La vérification automatique du locuteur est une tâche de classification qui vise à confirmer ou infirmer l’identité d’un individu d’après une étude des caractéristiques spécifiques de sa voix. L’intégration de systèmes de vérification du locuteur sur des appareils embarqués impose de respecter deux types de contraintes, liées à cet environnement : – les contraintes matérielles, qui limitent fortement les ressources disponibles en termes de mémoire de stockage et de puissance de calcul disponibles ; – les contraintes ergonomiques, qui limitent la durée et le nombre des sessions d’entraînement ainsi que la durée des sessions de test. En reconnaissance du locuteur, la structure temporelle du signal de parole n’est pas exploitée par les approches état-de-l’art. Nous proposons d’utiliser cette information, à travers l’utilisation de mots de passe personnels, afin de compenser le manque de données d’apprentissage et de test. Une première étude nous a permis d’évaluer l’influence de la dépendance au texte sur l’approche état-de-l’art GMM/UBM (Gaussian Mixture Model/ Universal Background Model). Nous avons montré qu’une contrainte lexicale imposée à cette approche, généralement utilisée pour la reconnaissance du locuteur indépendante du texte, permet de réduire de près de 30% (en relatif) le taux d’erreurs obtenu dans le cas où les imposteurs ne connaissent pas le mot de passe des clients. Dans ce document, nous présentons une architecture acoustique spécifique qui permet d’exploiter à moindre coût la structure temporelle des mots de passe choisis par les clients. Cette architecture hiérarchique à trois niveaux permet une spécialisation progressive des modèles acoustiques. Un modèle générique représente l’ensemble de l’espace acoustique. Chaque locuteur est représenté par une mixture de Gaussiennes qui dérive du modèle du monde générique du premier niveau. Le troisième niveau de notre architecture est formé de modèles de Markov semi-continus (SCHMM), qui permettent de modéliser la structure temporelle des mots de passe tout en intégrant l’information spécifique au locuteur, modélisée par le modèle GMM du deuxième niveau. Chaque état du modèle SCHMM d’un mot de passe est estimé, relativement au modèle indépendant du texte de ce locuteur, par adaptation des paramètres de poids des distributions Gaussiennes de ce GMM. Cette prise en compte de la structure temporelle des mots de passe permet de réduire de 60% le taux d’égales erreurs obtenu lorsque les imposteurs prononcent un énoncé différent du mot de passe des clients. Pour renforcer la modélisation de la structure temporelle des mots de passe, nous proposons d’intégrer une information issue d’un processus externe au sein de notre architecture acoustique hiérarchique. Des points de synchronisation forts, extraits du signal de parole, sont utilisés pour contraindre l’apprentissage des modèles de mots de passe durant la phase d’enrôlement. Les points de synchronisation obtenus lors de la phase de test, selon le même procédé, permettent de contraindre le décodage Viterbi utilisé, afin de faire correspondre la structure de la séquence avec celle du modèle testé. Cette approche a été évaluée sur la base de données audio-vidéo MyIdea grâce à une information issue d’un alignement phonétique. Nous avons montré que l’ajout d’une contrainte de synchronisation au sein de notre approche acoustique permet de dégrader les scores imposteurs et ainsi de diminuer le taux d’égales erreurs de 20% (en relatif) dans le cas où les imposteurs ignorent le mot de passe des clients tout en assurant des performances équivalentes à celles des approches état-de-l’art dans le cas où les imposteurs connaissent les mots de passe. L’usage de la modalité vidéo nous apparaît difficilement conciliable avec la limitation des ressources imposée par le contexte embarqué. Nous avons proposé un traitement simple du flux vidéo, respectant ces contraintes, qui n’a cependant pas permis d’extraire une information pertinente. L’usage d’une modalité supplémentaire permettrait néanmoins d’utiliser les différentes informations structurelles pour déjouer d’éventuelles impostures par play-back. Ce travail ouvre ainsi de nombreuses perspectives, relatives à l’utilisation d’information structurelle dans le cadre de la vérification du locuteur et aux approches de reconnaissance du locuteur assistée par la modalité vidéo / SPEAKER verification aims to validate or invalidate identity of a person by using his/her speech characteristics. Integration of an automatic speaker verification engine on embedded devices has to respect two types of constraint, namely : – limited material resources such as memory and computational power ; – limited speech, both training and test sequences. Current state-of-the-art systems do not take advantage of the temporal structure of speech. We propose to use this information through a user-customised framework, in order to compensate for the short duration speech signals that are common in the given scenario. A preliminary study allows us to evaluate the influence of text-dependency on the state-of-the-art GMM/UBM (Gaussian Mixture Model / Universal Background Model) approach. By constraining this approach, usually dedicated to text-independent speaker recognition, we show that a lexical constraint allows a relative reduction of 30% in error rate when impostors do not know the client password. We introduce a specific acoustic architecture which takes advantage of the temporal structure of speech through a low cost user-customised password framework. This three stage hierarchical architecture allows a layered specialization of the acoustic models. The upper layer, which is a classical UBM, aims to model the general acoustic space. The middle layer contains the text-independent specific characteristics of each speaker. These text-independent speaker models are obtained by a classical GMM/UBM adaptation. The previous text-independent speaker model is used to obtain a left-right Semi-Continuous Hidden Markov Model (SCHMM) with the goal of harnessing the Temporal Structure Information (TSI) of the utterance chosen by the given speaker. This TSI is shown to reduce the error rate by 60% when impostors do not know the client password. In order to reinforce the temporal structure of speech, we propose a new approach for speaker verification. The speech modality is reinforced by additional temporal information. Synchronisation points extracted from an additional process are used to constrain the acoustic decoding. Such an additional modality could be used in order to add different structural information and to thwart impostor attacks such as playback. Thanks to the specific aspects of our system, this aided-decoding shows an acceptable level of complexity. In order to reinforce the relaxed synchronisation between states and frames due to the SCHMM structure of the TSI modelling, we propose to embed an external information during the audio decoding by adding further time-constraints. This information is here labelled external to reflect that it is aimed to come from an independent process. Experiments were performed on the BIOMET part of the MyIdea database by using an external information gathered from an automatic phonetical alignment. We show that adding a synchronisation constraint to our acoustic approach allows to reduce impostor scores and to decrease the error rate from 20% when impostor do not know the client password. In others conditions, when impostors know the passwords, the performance remains similar to the original baseline. The extraction of the synchronisation constraint from a video stream seems difficult to accommodate with embedded limited resources. We proposed a first exploration of the use of a video stream in order to constrain the acoustic process. This simple video processing did not allow us to extract any pertinent information

Orientador: Jose Eduardo Ribeiro de Paiva / Dissertação (mestrado) - Universidade Estadual de Campinas, Instituto de Artes / Made available in DSpace on 2018-08-11T16:40:10Z (GMT). No. of bitstreams: 1 Tonelli_MarcioJose_M.pdf: 9221864 bytes, checksum: ef171c3087236cc6e9b1afc9f845ce22 (MD5) Previous issue date: 2007 / Resumo: Este trabalho tem como objetivo principal discutir o impacto das redes móveis de comunicação nas atividades de criação, produção, divulgação, distribuição e acesso de música digital pelos usuários, considerando-se a tendência predominante no século 21 de criação de conteúdo informacional e de entretenimento por parte dos próprios usuários do ciberespaço. São analisados fenômenos como redes sociais e comunidades virtuais, nomadismo e tribalização, cibercultura e ciberespaço, interatividade, podeasting e a transformação do telefone celular numa nova mídia. Adicionalmente foi apresentado o aplicativo MobiDJ de composição de tones polifônicos, como uma ferramenta alinhada com a filosofia de conteúdo gerado pelo usuário (UGC - User Generated Content) e desenvolvida como um estudo de caso para embasamento deste trabalho. / Abstract:This dissertation has as its main objéetive the diseussion of the impaet of mobile eommunieations networks on the aetivities of ereation, produetion, promotion, distribution and digital musie aeeess by users taking into eonsideration the main trend of the 21 eentury of ereation of information and entertainment eontent by the end-users of eyberspaee themselves. Phenomena sueh as social networks and virtual eommunities, nomadism e tribal groups, eyber eulture and eyberspaee, interaetivity, podeasting and the transformation of eellular phones into a new media are analyzed. In addition the applieation software MobiDJ, an online platform for polyphonie tone eomposition was presented as a tool aligned with the philosophy of eontent generated by the end-user (UGC - User Generated Content) and developed as a ease study as rationale for this dissertation. / Mestrado / Mestre em Multimeios

M.Ed. (Subject Didactics) / It is important that allowances be made for the Mathematics gifted pupil, who is seen as the problem solver of the future. Mathematics gifted pupils, on the average, use half their time to work thoroughly through the average syllabus and achieve 90% plus. The average subject teacher does not always have the necessary time for the gifted pupil. He is thus left to his own devices. The lack of facilities, sufficient qualified teachers, support from like-minded people, stimulating opportunities, resource centres and other stimulating factors add further to the pupils' frustrations. Enrichment of syllabi is seen as one of the most prominent provisional possibilities for the gifted child. The reason for this is found in the fact that the gifted child does not constitute even 5% of the population. They are kept mainly in the mainstream and are in no way identified as a group for any type of special educational need, like acceleration. Enrichment by means of educational computer programmes does not need individual teaching or a faster pace, but an adjustment in the activities within the classroom. Teaching, with the aid of computers, offers an educational aid which offers the opportunity for more effective provision for the gifted child ...

This study investigated the effects of a home-based, audio cassette marriage enrichment course on marital communication and marital adjustment. The marriage enrichment course evaluated in this study consisted of two audio cassette tapes, each containing two sessions of approximately 45 minutes in length, and one work booklet. The course contained exercises emphasizing the development of communication skills, encouragement of self-disclosure, learning of empathy skills, and the setting of personal and mutual goals. The unique aspects of the course were the home-based setting in which the couples completed the program, and the self-enclosed audio cassette nature of the course.

Cette thèse porte sur la classification audio sous contrainte de faible latence. La classification audio est un sujet qui a beaucoup mobilisé les chercheurs depuis plusieurs années. Cependant, on remarque qu’une grande majorité des systèmes de classification ne font pas état de contraintes temporelles : le signal peut être parcouru librement afin de rassembler les informations nécessaires pour la prise de décision (on parle alors d’une classification hors ligne). Or, on se place ici dans un contexte de classification audio pour des applications liées au domaine des télécommunications. Les conditions d’utilisation sont alors plus sévères : les algorithmes fonctionnent en temps réel et l’analyse du signal et le traitement associé se font à la volée, au fur et à mesure que le signal audio est transmis. De fait, l’étape de classification audio doit également répondre aux contraintes du temps réel, ce qui affecte son fonctionnement de plusieurs manières : l’horizon d’observation du signal se voit nécessairement réduit aux instants présents et à quelques éléments passés, et malgré cela, le système doit être fiable et réactif. Dès lors, la première question qui survient est : quelle stratégie de classification peut-on adopter afin de faire face aux exigences du temps réel ? On retrouve dans littérature deux grandes approches permettant de répondre à des contraintes temporelles plus ou moins fortes : la classification à la trame et la classification sur segment. Dans le cadre d’une classification à la trame, la décision est prise en se basant uniquement sur des informations issues de la trame audio courante. La classification sur segment, elle, exploite une information court-terme en utilisant les informations issues de la trame courante et de quelques trames précédentes. La fusion des données se fait via un processus d’intégration temporelle qui consiste à extraire une information pertinente basée sur l’évolution temporelle des descripteurs audio. À partir de là, on peut s’interroger pour savoir quelles sont les limites de ces stratégies de classification ? Une classification à la trame et une classification sur segment peuvent-elles être utilisées quel que soit le contexte ? Est-il possible d’obtenir des performances convenables avec ces deux approches ? Quelle mode de classification permet de produire le meilleur rapport entre performance de classification et réactivité ? Aussi, pour une classification sur segment, le processus d’intégration temporelle repose principalement sur des modélisation statistiques mais serait-il possible de proposer d’autres approches ? L’exploration de ce sujet se fera à travers plusieurs cas d’étude concrets. Tout d’abord, dans le cadre des projets de recherche à Orange Labs, nous avons pu contribuer au développement d’un nouvel algorithme de protection acoustique, visant à supprimer très rapidement des signaux potentiellement dangereux pour l’auditeur. La méthode mise au point, reposant sur la proposition de trois descripteurs audio, montre un taux de détection élevé tout en conservant un taux de fausse alarme très bas, et ce, quelles que soient les conditions d’utilisation. Par la suite, nous nous sommes intéressés plus en détail à l’utilisation de l’intégration temporelle des descripteurs dans un cadre de classification audio faible latence. Pour cela, nous avons proposé et évalué plusieurs méthodologies d’utilisation de l’intégration temporelle permettant d’obtenir le meilleur compromis entre performance globale et réactivité. Enfin, nous proposons une autre manière d’exploiter l’information temporelle des descripteurs. L’approche proposée s’appuie sur l’utilisation des représentations symboliques permettant de capter la structure temporelle des séries de descripteurs. L’idée étant ensuite de rechercher des motifs temporels caractéristiques des différentes classes audio. Les expériences réalisées montrent le potentiel de cette approche. / This thesis focuses on audio classification under low-latency constraints. Audio classification has been widely studied for the past few years, however, a large majority of the existing work presents classification systems that are not subject to temporal constraints : the audio signal can be scanned freely in order to gather the needed information to perform the decision (in that case, we may refer to an offline classification). Here, we consider audio classification in the telecommunication domain. The working conditions are now more severe : algorithms work in real time and the analysis and processing steps are now operated on the fly, as long as the signal is transmitted. Hence, the audio classification step has to meet the real time constraints, which can modify its behaviour in different ways : only the current and the past observations of the signal are available, and, despite this fact the classification system has to remain reliable and reactive. Thus, the first question that occurs is : what strategy for the classification can we adopt in order to tackle the real time constraints ? In the literature, we can find two main approaches : the frame-level classification and the segment-level classification. In the frame-level classification, the decision is performed using only the information extracted from the current audio frame. In the segment-level classification, we exploit a short-term information using data computed from the current and few past frames. The data fusion here is obtained using the process of temporal feature integration which consists of deriving relevant information based on the temporal evolution of the audio features. Based on that, there are several questions that need to be answered. What are the limits of these two classification framework ? Can an frame-level classification and a segment-level be used efficiently for any classification task ? Is it possible to obtain good performance with these approaches ? Which classification framework may lead to the best trade-off between accuracy and reactivity ? Furthermore, for the segment-level classification framework, the temporal feature integration process is mainly based on statistical models, but would it be possible to propose other methods ? Throughout this thesis, we investigate this subject by working on several concrete case studies. First, we contribute to the development of a novel audio algorithm dedicated to audio protection. The purpose of this algorithm is to detect and suppress very quickly potentially dangerous sounds for the listener. Our method, which relies on the proposition of three features, shows high detection rate and low false alarm rate in many use cases. Then, we focus on the temporal feature integration in a low-latency framework. To that end, we propose and evaluate several methodologies for the use temporal integration that lead to a good compromise between performance and reactivity. Finally, we propose a novel approach that exploits the temporal evolution of the features. This approach is based on the use of symbolic representation that can capture the temporal structure of the features. The idea is thus to find temporal patterns that are specific to each audio classes. The experiments performed with this approach show promising results.

M.Ed. / The subject English is compulsory in the researcher's students' field of study. The researcher offers the subject English for the National Teacher's Diploma Technical and the National Higher Diploma : Post-School Education. The students come from many different cultural backgrounds and for many of them English is a second or third language. The researcher has observed that there are social affects with which students have to contend, and has detected a definite need amongst students for more certainty pertaining to communication in distance education. The researcher has also perceived definite distinguishing features amongst cultural groups in her subject field English. A definite need arose to examine problems experienced by students studying through teletuition, such as trying to interact with a distant institution, problems concerning the study of English as second or third language and social effects experienced by students in distance education. After having done theoretical research, the researcher included an empirical questionnaire survey in which she attempted to gain biographical and other background information. Respondents were asked questions which concentrated on methods by which they had learned English, whether they experience an intrusion of their home language on English language performance and whether culture impinges on the acquisition of English. The research also attempted to determine whether students experience demotivation, fear of failure and situational problems, amongst others. In the light of the literature study, the questionnaire survey and interviews, the findings summarily were that students * have a need for more contact with their lecturers * revealed a need for emotional support from relatives and lecturers * are demotivated by the negative tone of comments on assignments * experience situational problems * experience fear of failure * experience an intrusion of their home language on English language performance * experience a cultural intrusion in the study of English * experience too many cross - cul,tural contrasts which have an effect on the understanding of English. If the educator in a distance education institution is aware of the students' needs and problems, it would result in a greater understanding of this didactic responsibility in helping students realize their full potential.

Undersökningens syfte var att jämföra två implementationsmetoder för ljud i spel. Dessa metoder är en lagerbaserad metod och en förrenderad metod. I den lagerbaserade metoden kombineras de olika ljudlager som ljudeffekten består av i realtid av spelmotorn, där den individuellt för varje lager applicerar volym- och filtreringskurvor över distans. I den förrenderade metoden har dessa ljudlager redan i förväg kombinerats ihop i en enda audiofil, d.v.s. att den enda behandlingen som sker i ljudmotorn är att volymen sänks över distans, på hela ljudeffekten. I bakgrunden beskrivs övergripande teorier kring perceptionen av ljud över distans och olika lyssningssätt. Även den lagerbaserade ljudmotorn Sound-O-Matic (Double Trouble Audio, 2016) beskrivs kort. Frågeställningen utvärderades genom en prototyp och en nätenkät. Prototypen bestod av ett spel med de två implementationsmetoderna, uppdelade i två versioner: Version A = Förrenderad metod Version B = Lagerbaserad metod Respondenterna spelade båda versionerna av ljudimplementationen och skulle i vardera Version gissa på avståndet och riktningen av gevärsskott som avfyrades runt dem. Resultaten tyder på att den förrenderade metoden i Version A med fördel kan användas i spel, då majoriteten av respondenterna föredrog den före Version B.

Electroacoustic music is often regarded as not being sufficiently accessible to the general public because of its sound-based abstract quality and the complexity of its language. Live electronic music introduces the figure of the performer as a gestural bodily agent that re-enables our multimodal perception of sound and seems to alleviate the accessibility dilemma. However, live electronic music generally lacks the level of detail found in studio-based fixed media works, and it can hardly be transferred outside the concert hall situation (e.g. as a video recording) without losing most of its fresh, dynamic and unpredictable nature. Recent developments in 3D simulation environments and game audio technologies suggest that alternative approaches to music composition and distribution are possible, presenting an opportunity to address some of these issues. In particular, this Portfolio of Compositions proposes the use of real and virtual space as a new medium for the creation and organisation of sound events via computer-simulated audio-sources. In such a context, the role of the performer is sometimes assumed by the listener itself, through the operation of an interactive-adaptive system, or it is otherwise replaced by a set of automated but flexible procedures. Although all of these works are sonic centric in nature, they often present a visual component that reinforces the multimodal perception of meaningful musical structures, either as real space locations for sonic navigation (locative audio), or live visualisations of physically-informed gestural agents in 3D virtual environments. Consequently, this thesis draws on general game-audio concepts and terminology, such as procedural sound, non-linearity, and generative music; but it also embraces game development tools (game engines) as a new methodological and technological approach to electroacoustic music composition. In such context, space and the real-time generation, control, and manipulation of assets combine to play an important role in broadening the routes of musical expression and the accessibility of the musical language. The portfolio consists of six original compositions. Three of these works–Swirls, Alice - Elegy to the Memory of an Unfortunate Lady, and Alcazabilla–are interactive in nature and they required the creation of custom software solutions (e.g. SonicMaps) in order to deal with open-form musical structures. The last three pieces–Singularity, Apollonian Gasket, and Boids–are based on fractal or emergent behaviour models and algorithms, and they propose a non-interactive linear organisation of sound materials via real-time manipulation of non-conventional 3D virtual instruments. These original instrumental models exhibit strong spatial and kinematic qualities with an abstract and minimal visual representation, resulting in an extremely efficient way to build spatialisation patterns, texture, and musical gesture, while preserving the sonic-centric essence of the pieces.

Interviews with experienced ESL language instructors were held to aid in determining how these instructors choose and design authentic materials and make them authentic to second language learners.

For a long time, the function of the musical score has been to support activity in video games, largely by reinforcing the drama and excitement. Rather than leave the score in the background, this project explores the interaction possibilities of an adaptive video game score using real-time modulation of granular synthesis. This study evaluates a vertically re-orchestrated musical score with elements of the score being played back with granular synthesis. A game level was created where parts of the musical score utilized one granular synthesis stem, the parameters of which were controlled by the player. A user experience study was conducted to evaluate the granular synthesis interaction. The results show a wide array of user responses, opinions, impression and recommendations about how the granular synthesis interaction was musically experienced. Some results show that the granular synthesis stem is regarded as an interactive feature and have a direct relationship to the background music. Other results show that interaction went unnoticed. In most cases, the granular synthesis score was experienced as comparable to a more conventional game score and so, granular synthesis can be seen a new interactive tool for the sounddesigner. The study shows that there is more to be explored regarding musical interactions within games. / <p>For contact with the author or request of videoclips, audio or other resources</p><p>Mail: olliver.andersson@gmail.com</p>