Thesis (M.S.)--Brigham Young University. Dept. of Statistics, 2006. / Includes bibliographical references (p. 45-46).

This study investigates Audio over IP. A stress test was used to see what impact asymmetric latency had on the audio signal in a closed network. The study was constructed into two parts. The first part is the stress test where two AoIP solutions were tested. The two solutions where exposed in two forms of asymmetric latency. First a fixed value was used, next, a custom script was used to simulate changing values of asymmetric latency. The second part of this study involved interviews that where conducted with representatives from the audio industry that are working with audio over IP on a dayto-day usage. The goal for these interviews was to figure out what knowledge the audio industry had about asymmetric latency, if the industry had experienced problems related to latency and what general knowledge the industry has about networks. It was found in the interviews that the limitation in AoIP isn’t the technology in itself but rather missing knowledge with the people that are using the systems.

Notre capacité à intégrer des informations venant de nos différents sens est fondamentale pour produire et réguler les mouvements de notre corps. L’objectif général de cette thèse est d’étudier les effets des déterminants de l’intégration multisensorielle sur nos comportements sensorimoteurs rythmiques. Les effets de l’intégration multisensorielle sur ce type de comportements sont peu connus car peu étudiés. Ces comportements caractérisent pourtant la majorité de nos activités quotidiennes comme marcher, écrire ou encore lors de la pratique d’activités sportives. Jusqu’à présent les processus multisensoriels ont été étudiés principalement dans le cadre de nos capacités de discrimination et de détection, révélant notamment la nécessaire synchronie temporelle entre les modalités pour leur intégration. Les conséquences de cette cohérence temporelle et des mécanismes associés n’ont jamais été testées sur les comportements sensorimoteurs rythmiques. Nous chercherons donc à généraliser leurs effets sur ces comportements. Par ailleurs, la littérature rapporte que les caractéristiques du mouvement modifient le traitement des informations sensorielles et semblent également influencer l’intégration multisensorielle. Nous testerons ainsi l’effet des caractéristiques de stabilité du système sensorimoteur, i.e. stabilité intrinsèque de la marche, sur l’intégration multisensorielle.Les deux contributions de cette thèse sont les suivantes : (1) Les comportements rythmiques obéissent aux mêmes principes que les comportements de discrimination temporelle ou de détection. Nos résultats généralisent les effets de cohérence temporelle et montrent pour la première fois l’effet de bénéfice multisensoriel sur la marche. (2) Nous formulons une hypothèse novatrice de compensation sensorimotrice, qui souligne l’utilisation adaptée des informations multisensorielles par le système sensorimoteur. Ce dernier compense la diminution de stabilité intrinsèque de la marche par une plus grande et/ou meilleure utilisation des informations externes audio-tactiles. / Our ability to merge information coming from several senses is crucial to produce and regulate our body movements. The main objective of this thesis is to study the effects of multisensory integration factors on our sensorimotor rhythmic behaviours. The multisensory integration effects on these behaviours are not well understood, being seldom studied. However, those behaviours characterize most of our daily activities such as walking, writing or doing sports. So far, multisensory processes have essentially been studied with regard to our discrimination and detection skills, highlighting the necessity of a temporal synchrony between senses for their integration. The consequences of this temporal coherence and the associated mechanisms have never been tested on sensorimotor rhythmic behaviours. Thus, we will extend their effects to these behaviours. Besides, literature shows that the movements’ features modify the processing of sensory information and can influence multisensory integration. We will test the effects of the stability of the sensorimotor system, i.e. intrinsic stability of gait, on the multisensory integration.The two main contributions of the thesis are the following ones: (1) Rhythmic behaviours obey the same principles as temporal discrimination and detection behaviours. Our results generalize the effects of temporal coherence and show for the first time a multisensory benefit on gait. (2) We suggest a novel sensorimotor compensatory assumption, highlighting the adaptive use of multisensory information by the sensorimotor system, which compensates the decrease of the intrinsic stability of the gait with a larger and/or better use of external audio-tactile information.

Le travail presenté dans cette thèse adresse le problème de génération des performances expressives audio-visuelles pour les acteurs virtuels. Un acteur virtuel est répresenté par une tête parlante en 3D et une performance audio-visuelle contient les expressions faciales, les mouvements de la tête, la direction du regard et le signal de parole.Si une importante partie de la littérature a été dediée aux émotions, nous explorons ici les comportements expressifs verbaux qui signalent les états mentaux, i.e. "ce que le locuteur sent par rapport à ce qu'il dit". Nous explorons les caractéristiques de ces attitudes dites dramatiques et la manière dont elles sont encodées par des signatures prosodiques spécifiques pour une personne i.e. des motifs spécifiques à l'état mental de trajectoires de paramètres prosodiques audio-visuels. / The work presented in this thesis addresses the problem of generating audio-visual expressive performances for virtual actors. A virtual actor is represented by a 3D talking head and an audio-visual performance refers to facial expressions, head movements, gaze direction and the speech signal.While an important amount of work has been dedicated to emotions, we explore here expressive verbal behaviors that signal mental states, i.e "how speakers feel about what they say". We explore the characteristics of these so-called dramatic attitudes and the way they are encoded with speaker-specific prosodic signatures i.e. mental state-specific patterns of trajectories of audio-visual prosodic parameters.

El presente trabajo de título tuvo como objetivo crear un sistema embebido capaz de transmitir y recibir audio profesional AES3 a través de Internet tolerando eventuales problemas en la red. La razón de realizar este sistema está fundada en el creciente surgimiento de la tecnología Audio sobre IP que permite transmitir y reproducir audio de alta fidelidad usando redes de computadores. Para desarrollar el sistema fue necesario adquirir previamente todos los conocimientos prácticos y teóricos que lo fundamentan, como son los principios acústicos, los sistemas embebidos, el procesamiento digital, las redes de computadores y el audio sobre IP. Una vez obtenidos los conocimientos, el desarrollo del sistema se llevó a cabo usando una metodología en cascada específica para sistemas embebidos. Esta metodología se basó en seis etapas secuenciales; especificación, diseño, implementación, pruebas, integración y validación. En la especificación se definieron todos los procesos que debían efectuarse dentro del sistema para conseguir su funcionalidad. Luego, en la etapa siguiente se diseñó el sistema con todos los bloques que debía tener para implementar los procesos especificados. En la implementación se llevó a cabo la creación, la codificación y la síntesis de los componentes diseñados, los que se fueron probando a medida que se iban desarrollando. Una vez que las pruebas terminaron de forma exitosa los componentes se integraron para formar el sistema embebido final. Posterior a la integración el sistema embebido se validó en su funcionalidad dando por terminado el proceso de desarrollo. La tecnología empleada para desarrollar el dispositivo fue una FPGA a la cual se le monto un procesador embebido. En la misma FPGA se desarrollaron todos los módulos HDL que realizan el procesamiento de audio del sistema. En cuanto al procesador embebido, éste se utilizó junto a un sistema operativo Linux para ejecutar rutinas programadas de difícil implementación en lógica digital. De esta forma el desarrollo de todo el sistema embebido involucró la creación de hardware electrónico, hardware HDL y software. Gracias al sistemático cumplimiento de los objetivos propuestos la creación del sistema embebido finalizó de forma exitosa. Además, dado que el trabajo fue patrocinado por una empresa chilena especializada en radiodifusión, el sistema desarrollado establece las bases de un primer prototipo industrial fabricado en Chile capaz de transmitir y recibir audio profesional AES3 sobre IP.

Mid/side processing techniques are commonly used in the music recording industry to widen the stereo image to create a more enveloping listening experience. Since the gaming industry is now in need of better audio solutions to stay on par with the recent visual advances intechnology; these mid/side techniques could potentially be a useful tool for sound designers to use. In this study, an experiment was conducted where 16 participants were asked to play 4 scenarios with different audio settings meant to enhance envelopment in different ways. After each scenario the participants were asked to rate their preference and perceived envelopment followed by a short survey after all 4 scenarios were completed. The quantitative data showed very little evidence suggesting the mid/side processing to be neither perceived more enveloping nor more preferred than the other versions, except for a group with gamers that played games less than 6 hours per week. The qualitative data on the other hand, showed hints at the mid/side version having envelopment as its defining attribute along with it making the sound design more exciting and making some sounds more powerful. The main problem with the mid/side technique seems to be that it has to exclude in-game spatialization for the widened stereo image to be perceived as enveloping. However, if it is applied on sounds that do not need to be spatialized then it might be able to improve the perceived envelopment of those sounds.

This thesis deals with the design and implementation of multi-platform web application based on HTML5, HTML5 Audio and Web Audio API, designed for elementary school students to develop their musical skills, especially musical hearing. The first part compares current educational programs designed to practise musical hearing and to educate musical notation. The following section develops design of music games and application interface, which is based on requirements, followed by analysis of web technologies for sound playback. Last part describes implementation of application in PHP (Nette Framework) with MySQL database and user testing.

O Sistema Brasileiro de Televisão Digital estabeleceu como padrão de codificação de áudio o algoritmo MPEG-4 Advanced Audio Coding, mais precisamente nos perfis Low Complexity, High Efficiency versão 1 e High Efficiency versão 2. O trabalho apresenta um estudo detalhado sobre o padrão, contendo desde alguns conceitos da psicoacústica como o mascaramento até a metodologia de decodificação do stream codificado, sempre voltado para o mercado do SBTVD. É proposta uma arquitetura em hardware para um decodificador compatível com o padrão MPEG-4 AAC LC. O decodificador é separado em dois grandes blocos mantendo em um deles o banco de filtros, considerado a parte mais custosa em termos de processamento. No bloco restante é realizada a decodificação do espectro, onde ocorre a decodificação dos códigos de Huffman, o segundo ponto crítico do algoritmo em termos de demandas computacionais. Por fim é descrita a implementação da arquitetura proposta em VHDL para prototipação em um FPGA da família Cyclone II da Altera. / MPEG-4 Advanced Audio Coding is the chosen algorithm for the Brazilian Digital Television System (SBTVD), supporting the Low Complexity, High Efficiency version 1 and High Efficiency version 2 profiles. A detailed study of the algorithm is presented, ranging from psychoacoustics concepts like masking to a review of the AAC bitstream decoding process, always keeping in mind the SBTVD. A digital hardware architecture is proposed, in which the algorithm is split in two separate blocks, one of them containing the Filter Bank, considered the most demanding task. The other block is responsible for decoding the coded spectrum, which contains the second most demanding task of the system: the Huffman decoding. In the final part of this work the conversion of the proposed architecture into VHDL modules meant to be prototyped with an Altera Cyclone II FPGA is described.

Dans cette thèse, nous abordons le problème de l’estimation de pose de visage dans le contexte des interactions homme-robot. Nous abordons la résolution de cette tâche à l’aide d’une approche en deux étapes. Tout d’abord en nous inspirant de [Deleforge 15], nous proposons une nouvelle façon d’estimer la pose d’un visage, en apprenant un lien entre deux espaces, l’espace des paramètres de pose et un espace de grande dimension représentant les observations perçues par une caméra. L’apprentissage de ce lien se fait à l’aide d’une approche probabiliste, utilisant un mélange de regressions affines. Par rapport aux méthodes d’estimation de pose de visage déjà existantes, nous incorporons de nouvelles informations à l’espace des paramètres de pose, ces additions sont nécessaires afin de pouvoir prendre en compte la diversité des observations, comme les differents visages et expressions mais aussi lesdécalages entre les positions des visages détectés et leurs positions réelles, cela permet d’avoir une méthode robuste aux conditions réelles. Les évaluations ont montrées que cette méthode permettait d’avoir de meilleurs résultats que les méthodes de regression standard et des résultats similaires aux méthodes de l’état de l’art qui pour certaines utilisent plus d’informations, comme la profondeur, pour estimer la pose. Dans un second temps, nous développons un modèle temporel qui utilise les capacités des traqueurs pour combiner l’information du présent avec celle du passé. Le but à travers cela est de produire une estimation de la pose plus lisse dans le temps, mais aussi de corriger les oscillations entre deux estimations consécutives indépendantes. Le modèle proposé intègre le précédent modèle de régression dans une structure de filtrage de Kalman. Cette extension fait partie de la famille des modèles dynamiques commutatifs et garde tous les avantages du mélange de regressionsaffines précédent. Globalement, le modèle temporel proposé permet d’obtenir des estimations de pose plus précises et plus lisses sur une vidéo. Le modèle dynamique commutatif donne de meilleurs résultats qu’un modèle de suivi utilsant un filtre de Kalman standard. Bien qu’appliqué à l’estimation de pose de visage le modèle presenté dans cette thèse est très général et peut être utilisé pour résoudre d’autres problèmes de régressions et de suivis. / In this thesis, we address the well-known problem of head-pose estimationin the context of human-robot interaction (HRI). We accomplish this taskin a two step approach. First, we focus on the estimation of the head pose from visual features. We design features that could represent the face under different orientations and various resolutions in the image. The resulting is a high-dimensional representation of a face from an RGB image. Inspired from [Deleforge 15] we propose to solve the head-pose estimation problem by building a link between the head-pose parameters and the high-dimensional features perceived by a camera. This link is learned using a high-to-low probabilistic regression built using probabilistic mixture of affine transformations. With respect to classic head-pose estimation methods we extend the head-pose parameters by adding some variables to take into account variety in the observations (e.g. misaligned face bounding-box), to obtain a robust method under realistic conditions. Evaluation of the methods shows that our approach achieve better results than classic regression methods and similar results thanstate of the art methods in head pose that use additional cues to estimate the head pose (e.g depth information). Secondly, we propose a temporal model by using tracker ability to combine information from both the present and the past. Our aim through this is to give a smoother estimation output, and to correct oscillations between two consecutives independent observations. The proposed approach embeds the previous regression into a temporal filtering framework. This extention is part of the family of switching dynamic models and keeps all the advantages of the mixture of affine regressions used. Overall the proposed tracker gives a more accurate and smoother estimation of the head pose on a video sequence. In addition, the proposed switching dynamic model gives better results than standard tracking models such as Kalman filter. While being applied to the head-pose estimation problem the methodology presented in this thesis is really general and can be used to solve various regression and tracking problems, e.g. we applied it to the tracking of a sound source in an image.

O Sistema Brasileiro de Televisão Digital estabeleceu como padrão de codificação de áudio o algoritmo MPEG-4 Advanced Audio Coding, mais precisamente nos perfis Low Complexity, High Efficiency versão 1 e High Efficiency versão 2. O trabalho apresenta um estudo detalhado sobre o padrão, contendo desde alguns conceitos da psicoacústica como o mascaramento até a metodologia de decodificação do stream codificado, sempre voltado para o mercado do SBTVD. É proposta uma arquitetura em hardware para um decodificador compatível com o padrão MPEG-4 AAC LC. O decodificador é separado em dois grandes blocos mantendo em um deles o banco de filtros, considerado a parte mais custosa em termos de processamento. No bloco restante é realizada a decodificação do espectro, onde ocorre a decodificação dos códigos de Huffman, o segundo ponto crítico do algoritmo em termos de demandas computacionais. Por fim é descrita a implementação da arquitetura proposta em VHDL para prototipação em um FPGA da família Cyclone II da Altera. / MPEG-4 Advanced Audio Coding is the chosen algorithm for the Brazilian Digital Television System (SBTVD), supporting the Low Complexity, High Efficiency version 1 and High Efficiency version 2 profiles. A detailed study of the algorithm is presented, ranging from psychoacoustics concepts like masking to a review of the AAC bitstream decoding process, always keeping in mind the SBTVD. A digital hardware architecture is proposed, in which the algorithm is split in two separate blocks, one of them containing the Filter Bank, considered the most demanding task. The other block is responsible for decoding the coded spectrum, which contains the second most demanding task of the system: the Huffman decoding. In the final part of this work the conversion of the proposed architecture into VHDL modules meant to be prototyped with an Altera Cyclone II FPGA is described.