Streaming multimedia content over the IP-network is poised to be the dominant Internet traffic for the coming decade, predicted to account for more than 91% of all consumer traffic in the coming years. Streaming multimedia content ranges from Internet television (IPTV), video on demand (VoD), peer-to-peer streaming, and 3D television over IP to name a few. Widespread acceptance, growth, and subscriber retention are contingent upon network providers assuring superior Quality of Experience (QoE) on top of todays Internet. This work presents the first empirical understanding of Internet’s video-QoE capabilities, and tools and protocols to efficiently infer and improve them. To infer video-QoE at arbitrary nodes in the Internet, we design and implement MintMOS: a lightweight, real-time, noreference framework for capturing perceptual quality. We demonstrate that MintMOS’s projections closely match with subjective surveys in accessing perceptual quality. We use MintMOS to characterize Internet video-QoE both at the link level and end-to-end path level. As an input to our study, we use extensive measurements from a large number of Internet paths obtained from various measurement overlays deployed using PlanetLab. Link level degradations of intra– and inter–ISP Internet links are studied to create an empirical understanding of their shortcomings and ways to overcome them. Our studies show that intra–ISP links are often poorly engineered compared to peering links, and that iii degradations are induced due to transient network load imbalance within an ISP. Initial results also indicate that overlay networks could be a promising way to avoid such ISPs in times of degradations. A large number of end-to-end Internet paths are probed and we measure delay, jitter, and loss rates. The measurement data is analyzed offline to identify ways to enable a source to select alternate paths in an overlay network to improve video-QoE, without the need for background monitoring or apriori knowledge of path characteristics. We establish that for any unstructured overlay of N nodes, it is sufficient to reroute key frames using a random subset of k nodes in the overlay, where k is bounded by O(lnN). We analyze various properties of such random subsets to derive simple, scalable, and an efficient path selection strategy that results in a k-fold increase in path options for any source-destination pair; options that consistently outperform Internet path selection. Finally, we design a prototype called source initiated frame restoration (SIFR) that employs random subsets to derive alternate paths and demonstrate its effectiveness in improving Internet video-QoE.

Les solutions de streaming vidéo adaptatives basées sur l’utilisation du protocol HTTP ont été largement plébiscitées dans les mondes de l’industrie et de la recherche, notamment pour les possibilités d’améliorations de qualité d’experience qu’elles offrent ainsi que pour leurs facilités de déploiement liées au protocol HTTP. Pour autant, bien que ces solutions permettent d’augmenter la qualité d’experience utilisateurs en diminuant la qualité de la vidéo transmise sur les réseaux pour minimiser les interruptions vidéo liées au temps de chargement, la qualité intrinsèque de la vidéo est limitée par les capacités physiques du chemin entre le serveur utilisé et le client. Dans l’objectif d’augmenter la qualité d’experience utilisateurs et de diminuer les couts de déploiements des services de streaming, les travaux de cette thèse de doctorat proposent de faire évoluer de façon pragmatique les solutions de streaming adaptatives actuelles vers l’utilisation en simultané de plusieurs sources (serveurs ou pairs). La première contribution de cette thèse présente MS-Stream, une technique évolutive de streaming adaptatif basé sur HTTP et utilisant plusieurs serveurs simultanément. MS-Stream offre la possibilité d’exploiter la bande passante disponible dans les infrastructures distribuées et les réseaux hétérogènes. La deuxieme contribution de ce document est MATHIAS, un groupe d’algorithmes d’adaptation centrés client, implémentés dans MS-Stream, qui a pour vocation d’optimiser l’utilisation des ressources réseau hétérogènes mises à disposition du client pour obtenir une qualité vidéo cible. MATHIAS permet à chaque client de controller le nombre de serveur utilisé en simultané, de faire face à l’hétérogeneité des resources disponibles, de réagir aux fluctuations soudaines et non-anticipées des capacités des serveurs tout en donnant à l’utilisateur une experience de streaming ininterrompu. Pour finir, nous allons plus loin dans les capacités de scalabilité et de qualité d’experience de MS-Stream et MATHIAS en tirant profit des ressources physiques des consommateurs. Nous proposons une solution hybride pair-à-pair/multi-server de streaming adaptative: PMS. Au sein de PMS, les logiques d’adaptation de la qualité vidéo et de la scalabilité sont distribuées pour permettre à chaque client de tendre vers une utilisation optimale de l’infrastructure de streaming. / Single-source HTTP Adaptive Streaming solutions (HAS) have become the de-facto solutions to deliver video over the Internet mostly due to their capabilities to increase end-user’s Quality of Experience (QoE) as well as their ease of deployment due to the usage of the HTTP protocol. Although HAS solutions can increase QoE by trading off the delivered video quality to minimize the number of video freezing events, they are limited by the bandwidth available on the considered communication channel between the client and the server. This thesis exposes our contributions in building lightweight pragmatic and evolving solutions advocating for the simultaneous usage of multiple sources with heterogeneous capacities so as to achieve high QoE content delivery at low cost. The first contribution of this work presents a streaming solution extending HAS capabilities to a pragmatic multi-server technique: MS-Stream. MS-Stream provides the means to exploit expanded bandwidth and link diversity in distributed heterogeneous network infrastructures. In our second contribution, we propose MATHIAS, a client-side two-phase consumption and adaptation algorithm implemented into MSStream. MATHIAS aims at increasing the end-user’s perceived streaming quality while utilizing the most of the heterogeneous capacities offered at the service and network environments. Finally, we further extend the QoE and scalability capabilities of MS-Stream and MATHIAS by leveraging on clients’ connectivity capacities and we expose our third contribution: a hybrid P2P/Multi-server live-Streaming system (PMS) incorporating distributed quality and scalability adaptation mechanisms.

Off-highway vehicles are on the verge of switching from analog to digital video camera technology (VCT), which offers better video quality and new features but adds complexity to the system. This thesis project aims to implement the digital VCT to the display computer CCpilot VA intended for off-highway vehicles. In this project the differences between analog and digital VCTs were reviewed and then a demo displaying a live digital camera video feed on the embedded Linux based display computer CCpilot VA was implemented with Qt and QML. More specifically, different GStreamer pipelines were tested, as Qt uses GStreamer to play video, and camera settings were changed using the ISO 17215 standard.  The demo displayed a live digital camera video feed with high quality, low latency and high frame rate on the VA by using a GStreamer pipeline utilizing hardware decoding. The results have shown that digital video cameras perform better than analog cameras, primarily because digital cameras have better video quality. The attempts to simultaneously display a video feed and a Graphical User Interface created by Qt have been made. However, they were only successful with poor video performance. A zero-copy link between the GStreamer pipeline’s decoder and sink element must be used to obtain good video performance.

The beginning of the thesis deals with history of interconnection between video and the internet from its origins to the present. The thesis has two main aims. The first aim is to provide an overview of main solutions how to provide stream video over the internet. In the thesis there are described three different solutions. The first solution is to buy a complex media server. The second one is to hire a special service which can provide video storage and its distribution over the internet. The service must allow integration to various information systems and web pages or web applications. The last part of the first aim speculates about possibilities how to solve the stream video distribution over the internet in a business and commercial domain without any expenses to provide the functionality. The second aim is a practical realization of a complete project. The content of the project is a solution of video conversion, storage and distribution over the internet in a real company. The main business activity of the company is the development and providing of web pages and applications on its own servers. At the beginning of the second part of the thesis there are evaluated possible ways how to solve the goal of the project. The possible ways are solutions that are mentioned in the first part of the thesis plus development of new a solution by the company itself. The last alternative has proved to be optimal from the economic point of view. The result of the project is a web service which is completely platform independent. So it is possible to use the web service in company's content management systems that are developed in PHP or .Net. In the thesis there is also mentioned an issue of mechanism called HTTP Pseudo Streaming.

The efficient scheduling of streaming data delivery in a peer-to-peer (P2P) network is a hard problem due to the Internet’s lack of support for resource allocation and performance guarantees. In particular, the bandwidth resources available to a peer is constantly in flux and the future bandwidth availability is very difficult, if not impossible, to predict accurately. This thesis proposes to tackle this problem from a different angle. We investigate the use of erasure codes to encode the media data and then schedule multiple peers to stream the encoded data simultaneously to a receiver. By exploiting the order-invariant property of erasure codes this approach enables the sending peers to fully utilize their available bandwidth resources and yet does not need to estimate or predict their bandwidth availability. Moreover, we develop distributed scheduling algorithms to juxtapose the data transmissions from multiple peers so that the coding and storage complexities can be kept at practical level in scaling up the system. / 在互聯網中變動的可用頻寬和網路的延遲變異等問題，對點對點網絡上的串流傳輸品質造成嚴重的影響。由於每一個用戶端的網絡傳輸速度都是不穩定，而且是難以預測的，這是很困難去制定一個高效率的傳輸排程。在本論文中，我們從另一個角度去解決這個問題。首先，利用抹除碼把訊息變成足夠數量的區塊，讓多個訊息源可以同時傳送不同部分的區塊給受信者。由於抹除碼擁有糾錯能力，訊息源之間便不再需要任何協調及傳輸排程，提升整體的傳輸速度。另外，我們開發了一個排程算法來演算每一個訊息源的區塊傳送次序，盡可能的節省計算抹除碼時所需的資源，以能夠在具規模的環境中運作。 / Ma, Man Lok. / Thesis (Ph.D.)--Chinese University of Hong Kong, 2014. / Includes bibliographical references (leaves 72-75). / Abstracts also in Chinese. / Title from PDF title page (viewed on 26, September, 2016). / Detailed summary in vernacular field only.

The explosive growth in on-demand access of video across all forms of delivery (Internet, traditional cable, IPTV, wireless) has renewed the interest in scalable delivery methods. Approaches using Content Delivery Networks (CDNs), Peer-to-Peer (P2P) approaches, and their combinations have been proposed as viable options to ease the load on servers and network links. However, there has been little focus on how to take advantage of user viewing patterns to understand their impact on existing mechanisms and to design new solutions that improve the streaming service quality. In this dissertation, we leverage on the observation that users watch only a small portion of videos to understand the limits of existing designs and to optimize two scalable approaches -- the content placement and P2P Video-on-Demand (VoD) streaming. Then, we present our novel scalable system called Joint-Family which enables adaptive bitrate streaming (ABR) in P2P VoD, supporting user viewing patterns. We first provide evidence of such user viewing behavior from data collected from a nationally deployed VoD service. In contrast to using a simplistic popularity-based placement and traditionally proposed caching strategies (such as CDNs), we use a Mixed Integer Programming formulation to model the placement problem and employ an innovative approach that scales well. We have performed detailed simulations using actual traces of user viewing sessions (including stream control operations such as pause, fast-forward, and rewind). Our results show that the use of segment-based placement strategy yields substantial savings in both disk storage requirements at origin servers/VHOs as well as network bandwidth use. For example, compared to a simple caching scheme using full videos, our MIP-based placement using segments can achieve up to 71% reduction in peak link bandwidth usage. Secondly, we note that the policies adopted in existing P2P VoD systems have not taken user viewing behavior -- that users abandon videos -- into account. We show that abandonment can result in increased interruptions and wasted resources. As a result, we reconsider the set of policies to use in the presence of abandonment. Our goal is to balance the conflicting needs of delivering videos without interruptions while minimizing wastage. We find that an Earliest-First chunk selection policy in conjunction with the Earliest-Deadline peer selection policy allows us to achieve high download rates. We take advantage of abandonment by converting peers to "partial seeds"; this increases capacity. We minimize wastage by using a playback lookahead window. We use analysis and simulation experiments using real-world traces to show the effectiveness of our approach. Finally, we propose Joint-Family, a protocol that combines P2P and adaptive bitrate (ABR) streaming for VoD. While P2P for VoD and ABR have been proposed previously, they have not been studied together because they attempt to tackle problems with seemingly orthogonal goals. We motivate our approach through analysis that overcomes a misconception resulting from prior analytical work, and show that the popularity of a P2P swarm and seed staying time has a significant bearing on the achievable per-receiver download rate. Specifically, our analysis shows that popularity affects swarm efficiency when seeds stay "long enough". We also show that ABR in a P2P setting helps viewers achieve higher playback rates and/or fewer interruptions. We develop the Joint-Family protocol based on the observations from our analysis. Peers in Joint-Family simultaneously participate in multiple swarms to exchange chunks of different bitrates. We adopt chunk, bitrate, and peer selection policies that minimize occurrence of interruptions while delivering high quality video and improving the efficiency of the system. Using traces from a large-scale commercial VoD service, we compare Joint-Family with existing approaches for P2P VoD and show that viewers in Joint-Family enjoy higher playback rates with minimal interruption, irrespective of video popularity.

Today's popular over-the-top (OTT) video streaming services such as YouTube, Netflix and Hulu deliver video contents to viewers using adaptive bitrate (ABR) technologies. In ABR streaming, a video player running on a viewer's device adaptively changes bitrates to match given network conditions. However, providing reliable streaming is challenging. First, an ABR player may select an inappropriate bitrate during playback due to the lack of direct knowledge of access networks, frequent user mobility and rapidly changing channel conditions. Second, OTT content is delivered to viewers without any cooperation with Internet service providers (ISPs). Last, there are no appropriate tools that evaluate the performance of ABR streaming along with video quality of experience (QoE). This thesis describes how to improve the video QoE of OTT video streaming services using ABR technologies. Our analysis starts from understanding ABR heuristics. How does ABR streaming work? What factors does an ABR player consider when switching bitrates during a download? Then, we propose our solutions to improve existing ABR streaming from the perspective of network operators who deliver video content through their networks and video service providers who build ABR players running on viewers' devices. From the network operators' point of view, we propose to find a better video content server based on round trip times (RTTs) between an edge node of a wireless network and available video content servers when a viewer requests a video. The edge node can be an Internet Service Provider (ISP) router in a Wi-Fi network and a packet data network gateway (P-GW) in a 4G network. During the experiments, our solution showed better TCP performance (e.g., higher TCP throughput during playback) 146 times out of 200 experiments (73%) over Wi-Fi networks and 162 times out of 200 experiments (81%) over 3G networks. In addition, we claim that the wireless edge nodes can assist an ABR video player in selecting the best available bitrate by controlling the available bandwidth in the radio access network between a base station and a viewer's device. In our Wi-Fi testbed, the proposed solution saved up to 21% of radio bandwidth on mobile devices and enhanced the viewing experience by reducing rebufferings during playback. Last, we assert that software-defined networking (SDN) can improve video QoE by dynamically controlling routing paths of video streaming flows based on the provisioned networking information collected from SDN-enabled networking devices. Using an off-the-shelf SDN platform, we showed that our proposed solution can reduce rebufferings by 50% and provide higher bitrates during a download. From the perspective of video service providers, higher video QoE can be achieved by improving ABR heuristics implemented in an ABR player. To support this idea, we investigated the role of playout buffer size in ABR streaming and its impact on video QoE. Through our video QoE survey, we proved that a large buffer does not always outperform a small buffer, especially under rapidly varying network conditions. Based on this finding, we suggest to dynamically change the maximum buffer size in an ABR player depending on the current capacity of its playout buffer for improving the QoE of viewers. During the experiments, our proposed solution improved the viewing experience by offering 15% higher average played bitrate, 70% fewer bitrate changes and 50% shorter rebuffering duration. Our experimental results show that even small changes of ABR heuristics and new features of network systems can greatly affect video QoE. However, it is still difficult for video service providers or network operators to evaluate new ABR heuristics or network system changes due to lack of accurate QoE monitoring systems. In order to solve this issue, we have developed YouSlow ("YouTube Too Slow!? - YouSlow") as a new approach to monitoring video QoE for the analysis of ABR performance. The lightweight web browser plug-in and mobile application are designed to monitor various playback events (e.g., rebuffering duration and frequency of bitrate changes) directly from within ABR video players and calculate statistics along with video QoE. Using YouSlow, we investigate the impact of the above playback events on video abandonment: about 10% of viewers abandoned the YouTube videos when the pre-roll ads lasted for 15 seconds. Even increasing the bitrate can annoy viewers; they prefer a high starting bitrate with no bitrate changes during playback. Our regression analysis shows that bitrate changes do not affect video abandonment significantly and the abandonment rate can be estimated accurately using the rebuffering ratio and the number of rebufferings. The thesis includes four main contributions. First, we investigate today's popular OTT video streaming services (e.g., YouTube and Netflix) that use ABR streaming technologies. Second, we propose to build QoS and QoE aware video streaming that can be implemented in existing wireless networks (e.g., Wi-Fi, 3G and 4G) and in SDN-enabled networks. Third, we propose to improve current ABR heuristics by dynamically changing the playout buffer size under varying network conditions. Last, we designed and implemented a new monitoring system for measuring video QoE.

In the first part, decoder based error concealment methods are discussed. An adaptive partition size (APS) temporal error concealment method is developed for H.264. We propose to use Weighted Double-Sided External Boundary Matching Error (WDS-EBME) to jointly measure the inter-MB boundary discontinuity, inter-partition boundary discontinuity and intra-partition block artifacts in the corrupted MB. By minimizing the WDS-EBME value of each partition, the best motion vectors of each candidate partition mode can be estimated, overall WDS-EBME of the MB concealed by each partition mode can then be evaluated and the best partition mode for the corrupted Macroblocks (MB) will be determined as the one with the smallest overall WDS-EBME. We also propose a progressive concealment order for the 4x4 partition mode. / In this thesis, techniques for efficient error resilient video coding are investigated. Three parts of the work are discussed in this thesis. / The last part of the thesis concerns the joint encoder-decoder error control method. A joint temporal error control method is proposed for H.264. It combines RDO-based macroblock (MB) classification at the encoder and adaptive partition size error concealment at the decoder. The encoder classifies the MBs by evaluating the sensitivity of the MBs as the RD cost between the concealment error and the bits needed for the additional motion information. Additional motion information such as the original motion vector or motion vector index can be transmitted for the error sensitive MBs. The decoder utilizes the additional motion information if any of these MBs get lost. Non-sensitive MBs and blocks are concealed by the APS method. / The second part of this thesis investigates encoder based error control techniques. Firstly, a VLC/FLC data partitioning method is proposed for MPEG-4. It disables intra AC prediction and groups appropriate fixed length coded (FLC) syntaxes in a video packet (or slice) together to form a new partition. With intra AC prediction disabled, errors occurring in these FLC syntaxes will not cause spatial error propagation. It essentially classifies the syntaxes into two categories according to whether that syntax will cause spatial error propagation when an error occurs. Secondly, a redundant macroblock strategy is proposed for H.264. MB Differential Mean Square Error (DMSE) is employed to evaluate the error sensitivity of MBs. The most sensitive MBs are transmitted separately in additional slices while coarsely quantized copies of the MBs are placed in the original slice. When working with chessboard style Flexible Macroblock reordering (FMO) and fixed length slice mode (FMO-slicing), the scheme performs well against packet loss errors with acceptable overhead and it is highly compatible with original H.264 bitstream. Thirdly, a joint optimal bit allocation and rate control scheme is proposed for H.264 with redundant slice. The optimum ratio between each primary and redundant picture pair is analytically deduced. Rate function and distortion model for both representations are developed, and a simple close-form solution is provided to achieve joint optimum bit allocation. / Video communication and other web-based video applications become popular in recent years. However, the transmission of the compressed video bit stream often suffers from imperfection of the communication channel, like path loss, multipath fading, co-channel interference, congestion, etc. Error resilient video coding techniques need to be employed to mitigate the channel errors, which include error concealment in the decoder, forward error correction in the encoder and joint encoder-decoder error control techniques. / Li, Jie. / Adviser: Ngan King Ngi. / Source: Dissertation Abstracts International, Volume: 73-01, Section: B, page: . / Thesis (Ph.D.)--Chinese University of Hong Kong, 2009. / Includes bibliographical references (leaves 137-146). / Electronic reproduction. Hong Kong : Chinese University of Hong Kong, [2012] System requirements: Adobe Acrobat Reader. Available via World Wide Web. / Electronic reproduction. [Ann Arbor, MI] : ProQuest Information and Learning, [201-] System requirements: Adobe Acrobat Reader. Available via World Wide Web. / Abstract also in Chinese.

Large classes that allow smaller amounts of instructor-student interaction have become more common in today's colleges. The best way to provide needed opportunities for students to overcome this lack of interaction with instructors remains unidentified. This research evaluated the use of video lecture capture (VLC) as a supplemental method for teacher-student interaction and what, if any, impact it and attendance have on student performance in large lecture courses. This ex post facto study conducted at a Northeastern research university utilized cognitive and andragogical frameworks to examine the relationships between the independent variables frequency of video viewing, quantity of videos viewed, and course attendance, as well as their impact on course performance in a large lecture course (N=329). Data sources included archival data from the learning management system and student survey responses. Analysis included a series of two-way ANOVA tests. The results indicated that the frequency of video viewing was found to have a significant positive effect on course performance (F = 3.018, p = .030). The number of VLC videos not viewed was also found to have a significant negative effect on course performance (F = 1.875, p = 0.016). Other independent variables were not found to have any significant main effect or interaction effect with the dependent variable, course performance. Findings from this research may be used by educators, students, and administrators planning course sizes and availability to better understand the relationship between these variables and how VLC can be used effectively in large lecture classes thus leading to improved efficacy in VLC use.

Voice over IP and video applications continue to increase the amount of traffic over the Internet. These applications utilize the UDP protocol because TCP is not suitable for streaming applications. The flow and congestion control mechanisms of TCP can change the connection transmission rate too drastically, affecting the user-perceived quality of the transmission. Also, the TCP protocol provides a level of reliability that may waste network resources, retransmitting packets that have no value. On the other hand, the use of end-to-end flow and congestion control mechanisms for streaming applications has been acknowledged as an important measure to ease or eliminate the unfairness problem that exist when TCP and UDP share the same congested bottleneck link. Actually, router-based and end-to-end solutions have been proposed to solve this problem. This thesis introduces a new end-to-end protocol based on TCP SACK called SF-SACK that promises to be smooth enough for streaming applications while implementing the known flow and congestion control mechanisms available in TCP. Through simulations, it is shown that in terms of smoothness the SF-SACK protocol is considerably better than TCP SACK and only slightly worse than TFRC. Regarding friendliness, SF-SACK is not completely fair to TCP but considerably fairer than UDP. Furthermore, if SF-SACK is used by both streaming and data-oriented applications, complete fairness is achieved. In addition, SF-SACK only needs sender side modifcations and it is simpler than TFRC.