La modélisation sinusoïdale est une des méthodes les plus largement utilisés paramétriques pour la parole et le traitement des signaux audio. Inspiré par le récemment introduit Modèle aQHM et Modèle aHM, nous la vue d’ensemble de la théorie de l’ adaptation sinusoïdale modélisation et nous proposons un modèle nommé la Modèle eaQHM, qui est un non modèle paramétrique de mesure d’ajuster les amplitudes et les phases instantanées de ses fonctions de base aux caractéristiques variant dans le temps de sous-jacents du signal de parole, ainsi atténuer significativement la dite hypothèse de stationnarité locale. Le eaQHM est montré à surperformer aQHM dans l’analyse et la resynthèse de la parole voisée. Sur la base de la eaQHM , un système hybride d’analyse / synthèse de la parole est présenté (eaQHNM), et aussi d’ une version hybride de l’ aHM (aHNM). En outre, nous présentons la motivation pour une représentation pleine bande de la parole en utilisant le eaQHM, c’est, représentant toutes les parties du discours comme haute résolution des sinusoıdes AM-FM. Les expériences montrent que l’adaptation et la quasi-harmonicité est suffisante pour fournir une qualité de transparence dans la parole non voisée resynthèse. La pleine bande analyse eaQHM et système de synthèse est présenté à côté, ce qui surpasse l’état de l’art des systèmes, hybride ou pleine bande, dans la reconstruction de la parole, offrant une qualité transparente confirmé par des évaluations objectives et subjectives. En ce qui concerne les applications, le eaQHM et l’ aHM sont appliquées sur les modifications de la parole (de temps et pas mise à l’échelle). Les modifications qui en résultent sont de haute qualité, et suivent des règles très simples, par rapport à d’autres systèmes de modification état de l’art. Les résultats montrent que harmonicité est préféré au quasi- harmonicité de modifications de la parole du fait de la simplicité de la représentation intégrée. En outre, la pleine bande eaQHM est appliquée sur le problème de la modélisation des signaux audio, et en particulier d’instrument de musique retentit. Le eaQHM est évaluée et comparée à des systèmes à la pointe de la technologie, et leur est montré surpasser en termes de qualité de resynthèse, représentant avec succès l’attaque , transitoire, et une partie stationnaire d’un son d’instruments de musique. Enfin, une autre application est suggéré, à savoir l’analyse et la classification des discours émouvant. Le eaQHM est appliqué sur l’analyse des discours émouvant, offrant à ses paramètres instantanés comme des caractéristiques qui peuvent être utilisés dans la reconnaissance et la quantification vectorielle à base classification du contenu émotionnel de la parole. Bien que les modèles sinusoidaux sont pas couramment utilisés dans ces tâches, les résultats sont prometteurs. / Sinusoidal Modeling is one of the most widely used parametric methods for speech and audio signal processing. The accurate estimation of sinusoidal parameters (amplitudes, frequencies, and phases) is a critical task for close representation of the analyzed signal. In this thesis, based on recent advances in sinusoidal analysis, we propose high resolution adaptive sinusoidal models for analysis, synthesis, and modifications systems of speech. Our goal is to provide systems that represent speech in a highly accurate and compact way. Inspired by the recently introduced adaptive Quasi-Harmonic Model (aQHM) and adaptive Harmonic Model (aHM), we overview the theory of adaptive Sinusoidal Modeling and we propose a model named the extended adaptive Quasi-Harmonic Model (eaQHM), which is a non-parametric model able to adjust the instantaneous amplitudes and phases of its basis functions to the underlying time-varying characteristics of the speech signal, thus significantly alleviating the so-called local stationarity hypothesis. The eaQHM is shown to outperform aQHM in analysis and resynthesis of voiced speech. Based on the eaQHM, a hybrid analysis/synthesis system of speech is presented (eaQHNM), along with a hybrid version of the aHM (aHNM). Moreover, we present motivation for a full-band representation of speech using the eaQHM, that is, representing all parts of speech as high resolution AM-FM sinusoids. Experiments show that adaptation and quasi-harmonicity is sufficient to provide transparent quality in unvoiced speech resynthesis. The full-band eaQHM analysis and synthesis system is presented next, which outperforms state-of-the-art systems, hybrid or full-band, in speech reconstruction, providing transparent quality confirmed by objective and subjective evaluations. Regarding applications, the eaQHM and the aHM are applied on speech modifications (time and pitch scaling). The resulting modifications are of high quality, and follow very simple rules, compared to other state-of-the-art modification systems. Results show that harmonicity is preferred over quasi-harmonicity in speech modifications due to the embedded simplicity of representation. Moreover, the full-band eaQHM is applied on the problem of modeling audio signals, and specifically of musical instrument sounds. The eaQHM is evaluated and compared to state-of-the-art systems, and is shown to outperform them in terms of resynthesis quality, successfully representing the attack, transient, and stationary part of a musical instrument sound. Finally, another application is suggested, namely the analysis and classification of emotional speech. The eaQHM is applied on the analysis of emotional speech, providing its instantaneous parameters as features that can be used in recognition and Vector-Quantization-based classification of the emotional content of speech. Although the sinusoidal models are not commonly used in such tasks, results are promising.

In this study, the understanding about multimodal perception from previous studies has been used to create a perceptually demanding visual search task inside a game. Also, a subtle multimodal cue was created to be in-directly informative about the visual search target’s location by attracting subjects’ attention. 20 subjects were divided equally among the experiments two conditions, one where the subjects had no access to the multimodal information and one where the subjects did have access to the multimodal information. The multimodal information conveyed to the subjects in this experiment was temporal synchrony between a visual light pulsating and a sound being modulated using level and low-pass filtering. Results showed that the subjects that were given the multimodal information improved more on the search task than the group without multimodal information, but the subjects in the multimodal group also perceived the pace of the task as higher. However, it is unclear exactly how the multimodal cue helped the subjects since the playing subjects did not seem to change their search movement pattern to favor the location of the search target, as was expected. Further, the difficulties and considerations of testing in a game environment is discussed and it is concluded that the gamer population is a very varied group which has big impacts on methodology of in-game experiments. To identify sub-groups, further research could study why different players use different search behaviors.

This two-part study employed an experimental design and interviews to examine the effects of videos on adult English as a Second Language (ESL) student listening comprehension. The purpose of the first part of the study was to compare the effects of video in two pre-listening activity conditions on beginner and upper-beginner adult ESL listening comprehension. The participants were 49 Language Instruction for Newcomers to Canada (LINC) students enrolled in two beginner and two upper-beginner ESL classes The beginner classes were randomly assigned either to the audio only (AO) condition or the audio with written script (AW) condition. The upper-beginner classes were similarly assigned to the two conditions, the AO condition or the AW condition. In the AO condition, the students listened to a tape recording of four pre-listening questions prior to viewing the video. In the AW condition, the students listened to the same tape recording and, in addition, were shown the same questions that were printed on large sheets of paper and held up at the front of the class by the teacher. The same pre-test was administered to all four classes. Twelve exercises including pre-listening questions, videos, and comprehension questions were completed. The beginner classes viewed videos from Learning English in the Community LINC 2 (Cameron et al., 1995), and the upper-beginner classes viewed videos from Learning English in the Community LINC 3 (Cameron et al., 1995). The same post-test was administered to all four classes. Results indicated that the written script in the AW condition significantly improved the listening comprehension ability of beginner and upper-beginner adult ESL students. The second part of the study was aimed at gaining insights into the pre-listening questions, the videos, and their value on teaching listening comprehension. All of the students were individually interviewed. Overall the students were positive about the showing of videos as aids for enhancing listening comprehension, the content of the videos, as well as the questions. Some of the students, however, found some of the instruments difficult to understand. They found that the vocabulary was unfamiliar or the speed of delivery was too fast. This lack of comprehension may have affected the results of Part One of the study. / Education, Faculty of / Language and Literacy Education (LLED), Department of / Graduate

D.Ed. / The aim of this study is to give an exposition of problems experienced by students studying through teletuition and to offer ways in which problems could be minimized or alleviated. From this exposition two sub-aims derive: to determine students' needs (with regard to various aspects of the teletuition experience); to develop a strategy for facilitating students' achievement in teletuition.

Les microphones sont des capteurs qui permettent à nos systèmes électroniques de prendre connaissance de notre environnement acoustique en fournissant un signal électrique représentatif des vibrations de l’air. Ils sont employés dans la plupart des systèmes multimédia, mais aussi dans les appareils auditifs. Dans l’implant auditif, le microphone se substitue à l’oreille humaine capable de détecter des pressions acoustiques variants de quelque μPa à quelques Pa. Les microphones, sont en général accompagnés d’un circuit électronique spécifique qui permet leur exploitation au coeur d’un système hétérogène. Depuis les toutes premières transductions acoustique-électriques, le microphone a été perfectionné avec la mise en oeuvre de nouveau principes de transduction et l’élaboration de circuit de conditionnement plus performants. Dernièrement, l’introduction de la technologie MEMS (Micro Electro Mechanical Systems) a permis de réaliser des microphones extrêmement compacts et peu couteux. Ces travaux de recherches concernent la réalisation d’un circuit électronique dédié à l’exploitation d’un transducteur M&NEMS (Micro & Nano Electro Mechanical Systems) survenant comme une évolution du MEMS. Pour commencer l’étude, le principe de transduction et l’application du microphone sont étudiés. Les circuits existants sont examinés en détail et adaptés au transducteur M&NEMS. Les résultats potentiels sont discutés et situés dans l’application. Dans un second temps, un circuit de conditionnement spécifique est proposé. Les résultats sont présentés puis le circuit électronique dédié est intégré sur silicium. Les performances des blocs fonctionnels intégrés sont mesurées et présentées. / Microphones are sensors which allow gauging acoustic environment through an electric representation of vibrations in the air. They can be found in most multimedia equipment and in hearing aids. In this particular application, microphone substitutes a human ear which is able to sense pressure level of sound ranging from a μPa to few Pa. The read-out circuit of microphones converts physical signal from transducer into electronic signals that can be used in any heterogeneous system involving audio processing. Transducers of microphones have known successive generation of improvement. The latest refinement is related to the emergence of MEMS (Micro Electro Mechanical Systems) technology which is suitable to build compact sensor. This thesis explores the design of a readout-circuit using an innovative M&NEMS (Micro & Nano Electro Mechanical Systems) technology derived from MEMS. The thesis is structured beginning with review of existing circuits for M&NEMS microphone. A comparative study is reported considering the proposed technical specifications using simulations and a prototype was realized using discrete components. In the second phase, an innovative circuit was proposed as an ASIC solution targeting M&NEMS technology developed at CEA-LETI. The performance evaluation and the physical measurements of the proposed ASIC are detailed.

Edge computing is the idea of moving computations away from the cloud andinstead perform them at the edge of the network. The benefits of edge computing arereduced latency, increased integrity, and less strain on networks. Edge AI is the practiceof deploying machine learning algorithms to perform computations on the edge.In this project, a pre-trained model YAMNet is retrained and used to perform audioclassification in real-time to detect gunshots, glass shattering, and speech. The modelis deployed onto the edge device both as a full TensorFlow model and as TensorFlowLite models. Comparing results of accuracy, inference time, and memory allocationfor full TensorFlow and TensorFlow Lite models with and without optimization. Resultsfrom this research were that it was a valid option to use both TensorFlow andTensorFlow Lite but there was a lot of performance to gain by using TensorFlow Litewith little downside.

The main goal of this thesis is design of simple digital audio synthesizer. The synthesis of piano tones are descrribed. Final part is devoted to the basic principle of sound sample processing.

This thesis deals with processing of video content on Android platform. Libraries for video encoding and decoding usable on this platform are described first. One of these libraries - FFmpeg - is used as core to design and implement middleware video-processing library. This library will provide simple but e fective API for multimedia Java applications.

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.

This master’s thesis deals with a modification of existing project which is used as a helpful tool for tracking and measuring digital audio interface I2S. The original design was created by Ing. Martin Stejskal, Polymorphic USB – I2S Interface. Modifications are based on practical one year experience when the device was tested and deals with new requirements for extension. This work describes and justify individual changes of hardware and software part of project.