671 research outputs found
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
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
Distributed video coding for wireless video sensor networks: a review of the state-of-the-art architectures
Distributed video coding (DVC) is a relatively new video coding architecture originated from two fundamental theorems namely, Slepian–Wolf and Wyner–Ziv. Recent research developments have made DVC attractive for applications in the emerging domain of wireless video sensor networks (WVSNs). This paper reviews the state-of-the-art DVC architectures with a focus on understanding their opportunities and gaps in addressing the operational requirements and application needs of WVSNs
Error resilient packet switched H.264 video telephony over third generation networks.
Real-time video communication over wireless networks is a challenging problem because
wireless channels suffer from fading, additive noise and interference, which translate
into packet loss and delay. Since modern video encoders deliver video packets with
decoding dependencies, packet loss and delay can significantly degrade the video quality
at the receiver. Many error resilience mechanisms have been proposed to combat packet
loss in wireless networks, but only a few were specifically designed for packet switched
video telephony over Third Generation (3G) networks.
The first part of the thesis presents an error resilience technique for packet switched
video telephony that combines application layer Forward Error Correction (FEC) with
rateless codes, Reference Picture Selection (RPS) and cross layer optimization. Rateless
codes have lower encoding and decoding computational complexity compared to traditional
error correcting codes. One can use them on complexity constrained hand-held
devices. Also, their redundancy does not need to be fixed in advance and any number of
encoded symbols can be generated on the fly. Reference picture selection is used to limit
the effect of spatio-temporal error propagation. Limiting the effect of spatio-temporal
error propagation results in better video quality. Cross layer optimization is used to
minimize the data loss at the application layer when data is lost at the data link layer.
Experimental results on a High Speed Packet Access (HSPA) network simulator for
H.264 compressed standard video sequences show that the proposed technique achieves
significant Peak Signal to Noise Ratio (PSNR) and Percentage Degraded Video Duration
(PDVD) improvements over a state of the art error resilience technique known as
Interactive Error Control (IEC), which is a combination of Error Tracking and feedback
based Reference Picture Selection. The improvement is obtained at a cost of higher
end-to-end delay.
The proposed technique is improved by making the FEC (Rateless code) redundancy
channel adaptive. Automatic Repeat Request (ARQ) is used to adjust the redundancy
of the Rateless codes according to the channel conditions. Experimental results show
that the channel adaptive scheme achieves significant PSNR and PDVD improvements
over the static scheme for a simulated Long Term Evolution (LTE) network.
In the third part of the thesis, the performance of the previous two schemes is
improved by making the transmitter predict when rateless decoding will fail. In this
case, reference picture selection is invoked early and transmission of encoded symbols
for that source block is aborted. Simulations for an LTE network show that this results
in video quality improvement and bandwidth savings.
In the last part of the thesis, the performance of the adaptive technique is improved
by exploiting the history of the wireless channel. In a Rayleigh fading wireless channel,
the RLC-PDU losses are correlated under certain conditions. This correlation is
exploited to adjust the redundancy of the Rateless code and results in higher Rateless
code decoding success rate and higher video quality. Simulations for an LTE network
show that the improvement was significant when the packet loss rate in the two wireless
links was 10%.
To facilitate the implementation of the proposed error resilience techniques in practical
scenarios, RTP/UDP/IP level packetization schemes are also proposed for each
error resilience technique.
Compared to existing work, the proposed error resilience techniques provide better
video quality. Also, more emphasis is given to implementation issues in 3G networks
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
Real-time interactive video streaming over lossy networks: high performance low delay error resilient algorithms
According to Cisco's latest forecast, two-thirds of the world's mobile data traffic and 62 percent of the consumer Internet traffic will be video data by the end of 2016. However, the wireless networks and Internet are unreliable, where the video traffic may undergo packet loss and delay. Thus robust video streaming over unreliable networks, i.e., Internet, wireless networks, is of great importance in facing this challenge. Specifically, for the real-time interactive video streaming applications, such as video conference and video telephony, the allowed end-to-end delay is limited, which makes the robust video streaming an even more difficult task. In this thesis, we are going to investigate robust video streaming for real-time interactive applications, where the tolerated end-to-end delay is limited. Intra macroblock refreshment is an effective tool to stop error propagations in the prediction loop of video decoder, whereas redundant coding is a commonly used method to prevent error from happening for video transmission over lossy networks. In this thesis two schemes that jointly use intra macroblock refreshment and redundant coding are proposed. In these schemes, in addition to intra coding, we proposed to add two redundant coding methods to enhance the transmission robustness of the coded bitstreams. The selection of error resilient coding tools, i.e., intra coding and/or redundant coding, and the parameters for redundant coding are determined using the end-to-end rate-distortion optimization. Another category of methods to provide error resilient capacity is using forward error correction (FEC) codes. FEC is widely studied to protect streamed video over unreliable networks, with Reed-Solomon (RS) erasure codes as its commonly used implementation method. As a block-based error correcting code, on the one hand, enlarging the block size can enhance the performance of the RS codes; on the other hand, large block size leads to long delay which is not tolerable for real-time video applications. In this thesis two sub-GOP (Group of Pictures, formed by I-frame and all the following P/B-frames) based FEC schemes are proposed to improve the performance of Reed-Solomon codes for real-time interactive video applications. The first one, named DSGF (Dynamic sub-GOP FEC Coding), is designed for the ideal case, where no transmission network delay is taken into consideration. The second one, named RVS-LE (Real-time Video Streaming scheme exploiting the Late- and Early-arrival packets), is more practical, where the video transmission network delay is considered, and the late- and early-arrival packets are fully exploited. Of the two approaches, the sub-GOP, which contains more than one video frame, is dynamically tuned and used as the RS coding block to get the optimal performance. For the proposed DSGF approach, although the overall error resilient performance is higher than the conventional FEC schemes, that protect the streamed video frame by frame, its video quality fluctuates within the Sub-GOP. To mitigate this problem, in this thesis, another real-time video streaming scheme using randomized expanding Reed-Solomon code is proposed. In this scheme, the Reed-Solomon coding block includes not only the video packets of the current frame, but also all the video packets of previous frames in the current group of pictures (GOP). At the decoding side, the parity-check equations of the current frameare jointly solved with all the parity-check equations of the previous frames. Since video packets of the following frames are not encompassed in the RS coding block, no delay will be caused for waiting for the video or parity packets of the following frames both at encoding and decoding sides. The main contribution of this thesis is investigating the trade-off between the video transmission delay caused by FEC encoding/decoding dependency, the FEC error-resilient performance, and the computational complexity. By leveraging the methods proposed in this thesis, proper error-resilient tools and system parameters could be selected based on the video sequence characteristics, the application requirements, and the available channel bandwidth and computational resources. For example, for the applications that can tolerate relatively long delay, sub-GOP based approach is a suitable solution. For the applications where the end-to-end delay is stringent and the computational resource is sufficient (e.g. CPU is fast), it could be a wise choice to use the randomized expanding Reed-Solomon code
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 kostengünstiger und
zuverlässiger
Kommunikation wächst stetig. Während wir uns selbst immer mehr von modernen
Kommunikationstechnologien abhängig machen, müssen die Ingenieure dieser
Technologien sowohl den Bedarf nach schneller Einführung neuer Produkte
befriedigen als auch die wachsende Komplexität der Systeme beherrschen.
Gerade die Übertragung multimedialer Inhalte wie Video und Audiodaten ist
nicht trivial. Einer der prominentesten Gründe dafür ist die
Unzuverlässigkeit heutiger Netzwerke, wie z.B.~dem Internet. Paketverluste
und schwankende Laufzeiten können die Darstellungsqualität massiv
beeinträchtigen. Wie jüngste Entwicklungen im Bereich der
Streaming-Protokolle zeigen, sind jedoch Qualität und Robustheit der
Übertragung effizient kontrollierbar, wenn Streamingprotokolle
Informationen über den Inhalt der transportierten Daten ausnutzen.
Existierende Ansätze, die den Inhalt von Multimediadatenströmen
beschreiben, sind allerdings meist auf einzelne Kompressionsverfahren
spezialisiert und verwenden berechnungsintensive Metriken. Das reduziert
ihren praktischen Nutzen deutlich. Außerdem erfordert der
Informationsaustausch eine enge Kooperation zwischen Applikationen und
Transportschichten. Da allerdings die Schnittstellen aktueller
Systemarchitekturen nicht darauf vorbereitet sind, müssen entweder die
Schnittstellen erweitert oder alternative Architekturkonzepte geschaffen
werden. Die Gefahr beider Varianten ist jedoch, dass sich die Komplexität
eines Systems dadurch weiter erhöhen kann.
Das zentrale Ziel dieser Dissertation ist es deshalb,
schichtenübergreifende Koordination bei gleichzeitiger Reduzierung der
Komplexität zu erreichen. Hier leistet die Arbeit zwei Beträge zum
aktuellen Stand der Forschung. Erstens definiert sie ein universelles
Modell zur Beschreibung von Inhaltsattributen, wie Wichtigkeiten und
Abhängigkeitsbeziehungen innerhalb eines Datenstroms. Transportschichten
können dieses Wissen zur effizienten Fehlerkontrolle verwenden. Zweitens
beschreibt die Arbeit das Noja Programmiermodell für multimediale
Middleware. Noja definiert Abstraktionen zur Übertragung und Kontrolle
multimedialer Ströme, die die Koordination von Streamingprotokollen mit
Applikationen ermöglichen. Zum Beispiel können Programmierer geeignete
Fehlersemantiken und Kommunikationstopologien auswählen und den konkreten
Fehlerschutz dann zur Laufzeit verfeinern und kontrolliere
- …