Efficient and Effective Schemes for Streaming Media Delivery

The rapid expansion of the Internet and the increasingly wide deployment of wireless networks provide opportunities to deliver streaming media content to users at anywhere, anytime. To ensure good user experience, it is important to battle adversary effects, such as delay, loss and jitter. In this thesis, we first study efficient loss recovery schemes, which require pure XOR operations. In particular, we propose a novel scheme capable of recovering up to 3 packet losses, and it has the lowest complexity among all known schemes. We also propose an efficient algorithm for array codes decoding, which achieves significant throughput gain and energy savings over conventional codes. We believe these schemes are applicable to streaming applications, especially in wireless environments. We then study quality adaptation schemes for client buffer management. Our control-theoretic approach results in an efficient online rate control algorithm with analytically tractable performance. Extensive experimental results show that three goals are achieved: fast startup, continuous playback in the face of severe congestion, and maximal quality and smoothness over the entire streaming session. The scheme is later extended to streaming with limited quality levels, which is then directly applicable to existing systems

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

Multimedia streaming over wireless channels

The improvements in mobile communication systems have accelerated the development of new multimedia streaming techniques to increase the quality of streaming data over time varying wireless channels. In order to increase multimedia quality, error control schemes are indispensable due to time-varying and erroneous nature of the channel. However, relatively low channel capacity of wireless channels, and dependency structure in multimedia limit the eectiveness of existing error control schemes and require more sophisticated techniques to provide quality improvement on the streaming data. In this thesis, we propose sender driven multimedia streaming algorithms that incorporate error control schemes of FEC, ARQ, and packet scheduling by considering media and channel parameters such as packet importance, packet dependencies, decoding deadlines, channel state information, and channel capacity. Initially, we have proposed a multi-rate distortion optimization framework so as to jointly optimize FEC rate and packet selection by minimizing end-to-end distortion to satisfy a specified Quality of Service under channel capacity constraint. Minimization of end-to-end distortion causes computational complexity in the rate distortion optimization framework due to dependency in encoded multimedia. Therefore, we propose multimedia streaming algorithms that select packet and FEC rate with reduced computational complexity and high quality as compared with multi-rate distortion optimization framework. Additionally, protocol stack of a UMTS cellular network system with W-CDMA air interface is presented in order to clarify the relation between proposed multimedia streaming algorithms and UMTS system that is used in simulations. Finally, proposed algorithms are simulated and results demonstrate that proposed algorithms improve multimedia quality significantly as compared to existing methods

ENSURE: A Time Sensitive Transport Protocol to Achieve Reliability Over Wireless in Petrochemical Plants

As society becomes more reliant on the resources extracted in petroleum refinement the production demand for petrochemical plants increases. A key element is producing efficiently while maintaining safety through constant monitoring of equipment feedback. Currently, temperature and flow sensors are deployed at various points of production and 10/100 Ethernet cable is installed to connect them to a master control unit. This comes at a great monetary cost, not only at the time of implementation but also when repairs are required. The capability to provide plant wide wireless networks would both decrease investment cost and downtime needed for repairs. However, the current state of wireless networks does not provide any guarantee of reliability, which is critical to the industry. When factoring in the need for real-time information, network reliability further decreases. This work presents the design and development of a series of transport layer protocols (coined ENSURE) to provide time-sensitive reliability. More specifically three versions were developed to meet specific needs of the data being sent. ENSURE 1.0 addresses reliability, 2.0 enforces a time limit and the final version, 3.0, provides a balance of the two. A network engineer can set each specific area of the plant to use a different version of ENSURE based network performance needs for the data it produces. The end result being a plant wide wireless network that performs in a timely and reliable fashion

Source video rate allocation and scheduling policy design for wireless networks

In this work we address the problem of designing an efficient algorithm that allows high quality real time video streaming, from the source coding and scheduling perspectives. We take into account some key quantities like frame correlation, channel constraints and QoS metrics such as distortion and PSNR. We first formalize the problem in a mathematical fashion and then propose a Markov-chain based solution that applies to quantized values of the metrics involved in the framewor

Design of interface selection protocols for multi-homed wireless networks

This thesis was submitted for the degree of Doctor of Philosophy and was awarded by Brunel University on 10 December 2010.The IEEE 802.11/802.16 standards conformant wireless communication stations have multi-homing transmission capability. To achieve greater communication efficiency, multi-homing capable stations use handover mechanism to select appropriate transmission channel according to variations in the channel quality. This thesis presents three internal-linked handover schemes, (1) Interface Selection Protocol (ISP), belonging to Wireless Local Area Network (WLAN)- Worldwide Interoperability for Microwave Access (WiMAX) environment (2) Fast Channel Scanning (FCS) and (3) Traffic Manager (TM), (2) and (3) belonging to WiMAX Environment. The proposed schemes in this thesis use a novel mechanism of providing a reliable communication route. This solution is based on a cross-layer communication framework, where the interface selection module uses various network related parameters from Medium Access Control (MAC) sub-layer/Physical Layer (PHY) across the protocol suite for decision making at the Network layer. The proposed solutions are highly responsive when compared with existing multi-homed schemes; responsiveness is one of the key factors in the design of such protocols. Selected route under these schemes is based on the most up to date link-layer information. Therefore, such a route is not only reliable in terms of route optimization but it also fulfils the application demands in terms of throughput and delay. Design of ISP protocol use probing frames during the route discovery process. The 802.11 mandates the use of different rates for data transmission frames. The ISP-metric can be incorporated into various routing aspects and its applicability is determined by the possibility of provision of MAC dependent parameters that are used to determine the best path metric values. In many cases, higher device density, interference and mobility cause variable medium access delays. It causes creation of ‘unreachable zones’, where destination is marked as unreachable. However, by use of the best path metric, the destination has been made reachable, anytime and anywhere, because of the intelligent use of the probing frames and interface selection algorithm implemented. The IEEE 802.16e introduces several MAC level queues for different access categories, maintaining service requirement within these queues; which imply that frames from a higher priority queue, i.e. video frames, are serviced more frequently than those belonging to lower priority queues. Such an enhancement at the MAC sub-layer introduces uneven queuing delays. Conventional routing protocols are unaware of such MAC specific constraints and as a result, these factors are not considered which result in channel performance degradation. To meet such challenges, the thesis presents FCS and TM schemes for WiMAX. For FCS, Its solution is to improve the mobile WiMAX handover and address the scanning latency. Since minimum scanning time is the most important issue in the handover process. This handover scheme aims to utilize the channel efficiently and apply such a procedure to reduce the time it takes to scan the neighboring access stations. TM uses MAC and physical layer (PHY) specific information in the interface metric and maintains a separate path to destination by applying an alternative interface operation. Simulation tests and comparisons with existing multi-homed protocols and handover schemes demonstrate the effectiveness of incorporating the medium dependent parameters. Moreover, show that suggested schemes, have shown better performance in terms of end-to-end delay and throughput, with efficiency up to 40% in specific test scenarios

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 multimedialità, diffusa sulle reti cablate e wireless, sta diventando parte integrante del modo di utilizzare la rete diventando di fatto il fenomeno più 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 più 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 più 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 qualità 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

Progressive Source-Channel Coding for Multimedia Transmission over Noisy and Lossy Channels with and without Feedback

Rate-scalable or layered lossy source-coding is useful for progressive transmission of multimedia sources, where the receiver can reconstruct the source incrementally. This thesis considers ``joint source-channel'' schemes for such a progressive transmission, in the presence of noise or loss, with and without the use of a feedback link. First we design image communication schemes for memoryless and finite state channels using limited and explicitly constrained use of the feedback channel in the form of a variable incremental redundancy Hybrid ARQ protocol. Constraining feedback allows a direct comparison with schemes without feedback. Optimized feedback based systems are shown to have useful gains. Second, we develop a controlled Markov chain approach for constrained feedback Hybrid ARQ protocol design. The proposed methodology allows the protocol to be chosen from a collection of signal flow graphs, and also allows explicit control over the tradeoffs in throughput, reliability and complexity. Next we consider progressive image transmission in the absence of feedback. We assign unequal error protection to the bits of a rate-scalable source-coder using rate compatible channel codes. We show that, under the framework, the source and channel bits can be ``scheduled'' in a single bitstream in such a way that operational optimality is retained for different transmission budgets, creating a rate-scalable joint source-channel coder. Next we undertake the design of a joint source-channel decoder that uses ``distortion aware'' ACK/NACK feedback generation. For memoryless channels, and Type-I HARQ, the design of optimal ACK/NACK generation and decoding by packet combining is cast and solved as a sequential decision problem. We obtain dynamic programming based optimal solutions and also propose suboptimal, lower complexity distortion-aware decoders and feedback generation rules which outperform conventional BER based rules such as CRC-check. Finally we design operational rate-distortion optimal ACK/NACK feedback generation rules for transmitting a tree structured quantizer over a memoryless channel. We show that the optimal feedback generation rules are embedded, that is, they allow incremental switching to higher rates during the transmission. Also, we obtain the structure of the feedback generation rules in terms of a feedback threshold function that simplifies the implementation