Internet applications are becoming more and more flexible to support diverge user demands and network conditions. This is reflected by technical concepts, which provide new adaptation mechanisms to allow fine grained adjustment of the application quality and the corresponding bandwidth requirements. For the case of video streaming, the scalable video codec H.264/SVC allows the flexible adaptation of frame rate, video resolution and image quality with respect to the available network resources. In order to guarantee a good user-perceived quality (Quality of Experience, QoE) it is necessary to adjust and optimize the video quality accurately. But not only have the applications of the current Internet changed. Within network and transport, new technologies evolved during the last years providing a more flexible and efficient usage of data transport and network resources. One of the most promising technologies is Network Virtualization (NV) which is seen as an enabler to overcome the ossification of the Internet stack. It provides means to simultaneously operate multiple logical networks which allow for example application-specific addressing, naming and routing, or their individual resource management. New transport mechanisms like multipath transmission on the network and transport layer aim at an efficient usage of available transport resources. However, the simultaneous transmission of data via heterogeneous transport paths and communication technologies inevitably introduces packet reordering. Additional mechanisms and buffers are required to restore the correct packet order and thus to prevent a disturbance of the data transport. A proper buffer dimensioning as well as the classification of the impact of varying path characteristics like bandwidth and delay require appropriate evaluation methods. Additionally, for a path selection mechanism real time evaluation mechanisms are needed. A better application-network interaction and the corresponding exchange of information enable an efficient adaptation of the application to the network conditions and vice versa. This PhD thesis analyzes a video streaming architecture utilizing multipath transmission and scalable video coding and develops the following optimization possibilities and results: Analysis and dimensioning methods for multipath transmission, quantification of the adaptation possibilities to the current network conditions with respect to the QoE for H.264/SVC, and evaluation and optimization of a future video streaming architecture, which allows a better interaction of application and network. / Die Applikationen im Internet passen sich immer besser an unterschiedliche Anforderungen der Nutzer und variierende Netzwerkbedingungen an. Neue Mechanismen ermöglichen die zielgerichtete Anpassung der Anwendungsqualität und damit der benötigten Bandbreite. Im Falle von Videostreaming ermöglicht der skalierbare Videocodec H.264/SVC, die flexible Veränderung der Bildwiederholungsrate, der Auflösung des Videos und der Bildqualität an die vorhandenen Ressourcen im Netzwerk. Um eine gute vom Nutzer erfahrene Dienstgüte (Quality of Experience, QoE) zu garantieren, muss die Videoqualität richtig angepasst und optimiert werden. Aber nicht nur die Anwendungen des heutigen Internets haben sich verändert. Gerade in den letzten Jahren entstanden neue Netzwerk- und Transporttechnologien, welche eine flexiblere und effizientere Nutzung der Kommunikationsnetze erlauben. Eine dieser Techniken ist die Virtualisierung von Netzwerken. Sie erlaubt es auf einem gemeinsamen physikalischen Netz verschiedene logische Netze zu betreiben, die zum Beispiel Anwendungs-abhängige Adressierung unterstützen, eigene Namensgebung erlauben oder ein individuelles Ressourcen Management ermöglichen. Neuartige Transportmechanismen wie Mehrpfadübertragung auf Netzwerk- und Transportebene des ISO/OSI Stacks streben eine effiziente Ausnutzung der zur Verfügung stehenden Übertragungsmöglichkeiten an. Doch die simultane Übertragung von Daten über heterogene Kommunikationspfade und –technologien führt unausweichlich zu einer Veränderung der Reihenfolge, in der die Pakete ankommen. Es werden zusätzliche Mechanismen und Puffer benötigt, um die ursprüngliche Paketreihenfolge wieder herzustellen und so einen störenden Einfluss auf den Datentransport zu verhindern. Die richtige Dimensionierung dieser Puffer sowie die Klassifizierung des Einflusses von variierenden Pfadparametern wie Bandbreite und Verzögerungen setzen passende Evaluierungsmethoden voraus. Darüber hinaus werden für die Auswahl von geeigneten Pfaden aus einer Menge vorhandener Pfade echtzeitfähige Bewertungsmechanismen benötigt. Eine bessere Interaktion zwischen Applikationen und Netzwerk und der damit verbundene Informationsaustausch ermöglicht die effiziente Anpassung der Applikationsqualität an das Netzwerk und umgekehrt. Diese Doktorarbeit analysiert eine auf Mehrpfadübertragung und skalierbarer Videokodierung basierende Videostreaming Architektur und erarbeitet die folgenden Optimierungsmöglichkeiten und Auswertungen: Analyse- und Dimensionierungsmethoden für Mehrpfadübertragung, Quantifizierung der Anpassungsmöglichkeiten von SVC an das Netzwerk unter Berücksichtigung der QoE und Evaluierung und Optimierung einer zukünftigen Videostreaming Architektur, welche eine stärkere Interaktion zwischen Applikation und Netzwerk ermöglicht.

Le but de cette thèse est de proposer des mécanismes efficaces pour l'ordonnancement des chunks et l'allocation de la bande passante dans le contexte de la transmission vidéo sur les réseaux P2P,afin d'offrir une meilleure qualité de service pour l'utilisateur final. Dans un premier temps nousavons proposé un mécanisme d'ordonnancement des chunks pour la transmission de vidéomulticouche dans les réseaux P2P. Le mécanisme proposé est basé sur une nouvelle technique quipermet de sélectionner les chunks adéquats et les demander des pairs les plus appropriés. Ensuitenous avons proposé un mécanisme d'allocation de la bande passante, toujours dans le cadre detransmission de vidéo multicouche dans les réseaux P2P. Le pair émetteur organise une enchère pour«vendre » sa bande passante. L'allocation tient en considération la priorité des pairs et l'importancedes couches demandées. Finalement nous avons proposé un mécanisme d'adaptation lisse « smooth» d'une vidéo multicouche transportée sur un réseau P2P.Après une introduction, nous présentons dans le chapitre 2 les motivations du travail le but du travailet les problèmes recherche qui demeurent. Dans ce chapitre nous présentons les composants dessystèmes P2P et tout particulièrement la distribution et l'adaptation de contenus. Dans ce cadre,nous proposons une classification des applications de streaming vidéo P2P ainsi que des mécanismesd'allocation de bande passante et d'ordonnancement pour le streaming pair-à-pair. Nous nousintéressons également aux techniques d'adaptation de la qualité en se focalisant plusparticulièrement sur la norme SVC (Scalable Video Coding).Le chapitre 3 propose des mécanismes de priorisation pour la planification de streaming P2P multicouches.Nous proposons une heuristique pour résoudre un problème général d'affectationgénéralisé (Generalized Assignment Problem – GAP). La solution présentée est ensuite adaptée aucas du streaming non multicouches. Les résultats issus des simulations montrent que les solutionsproposées donnent de meilleurs résultats que les solutions traditionnelles.Le chapitre 4 décrit un mécanisme d'allocation dynamique de la bande passante pour les réseaux destreaming P2P multicouches qui se base sur l'allocation d'une bande passante aux pairs tout enassurant un minimum de qualité de service à l'ensemble des pairs. Les bonnes performances desmécanismes proposés, qui sont détaillées à travers l'étude du ratio concernant l'utilisation de labande passante ainsi que du niveau de satisfaction des pairs, montrent que ces derniers permettentd'obtenir une utilisation optimale de la bande passante.Le chapitre 5 porte sur le lissage du streaming multicouches dans les réseaux P2P en se basant sur lesmétriques liées à la variation de la fréquence et de l'amplitude. Les mécanismes proposés ont étéimplémentés dans un banc d'essai réel et l'évaluation des performances montrent l'efficacité desmécanismes pour le lissage du streaming.Dans le chapitre 6 (conclusion and perspectives), nous résumons les contributions proposées danscette thèse ainsi qu’une ouverture sur les travaux futures / Recently we witnessed an increasing demand for scalable deployment of real-time multimediastreaming applications over Internet. In this context, Peer-to-Peer (P2P) networks are playing asignificant role for supporting large-scale and robust distribution of multimedia content to end-users.However, due to peers’ dynamicity, heterogeneity of terminals and access networks, the deploymentof real-time video streaming applications over P2P networks arises lot of challenges. Indeed, animportant issue in P2P overlays is the capacity to self-organize in the face of the dynamic behavior ofpeers in order to ensure content availability and continuity. In addition, the heterogeneity in networks,terminals, and P2P characteristics make the situation more challenging. In this context, layered videostreaming in P2P networks has drawn great interest to overcome these challenges, since it can notonly accommodate large numbers of users, but also handle heterogeneity of peers. However, there isstill a lack of comprehensive studies on video data blocks (chunks) scheduling and bandwidthallocation for the smooth playout in layered streaming over P2P networks.The aim of this thesis is to analyze these concerns and to propose an efficient real-time chunksscheduling and bandwidth allocation mechanisms for QoS provisioning of layered streamingapplications over P2P networks. Our contributions in this thesis are threefold. First, we propose ascheduling mechanism for layered P2P streaming. The proposed mechanism relies on a novelscheduling algorithm that enables each peer to select appropriate stream layers, along withappropriate peers to provide them. The presented mechanism makes efficient use of networkresources and provides high system throughput. Second, we propose a bandwidth allocation modelfor P2 layered streaming systems based on auction mechanisms to optimize the allocation of senderpeers’ uploads bandwidth. The upstream peers organize auctions to “sell” theirs items (links’bandwidth) according to bids submitted by the downstream peers taking into consideration the peerspriorities and the requested layers importance. The ultimate goal is to satisfy the quality levelrequirement for each peer, while reducing the overall streaming cost. Finally, we present a smoothingmechanism for layered streaming in P2P networks. The mechanism aims to reduce the number oflayer changes under varying network conditions, and ensure a smooth playout for the end-user.

"A mobile ultrasound system has been developed, which makes ultrasound examinations possible in harsh environments without reliable power sources, such as ambulances, helicopters, war zones, and disaster sites. The goal of this project was to analyze three different wireless communication technologies that could be integrated into the ultrasound system for possible utilization in remote data applications where medical information may be transmitted from the mobile unit to some centralized base station, such as an emergency room or field hospital. By incorporating wireless telecommunication technology into the design, on site medical personnel can be assisted in diagnostic decisions by remote medical experts. The wireless options that have been tested include the IEEE 802.11g standard, mobile broadband cards on a 3G cellular network, and a mobile satellite terminal. Each technology was tested in two phases. In the first phase, a client/server application was developed to measure and record general information about the quality of each link. Four different types of tests were developed to measure channel properties such as data rate, latency, inter-arrival jitter, and packet loss using various signal strengths, packet sizes, network protocols, and traffic loads. In the second phase of testing, the H.264 Scalable Video Codec (SVC) was used to transmit real-time ultrasound video streams over each of the wireless links to observe the image quality as well as the diagnostic value of the received video stream. The information gathered during both testing phases revealed the abilities and limitations of the different wireless technologies. The results from the performance testing will be valuable in the future for those trying to develop network applications for telemedicine procedures over these wireless telecommunication options. Additionally, the testing demonstrated that the system is currently capable of using H.264 SVC compression to transmit VGA quality ultrasound video at 30 frames per second (fps) over 802.11g while QVGA resolution at frame rates between 10 and 15 fps is possible over 3G and satellite networks."

Le besoin des applications de streaming multimédia en temps réel sur Internet ne cesse d’accroître. Dans ce contexte, les réseaux Pair-à-Pair (P2P) jouent un rôle prépondérant afin d’assurer une transmission robuste et extensible (scalable) de contenu multimédia à grand-échelle. Cependant, le déploiement des applications de streaming multimédia à travers les réseaux P2P présente plusieurs verrous. En effet, les applications de streaming vidéo nécessitent une garantie de performances temps réel en termes de : délai minimal de bout-en-bout, faible taux de perte de paquets et garantie de la bande passante. Par ailleurs, un réseau P2P est composé d’un ensemble de terminaux interconnectés à travers des réseaux hétérogènes, n’ayant pas forcément les mêmes caractéristiques requises afin d’offrir la même qualité vidéo aux clients finaux.Dans cette thèse, nous analysons différents problèmes liés au streaming en-direct (Live) ou à la demande (VoD) de la vidéo sur les réseaux P2P, et nous proposons un ensemble de mécanismes pour y remédier. / There is an increasing demand for efficient deployment of real-time multimedia streaming applications over Internet. In this context, Peer-to-Peer (P2P) networks are playing an important role for supporting robust and scalable transmission of multimedia content to large-scale and massive receivers. P2P networks consist of different heterogeneous networks and devices, which may not have symmetric characteristics to offer the same video quality to end clients. The deployment of streaming applications such as live and on-demand over P2P networks is challenging. Indeed, video streaming applications require real-time performance guarantee in terms of low end-to-end delay, low packet loss and guaranteed available bandwidth. Moreover, P2P video streaming also experiences the problems of long start-time and churn-induced instability. The situation leads to affect the viewing experience of the users in P2P networks.The aim of this thesis is to analyze these issues and to propose quality-aware mechanisms for both live and on-demand streaming over P2P networks. Our contributions in this dissertation are threefold. First, we propose a small world overlay organization mechanism allowing efficient organization of peers to reduce the end-to-end delay for live streaming applications. Second, we propose a mechanism for cooperative prefetching with differentiated chunks scheduling suitable for P2P video-on-demand (VoD) to efficiently handle seek operations while reducing the response latency and increasing the hit ratio. Finally, we present a smoothing mechanism for layered streaming in P2P networks. The mechanism aims to reduce the number of layer changes under varying network conditions while at the same time achieving a high delivery ratio.

Actuellement, nous constatons une augmentation de demande de services vidéo sur les réseaux P2P. Ces réseaux jouent un rôle primordial dans la transmission de contenus multimédia à grande échelle pour des clients hétérogènes. Cependant, le déploiement de services vidéo temps réel sur les réseaux P2P a suscité un grand nombre de défis dû à l’hétérogénéité des terminaux et des réseaux d’accès, aux caractéristiques dynamiques des pairs, et aux autres problèmes hérités des protocoles TCP/IP, à savoir les pertes de paquets, les délais de transfert et la variation de la bande passante de bout-en-bout. Dans ce contexte, l’objectif de cette thèse est d’analyser les différents problèmes et de proposer un mécanisme de transport vidéo temps réel avec le provisionnement de la qualité de Service (QoS). Ainsi, nous proposons trois contributions majeures. Premièrement, il s’agit d’un mécanisme de streaming vidéo adaptatif permettant de sélectionner les meilleurs pair émetteurs. Ce mécanisme permet de structurer les pairs dans des réseaux overlay hybrides avec une prise en charge des caractéristiques sémantiques du contenu et des caractéristiques physiques du lien de transport. Nous présentons ensuite un mécanisme d’ordonnancement de paquets vidéo combiné à une commutation de pairs et/ou de flux pour assurer un transport lisse. Finalement, une architecture de collaboration entre les applications P2P et les fournisseurs de services / réseaux est proposée pour supporter un contrôle d’admission de flux. / There is an increasing demand for scalable deployment of real-time multimedia streaming applications over Internet. In this context, Peer-to-Peer (P2P) networks are playing an important role for supporting robust and large-scale transmission of multimedia content to heterogeneous clients. However, the deployment of real-time video streaming applications over P2P networks arises lot of challenges due to heterogeneity of terminals and access networks, dynamicity of peers, and other problems inherited from IP network. Real-time streaming applications are very sensitive to packet loss, jitter / transmission delay, and available end-to-end bandwidth. These elements have key importance in QoS provisioning and need extra consideration for smooth delivery of video streaming applications over P2P networks. Beside the abovementioned issues, P2P applications lack of awareness in constructing their overlay topologies and do not have any explicit interaction with service and network providers. This situation leads to inefficient utilization of network resources and may cause potential violation of peering agreements between providers. The aim of this thesis is to analyze these issues and to propose an adaptive real-time transport mechanism for QoS provisioning of Scalable Video Coding (SVC) applications over P2P networks. Our contributions in this dissertation are threefold. First, we propose a hybrid overlay organization mechanism allowing intelligent organization of sender peers based on network-awareness, media- awareness, and quality-awareness. This overlay organization is further used for an appropriate selection of best sender peers, and the efficient switching of peers to ensure a smooth video delivery when any of the sender peers is no more reliable. Second, we propose a packet video scheduling mechanism to assign different parts of the video content to specific peers. Third, we present a service provider driven P2P network framework that enables effective interaction between service / network providers and P2P applications to perform QoS provisioning mechanism for the video streaming.

abstract: With tremendous increase in the popularity of networked multimedia applications, video data is expected to account for a large portion of the traffic on the Internet and more importantly next-generation wireless systems. To be able to satisfy a broad range of customers requirements, two major problems need to be solved. The first problem is the need for a scalable representation of the input video. The recently developed scalable extension of the state-of-the art H.264/MPEG-4 AVC video coding standard, also known as H.264/SVC (Scalable Video Coding) provides a solution to this problem. The second problem is that wireless transmission medium typically introduce errors in the bit stream due to noise, congestion and fading on the channel. Protection against these channel impairments can be realized by the use of forward error correcting (FEC) codes. In this research study, the performance of scalable video coding in the presence of bit errors is studied. The encoded video is channel coded using Reed Solomon codes to provide acceptable performance in the presence of channel impairments. In the scalable bit stream, some parts of the bit stream are more important than other parts. Parity bytes are assigned to the video packets based on their importance in unequal error protection scheme. In equal error protection scheme, parity bytes are assigned based on the length of the message. A quantitative comparison of the two schemes, along with the case where no channel coding is employed is performed. H.264 SVC single layer video streams for long video sequences of different genres is considered in this study which serves as a means of effective video characterization. JSVM reference software, in its current version, does not support decoding of erroneous bit streams. A framework to obtain H.264 SVC compatible bit stream is modeled in this study. It is concluded that assigning of parity bytes based on the distribution of data for different types of frames provides optimum performance. Application of error protection to the bit stream enhances the quality of the decoded video with minimal overhead added to the bit stream. / Dissertation/Thesis / M.S. Electrical Engineering 2011

Visando atuação flexível em redes heterogêneas, modernos sistemas multimídia podem adotar o conceito da codificação escalável, onde o fluxo de vídeo é composto por múltiplas camadas, cada qual complementando e aprimorando gradualmente as características de exibição, de forma adaptativa às capacidades de cada receptor. Atualmente, a especificação H.264/SVC representa o estado da arte da área, por sua eficiência de codificação aprimorada, porém demanda recursos computacionais extremamente elevados. Neste contexto, o presente trabalho apresenta uma arquitetura de projeto colaborativo de hardware e software, que explora as características dos diversos algoritmos internos do codificador H.264/SVC, buscando um adequado balanceamento entre as duas tecnologias (hardware e software) para a implementação prática de um codificador escalável de até 16 camadas em formato de 1920x1080 pixels. A partir de um modelo do código de referência H.264/SVC, refinado para reduzir tempos de codificação, foram definidas estratégias de particionamento de módulos e integração entre entidades de software e hardware, avaliando-se questões como dependência de dados e potencial de paralelismo dos algoritmos, assim como restrições práticas das interfaces de comunicação e acessos à memória. Em hardware foram implementados módulos de transformadas, quantização, filtro anti-blocagem e predição entre camadas, permanecendo em software funções de gerência do sistema, entropia, controle de taxa e interface com usuário. A solução completa obtida, integrando módulos em hardware, sintetizados em uma placa de desenvolvimento, com o software de referência refinado, comprova a validade da proposta, pelos significativos ganhos de desempenho registrados, mostrando-se como uma solução adequada para aplicações que exijam codificação escalável tempo real. / In order to support heterogeneous networks and distinct devices simultaneously, modern multimedia systems can adopt the scalability concept, when the video stream is composed by multiple layers, each one being responsible for gradually enhance the video exhibition quality, according to specific receiver capabilities. Currently the H.264/SVC specification can be considered the state-of-art in this area, by improving the coding efficiency, but, in the other hand, impacting in extremely high computational demands. Based on that, this work presents a hardware/software co-design architecture, which explores the characteristics of H.264/SVC internal algorithms, aiming the right balancing between both technologies (hardware and software) in order to generate a practical scalable encoder implementation, able to process up to 16 layers in 1920x1080 pixels format. Based in an H.264/SVC reference code model, which was refined in order to reduce global encoding time, the approaches for module partitioning and data integration between hardware and software were defined. The proposed methodology took into account characteristics like data dependency and inherent possibility of parallelism, as well practical restrictions like influence of communication interfaces and memory accesses. Particularly, the modules of transforms, quantization, deblocking and inter-layer prediction were implemented in hardware, while the functions of system management, entropy, rate control and user interface were kept in software. The whole solution, which was obtained integrating hardware modules, synthesized in a development board, with the refined H.264/SVC reference code, validates the proposal, by the significant performance gains registered, indicating it as an adequate solution for applications which require real-time video scalable coding.

Os constantes avanços nas áreas de transmissão e processamento de dados ao longo dos últimos anos permitiram a criação de diversas aplicações e serviços baseados em dados multimídia, como streaming de vídeo, videoconferências, aulas remotas e IPTV. Além disso, avanços nas demais áreas da computação e engenharias, possibilitaram a construção de uma enorme diversidade de dispositivos de acesso a esses serviços, desde computadores pessoais até celulares, para citar os mais utilizados atualmente. Muitas dessas aplicações e dispositivos estão amplamente difundidos hoje em dia, e, ao mesmo tempo em que a tecnologia avança, os usuários tornam-se mais exigentes, buscando sempre melhor qualidade nos serviços que utilizam. Devido à grande variedade de redes e dispositivos atuais, uma dificuldade existente é possibilitar o acesso universal a uma transmissão. Uma alternativa criada é utilizar transmissão de vídeo escalável com IP multicast e controlada por mecanismos para adaptabilidade e controle de congestionamento. O produto final dessas transmissões mulimídia são os próprios dados multimídia (vídeo e áudio, principalmente) que o usuário está recebendo, portanto a qualidade destes dados é fundamental para um bom desempenho do sistema e satisfação dos usuários. Este trabalho apresenta um estudo de avaliações subjetivas de qualidade aplicadas em sequências de vídeo codificadas através da extensão escalável do padrão H.264 (SVC). Foi executado um conjunto de testes para avaliar, principalmente, os efeitos da instabilidade da transmissão (variação do número de camadas de vídeo recebidas) e a influência dos três métodos de escalabilidade (espacial, temporal e de qualidade) na qualidade dos vídeos. As definições foram baseadas em um sistema de transmissão em camadas com utilização de protocolos para adaptabilidade e controle de congestionamento. Para execução das avaliações subjetivas foi feito o uso da metodologia ACR-HRR e recomendações das normas ITU-R Rec. BT.500 e ITU-T Rec. P.910. Os resultados mostram que, diferente do esperado, a instabilidade não provoca grandes alterações na qualidade subjetiva dos vídeos e que o método de escalabilidade temporal tende a apresentar qualidade bastante inferior aos outros métodos. As principais contribuições deste trabalho estão nos resultados obtidos nas avaliações, além da metodologia utilizada durante o desenvolvimento do trabalho (definição do plano de avaliação, uso das ferramentas como o JSVM, seleção do material de teste, execução das avaliações, entre outros), das aplicações desenvolvidas, da definição de alguns trabalhos futuros e de possíveis objetivos para avaliações de qualidade. / The constant advances in multimedia processing and transmission over the past years have enabled the creation of several applications and services based on multimedia data, such as video streaming, teleconference, remote classes and IPTV. Futhermore, a big variety of devices, that goes from personal computers to mobile phones, are now capable of receiving these transmissions and displaying the multimedia data. Most of these applications are widely adopted nowadays and, at the same time the technology advances, the user are becoming more demanding about the quality of the services they use. Given the diversity of devices and networks available today, one of the big challenges of these multimedia systems is to be able to adapt the transmission to the receivers' characteristics and conditions. A suitable solution to provide this adaptation is the integration of scalable video coding with layered transmission. As the final product in these multimedia systems are the multimedia data that is presented to the user, the quality of these data will define the performace of the system and the users' satisfaction. This paper presents a study of subjective quality of scalable video sequences, coded using the scalable extension of the H.264 standard (SVC). A group of experiments was performed to measure, primarily, the efeects that the transmission instability (variations in the number of video layers received) has in the video quality and the relationship between the three scalability methods (spatial, temporal and quality) in terms of subjective quality. The decisions taken to model the tests were based on layered transmission systems that use protocols for adaptability and congestion control. To run the subjective assessments we used the ACR-HRR methodology and recommendations given by ITU-R Rec. BT.500 and ITU-T Rec. P.910. The results show that the instability modelled does not causes significant alterations on the overall video subjective quality if compared to a stable video and that the temporal scalability usually produces videos with worse quality than the spatial and quality methods, the latter being the one with the better quality. The main contributions presented in this work are the results obtained in the subjective assessments. Moreover, are also considered as contributions the methodology used throughout the entire work (including the test plan definition, the use of tools as JSVM, the test material selection and the steps taken during the assessment), some applications that were developed, the definition of future works and the specification of some problems that can also be solved with subjective quality evaluations.

Os constantes avanços nas áreas de transmissão e processamento de dados ao longo dos últimos anos permitiram a criação de diversas aplicações e serviços baseados em dados multimídia, como streaming de vídeo, videoconferências, aulas remotas e IPTV. Além disso, avanços nas demais áreas da computação e engenharias, possibilitaram a construção de uma enorme diversidade de dispositivos de acesso a esses serviços, desde computadores pessoais até celulares, para citar os mais utilizados atualmente. Muitas dessas aplicações e dispositivos estão amplamente difundidos hoje em dia, e, ao mesmo tempo em que a tecnologia avança, os usuários tornam-se mais exigentes, buscando sempre melhor qualidade nos serviços que utilizam. Devido à grande variedade de redes e dispositivos atuais, uma dificuldade existente é possibilitar o acesso universal a uma transmissão. Uma alternativa criada é utilizar transmissão de vídeo escalável com IP multicast e controlada por mecanismos para adaptabilidade e controle de congestionamento. O produto final dessas transmissões mulimídia são os próprios dados multimídia (vídeo e áudio, principalmente) que o usuário está recebendo, portanto a qualidade destes dados é fundamental para um bom desempenho do sistema e satisfação dos usuários. Este trabalho apresenta um estudo de avaliações subjetivas de qualidade aplicadas em sequências de vídeo codificadas através da extensão escalável do padrão H.264 (SVC). Foi executado um conjunto de testes para avaliar, principalmente, os efeitos da instabilidade da transmissão (variação do número de camadas de vídeo recebidas) e a influência dos três métodos de escalabilidade (espacial, temporal e de qualidade) na qualidade dos vídeos. As definições foram baseadas em um sistema de transmissão em camadas com utilização de protocolos para adaptabilidade e controle de congestionamento. Para execução das avaliações subjetivas foi feito o uso da metodologia ACR-HRR e recomendações das normas ITU-R Rec. BT.500 e ITU-T Rec. P.910. Os resultados mostram que, diferente do esperado, a instabilidade não provoca grandes alterações na qualidade subjetiva dos vídeos e que o método de escalabilidade temporal tende a apresentar qualidade bastante inferior aos outros métodos. As principais contribuições deste trabalho estão nos resultados obtidos nas avaliações, além da metodologia utilizada durante o desenvolvimento do trabalho (definição do plano de avaliação, uso das ferramentas como o JSVM, seleção do material de teste, execução das avaliações, entre outros), das aplicações desenvolvidas, da definição de alguns trabalhos futuros e de possíveis objetivos para avaliações de qualidade. / The constant advances in multimedia processing and transmission over the past years have enabled the creation of several applications and services based on multimedia data, such as video streaming, teleconference, remote classes and IPTV. Futhermore, a big variety of devices, that goes from personal computers to mobile phones, are now capable of receiving these transmissions and displaying the multimedia data. Most of these applications are widely adopted nowadays and, at the same time the technology advances, the user are becoming more demanding about the quality of the services they use. Given the diversity of devices and networks available today, one of the big challenges of these multimedia systems is to be able to adapt the transmission to the receivers' characteristics and conditions. A suitable solution to provide this adaptation is the integration of scalable video coding with layered transmission. As the final product in these multimedia systems are the multimedia data that is presented to the user, the quality of these data will define the performace of the system and the users' satisfaction. This paper presents a study of subjective quality of scalable video sequences, coded using the scalable extension of the H.264 standard (SVC). A group of experiments was performed to measure, primarily, the efeects that the transmission instability (variations in the number of video layers received) has in the video quality and the relationship between the three scalability methods (spatial, temporal and quality) in terms of subjective quality. The decisions taken to model the tests were based on layered transmission systems that use protocols for adaptability and congestion control. To run the subjective assessments we used the ACR-HRR methodology and recommendations given by ITU-R Rec. BT.500 and ITU-T Rec. P.910. The results show that the instability modelled does not causes significant alterations on the overall video subjective quality if compared to a stable video and that the temporal scalability usually produces videos with worse quality than the spatial and quality methods, the latter being the one with the better quality. The main contributions presented in this work are the results obtained in the subjective assessments. Moreover, are also considered as contributions the methodology used throughout the entire work (including the test plan definition, the use of tools as JSVM, the test material selection and the steps taken during the assessment), some applications that were developed, the definition of future works and the specification of some problems that can also be solved with subjective quality evaluations.

Visando atuação flexível em redes heterogêneas, modernos sistemas multimídia podem adotar o conceito da codificação escalável, onde o fluxo de vídeo é composto por múltiplas camadas, cada qual complementando e aprimorando gradualmente as características de exibição, de forma adaptativa às capacidades de cada receptor. Atualmente, a especificação H.264/SVC representa o estado da arte da área, por sua eficiência de codificação aprimorada, porém demanda recursos computacionais extremamente elevados. Neste contexto, o presente trabalho apresenta uma arquitetura de projeto colaborativo de hardware e software, que explora as características dos diversos algoritmos internos do codificador H.264/SVC, buscando um adequado balanceamento entre as duas tecnologias (hardware e software) para a implementação prática de um codificador escalável de até 16 camadas em formato de 1920x1080 pixels. A partir de um modelo do código de referência H.264/SVC, refinado para reduzir tempos de codificação, foram definidas estratégias de particionamento de módulos e integração entre entidades de software e hardware, avaliando-se questões como dependência de dados e potencial de paralelismo dos algoritmos, assim como restrições práticas das interfaces de comunicação e acessos à memória. Em hardware foram implementados módulos de transformadas, quantização, filtro anti-blocagem e predição entre camadas, permanecendo em software funções de gerência do sistema, entropia, controle de taxa e interface com usuário. A solução completa obtida, integrando módulos em hardware, sintetizados em uma placa de desenvolvimento, com o software de referência refinado, comprova a validade da proposta, pelos significativos ganhos de desempenho registrados, mostrando-se como uma solução adequada para aplicações que exijam codificação escalável tempo real. / In order to support heterogeneous networks and distinct devices simultaneously, modern multimedia systems can adopt the scalability concept, when the video stream is composed by multiple layers, each one being responsible for gradually enhance the video exhibition quality, according to specific receiver capabilities. Currently the H.264/SVC specification can be considered the state-of-art in this area, by improving the coding efficiency, but, in the other hand, impacting in extremely high computational demands. Based on that, this work presents a hardware/software co-design architecture, which explores the characteristics of H.264/SVC internal algorithms, aiming the right balancing between both technologies (hardware and software) in order to generate a practical scalable encoder implementation, able to process up to 16 layers in 1920x1080 pixels format. Based in an H.264/SVC reference code model, which was refined in order to reduce global encoding time, the approaches for module partitioning and data integration between hardware and software were defined. The proposed methodology took into account characteristics like data dependency and inherent possibility of parallelism, as well practical restrictions like influence of communication interfaces and memory accesses. Particularly, the modules of transforms, quantization, deblocking and inter-layer prediction were implemented in hardware, while the functions of system management, entropy, rate control and user interface were kept in software. The whole solution, which was obtained integrating hardware modules, synthesized in a development board, with the refined H.264/SVC reference code, validates the proposal, by the significant performance gains registered, indicating it as an adequate solution for applications which require real-time video scalable coding.