<p>The purpose of this thesis is to examine the possibilities of creating a MPEG-4-compatible set-top box for IP-networks based on open standards. </p><p>Existing alternatives for transporting MPEG-4 over IP are evaluated and ISMA is found to be an important actor within the area. ISMA is a non-profit corporation formed to provide a forum for the creation of specifications that define an interoperable implementation for streaming rich media over IP-networks. </p><p>Two different designs based on ISMA's recommendation are constructed and evaluated. The designs have different levels of complexity and the more complex design is found to be better due to its extended functionality. </p><p>During the design process a number of problems related to this kind of set-top box are discovered. It is believed, however, that many of these problems will be solved within the near future.</p>

<p>Computer animated faces have found their way into a wide variety of areas. Starting from entertainment like computer games, through television and films to user interfaces using “talking heads”. Animated faces are also becoming popular in web applications in form of human-like assistants or newsreaders. </p><p>This thesis presents a few aspects of dealing with human face animations, namely: editing, playing and transmitting such animations. It describes a standard for handling human face animations, the MPEG-4 Face Animation, and shows the process of designing, implementing and evaluating applications compliant to this standard. </p><p>First, it presents changes introduced to the existing components of the Visage|toolkit package for dealing with facial animations, offered by the company Visage Technologies AB. It also presents the process of designing and implementing of an application for editing facial animations compliant to the MPEG-4 Face Animation standard. Finally, it discusses several approaches to the problem of streaming facial animations over the Internet or the Local Area Network (LAN).</p>

<p>The third generation, 3G, mobile telephone systems are designed for multimedia communication and will offer us similar services as in our stationary computers. This will involve large traffic loads, especially in the downlink direction, i.e. from base station to terminal. To improve the downlink capacity for packet data services a new concept is included in evolved 3G networks. The concept is called High Speed Data Packet Access, HSDPA, and provides peak bit rates of 14 Mbps. HSDPA uses a so-called best effort channel, i.e. it is developed for services that do not require guaranteed bit rates. The channel is divided in time between the users and a scheduling algorithm is used to allocate the channel among them. </p><p>Streaming is a common technology for video transmission over the Internet and with 3G it is supposed to become popular also in our mobiles. Streaming generates lots of data traffic in the downlink direction and it would thus be satisfying to make use of the high bit rates HSDPA provides. The problem is that streaming requires reasonable stable bit rates, which is not guaranteed using HSDPA. The aim of this study is to modify the scheduling algorithms to prioritise streaming over web users and provide streaming Quality of Service, QoS. QoS is the ability to guarantee certain transmission characteristics. </p><p>The results of the study show that it is hard to improve the streaming capacity by modifications of the scheduling. Of course, a consequence is that the web user throughput is decreased and to avoid this, new users have to be rejected by the admission control.The solution is to prioritise the streaming users both in the scheduling algorithm and in the admission control, i.e. when the system is nearly full new web users are rejected. By doing so the results are significantly improved.</p>

<p>This project investigates the importance of availability in the scope of dictation programs. Using speech recognition technology for dictating has not reached the public, and that may very well be a result of poor availability in today’s technical solutions. </p><p>I have constructed a persona character, Johanna, who personalizes the target user. I have also developed a solution that streams audio into a speech recognition server and sends back interpreted text. Johanna affirmed that the solution was successful in theory. </p><p>I then incorporated test users that tried out the solution in practice. Half of them do indeed claim that their usage has been and will continue to be increased thanks to the new level of availability.</p>

<p>The purpose of this thesis is to develop software to stream input and output audio from a large number of computers in a network to one specific computer in the same network. This computer will save the audio to disk. The audio that is to be saved will consist mostly of spoken communication. The saved audio is to be used in a framework for modeling and visualization.</p><p>There are three major problems involved in designing a software to fill this purpose: recording both input and output audio at the same time, efficiently receiving multiple audio-streams at once and designing an interface where finding and organizing the computers to record audio from is easy.</p><p>The software developed to solve these problems consists of two parts; a server and a client. The server captures the input (microphone) and output (speaker) audio from a computer. To capture the output and input audio simultaneously an external application named Virtual Audio Cable (VAC) is used. The client connects to multiple servers and receives the captured audio. Each one of the client’s server-connections is handled by its own thread. To make it easy to find available servers an Automatic Server Discovery System has been developed. To simplify the organization of the servers they are displayed in a tree-view specifically designed for this purpose.</p>

With the increasing miniaturization of transistors, wire delays are becoming a dominant factor in microprocessor performance. To address this issue, a number of emerging architectures contain replicated processing units with software-exposed communication between one unit and another (e.g., Raw, SmartMemories, TRIPS). However, for their use to be widespread, it will be necesary to develop a common machine language to allow programmers to express an algorithm in a way that can be efficiently mapped across these architectures. We propose a new common machine language for grid-based software-exposed architectures: StreamIt. StreamIt is a high-level programming language with explicit support for streaming computation. Unlike sequential programs with obscured dependence information and complex communication patterns, a stream program is naturally written as a set of concurrent filters with regular steady-state communication. The language imposes a hierarchical structure on the stream graph that enables novel representations and optimizations within the StreamIt compiler. We have implemented a fully functional compiler that parallelizes StreamIt applications for Raw, including several load-balancing transformations. Though StreamIt exposes the parallelism and communication patterns of stream programs, analysis is needed to adapt a stream program to a software-exposed processor. We describe a partitioning algorithm that employs fission and fusion transformations to adjust the granularity of a stream graph, a layout algorithm that maps a stream graph to a given network topology, and a scheduling strategy that generates a fine-grained static communication pattern for each computational element. Using the cycle-accurate Raw simulator, we demonstrate that the StreamIt compiler can automatically map a high-level stream abstraction to Raw. We consider this work to be a first step towards a portable programming model for communication-exposed architectures. / Singapore-MIT Alliance (SMA)

As mobile devices still have limited battery life, processing power, memory, and display size, they cannot yet execute gaming applications with the same fidelity and quality as their PC counterparts. In response, researchers have recently performed research with the goal of the real-time delivery of game content specifically to fit within mobile devices’ limitations. In this thesis, we present a novel approach to tackling the streaming of objects to mobile devices. Our goal is to reduce the number of objects subject to streaming from the server to the target devices, while not violating the user-defined limitations through an efficient, context-aware 3D object selection and prioritization scheme. We take advantage of the game context to stream only the most relevant objects. Our evaluations have shown that this technique not only leads to better performance in general, but also increases the gameplay experience by helping the player to achieve a higher score.

Fourth Generation (4G) cellular technology Third Generation Partnership Project (3GPP) Long Term Evolution (LTE) offers high data rate capabilities to mobile users; and, operators are trying to deliver a true mobile broadband experience over LTE networks. Mobile TV and Video on Demand (VoD) are expected to be the main revenue generators in the near future [36] and efficient video streaming over wireless is the key to enabling this. 3GPP recommends the use of H.264 baseline profiles for all video based services in Third Generation (3G) Universal Mobile Telecommunication System (UMTS) networks. However, LTE networks need to support mobile devices with different display resolution requirements like small resolution mobile phones and high resolution laptops. Scalable Video Coding (SVC) is required to achieve this goal. Feasibility study of SVC for LTE is one of the main agenda of 3GPP Release10. SVC enhances H.264 with a set of new profiles and encoding tools that may be used to produce scalable bit streams. Efficient adaptation methods for SVC video transmission over LTE networks are proposed in this thesis. Advantages of SVC over H.264 are analyzed using real time use cases of mobile video streaming. Further, we study the cross layer adaptation and scheduling schemes for delivering SVC video streams most efficiently to the users in LTE networks in unicast and multicast transmissions. We propose SVC based video streaming scheme for unicast and multicast transmissions in the downlink direction, with dynamic adaptations and a scheduling scheme based on channel quality information from users. Simulation results indicate improved video quality for more number of users in the coverage area and efficient spectrum usage with the proposed methods.

La squelettisation est une méthode de reconnaissance et de caractérisation des formes utilisant des notions de topologie discrète, qui permet d'extraire le squelette d'une forme binaire. Ce squelette (un sous ensemble de la forme centré, fin, topologiquement équivalent à la forme et reconstructible) se calcule grâce à plusieurs balayages de l'image. Dans ce travail de thèse nous nous sommes intéressé au calcul du squelette en une seule passe (mode " streaming "). Pour cela nous avons proposé un nouvel algorithme qui ré-ordonne les calculs sur chaque pixels dans leur ordre d'apparition dans l'image (sans changer le sens de propagation de l'information dans l'image) en calculant une carte de distance asymétrique puis qui trouve les points de l'axe médian. Selon un ordre particulier, qui garantit l'identité de nos résultats avec ceux d'un algorithme de référence, nous pouvons supprimer les points de la carte de distance en ne conservant que les points du squelette dont les points de l'axe médian. En plus du fait que nos squelettes soient identiques au pixel près à ceux obtenus par l'algorithme de référence, notre algorithme les calcule plus rapidement. Bien que notre approche soit générique, nous avons utilisé cet algorithme pour développer un outil logiciel en imagerie médicale permettant d'extraire les caractéristiques osseuses dans le but de mesurer la structure osseuse à partir du squelette obtenu sur des images de micro-scanner. L'extraction de zones d'os trabéculaire 2D sur des images binarisées, permet bien une caractérisation de l'os en accord avec la vérité terrain obtenue par d'autres logiciels.

Peer-to-peer (P2P) streaming has recently received much research attention, with successful commercial systems showing its viability in the Internet. Despite the remarkable popularity in real-world systems, the fundamental properties and limitations are not yet well understood from a theoretical perspective, as there exists a significant gap between the fundamental limits and the performance achieved in practice. In this thesis, we seek to provide an in-depth analytical understanding of fundamental properties and limitations of P2P streaming systems, with a particular spotlight on the performance gap. We first identify the major problem in existing streaming protocols and show that this problem accounts for most of the gap separating the actual and optimal performances of the streaming systems. We then propose a remedy based on network coding to address this problem and show that the gap to the fundamental limits can be significantly reduced.