131 research outputs found
Adaptive network abstraction layer packetization for low bit rate H.264/AVC video transmission over wireless mobile networks under cross layer optimization
Master'sMASTER OF ENGINEERIN
Error and Congestion Resilient Video Streaming over Broadband Wireless
In this paper, error resilience is achieved by adaptive, application-layer rateless channel coding, which is used to protect H.264/Advanced Video Coding (AVC) codec data-partitioned videos. A packetization strategy is an effective tool to control error rates and, in the paper, source-coded data partitioning serves to allocate smaller packets to more important compressed video data. The scheme for doing this is applied to real-time streaming across a broadband wireless link. The advantages of rateless code rate adaptivity are then demonstrated in the paper. Because the data partitions of a video slice are each assigned to different network packets, in congestion-prone wireless networks the increased number of packets per slice and their size disparity may increase the packet loss rate from buffer overflows. As a form of congestion resilience, this paper recommends packet-size dependent scheduling as a relatively simple way of alleviating the buffer-overflow problem arising from data-partitioned packets. The paper also contributes an analysis of data partitioning and packet sizes as a prelude to considering scheduling regimes. The combination of adaptive channel coding and prioritized packetization for error resilience with packet-size dependent packet scheduling results in a robust streaming scheme specialized for broadband wireless and real-time streaming applications such as video conferencing, video telephony, and telemedicine
Recommended from our members
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
Survey of Transportation of Adaptive Multimedia Streaming service in Internet
[DE] World Wide Web is the greatest boon towards the technological advancement of modern era. Using the benefits of Internet globally, anywhere and anytime, users can avail the benefits of accessing live and on demand video services. The streaming media systems such as YouTube, Netflix, and Apple Music are reining the multimedia world with frequent popularity among users. A key concern of quality perceived for video streaming applications over Internet is the Quality of Experience (QoE) that users go through. Due to changing network conditions, bit rate and initial delay and the multimedia file freezes or provide poor video quality to the end users, researchers across industry and academia are explored HTTP Adaptive Streaming (HAS), which split the video content into multiple segments and offer the clients at varying qualities. The video player at the client side plays a vital role in buffer management and choosing the appropriate bit rate for each such segment of video to be transmitted. A higher bit rate transmitted video pauses in between whereas, a lower bit rate video lacks in quality, requiring a tradeoff between them. The need of the hour was to adaptively varying the bit rate and video quality to match the transmission media conditions. Further, The main aim of this paper is to give an overview on the state of the art HAS techniques across multimedia and networking domains. A detailed survey was conducted to analyze challenges and solutions in adaptive streaming algorithms, QoE, network protocols, buffering and etc. It also focuses on various challenges on QoE influence factors in a fluctuating network condition, which are often ignored in present HAS methodologies. Furthermore, this survey will enable network and multimedia researchers a fair amount of understanding about the latest happenings of adaptive streaming and the necessary improvements that can be incorporated in future developments.Abdullah, MTA.; Lloret, J.; Canovas Solbes, A.; GarcĂa-GarcĂa, L. (2017). Survey of Transportation of Adaptive Multimedia Streaming service in Internet. Network Protocols and Algorithms. 9(1-2):85-125. doi:10.5296/npa.v9i1-2.12412S8512591-
Poor Man's Content Centric Networking (with TCP)
A number of different architectures have been proposed in support of data-oriented or information-centric networking. Besides a similar visions, they share the need for designing a new networking architecture. We present an incrementally deployable approach to content-centric networking based upon TCP. Content-aware senders cooperate with probabilistically operating routers for scalable content delivery (to unmodified clients), effectively supporting opportunistic caching for time-shifted access as well as de-facto synchronous multicast delivery. Our approach is application protocol-independent and provides support beyond HTTP caching or managed CDNs. We present our protocol design along with a Linux-based implementation and some initial feasibility checks
Scalable Multiple Description Coding and Distributed Video Streaming over 3G Mobile Networks
In this thesis, a novel Scalable Multiple Description Coding (SMDC) framework is proposed. To address the bandwidth fluctuation, packet loss and heterogeneity problems in the wireless networks and further enhance the error resilience tools in Moving Pictures Experts Group 4 (MPEG-4), the joint design of layered coding (LC) and multiple description coding (MDC) is explored. It leverages a proposed distributed multimedia delivery mobile network (D-MDMN) to provide path diversity to combat streaming video outage due to handoff in Universal Mobile Telecommunications System (UMTS). The corresponding intra-RAN (Radio Access Network) handoff and inter-RAN handoff procedures in D-MDMN are studied in details, which employ the principle of video stream re-establishing to replace the principle of data forwarding in UMTS. Furthermore, a new IP (Internet Protocol) Differentiated Services (DiffServ) video marking algorithm is proposed to support the unequal error protection (UEP) of LC components of SMDC. Performance evaluation is carried through simulation using OPNET Modeler 9. 0. Simulation results show that the proposed handoff procedures in D-MDMN have better performance in terms of handoff latency, end-to-end delay and handoff scalability than that in UMTS. Performance evaluation of our proposed IP DiffServ video marking algorithm is also undertaken, which shows that it is more suitable for video streaming in IP mobile networks compared with the previously proposed DiffServ video marking algorithm (DVMA)
Multimedia over wireless ip networks:distortion estimation and applications.
2006/2007This thesis deals with multimedia communication over unreliable and resource
constrained IP-based packet-switched networks. The focus is on estimating, evaluating
and enhancing the quality of streaming media services with particular regard
to video services. The original contributions of this study involve mainly the
development of three video distortion estimation techniques and the successive
definition of some application scenarios used to demonstrate the benefits obtained
applying such algorithms. The material presented in this dissertation is the result
of the studies performed within the Telecommunication Group of the Department
of Electronic Engineering at the University of Trieste during the course of Doctorate
in Information Engineering.
In recent years multimedia communication over wired and wireless packet based
networks is exploding. Applications such as BitTorrent, music file sharing, multimedia
podcasting are the main source of all traffic on the Internet. Internet radio
for example is now evolving into peer to peer television such as CoolStreaming.
Moreover, web sites such as YouTube have made publishing videos on demand
available to anyone owning a home video camera. Another challenge in the multimedia
evolution is inside the house where videos are distributed over local WiFi
networks to many end devices around the house. More in general we are assisting
an all media over IP revolution, with radio, television, telephony and stored media
all being delivered over IP wired and wireless networks. All the presented applications
require an extreme high bandwidth and often a low delay especially for
interactive applications. Unfortunately the Internet and the wireless networks provide
only limited support for multimedia applications. Variations in network conditions
can have considerable consequences for real-time multimedia applications
and can lead to unsatisfactory user experience. In fact, multimedia applications
are usually delay sensitive, bandwidth intense and loss tolerant applications. In order
to overcame this limitations, efficient adaptation mechanism must be derived
to bridge the application requirements with the transport medium characteristics.
Several approaches have been proposed for the robust transmission of multimedia
packets; they range from source coding solutions to the addition of redundancy with forward error correction and retransmissions. Additionally, other techniques
are based on developing efficient QoS architectures at the network layer or at the
data link layer where routers or specialized devices apply different forwarding
behaviors to packets depending on the value of some field in the packet header.
Using such network architecture, video packets are assigned to classes, in order
to obtain a different treatment by the network; in particular, packets assigned to
the most privileged class will be lost with a very small probability, while packets
belonging to the lowest priority class will experience the traditional bestâeffort
service. But the key problem in this solution is how to assign optimally video
packets to the network classes. One way to perform the assignment is to proceed
on a packet-by-packet basis, to exploit the highly non-uniform distortion impact
of compressed video. Working on the distortion impact of each individual video
packet has been shown in recent years to deliver better performance than relying
on the average error sensitivity of each bitstream element. The distortion impact
of a video packet can be expressed as the distortion that would be introduced at
the receiver by its loss, taking into account the effects of both error concealment
and error propagation due to temporal prediction.
The estimation algorithms proposed in this dissertation are able to reproduce accurately
the distortion envelope deriving from multiple losses on the network and
the computational complexity required is negligible in respect to those proposed in
literature. Several tests are run to validate the distortion estimation algorithms and
to measure the influence of the main encoder-decoder settings. Different application scenarios are described and compared to demonstrate the benefits obtained
using the developed algorithms. The packet distortion impact is inserted in each
video packet and transmitted over the network where specialized agents manage
the video packets using the distortion information. In particular, the internal structure of the agents is modified to allow video packets prioritization using primarily
the distortion impact estimated by the transmitter. The results obtained will show
that, in each scenario, a significant improvement may be obtained with respect to
traditional transmission policies.
The thesis is organized in two parts. The first provides the background material
and represents the basics of the following arguments, while the other is dedicated
to the original results obtained during the research activity.
Referring to the first part in the first chapter it summarized an introduction to
the principles and challenges for the multimedia transmission over packet networks.
The most recent advances in video compression technologies are detailed
in the second chapter, focusing in particular on aspects that involve the resilience
to packet loss impairments. The third chapter deals with the main techniques
adopted to protect the multimedia flow for mitigating the packet loss corruption due to channel failures. The fourth chapter introduces the more recent advances in
network adaptive media transport detailing the techniques that prioritize the video
packet flow. The fifth chapter makes a literature review of the existing distortion
estimation techniques focusing mainly on their limitation aspects.
The second part of the thesis describes the original results obtained in the modelling
of the video distortion deriving from the transmission over an error prone
network. In particular, the sixth chapter presents three new distortion estimation
algorithms able to estimate the video quality and shows the results of some validation
tests performed to measure the accuracy of the employed algorithms. The
seventh chapter proposes different application scenarios where the developed algorithms may be used to enhance quickly the video quality at the end user side.
Finally, the eight chapter summarizes the thesis contributions and remarks the
most important conclusions. It also derives some directions for future improvements.
The intent of the entire work presented hereafter is to develop some video distortion
estimation algorithms able to predict the user quality deriving from the loss on the network as well as providing the results of some useful applications able to enhance the user experience during a video streaming session.Questa tesi di dottorato affronta il problema della trasmissione efficiente di contenuti
multimediali su reti a pacchetto inaffidabili e con limitate risorse di banda.
Lâobiettivo è quello di ideare alcuni algoritmi in grado di predire lâandamento
della qualitĂ del video ricevuto da un utente e successivamente ideare alcune tecniche in grado di migliorare lâesperienza dellâutente finale nella fruizione dei servizi video. In particolare i contributi originali del presente lavoro riguardano lo sviluppo di algoritmi per la stima della distorsione e lâideazione di alcuni scenari applicativi in molto frequenti dove poter valutare i benefici ottenibili applicando gli algoritmi di stima.
I contributi presentati in questa tesi di dottorato sono il risultato degli studi compiuti con il gruppo di Telecomunicazioni del Dipartimento di Elettrotecnica Elettronica ed Informatica (DEEI) dellâUniversitĂ degli Studi di Trieste durante il corso di dottorato in Ingegneria dellâInformazione.
Negli ultimi anni la multimedialitĂ , diffusa sulle reti cablate e wireless, sta diventando
parte integrante del modo di utilizzare la rete diventando di fatto il fenomeno piĂš imponente. Applicazioni come BitTorrent, la condivisione di file musicali e multimediali e il podcasting ad esempio costituiscono una parte significativa del traffico attuale su Internet. Quelle che negli ultimi anni erano le prime radio che trsmettevano sulla rete oggi si stanno evolvendo nei sistemi peer
to peer per piĂš avanzati per la diffusione della TV via web come CoolStreaming.
Inoltre siti web come YouTube hanno costruito il loro business sulla memorizzazione/
distribuzione di video creati da chiunque abbia una semplice video camera.
Unâaltra caratteristica dellâimponente rivoluzione multimediale a cui stiamo
assistendo è la diffusione dei video anche allâinterno delle case dove i contenuti
multimediali vengono distribuiti mediante delle reti wireless locali tra i vari dispositivi finali. Tuttâoggi è in corso una rivoluzione della multimedialitĂ sulle reti
IP con le radio, i televisioni, la telefonia e tutti i video che devono essere distribuiti
sulle reti cablate e wireless verso utenti eterogenei. In generale la gran parte delle
applicazioni multimediali richiedono una banda elevata e dei ritardi molto contenuti specialmente se le applicazioni sono di tipo interattivo. Sfortunatamente le reti wireless e Internet piĂš in generale sono in grado di fornire un supporto limitato alle applicazioni multimediali. La variabilitĂ di banda, di ritardo e nella perdita possono avere conseguenze gravi sulla qualitĂ con cui viene ricevuto il video e questo può portare a una parziale insoddisfazione o addirittura alla rinuncia della fruizione da parte dellâutente finale.
Le applicazioni multimediali sono spesso sensibili al ritardo e con requisiti di
banda molto stringenti ma di fatto rimango tolleranti nei confronti delle perdite
che possono avvenire durante la trasmissione. Al fine di superare le limitazioni è necessario sviluppare dei meccanismi di adattamento in grado di fare da ponte fra i requisiti delle applicazioni multimediali e le caratteristiche offerte dal livello di trasporto. Diversi approcci sono stati proposti in passato in letteratura per
migliorare la trasmissione dei pacchetti riducendo le perdite; gli approcci variano
dalle soluzioni di compressione efficiente allâaggiunta di ridondanza con tecniche
di forward error correction e ritrasmissioni. Altre tecniche si basano sulla creazione di architetture di rete complesse in grado di garantire la QoS a livello rete dove router oppure altri agenti specializzati applicano diverse politiche di gestione del traffico in base ai valori contenuti nei campi dei pacchetti. Mediante queste architetture il traffico video viene marcato con delle classi di prioritĂ al fine di creare una differenziazione nel traffico a livello rete; in particolare i pacchetti con i privilegi maggiori vengono assegnati alle classi di prioritĂ piĂš elevate e verranno persi con probabilitĂ molto bassa mentre i pacchetti appartenenti alle classi di prioritĂ inferiori saranno trattati alla stregua dei servizi di tipo best-effort. Uno dei principali problemi di questa soluzione riguarda come assegnare in maniera ottimale i singoli pacchetti video alle diverse classi di prioritĂ . Un modo per effettuare questa classificazione è quello di procedere assegnando i pacchetti alle varie classi sulla base dellâimportanza che ogni pacchetto ha sulla qualitĂ finale.
Eâ stato dimostrato in numerosi lavori recenti che utilizzando come meccanismo
per lâadattamento lâimpatto sulla distorsione finale, porta significativi miglioramenti
rispetto alle tecniche che utilizzano come parametro la sensibilitĂ media del flusso nei confronti delle perdite. Lâimpatto che ogni pacchetto ha sulla qualitĂ può essere espresso come la distorsione che viene introdotta al ricevitore se il pacchetto viene perso tenendo in considerazione gli effetti del recupero (error concealment) e la propagazione dellâerrore (error propagation) caratteristica dei piĂš recenti codificatori video.
Gli algoritmi di stima della distorsione proposti in questa tesi sono in grado di riprodurre in maniera accurata lâinviluppo della distorsione derivante sia da perdite isolate che da perdite multiple nella rete con una complessitĂ computazionale minima se confrontata con le piĂš recenti tecniche di stima. Numerose prove sono stati effettuate al fine di validare gli algoritmi di stima e misurare lâinfluenza dei principali parametri di codifica e di decodifica. Al fine di enfatizzare i benefici ottenuti applicando gli algoritmi di stima della distorsione, durante la tesi verranno presentati alcuni scenari applicativi dove lâapplicazione degli algoritmi proposti migliora sensibilmente la qualitĂ finale percepita dagli utenti. Tali scenari verranno descritti, implementati e accuratamente valutati. In particolare, la distorsione stimata dal trasmettitore verrĂ incapsulata nei pacchetti video e, trasmessa
nella rete dove agenti specializzati potranno agevolmente estrarla e utilizzarla come meccanismo rate-distortion per privilegiare alcuni pacchetti a discapito di altri. In particolare la struttura interna di un agente (un router) verrĂ modificata al fine di consentire la differenziazione del traffico utilizzando lâinformazione dellâimpatto che ogni pacchetto ha sulla qualitĂ finale. I risultati ottenuti anche in termini di ridotta complessitĂ computazionale in ogni scenario applicativo proposto mettono in luce i benefici derivanti dallâimplementazione degli algoritmi di stima.
La presenti tesi di dottorato è strutturata in due parti principali; la prima fornisce
il background e rappresenta la base per tutti gli argomenti trattati nel seguito mentre
la seconda parte è dedicata ai contributi originali e ai risultati ottenuti durante
lâintera attivitĂ di ricerca.
In riferimento alla prima parte in particolare unâintroduzione ai principi e alle opportunitĂ offerte dalla diffusione dei servizi multimediali sulle reti a pacchetto
viene esposta nel primo capitolo. I progressi piĂš recenti nelle tecniche di compressione
video vengono esposti dettagliatamente nel secondo capitolo che si focalizza in particolare solo sugli aspetti che riguardano le tecniche per la mitigazione delle perdite. Il terzo capitolo introduce le principali tecniche per proteggere i flussi multimediali e ridurre le perdite causate dai fenomeni caratteristici del canale. Il quarto capitolo descrive i recenti avanzamenti nelle tecniche di network adaptive media transport illustrando i principali metodi utilizzati per differenziare il traffico video. Il quinto capitolo analizza i principali contributi nella letteratura sulle
tecniche di stima della distorsione e si focalizza in particolare sulle limitazioni dei metodi attuali.
La seconda parte della tesi descrive i contributi originali ottenuti nella modellizzazione della distorsione video derivante dalla trasmissione sulle reti con perdite.
In particolare il sesto capitolo presenta tre nuovi algoritmi in grado di riprodurre
fedelmente lâinviluppo della distorsione video. I numerosi test e risultati verranno
proposti al fine di validare gli algoritmi e misurare lâaccuratezza nella stima. Il settimo capitolo propone diversi scenari applicativi dove gli algoritmi sviluppati
possono essere utilizzati per migliorare in maniera significativa la qualitĂ percepita
dallâutente finale. Infine lâottavo capitolo sintetizza lâintero lavoro svolto e i principali risultati ottenuti. Nello stesso capitolo vengono inoltre descritti gli
sviluppi futuri dellâattivitĂ di ricerca.
Lâobiettivo dellâintero lavoro presentato è quello di mostrare i benefici derivanti
dallâutilizzo di nuovi algoritmi per la stima della distorsione e di fornire alcuni
scenari applicativi di utilizzo.XIX Ciclo197
Low-complexity video coding for receiver-driven layered multicast
In recent years, the âInternet Multicast Backbone,â or MBone, has risen from a small, research curiosity to a large- scale and widely used communications infrastructure. A driving force behind this growth was the development of multipoint audio, video, and shared whiteboard conferencing applications. Because these real-time media are transmitted at a uniform rate to all of the receivers in the network, a source must either run at the bottleneck rate or overload portions of its multicast distribution tree. We overcome this limitation by moving the burden of rate adaptation from the source to the receivers with a scheme we call receiver-driven layered multicast, or RLM. In RLM, a source distributes a hierarchical signal by striping the different layers across multiple multicast groups, and receivers adjust their reception rate by simply joining and leaving multicast groups. In this paper, we describe a layered video compression algorithm which, when combined with RLM, provides a comprehensive solution for scalable multicast video transmission in heterogeneous networks. In addition to a layered representation, our coder has low complexity (admitting an effi- cient software implementation) and high loss resilience (admitting robust operation in loosely controlled environments like the Inter- net). Even with these constraints, our hybrid DCT/wavelet-based coder exhibits good compression performance. It outperforms all publicly available Internet video codecs while maintaining comparable run-time performance. We have implemented our coder in a ârealâ applicationâthe UCB/LBL videoconferencing tool vic. Unlike previous work on layered video compression and transmission, we have built a fully operational system that is currently being deployed on a very large scale over the MBone
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