24 research outputs found
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
An adaptive error resilient scheme for packet-switched H.264 video transmission
2010-2011 > Academic research: refereed > Chapter in an edited book (author)Version of RecordPublishe
Rate-distortion analysis and traffic modeling of scalable video coders
In this work, we focus on two important goals of the transmission of scalable video over the Internet. The first goal is to provide high quality video to end users and the second one is to properly design networks and predict network performance for video transmission based on the characteristics of existing video traffic. Rate-distortion (R-D) based schemes are often applied to improve and stabilize video quality; however, the lack of R-D modeling of scalable coders limits their applications in scalable streaming.
Thus, in the first part of this work, we analyze R-D curves of scalable video coders and propose a novel operational R-D model. We evaluate and demonstrate the accuracy of our R-D function in various scalable coders, such as Fine Granular Scalable (FGS) and Progressive FGS coders. Furthermore, due to the time-constraint nature of Internet streaming, we propose another operational R-D model, which is accurate yet with low computational cost, and apply it to streaming applications for quality control purposes.
The Internet is a changing environment; however, most quality control approaches only consider constant bit rate (CBR) channels and no specific studies have been conducted for quality control in variable bit rate (VBR) channels. To fill this void, we examine an asymptotically stable congestion control mechanism and combine it with our R-D model to present smooth visual quality to end users under various network conditions.
Our second focus in this work concerns the modeling and analysis of video traffic, which is crucial to protocol design and efficient network utilization for video transmission. Although scalable video traffic is expected to be an important source for the Internet, we find that little work has been done on analyzing or modeling it. In this regard, we develop a frame-level hybrid framework for modeling multi-layer VBR video traffic. In the proposed framework, the base layer is modeled using a combination of wavelet and time-domain methods and the enhancement layer is linearly predicted from the base layer using the cross-layer correlation
Complexity management of H.264/AVC video compression.
The H. 264/AVC video coding standard offers significantly improved compression efficiency and flexibility compared to previous standards. However, the high computational complexity of H. 264/AVC is a problem for codecs running on low-power hand held devices and general purpose computers. This thesis presents new techniques to reduce, control and manage the computational complexity of an H. 264/AVC codec. A new complexity reduction algorithm for H. 264/AVC is developed. This algorithm predicts "skipped" macroblocks prior to motion estimation by estimating a Lagrange ratedistortion cost function. Complexity savings are achieved by not processing the macroblocks that are predicted as "skipped". The Lagrange multiplier is adaptively modelled as a function of the quantisation parameter and video sequence statistics. Simulation results show that this algorithm achieves significant complexity savings with a negligible loss in rate-distortion performance. The complexity reduction algorithm is further developed to achieve complexity-scalable control of the encoding process. The Lagrangian cost estimation is extended to incorporate computational complexity. A target level of complexity is maintained by using a feedback algorithm to update the Lagrange multiplier associated with complexity. Results indicate that scalable complexity control of the encoding process can be achieved whilst maintaining near optimal complexity-rate-distortion performance. A complexity management framework is proposed for maximising the perceptual quality of coded video in a real-time processing-power constrained environment. A real-time frame-level control algorithm and a per-frame complexity control algorithm are combined in order to manage the encoding process such that a high frame rate is maintained without significantly losing frame quality. Subjective evaluations show that the managed complexity approach results in higher perceptual quality compared to a reference encoder that drops frames in computationally constrained situations. These novel algorithms are likely to be useful in implementing real-time H. 264/AVC standard encoders in computationally constrained environments such as low-power mobile devices and general purpose computers
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Multi-scale edge-guided image gap restoration
This thesis was submitted for the degree of Doctor of Philosophy and awarded by Brunel University London.The focus of this research work is the estimation of gaps (missing blocks) in digital images. To progress the research two main issues were identified as (1) the appropriate domains for image gap restoration and (2) the methodologies for gap interpolation. Multi-scale transforms provide an appropriate framework for gap restoration. The main advantages are transformations into a set of frequency and scales and the ability to progressively reduce the size of the gap to one sample wide at the transform apex. Two types of multi-scale transform were considered for comparative evaluation; 2-dimensional (2D) discrete cosines (DCT) pyramid and 2D discrete wavelets (DWT). For image gap estimation, a family of conventional weighted interpolators and directional edge-guided interpolators are developed and evaluated. Two types of edges were considered; ‘local’ edges or textures and ‘global’ edges such as the boundaries between objects or within/across patterns in the image. For local edge, or texture, modelling a number of methods were explored which aim to reconstruct a set of gradients across the restored gap as those computed from the known neighbourhood. These differential gradients are estimated along the geometrical vertical, horizontal and cross directions for each pixel of the gap. The edge-guided interpolators aim to operate on distinct regions confined within edge lines. For global edge-guided interpolation, two main methods explored are Sobel and Canny detectors. The latter provides improved edge detection. The combination and integration of different multi-scale domains, local edge interpolators, global edge-guided interpolators and iterative estimation of edges provided a variety of configurations that were comparatively explored and evaluated. For evaluation a set of images commonly used in the literature work were employed together with simulated regular and random image gaps at a variety of loss rate. The performance measures used are the peak signal to noise ratio (PSNR) and structure similarity index (SSIM). The results obtained are better than the state of the art reported in the literature
Mobile Ad-Hoc Networks
Being infrastructure-less and without central administration control, wireless ad-hoc networking is playing a more and more important role in extending the coverage of traditional wireless infrastructure (cellular networks, wireless LAN, etc). This book includes state-of-the-art techniques and solutions for wireless ad-hoc networks. It focuses on the following topics in ad-hoc networks: quality-of-service and video communication, routing protocol and cross-layer design. A few interesting problems about security and delay-tolerant networks are also discussed. This book is targeted to provide network engineers and researchers with design guidelines for large scale wireless ad hoc networks
Method for performance-complexity analyses in SOC-based designs
Master'sMASTER OF ENGINEERIN
Selected topics on distributed video coding
Distributed Video Coding (DVC) is a new paradigm for video compression based on the information theoretical results of Slepian and Wolf (SW), and Wyner and Ziv (WZ). While conventional coding has a rigid complexity allocation as most of the complex tasks are performed at the encoder side, DVC enables a flexible complexity allocation between the encoder and the decoder. The most novel and interesting case is low complexity encoding and complex decoding, which is the opposite of conventional coding. While the latter is suitable for applications where the cost of the decoder is more critical than the encoder's one, DVC opens the door for a new range of applications where low complexity encoding is required and the decoder's complexity is not critical. This is interesting with the deployment of small and battery-powered multimedia mobile devices all around in our daily life. Further, since DVC operates as a reversed-complexity scheme when compared to conventional coding, DVC also enables the interesting scenario of low complexity encoding and decoding between two ends by transcoding between DVC and conventional coding. More specifically, low complexity encoding is possible by DVC at one end. Then, the resulting stream is decoded and conventionally re-encoded to enable low complexity decoding at the other end. Multiview video is attractive for a wide range of applications such as free viewpoint television, which is a system that allows viewing the scene from a viewpoint chosen by the viewer. Moreover, multiview can be beneficial for monitoring purposes in video surveillance. The increased use of multiview video systems is mainly due to the improvements in video technology and the reduced cost of cameras. While a multiview conventional codec will try to exploit the correlation among the different cameras at the encoder side, DVC allows for separate encoding of correlated video sources. Therefore, DVC requires no communication between the cameras in a multiview scenario. This is an advantage since communication is time consuming (i.e. more delay) and requires complex networking. Another appealing feature of DVC is the fact that it is based on a statistical framework. Moreover, DVC behaves as a natural joint source-channel coding solution. This results in an improved error resilience performance when compared to conventional coding. Further, DVC-based scalable codecs do not require a deterministic knowledge of the lower layers. In other words, the enhancement layers are completely independent from the base layer codec. This is called the codec-independent scalability feature, which offers a high flexibility in the way the various layers are distributed in a network. This thesis addresses the following topics: First, the theoretical foundations of DVC as well as the practical DVC scheme used in this research are presented. The potential applications for DVC are also outlined. DVC-based schemes use conventional coding to compress parts of the data, while the rest is compressed in a distributed fashion. Thus, different conventional codecs are studied in this research as they are compared in terms of compression efficiency for a rich set of sequences. This includes fine tuning the compression parameters such that the best performance is achieved for each codec. Further, DVC tools for improved Side Information (SI) and Error Concealment (EC) are introduced for monoview DVC using a partially decoded frame. The improved SI results in a significant gain in reconstruction quality for video with high activity and motion. This is done by re-estimating the erroneous motion vectors using the partially decoded frame to improve the SI quality. The latter is then used to enhance the reconstruction of the finally decoded frame. Further, the introduced spatio-temporal EC improves the quality of decoded video in the case of erroneously received packets, outperforming both spatial and temporal EC. Moreover, it also outperforms error-concealed conventional coding in different modes. Then, multiview DVC is studied in terms of SI generation, which differentiates it from the monoview case. More specifically, different multiview prediction techniques for SI generation are described and compared in terms of prediction quality, complexity and compression efficiency. Further, a technique for iterative multiview SI is introduced, where the final SI is used in an enhanced reconstruction process. The iterative SI outperforms the other SI generation techniques, especially for high motion video content. Finally, fusion techniques of temporal and inter-view side informations are introduced as well, which improves the performance of multiview DVC over monoview coding. DVC is also used to enable scalability for image and video coding. Since DVC is based on a statistical framework, the base and enhancement layers are completely independent, which is an interesting property called codec-independent scalability. Moreover, the introduced DVC scalable schemes show a good robustness to errors as the quality of decoded video steadily decreases with error rate increase. On the other hand, conventional coding exhibits a cliff effect as the performance drops dramatically after a certain error rate value. Further, the issue of privacy protection is addressed for DVC by transform domain scrambling, which is used to alter regions of interest in video such that the scene is still understood and privacy is preserved as well. The proposed scrambling techniques are shown to provide a good level of security without impairing the performance of the DVC scheme when compared to the one without scrambling. This is particularly attractive for video surveillance scenarios, which is one of the most promising applications for DVC. Finally, a practical DVC demonstrator built during this research is described, where the main requirements as well as the observed limitations are presented. Furthermore, it is defined in a setup being as close as possible to a complete real application scenario. This shows that it is actually possible to implement a complete end-to-end practical DVC system relying only on realistic assumptions. Even though DVC is inferior in terms of compression efficiency to the state of the art conventional coding for the moment, strengths of DVC reside in its good error resilience properties and the codec-independent scalability feature. Therefore, DVC offers promising possibilities for video compression with transmission over error-prone environments requirement as it significantly outperforms conventional coding in this case
Terrain Representation And Reasoning In Computer Generated Forces : A Survey Of Computer Generated Forces Systems And How They Represent And Reason About Terrain
Report on a survey of computer systems used to produce realistic or intelligent behavior by autonomous entities in simulation systems. In particular, it is concerned with the data structures used by computer generated forces systems to represent terrain and the algorithmic approaches used by those systems to reason about terrain