13 research outputs found
Compressed-domain techniques for error-resilient video transcoding using RPS
Centre for Signal Processing, Department of Electronic and Information Engineering2008-2009 > Academic research: refereed > Publication in refereed journalVersion of RecordPublishe
Reference picture selection in an already MPEG encoded bitstream
Centre for Multimedia Signal Processing, Department of Electronic and Information EngineeringRefereed conference paper2005-2006 > Academic research: refereed > Refereed conference paperVersion of RecordPublishe
Encryption for high efficiency video coding with video adaptation capabilities
Video encryption techniques enable applications like digital rights management and video scrambling. Applying encryption on the entire video stream can be computationally costly and prevents advanced video modifications by an untrusted middlebox in the network, like splicing, quality monitoring, watermarking, and transcoding. Therefore, encryption techniques are proposed which influence a small amount of the video stream while keeping the video compliant with its compression standard, High Efficiency Video Coding. Encryption while guaranteeing standard compliance can cause degraded compression efficiency, so depending on their bitrate impact, a selection of encrypted syntax elements should be made. Each element also impacts the quality for untrusted decoders differently, so this aspect should also be considered. In this paper, multiple techniques for partial video encryption are investigated, most of them having a low impact on rate-distortion performance and having a broad range in scrambling performance(1)
Intra Coding Strategy for Video Error Resiliency: Behavioral Analysis
One challenge in video transmission is to deal with packet loss. Since the compressed video streams are sensitive to data loss, the error resiliency of the encoded video becomes important. When video data is lost and retransmission is not possible, the missed data should be concealed. But loss concealment causes distortion in the lossy frame which also propagates into the next frames even if their data are received correctly. One promising solution to mitigate this error propagation is intra coding. There are three approaches for intra coding: intra coding of a number of blocks selected randomly or regularly, intra coding of some specific blocks selected by an appropriate cost function, or intra coding of a whole frame. But Intra coding reduces the compression ratio; therefore, there exists a trade-off between bitrate and error resiliency achieved by intra coding. In this paper, we study and show the best strategy for getting the best rate-distortion performance. Considering the error propagation, an objective function is formulated, and with some approximations, this objective function is simplified and solved. The solution demonstrates that periodical I-frame coding is preferred over coding only a number of blocks as intra mode in P-frames. Through examination of various test sequences, it is shown that the best intra frame period depends on the coding bitrate as well as the packet loss rate. We then propose a scheme to estimate this period from curve fitting of the experimental results, and show that our proposed scheme outperforms other methods of intra coding especially for higher loss rates and coding bitrates
Reference picture selection using checkerboard pattern for resilient video coding
The improved compression efficiency achieved by
the High Efficiency Video Coding (HEVC) standard has the
counter-effect of decreasing error resilience in transmission over
error-prone channels. To increase the error resilience of HEVC
streams, this paper proposes a checkerboard reference picture
selection method in order to reduce the prediction mismatch at
the decoder in case of frame losses. The proposed approach not
only allows to reduce the error propagation at the decoder, but
also enhances the quality of reconstructed frames by selectively
constraining the choice of reference pictures used for temporal
prediction. The underlying approach is to increase the amount of
accurate temporal information at the decoder when transmission
errors occur, to improve the video quality by using an efficient
combination of diverse motion fields. The proposed method
compensates for the small loss of coding efficiency at frame loss
rates as low as 3%. For a single frame-loss event the proposed
method can achieve up to 2 dB of gain in the affected frames
and an average quality gain of 0:84 dB for different error prone
conditions
Error resilience and concealment techniques for high-efficiency video coding
This thesis investigates the problem of robust coding and error concealment in High Efficiency Video Coding (HEVC). After a review of the current state of the art, a simulation study about error robustness, revealed that the HEVC has weak protection against network losses with significant impact on video quality degradation. Based on this evidence, the first contribution of this work is a new method to reduce the temporal dependencies between motion vectors, by improving the decoded video quality without compromising the compression efficiency. The second contribution of this thesis is a two-stage approach for reducing the mismatch of temporal predictions in case of video streams received with errors or lost data. At the encoding stage, the reference pictures are dynamically distributed based on a constrained Lagrangian rate-distortion optimization to reduce the number of predictions from a single reference. At the streaming stage, a prioritization algorithm, based on spatial dependencies, selects a reduced set of motion vectors to be transmitted, as side information, to reduce mismatched motion predictions at the decoder. The problem of error concealment-aware video coding is also investigated to enhance the overall error robustness. A new approach based on scalable coding and optimally error concealment selection is proposed, where the optimal error concealment modes are found by simulating transmission losses, followed by a saliency-weighted optimisation. Moreover, recovery residual information is encoded using a rate-controlled enhancement layer. Both are transmitted to the decoder to be used in case of data loss. Finally, an adaptive error resilience scheme is proposed to dynamically predict the video stream that achieves the highest decoded quality for a particular loss case. A neural network selects among the various video streams, encoded with different levels of compression efficiency and error protection, based on information from the video signal, the coded stream and the transmission network. Overall, the new robust video coding methods investigated in this thesis yield consistent quality gains in comparison with other existing methods and also the ones implemented in the HEVC reference software. Furthermore, the trade-off between coding efficiency and error robustness is also better in the proposed methods
Reference picture selection using checkerboard pattern for resilient video coding
The improved compression efficiency achieved by
the High Efficiency Video Coding (HEVC) standard has the
counter-effect of decreasing error resilience in transmission over
error-prone channels. To increase the error resilience of HEVC
streams, this paper proposes a checkerboard reference picture
selection method in order to reduce the prediction mismatch at
the decoder in case of frame losses. The proposed approach not
only allows to reduce the error propagation at the decoder, but
also enhances the quality of reconstructed frames by selectively
constraining the choice of reference pictures used for temporal
prediction. The underlying approach is to increase the amount of
accurate temporal information at the decoder when transmission
errors occur, to improve the video quality by using an efficient
combination of diverse motion fields. The proposed method
compensates for the small loss of coding efficiency at frame loss
rates as low as 3%. For a single frame-loss event the proposed
method can achieve up to 2 dB of gain in the affected frames
and an average quality gain of 0:84 dB for different error prone
conditions
A two-stage approach for robust HEVC coding and streaming
The increased compression ratios achieved by the High Efficiency Video Coding (HEVC) standard lead to reduced
robustness of coded streams, with increased susceptibility to
network errors and consequent video quality degradation. This
paper proposes a method based on a two-stage approach to
improve the error robustness of HEVC streaming, by reducing
temporal error propagation in case of frame loss. The prediction mismatch that occurs at the decoder after frame loss is reduced through the following two stages: (i) at the encoding stage, the reference pictures are dynamically selected based on constraining conditions and Lagrangian optimisation, which distributes the use of reference pictures, by reducing the number of prediction units (PUs) that depend on a single reference; (ii) at the streaming stage, a motion vector (MV) prioritisation algorithm, based on spatial dependencies, selects an optimal sub-set of MVs to be transmitted, redundantly, as side information to reduce mismatched MV predictions at the decoder. The simulation results show that the proposed method significantly reduces the effect of temporal error propagation. Compared to the reference HEVC, the proposed reference picture selection method is able to improve the video quality at low packet loss rates (e.g., 1%) using
the same bitrate, achieving quality gains up to 2.3 dB for 10%
of packet loss ratio. It is shown, for instance, that the redundant MVs are able to boost the performance achieving quality gains of 3 dB when compared to the reference HEVC, at the cost using 4% increase in total bitrate
Scalable Multiple Description Coding and Distributed Video Streaming over 3G Mobile Networks
In this thesis, a novel Scalable Multiple Description Coding (SMDC) framework is proposed. To address the bandwidth fluctuation, packet loss and heterogeneity problems in the wireless networks and further enhance the error resilience tools in Moving Pictures Experts Group 4 (MPEG-4), the joint design of layered coding (LC) and multiple description coding (MDC) is explored. It leverages a proposed distributed multimedia delivery mobile network (D-MDMN) to provide path diversity to combat streaming video outage due to handoff in Universal Mobile Telecommunications System (UMTS). The corresponding intra-RAN (Radio Access Network) handoff and inter-RAN handoff procedures in D-MDMN are studied in details, which employ the principle of video stream re-establishing to replace the principle of data forwarding in UMTS. Furthermore, a new IP (Internet Protocol) Differentiated Services (DiffServ) video marking algorithm is proposed to support the unequal error protection (UEP) of LC components of SMDC. Performance evaluation is carried through simulation using OPNET Modeler 9. 0. Simulation results show that the proposed handoff procedures in D-MDMN have better performance in terms of handoff latency, end-to-end delay and handoff scalability than that in UMTS. Performance evaluation of our proposed IP DiffServ video marking algorithm is also undertaken, which shows that it is more suitable for video streaming in IP mobile networks compared with the previously proposed DiffServ video marking algorithm (DVMA)
Multimedia over wireless ip networks:distortion estimation and applications.
2006/2007This thesis deals with multimedia communication over unreliable and resource
constrained IP-based packet-switched networks. The focus is on estimating, evaluating
and enhancing the quality of streaming media services with particular regard
to video services. The original contributions of this study involve mainly the
development of three video distortion estimation techniques and the successive
definition of some application scenarios used to demonstrate the benefits obtained
applying such algorithms. The material presented in this dissertation is the result
of the studies performed within the Telecommunication Group of the Department
of Electronic Engineering at the University of Trieste during the course of Doctorate
in Information Engineering.
In recent years multimedia communication over wired and wireless packet based
networks is exploding. Applications such as BitTorrent, music file sharing, multimedia
podcasting are the main source of all traffic on the Internet. Internet radio
for example is now evolving into peer to peer television such as CoolStreaming.
Moreover, web sites such as YouTube have made publishing videos on demand
available to anyone owning a home video camera. Another challenge in the multimedia
evolution is inside the house where videos are distributed over local WiFi
networks to many end devices around the house. More in general we are assisting
an all media over IP revolution, with radio, television, telephony and stored media
all being delivered over IP wired and wireless networks. All the presented applications
require an extreme high bandwidth and often a low delay especially for
interactive applications. Unfortunately the Internet and the wireless networks provide
only limited support for multimedia applications. Variations in network conditions
can have considerable consequences for real-time multimedia applications
and can lead to unsatisfactory user experience. In fact, multimedia applications
are usually delay sensitive, bandwidth intense and loss tolerant applications. In order
to overcame this limitations, efficient adaptation mechanism must be derived
to bridge the application requirements with the transport medium characteristics.
Several approaches have been proposed for the robust transmission of multimedia
packets; they range from source coding solutions to the addition of redundancy with forward error correction and retransmissions. Additionally, other techniques
are based on developing efficient QoS architectures at the network layer or at the
data link layer where routers or specialized devices apply different forwarding
behaviors to packets depending on the value of some field in the packet header.
Using such network architecture, video packets are assigned to classes, in order
to obtain a different treatment by the network; in particular, packets assigned to
the most privileged class will be lost with a very small probability, while packets
belonging to the lowest priority class will experience the traditional 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