304 research outputs found
Loss-resilient Coding of Texture and Depth for Free-viewpoint Video Conferencing
Free-viewpoint video conferencing allows a participant to observe the remote
3D scene from any freely chosen viewpoint. An intermediate virtual viewpoint
image is commonly synthesized using two pairs of transmitted texture and depth
maps from two neighboring captured viewpoints via depth-image-based rendering
(DIBR). To maintain high quality of synthesized images, it is imperative to
contain the adverse effects of network packet losses that may arise during
texture and depth video transmission. Towards this end, we develop an
integrated approach that exploits the representation redundancy inherent in the
multiple streamed videos a voxel in the 3D scene visible to two captured views
is sampled and coded twice in the two views. In particular, at the receiver we
first develop an error concealment strategy that adaptively blends
corresponding pixels in the two captured views during DIBR, so that pixels from
the more reliable transmitted view are weighted more heavily. We then couple it
with a sender-side optimization of reference picture selection (RPS) during
real-time video coding, so that blocks containing samples of voxels that are
visible in both views are more error-resiliently coded in one view only, given
adaptive blending will erase errors in the other view. Further, synthesized
view distortion sensitivities to texture versus depth errors are analyzed, so
that relative importance of texture and depth code blocks can be computed for
system-wide RPS optimization. Experimental results show that the proposed
scheme can outperform the use of a traditional feedback channel by up to 0.82
dB on average at 8% packet loss rate, and by as much as 3 dB for particular
frames
Error-resilient performance of Dirac video codec over packet-erasure channel
Video transmission over the wireless or wired network requires error-resilient mechanism since compressed video bitstreams are sensitive to transmission errors because of the use of predictive coding and variable length coding. This paper investigates the performance of a simple and low complexity error-resilient coding scheme which combines source and channel coding to protect compressed bitstream of wavelet-based Dirac video codec in the packet-erasure channel. By partitioning the wavelet transform coefficients of the motion-compensated residual frame into groups and independently processing each group using arithmetic and Forward Error Correction (FEC) coding, Dirac could achieves the robustness to transmission errors by giving the video quality which is gracefully decreasing over a range of packet loss rates up to 30% when compared with conventional FEC only methods. Simulation results also show that the proposed scheme using multiple partitions can achieve up to 10 dB PSNR gain over its existing un-partitioned format. This paper also investigates the error-resilient performance of the proposed scheme in comparison with H.264 over packet-erasure channel
Error Resilient Video Coding Using Bitstream Syntax And Iterative Microscopy Image Segmentation
There has been a dramatic increase in the amount of video traffic over the Internet in past several years. For applications like real-time video streaming and video conferencing, retransmission of lost packets is often not permitted. Popular video coding standards such as H.26x and VPx make use of spatial-temporal correlations for compression, typically making compressed bitstreams vulnerable to errors. We propose several adaptive spatial-temporal error concealment approaches for subsampling-based multiple description video coding. These adaptive methods are based on motion and mode information extracted from the H.26x video bitstreams. We also present an error resilience method using data duplication in VPx video bitstreams.
A recent challenge in image processing is the analysis of biomedical images acquired using optical microscopy. Due to the size and complexity of the images, automated segmentation methods are required to obtain quantitative, objective and reproducible measurements of biological entities. In this thesis, we present two techniques for microscopy image analysis. Our first method, “Jelly Filling” is intended to provide 3D segmentation of biological images that contain incompleteness in dye labeling. Intuitively, this method is based on filling disjoint regions of an image with jelly-like fluids to iteratively refine segments that represent separable biological entities. Our second method selectively uses a shape-based function optimization approach and a 2D marked point process simulation, to quantify nuclei by their locations and sizes. Experimental results exhibit that our proposed methods are effective in addressing the aforementioned challenges
Machine Learning for Multimedia Communications
Machine learning is revolutionizing the way multimedia information is processed and transmitted to users. After intensive and powerful training, some impressive efficiency/accuracy improvements have been made all over the transmission pipeline. For example, the high model capacity of the learning-based architectures enables us to accurately model the image and video behavior such that tremendous compression gains can be achieved. Similarly, error concealment, streaming strategy or even user perception modeling have widely benefited from the recent learningoriented developments. However, learning-based algorithms often imply drastic changes to the way data are represented or consumed, meaning that the overall pipeline can be affected even though a subpart of it is optimized. In this paper, we review the recent major advances that have been proposed all across the transmission chain, and we discuss their potential impact and the research challenges that they raise
Resilient Digital Video Transmission over Wireless Channels using Pixel-Level Artefact Detection Mechanisms
Recent advances in communications and video coding technology have brought multimedia communications into everyday life, where a variety of services and applications are being integrated within different devices such that multimedia content is provided everywhere and on any device. H.264/AVC provides a major advance on preceding video coding standards obtaining as much as twice the coding efficiency over these standards (Richardson I.E.G., 2003, Wiegand T. & Sullivan G.J., 2007). Furthermore, this new codec inserts video related information within network abstraction layer units (NALUs), which facilitates the transmission of H.264/AVC coded sequences over a variety of network environments (Stockhammer, T. & Hannuksela M.M., 2005) making it applicable for a broad range of applications such as TV broadcasting, mobile TV, video-on-demand, digital media storage, high definition TV, multimedia streaming and conversational applications. Real-time wireless conversational and broadcast applications are particularly challenging as, in general, reliable delivery cannot be guaranteed (Stockhammer, T. & Hannuksela M.M., 2005). The H.264/AVC standard specifies several error resilient strategies to minimise the effect of transmission errors on the perceptual quality of the reconstructed video sequences. However, these methods assume a packet-loss scenario where the receiver discards and conceals all the video information contained within a corrupted NALU packet. This implies that the error resilient methods adopted by the standard operate at a lower bound since not all the information contained within a corrupted NALU packet is un-utilizable (Stockhammer, T. et al., 2003).peer-reviewe
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
Recommended from our members
Research and developments of Dirac video codec
This thesis was submitted for the degree of Doctor of Philosophy and was awarded by Brunel University.In digital video compression, apart from storage, successful transmission of the compressed video
data over the bandwidth limited erroneous channels is another important issue. To enable a video
codec for broadcasting application, it is required to implement the corresponding coding tools (e.g.
error-resilient coding, rate control etc.). They are normally non-normative parts of a video codec and
hence their specifications are not defined in the standard. In Dirac as well, the original codec is
optimized for storage purpose only and so, several non-normative part of the encoding tools are still
required in order to be able to use in other types of application.
Being the "Research and Developments of the Dirac Video Codec" as the research title, phase I of
the project is mainly focused on the error-resilient transmission over a noisy channel. The error-resilient
coding method used here is a simple and low complex coding scheme which provides the
error-resilient transmission of the compressed video bitstream of Dirac video encoder over the packet
erasure wired network. The scheme combines source and channel coding approach where error-resilient
source coding is achieved by data partitioning in the wavelet transformed domain and
channel coding is achieved through the application of either Rate-Compatible Punctured
Convolutional (RCPC) Code or Turbo Code (TC) using un-equal error protection between header plus
MV and data. The scheme is designed mainly for the packet-erasure channel, i.e. targeted for the
Internet broadcasting application.
But, for a bandwidth limited channel, it is still required to limit the amount of bits generated from
the encoder depending on the available bandwidth in addition to the error-resilient coding. So, in the
2nd phase of the project, a rate control algorithm is presented. The algorithm is based upon the Quality
Factor (QF) optimization method where QF of the encoded video is adaptively changing in order to
achieve average bitrate which is constant over each Group of Picture (GOP). A relation between the
bitrate, R and the QF, which is called Rate-QF (R-QF) model is derived in order to estimate the
optimum QF of the current encoding frame for a given target bitrate, R.
In some applications like video conferencing, real-time encoding and decoding with minimum
delay is crucial, but, the ability to do real-time encoding/decoding is largely determined by the
complexity of the encoder/decoder. As we all know that motion estimation process inside the encoder
is the most time consuming stage. So, reducing the complexity of the motion estimation stage will
certainly give one step closer to the real-time application. So, as a partial contribution toward realtime
application, in the final phase of the research, a fast Motion Estimation (ME) strategy is designed
and implemented. It is the combination of modified adaptive search plus semi-hierarchical way of
motion estimation. The same strategy was implemented in both Dirac and H.264 in order to
investigate its performance on different codecs. Together with this fast ME strategy, a method which
is called partial cost function calculation in order to further reduce down the computational load of the
cost function calculation was presented. The calculation is based upon the pre-defined set of patterns
which were chosen in such a way that they have as much maximum coverage as possible over the
whole block.
In summary, this research work has contributed to the error-resilient transmission of compressed
bitstreams of Dirac video encoder over a bandwidth limited error prone channel. In addition to this,
the final phase of the research has partially contributed toward the real-time application of the Dirac
video codec by implementing a fast motion estimation strategy together with partial cost function
calculation idea.BBC R&D and Brunel University
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