Rozprawy doktorskie: „Adaptive video coding”

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Hosking, Brett. "Adaptive resolution video coding". Thesis, University of Bristol, 2016. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.707749.

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Su, Jonathan K. "Adaptive rate-constrained transform video coding". Diss., Georgia Institute of Technology, 1997. http://hdl.handle.net/1853/13364.

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Abrahamsson, Anna. "Variance Adaptive Quantization and Adaptive Offset Selection in High Efficiency Video Coding". Thesis, Uppsala universitet, Avdelningen för systemteknik, 2016. http://urn.kb.se/resolve?urn=urn:nbn:se:uu:diva-278155.

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Video compression uses encoding to reduce the number of bits that are used forrepresenting a video file in order to store and transmit it at a smaller size. Adecoder reconstructs the received data into a representation of the original video.Video coding standards determines how the video compression should beconducted and one of the latest standards is High Efficiency Video Coding (HEVC).One technique that can be used in the encoder is variance adaptive quantizationwhich improves the subjective quality in videos. The technique assigns lowerquantization parameter values to parts of the frame with low variance to increasequality, and vice versa. Another part of the encoder is the sample adaptive offsetfilter, which reduces pixel errors caused by the compression. In this project, thevariance adaptive quantization technique is implemented in the Ericsson researchHEVC encoder c65. Its functionality is verified by subjective evaluation. It isinvestigated if the sample adaptive offset can exploit the adjusted quantizationparameters values when reducing pixel errors to improve compression efficiency. Amodel for this purpose is developed and implemented in c65. Data indicates thatthe model can increase the error reduction in the sample adaptive offset. However,the difference in performance of the model compared to a reference encoder is notsignificant.

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Jónsson, Ragner H. "Adaptive subband coding of video using probability distribution models". Diss., Georgia Institute of Technology, 1994. http://hdl.handle.net/1853/14453.

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Gu, Ye. "Wavelet-based adaptive video coding for packet-switching networks". Thesis, Massachusetts Institute of Technology, 1995. http://hdl.handle.net/1721.1/37021.

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Chung, Wilson C. "Adaptive subband video coding in a rate-distortion-constrained framework". Diss., Georgia Institute of Technology, 1996. http://hdl.handle.net/1853/15459.

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De, La Rocha Gomes-Arevalillo Alfonso. "Investigating the Adaptive Loop Filter in Next Generation Video Coding". Thesis, KTH, Skolan för elektro- och systemteknik (EES), 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-201141.

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Current trends on video technologies and services are demanding higher bit rates, highervideo resolutions and better video qualities. This issue results in the need of a new generationof video coding techniques to increase the quality and compression rates of previous standards.Since the release of HEVC, ITU-T VCEG and ISO/IEC MPEG have been studying the potentialneed for standardization of future video coding technologies with a compression capability thatsignificantly exceeds the ones from current standards. These new e↵orts of standardization andcompression enhancements are being implemented and evaluated over a software test modelknown under the name of Joint Exploration Model (JEM). One of the blocks being explored inJEM is an Adaptive Loop Filter (ALF) at the end of each frame’s processing flow. ALF aimsto minimize the error between original pixels and decoded pixels using Wiener-based adaptivefilter coefficients, reporting, in its JEM’s implementation, improvements of around a 1% in theBD MS-SSIM rate. A lot of e↵orts have been devoted on improving this block over the pastyears. However, current ALF implementations do not consider the potential use of adaptive QPalgorithms at the encoder. Adaptive QP algorithms enable the use of di↵erent quality levels forthe coding of di↵erent parts of a frame to enhance its subjective quality.In this thesis, we explore potential improvements over di↵erent dimensions of JEM’s AdaptiveLoop Filter block considering the potential use of adaptive QP algorithms. In the document, weexplore a great gamut of modification over ALF processing stages, being the ones with betterresults (i) a QP-aware implementation of ALF were the filter coefficients estimation, the internalRD-optimization and the CU-level flag decision process are optimized for the use of adaptiveQP, (ii) the optimization of ALF’s standard block activity classification stage through the useof CU-level information given by the di↵erent QPs used in a frame, and (iii) the optimizationof ALF’s standard block activity classification stage in B-frames through the application of acorrection weight on coded, i.e. not predicted, blocks of B-frames. These ALF modificationscombined obtained improvements of a 0.419% on average for the BD MS-SSIM rate in the lumachannel, showing each modification individual improvements of a 0.252%, 0.085% and 0.082%,respectively. Thus, we concluded the importance of optimizing ALF for the potential use ofadaptive-QP algorithms in the encoder, and the benefits of considering CU-level and frame-levelmetrics in ALF’s block classification stage.
Utvecklingen inom video-teknologi och service kräver högre bithastighet, högre videoupplösningoch bättre kvalitet. Problemet kräver en ny generation av kodning och tekniker för att ökakvaliteten och komprimeringsgraden utöver vad tidigare teknik kunnat prestera. Sedan lanseringenav HEVC har ITU-T VCEG och ISO/IEC MPEG studerat ett eventuellt behov av standardiseringav framtida video-kodings tekniker med kompressions kapacitet som vida överstigerdagens system. Dessa försök till standardisering och kompressionsframsteg har implementeratsoch utvärderats inom ramen för en mjukvara testmodell som kallas Joint Exploration Model(JEM). Ett av områdena som undersöks inom ramen för JEM är adaptiva loopfilter (ALF) somläggs till i slutet av varje bilds processflöde. ALF har som mål att minimera felet mellan originalpixel och avkodad pixel genom Wiener-baserade adaptiva filter-koefficienter. Mycket kraft harlagts på att förbättra detta område under de senaste åren. Men, nuvarande ALF-appliceringbeaktar inte potentialen av att använda adaptiva QP algoritmer i videokodaren. Adaptiva QPalgoritmer tillåter användningen av olika kvalitet på kodning av olika delar av bilden för attförbättra den subjektiva kvaliteten.I föreliggande uppsats kommer vi undersöka den potentiella förbättringen av JEM:s adaptivaloopfilter som kan uppnås genom att använda adaptiva QP algoritmer. I uppsatsen kommervi undersöka ett stort antal modifikationer i ALF:s process-stadier, för att ta reda på vilkenmodifikationer som har bäst resultat: (i) en QP-medveten implementering av ALF där filterkoefficientensuppskattning av den interna RD-optimeringen och CU-nivåns flaggbeslutsprocessär optimerade för användnngen av adaptiv QP, (ii) optimeringen av ALF:s standard blockaktivitets klassificerings stadie genom användning av CU-nivå-information producerad av deolika QP:n som används i en bild, och (iii) optimeringen av ALF:s standard block aktivitetsklassificerings stadier i B-bilders genom applicering av en korrektursvikt i tidigare kod, d.v.sej förutsedda, block av B-bilder. När dessa ALF modifikationer kombinerades förbättradesi genomsnitt BD MS-SSIM hastigheten i luma kanalen med 0.419%, där varje modifikationförbättrade med 0.252%, 0.085% och 0.082% var. Därigenom drog vi slutstatsen att det är viktigtatt optimera ALF för det potentiella användandet av adaptiva QP-algoritmer i kodaren, ochfördelarna av att beakta CU-nivåmätningar och bild-nivåmätningar i ALF:s block klassificeringsstadie.

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MORALES, FIGUEROA AMPARITO ALEXANDRA. "Adaptive Video Coding to Optimize Video Streaming in Internet and Wireless Ad-hoc Networks". Doctoral thesis, Università degli studi di Pavia, 2017. http://hdl.handle.net/11571/1203302.

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Abraham, Arun S. "Bandwidth-aware video transmission with adaptive image scaling". [Gainesville, Fla.] : University of Florida, 2003. http://purl.fcla.edu/fcla/etd/UFE0001221.

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Shu, Yan. "Coding of motion picture residues without adaptive spatial transforms". Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1999. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape7/PQDD_0030/MQ47477.pdf.

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Drevery, Chad W. J. "Adaptive sampling and interpolation methods for digital image and video coding". Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1999. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape8/PQDD_0015/MQ48060.pdf.

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Sprljan, Nikola. "A flexible scalable video coding framework with adaptive spatio-temporal decompositions". Thesis, Queen Mary, University of London, 2006. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.434834.

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Wan, Wade K. (Wade Keith) 1973. "Adaptive format conversion information as enhancement data for scalable video coding". Thesis, Massachusetts Institute of Technology, 2002. http://hdl.handle.net/1721.1/29903.

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Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2002.
Includes bibliographical references (p. 143-145).
Scalable coding techniques can be used to efficiently provide multicast video service and involve transmitting a single independently coded base layer and one or more dependently coded enhancement layers. Clients can decode the base layer bitstream and none, some or all of the enhancement layer bitstreams to obtain video quality commensurate with their available resources. In many scalable coding algorithms, residual coding information is the only type of data that is coded in the enhancement layers. However, since the transmitter has access to the original sequence, it can adaptively select different format conversion methods for different regions in an intelligent manner. This adaptive format conversion information can then be transmitted as enhancement data to assist processing at the decoder. The use of adaptive format conversion has not been studied in detail and this thesis examines when and how it can be used for scalable video compression. A new scalable codec is developed in this thesis that can utilize adaptive format conversion information and/or residual coding information as enhancement data. This codec was used in various simulations to investigate different aspects of adaptive format conversion such as the effect of the base layer, a comparison of adaptive format conversion and residual coding, and the use of both adaptive format conversion and residual coding.
(cont.) The experimental results show adaptive format conversion can provide video scalability at low enhancement bitrates not possible with residual coding and also assist residual coding at higher enhancement layer bitrates. This thesis also discusses the application of adaptive format conversion to the migration path for digital television. Adaptive format conversion is well-suited to the unique problems of the migration path and can provide initial video scalability as well as assist a future migration path.
by Wade K. Wan.
Ph.D.

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Lawan, Sagir. "Adaptive intra refresh for robust wireless multi-view video". Thesis, Brunel University, 2016. http://bura.brunel.ac.uk/handle/2438/13078.

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Mobile wireless communication technology is a fast developing field and every day new mobile communication techniques and means are becoming available. In this thesis multi-view video (MVV) is also refers to as 3D video. Thus, the 3D video signals through wireless communication are shaping telecommunication industry and academia. However, wireless channels are prone to high level of bit and burst errors that largely deteriorate the quality of service (QoS). Noise along the wireless transmission path can introduce distortion or make a compressed bitstream lose vital information. The error caused by noise progressively spread to subsequent frames and among multiple views due to prediction. This error may compel the receiver to pause momentarily and wait for the subsequent INTRA picture to continue decoding. The pausing of video stream affects the user's Quality of Experience (QoE). Thus, an error resilience strategy is needed to protect the compressed bitstream against transmission errors. This thesis focuses on error resilience Adaptive Intra Refresh (AIR) technique. The AIR method is developed to make the compressed 3D video more robust to channel errors. The process involves periodic injection of Intra-coded macroblocks in a cyclic pattern using H.264/AVC standard. The algorithm takes into account individual features in each macroblock and the feedback information sent by the decoder about the channel condition in order to generate an MVV-AIR map. MVV-AIR map generation regulates the order of packets arrival and identifies the motion activities in each macroblock. Based on the level of motion activity contained in each macroblock, the MVV-AIR map classifies frames as high or low motion macroblocks. A proxy MVV-AIR transcoder is used to validate the efficiency of the generated MVV-AIR map. The MVV-AIR transcoding algorithm uses spatial and views downscaling scheme to convert from MVV to single view. Various experimental results indicate that the proposed error resilient MVV-AIR transcoder technique effectively improves the quality of reconstructed 3D video in wireless networks. A comparison of MVV-AIR transcoder algorithm with some traditional error resilience techniques demonstrates that MVV-AIR algorithm performs better in an error prone channel. Results of simulation revealed significant improvements in both objective and subjective qualities. No additional computational complexity emanates from the scheme while the QoS and QoE requirements are still fully met.

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Zhao, Jing. "Information theoretic approach for low-complexity adaptive motion estimation". [Gainesville, Fla.] : University of Florida, 2005. http://purl.fcla.edu/fcla/etd/UFE0013068.

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Rasch, Jennifer [Verfasser]. "Signal Adaptive Methods To Optimize Prediction Signals in Video Coding / Jennifer Rasch". Berlin : epubli, 2019. http://d-nb.info/1199686522/34.

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Andelin, Travis L. "Quality Selection for Dynamic Adaptive Streaming over HTTP with Scalable Video Coding". BYU ScholarsArchive, 2011. https://scholarsarchive.byu.edu/etd/2683.

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Video streaming on the Internet is increasingly using Dynamic Adaptive Streaming over HTTP (DASH), in which the video is converted into various quality levels and divided into two-second segments. A client can then adjust its video quality over time by choosing to download the appropriate quality level for a given segment using standard HTTP. Scalable Video Coding (SVC) is a promising enhancement to the DASH protocol. With SVC, segments are divided into subset bitstream blocks. At playback, blocks received for a given segment are combined to additively increase the current quality. Unlike traditional DASH, which downloads segments serially, this encoding creates a large space of possible ways to download a video; for example, if given a variable download rate, when should the client try to maximize the current segment's video quality, and when should it instead play it safe and ensure a minimum level of quality for future segments? In this work, we examine the impact of SVC on the client's quality selection policy, with the goal of maximizing a performance metric quantifying user satisfaction. We use acombination of analysis, dynamic programming, and simulation to show that, in many cases, a client should use a diagonal quality selection policy, balancing both of the aforementioned concerns, and that the slope of the best policy flattens out as the variation in download rateincreases.

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Silva, Cauane Blumenberg. "Adaptive tiling algorithm based on highly correlated picture regions for the HEVC standard". reponame:Biblioteca Digital de Teses e Dissertações da UFRGS, 2014. http://hdl.handle.net/10183/96040.

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Esta dissertação de mestrado propõe um algoritmo adaptativo que é capaz de dinamicamente definir partições tile para quadros intra- e inter-preditos com o objetivo de reduzir o impacto na eficiência de codificação. Tiles são novas ferramentas orientadas ao paralelismo que integram o padrão de codificação de vídeos de alta eficiência (HEVC – High Efficiency Video Coding standard), as quais dividem o quadro em regiões retangulares independentes que podem ser processadas paralelamente. Para viabilizar o paralelismo, os tiles quebram as dependências de codificação através de suas bordas, gerando impactos na eficiência de codificação. Este impacto pode ser ainda maior caso os limites dos tiles dividam regiões altamente correlacionadas do quadro, porque a maior parte das ferramentas de codificação usam informações de contexto durante o processo de codificação. Assim, o algoritmo proposto agrupa as regiões do quadro que são altamente correlacionadas dentro de um mesmo tile para reduzir o impacto na eficiência de codificação que é inerente ao uso de tiles. Para localizar as regiões altamente correlacionadas do quadro de uma maneira inteligente, as características da imagem e também as informações de codificação são analisadas, gerando mapas de particionamento que servem como parâmetro de entrada para o algoritmo. Baseado nesses mapas, o algoritmo localiza as quebras naturais de contexto presentes nos quadros do vídeo e define os limites dos tiles nessas regiões. Dessa maneira, as quebras de dependência causadas pelas bordas dos tiles coincidem com as quebras de contexto naturais do quadro, minimizando as perdas na eficiência de codificação causadas pelo uso dos tiles. O algoritmo proposto é capaz de reduzir mais de 0.4% e mais de 0.5% o impacto na eficiência de codificação causado pelos tiles em quadros intra-preditos e inter-preditos, respectivamente, quando comparado com tiles uniformes.
This Master Thesis proposes an adaptive algorithm that is able to dynamically choose suitable tile partitions for intra- and inter-predicted frames in order to reduce the impact on coding efficiency arising from such partitioning. Tiles are novel parallelismoriented tools that integrate the High Efficiency Video Coding (HEVC) standard, which divide the frame into independent rectangular regions that can be processed in parallel. To enable the parallelism, tiles break the coding dependencies across their boundaries leading to coding efficiency impacts. These impacts can be even higher if tile boundaries split highly correlated picture regions, because most of the coding tools use context information during the encoding process. Hence, the proposed algorithm clusters the highly correlated picture regions inside the same tile to reduce the inherent coding efficiency impact of using tiles. To wisely locate the highly correlated picture regions, image characteristics and encoding information are analyzed, generating partitioning maps that serve as the algorithm input. Based on these maps, the algorithm locates the natural context break of the picture and defines the tile boundaries on these key regions. This way, the dependency breaks caused by the tile boundaries match the natural context breaks of a picture, then minimizing the coding efficiency losses caused by the use of tiles. The proposed adaptive tiling algorithm, in some cases, provides over 0.4% and over 0.5% of BD-rate savings for intra- and inter-predicted frames respectively, when compared to uniform-spaced tiles, an approach which does not consider the picture context to define the tile partitions.

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Indra, Isara. "Very low bit rate video coding using adaptive nonuniform sampling and matching pursuit". Diss., Georgia Institute of Technology, 2001. http://hdl.handle.net/1853/15779.

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Luo, Yi. "Fast Adaptive Block Based Motion Estimation for Video Compression". Ohio University / OhioLINK, 2009. http://rave.ohiolink.edu/etdc/view?acc_num=ohiou1240002690.

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Amiri, Delaram. "Bilateral and adaptive loop filter implementations in 3D-high efficiency video coding standard". Thesis, Purdue University, 2016. http://pqdtopen.proquest.com/#viewpdf?dispub=10109195.

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In this thesis, we describe a different implementation for in loop filtering method for 3D-HEVC. First we propose the use of adaptive loop filtering (ALF) technique for 3D-HEVC standard in-loop filtering. This filter uses Wiener-based method to minimize the Mean Squared Error between filtered pixel and original pixels. The performance of adaptive loop filter in picture based level is evaluated. Results show up to of 0.2 dB PSNR improvement in Luminance component for the texture and 2.1 dB for the depth. In addition, we obtain up to 0.1 dB improvement in Chrominance component for the texture view after applying this filter in picture based filtering. Moreover, a design of an in-loop filtering with Fast Bilateral Filter for 3D-HEVC standard is proposed. Bilateral filter is a filter that smoothes an image while preserving strong edges and it can remove the artifacts in an image. Performance of the bilateral filter in picture based level for 3D-HEVC is evaluated. Test model HTM- 6.2 is used to demonstrate the results. Results show up to of 20 percent of reduction in processing time of 3D-HEVC with less than affecting PSNR of the encoded 3D video using Fast Bilateral Filter.

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Aklouf, Mourad. "Video for events : Compression and transport of the next generation video codec". Electronic Thesis or Diss., université Paris-Saclay, 2022. http://www.theses.fr/2022UPASG029.

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L'acquisition et la diffusion de contenus avec une latence minimale sont devenus essentiel dans plusieurs domaines d'activités tels que la diffusion d'évènements sportifs, la vidéoconférence, la télé-présence, la télé-opération de véhicules ou le contrôle à distance de systèmes. L'industrie de la diffusion en direct a connu une croissance en 2020, et son importance va encore croitre au cours des prochaines années grâce à l'émergence de nouveaux codecs vidéo à haute efficacité reposant sur le standard Versatile Video Coding(VVC)et à la cinquième génération de réseaux mobiles (5G).Les méthodes de streaming de type HTTP Adaptive Streaming (HAS) telles que MPEG-DASH, grâce aux algorithmes d'adaptation du débit de transmission de vidéo compressée, se sont révélées très efficaces pour améliorer la qualité d'expérience (QoE) dans un contexte de vidéo à la demande (VOD).Cependant, dans les applications où la latence est critique, minimiser le délai entre l'acquisition de l'image et son affichage au récepteur est essentiel. La plupart des algorithmes d'adaptation de débit sont développés pour optimiser la transmission vidéo d'un serveur situé dans le cœur de réseau vers des clients mobiles. Dans les applications nécessitant un streaming à faible latence, le rôle du serveur est joué par un terminal mobile qui va acquérir, compresser et transmettre les images via une liaison montante comportant un canal radio vers un ou plusieurs clients. Les approches d'adaptation de débit pilotées par le client sont par conséquent inadaptées. De plus, les HAS, pour lesquelles la prise de décision se fait avec une périodicité de l'ordre de la seconde ne sont pas suffisamment réactives lors d'une mobilité importante du serveur et peuvent engendrer des délais importants. Il est donc essentiel d'utiliser une granularité d'adaptation très fine afin de réduire le délai de bout-en-bout. En effet, la taille réduite des tampons d'émission et de réception afin de minimiser la latence rend plus délicate l'adaptation du débit dans notre cas d'usage. Lorsque la bande passante varie avec une constante de temps plus petite que la période avec laquelle la régulation est faite, les mauvaises décisions de débit de transmission peuvent induire un surcroit de latence important.L'objet de cette thèse est d'apporter des éléments de réponse à la problématique de la transmission vidéo à faible latence depuis des terminaux (émetteurs) mobiles. Nous présentons d'abord un algorithme d'adaptation de débit image-par-image pour la diffusion à faible latence. Une approche de type Model Predictive Control (MPC) est proposée pour déterminer le débit de codage de chaque image à transmettre. Cette approche utilise des informations relatives au niveau de tampon de l'émetteur et aux caractéristiques du canal de transmission. Les images étant codées en direct, un modèle reliant le paramètre de quantification (QP) au débit de sortie du codeur vidéo est nécessaire. Nous avons donc proposé un nouveau modèle reliant le débit au paramètre de quantification et à la distorsion de l'image précédente. Ce modèle fournit de bien meilleurs résultats dans le contexte d'une décision prise image par image du débit de codage que les modèle de référence de la littérature.En complément des techniques précédentes, nous avons également proposé des outils permettant de réduire la complexité de codeurs vidéo tels que VVC. La version actuelle du codeur VVC (VTM10) a un temps d'exécution neuf fois supérieur à celui du codeur HEVC. Par conséquent, le codeur VVC n'est pas adapté aux applications de codage et diffusion en temps réel sur les plateformes actuellement disponibles. Dans ce contexte, nous présentons une méthode systématique, de type branch-and-prune, permettant d'identifier un ensemble d'outils de codage pouvant être désactivés tout en satisfaisant une contrainte sur l'efficacité de codage. Ce travail contribue à la réalisation d'un codeur VVC temps réel
The acquisition and delivery of video content with minimal latency has become essential in several business areas such as sports broadcasting, video conferencing, telepresence, remote vehicle operation, or remote system control. The live streaming industry has grown in 2020 and it will expand further in the next few years with the emergence of new high-efficiency video codecs based on the Versatile Video Coding (VVC) standard and the fifth generation of mobile networks (5G).HTTP Adaptive Streaming (HAS) methods such as MPEG-DASH, using algorithms to adapt the transmission rate of compressed video, have proven to be very effective in improving the quality of experience (QoE) in a video-on-demand (VOD) context.Nevertheless, minimizing the delay between image acquisition and display at the receiver is essential in applications where latency is critical. Most rate adaptation algorithms are developed to optimize video transmission from a server situated in the core network to mobile clients. In applications requiring low-latency streaming, such as remote control of drones or broadcasting of sports events, the role of the server is played by a mobile terminal. The latter will acquire, compress, and transmit the video and transmit the compressed stream via a radio access channel to one or more clients. Therefore, client-driven rate adaptation approaches are unsuitable in this context because of the variability of the channel characteristics. In addition, HAS, for which the decision-making is done with a periodicity of the order of a second, are not sufficiently reactive when the server is moving, which may generate significant delays. It is therefore important to use a very fine adaptation granularity in order to reduce the end-to-end delay. The reduced size of the transmission and reception buffers (to minimize latency) makes it more difficult to adapt the throughput in our use case. When the bandwidth varies with a time constant smaller than the period with which the regulation is made, bad transmission rate decisions can induce a significant latency overhead.The aim of this thesis is to provide some answers to the problem of low-latency delivery of video acquired, compressed, and transmitted by mobile terminals. We first present a frame-by-frame rate adaptation algorithm for low latency broadcasting. A Model Predictive Control (MPC) approach is proposed to determine the coding rate of each frame to be transmitted. This approach uses information about the buffer level of the transmitter and about the characteristics of the transmission channel. Since the frames are coded live, a model relating the quantization parameter (QP) to the output rate of the video encoder is required. Hence, we have proposed a new model linking the rate to the QP of the current frame and to the distortion of the previous frame. This model provides much better results in the context of a frame-by-frame decision on the coding rate than the reference models in the literature.In addition to the above techniques, we have also proposed tools to reduce the complexity of video encoders such as VVC. The current version of the VVC encoder (VTM10) has an execution time nine times higher than that of the HEVC encoder. Therefore, the VVC encoder is not suitable for real-time encoding and streaming applications on currently available platforms. In this context, we present a systematic branch-and-prune method to identify a set of coding tools that can be disabled while satisfying a constraint on coding efficiency. This work contributes to the realization of a real-time VVC coder

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Liu, Sam J. "Low bit-rate image and video compression using adaptive segmentation and quantization". Diss., Georgia Institute of Technology, 1993. http://hdl.handle.net/1853/14850.

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Man, Hong. "On efficiency and robustness of adaptive quantization for subband coding of images and video sequences". Diss., Georgia Institute of Technology, 1999. http://hdl.handle.net/1853/15003.

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Ye, Zakaria. "Analyse de Performance des Services de Vidéo Streaming Adaptatif dans les Réseaux Mobiles". Thesis, Avignon, 2017. http://www.theses.fr/2017AVIG0219/document.

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Le trafic vidéo a subi une augmentation fulgurante sur Internet ces dernières années. Pour pallier à cette importante demande de contenu vidéo, la technologie du streaming adaptatif sur HTTP est utilisée. Elle est devenue par ailleurs très populaire car elle a été adoptée par les différents acteurs du domaine de la vidéo streaming. C’est une technologie moins couteuse qui permet aux fournisseurs de contenu, la réutilisation des serveurs web et des caches déjà déployés. En plus, elle est exempt de tout blocage car elle traverse facilement les pare-feux et les translations d’adresses sur Internet. Dans cette thèse, nous proposons une nouvelle méthode de vidéo streaming adaptatif appelé “Backward-Shifted Coding (BSC)”. Il se veut être une solution complémentaire au standard DASH, le streaming adaptatif et dynamique utilisant le protocole HTTP. Nous allons d’abord décrire ce qu’est la technologie BSC qui se base sur le codec (encodeur décodeur) à multi couches SVC, un algorithme de compression extensible ou évolutif. Nous détaillons aussi l’implémentation de BSC dans un environnement DASH. Ensuite,nous réalisons une évaluation analytique de BSC en utilisant des résultats standards de la théorie des files d’attente. Les résultats de cette analyse mathématique montrent que le protocole BSC permet de réduire considérablement le risque d’interruption de la vidéo pendant la lecture, ce dernier étant très pénalisant pour les utilisateurs. Ces résultats vont nous permettre de concevoir des algorithmes d’adaptation de qualité à la bande passante en vue d’améliorer l’expérience utilisateur. Ces algorithmes permettent d’améliorer la qualité de la vidéo même étant dans un environnement où le débit utilisateur est très instable.La dernière étape de la thèse consiste à la conception de stratégies de caching pour optimiser la transmission de contenu vidéo utilisant le codec SVC. En effet, dans le réseau, des serveurs de cache sont déployés dans le but de rapprocher le contenu vidéo auprès des utilisateurs pour réduire les délais de transmission et améliorer la qualité de la vidéo. Nous utilisons la programmation linéaire pour obtenir la solution optimale de caching afin de le comparer avec nos algorithmes proposés. Nous montrons que ces algorithmes augmentent la performance du système tout en permettant de décharger les liens de transmission du réseau cœur
Due to the growth of video traffic over the Internet in recent years, HTTP AdaptiveStreaming (HAS) solution becomes the most popular streaming technology because ithas been succesfully adopted by the different actors in Internet video ecosystem. Itallows the service providers to use traditional stateless web servers and mobile edgecaches for streaming videos. Further, it allows users to access media content frombehind Firewalls and NATs.In this thesis we focus on the design of a novel video streaming delivery solutioncalled Backward-Shifted Coding (BSC), a complementary solution to Dynamic AdaptiveStreaming over HTTP (DASH), the standard version of HAS. We first describe theBackward-Shifted Coding scheme architecture based on the multi-layer Scalable VideoCoding (SVC). We also discuss the implementation of BSC protocol in DASH environment.Then, we perform the analytical evaluation of the Backward-Sihifted Codingusing results from queueing theory. The analytical results show that BSC considerablydecreases the video playback interruption which is the worst event that users can experienceduring the video session. Therefore, we design bitrate adaptation algorithms inorder to enhance the Quality of Experience (QoE) of the users in DASH/BSC system.The results of the proposed adaptation algorithms show that the flexibility of BSC allowsus to improve both the video quality and the variations of the quality during thestreaming session.Finally, we propose new caching policies to be used with video contents encodedusing SVC. Indeed, in DASH/BSC system, cache servers are deployed to make contentsclosed to the users in order to reduce network latency and improve user-perceived experience.We use Linear Programming to obtain optimal static cache composition tocompare with the results of our proposed algorithms. We show that these algorithmsincrease the system overall hit ratio and offload the backhaul links by decreasing thefetched content from the origin web servers

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Yu, Lang. "Evaluating and Implementing JPEG XR Optimized for Video Surveillance". Thesis, Linköping University, Computer Engineering, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-54307.

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This report describes both evaluation and implementation of the new coming image compression standard JPEG XR. The intention is to determine if JPEG XR is an appropriate standard for IP based video surveillance purposes. Video surveillance, especially IP based video surveillance, currently has an increasing role in the security market. To be a good standard for surveillance, the video stream generated by the camera is required to be low bit-rate, low latency on the network and at the same time keep a high dynamic display range. The thesis start with a deep insightful study of JPEG XR encoding standard. Since the standard could have different settings,optimized settings are applied to JPEG XR encoder to fit the requirement of network video surveillance. Then, a comparative evaluation of the JPEG XR versusthe JPEG is delivered both in terms of objective and subjective way. Later, part of the JPEG XR encoder is implemented in hardware as an accelerator for further evaluation. SystemVerilog is the coding language. TSMC 40nm process library and Synopsys ASIC tool chain are used for synthesize. The throughput, area, power ofthe encoder are given and analyzed. Finally, the system integration of the JPEGXR hardware encoder to Axis ARTPEC-X SoC platform is discussed.

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Sharma, Naresh. "Arbitrarily Shaped Virtual-Object Based Video Compression". Columbus, Ohio : Ohio State University, 2009. http://rave.ohiolink.edu/etdc/view?acc%5Fnum=osu1238165271.

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Ye, Fangping. "Slepian-Wolf source coding using LDPC codes for free viewpoint television". Thesis, Ecole nationale supérieure Mines-Télécom Atlantique Bretagne Pays de la Loire, 2019. http://www.theses.fr/2019IMTA0162.

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La Télévision Interactive (FTV) est un service de vidéo à la demande qui permet au client de choisir l’angle de vue de la vidéo. Le défi principal est de stocker un d’énorme volume de données et d’extraire une petite partie de ces données à la demande et en temps réel. Pour améliorer le décodage de l’information, on peut supposer que les vues précédemment reçues sont conservées par l’utilisateur. Le problème ainsi posé devient un problème de codage de sources avec information adjacente du côté de l’utilisateur. Cette thèse s’inscrit dans ce contexte. Elle s’intègre au projet CominLabs InterCom dont l’objectif est de proposer des solutions pour l’accès massif aléatoire à des sous-ensembles de données corrélées. Dans cette thèse, nous proposons des schémas pratiques de codage de sources sans perte sous l’hypothèse d’une information adjacente. Ces schémas sont basés sur des codes de type Low Density Parity Check (LDPC). Les méthodes de l’état de l’art utilisent des solutions adaptatives en débit s’appuyant sur des codes LDPC de type Rateless ou LDPC Accumulés (LDPCA). Mais les codes Rateless fonctionnent mal aux débits faibles, et les codes LDPCA ne sont pas adaptés aux débits élevés. Dans cette thèse, on combine les deux méthodes pour construire des codes LDPC aux débits adaptables offrant une large gamme de débits. Cependant, la technique LDPCA ne permet pas d’optimiser la distribution des degrés du code, ni de contrôler le nombre de cycles courts pour tous les rendements. C’est pourquoi nous proposons deux nouvelles méthodes de construction pour remplacer le code LDPCA. Les résultats de la simulation montrent une amélioration des performances par rapport à LDPCA. Enfin, nous intégrons la construction de codes sans perte dans un schéma complet de codage de source avec pertes développé pour l’application FTV dans le cadre du project InterCom
Many multimedia applications such as Free Viewpoint Television (FTV) use a distant service provider that offers customized services depending on the user request. The main challenge is the efficient storage of a huge amount of data and the real-time extraction of a small fraction of these data upon request. In some applications such as FTV, the requests previously addressed by the user can help to optimize both the storage and the extraction. The problem can thus be seen as a source coding problem with side information at the user side. This PhD thesis fits into this context. It is part of the CominLabs project InterCom that focuses on solutions for massive random access to subsets of correlated data. In this thesis, we investigate practical lossless source coding schemes with side information based on Low Density Parity Check (LDPC) codes. State-of-the-art approaches use rate-adaptive LDPC codes such as rateless codes and LDPC accumulate (LDPCA) codes. Rateless codes perform poorly at low coding rates while LDPCA is not adapted to high-rates. In this thesis, we combine both methods to constructrate-adaptive LDPC codes offering a wide range of rates. However LDPCA does not allow to optimize the code degree distribution, nor to control the amount of short cycles at all rates. This is why we propose two novel rate-adaptive LDPC code constructions to replace the LDPCA part. Simulation results show improved performance compared to LDPCA. Finally, we incorporate the proposed lossless code construction into a complete lossy source coding scheme that was developped for FTV in the framework of the InterCom project

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Trioux, Anthony. "Étude et optimisation d'un système de vidéotransmission conjoint source-canal basé "SoftCast". Thesis, Valenciennes, Université Polytechnique Hauts-de-France, 2019. http://www.theses.fr/2019UPHF0018.

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Des nouveaux schémas de Codage Vidéo Linéaire (CVL) ont démontré ces dernières années un potentiel élevé pour la diﬀusion de contenus vidéo sur des canaux de transmission sans-ﬁl sévères. SoftCast représente le pionnier des schémas CVL. Diﬀérent des standards de transmission vidéo actuels et particulièrement utile en situation de broadcast, SoftCast est un système de codage conjoint source-canal où les pixels sont traités par des opérations linéaires successives (transformée DCT, allocation de puissance, modulation quasi-analogique) et directement transmis sans quantiﬁcation ni codage (entropique ou de canal). SoftCast permet ainsi d’offrir une qualité vidéo reçue directement proportionnelle à la qualité du canal de transmission, sans aucune information de retour et tout en évitant les mécanismes d’adaptation complexes des schémas classiques. Un premier objectif de ces travaux de thèse concerne l’étude des performances de bout en bout de SoftCast. Des modèles théoriques sont ainsi proposés prenant en compte les contraintes de bande passante de l’application, l’allocation de puissance, ainsi que le type de décodeur utilisé à la réception (LLSE, ZF). Une deuxième partie basée sur une campagne de tests subjectifs concerne une étude originale de la qualité vidéo et des artefacts spéciﬁques associés à SoftCast. Dans une troisième partie, des méthodes de prétraitement permettant d’accroître la qualité reçue sont proposées avec un gain moyen en PSNR de l’ordre de 3 dB. Finalement, un algorithme adaptatif modiﬁant la taille du groupe d’images (GoP) en fonction des caractéristiques du contenu vidéo transmis est proposé. Cette solution permet d’obtenir des gains supplémentaires en PSNR de l’ordre de 1 dB
Linear video coding (LVC) schemes have recently demonstrated a high potential for delivering video content over challenging wireless channels. SoftCast represents the pioneer of the LVC schemes. Diﬀerent from current video transmission standards and particularly useful in broadcast situation, SoftCast is a joint source-channel coding system where pixels are processed by successive linear operations (DCT transform, power allocation, quasi-analog modulation) and directly transmitted without quantization or coding (entropic or channel). This allows to provide a received video quality directly proportional to the transmission channel quality, without any feedback information, while avoiding the complex adaptation mechanisms of conventional schemes. A ﬁrst contribution of this thesis is the study of the end-to-end performances of SoftCast. Theoretical models are thus proposed taking into account the bandwidth constraints of the application, the power allocation, as well as the type of decoder used at the reception (LLSE, ZF). Based on a subjective test campaign, a second part concern an original study of the video quality and speciﬁc artifacts related to SoftCast. In a third part, preprocessing methods are proposed to increase the received quality in terms of PSNR scores with an average gain of 3 dB. Finally, an adaptive algorithm modifying the size of the group of pictures (GoP) according to the characteristics of the transmitted video content is proposed. This solution allows to obtain about 1 dB additional gains in terms of PSNR scores

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Al, Hasrouty Christelle. "Adaptive Multicast Live Streaming for A/V Conferencing Systems over Software-Defined Networks". Thesis, Bordeaux, 2018. http://www.theses.fr/2018BORD0267/document.

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Les applications en temps réel, telles que les systèmes de conférence multi-utilisateurs, ont des exigences de qualité de service élevées pour garantir une qualité d'expérience décente. De nos jours, la plupart de ces conférences sont effectuées sur des appareils sans fil. Ainsi, les appareils mobiles hétérogènes et la dynamique du réseau doivent être correctement gérés pour fournir une bonne qualité d’expérience. Dans cette thèse, nous proposons deux algorithmes pour construire et gérer des sessions de conférence basées sur un réseau défini par logiciel qui utilise à la fois la distribution multicast et l’adaptation de flux. Le premier algorithme configure la conférence téléphonique en créant des arborescences de multidiffusion pour chaque participant. Ensuite, il place de manière optimale les emplacements et les règles d’adaptation des flux sur le réseau afin de minimiser la consommation de bande passante. Nous avons créé deux versions de cet algorithme: le premier, basé sur les arborescences les plus courtes, minimise la latence, tandis que le second, basé sur les arborescences, minimise la consommation de bande passante. Le deuxième algorithme adapte les arborescences de multidiffusion en fonction des modifications du réseau qui se produisent pendant un appel. Il ne recalcule pas les arbres, mais ne déplace que les emplacements et les règles d’adaptation des flux. Cela nécessite un calcul très faible au niveau du contrôleur, ce qui rend notre proposition rapide et hautement réactive. Des résultats de simulation étendus confirment l'efficacité de notre solution en termes de temps de traitement et d'économies de bande passante par rapport aux systèmes de conférence existants basés sur une unité de contrôle multipoint et une multidiffusion de couche d'application
Real-time applications, such as Multi-party conferencing systems, have strong Quality of Service requirements for ensuring a decent Quality of Experience. Nowadays, most of these conferences are performed on wireless devices. Thus, heterogeneous mobile devices and network dynamics must be properly managed to provide a good quality of experience. In this thesis, we propose two algorithms for building and maintaining conference sessions based on Software-Defined Network that uses both multicast distribution and streams adaptation. The first algorithm set up the conference call by building multicast trees for each participant. Then, it optimally places the stream adaptation locations and rules inside the network in order to minimize the bandwidth consumption. We have created two versions of this algorithm: the first one, based on the shortest path trees is minimizing the latency, while the second one, based on spanning trees is minimizing the bandwidth consumption. The second algorithm adapts the multicast trees according to the network changes occurring during a call. It does not recompute the trees, but only relocates the locations and rules of stream adaptation. It requires very low computation at the controller, thus making our proposal fast and highly reactive. Extensive simulation results confirm the efficiency of our solution in terms of processing time and bandwidth savings compared to existing conferencing systems based on a Multipoint Control Unit and Application Layer Multicast

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Herrou, Glenn. "Résolution Spatio-temporelle Adaptative pour un Codage à Faible Complexité des Formats Vidéo Émergents". Thesis, Rennes, INSA, 2019. http://www.theses.fr/2019ISAR0020.

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La standardisation du dernier format vidéo en date, appelé Ultra-High Definition TV (UHDTV), vise à améliorer la qualité l’expérience des utilisateurs en introduisant de nouvelles technologies telles que la 4K ou le High Frame-Rate (HFR). Cependant, ces améliorations multiplient la quantité de données à traiter avant transmission du signal par un facteur 8. En plus de ce nouveau format, les fournisseurs de contenu doivent aussi encoder les vidéos dans des formats et à des débits différents du fait de la grande variété des systèmes et réseaux utilisés par les consommateurs. SHVC, l’extension scalable du dernier standard de compression video High Efficiency Video Coding (HEVC) est une solution prometteuse pour adresser ces problématiques. En revanche, son architecture, très demandeuse en termes de calculs, atteint ses limites lors de l’encodage des nouveaux formats vidéo immersifs tels que le standard UHDTV.L’objectif de cette thèse est donc d’étudier des approches de codage scalables et légères basées sur l’adaptation de la résolution spatio-temporelle des vidéos. La première partie de cette thèse propose deux algorithmes de pré-traitement, utilisant respectivement des approches polyphase et ondelette basées image, afin de permettre la scalabilité spatiale avec une faible augmentation de la complexité.Ensuite, dans un second lieu, le design d’une architecture scalable à deux couches, plus conventionnelle, est étudié. Celle-ci est composée d’un encodeur HEVC standard dans la couche de base pour assurer la compatibilité avec les systèmes existants. Pour la couche d’amélioration, un encodeur basse complexité, se basant sur l’adaptation locale de la résolution spatiale, est proposé. Enfin, la dernière partie de cette thèse se focalise sur l’adaptation de la résolution spatio-temporelle. Un algorithme faisant varier la fréquence image est d’abord proposé. Cet algorithme est capable de détecter localement et de façon dynamique la fréquence image la plus basse n’introduisant pas d’artefacts visibles liés au mouvement. Les algorithmes de fréquence image variable et de résolution spatiale adaptative sont ensuite combinés afin d’offrir un codage scalable à faible complexité des contenus 4KHFR
The definition of the latest Ultra-High Definition TV (UHDTV) standard aims to increase the user’s quality of experience by introducing new video signal features such as 4K and High Frame-Rate (HFR). However, these new features multiply by a factor 8 the amount of data to be processed before transmission to the end user.In addition to this new format, broadcasters and Over-The-Top (OTT) content providers have to encode videos in different formats and at different bitrates due to the wide variety of devices with heterogeneous video format and network capacities used by consumers.SHVC, the scalable extension of the latest video coding standard High Efficiency Video Coding (HEVC) is a promising solution to address these issues but its computationally demanding architecture reaches its limit with the encoding and decoding of the data-heavy newly introduced immersive video features of the UHDTV video format.The objective of this thesis is thus to investigate lightweight scalable encoding approaches based on the adaptation of the spatio-temporal resolution. The first part of this document proposes two pre-processing tools, respectively using polyphase and wavelet frame-based approaches, to achieve spatial scalability with a slight complexity overhead.Then, the second part of this thesis addresses the design of a more conventional dual-layer scalable architecture using an HEVC encoder in the Base Layer (BL) for backward compatibility and a proposed low-complexity encoder, based on the local adaptation of the spatial resolution, for the Enhancement Layer (EL).Finally, the last part of this thesis investigates spatiotemporal resolution adaptation. A variable frame-rate algorithm is first proposed as pre-processing. This solution has been designed to locally and dynamically detect the lowest frame-rate that does not introduce visible motion artifacts. The proposed variable frame-rate and adaptive spatial resolution algorithms are then combined to offer a lightweight scalable coding of 4K HFR video contents

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Martins, André Luis Del Mestre. "Projeto da arquitetura de hardware para binarização e modelagem de contextos para o CABAC do padrão de compressão de vídeo H.264/AVC". reponame:Biblioteca Digital de Teses e Dissertações da UFRGS, 2011. http://hdl.handle.net/10183/28742.

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O codificador aritmético binário adaptativo ao contexto adotado (CABAC – Context-based Adaptive Binary Arithmetic Coding) pelo padrão H.264/AVC a partir de perfil Main é o estado-da-arte em termos de eficiência de taxa de bits. Entretanto, o CABAC ocupa 9.6% do tempo total de processamento e seu throughput é limitado pelas dependências de dados no nível de bit (LIN, 2010). Logo, atingir os requisitos de desempenho em tempo real nos níveis mais altos do padrão H.264/AVC se torna uma tarefa árdua em software, sendo necesário então, a aceleração do CABAC através de implementações em hardware. As arquiteturas de hardware encontradas na literatura para o CABAC focam no Codificador Aritmético Binário (BAE - Binary Arithmetic Encoder) enquanto que a Binarização e Modelagem de Contextos (BCM – Binarization and Context Modeling) fica em segundo plano ou nem é apresentada. O BCM e o BAE juntos constituem o CABAC. Esta dissertação descreve detalhadamente o conjunto de algoritmos que compõem o BCM do padrão H.264/AVC. Em seguida, o projeto de uma arquitetura de hardware específica para o BCM é apresentada. A solução proposta é descrita em VHDL e os resultados de síntese mostram que a arquitetura alcança desempenho suficiente, em FPGA e ASIC, para processar vídeos no nível 5 do padrão H.264/AVC. A arquitetura proposta é 13,3% mais rápida e igualmente eficiente em área que os melhores trabalhos relacionados nestes quesitos.
Context-based Adaptive Binary Arithmetic Coding (CABAC) adopted in the H.264/AVC main profile is the state-of-art in terms of bit-rate efficiency. However, CABAC takes 9.6% of the total encoding time and its throughput is limited by bit-level data dependency (LIN, 2010). Moreover, meeting real-time requirement for a pure software CABAC encoder is difficult at the highest levels of the H.264/AVC standard. Hence, speeding up the CABAC by hardware implementation is required. The CABAC hardware architectures found in the literature focus on the Binary Arithmetic Encoder (BAE), while the Binarization and Context Modeling (BCM) is a secondary issue or even absent in the literature. Integrated, the BCM and the BAE constitute the CABAC. This dissertation presents the set of algorithms that describe the BCM of the H.264/AVC standard. Then, a novel hardware architecture design for the BCM is presented. The proposed design is described in VHDL and the synthesis results show that the proposed architecture reaches sufficiently high performance in FPGA and ASIC to process videos in real-time at the level 5 of H.264/AVC standard. The proposed design is 13.3% faster than the best works in these items, while being equally efficient in area.

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Abdallah, Alaeddine. "Mécanismes Cross-Layer pour le streaming vidéo dans les réseaux WIMAX". Thesis, Bordeaux 1, 2010. http://www.theses.fr/2010BOR14142/document.

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Poussé par la demande croissante de services multimédia dans les réseaux Internet à haut débit, la technologie WIMAX a émergé comme une alternative compétitive à la solution filaire d’accès à haut débit. L’IEEE 802.16 constitue une solution qui offre des débits élevés en assurant une qualité de service (QoS) satisfaisante. En particulier, elle est adaptée aux applications multimédia qui ont des contraintes de QoS à satisfaire. Cependant, avec la présence d’utilisateurs hétérogènes qui ont des caractéristiques diverses en termes de bande passante, de conditions radio et de ressources disponibles, de nouveaux défis posés doivent être résolus. En effet, les applications multimédia doivent interagir avec leur environnement pour informer le réseau d’accès de leurs besoins en QoS et s’adapter dynamiquement aux variations des conditions du réseau.Dans ce contexte, nous proposons deux solutions pour la transmission des flux vidéo sur les réseaux 802.16 sur la base de l’approche Cross-layer. Nous nous intéressons à la fois à la transmission unicast et multicast sur le lien montant et descendant d’une ou plusieurs cellules WIMAX.Premièrement, nous proposons une architecture Cross-Layer qui permet l’adaptation et l’optimisation du streaming vidéo en fonction des ressources disponibles. Nous avons défini une entité CLO (Cross-Layer Optimizer) qui exploite des messages de gestion des flux de service, échangés entre BS et SS, au niveau MAC, pour déterminer l’adaptation nécessaire et optimale afin d’assurer le bon fonctionnement de l’application. Les adaptations se produisent en deux temps, lors de l'admission du flux et au cours de la session de streaming. L’analyse des performances, par simulations, de notre solution montre l’efficacité du CLO à adapter, d’une façon dynamique, le débit vidéo en fonction des conditions du réseau afin d’assurer une QoS optimale.Deuxièmement, nous proposons une solution de streaming multicast des flux vidéo dans les réseaux WIMAX. Cette solution permet de trouver un compromis entre la diversité des clients, en termes de conditions radio, de schémas de modulation et de ressources disponibles, ainsi que le format de codage vidéo hiérarchique SVC, pour offrir la meilleure qualité vidéo y compris pour les clients ayant de faibles conditions radio. En effet, cette solution permet à chaque utilisateur d’obtenir une qualité vidéo proportionnellement à ses conditions radio et à sa bande passante disponible. Pour atteindre cet objectif, plusieurs groupes multicast sont formés par couches vidéo SVC. Cette solution permet d’optimiser davantage les ressources radio et ainsi d’augmenter la capacité globale du système
Driven by the increasing demand for multimedia services in broadband Internet networks, WIMAX technology has emerged as a competitive alternative to the wired broadband access solutions. The IEEE 802.16 is a solution that provides high throughput by ensuring a satisfactory QoS. In particular, it is suitable for multimedia applications that have strict QoS constraints. However, the users’ heterogeneity and diversity in terms of bandwidth, radio conditions and available resources, pose new deployment challenges. Indeed, multimedia applications need to interact with their environment to inform the access network about their QoS requirements and dynamically adapt to changing network conditions.In this context, we propose two solutions for video streaming over 802.16 networks based on Cross-Layer approach. We are interested in both unicast and multicast transmissions in uplink and downlink of one or more WIMAX cells.First, we proposed an architecture that enables Cross-Layer adaptation and optimization of video streaming based on available resources. We defined the entity CLO (Cross-Layer Optimizer) that takes benefits from service flow management messages, exchanged between BS and SS, at the MAC level, to determine the necessary adaptations / adjustment to ensure optimal delivery of the application. Adaptations occur at two epochs, during the admission of the video stream and during the streaming phase. The performance analysis, performed through simulations, shows the effectiveness of the CLO to adapt in a dynamic way, the video data rate depending on network conditions, and thus guarantee an optimal QoS.Second, we proposed a solution that enables IP multicast video delivery in WIMAX network. This solution allows finding the compromise between the diversity of end-user requirements, in terms of radio conditions, modulation schemes and available resources, along with the SVC hierarchy video format, to offer the best video quality even for users with low radio conditions. Indeed, we define a multicast architecture that allows each user to get a video quality proportionally to its radio conditions and its available bandwidth. Towards this end, several IP multicast groups are created depending on the SVC video layers. Subsequently, our solution allows optimizing the use of radio resources by exploiting the different modulations that can be selected by the end-users

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ACOCELLA, EMILIO CARLOS. "MATHEMATICAL AND EXPERIMENTAL APPROACHES IN SHAPE-ADAPTATIVE VIDEO CODING". PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 2000. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=7568@1.

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CONSELHO NACIONAL DE DESENVOLVIMENTO CIENTÍFICO E TECNOLÓGICO
Esta tese aborda teórica e experimentalmente diversos tópicos de Codificação Adaptativa à Forma de objetos de forma arbitrária. Aspectos associados à representação e à codificação eficiente da intensidade e do contorno de objetos são analisados e são propostas soluções para os problemas identificados. Os métodos introduzidos são testados valendo-se de seqüências de imagens empregadas em trabalhos congêneres. Inicialmente é desenvolvida uma formulação matemática das transformadas adaptativas à forma utilizando operadores lineares e, com base nela, é obtida uma métrica que possibilita a avaliação teórica do desempenho dessas transformadas. A comparação das grandezas obtidas com resultados de experimentos mostram a validade dessa métrica para a finalidade visada. Em seguida é analisada a questão do melhor alinhamento dos coeficientes das transformadas unidimensionais de duas colunas com dimensões distintas e é proposto um método de alinhamento pela fase. Esse método caracteriza-se pela baixa complexidade e os resultados experimentais demonstram o seu desempenho superior ao de outros encontrados na literatura. Problemas específicos da codificação adaptativa à forma referentes à quantização dos coeficientes da transformada empregada são abordados matematicamente para diversas e freqüentes versões de sua implementação. Apresenta-se um método para solucionar simultaneamente os problemas da distorção do valor médio e da correlação do erro do sinal introduzido pela quantização. Constata-se experimentalmente sua maior eficiência de codificação em relação à de outros métodos propostos em trabalhos recentes. Um grande número de possíveis modificações de um codificador de cadeia diferencial, método bastante empregado para a codificação de contorno sem perda, é identificado e avaliado, concluindo-se com a implementação de um método que introduz aquelas mudanças que resultaram em aumento significativo da eficiência de codificação da forma de objetos. Por fim, propõe-se um esquema genérico de decomposição em subbandas através de uma transformada wavelet discreta adaptativa à forma. Os resultados dos experimentos realizados permitem concluir que o esquema oferece perspectivas de obtenção de eficiência de codificação superior à da transformada cosseno discreta adaptativa à forma, sobreturde em baixas taxas de bits por pixel.
This thesis investigates shape adaptative coding of arbitrarily shaped segments. The texture and contour coding efficiency is discussed and solutions to tackle the associated problems are proposed. The presented methods are evaluated using standard image sequences. A mathematical approach for shape-adaptative transforms using linear operators is developed, followed by a metric that theoretically evaluates the transform performances. Experimental results show that the proposed metric is an efficient tool for such purposes. The proper way for grouping the 1-D transform coefficients of two image segments of different sizes is analyzed. Based on this analysis, a new low complexity method for grouping the coefficients is proposed. A better performance than other reported methods in the literature is attested by the experimental results. A mathematical analysis of the performance limitations of shape-adaptative transforms due to coefficients quantization is presented. The drawbacks discussed are the mean weighting distortion and the signal error correlation produced by the quantization process. An efficient method to simultaneously overcome both problems is proposed. The differential chain coder is an efficient and frequently employed structure for lossless encoding of object boundaries. Many modifications in the differential chain coders are investigated and evaluated, resulting in a method that reduces the bit rate to encode the object shape. Finally, a generic scheme for sub-band decomposition using shape-adaptative discrete wavelet transform is proposed. The experimental results show that such a scheme is able to provide a performance gain over the shape-adptative discrete cosine transform at low bit rates. The preliminary results suggest that this scheme could be a promising new approach for shape adaptative video coding.

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Fatani, Imade Fahd Eddine. "Contribution à l’étude de l’optimisation conjointe source-canal d’une transmission vidéo dans un contexte MIMO sans fil : application à la vidéosurveillance embarquée pour les transports publics". Valenciennes, 2010. http://ged.univ-valenciennes.fr/nuxeo/site/esupversions/f1e3d785-7cbb-4d39-86d8-eec5433f62a0.

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Les applications de vidéosurveillance pour les transports publics s’appuient sur des systèmes de communication sans fil qui requièrent des débits élevés entre les véhicules et le sol et des critères de qualité de service élevés. Afin de répondre à ces contraintes, dans ce travail nous avons proposé de tenir compte à la fois des paramètres de transmission et de d’encodage vidéo en combinant les techniques de codage MDC (Multiple Description Coding) et de codage par zone d'intérêt (ROI, Region Of Interest) avec différentes schémas MIMO (Mulitple Input Multiple Output) sur la base de la couche PHY du standard Wifi IEEE802. 11n dans un environnement de type métro (tunnel). Dans un premier temps, nous avons montré qu'il est possible d'améliorer les performances d'un système MIMO en optimisant l'allocation des bits et des puissances indépendamment de l'information à transmettre. Nous proposons deux approches aboutissant à la répartition optimale des ressources qui permettent d'atteindre un ordre de diversité maximal et offrent de meilleures performances que le précodeur max-SNR dans le cas d’un canal corrélé ou non. Nous montrons ensuite que l’association d’un codage MDC avec des schémas MIMO constitue une stratégie intéressante afin d’adapter le contenu vidéo à la structure multi-antennes, en particulier lorsqu’aucune connaissance de l’état du canal n’est pas disponible en émission. En outre, il est possible d'améliorer les performances en utilisant un lien retour à faible débit grâce aux techniques OSM (Orthogonalized Spatial Multiplexing) et à l’OSM précodé. Enfin, dans le cas où la connaissance du canal à l’émission est parfaite, pour un lien retour offrant un débit suffisant, nous avons associé les techniques MIMO et un mécanisme de codage vidéo hiérarchique qui consiste en la séparation de la scène en régions d'intérêt. Le flux correspondant à la zone d’intérêt maximal est transmis sur le canal propre de plus grand gain. Ceci permet d'assurer une meilleure robustesse de transmission et garantit ainsi un niveau acceptable pour la QoS perçue par le centre de contrôle. La création des différentes régions d’intérêt s’appuie sur l’outil FMO (Flexible Macroblock Ordering) introduit dans le nouveau standard de compression H. 264/AVC. Ainsi, les différents schémas de transmission proposés permettent d’accroître la qualité de service d’un flux vidéo sans augmenter la puissance émise ni multiplier le nombre de points d’accès radio de l’infrastructure
Video monitoring applications in the Public Transport field rely on wireless telecommunication systems which require high data rate between vehicles and the ground and high Quality of Service (QoS). In order to satisfy these constraints we have proposed to take into account both transmission parameters and video coding by combining Multiple Description Coding (MDC) and Region Of Interest coding with different MIMO (Mulitple Input Multiple Output) schemes on the basis of the PHY layer of IEEE802. 11n Wifi standard in a metro environment (tunnel). First, we have shown that it is possible to increase the performance of a MIMO system by optimizing bits and power allocation independently of the type of information to be transmitted. Two approaches are proposed. They lead to an optimal repartition of resources, reach maximal diversity order and they outperform the max-SNR precoder performances. Secondly, the association of MDC with MIMO schemes is introduced to adapt the video content to the multi antenna structure particularly when the channel knowledge is not available at transmitter side. Furthermore, the performances can be enhanced using a low data rate return link and considering the Orthogonalized Spatial Multiplexing (OSM) and the precoded OSM. When perfect channel information is available at transmitter side thanks to a high data rate return link, MIMO schemes are associated with hierarchic video coding consisting in the separation of regions of interest in the scene. The stream associated to the area with the maximal interest is transmitted on the eigen channel with the higher gain. This strategy allows to guaranty better robustness and acceptable QoS of the video streams observed in the control-center. The creation of the different regions of interest is based on the Flexible Macroblock Ordering (FMO) technique introduced in the new compression standard H. 264/AVC. We have shown the interest of the different transmission schemes proposed in order to enhance the QoS of a video stream with no increase of the transmitted power and of the number of radio access points along the infrastructure

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36

"Adaptive coding and rate control of video signals". 2015. http://repository.lib.cuhk.edu.hk/en/item/cuhk-1291501.

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As the bandwidth has become much cheaper in the recent years, video applications are more popular than before. However, the demand of high video resolution, high frame rate, or high bit-depth has continued to increase more rapidly than the cost of video transmission and storage bandwidth. It requires more efficient compression techniques, and hence many international video coding standards are developed in the past decades such as the MPEG-1/2/4 part 2, H.264/MPEG-4 part 10 AVC and the latest High Efficiency Video Coding (HEVC) standards. The main objective of this thesis is to consider the problems in analyzing the characteristics of video signals and providing efficient compression and transmission solutions in both H.264/AVC and HEVC video systems. Three main parts of this work are briey summarized below.
The first part concerns transform coding. Transform coding has been widely used to remove spatial redundancy of prediction residuals in the modern video coding standards. However, since the residual blocks exhibit diverse characteristics in a video sequence, conventional sinusoidal transforms with fixed transform kernels may result in low coding efficiency. To tackle this problem, we propose a novel content adaptive transform framework for H.264/AVC-based video coding. We propose to utilize pixel rearrangement to dynamically adjust the transform kernels to adapt to the video signals. In addition, unlike the traditional adaptive transforms, the proposed method obtains the transform kernels from the reconstructed block, and hence it consumes only one logic indicator for each transform unit. Moreover, a spiral-scanning method is developed to reorder the transform coefficients for better entropy coding. Experimental results on the Key Technical Area (KTA) platform show that the proposed method can achieve a significant bits reduction under both the all-intra and low-delay configurations.
The second part investigates the next-generation video coding. Due to increase of display resolution from High-definition (HD) to Ultra-HD, how to efficiently compress the Ultra-HD signals are essential in the development of future video compression systems. High-resolution video coding benefits from a larger prediction block size and thereof transform and quantization of prediction residues. However, in the current HEVC video coding standard, the maximum coding tree unit (CTU) size is 64x64, which can limit a possible larger prediction block in Ultra-HD video coding, and hence cause negative effects on coding efficiency. Thus, we propose to extend CTU to a super coding unit (SCU) for next-generation video coding, and two separate coding structures are designed to encode a SCU, including Direct-CTU and SCU-to-CTU modes. In Direct-CTU, an SCU is first split into a number of predefined CTUs, and then, the best encoding parameters are searched from the current CTU to the possible minimum coding unit (MCU). Similarly, in SCU-to-CTU, the best encoding parameters are searched from SCU to CTU. In addition, the adaptive loop filter (ALF) and sample adaptive offset (SAO) methods are investigated in SCU based video coding framework. We propose to change the filtering control from SCU level to the coding unit (CU) level, and an improved CU level ALF signaling method is also proposed to further improve the coding efficiency. Furthermore, an adaptive SAO block method is also proposed, and this flexibility of SAO blocks can further improve the performance of the traditional method in the Ultra HD video coding.
In the last part, we explore the bit rate control of video transmission. Rate control serves as an important technique to regulate the bit rate of video transmission over a limited bandwidth and to maximize the overall video quality. Video quality fluctuation plays a key role in human visual perception, and hence many rate control algorithms have been widely developed to maintain a consistent quality for video communication. We propose a novel rate control framework based on the Lagrange multiplier in HEVC. With the assumption of constant quality control, a new relationship between the distortion and the Lagrange multiplier is established. Based on the proposed distortion model and buffer status, we obtain a computationally feasible solution to the problem of minimizing the distortion variation across video frames at the coding tree unit level. Extensive simulation results show that our method outperforms the HEVC rate control by providing a more accurate rate regulation, lower video quality fluctuation and stabler buffer fullness.
近些年，隨著帶寬費用變得越來越便宜，各種視頻應用比以前更為流行了。然而，人們對于高視頻分辨率，高幀率，或更高比特像素的需求增加了視頻傳輸和存儲帶寬的成本。滿足這樣的需求需要更有效的壓縮技術，因此在過去的幾十年裏，很多國際視頻編碼標准被開發出來，例如MPEG-1/2/4 part2, H264/MPEG-4 part 10 AVC和最新高效視頻編碼標准(HEVC)。本論文的主要目的是研究視頻信號的特點，在H.264和HEVC視頻系統中提供高效的壓縮和傳輸解決方案。論文分三部分，簡要總結如下。
第壹部分涉及變換編碼。在現代視頻編碼標准中，變換編碼已被廣泛用于消除預測殘差的空間冗余度。然而，由于在視頻序列中的預測殘差塊有著不同的特性，傳統的變換采用固定變換矩陣可能會導致低的編碼效率。為了解決這個問題，我們提出了壺種新的基于內容自適應變換方案的視頻編碼框架。我們利用重排像素，動態調整的變換矩陣以適應當前的視頻信號。此外，與傳統的自適應變換不同之處在于，我們所提出的方法得到的變換矩陣不需要傳輸到解碼端，而它僅消耗壺個邏輯單元指示當前變換矩陣。此外，我們提出了相應的變換系數掃描方法以達到更有效的熵編碼。在關鍵技術領域(KTA)平台，實驗結果表明本方法可以有效的改善幀內和低延遲的配置下的編碼效率。
第二部分探討了新壹代視頻編碼。由于主流顯示分辨率從高清到超高清的變化，如何有效地壓縮超高清視頻信號是未來視頻壓縮技術發展的關鍵。超高分辨率視頻編碼的好處在于可從壹個更大的預測塊對其預測殘差進行變換和量化。然而，在目前HEVC視頻編碼標準，最大編碼榭單元尺寸（CTU）是64x64，其可能限制較大的預測塊，從而影響編碼效率。因此，我們提出了擴展CTU為SCU。其中編碼壹個SCU可能用到兩個獨立的編碼模式，包括Direct-CTU和SCU-to-CTU。在Direct-CTU模式中，SCU被分割成許多預定義的CTUs，然後，最佳的編碼參數搜索範圍為CTU到MCU。同樣，在SCU-to-CTU模式中，最佳的編碼參數搜索範圍是SCU到CTU。此外，自適應環路濾波器（ALF）和自適應采偏移（SAO）在新的SCU編碼框架下進行了研究。我們提出濾波控制從SCU級別更改為CU級別，並提出了新的ALF信號傳送方法進壹步提高傳統的方法在超高清視頻編碼的中性能。
在最後壹部分，我們探討了視頻傳輸中的碼率控制。碼率控制作為壹種重要的技術，在有限的帶寬條件下，以最大限度地提高整體的視頻質量。視頻質量波動在人眼視覺感知中起著至關重要的作用，因此許多碼率控制方法得到了廣泛的發展，以追求提供穩定的視頻通信質量。我們提出了壹個新基于HEVC拉格日乘數碼率控制框架。在平穩視頻質量的假設下，我們提出了壹種新的失真和拉格日乘子之間的關係。基于新提出的失真模型和緩沖區的狀態，我們得到壹個計算上可行的解決方案，以最大限度地減少在編碼榭單元級的視頻幀的失真變化。大量的仿真結果表明，我們的方法優于HEVC的碼率控制，它可以提供更精確的碼率調節，降低視頻質量波動，以及維護穩定的緩沖區占有率。
Wang, Miaohui.
Thesis Ph.D. Chinese University of Hong Kong 2015.
Includes bibliographical references (leaves 158-164).
Abstracts and acknowledgements also in Chinese.
Title from PDF title page (viewed on 11, October, 2016).
Detailed summary in vernacular field only.
Detailed summary in vernacular field only.
Detailed summary in vernacular field only.
Detailed summary in vernacular field only.

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Lai, Hsueh-Han, i 賴學翰. "An Illumination Adaptive Video Coding Scheme for In-vehicle Video Applications". Thesis, 2008. http://ndltd.ncl.edu.tw/handle/92614027833348050233.

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碩士
國立中央大學
通訊工程研究所
96
With the advance of intelligent vehicle systems, drivers or passengers can keep interaction with people in fixed offices or other vehicles through visual communications. However, the illumination variations due to the changes of environments or weather conditions may significantly change the appearance of in-vehicle videos. Accordingly, the compression efficiency is much reduced even though the bandwidth of such wireless communications has been quite limited. There is pretty few previous work designed for efficient in-vehicle video compressions. Thus, in this paper, we propose an illumination adaptive video coding scheme for in-vehicle video applications. Since human faces are usually the most visually attended regions in such applications, this scheme consists of illumination correction, face detection, and the visual attention based video codec. The proposed illumination correction strategy combines the advantages of the single-scale Retinex (SSR) and the Interval weighted histogram separation (IWHS). The experimental results show that our illumination correction strategy effectively improves the face detection performance of in-vehicle videos. Moreover, the subjective visual quality of the proposed scheme outperforms that of H.264 with rate control since our scheme allocates bits by incorporating the human visual characteristics.

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Chen, Hung-Wei, i 陳泓瑋. "Adaptive GOP Structure Determination for Scalable Video Coding". Thesis, 2008. http://ndltd.ncl.edu.tw/handle/h5qd6r.

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碩士
國立東華大學
電子工程研究所
96
The hierarchical B picture coding is introduced into the extension of H.264/AVC in order to improve coding performance and provide strong temporal scalability as well. In general, coding performance is affected by the content variation in GOP. Therefore, the ways to determine the size of GOP (Group of Picture) is a critical problem for video coding. In this thesis, the adaptive GOP structure determination scheme is proposed to select the appropriate GOP size with content complexity consideration. Experimental results show that the proposed method can provide better PSNR performance compared to the fixed GOP setting in the existing hierarchical B picture coding of H.264/AVC. In this thesis, an area-weighted motion vectors prediction algorithm is proposed to increase motion vector prediction accuracy via using temporal information. Experimental results show that our proposed algorithms can get better motion vector prediction center.

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Chang, Chih-Chun, i 張智鈞. "Adaptive Error Concealment Algorithm for Scalable Video Coding". Thesis, 2014. http://ndltd.ncl.edu.tw/handle/5s96p8.

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碩士
國立東華大學
電機工程學系
102
As the development of multimedia, the transmission for video data becomes more and more necessary. However, the transmission may occur error due to interference or congestion. The international coding standard can reduce large amount of bit-rate by employing strong temporal relationship between current and reference frames. When transmission error occurs, it may cause great subjective and objective quality degeneration. The goal of this thesis is to keep good subjective and objective qualities by error concealment. This thesis presents a new error concealment algorithm for continuous error frames for the enhancement layer of H.264/SVC. The first part of our proposal is to adaptively decide the switching point for different error concealment methods in each group of picture. Second, we propose new error concealment methods. Simulation results demonstrate that our proposed algorithm provides better subjective and objective qualities on the error affected frames.

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Li, Chien-Hsiu, i 李鑑修. "Adaptive Subband Video Coding Using Diamond-Shaped Filter". Thesis, 1993. http://ndltd.ncl.edu.tw/handle/73983434940890073277.

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碩士
國立清華大學
電機工程研究所
81
Subband coding (SBC) has been a major technique in data compression of speeches and images. Instead of conventional rec- tangular bands in two dimensional subband coding of images, a non-separable diamond-shaped frequency decomposition which splits the original signal into four uniform subbands along the diagonal directions is discussed in this work. This diamond- shaped frequency decomposition matches the human visual system more closely than the conventional rectangular bands. The sub- band video coding system proposed is built up based on this diamond-shaped subband filtering system. One motivation of adopting subband structures is that each subband can be coded by the optimal coding algorithm according to its own characteristics. The coding algorithms which are adaptive to the different properties of the four subbands from diamond-shaped analysis filter bank are discussed in this work. Subband coding system is very suitable for the transmission of video or image in the broadband ISDN for its hierarchical structure. In the ATM network the priority scheme is used to provide the basic service quality. An adaptive layered coding system which is based on the diamond-shaped subband structure is proposed. This adaptive scheme can provide a superior ability of cell loss resilience. Some other processings which can reduce the reconstruction errors introduced by cell loss or transmission errors are also presented.

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Tsai, KunChih, i 蔡昆池. "Adaptive Quantization for Alleviating Blocking Effects in Video Coding". Thesis, 2002. http://ndltd.ncl.edu.tw/handle/26534173196415520330.

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碩士
國立海洋大學
電機工程學系
90
This thesis presents a novel method to reduce the blocking effects in block-based coding schemes. From our observations, the blocking effects occur mainly in regions that contain fast movements or complex structures; these regions need more bits to encode. However, this problem has not been taken care of in TM5 of MPEG2 [7]. In order to achieve fine quantization in these regions, our method employs a special measure of spatial activity to detect whether a region is complex or simple and thereby to decide whether or not to perform adaptive quantization. In order to satisfy the target bits allocation, we propose an algorithm to calculate the coding bits and to obtain a better quantization parameter. Experimental results show that our method achieve better picture quality in comparison to TM5 of MPEG2.Particuly in fast or complex motion scenes, the blocking effects in the object boundaries are far less vivid than that of TM5 of MPEG2, thereby generating much more smooth decoded images (with higher PSNR) for comfortable viewing.

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Hu, Yu-Chih, i 胡育誌. "Adaptive Rate Control in Pixel-Domain Distributed Video Coding". Thesis, 2007. http://ndltd.ncl.edu.tw/handle/28669564055587840865.

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碩士
國立中正大學
電機工程所
96
Because of the popularity of the Distributed Video Coding, the essay will transmitted a sequence of bit stream with the Laplacian noise through the Wyner-Ziv’s architecture. The difference of side information and the original image data will be assumed the Laplacian distribution. In implementation, it uses the effective way to store H matrix by recording the index of the position of 1’s. It saves a lot of memory space and the computations for LDPC code. By inputting the deviation, we can find out there is a determinative range of deviation that let the computation of LDPC converges to the original data.

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Wang, Jen-Chieh, i 王仁傑. "Adaptive Down-sampling Coding Scheme for High-difinition Video". Thesis, 2009. http://ndltd.ncl.edu.tw/handle/30627909629687213154.

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碩士
國立中央大學
通訊工程研究所
97
High-definition (HD) video that provides enhanced viewing experience is becoming increasingly popular. However, HD video requires large transmission bandwidth and computation complexity. This thesis proposes an efficient coding scheme for HD video by utilizing sub-sampling technique. First, we propose a down-sampling coding scheme with adaptive resolution-ratio to achieve better rate-distortion performance for video signals. And then, the computational complexity is greatly reduced by skipping un-necessary encoding modes. The Down-sampling coding, which sub-samples the image and encodes the smaller sized images, is one of the solutions to raise the image quality under insufficient rates. An adaptive resolution-ratio for down-sampling coding is utilized instead of fixed resolution-ratio. The optimum resolution-ratio is derived based on the models of down-sampling distortion and coding distortion. Simulation results show that the rate-distortion performance of adaptive resolution-ratio is higher than H.264 by 2 to 4 dB at low to medium rates. The complexity analysis of encoding tools for video at different resolutions has not been addressed much. This work analyzed quality gain of high complexity tools at different resolutions. Based on this analysis, we propose an adaptive encoding configuration scheme to reduce the computation complexity by skipping modes with low quality gains. As simulation results are shown, with almost the same rate-distortion performance, the proposed scheme further reduces complexity of the down-sampling coding. Compared with H.264, it has 90% complexity reduction at low to medium bitrates.

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Wang, Hongqiang. "Robust image and video coding with adaptive rate control". 2009. http://proquest.umi.com/pqdweb?did=1697720071&sid=3&Fmt=2&clientId=14215&RQT=309&VName=PQD.

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Thesis (Ph.D.)--University of Nebraska-Lincoln, 2009.
Title from title screen (site viewed July 21, 2009). PDF text: xii, 170 p. : ill. (some col.) ; 3 Mb. UMI publication number: AAT 3350500. Includes bibliographical references. Also available in microfilm and microfiche formats.

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45

Yang, Fengming, i 楊峰銘. "An Adaptive Frame Buffer Compression Method for Video Coding". Thesis, 2012. http://ndltd.ncl.edu.tw/handle/64455478889930286988.

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碩士
國立中正大學
資訊工程研究所
100
In this thesis, we study an adaptive data compression for H.264 and MVC (Multi-view Video Coding) encoders. We propose an adaptive data compression according to intra mode that completely lowers data bandwidth. Based on the method, we developed data compression algorithms for H.264/MVC. By using high resolution video like Full-HD and QFHD, the H.264/MVC encoders achieve a huge data reduction proportional to the reference frame used. Finally, we integrate the proposed data compression method into H.264 and MVC video encoding flow for the evaluation of performance. The experimental results show that it can save about 50% data amount of the reference frames under acceptable quality loss in PSNR. As compared to the existing lossy data compression methods, the proposed adaptive data compression algorithm has achieved fixed compression ratio (2 times data compression) with least quality loss for Full-HD video encoding systems.

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吳文棋. "Adaptive motion estimation with partial distortion search for video coding". Thesis, 2008. http://ndltd.ncl.edu.tw/handle/83977466635137439205.

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碩士
國立彰化師範大學
電子工程學系
96
With the advances of digital technologies and the proliferation of communication networks, multi-media contents consisting of both audio and video recordings continue to grow rapidly. Multi-media data generally takes huge storage space and also requires substantial amount of channel bandwidth to transmit. Many video coding standards exploit motion estimation (ME) to compress the video data by removing the inter-frame redundancy. This paper presents a new motion estimation algorithm, called the adaptive motion estimation with partial distortion search (AMEPDS). The AMEPDS algorithm exploits the information gathered from the previous frame to derive a parameter, called CP (Correlation parameter), and employs CP to classify the blocks in the current frame into potentially dependent blocks and potentially independent blocks. AMEPDS applies different motion estimation methods and adaptive search areas for different types of blocks to achieve better estimation accuracy and lower computational complexity. Early search termination is also introduced in AMEPDS to further speed up the process of motion estimation. Experimental results showed that the proposed algorithm AMEPDS can achieve a speedup of 15.58 to 150.5 times compared to the traditional full search algorithm, while maintaining similar visual quality.

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47

Lai, Hung-Liang, i 賴宏亮. "Adaptive Leaky Prediction Technique under Multiple Description Video Coding Framework". Thesis, 2005. http://ndltd.ncl.edu.tw/handle/22339505678368670002.

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碩士
國立清華大學
電機工程學系
93
In recent years, with the development of the internet, video streaming across packet-lossy networks has received much attention. But the problem of packet loss is still difficult to deal with. Since video coding uses motion compensation prediction, there will be error propagation problems in the decoding if some packets of previous frames are lost. To solve the problem of the error propagation, the well-known approach is leaky prediction. However, how to find an adaptive optimal leaky factor in the leaky prediction remains a challenging task. For such a task, we propose a new solution for the leaky factor of leaky prediction under Base MDSQ. Generally speaking, our optimal leaky factor is still depending on the packet loss rate. But the natural property of videos such as the complexity or amount of movement also affects the decision of the leaky factor. Moreover, it is necessary that the whole framework of coder must be considered because some properties of coder such as multiple description coding or Error Concealment technique can help for the reconstruction of lost frames. Therefore, we propose a new method about how to find the optimal leaky factors depending on the loss rate, the natural property of videos and the whole framework of the coder. From the simulation results, we can see that the proposed algorithm provides better performance than other leaky factors. And our method can have fine performance for the videos with different properties.

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48

Chiu, Yung-Lung, i 邱永龍. "Adaptive Coding Mode Decisions for MPEG-4 FGS Video Communication". Thesis, 2003. http://ndltd.ncl.edu.tw/handle/88276438295976737457.

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碩士
國立清華大學
電機工程學系
91
Streaming multimedia contents over the Internet is becoming more and more popular in recent years, partially due to the extraordinary presentation capability of multimedia data and partially due to the wider and wider deployment of broadband networking services. However, network heterogeneity and competitions among traffic have made the available bandwidth fluctuating for multimedia streaming application. Furthermore, the delivery is not error-free, due to the best effort nature of the current Internet. In order to provide the best possible quality of service(QoS) to the end users and satisfy the necessary requirements for the effective delivery of multimedia streams, an Amendment of MPEG-4 is developed in response to the growing need on a video-coding standard for streaming video over the Internet. It provides fine granularity scalability (FGS), and its combination with temporal scalability addresses a variety of challenging problems in delivering video over the Internet. In this thesis, we develop a mode selection method under network transmission simulation that may find the most suitable scalable coding from four coding schemes: FGS, FGST, FGS-SE and FGST with background composition based on the information that can be easily extracted from the base layer encoder.

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49

Hsiao, Yi-Mao, i 蕭詣懋. "Design of Bandwidth Adaptive Streaming System for Scalable Video Coding". Thesis, 2012. http://ndltd.ncl.edu.tw/handle/47013988794948558047.

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博士
國立中正大學
電機工程研究所
101
Designing a real-time video-streaming system involves the challenges, which are network congestion and bandwidth fluctuations, of lowering the video quality. Because of the increasing heavy network traffic, the backbone network has to handle routing lookup more quickly so that the transmission efficiency of multimedia streaming on the Internet can be improved. We design and implement a real-time video-streaming system for scalable video coding (SVC) and propose an adaptive control scheme, containing macro- and micro-control, to resolve the bandwidth fluctuation and network congestion. Under macro-control, the server allocates appropriated video-streaming of SVC for heterogeneous clients based on terminal capacity and network conditions. Under micro-control, the server changes the video frame rate based on video bit rates and network bandwidth. The streaming system with the adaptive control scheme is implemented to improve the peak signal noise ratio (PSNR) by 4.01 % to 10.89 % of the Foreman and Stefan videos. A routing lookup system is presented to manage connection, and it is composed of routing lookup ASIC and off-chip memory set. The off-chip memory set is two-level hierarchical memory architecture. 91.89% routing entries of the routing table can be searched in one memory access, and the worst case about 10% in this system is two memory accesses .The ASIC includes a function unit and a Binary Content Addressable Memory (BCAM) . The BCAM is used as cache memory with FIFO replacement algorithm .There are 1024 cache entries in the BCAM with 80% hit ratio. The routing lookup system approaches 260 Mega lookups per second (Mlps), which is sufficient for 100 Gbps networks. The memory density is good, with each routing entry requiring only 64 bits. Moreover, the routing table only needs 10.24KB on-chip BCAM, 20.04KB off-chip TCAM and 29.29MB DRAM for 3.6M routing entries in the proposed system. The proposed system is designed for real-time streaming transmission over Internet to resolve the bandwidth fluctuation and network congestion. The system is also designed for backbone network to resolve routing lookup and heavy network traffic.

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50

Lin, Chun-Han, i 林俊翰. "content-based classification for scalable video coding using adaptive GOP". Thesis, 2007. http://ndltd.ncl.edu.tw/handle/51308715962968851026.

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碩士
國立交通大學
資訊科學與工程研究所
95
Scalable video coding (SVC) is the extension of H.264/AVC standard. In SVC, it provides three types of scalability, including temporal scalability, spatial scalability, and quality scalability, with which great flexibility could be provided to fit different needs of transmission. Scalability is a great feature for transmitting compressed video data adaptively in variant bandwidth. However, manipulating the three types of scalability in different ways to encode the video sequence can yield a wide range of quality and coding efficiency. This paper exploits the characteristics of different scalabilities on the encoding of video sequence with varying features, in order to achieve satisfactory video quality at the desirable bit-rate. This work focuses on how to dynamically decide GOP size and the coding type adaptively according to the features of video content. With adaptive GOP, the most suitable scalable coding type can be applied for each GOP, and the quality under low bandwidth can be satisfying. We classify the content into four types, including simple frame with slow motion (SS), simple frame with high motion (SH), complex frame with slow motion (CS), and complex frame with high motion (CH). The idea is based upon the human perception that simple frame can present better III visual quality than complex frame after scaling from low resolution to high one, and thus make it a good choice for coding in spatial scalability; on the other hand, it is easier to reconstruct a lost frame with acceptable quality if the lost frame is in a slow motion sequence than in a high motion one, and thus make it a good choice for using temporal scalability. Under good consideration of the content of the video sequence, a better quality can be obtained in low bandwidth compared to standard scalable coding way. In high bandwidth, the algorithm still can adaptively choose the most suitable scalable coding technique for the video sequence. Therefore it works well in variant bandwidth.

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