414 research outputs found
Packet loss visibility across SD, HD, 3D, and UHD video streams
The trend towards video streaming with increased spatial resolutions and dimensions, SD, HD, 3D, and 4kUHD, even for portable devices has important implications for displayed video quality. There is an interplay between packetization, packet loss visibility, choice of codec, and viewing conditions, which implies that prior studies at lower resolutions may not be as relevant. This paper presents two sets of experiments, the one at a Variable BitRate (VBR) and the other at a Constant BitRate (CBR), which highlight different aspects of the interpretation. The latter experiments also compare and contrast encoding with either an H.264 or an High Efficiency Video Coding (HEVC) codec, with all results recorded as objective Mean Opinion Score (MOS). The video quality assessments will be of interest to those considering: the bitrates and expected quality in error-prone environments; or, in fact, whether to use a reliable transport protocol to prevent all errors, at a cost in jitter and latency, rather than tolerate low levels of packet errors
Effect of Video Streaming SpaceâTime Characteristics on Quality of Transmission over Wireless Telecommunication Networks
The spate in popularity of multimedia applications
has led to the need for optimization of bandwidth allocation
and usage in telecommunication networks. Modern
telecommunication networks should by their definition be able to maintain the quality of different applications with different Quality of Service (QoS) levels. QoS requirements are generally dependent on the parameters of network and
application layers of the OSI model. At the application layer QoS depends on factors such as resolution, bit rate, frame rate, video type, audio codecs, etc. At the network layer, distortions such as delay, jitter, packet loss, etc. are introduced. This paper presents simulation results of modeling video streaming over wireless communications networks. The differences in spatial and time characteristics of the different subject groups
were taken into account. Analysis of the influence of bit error rate (BER) and bit rate for video quality is also presented.
Simulation showed that different video subject groups affect
the perceived quality differently when transmitted over
networks. We show conclusively that in a transmission network
with a small error probabilities (BER = 10-6, BER = 10-5), the
minimum bit rate (128 kbps) guarantees an acceptable video
quality, corresponding to MOS > 3 for all types of frames
Video Tester -- A multiple-metric framework for video quality assessment over IP networks
This paper presents an extensible and reusable framework which addresses the
problem of video quality assessment over IP networks. The proposed tool
(referred to as Video-Tester) supports raw uncompressed video encoding and
decoding. It also includes different video over IP transmission methods (i.e.:
RTP over UDP unicast and multicast, as well as RTP over TCP). In addition, it
is furnished with a rich set of offline analysis capabilities. Video-Tester
analysis includes QoS and bitstream parameters estimation (i.e.: bandwidth,
packet inter-arrival time, jitter and loss rate, as well as GOP size and
I-frame loss rate). Our design facilitates the integration of virtually any
existing video quality metric thanks to the adopted Python-based modular
approach. Video-Tester currently provides PSNR, SSIM, ITU-T G.1070 video
quality metric, DIV and PSNR-based MOS estimations. In order to promote its use
and extension, Video-Tester is open and publicly available.Comment: 5 pages, 5 figures. For the Google Code project, see
http://video-tester.googlecode.com
Content-Aware Multimedia Communications
The demands for fast, economic and reliable dissemination of multimedia
information are steadily growing within our society. While people and
economy increasingly rely on communication technologies, engineers still
struggle with their growing complexity.
Complexity in multimedia communication originates from several sources. The
most prominent is the unreliability of packet networks like the Internet.
Recent advances in scheduling and error control mechanisms for streaming
protocols have shown that the quality and robustness of multimedia delivery
can be improved significantly when protocols are aware of the content they
deliver. However, the proposed mechanisms require close cooperation between
transport systems and application layers which increases the overall system
complexity. Current approaches also require expensive metrics and focus on
special encoding formats only. A general and efficient model is missing so
far.
This thesis presents efficient and format-independent solutions to support
cross-layer coordination in system architectures. In particular, the first
contribution of this work is a generic dependency model that enables
transport layers to access content-specific properties of media streams,
such as dependencies between data units and their importance. The second
contribution is the design of a programming model for streaming
communication and its implementation as a middleware architecture. The
programming model hides the complexity of protocol stacks behind simple
programming abstractions, but exposes cross-layer control and monitoring
options to application programmers. For example, our interfaces allow
programmers to choose appropriate failure semantics at design time while
they can refine error protection and visibility of low-level errors at
run-time.
Based on some examples we show how our middleware simplifies the
integration of stream-based communication into large-scale application
architectures. An important result of this work is that despite cross-layer
cooperation, neither application nor transport protocol designers
experience an increase in complexity. Application programmers can even
reuse existing streaming protocols which effectively increases system
robustness.Der Bedarf unsere Gesellschaft nach kostengĂŒnstiger und
zuverlÀssiger
Kommunikation wÀchst stetig. WÀhrend wir uns selbst immer mehr von modernen
Kommunikationstechnologien abhĂ€ngig machen, mĂŒssen die Ingenieure dieser
Technologien sowohl den Bedarf nach schneller EinfĂŒhrung neuer Produkte
befriedigen als auch die wachsende KomplexitÀt der Systeme beherrschen.
Gerade die Ăbertragung multimedialer Inhalte wie Video und Audiodaten ist
nicht trivial. Einer der prominentesten GrĂŒnde dafĂŒr ist die
UnzuverlÀssigkeit heutiger Netzwerke, wie z.B.~dem Internet. Paketverluste
und schwankende Laufzeiten können die DarstellungsqualitÀt massiv
beeintrĂ€chtigen. Wie jĂŒngste Entwicklungen im Bereich der
Streaming-Protokolle zeigen, sind jedoch QualitÀt und Robustheit der
Ăbertragung effizient kontrollierbar, wenn Streamingprotokolle
Informationen ĂŒber den Inhalt der transportierten Daten ausnutzen.
Existierende AnsÀtze, die den Inhalt von Multimediadatenströmen
beschreiben, sind allerdings meist auf einzelne Kompressionsverfahren
spezialisiert und verwenden berechnungsintensive Metriken. Das reduziert
ihren praktischen Nutzen deutlich. AuĂerdem erfordert der
Informationsaustausch eine enge Kooperation zwischen Applikationen und
Transportschichten. Da allerdings die Schnittstellen aktueller
Systemarchitekturen nicht darauf vorbereitet sind, mĂŒssen entweder die
Schnittstellen erweitert oder alternative Architekturkonzepte geschaffen
werden. Die Gefahr beider Varianten ist jedoch, dass sich die KomplexitÀt
eines Systems dadurch weiter erhöhen kann.
Das zentrale Ziel dieser Dissertation ist es deshalb,
schichtenĂŒbergreifende Koordination bei gleichzeitiger Reduzierung der
KomplexitÀt zu erreichen. Hier leistet die Arbeit zwei BetrÀge zum
aktuellen Stand der Forschung. Erstens definiert sie ein universelles
Modell zur Beschreibung von Inhaltsattributen, wie Wichtigkeiten und
AbhÀngigkeitsbeziehungen innerhalb eines Datenstroms. Transportschichten
können dieses Wissen zur effizienten Fehlerkontrolle verwenden. Zweitens
beschreibt die Arbeit das Noja Programmiermodell fĂŒr multimediale
Middleware. Noja definiert Abstraktionen zur Ăbertragung und Kontrolle
multimedialer Ströme, die die Koordination von Streamingprotokollen mit
Applikationen ermöglichen. Zum Beispiel können Programmierer geeignete
Fehlersemantiken und Kommunikationstopologien auswÀhlen und den konkreten
Fehlerschutz dann zur Laufzeit verfeinern und kontrolliere
Service Platform for Converged Interactive Broadband Broadcast and Cellular Wireless
A converged broadcast and telecommunication
service platform is presented that is able to create, deliver, and
manage interactive, multimedia content and services for consumption
on three different terminal types. The motivations of
service providers for designing converged interactive multimedia
services, which are crafted for their individual requirements, are
investigated. The overall design of the system is presented with
particular emphasis placed on the operational features of each
of the sub-systems, the flows of media and metadata through the
sub-systems and the formats and protocols required for inter-communication
between them. The key features of tools required for
creating converged interactive multimedia content for a range of
different end-user terminal types are examined. Finally possible
enhancements to this system are discussed. This study is of particular
interest to those organizations currently conducting trials
and commercial launches of DVB-H services because it provides
them with an insight of the various additional functions required
in the service provisioning platforms to provide fully interactive
services to a range of different mobile terminal types
EXPERIMENTS ON VIDEO STREAMING OVER COMPUTER NETWORKS
Video traffic (including streaming video service) is dominating the Internet traffic today. Video can be streamed using a dedicated server, a content delivery network (CDN), or peer-to-peer (P2P) overlays across a network. Video can be transmitted in multiple formats and at different resolutions. Video is also being distributed to a variety of devices (fixed and mobile)
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