaknesses. First, it fails to take into account the temporal behavior of the voice, which is known to contain speaker discriminant information. Secondly, speech events rare in a large population of speakers although very present for a given speaker are hardly taken into account by these approaches, which is contradictory when the goal is to discriminate among speakers.In order to overpass these limitations, we propose in this thesis a new speech representation for speaker recognition. This method represents each acoustic vector in a a binary space which is intrinsically speaker discriminant. A similarity measure associated with a global representation (cumulative vectors) is also proposed. This new speech utterance representation is able to represent infrequent but discriminant events and to work on temporal information. It allows also to take advantage of existing « session » variability compensation approaches (« session » variability represents all the negative variability factors). In this area, we proposed also several improvements to the usual session compensation algorithms. An original solution to deal with the temporal information inside the binary speech representation was also proposed. Thanks to a linear fusion approach between the two sources of information, we demonstrated the complementary nature of the temporal information versus the classical time independent representations
El reconocimiento automático del locutor independiente del texto, es un método dereciente incorporación en los sistemas biométricos. El desarrollo y auge del mismo serefleja en las competencias internacionales, pero aun la eficacia de los métodos de reconocimientose encuentra afectada por la cantidad de información discriminatoria dellocutor que esta presente en las representaciones actuales de las expresiones de voz.En esta tesis se realizó un estudio donde se identificaron dos principales debilidadespresentes en las representaciones actuales del locutor. En primer lugar, no se tiene encuenta el comportamiento temporal de la voz, siendo este un rasgo discriminatorio dellocutor y en segundo lugar los eventos pocos frecuentes dentro de una población delocutores pero frecuentes en un locutor dado, apenas son tenidos en cuenta por estosenfoques, lo cual es contradictorio cuando el objetivo es discriminar los locutores. Motivadopor la solución de estos problemas, se confirmó la redundancia de informaciónexistente en las representaciones actuales y la necesidad de emplear nuevas representacionesde las expresiones de voz. Se propuso un nuevo enfoque con el desarrollo de unmétodo para la obtención de un modelo generador capaz de transformar la representación actual del espacio acústico a una representación en un espacio binario, dondese propuso una medida de similitud asociada con una representación global (vectoracumulativo) que contiene tanto los eventos frecuentes como los pocos frecuentes enuna expresión de voz. Para la compensación de la variabilidad de sesión se incorporóen la matriz de dispersión intra-clase, la información común de la población de locutores,lo que implicó la modificación de tres algoritmos de la literatura que mejoraronsu desempeño respecto a la eficacia en el reconocimiento del locutor, tanto utilizandoel nuevo enfoque propuesto como el enfoque actual de referencia. La información temporalexistente en las expresiones de voz fue capturada e incorporada en una nuevarepresentación, mejorando aun más la eficacia del enfoque propuesto. Finalmente sepropuso y evaluó una fusión lineal entre los dos enfoques que demostró la informacióncomplementaria existente entre ellos, obteniéndose los mejores resultados de eficaciaen el reconocimiento del locutor

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4

Sun, Felix (Felix W. ). "Speech Representation Models for Speech Synthesis and Multimodal Speech Recognition." Thesis, Massachusetts Institute of Technology, 2016. http://hdl.handle.net/1721.1/106378.

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Thesis: M. Eng., Massachusetts Institute of Technology, Department of Electrical Engineering and Computer Science, 2016.
This electronic version was submitted by the student author. The certified thesis is available in the Institute Archives and Special Collections.
Cataloged from student-submitted PDF version of thesis.
Includes bibliographical references (pages 59-63).
The field of speech recognition has seen steady advances over the last two decades, leading to the accurate, real-time recognition systems available on mobile phones today. In this thesis, I apply speech modeling techniques developed for recognition to two other speech problems: speech synthesis and multimodal speech recognition with images. In both problems, there is a need to learn a relationship between speech sounds and another source of information. For speech synthesis, I show that using a neural network acoustic model results in a synthesizer that is more tolerant of noisy training data than previous work. For multimodal recognition, I show how information from images can be effectively integrated into the recognition search framework, resulting in improved accuracy when image data is available.
by Felix Sun.
M. Eng.

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5

Mansfield, Rachel. "Temporal Abstract Behavioral Representation Model." Honors in the Major Thesis, University of Central Florida, 2007. http://digital.library.ucf.edu/cdm/ref/collection/ETH/id/1181.

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This item is only available in print in the UCF Libraries. If this is your Honors Thesis, you can help us make it available online for use by researchers around the world by following the instructions on the distribution consent form at http://library.ucf.edu/Systems/DigitalInitiatives/DigitalCollections/InternetDistributionConsentAgreementForm.pdf You may also contact the project coordinator, Kerri Bottorff, at kerri.bottorff@ucf.edu for more information.
Bachelors
Engineering and Computer Science
Electrical Engineering

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6

Howard, John Graham. "Temporal aspects of auditory-visual speech and non-speech perception." Thesis, University of Reading, 2001. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.553127.

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This thesis concentrates on the temporal aspects of the auditory-visual integratory perceptual experience described above. It is organized in two parts, a literature review, followed by an experimentation section. After a brief introduction (Chapter One), Chapter Two begins by considering the evolution of the earliest biological structures to exploit information in the acoustic and optic environments. The second part of the chapter proposes that the auditory-visual integratory experience might be a by-product of the earliest emergence of spoken language. Chapter Three focuses on human auditory and visual neural structures. It traces the auditory and visual systems of the modem human brain through the complex neuroanatomical forms that construct their pathways, through to where they finally integrate into the high-level multi-sensory association areas. Chapter Four identifies two distinct investigative schools that have each reported on the auditory-visual integratory experience. We consider their different experimental methodologies and a number of architectural and information processing models that have sought to emulate human sensory, cognitive and perceptual processing, and ask how far they can accommodate a bi-sensory integratory processing. Chapter Five draws upon empirical data to support the importance of the temporal dimension of sensory forms in information processing, especially bimodal processing. It considers the implications of different modalities processing differently discontinuous afferent information within different time-frames. It concludes with a discussion of a number of models of biological clocks that have been proposed as essential temporal regulators of human sensory experience. In Part Two, the experiments are presented. Chapter Six provides the general methodology, and in the following Chapters a series of four experiments is reported upon. The experiments follow a logical sequence, each being built upon information either revealed or confirmed in results previously reported. Experiments One, Three, and Four required a radical reinterpretation of the 'fast-detection' paradigm developed for use in signal detection theory. This enables the work of two discrete investigative schools in auditory-visual processing to be brought together. The use of this modified paradigm within an appropriately designed methodology produces experimental results that speak directly to both the 'speech versus non-speech' debate and also to gender studies.

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7

Payne, Nicole, and Saravanan Elangovan. "Musical Training Influences Temporal Processing of Speech and Non-Speech Contrasts." Digital Commons @ East Tennessee State University, 2012. https://dc.etsu.edu/etsu-works/1565.

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8

Schramm, Cheryl (Cheryl Joanne) Carleton University Dissertation Engineering Electrical. "A temporal representation for multimedia radiological reports." Ottawa, 1989.

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9

Igualada, Pérez Alfonso. "Gesture-speech temporal integration in language development." Doctoral thesis, Universitat Pompeu Fabra, 2017. http://hdl.handle.net/10803/670094.

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In everyday interactions, speakers integrate gestures and speech sounds at a temporal level. One of the linguistic functions of temporally synchronous gesture-speech combinations is to provide prominence to specific parts of a discourse. While a bulk of evidence has explored the gesture-speech co-expressiveness at a semantic level, little is known about the children’s ability to use synchronized gestural and prosodic prominences in the benefit of language. This PhD thesis investigates gesture-speech temporal integration abilities in development and its beneficial impact for children’s language. The dissertation includes three independent studies at different time points in development, each one described in one chapter. The first two studies aim at investigating the role of perceiving gesture-speech temporal synchronizations functioning as markers of prominence, and its linkage to language abilities. First, a study investigated whether three- to- five- year- old children responded better to a word recall task when the word was presented with a contrast of prominence expressed with a synchronous beat gesture (i.e., a hand gesture synchronized with prominence in speech). The results indicated a beneficial local effect of the beat gesture on the recall of the temporally synchronous word. Second, a study examined whether six- to- eight- year-old children processed pragmatic inferences online more rapidly when the relevant information was presented together with a beat gesture. Additionally, this study investigated whether these potential benefits were due to the prominence expressed in the gesture or to its concomitant prosodic prominence. Results showed that children’s processing of a pragmatic inference was improved by both prosodic and beat gesture prominence contributions to the discourse. The last study focused on the predictive role of the first infant’s uses of temporally synchronous gesture-speech combinations on later language development. To do so, a longitudinal study correlated the infants’ production of synchronous pointing gesture-speech combinations during controlled socio-communicative interactions at 12 months with linguistic measures at 18 months. Results demonstrated that synchronous productions positively correlated with lexical and grammatical development at 18 months of age. Overall, the three studies show evidence that infant's synchronous gesture-speech abilities (a) function as multimodal markers of prominence; (b) when perceived in a discourse context synchronies have positive impact on children’s word recall (Study 1) and pragmatic inference resolution (Study 2); and (c) infants’ first productions of synchronous gesture-speech combinations serve a communicative strategy which is correlated to later language abilities (Study 3). The findings of the studies presented in this thesis point out the importance of synchronous gesture-speech combinations in highlighting information, as well as their beneficial effects in language acquisition.

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10

Wlodarczak, Marcin [Verfasser]. "Temporal entrainment in overlapping speech / Marcin Wlodarczak." Bielefeld : Universitätsbibliothek Bielefeld, 2014. http://d-nb.info/1047666359/34.

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11

Warren, P. "The temporal organisation and perception of speech." Thesis, University of Cambridge, 1985. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.355053.

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12

Brown, Stephanie Danielle. "Speech-in-Speech Recognition: Understanding the Effect of Different Talker Maskers." Case Western Reserve University School of Graduate Studies / OhioLINK, 2019. http://rave.ohiolink.edu/etdc/view?acc_num=case1556649651028033.

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13

Mesgarani, Nima. "Discrimination of speech from non-speech based on multiscale spectro-temporal modulations." College Park, Md. : University of Maryland, 2005. http://hdl.handle.net/1903/3044.

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Thesis (M.S.) -- University of Maryland, College Park, 2005.
Thesis research directed by: Dept. of Electrical and Computer Engineering. Title from t.p. of PDF. Includes bibliographical references. Published by UMI Dissertation Services, Ann Arbor, Mich. Also available in paper.

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14

Theron, Karin. "Temporal aspects of speech production in bilingual speakers with neurogenic speech disorders." Diss., Pretoria : [s.n.], 2003. http://upetd.up.ac.za/thesis/available/etd-08072003-152242.

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Payne, N., Saravanan Elangovan, and Jacek Smurzynski. "Auditory Temporal Processing of Speech and Non-speech Contrasts in Specialized Listeners." Digital Commons @ East Tennessee State University, 2012. https://dc.etsu.edu/etsu-works/2216.

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16

Hostetter, Michael. "Analogical representation in temporal, spatial, and mnemonic reasoning." Thesis, This resource online, 1990. http://scholar.lib.vt.edu/theses/available/etd-03242009-040545/.

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17

Hamilton, A. C. J. "Parameters of the analytic vector representation of speech." Thesis, University of Canterbury. Electrical Engineering, 1985. http://hdl.handle.net/10092/5833.

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The instantaneous amplitude (magnitude) and instantaneous frequency (phase derivative) waveforms of analytic vector representations of speech, models of speech and sub-bands of speech are generated and analysed. Important characteristics of the instantaneous parameter waveforms are identified. These are related to attributes of vector loci and. the spectral and zero structure of the associated real and analytic signals. Real time generation and display of the analytic vector and its parameters is achieved and associated difficulties identified. The problem of reconstruction of speech from its instantaneous parameters and from modified instantaneous parameters is addressed. Resulting distortions are classified and an application to low bit rate speech transmission investigated.

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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.
Includes bibliographical references (leaves 95-97).
by Ellen Marie Eide.
Ph.D.

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19

Karnebäck, Stefan. "Spectro-temporal properties of the acoustic speech signal used for speech/music discrimination /." Stockholm : Department of Speech, Music and Hearing, Royal Institute of Technology, 2004. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-501.

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20

Domont, Xavier. "Hierarchical spectro-temporal features for robust speech recognition." Münster Verl.-Haus Monsenstein und Vannerdat, 2009. http://d-nb.info/1001282655/04.

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21

Kleinschmidt, Michael. "Robust speech recognition based on spectro-temporal processing." [S.l. : s.n.], 2002. http://deposit.ddb.de/cgi-bin/dokserv?idn=965610276.

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22

Shatzer, Hannah Elizabeth. "Visual and Temporal Influences on Multimodal Speech Integration." The Ohio State University, 2015. http://rave.ohiolink.edu/etdc/view?acc_num=osu1437403560.

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23

Kirchhuebel, Christin. "The acoustic and temporal characteristics of deceptive speech." Thesis, University of York, 2013. http://etheses.whiterose.ac.uk/4790/.

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Voice stress analysis based technologies, commonly known as Voice Stress Analysers (VSA), which are said to measure peoples’ veracity based on the speech signal, have come under a large amount of scientific scrutiny in recent years. Scientific reliability testing of these products has exclusively resulted in negative evaluations. While testing of these products is a necessary part of their evaluation, it is believed that a more fundamental step has been overlooked. Prior to examining the reliability of a test it should be ascertained whether the assumptions on which the test is based are valid. In other words, whether a relationship exists between deception, truth and speech, and if so, what the nature of this relationship is. Using two empirical studies, the present research explores the viability of using speech analysis as a means of differentiating between deceptive and non-deceptive speech. Experiment 1 takes the form of a laboratory-based study which employs a mock-theft paradigm in conjunction with a ‘security interview’ to elicit baseline/control and deceptive speech from a total of ten male native British English speakers. The data for experiment 2 is taken from audio recordings of the interrogation sessions that were part of a broader research study investigating a number of human deception responses across biological, physiological, psychological and behavioural dimensions. The recordings provided control, truthful and deceptive speech from a total of 37 speakers. The speech samples were analysed on a range of acoustic and temporal parameters. For the majority of acoustic parameters no significant differences or trends can be discerned that would allow for a reliable differentiation between deceptive and non-deceptive speech. The changes that do occur are very minute and not consistent across speakers. The results of the temporal measurements seem more promising with respect to highlighting differences between deceptive and non-deceptive samples. However, it transpired that contextual factors such as type of interview and nature of questions asked have conflicting effects on the temporal aspects of liars’ speech. Moreover, context appears to affect the speech of truth-tellers in similar ways to that of liars. Deceptive behaviour is individualised, very multifaceted and far from being clear cut. The successful development of a method that would reliably and consistently differentiate between truths and lies, based on speech analysis, is highly questionable.

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Haas, B., Jacek Smurzynski, and Marc A. Fagelson. "Temporal Processing in Patients with Tinnitus." Digital Commons @ East Tennessee State University, 2011. https://dc.etsu.edu/etsu-works/1642.

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Parkinson, Jon. "Representation learning with a temporally coherent mixed-representation." Thesis, University of Manchester, 2017. https://www.research.manchester.ac.uk/portal/en/theses/representation-learning-with-a-temporally-coherent-mixedrepresentation(ba48bd9e-80ed-4d37-b743-cb149bc498ee).html.

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Guiding a representation towards capturing temporally coherent aspects present invideo improves object identity encoding. Existing models apply temporal coherenceuniformly over all features based on the assumption that optimal encoding of objectidentity only requires temporally stable components. We test the validity of this assumptionby exploring the effects of applying a mixture of temporally coherent invariantfeatures, alongside variable features, in a single 'mixed' representation. Applyingtemporal coherence to different proportions of the available features, we evaluate arange of models on a supervised object classification task. This series of experimentswas tested on three video datasets, each with a different complexity of object shape andmotion. We also investigated whether a mixed-representation improves the capture ofinformation components associated with object position, alongside object identity, ina single representation. Tests were initially applied using a single layer autoencoderas a test bed, followed by subsequent tests investigating whether similar behaviouroccurred in the more abstract features learned by a deep network. A representationapplying temporal coherence in some fashion produced the best results in all tests,on both single layered and deep networks. The majority of tests favoured a mixed representation,especially in cases where the quantity of labelled data available to thesupervised task was plentiful. This work is the first time a mixed-representation hasbeen investigated, and demonstrates its use as a method for representation learning.

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Nilsson, Mattias. "Entropy and Speech." Doctoral thesis, Stockholm : Sound and Image Processing Laboratory, School of Electrical Engineering, Royal Institute of Technology, 2006. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-3990.

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Viera, Gerardo. "Time in mind : the cognitive science of temporal representation." Thesis, University of British Columbia, 2016. http://hdl.handle.net/2429/59975.

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Philosophers and cognitive scientists have always been interested in how people come to mentally represent time. Surprisingly though, contemporary philosophers have largely neglected the wealth of relevant empirical research coming from neuroscience, computational psychology, zoology and related fields. My dissertation is meant to remedy this neglect by bringing together major strands in the philosophical and empirical literatures on temporal representation in order to show how both fields can mutually benefit one another. Chapter 1 describes what I call the temporal coordination problem and provides the needed philosophical background on mental representation that frames the majority of the thesis. Chapter 2 provides a taxonomy of the general approaches to explaining how animals coordinate their behaviors with the temporal structure of the world around them. Chapter 3 argues that part of the explanation for how animals come to mentally represent time is through the operation of a genuine sense of time centered on the circadian systems that provides animals with information about the approximate time of day. Chapters 4 and 5 argue for what I call the fragmentary model of temporal perception – temporal perception is not a unified capacity but is importantly fragmented. Chapter 4 argues that the fragmentary model undermines the central debate in the philosophical literature over the mirroring constraint. I conclude that there simply is no single story to be told about how the temporal structure of experience itself relates to the temporal content of experience. While chapter 4 emphasizes the fragmentary nature of temporal perception, chapter 5 emphasizes the way in which time appears unified in perception and cognition and proposes an explanation of how this apparent unity comes about. Here I highlight how literature more commonly found in the history and philosophy of science on the unitization of measurement actually informs current understanding of the mind. In particular, I argue that the brain comes to integrate the temporal information encoded in various time keeping devices by unitizing time in a manner that parallels how our cultural time keeping practices have unitized time. Finally, chapter 6 concludes by recapping many of the major conclusions of the thesis.
Arts, Faculty of
Graduate

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Shunmugam, Tamindran. "Adoption of a visual model for temporal database representation." Master's thesis, University of Cape Town, 2016. http://hdl.handle.net/11427/20875.

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Today, in the world of information technology, conceptual model representation of database schemas is challenging for users both in the Software Development Life Cycle (SDLC) and the Human-Computer Interaction (HCI) domain. The primary way to resolve this issue, in both domains, is to use a model that is concise, interpretable and clear to understand, yet encompasses all of the required information to be able to clearly define the database. A temporal database is understood as a database capable of supporting reasoning of time-based data for e.g.: a temporal database can answer questions such as: - for what period was Mrs Jones single before she got married? On the other hand, an atemporal database stores data that is valid today and has no history. In the thesis, I looked at different theoretical temporal visual conceptual models proposed by temporal researchers and aimed, by means of a user-survey consisting of business users, to ascertain towards which models users a preference has. I further asked the users for firstly; whether they prefer textual or graphical representations for the entities, attributes and constraints represented by the visual models, or secondly; whether there is a preference for a specific graphical icon for the temporal entities and lastly; to ascertain if the users show a preference towards a specific theoretical temporal conceptual model. The methodology employed to reach my goal in this thesis, is one of experiments on business users with knowledge enhancements after each experiment. Users were to perform a task, and then based on analysis of the task results, they are taught additional temporal aspects so as improve their knowledge before the next experiment commences. The ultimate aim was to extract a visual conceptual model preference from business users with enhanced knowledge of temporal aspects. This is the first work done in this field and thus will aid researchers in future work, as they will have a temporal conceptual model that promotes effective communication, understandability and interpretability.

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Quick, Donya. "Applications and parameter analysis of temporal chaos game representation." Ann Arbor, Mich. : ProQuest, 2008. http://gateway.proquest.com/openurl?url_ver=Z39.88-2004&rft_val_fmt=info:ofi/fmt:kev:mtx:dissertation&res_dat=xri:pqdiss&rft_dat=xri:pqdiss:1454350.

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Thesis (M.S. in Computer Science)--S.M.U.
Title from PDF title page (viewed Mar. 16, 2009). Source: Masters Abstracts International, Volume: 47-01, page: 0419. Adviser: Margaret H. Dunham. Includes supplementary digital materials in a .zip file available from ProQuest website. Includes bibliographical references.

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Alexopoulos, Kyriakos. "Phase spectral representation for low bit rate speech coding." Thesis, Imperial College London, 2001. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.249314.

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Strange, John. "VOICE AUTHENTICATIONA STUDY OF POLYNOMIAL REPRESENTATION OF SPEECH SIGNALS." Master's thesis, University of Central Florida, 2005. http://digital.library.ucf.edu/cdm/ref/collection/ETD/id/4015.

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A subset of speech recognition is the use of speech recognition techniques for voice authentication. Voice authentication is an alternative security application to the other biometric security measures such as the use of fingerprints or iris scans. Voice authentication has advantages over the other biometric measures in that it can be utilized remotely, via a device like a telephone. However, voice authentication has disadvantages in that the authentication system typically requires a large memory and processing time than do fingerprint or iris scanning systems. Also, voice authentication research has yet to provide an authentication system as reliable as the other biometric measures. Most voice recognition systems use Hidden Markov Models (HMMs) as their basic probabilistic framework. Also, most voice recognition systems use a frame based approach to analyze the voice features. An example of research which has been shown to provide more accurate results is the use of a segment based model. The HMMs impose a requirement that each frame has conditional independence from the next. However, at a fixed frame rate, typically 10 ms., the adjacent feature vectors might span the same phonetic segment and often exhibit smooth dynamics and are highly correlated. The relationship between features of different phonetic segments is much weaker. Therefore, the segment based approach makes fewer conditional independence assumptions which are also violated to a lesser degree than for the frame based approach. Thus, the HMMs using segmental based approaches are more accurate. The speech polynomials (feature vectors) used in the segmental model have been shown to be Chebychev polynomials. Use of the properties of these polynomials has made it possible to reduce the computation time for speech recognition systems. Also, representing the spoken word waveform as a Chebychev polynomial allows for the recognition system to easily extract useful and repeatable features from the waveform allowing for a more accurate identification of the speaker. This thesis describes the segmental approach to speech recognition and addresses in detail the use of Chebychev polynomials in the representation of spoken words, specifically in the area of speaker recognition. .
M.S.
Department of Mathematics
Arts and Sciences
Mathematics

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32

Rusnak, John Joseph 1975. "Pitch period segmentation and spectral representation of speech signals." Thesis, Massachusetts Institute of Technology, 1999. http://hdl.handle.net/1721.1/80641.

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Thesis (M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1999.
Includes bibliographical references (leaves 87-88).
by John Joseph Rusnak, Junior.
M.Eng.

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Lee, Chia-ying (Chia-ying Jackie). "Closed-loop auditory-based representation for robust speech recognition." Thesis, Massachusetts Institute of Technology, 2010. http://hdl.handle.net/1721.1/60176.

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Thesis (S.M.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2010.
Includes bibliographical references (p. 93-96).
A closed-loop auditory based speech feature extraction algorithm is presented to address the problem of unseen noise for robust speech recognition. This closed-loop model is inspired by the possible role of the medial olivocochlear (MOC) efferent system of the human auditory periphery, which has been suggested in [6, 13, 42] to be important for human speech intelligibility in noisy environment. We propose that instead of using a fixed filter bank, the filters used in a feature extraction algorithm should be more flexible to adapt dynamically to different types of background noise. Therefore, in the closed-loop model, a feedback mechanism is designed to regulate the operating points of filters in the filter bank based on the background noise. The model is tested on a dataset created from TIDigits database. In this dataset, five kinds of noise are added to synthesize noisy speech. Compared with the standard MFCC extraction algorithm, the proposed closed-loop form of feature extraction algorithm provides 9.7%, 9.1% and 11.4% absolution word error rate reduction on average for three kinds of filter banks respectively.
by Chia-ying Lee.
S.M.

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34

Stuttle, Matthew Nicholas. "A gaussian mixture model spectral representation for speech recognition." Thesis, University of Cambridge, 2003. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.620077.

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35

Elangovan, Saravanan, Nicole Payne, Jacek Smurzynski, and Marc A. Fagelson. "Musical Training Influences Auditory Temporal Processing." Digital Commons @ East Tennessee State University, 2016. https://dc.etsu.edu/etsu-works/1551.

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Background: A link between musical expertise and auditory temporal processing abilities was examined. Material and methods: Trained musicians (n=13) and non-musicians (n=12) were tested on speech tasks (phonetic identification, speech recognition in noise) and non-speech tasks (temporal gap detection). Results: Results indicated musicians had shorter between-channel gap detection thresholds and sharper phonetic identification functions, suggesting that perceptual reorganization following musical training assists basic temporal auditory processes. Conclusions: In general, our results provide a conceptual advance in understanding how musical training influences speech processing, an ability which, when impaired, can affect speech and reading competency.

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Elangovan, Saravanan, and Andrew Stuart. "Auditory Temporal Processing in the Perception of Voicing." Digital Commons @ East Tennessee State University, 2006. https://dc.etsu.edu/etsu-works/1559.

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37

Mesgarani, Nima. "Representation of speech in the primary auditory cortex and its implications for robust speech processing." College Park, Md.: University of Maryland, 2008. http://hdl.handle.net/1903/8586.

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Thesis (Ph. D.) -- University of Maryland, College Park, 2008.
Thesis research directed by: Dept. of Electrical and Computer Engineering. Title from t.p. of PDF. Includes bibliographical references. Published by UMI Dissertation Services, Ann Arbor, Mich. Also available in paper.

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Moghimi, Amir Reza. "Array-based Spectro-temporal Masking For Automatic Speech Recognition." Research Showcase @ CMU, 2014. http://repository.cmu.edu/dissertations/334.

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Over the years, a variety of array processing techniques have been applied to the problem of enhancing degraded speech to improve automatic speech recognition. In this context, linear beamforming has long been the approach of choice, for reasons including good performance, robustness and analytical simplicity. While various non-linear techniques - typically based to some extent on the study of auditory scene analysis - have also been of interest, they tend to lag behind their linear counterparts in terms of simplicity, scalability and exibility. Nonlinear techniques are also more difficult to analyze and lack the systematic descriptions available in the study of linear beamformers. This work focuses on a class of nonlinear processing, known as time-frequency (T-F) masking - a.k.a. spectro-temporal masking { whose variants comprise a significant portion of the existing techniques. T-F masking is based on accepting or rejecting individual time-frequency cells based on some estimate of local signal quality. Analyses are developed that attempt to mirror the beam patterns used to describe linear processing, leading to a view of T-F masking as "nonlinear beamforming". Two distinct formulations of these "nonlinear beam patterns" are developed, based on different metrics of the algorithms behavior; these formulations are modeled in a variety of scenarios to demonstrate the flexibility of the idea. While these patterns are not quite as simple or all-encompassing as traditional beam patterns in microphone-array processing, they do accurately represent the behavior of masking algorithms in analogous and intuitive ways. In addition to analyzing this class of nonlinear masking algorithm, we also attempt to improve its performance in a variety of ways. Improvements are proposed to the baseline two-channel version of masking, by addressing both the mask estimation and the signal reconstruction stages; the latter more successfully than the former. Furthermore, while these approaches have been shown to outperform linear beamforming in two-sensor arrays, extensions to larger arrays have been few and unsuccessful. We find that combining beamforming and masking is a viable method of bringing the benefits of masking to larger arrays. As a result, a hybrid beamforming-masking approach, called "post-masking", is developed that improves upon the performance of MMSE beamforming (and can be used with any beamforming technique), with the potential for even greater improvement in the future.

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39

Tarr, Eric William. "Processing Perceptually Important Temporal and Spectral Characteristics of Speech." The Ohio State University, 2013. http://rave.ohiolink.edu/etdc/view?acc_num=osu1376913300.

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40

El-Geresy, Baher. "Qualitative representation and reasoning for spatial and spatio-temporal systems." Thesis, University of South Wales, 2004. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.403330.

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41

Lucke, Helmut. "On the representation of temporal data for connectionist word recognition." Thesis, University of Cambridge, 1991. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.239520.

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42

Ciotti, Giovanni. "The representation of Sanskrit speech-sounds : philological and linguistic historiographies." Thesis, University of Cambridge, 2013. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.608079.

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43

Willi, Megan, and Megan Willi. "The Perceptual Significance of a Relative Acoustic Representation of Speech." Diss., The University of Arizona, 2017. http://hdl.handle.net/10150/624495.

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The acoustic signature of a particular speech sound varies according to its surrounding phonetic context. How listeners overcome these context dependent acoustic differences is not well understood. Researchers have attempted to find a one-to-one relationship between speech sounds and their acoustic realizations for over 50 years. More recently, relative acoustic representations have been proposed as a means of addressing the lack of acoustic invariance problem. Unlike absolute acoustic cues that are dependent on the context of a given speech signal, relative acoustic representations consist of abstract acoustic patterns that are context independent. The primary goal of this dissertation was to evaluate the perceptual significance of a relative acoustic representation of speech called a relative formant deflection pattern and formally test the relative formant deflection pattern hypothesis. A series of experiments were used to 1) establish how the relative formant deflection pattern hypothesis is different from other relative acoustic hypotheses of place of articulation perception and 2) explicitly test the ability of the relative formant deflection pattern hypothesis to predict listeners' place of articulation perceptions. Chapter 2 established that the relative formant deflection pattern hypothesis significantly predicted listeners' identifications more often than a locus-equation-inspired hypothesis in three out of four vowel contexts. Chapter 3 validated a novel acoustic measurement capable of extracting relative formant deflection patterns from recorded speech and then used the developed tool to evaluate listeners' perceptions of relative formant deflection patterns in natural, reduced speech. The experiment found that the relative formant deflection pattern hypothesis correctly predicted the dominant relative formant deflection pattern that solicited listeners' %/b/ identifications and %/g/ identifications, but not listeners' %/d/ identifications. Chapter 4 established that in a more controlled experimental setting the relative formant deflection pattern hypothesis accounted for listeners’ phonetic identifications across vowel contexts and speech manipulation conditions (i.e. simulated versus sine-wave speech). Additionally, the final experiment in Chapter 4 established that relative deflection in all three formant frequencies was necessary to fully account for the predictive power of the relative formant deflection pattern hypothesis. Taken together, the research provides insight into the perceptual significance of different relative acoustic representations of speech and proposes that a relative acoustic representation called a relative formant deflection pattern may be a perceptually significant solution to the lack of acoustic invariance problem.

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Wang, Yao Electrical Engineering &amp Telecommunications Faculty of Engineering UNSW. "Single channel speech enhancement based on perceptual temporal masking model." Awarded by:University of New South Wales. Electrical Engineering & Telecommunications, 2007. http://handle.unsw.edu.au/1959.4/40454.

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In most speech communication systems, the presence of background noise causes the quality and intelligibility of speech to degrade, especially when the Signal-to-Noise Ratio (SNR) is low. Numerous speech enhancement techniques have been employed successfully in many applications. However, at low signal-to-noise ratios most of these speech enhancement techniques tend to introduce a perceptually annoying residual noise known as "musical noise". The research presented in this thesis aims to minimize this musical noise and maximize the noise reduction ability of speech enhancement algorithms to improve speech quality in low SNR environments. This thesis proposes two novel speech enhancement algorithms based on Weiner and Kalman filters, and exploit the masking properties of the human auditory system to reduce background noise. The perceptual Wiener filter method uses either temporal or simultaneous masking to adjust the Wiener gain in order to suppress noise below the masking thresholds. The second algorithm involves reshaping the corrupted signal according to the masking threshold in each critical band, followed by Kalman filtering. A comparison of the results from these proposed techniques with those obtained from traditional methods suggests that the proposed algorithms address the problem of noise reduction effectively while decreasing the level of the musical noise. In this thesis, many other existing competitive noise suppression methods have also been discussed and their performance evaluated under different types of noise environments. The performances were evaluated and compared to each other using both objective PESQ measures (ITU-T P.862) and subjective listening tests (ITU-T P.835). The proposed speech enhancement schemes based on the auditory masking model outperformed the other methods that were tested.

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Monteiro, Axel. "Spatial and temporal replication in visual and audiovisual speech recognition." Thesis, University of Nottingham, 2003. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.410421.

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46

Davis, Samantha N. "Assessing Temporal Compensation of Speech due to Delayed Auditory Feedback." Ohio University Honors Tutorial College / OhioLINK, 2017. http://rave.ohiolink.edu/etdc/view?acc_num=ouhonors1492779500766271.

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Todman, Christopher Derek. "The representation of time in data warehouses." Thesis, Open University, 1999. http://oro.open.ac.uk/58004/.

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This thesis researches the problems concerning the specification and implementation of the temporal requirements in data warehouses. The thesis focuses on two areas, firstly, the methods for identifying and capturing the business information needs and associated temporal requirements at the conceptual level and; secondly, methods for classifying and implementing the requirements at the logical level using the relational model. At the conceptual level, eight candidate methodologies were investigated to examine their suitability for the creation of data models that are appropriate for a data warehouse. The methods were evaluated to assess their representation of time, their ability to reflect the dimensional nature of data warehouse models and their simplicity of use. The research found that none of the methods under review fully satisfied the criteria. At the logical level, the research concluded that the methods widely used in current practice result in data structures that are either incapable of answering some very basic questions involving history or that return inaccurate results. Specific proposals are made in three areas. Firstly, a new conceptual model is described that is designed to capture the information requirements for dimensional models and has full support for time. Secondly, a new approach at the logical level is proposed. It provides the data structures that enable the requirements captured in the conceptual model to be implemented, thus enabling the historical questions to be answered simply and accurately. Thirdly, a set of rules is developed to help minimise the inaccuracy caused by time. A guide has been produced that provides practitioners with the tools and instructions on how to implement data warehouses using the methods developed in the thesis.

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Dossoumon, Mafoya. "Class and Gender Representation in Nollywood Movies." Thesis, Southern Illinois University at Edwardsville, 2014. http://pqdtopen.proquest.com/#viewpdf?dispub=1549813.

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This study examines class and gender representations in Nollywood films through textual analysis of a sample of films retrieved from the website of the largest Nollywood streaming service, irokoTV. The study investigates patterns in class and gender representations in terms of similarities in portrayals, instances of stereotypes, and value assumptions in terms of who has power by answering the following questions: (1) What class stereotypes are portrayed in Nollywood films? (2) What gender stereotypes are portrayed in Nollywood films? (3) What hegemonic ideas of power are portrayed in Nollywood films as a result of class and gender representations? The study uses an exposure approach to select a sample of convenience of the top 5 films most attended to by the audience on iROKOtv and relies on close reading and a distancing technique called the "commutation test" to discuss the meaning of class and gender representations in the films. Findings indicate that even when they appear to subvert dominant ideologies, the films still reinforce long established societal norms about the importance of wealth and female gender stereotypes such as submissiveness in domestic households. The tales are often aspirational but the films lack grand ideological narratives to make them relevant to social transformation. These findings support Stuart Hall's Theory of Ideology which allows for a subversive agenda in media texts while retaining the flexibility needed to critique connections between dominant ideologies and social practices and structures.

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Hasselmo, M. E. "The representation and storage of visual information in the temporal lobe." Thesis, University of Oxford, 1987. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.379950.

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Prendergast, Garreth. "The representation of temporal dynamics : psychophysics, neural activity and natural sounds." Thesis, University of York, 2008. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.495906.

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