274 research outputs found
Design of a transport coding scheme for high-quality video over ATM networks
Caption title.Includes bibliographical references (p. 38-39).Supported by ARPA. F30602-92-C-0030 Supported by the Laboratory for Information and Decision Systems, Massachusetts Institute of Technology. DAAH04-95-1-0103V. Parthasarathy, J.W. Modestino and K.S. Vastola
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 best–effort
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.
L’obiettivo è quello di ideare alcuni algoritmi in grado di predire l’andamento
della qualità del video ricevuto da un utente e successivamente ideare alcune tecniche in grado di migliorare l’esperienza dell’utente 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 l’ideazione 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) dell’Università degli Studi di Trieste durante il corso di dottorato in Ingegneria dell’Informazione.
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 piĂą 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 piĂą 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.
Un’altra caratteristica dell’imponente rivoluzione multimediale a cui stiamo
assistendo è la diffusione dei video anche all’interno delle case dove i contenuti
multimediali vengono distribuiti mediante delle reti wireless locali tra i vari dispositivi finali. Tutt’oggi è in corso una rivoluzione della multimedialità 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 più in generale sono in grado di fornire un supporto limitato alle applicazioni multimediali. La variabilità di banda, di ritardo e nella perdita possono avere conseguenze gravi sulla qualità con cui viene ricevuto il video e questo può portare a una parziale insoddisfazione o addirittura alla rinuncia della fruizione da parte dell’utente 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 all’aggiunta 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 priorità al fine di creare una differenziazione nel traffico a livello rete; in particolare i pacchetti con i privilegi maggiori vengono assegnati alle classi di priorità più elevate e verranno persi con probabilità molto bassa mentre i pacchetti appartenenti alle classi di priorità 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 priorità . Un modo per effettuare questa classificazione è quello di procedere assegnando i pacchetti alle varie classi sulla base dell’importanza che ogni pacchetto ha sulla qualità finale.
E’ stato dimostrato in numerosi lavori recenti che utilizzando come meccanismo
per l’adattamento l’impatto sulla distorsione finale, porta significativi miglioramenti
rispetto alle tecniche che utilizzano come parametro la sensibilità media del flusso nei confronti delle perdite. L’impatto che ogni pacchetto ha sulla qualità 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 dell’errore (error propagation) caratteristica dei più recenti codificatori video.
Gli algoritmi di stima della distorsione proposti in questa tesi sono in grado di riprodurre in maniera accurata l’inviluppo della distorsione derivante sia da perdite isolate che da perdite multiple nella rete con una complessità computazionale minima se confrontata con le più recenti tecniche di stima. Numerose prove sono stati effettuate al fine di validare gli algoritmi di stima e misurare l’influenza 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 l’applicazione degli algoritmi proposti migliora sensibilmente la qualità finale percepita dagli utenti. Tali scenari verranno descritti, implementati e accuratamente valutati. In particolare, la distorsione stimata dal trasmettitore verrà 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) verrà modificata al fine di consentire la differenziazione del traffico utilizzando l’informazione dell’impatto che ogni pacchetto ha sulla qualità finale. I risultati ottenuti anche in termini di ridotta complessità computazionale in ogni scenario applicativo proposto mettono in luce i benefici derivanti dall’implementazione 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
l’intera attività di ricerca.
In riferimento alla prima parte in particolare un’introduzione ai principi e alle opportunità offerte dalla diffusione dei servizi multimediali sulle reti a pacchetto
viene esposta nel primo capitolo. I progressi piĂą 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 l’inviluppo della distorsione video. I numerosi test e risultati verranno
proposti al fine di validare gli algoritmi e misurare l’accuratezza nella stima. Il settimo capitolo propone diversi scenari applicativi dove gli algoritmi sviluppati
possono essere utilizzati per migliorare in maniera significativa la qualitĂ percepita
dall’utente finale. Infine l’ottavo capitolo sintetizza l’intero lavoro svolto e i principali risultati ottenuti. Nello stesso capitolo vengono inoltre descritti gli
sviluppi futuri dell’attività di ricerca.
L’obiettivo dell’intero lavoro presentato è quello di mostrare i benefici derivanti
dall’utilizzo di nuovi algoritmi per la stima della distorsione e di fornire alcuni
scenari applicativi di utilizzo.XIX Ciclo197
Optimal packet loss protection of progressively compressed 3D meshes
©20009 IEEE. Personal use of this material is permitted. However, permission to reprint/republish this material for advertising or promotional purposes or for creating new collective works for resale or redistribution to servers or lists, or to reuse any copyrighted component of this work in other works must be obtained from the IEEE.This material is presented to ensure timely dissemination of scholarly and technical work. Copyright and all rights therein are retained by authors or by other copyright holders. All persons copying this information are expected to adhere to the terms and constraints invoked by each author's copyright. In most cases, these works may not be reposted without the explicit permission of the copyright holder.We consider a state of the art system that uses layered source coding and forward error correction with Reed-
Solomon codes to efficiently transmit 3D meshes over lossy packet networks. Given a transmission bit budget, the
performance of this system can be optimized by determining how many layers should be sent, how each layer
should be packetized, and how many parity bits should be allocated to each layer such that the expected distortion
at the receiver is minimum. The previous solution for this optimization problem uses exhaustive search, which is not
feasible when the transmission bit budget is large.We propose instead an exact algorithm that solves this optimization
problem in linear time and space. We illustrate the advantages of our approach by providing experimental results
for the CPM (Compressed Progressive Meshes) mesh compression techniqueDFG Research Training Group GK-1042
Practical Distributed Video Coding in Packet Lossy Channels
Improving error resilience of video communications over packet lossy channels is an important and tough task. We present a framework to optimize the quality of video communications based on distributed video coding (DVC) in practical packet lossy network scenarios. The peculiar characteristics of DVC indeed require a number of adaptations to take full advantage of its intrinsic robustness when dealing with data losses of typical real packet networks. This work proposes a new packetization scheme, an investigation of the best error-correcting codes to use in a noisy environment, a practical rate-allocation mechanism, which minimizes decoder feedback, and an improved side-information generation and reconstruction function. Performance comparisons are presented with respect to a conventional packet video communication using H.264/advanced video coding (AVC). Although currently the H.264/AVC rate-distortion performance in case of no loss is better than state-of-the-art DVC schemes, under practical packet lossy conditions, the proposed techniques provide better performance with respect to an H.264/AVC-based system, especially at high packet loss rates. Thus the error resilience of the proposed DVC scheme is superior to the one provided by H.264/AVC, especially in the case of transmission over packet lossy networks
Downlink Video Streaming for Users Non-Equidistant from Base Station
We consider multiuser video transmission for users that are non-equidistantly positioned from base station. We propose a greedy algorithm for video streaming in a wireless system with capacity achieving channel coding, that implements the cross-layer principle by partially separating the physical and the application layer. In such a system the parameters at the physical layer are dependent on the packet length and the conditions in the wireless channel and the parameters at the application layer are dependent on the reduction of the expected distortion assuming no packet errors in the system. We also address the fairness in the multiuser video system with non-equidistantly positioned users. Our fairness algorithm is based on modified opportunistic round robin scheduling. We evaluate the performance of the proposed algorithms by simulating the transmission of H.264/AVC video signals in a TDMA wireless system
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Multicast networks : capacity, algorithms, and implementation
textIn this dissertation, we investigate the capacity and performance of wireless networks with an emphasis on multicast traffic. The defining characteristic of a multicast network is a network where a number of different destinations all require the information generated by a single source. The models that we explore differ in the nature of the nodes from all-mobile case where all nodes are mobile to hybrid case where some nodes are mobile and some are static. We investigate different performance measure for these wireless multicast networks: upper bounds, capacity scaling laws, and achievable rates. The understanding of these measures for such networks helps in the development of efficient algorithms for operating these networks.
In addition, we study the practical realization of algorithms for real-time streaming of rich multimedia content in the context of mobile wireless networks for embedded and cyberphysical systems. Our initial work is in the context of unicast and multiple unicast systems over an autonomous aerial vehicle (AAV) network. Bandwidth requirements and stringent delay constraints of real-time video streaming, paired with limitations on computational complexity and power consumptions imposed by the underlying implementation platform, make cross-layer and cross-domain co-design approaches a necessity. In this dissertation, we propose a novel, low-complexity rate-distortion optimized (RDO) protocol specifically targeted at video streaming over mobile embedded networks. First, we test the performance of our RDO algorithm on simulation models developed for aerial mobility of multiple wirelessly communicating AAVs. Second, we test the performance of our RDO algorithm and other proposed adaptive algorithms on a real network of AAVs and present a comparative study between these different algorithms. Note that generalizing these algorithms to multicast settings is relatively straightforward and thus is not highlighted to a great degree in this thesis.Electrical and Computer Engineerin
Joint Source-Channel Coding Optimized On End-to-End Distortion for Multimedia Source
In order to achieve high efficiency, multimedia source coding usually relies on the use of predictive coding. While more efficient, source coding based on predictive coding has been considered to be more sensitive to errors during communication. With the current volume and importance of multimedia communication, minimizing the overall distortion during communication over an error-prone channel is critical. In addition, for real-time scenarios, it is necessary to consider additional constraints such as fix and small delay for a given bit rate. To comply with these requirements, we seek an efficient joint source-channel coding scheme.
In this work, end-to-end distortion is studied for a first order autoregressive synthetic source that represents a general multimedia traffic. This study reveals that predictive coders achieve the same channel-induced distortion performance as memoryless codecs when applying optimal error concealment. We propose a joint source-channel system based on incremental redundancy that satisfies the fixed delay and error-prone channel constraints and combines DPCM as a source encoder and a rate-compatible punctured convolutional (RCPC) error control codec. To calculate the joint source-channel coding rate allocation that minimizes end-to-end distortion, we develop a Markov Decision Process (MDP) approach for delay constrained feedback Hybrid ARQ, and we use a Dynamic Programming (DP) technique. Our simulation results support the improvement in end-to-end distortion compared to a conventional Forward Error Control (FEC) approach with no feedback
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