788 resultados para Video coding
Resumo:
El esquema actual que existe en el ámbito de la normalización y el diseño de nuevos estándares de codificación de vídeo se está convirtiendo en una tarea difícil de satisfacer la evolución y dinamismo de la comunidad de codificación de vídeo. El problema estaba centrado principalmente en poder explotar todas las características y similitudes entre los diferentes códecs y estándares de codificación. Esto ha obligado a tener que rediseñar algunas partes comunes a varios estándares de codificación. Este problema originó la aparición de una nueva iniciativa de normalización dentro del comité ISO/IEC MPEG, llamado Reconfigurable Video Coding (RVC). Su principal idea era desarrollar un estándar de codificación de vídeo que actualizase e incrementase progresivamente una biblioteca de los componentes, aportando flexibilidad y la capacidad de tener un código reconfigurable mediante el uso de un nuevo lenguaje orientado a flujo de Actores/datos denominado CAL. Este lenguaje se usa para la especificación de la biblioteca estándar y para la creación de instancias del modelo del decodificador. Más tarde, se desarrolló un nuevo estándar de codificación de vídeo denominado High Efficiency Video Coding (HEVC), que actualmente se encuentra en continuo proceso de actualización y desarrollo, que mejorase la eficiencia y compresión de la codificación de vídeo. Obviamente se ha desarrollado una visión de HEVC empleando la metodología de RVC. En este PFC, se emplean diferentes implementaciones de estándares empleando RVC. Por ejemplo mediante los decodificadores Mpeg 4 Part 2 SP y Mpeg 4 Part 10 CBP y PHP así como del nuevo estándar de codificación HEVC, resaltando las características y utilidad de cada uno de ellos. En RVC los algoritmos se describen mediante una clase de actores que intercambian flujos de datos (tokens) para realizar diferentes acciones. El objetivo de este proyecto es desarrollar un programa que, partiendo de los decodificadores anteriormente mencionados, una serie de secuencia de vídeo en diferentes formatos de compresión y una distribución estándar de los actores (para cada uno de los decodificadores), sea capaz de generar diferentes distribuciones de los actores del decodificador sobre uno o varios procesadores del sistema sobre el que se ejecuta, para conseguir la mayor eficiencia en la codificación del vídeo. La finalidad del programa desarrollado en este proyecto es la de facilitar la realización de las distribuciones de los actores sobre los núcleos del sistema, y obtener las mejores configuraciones posibles de una manera automática y eficiente. ABSTRACT. The current scheme that exists in the field of standardization and the design of new video coding standards is becoming a difficult task to meet the evolving and dynamic community of video encoding. The problem was centered mainly in order to exploit all the features and similarities between different codecs and encoding standards. This has forced redesigning some parts common to several coding standards. This problem led to the emergence of a new initiative for standardization within the ISO / IEC MPEG committee, called Reconfigurable Video Coding (RVC). His main idea was to develop a video coding standard and gradually incrementase to update a library of components, providing flexibility and the ability to have a reconfigurable code using a new flow -oriented language Actors / data called CAL. This language is used for the specification of the standard library and to the instantiation model decoder. Later, a new video coding standard called High Efficiency Video Coding (HEVC), which currently is in continuous process of updating and development, which would improve the compression efficiency and video coding is developed. Obviously has developed a vision of using the methodology HEVC RVC. In this PFC, different implementations using RVC standard are used. For example, using decoders MPEG 4 Part 2 SP and MPEG 4 Part 10 CBP and PHP and the new coding standard HEVC, highlighting the features and usefulness of each. In RVC, the algorithms are described by a class of actors that exchange streams of data (tokens) to perform different actions. The objective of this project is to develop a program that, based on the aforementioned decoders, a series of video stream in different compression formats and a standard distribution of actors (for each of the decoders), is capable of generating different distributions decoder actors on one or more processors of the system on which it runs, to achieve greater efficiency in video coding. The purpose of the program developed in this project is to facilitate the realization of the distributions of the actors on the cores of the system, and get the best possible settings automatically and efficiently.
Resumo:
We present a framework for the analysis of the decoding delay in multiview video coding (MVC). We show that in real-time applications, an accurate estimation of the decoding delay is essential to achieve a minimum communication latency. As opposed to single-view codecs, the complexity of the multiview prediction structure and the parallel decoding of several views requires a systematic analysis of this decoding delay, which we solve using graph theory and a model of the decoder hardware architecture. Our framework assumes a decoder implementation in general purpose multi-core processors with multi-threading capabilities. For this hardware model, we show that frame processing times depend on the computational load of the decoder and we provide an iterative algorithm to compute jointly frame processing times and decoding delay. Finally, we show that decoding delay analysis can be applied to design decoders with the objective of minimizing the communication latency of the MVC system.
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Thesis (Ph.D.)--University of Washington, 2016-06
Resumo:
The advent of the Integrated Services Digital Network (ISDN) led to the standardisation of the first video codecs for interpersonal video communications, followed closely by the development of standards for the compression, storage and distribution of digital video in the PC environment, mainly targeted at CD-ROM storage. At the same time the second-generation digital wireless networks, and the third-generation networks being developed, have enough bandwidth to support digital video services. The radio propagation medium is a difficult environment in which to deploy low bit error rate, real time services such as video. The video coding standards designed for ISDN and storage applications, were targeted at low bit error rate levels, orders of magnitude lower than the typical bit error rates experienced on wireless networks. This thesis is concerned with the transmission of digital, compressed video over wireless networks. It investigates the behaviour of motion compensated, hybrid interframe DPCM/DCT video coding algorithms, which form the basis of current coding algorithms, in the presence of high bit error rates commonly found on digital wireless networks. A group of video codecs, based on the ITU-T H.261 standard, are developed which are robust to the burst errors experienced on radio channels. The radio link is simulated at low level, to generate typical error files that closely model real world situations, in a Rayleigh fading environment perturbed by co-channel interference, and on frequency selective channels which introduce inter symbol interference. Typical anti-multipath techniques, such as antenna diversity, are deployed to mitigate the effects of the channel. Link layer error control techniques are also investigated.
Resumo:
In this work, we present an adaptive unequal loss protection (ULP) scheme for H264/AVC video transmission over lossy networks. This scheme combines erasure coding, H.264/AVC error resilience techniques and importance measures in video coding. The unequal importance of the video packets is identified in the group of pictures (GOP) and the H.264/AVC data partitioning levels. The presented method can adaptively assign unequal amount of forward error correction (FEC) parity across the video packets according to the network conditions, such as the available network bandwidth, packet loss rate and average packet burst loss length. A near optimal algorithm is developed to deal with the FEC assignment for optimization. The simulation results show that our scheme can effectively utilize network resources such as bandwidth, while improving the quality of the video transmission. In addition, the proposed ULP strategy ensures graceful degradation of the received video quality as the packet loss rate increases. © 2010 IEEE.
Resumo:
With the rapid development of Internet technologies, video and audio processing are among the most important parts due to the constant requirements of high quality media contents. Along with the improvement of network environment and the hardware equipment, this demand is becoming more and more imperious, people prefer high quality videos and audios as well as the net streaming media resources. FFmpeg is a set of open source program about the A/V decoding. Many commercial players use FFmpeg as their displaying cores. This paper designed a simple and easy-to-use video player based on FFmpeg. The first part is about the basic theories and related knowledge of video displaying, including some concepts like data formats, streaming media data, video coding and decoding. In a word, the realization of the video player depend on the a set of video decoding process. The general idea about the process is to get the video packets from the Internet, to read the related protocols and de-encapsulate the protocols, to de-encapsulate the packaging data and to get encoded formats data, to decode them to pixel data that can be displayed directly through graphics cards. During the coding and decoding process, there could be different degrees of data losing, which is called lossy compression, but it usually does not influence the quality of user experiences. The second part is about the principle of the FFmpeg decoding process, that is one of the key point of the paper. In this project, FFmpeg is used for the main decoding task, by call some main functions and structures from FFmpeg class libraries, packaging video formats could be transfer to pixel data, after getting the pixel data, SDL is used for the displaying process. The third part is about the SDL displaying flow. Similarly, it would invoke some important displaying functions from SDL class libraries to realize the function, though SDL is able to do not only displaying task, but also many other game playing process. After that, a independent video displayer is completed, it is provided with all the key function of a player. The fourth part make a simple users interface for the player based on the MFC program, it enable the player could be used by most people. At last, in consideration of the mobile Internet’s blossom, people nowadays can hardly ever drop their mobile phones, there is a brief introduction about how to transplant the video player to Android platform which is one of the most used mobile systems.
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Hoje em dia, há cada vez mais informação audiovisual e as transmissões ou ficheiros multimédia podem ser partilhadas com facilidade e eficiência. No entanto, a adulteração de conteúdos vídeo, como informação financeira, notícias ou sessões de videoconferência utilizadas num tribunal, pode ter graves consequências devido à importância desse tipo de informação. Surge então, a necessidade de assegurar a autenticidade e a integridade da informação audiovisual. Nesta dissertação é proposto um sistema de autenticação de vídeo H.264/Advanced Video Coding (AVC), denominado Autenticação de Fluxos utilizando Projecções Aleatórias (AFPA), cujos procedimentos de autenticação, são realizados ao nível de cada imagem do vídeo. Este esquema permite um tipo de autenticação mais flexível, pois permite definir um limite máximo de modificações entre duas imagens. Para efectuar autenticação é utilizada uma nova técnica de autenticação de imagens, que combina a utilização de projecções aleatórias com um mecanismo de correcção de erros nos dados. Assim é possível autenticar cada imagem do vídeo, com um conjunto reduzido de bits de paridade da respectiva projecção aleatória. Como a informação de vídeo é tipicamente, transportada por protocolos não fiáveis pode sofrer perdas de pacotes. De forma a reduzir o efeito das perdas de pacotes, na qualidade do vídeo e na taxa de autenticação, é utilizada Unequal Error Protection (UEP). Para validação e comparação dos resultados implementou-se um sistema clássico que autentica fluxos de vídeo de forma típica, ou seja, recorrendo a assinaturas digitais e códigos de hash. Ambos os esquemas foram avaliados, relativamente ao overhead introduzido e da taxa de autenticação. Os resultados mostram que o sistema AFPA, utilizando um vídeo com qualidade elevada, reduz o overhead de autenticação em quatro vezes relativamente ao esquema que utiliza assinaturas digitais e códigos de hash.
Resumo:
A new high performance architecture for the computation of all the DCT operations adopted in the H.264/AVC and HEVC standards is proposed in this paper. Contrasting to other dedicated transform cores, the presented multi-standard transform architecture is supported on a completely configurable, scalable and unified structure, that is able to compute not only the forward and the inverse 8×8 and 4×4 integer DCTs and the 4×4 and 2×2 Hadamard transforms defined in the H.264/AVC standard, but also the 4×4, 8×8, 16×16 and 32×32 integer transforms adopted in HEVC. Experimental results obtained using a Xilinx Virtex-7 FPGA demonstrated the superior performance and hardware efficiency levels provided by the proposed structure, which outperforms its more prominent related designs by at least 1.8 times. When integrated in a multi-core embedded system, this architecture allows the computation, in real-time, of all the transforms mentioned above for resolutions as high as the 8k Ultra High Definition Television (UHDTV) (7680×4320 @ 30fps).
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A unified architecture for fast and efficient computation of the set of two-dimensional (2-D) transforms adopted by the most recent state-of-the-art digital video standards is presented in this paper. Contrasting to other designs with similar functionality, the presented architecture is supported on a scalable, modular and completely configurable processing structure. This flexible structure not only allows to easily reconfigure the architecture to support different transform kernels, but it also permits its resizing to efficiently support transforms of different orders (e. g. order-4, order-8, order-16 and order-32). Consequently, not only is it highly suitable to realize high-performance multi-standard transform cores, but it also offers highly efficient implementations of specialized processing structures addressing only a reduced subset of transforms that are used by a specific video standard. The experimental results that were obtained by prototyping several configurations of this processing structure in a Xilinx Virtex-7 FPGA show the superior performance and hardware efficiency levels provided by the proposed unified architecture for the implementation of transform cores for the Advanced Video Coding (AVC), Audio Video coding Standard (AVS), VC-1 and High Efficiency Video Coding (HEVC) standards. In addition, such results also demonstrate the ability of this processing structure to realize multi-standard transform cores supporting all the standards mentioned above and that are capable of processing the 8k Ultra High Definition Television (UHDTV) video format (7,680 x 4,320 at 30 fps) in real time.
Resumo:
With the shift towards many-core computer architectures, dataflow programming has been proposed as one potential solution for producing software that scales to a varying number of processor cores. Programming for parallel architectures is considered difficult as the current popular programming languages are inherently sequential and introducing parallelism is typically up to the programmer. Dataflow, however, is inherently parallel, describing an application as a directed graph, where nodes represent calculations and edges represent a data dependency in form of a queue. These queues are the only allowed communication between the nodes, making the dependencies between the nodes explicit and thereby also the parallelism. Once a node have the su cient inputs available, the node can, independently of any other node, perform calculations, consume inputs, and produce outputs. Data ow models have existed for several decades and have become popular for describing signal processing applications as the graph representation is a very natural representation within this eld. Digital lters are typically described with boxes and arrows also in textbooks. Data ow is also becoming more interesting in other domains, and in principle, any application working on an information stream ts the dataflow paradigm. Such applications are, among others, network protocols, cryptography, and multimedia applications. As an example, the MPEG group standardized a dataflow language called RVC-CAL to be use within reconfigurable video coding. Describing a video coder as a data ow network instead of with conventional programming languages, makes the coder more readable as it describes how the video dataflows through the different coding tools. While dataflow provides an intuitive representation for many applications, it also introduces some new problems that need to be solved in order for data ow to be more widely used. The explicit parallelism of a dataflow program is descriptive and enables an improved utilization of available processing units, however, the independent nodes also implies that some kind of scheduling is required. The need for efficient scheduling becomes even more evident when the number of nodes is larger than the number of processing units and several nodes are running concurrently on one processor core. There exist several data ow models of computation, with different trade-offs between expressiveness and analyzability. These vary from rather restricted but statically schedulable, with minimal scheduling overhead, to dynamic where each ring requires a ring rule to evaluated. The model used in this work, namely RVC-CAL, is a very expressive language, and in the general case it requires dynamic scheduling, however, the strong encapsulation of dataflow nodes enables analysis and the scheduling overhead can be reduced by using quasi-static, or piecewise static, scheduling techniques. The scheduling problem is concerned with nding the few scheduling decisions that must be run-time, while most decisions are pre-calculated. The result is then an, as small as possible, set of static schedules that are dynamically scheduled. To identify these dynamic decisions and to find the concrete schedules, this thesis shows how quasi-static scheduling can be represented as a model checking problem. This involves identifying the relevant information to generate a minimal but complete model to be used for model checking. The model must describe everything that may affect scheduling of the application while omitting everything else in order to avoid state space explosion. This kind of simplification is necessary to make the state space analysis feasible. For the model checker to nd the actual schedules, a set of scheduling strategies are de ned which are able to produce quasi-static schedulers for a wide range of applications. The results of this work show that actor composition with quasi-static scheduling can be used to transform data ow programs to t many different computer architecture with different type and number of cores. This in turn, enables dataflow to provide a more platform independent representation as one application can be fitted to a specific processor architecture without changing the actual program representation. Instead, the program representation is in the context of design space exploration optimized by the development tools to fit the target platform. This work focuses on representing the dataflow scheduling problem as a model checking problem and is implemented as part of a compiler infrastructure. The thesis also presents experimental results as evidence of the usefulness of the approach.
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This paper presents a paralleled Two-Pass Hexagonal (TPA) algorithm constituted by Linear Hashtable Motion Estimation Algorithm (LHMEA) and Hexagonal Search (HEXBS) for motion estimation. In the TPA., Motion Vectors (MV) are generated from the first-pass LHMEA and are used as predictors for second-pass HEXBS motion estimation, which only searches a small number of Macroblocks (MBs). We introduced hashtable into video processing and completed parallel implementation. We propose and evaluate parallel implementations of the LHMEA of TPA on clusters of workstations for real time video compression. It discusses how parallel video coding on load balanced multiprocessor systems can help, especially on motion estimation. The effect of load balancing for improved performance is discussed. The performance or the algorithm is evaluated by using standard video sequences and the results are compared to current algorithms.
Resumo:
This paper presents a paralleled Two-Pass Hexagonal (TPA) algorithm constituted by Linear Hashtable Motion Estimation Algorithm (LHMEA) and Hexagonal Search (HEXBS) for motion estimation. In the TPA, Motion Vectors (MV) are generated from the first-pass LHMEA and are used as predictors for second-pass HEXBS motion estimation, which only searches a small number of Macroblocks (MBs). We introduced hashtable into video processing and completed parallel implementation. We propose and evaluate parallel implementations of the LHMEA of TPA on clusters of workstations for real time video compression. It discusses how parallel video coding on load balanced multiprocessor systems can help, especially on motion estimation. The effect of load balancing for improved performance is discussed. The performance of the algorithm is evaluated by using standard video sequences and the results are compared to current algorithms.
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As digital systems move away from traditional desktop setups, new interaction paradigms are emerging that better integrate with users’ realworld surroundings, and better support users’ individual needs. While promising, these modern interaction paradigms also present new challenges, such as a lack of paradigm-specific tools to systematically evaluate and fully understand their use. This dissertation tackles this issue by framing empirical studies of three novel digital systems in embodied cognition – an exciting new perspective in cognitive science where the body and its interactions with the physical world take a central role in human cognition. This is achieved by first, focusing the design of all these systems on a contemporary interaction paradigm that emphasizes physical interaction on tangible interaction, a contemporary interaction paradigm; and second, by comprehensively studying user performance in these systems through a set of novel performance metrics grounded on epistemic actions, a relatively well established and studied construct in the literature on embodied cognition. The first system presented in this dissertation is an augmented Four-in-a-row board game. Three different versions of the game were developed, based on three different interaction paradigms (tangible, touch and mouse), and a repeated measures study involving 36 participants measured the occurrence of three simple epistemic actions across these three interfaces. The results highlight the relevance of epistemic actions in such a task and suggest that the different interaction paradigms afford instantiation of these actions in different ways. Additionally, the tangible version of the system supports the most rapid execution of these actions, providing novel quantitative insights into the real benefits of tangible systems. The second system presented in this dissertation is a tangible tabletop scheduling application. Two studies with single and paired users provide several insights into the impact of epistemic actions on the user experience when these are performed outside of a system’s sensing boundaries. These insights are clustered by the form, size and location of ideal interface areas for such offline epistemic actions to occur, as well as how can physical tokens be designed to better support them. Finally, and based on the results obtained to this point, the last study presented in this dissertation directly addresses the lack of empirical tools to formally evaluate tangible interaction. It presents a video-coding framework grounded on a systematic literature review of 78 papers, and evaluates its value as metric through a 60 participant study performed across three different research laboratories. The results highlight the usefulness and power of epistemic actions as a performance metric for tangible systems. In sum, through the use of such novel metrics in each of the three studies presented, this dissertation provides a better understanding of the real impact and benefits of designing and developing systems that feature tangible interaction.
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Motion estimation is the main responsible for data reduction in digital video encoding. It is also the most computational damanding step. H.264 is the newest standard for video compression and was planned to double the compression ratio achievied by previous standards. It was developed by the ITU-T Video Coding Experts Group (VCEG) together with the ISO/IEC Moving Picture Experts Group (MPEG) as the product of a partnership effort known as the Joint Video Team (JVT). H.264 presents novelties that improve the motion estimation efficiency, such as the adoption of variable block-size, quarter pixel precision and multiple reference frames. This work defines an architecture for motion estimation in hardware/software, using a full search algorithm, variable block-size and mode decision. This work consider the use of reconfigurable devices, soft-processors and development tools for embedded systems such as Quartus II, SOPC Builder, Nios II and ModelSim
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In recent years, due to the rapid convergence of multimedia services, Internet and wireless communications, there has been a growing trend of heterogeneity (in terms of channel bandwidths, mobility levels of terminals, end-user quality-of-service (QoS) requirements) for emerging integrated wired/wireless networks. Moreover, in nowadays systems, a multitude of users coexists within the same network, each of them with his own QoS requirement and bandwidth availability. In this framework, embedded source coding allowing partial decoding at various resolution is an appealing technique for multimedia transmissions. This dissertation includes my PhD research, mainly devoted to the study of embedded multimedia bitstreams in heterogenous networks, developed at the University of Bologna, advised by Prof. O. Andrisano and Prof. A. Conti, and at the University of California, San Diego (UCSD), where I spent eighteen months as a visiting scholar, advised by Prof. L. B. Milstein and Prof. P. C. Cosman. In order to improve the multimedia transmission quality over wireless channels, joint source and channel coding optimization is investigated in a 2D time-frequency resource block for an OFDM system. We show that knowing the order of diversity in time and/or frequency domain can assist image (video) coding in selecting optimal channel code rates (source and channel code rates). Then, adaptive modulation techniques, aimed at maximizing the spectral efficiency, are investigated as another possible solution for improving multimedia transmissions. For both slow and fast adaptive modulations, the effects of imperfect channel estimation errors are evaluated, showing that the fast technique, optimal in ideal systems, might be outperformed by the slow adaptive modulation, when a real test case is considered. Finally, the effects of co-channel interference and approximated bit error probability (BEP) are evaluated in adaptive modulation techniques, providing new decision regions concepts, and showing how the widely used BEP approximations lead to a substantial loss in the overall performance.