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Multimedia delivery in the future internet
The term âNetworked Mediaâ 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 citizensâ 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 âon the moveâ, 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
MIJ2K: Enhanced video transmission based on conditional replenishment of JPEG2000 tiles with motion compensation
A video compressed as a sequence of JPEG2000 images can achieve the scalability, flexibility, and accessibility that is lacking in current predictive motion-compensated video coding standards. However, streaming JPEG2000-based sequences would consume considerably more bandwidth. With the aim of solving this problem, this paper describes a new patent pending method, called MIJ2K. MIJ2K reduces the inter-frame redundancy present in common JPEG2000 sequences (also called MJP2). We apply a real-time motion detection system to perform conditional tile replenishment. This will significantly reduce the bit rate necessary to transmit JPEG2000 video sequences, also improving their quality. The MIJ2K technique can be used both to improve JPEG2000-based real-time video streaming services or as a new codec for video storage. MIJ2K relies on a fast motion compensation technique, especially designed for real-time video streaming purposes. In particular, we propose transmitting only the tiles that change in each JPEG2000 frame. This paper describes and evaluates the method proposed for real-time tile change detection, as well as the overall MIJ2K architecture. We compare MIJ2K against other intra-frame codecs, like standard Motion JPEG2000, Motion JPEG, and the latest H.264-Intra, comparing performance in terms of compression ratio and video quality, measured by standard peak signal-to-noise ratio, structural similarity and visual quality metric metrics.This work was supported in part by Projects CICYT TIN2008â
06742-C02â02/TSI, CICYT TEC2008â06732-C02â02/TEC, SINPROB,
CAM MADRINET S-0505/TIC/0255 and DPS2008â07029-C02â02.Publicad
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
lâimplementazione 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 compatibilitĂ allâindietro con lâMPEG-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 dallâATSC, ed è lo standard di codifica nellâIPTV.
Lâobiettivo 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, LocalitĂ Piscinamanna 09010 Pula (CA) - ItaliaSeminario sullo Standard MPEG-4: utilizzo ed aspetti implementativ
Error resilient packet switched H.264 video telephony over third generation networks.
Real-time video communication over wireless networks is a challenging problem because
wireless channels suffer from fading, additive noise and interference, which translate
into packet loss and delay. Since modern video encoders deliver video packets with
decoding dependencies, packet loss and delay can significantly degrade the video quality
at the receiver. Many error resilience mechanisms have been proposed to combat packet
loss in wireless networks, but only a few were specifically designed for packet switched
video telephony over Third Generation (3G) networks.
The first part of the thesis presents an error resilience technique for packet switched
video telephony that combines application layer Forward Error Correction (FEC) with
rateless codes, Reference Picture Selection (RPS) and cross layer optimization. Rateless
codes have lower encoding and decoding computational complexity compared to traditional
error correcting codes. One can use them on complexity constrained hand-held
devices. Also, their redundancy does not need to be fixed in advance and any number of
encoded symbols can be generated on the fly. Reference picture selection is used to limit
the effect of spatio-temporal error propagation. Limiting the effect of spatio-temporal
error propagation results in better video quality. Cross layer optimization is used to
minimize the data loss at the application layer when data is lost at the data link layer.
Experimental results on a High Speed Packet Access (HSPA) network simulator for
H.264 compressed standard video sequences show that the proposed technique achieves
significant Peak Signal to Noise Ratio (PSNR) and Percentage Degraded Video Duration
(PDVD) improvements over a state of the art error resilience technique known as
Interactive Error Control (IEC), which is a combination of Error Tracking and feedback
based Reference Picture Selection. The improvement is obtained at a cost of higher
end-to-end delay.
The proposed technique is improved by making the FEC (Rateless code) redundancy
channel adaptive. Automatic Repeat Request (ARQ) is used to adjust the redundancy
of the Rateless codes according to the channel conditions. Experimental results show
that the channel adaptive scheme achieves significant PSNR and PDVD improvements
over the static scheme for a simulated Long Term Evolution (LTE) network.
In the third part of the thesis, the performance of the previous two schemes is
improved by making the transmitter predict when rateless decoding will fail. In this
case, reference picture selection is invoked early and transmission of encoded symbols
for that source block is aborted. Simulations for an LTE network show that this results
in video quality improvement and bandwidth savings.
In the last part of the thesis, the performance of the adaptive technique is improved
by exploiting the history of the wireless channel. In a Rayleigh fading wireless channel,
the RLC-PDU losses are correlated under certain conditions. This correlation is
exploited to adjust the redundancy of the Rateless code and results in higher Rateless
code decoding success rate and higher video quality. Simulations for an LTE network
show that the improvement was significant when the packet loss rate in the two wireless
links was 10%.
To facilitate the implementation of the proposed error resilience techniques in practical
scenarios, RTP/UDP/IP level packetization schemes are also proposed for each
error resilience technique.
Compared to existing work, the proposed error resilience techniques provide better
video quality. Also, more emphasis is given to implementation issues in 3G networks
Viewpoint switching in multiview videos using SP-frames
Centre for Signal Processing, Department of Electronic and Information EngineeringRefereed conference paper2008-2009 > Academic research: refereed > Refereed conference paperVersion of RecordPublishe
Description-driven Adaptation of Media Resources
The current multimedia landscape is characterized by a significant diversity in terms of available media formats, network technologies, and device properties. This heterogeneity has resulted in a number of new challenges, such as providing universal access to multimedia content. A solution for this diversity is the use of scalable bit streams, as well as the deployment of a complementary system that is capable of adapting scalable bit streams to the constraints imposed by a particular usage environment (e.g., the limited screen resolution of a mobile device). This dissertation investigates the use of an XML-driven (Extensible Markup Language) framework for the format-independent adaptation of scalable bit streams. Using this approach, the structure of a bit stream is first translated into an XML description. In a next step, the resulting XML description is transformed to reflect a desired adaptation of the bit stream. Finally, the transformed XML description is used to create an adapted bit stream that is suited for playback in the targeted usage environment. The main contribution of this dissertation is BFlavor, a new tool for exposing the syntax of binary media resources as an XML description. Its development was inspired by two other technologies, i.e. MPEG-21 BSDL (Bitstream Syntax Description Language) and XFlavor (Formal Language for Audio-Visual Object Representation, extended with XML features). Although created from a different point of view, both languages offer solutions for translating the syntax of a media resource into an XML representation for further processing. BFlavor (BSDL+XFlavor) harmonizes the two technologies by combining their strengths and eliminating their weaknesses. The expressive power and performance of a BFlavor-based content adaptation chain, compared to tool chains entirely based on either BSDL or XFlavor, were investigated by several experiments. One series of experiments targeted the exploitation of multi-layered temporal scalability in H.264/AVC, paying particular attention to the use of sub-sequences and hierarchical coding patterns, as well as to the use of metadata messages to communicate the bit stream structure to the adaptation logic. BFlavor was the only tool to offer an elegant and practical solution for XML-driven adaptation of H.264/AVC bit streams in the temporal domain
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