Användningen av strömmande video ökar snabbt just nu. Ett populärt konceptär adaptive bitrate streaming som går ut på att en video kodas i flera olikabithastigheter. Dessa videor tas sedan och delas upp i små filer och görstillgänglig via internet. När du vill spela upp en sådan video laddar du först hemen fil som beskriver vart filerna finns och i vilka bithastigheter de är kodade i.Mediaspelaren kan där efter börja ladda hem filerna och spela upp dom. Om defysiska förutsättningarna, som exempelvis nedladdningshastighet eller CPUbelastning,ändras under uppspelningen kan mediaspelaren enkelt byta kvalitépå videon genom att börja ladda filer av en annan bithastighet och slippa attvideon laggar. Denna rapport tar därför en närmare titt på fyra tekniker inomadaptive bitrate streaming. De som undersöks är HTTP Live Streaming,Dynamic Adaptive Streaming over HTTP, HTTP Dynamic Streaming ochSmooth Streaming med avseende på vilka protokoll som dom använder.Rapporten undersöker även hur Apple och FFmpeg har implementerat HTTPLive streaming med avseende på hur mycket data som behövs läsas i en filinnan videon kan börja spelas upp. Rapporten visar att det inte är så storaskillnader mellan de fyra teknikerna. Dock sticker Dynamic AdaptiveStreaming over HTTP ut lite genom att vara helt oberoende av vilket ljud ellervideoprotokoll som används. Rapporten visar också på en brist i specificeringenav HTTP Live Streaming då det inte är specificerat att första komplettabildrutan i videoströmmen bör ligga i början av filen. I Apples implementationbehövs upp till 30 kB data läsas innan uppspelning kan påbörjas medan iFFmpegs implementation är det ca 600 byte. / The use of streaming video is growing rapidly at the moment. A popular conceptis adaptive bitrate streaming, which is when a video gets encoded in severaldifferent bit rates. These videos are then split into small files and made availablevia the internet. When you want to play such a video, you first download afile that describes where the files are located and in what bitrates they are encodedin. The media player then begin downloading the files and play them. Ifthe physical conditions, such as the download speed or CPU load, changes duringplayback, the media player can easily change the quality of the video bystarting to downloading files of a different bit rate and avoid that the video lags.This report will take a closer look at four techniques in adaptive bitrate streaming.They examined techniques are HTTP Live Streaming, Dynamic AdaptiveStreaming over HTTP, HTTP Dynamic Streaming and Smooth Streaming andwhich protocols they use. The report also examines how Apple and FFmpeg hasimplemented HTTP Live Streaming with respect to how much data is needed toread a file before the video can begin to be played. The report shows that thereare no large differences between the four techniques. However, Dynamic AdaptiveStreaming over HTTP stood out a bit by being completely independent ofany audio or video protocols. The report also shows a shortcoming in the specificationof HTTP Live Streaming as it is not specified that the first completeframe of the video stream should be at the beginning of the file. In Apple's implementationits needed to read up to 30 KB of data before playback can bestarted while in FFmpeg's implementation its about 600 bytes.

Le trafic de vidéo streaming étant en très forte augmentation dans les réseaux mobiles, il devient essentiel pour les opérateurs de tenir compte des spécificités de ce trafic pour bien dimensionner et configurer le réseau. Dans cette thèse, nous nous intéressons à la modélisation du trafic de vidéo streaming dans les réseaux mobiles. Pour le trafic de vidéo streaming en temps-réel, nous obtenons une forme analytique pour une métrique de qualité de service (QoS) importante, le taux de perte de paquets, et utilisons ce modèle à faire du dimensionnement. Pour le trafic de vidéo streaming de type HTTP adaptatif, nous proposons et analysons d’autres métriques de QoS comme le bitrate moyen, le taux de déficit vidéo et le surplus de buffer, afin de trouver le bon compromis entre résolution de la vidéo et fluidité de la diffusion vidéo. Nous étudions par simulation l’impact de quelque paramètres clés du systéme. Nous montrons que l’utilisation de segments de vidéo courts, d’un nombre réduit d’encodages vidéos et de l’ordonnancement de type round robin améliore la fluidité de la vidéo tout en diminuant sa résolution. Nous proposons par ailleurs d’adapter le nombre des segments téléchargés dans une requête HTTP de sorte que chaque requête corresponde au même volume de données. Enfin, nous appliquons les techniques de l’apprentissage automatique pour analyser la corrélation entre les caracteristiques du système et la qualité d’expérience (QoE) des utilisateurs. / As the traffic of video streaming increases significantly in mobile networks, it is essential for oper- ators to account for the features of this traffic when dimensioning and configuring the network. The focus of this thesis is on traffic models of video streaming in mobile networks. For real-time video streaming traffic, we derive an analytical form for an important Quality-of-Service (QoS) metric, the packet outage rate, and utilize the model for dimensioning. For HTTP adaptive video streaming traffic, we propose and evaluate other QoS metrics such as the mean video bit rate, the deficit rate and the buffer surplus, so as to find the good trade-off between video resolution and playback smoothness. We study by simulation the impacts of some key parameters of the system. We show that using smaller chunk durations, fewer video coding rates and round-robin scheduling may provide a smoother video playback but decrease the mean video resolution. We also propose to adapt the number of chunks downloaded in an HTTP request so that each HTTP request has the same data volume. Finally, we apply machine learning techniques to analyze the correlation between system characteristics and the quality of experience (QoE) of users.

Master of Science / Department of Computing and Information Sciences / Gurdip Singh / A streaming video application allows a sequence of "moving images" to be sent over the Internet and displayed by a viewer as they arrive. This application is meant for viewing live videos on handheld devices such as PDAs and iPAQs. It captures video data from a webcam installed on a tablet pc, which is then sent over a UDP socket to an iPAQ via an ad hoc network where live video can be viewed in real time. This is achieved by sending video data, frame by frame, and displaying on iPAQ as it arrives. This application also allows taking a snapshot of the video which can be saved for later viewing and also allows the user to dynamically change the resolution of the video as being viewed. Two versions of this application have been developed, one using a TCP connection for video transfer between a tablet pc (server) and an iPAQ (client) and the other using a UDP connection. This report studies the trade-off between distance and time as each frame arrives at the client for both the versions. This implementation also supports multiple clients to connect to the server and allows video to be viewed simultaneously on more than one client and thus studies the trade-off between distance and time for multiple clients. This project is implemented using C#.NET on Microsoft Visual Studio 2005. It uses Microsoft .NET framework 2.0 for server (tablet pc) and Microsoft .NET Compact Framework 2.0 for client (iPAQ). Video streaming is useful in several areas such as entertainment media, live conferences, surveillance and security field. For entertainment media, streaming video avoids having a web user wait for a large file to be downloaded before viewing the video. Instead, the media is sent in a continuous stream and is played as it arrives. For surveillance purposes, the streamed video gives a real time view of the field. The primary application of this implementation is to be used in the field of sensor networks.

Denna studie syftar till att undersöka den förändring som den svenska musikbranschen hargenomgått i och med det teknologiskifte som streamingens inträde på marknaden har medfört.Genom en kvalitativ intervjubaserad studie identifieras både specifika förändringar hosskivbolagen i fråga men även deras syn på hur branschen förändrats i stort och vilkavärdeförändringar som ligger bakom branschförändringen. Studien utgår från ettvärdeförändrande perspektiv och ämnar identifiera värdeförändringarna på en företags- ochbranschnivå, som skett efter att branschen genomgått ett teknologiskifte. Studiens resultat visaratt skivbolagen har genomgått en stor organisatorisk påfrestning i och med det teknologiska skiftesom streamingen har inneburit för branschen och identifierar en stark anpassningsförmåga somen vitalitet för att överleva större branschförändringar.

Movies, songs, conferences, video streams and other multimedia services are spreading in everybody’s life, needs and work. Different types of audio and video codecs of various qualities appear to be used for different purposes. However, each codec can be suitable for a special usage, as for example a codec that can be use for high quality film may not be suitable for streaming it into network. The main purpose of this thesis is to compare the performance of different codecs in one particular scenario and indicate which codec performance is the best. The thesis work includes the investigation of several codec aspects (audio and video) and problems related to choosing a multimedia codec (coder/decoder) suitable for large-scale multimedia distribution over the Internet. More specifically, the work focuses on various features of modern media codecs, compares them and finds the best application for each of them. Furthermore, the thesis provides an overview of available research results related to this work and codec comparison tables for codecs and their features. Additionally, as part of this work, a testbed was developed to measure the performance of codecs when used to stream media over a network. The testbed enabled the collection of various QoS parameters, such as bandwidth usage and burst, size for each codec considered in this thesis. / Address: Minervavägen 22A, Room 2, 371 41 Karlskrona, sweden. Cel: +46706456674

Enabling efficient media streaming over wireless channels requires efficient utilization of the limited wireless spectrum while satisfying multimedia applications' quality of service (QoS) requirements. In this dissertation, we provide insights into network and application-centric approaches for media streaming over wireless channels. In network-centric approaches, the fundamental problem is how to model network variations at the different layers and optimize the total quality across these layers. We use Finite-state Markov chain (FSMC) models to investigate the packet loss and delay performance over a wireless link. We propose a new method for partitioning the received SNR space that results in a FSMC model with tractable queueing performance. We then use this model to derive closed-form expressions for the {\em Effective Bandwidth\/} subject to either packet loss or packet delay constraints. In application-centric approaches, we take into account the VBR nature of video frames and channel dynamics and integrate in the analysis the dynamics of the playback buffer occupancy. We introduce a mixture of sourec/channel rate adaptation schemes that target efficient utilization of the wireless spectrum and safeguard the continuity of media streaming over wireless channels. First, we propose two source-rate control schemes for streaming video over wireless channels that provide gracefully degraded quality and soft guarantees on frame delay. The schemes are designed to maximize the source bit rate at the encoder while preventing/reducing events of starvation at the decoder. Second, we present a novel cycle-based rate adaptation scheme. The scheme is designed to maximize the source bit rate at the encoder while guaranteeing an upper bound on the probability of starvation at the playback buffer. This approach can be applied to both {\em one-way} and {\em interactive} video. Finally, we propose a playback-adaptive source/channel rate control (SCRC) for video streaming over wireless channels. We exploit the so-called playback adaptation margin and the playback buffer occupancy to control the source and channel rates. The SCRC scheme is designed to limit potential playback discontinuities that may occur due to variations in the wireless link.

In the present study, the researcher examined the motivational variables of students within the Ontario Secondary School system, where groups based on course stream, LD status, gender, and achievement level were compared. This research was partially exploratory, where the researcher aimed to validate existing research on gender, LD status, and achievement, as well as to examine how motivational variables relate to course stream. Past research has shown that endorsing an entity theory of intelligence, having a preference for performance goals, attributing success to ability rather than effort, and having low confidence in one’s ability to self-regulate are all associated with lower achievement, and with the presence of a learning disability (e.g., Baird, Scott, Dearing & Hamill, 2009). A total of 243 secondary school students (127 male, 116 female) from one rural school in Ontario participated in this study. The participants completed a 28-item questionnaire made up of several subscales, including the Implicit Theories of Intelligence Scale (Dweck, 1999), the Learning vs. Performance Goal Preference Scale (Dweck, 1999), the Effort Attribution Scale (Dweck & Leggett, 1988), and the Self-Efficacy for Self-Regulated Learning Scale (Zimmerman et al., 1992). Regarding stream, an independent t-test revealed that students in the university English course had significantly more confidence in their ability to self-regulate than their peers in the college English course. A MANOVA revealed that individuals who had received a mid-term English grade of A (80-100%), regardless of stream, had significantly more confidence their ability to self-regulate than students who received lower grades. Finally, an independent t-test revealed that participants with a learning disability were more concerned with their personal growth than their performance when compared to participants without a learning disability. No differences in motivational variables were found between males and females. These findings suggest that educators and parents should place more emphasis on helping students to develop confidence in their self-regulatory abilities, as this appears to be an important variable in students’ achievement, as well as related to the academic stream in which they are enrolled. / Thesis (Master, Education) -- Queen's University, 2013-08-12 09:01:56.34

The aim of this study is to understand how leaders in the Canadian radio industry perceive the nascent personalized music streaming service industry, and how those perceptions inform their current business decisions. Over the last few years, Internet-based music services like Deezer and Rdio have launched in Canada, providing listeners with the opportunity of an alternative music experience to conventional broadcast radio. Through five interviews with experts representing terrestrial radio, on-demand services and regulation, three scenarios are presented for the future of conventional radio. In addition, a conceptualized listener profile is created using Grounded Theory Method. This profile buttresses the central finding of this research: should key political, economic and social factors remain in place, conventional radio is likely to continue to dominate as the leading commercial audio platform for Canadians in the years to come.

Ingeniero Civil en Computación / Este trabajo continúa el estudio presentado en el paper Frame Allocation Algorithms for Multi-threaded Network Cameras (2010) de J. Piquer y J. Bustos-Jiménez. Este paper indaga sobre el efecto que tiene la selección de imágenes o frames de video en un servidor, para su transmisión a clientes a través de la red. Esta selección de frames ocurre al momento de asignar un frame para ser enviado a un cliente, y cuando se debe reemplazar uno de los frames guardados en la memoria del servidor por un nuevo frame obtenido desde la fuente de video. Las pruebas presentadas en el paper muestran que los algoritmos de asignación basados en contadores de referencias son los más efectivos, ya que permiten desocupar más rápidamente los frames que están en memoria para que puedan ser reemplazados por nuevos frames. En el presente trabajo se implementaron los algoritmos de asignación y reemplazo de frames en la plataforma de streaming VLC, con el objetivo de usarlos para hacer streaming de video desde una fuente de video real hacia clientes reales, y comparar el desempeño de los distintos algoritmos con el rendimiento que actualmente ofrece VLC. Se realizaron pruebas para medir el número de frames por segundo (FPS) observado en los clientes al usar los algoritmos, y el uso de memoria y CPU en el servidor durante la transmisión de video. Los resultados obtenidos muestran que, para los formatos de video que la implementación soporta, se obtiene una mejora importante en el rendimiento de VLC al usar los algoritmos de asignación y reemplazo adecuados, tanto en términos de FPS en los clientes como en uso de CPU en el servidor, utilizando en promedio una cantidad de memoria muy similar a la que usa VLC en condiciones normales. Se confirma además lo que la simulación realizada en el paper sugiere sobre el rendimiento de los algoritmos de asignación basados en los contadores de referencias, que en VLC ofrecen también el mejor desempeño.

Context. Big Data and Cloud Computing nowadays require large amounts of storage that are accessible by many servers. To overcome the performance bottlenecks and single point of failure distributed storage systems came into force. So, our main aim in this thesis is evaluating the performance of these storage systems. A file coding technique is used that is the erasure coding which will help in data protection for the storage systems. Objectives. In this study, we investigate the performance evaluation of distributed storage system and understand the effect on performance for various patterns of I/O operations that is the read and write and also different measurement approaches for storage performance. Methods. The method is to use synthetic workload generator by streaming and transcoding video data as well as benchmark tool which generates the workload like SPECsfs2014 is used to evaluate the performance of distributed storage systems of GlusterFS and Compuverde which are file based storage. Results. In terms of throughput results, Gluster and Compuverde perform similar for both NFS and SMB server. The average latency results for both NFS and SMB shares indicate that Compuverde has lower latency. When comparing results of both Compuverde and Gluster, Compuverde delivers 100% IOPS with NFS server and Gluster delivers relatively near to the requested OP rate and with SMB server Gluster delivers 100% IOPS and Compuverde delivers more than the requested OP rate.