1,349 research outputs found
Adaptive filtering of MPEG system streams in IP networks
Congestion and large differences in available link bandwidth create challenges for the design of applications that want to deliver high quality video over the Internet. We present an efficient adaptive filter for MPEG System streams that can be placed in the network (e.g., as an active service). This filter adjusts the bandwidth demands of an MPEG System stream to the available bandwidth without transcoding while maintaining synchronization between the streams embedded in the MPEG System. The filter is network-friendly: it is fair with respect to other (TCP) competing streams and it avoids generating bursty traffic. This paper presents the system architecture and an evaluation of our implementation in three different operating environments: a networking testbed in a laboratory environment, a home-user scenario (DSL line with 640Kbit/s), and a wide area network covering the Atlantic (server in Europe, client in the US). Moreover we examine the network-friendliness of the adaptation protocol and the relationship between the quality of the received continuous media and the protocol's aggressiveness. Our architecture is based on efficient MPEG System filtering to achieve high-quality video over best-effort network
Design issues for the Generic Stream Encapsulation (GSE) of IP datagrams over DVB-S2
The DVB-S2 standard has brought an unprecedented degree of novelty and flexibility in the way IP datagrams or other network level packets can be transmitted over DVB satellite links, with the introduction of an IP-friendly link layer - he continuous Generic Streams - and the adaptive combination of advanced error coding, modulation and spectrum management techniques. Recently approved by the DVB, the Generic Stream Encapsulation (GSE) used for carrying IP datagrams over DVBS2 implements solutions stemmed from a design rationale quite different from the one behind IP encapsulation schemes over its predecessor DVB-S. This paper highlights GSE's original design choices under the perspective of DVB-S2's innovative features and possibilities
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Multimedia delivery in the future internet
The term “Networked Media” implies that all kinds of media including text, image, 3D graphics, audio
and video are produced, distributed, shared, managed and consumed on-line through various networks,
like the Internet, Fiber, WiFi, WiMAX, GPRS, 3G and so on, in a convergent manner [1]. This white
paper is the contribution of the Media Delivery Platform (MDP) cluster and aims to cover the Networked
challenges of the Networked Media in the transition to the Future of the Internet.
Internet has evolved and changed the way we work and live. End users of the Internet have been confronted
with a bewildering range of media, services and applications and of technological innovations concerning
media formats, wireless networks, terminal types and capabilities. And there is little evidence that the pace
of this innovation is slowing. Today, over one billion of users access the Internet on regular basis, more
than 100 million users have downloaded at least one (multi)media file and over 47 millions of them do so
regularly, searching in more than 160 Exabytes1 of content. In the near future these numbers are expected
to exponentially rise. It is expected that the Internet content will be increased by at least a factor of 6, rising
to more than 990 Exabytes before 2012, fuelled mainly by the users themselves. Moreover, it is envisaged
that in a near- to mid-term future, the Internet will provide the means to share and distribute (new)
multimedia content and services with superior quality and striking flexibility, in a trusted and personalized
way, improving citizens’ quality of life, working conditions, edutainment and safety.
In this evolving environment, new transport protocols, new multimedia encoding schemes, cross-layer inthe
network adaptation, machine-to-machine communication (including RFIDs), rich 3D content as well as
community networks and the use of peer-to-peer (P2P) overlays are expected to generate new models of
interaction and cooperation, and be able to support enhanced perceived quality-of-experience (PQoE) and
innovative applications “on the move”, like virtual collaboration environments, personalised services/
media, virtual sport groups, on-line gaming, edutainment. In this context, the interaction with content
combined with interactive/multimedia search capabilities across distributed repositories, opportunistic P2P
networks and the dynamic adaptation to the characteristics of diverse mobile terminals are expected to
contribute towards such a vision.
Based on work that has taken place in a number of EC co-funded projects, in Framework Program 6 (FP6)
and Framework Program 7 (FP7), a group of experts and technology visionaries have voluntarily
contributed in this white paper aiming to describe the status, the state-of-the art, the challenges and the way
ahead in the area of Content Aware media delivery platforms
Seminario sullo Standard MPEG-4: utilizzo ed aspetti implementativi
Una delle tecnologie chiave che hanno permesso il grande sviluppo della televisione digitale è la compressione video. La tecnologia di codifica video nota come MPEG-2, sviluppata nei primi anni novanta, è diventata lo standard di trasmissione DTV (Digital TV) sia satellitare sia terrestre in quasi tutti i paesi del mondo. Da allora la velocità dei microprocessori e le capacità di memoria dei dispositivi hardware per la codifica e la decodifica sono migliorate significativamente rendendo possibile lo sviluppo e
l’implementazione di algoritmi di codifica innovativi in grado di abbattere significativamente i limiti di compressione dello standard MPEG-2. Tali innovazioni, sfociate nel 2003 nello standard MPEG-4 AVC (Advanced Video Coding), non hanno permesso di mantenere la compatibilità all’indietro con l’MPEG-2, e questo ha inizialmente costituito un limite alla loro introduzione nei sistemi di trasmissione DTV. Tuttavia, negli ultimi anni la codifica MPEG-4 AVC si è diffusa rapidamente, è stata adottata dal progetto DVB, recentemente dall’ATSC, ed è lo standard di codifica nell’IPTV.
L’obiettivo di questo seminario, che si articola in due giornate, è quello di presentare lo standard di codifica MPEG-4 AVC con particolare attenzione agli aspetti implementativi del livello di codifica video.2008-11-18Sardegna Ricerche, Edificio 2, Località Piscinamanna 09010 Pula (CA) - ItaliaSeminario sullo Standard MPEG-4: utilizzo ed aspetti implementativ
COSMOS-7: Video-oriented MPEG-7 scheme for modelling and filtering of semantic content
MPEG-7 prescribes a format for semantic content models for multimedia to ensure interoperability across a multitude of platforms and application domains. However, the standard leaves it open as to how the models should be used and how their content should be filtered. Filtering is a technique used to retrieve only content relevant to user requirements, thereby reducing the necessary content-sifting effort of the user. This paper proposes an MPEG-7 scheme that can be deployed for semantic content modelling and filtering of digital video. The proposed scheme, COSMOS-7, produces rich and multi-faceted semantic content models and supports a content-based filtering approach that only analyses content relating directly to the preferred content requirements of the user
Towards a new generation of transport services adapted to multimedia application
Une connexion d'ordre et de fiabilité partiels (POC, partial order connection) est une connexion de transport autorisée à perdre certains objets mais également à les délivrer dans un ordre éventuellement différent de celui d'émission. L'approche POC établit un lien conceptuel entre les protocoles sans connexion au mieux et les protocoles fiables avec connexion. Le concept de POC est motivé par le fait que dans les réseaux hétérogènes sans connexion tels qu'Internet, les paquets transmis sont susceptibles de se perdre et d'arriver en désordre, entraînant alors une réduction des performances des protocoles usuels. De plus, on montre qu'un protocole associé au transport d'un flux multimédia permet une réduction très sensible de l'utilisation des ressources de communication et de mémorisation ainsi qu'une diminution du temps de transit moyen. Dans cet article, une extension temporelle de POC, nommée TPOC (POC temporisé), est introduite. Elle constitue un cadre conceptuel permettant la prise en compte des exigences de qualité de service des applications multimédias réparties. Une architecture offrant un service TPOC est également introduite et évaluée dans le cadre du transport de vidéo MPEG. Il est ainsi démontré que les connexions POC comblent, non seulement le fossé conceptuel entre les protocoles sans connexion et avec connexion, mais aussi qu'ils surpassent les performances des ces derniers lorsque des données multimédias (telles que la vidéo MPEG) sont transportées
Transport of video over partial order connections
A Partial Order and partial reliable Connection (POC) is an end-to-end transport connection authorized to deliver objects in an order that can differ from the transmitted one. Such a connection is also authorized to lose some objects. The POC concept is motivated by the fact that heterogeneous best-effort networks such as Internet are plagued by unordered delivery of packets and losses, which tax the performances of current applications and protocols. It has been shown, in several research works, that out of order delivery is able to alleviate (with respect to CO service) the use of end systems’ communication resources. In this paper, the efficiency of out-of-sequence delivery on MPEG video streams processing is studied. Firstly, the transport constraints (in terms of order and reliability) that can be relaxed by MPEG video decoders, for improving video transport, are detailed. Then, we analyze the performance gain induced by this approach in terms of blocking times and recovered errors. We demonstrate that POC connections fill not only the conceptual gap between TCP and UDP but also provide real performance improvements for the transport of multimedia streams such MPEG video
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Scalable and network aware video coding for advanced communications over heterogeneous networks
This thesis was submitted for the degree of Doctor of Philosophy and was awarded by Brunel UniversityThis work addresses the issues concerned with the provision of scalable video services over heterogeneous networks particularly with regards to dynamic adaptation and user’s acceptable quality of service.
In order to provide and sustain an adaptive and network friendly multimedia communication service, a suite of techniques that achieved automatic scalability and adaptation are developed. These techniques are evaluated objectively and subjectively to assess the Quality of Service (QoS) provided to diverse users with variable constraints and dynamic resources. The research ensured the consideration of various levels of user acceptable QoS The techniques are further evaluated with view to establish their performance against state of the art scalable and non-scalable techniques.
To further improve the adaptability of the designed techniques, several experiments and real time simulations are conducted with the aim of determining the optimum performance with various coding parameters and scenarios. The coding parameters and scenarios are evaluated and analyzed to determine their performance using various types of video content and formats. Several algorithms are developed to provide a dynamic adaptation of coding tools and parameters to specific video content type, format and bandwidth of transmission.
Due to the nature of heterogeneous networks where channel conditions, terminals, users capabilities and preferences etc are unpredictably changing, hence limiting the adaptability of a specific technique adopted, a Dynamic Scalability Decision Making Algorithm (SADMA) is developed. The algorithm autonomously selects one of the designed scalability techniques basing its decision on the monitored and reported channel conditions. Experiments were conducted using a purpose-built heterogeneous network simulator and the network-aware selection of the scalability techniques is based on real time simulation results. A technique with a minimum delay, low bit-rate, low frame rate and low quality is adopted as a reactive measure to a predicted bad channel condition. If the use of the techniques is not favoured due to deteriorating channel conditions reported, a reduced layered stream or base layer is used. If the network status does not allow the use of the base layer, then the stream uses parameter identifiers with high efficiency to improve the scalability and adaptation of the video service.
To further improve the flexibility and efficiency of the algorithm, a dynamic de-blocking filter and lambda value selection are analyzed and introduced in the algorithm. Various methods, interfaces and algorithms are defined for transcoding from one technique to another and extracting sub-streams when the network conditions do not allow for the transmission of the entire bit-stream
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