Currently different types of transform techniques are used by different video codecs to achieve data compression during video frame transmission. Among them, Discrete Cosine Transform (DCT) is supported by most of modern video standards. The integer DCT (Int-DCT) is an integer approximation of DCT. It can be implemented exclusively with integer arithmetic. Int-DCT proves to be highly advantageous in cost and speed for hardware implementations. In particular, the 4x4 and 8x8 block size Int-DCTs have the increased applicability at the current multimedia industry because of their simpler implementation and better de-correlation performance for high definition (HD) video signals. In this thesis, we present a fast and cost-shared reconfigurable architecture to compute variable block size Int-DCT for four modern video codecs – AVS, H.264/AVC, VC-1 and HEVC (under development). Based on the symmetric structure of the transform matrices and the similarity in matrix operations, we have developed a generalized “decompose and share” algorithm to compute the 4x4 and 8x8 block size Int-DCT. The algorithm is later applied to those four video codecs. Our shared hardware approach ensures the maximum circuit reuse during the computation. The entire architecture is multiplier free and designed with only adders and shifters to minimize hardware cost and improve working frequency. Finally, the design is implemented on a FPGA and later synthesized in CMOS 0.18um technology to compare the cost and performance with existing designs. The results show significant reduction in hardware cost and meet the requirements of real time video coding applications.

The thesis is focused on implementation of aplication for secure telephone communication on data network. Application is developed for operating system Android. For call management is responsible signaling protocol SIP and for transfer of voice data is used protocol RTP. For security of call is first created cryptografic key for symetric cryptography. After generating key is established call, which is encrypted by symetric cipher AES. Encrypting between communicating sides is provided in application or on microSD card. Part of solution is measurement of speed of cryptographic primitives, which are used for secure call.

Dans cette thèse, nous nous intéressons au déploiement à grande échelle de la Voix sur IP (VoIP) dans des environnements Internet hétérogènes. Après une description des mécanismes de codage et de transmission de la voix sur l'Internet, nous étudions dans une première partie de la thèse, les limites de performance dans le cas d'une transmission d'un grand nombre de flux de voix sur IP entre deux passerelles téléphoniques. Nous discutons le besoin d'utilisation de mécanismes de contrôle de congestion pour le trafic de voix sur IP qui est en croissance continue sur l'Internet. Nous proposons un nouveau schéma de contrôle de congestion de la voix sur IP. Ce schéma combine le multiplexage de flux RTP et le mécanisme de contrôle TCP-amical (TCP-friendly) afin d'améliorer l'efficacité et la performance de la transmission des flux de voix sur IP et de garantir l'équité avec les autres types de trafic coexistant sur l'Internet. La deuxième partie de la thèse est consacrée à l'étude de la transmission de la voix dans des environnements de réseaux locaux sans fil IEEE 802.11e. Nous développons un modèle analytique permettant d'évaluer la capacité d'un réseau 802.11e en nombre de communications de voix sur IP en fonction des paramètres de l'application (codage audio utilisé) ainsi que des paramètres relatifs aux canal de transmission sans fil. Ce modèle peut être utilisé pour ajuster ces paramètres afin d'augmenter la capacité du réseau sans fil tout en considérant les contraintes strictes des communications interactives de la voix sur IP. Dans la dernière partie de la thèse, nous étudions le cas de la transmission de la voix sur IP dans des environnements Internet hétérogènes (filaires/sans fil) constitués en partie par des liens d'accès sans fil. Nous proposons une architecture basée sur une passerelle de voix sur IP placée au bord du réseau sans fil. Cette passerelle est utilisée pour adapter les flux de voix aux caractéristiques du réseau sans fil. Le mécanisme d'adaptation proposé estime dynamiquement l'état de congestion du canal sans fil et permet la différentiation entre les pertes de paquets causées par la congestion et celles dûes aux erreurs de transmission sur le canal sans fil. L'adaptation appropriée est alors appliquée. Le mécanisme d'adaptation proposé, ne nécessite pas de modifications du protocole de contrôle d'accès au canal sans fil (MAC), ce qui facilite son déploiement sur l'infrastructure réseau existante.

<p>Over the last three decades, computer architects have been able to achieve an increase in performance for single processors by, e.g., increasing clock speed, introducing cache memories and using instruction level parallelism. However, because of power consumption and heat dissipation constraints, this trend is going to cease. In recent times, hardware engineers have instead moved to new chip architectures with multiple processor cores on a single chip. With multi-core processors, applications can complete more total work than with one core alone. To take advantage of multi-core processors, we have to develop parallel applications that assign tasks to different cores. On each core, pipeline, data and task parallelization can be used to achieve higher performance. Dataflow programming languages are attractive for achieving parallelism because of their high-level, machine-independent, implicitly parallel notation and because of their fine-grain parallelism. These features are essential for obtaining effective, scalable utilization of multi-core processors.</p><p>In this thesis work we have parallelized an existing audio codec - Adaptive Multi-Rate Wide Band (AMR-WB) - written in the C language for single core processor. The target platform is a multi-core AMR11 MP developer board. The final result of the efforts is a working AMR-WB encoder implemented in CAL and running in the OpenDF simulator. The C specification of the AMR-WB encoder was analysed with respect to dataflow and parallelism. The final implementation was developed in the CAL Actor Language, with the goal of exposing available parallelism - different dataflows - as well as removing unwanted data dependencies. Our thesis work discusses mapping techniques and guidelines that we followed and which can be used in any future work regarding mapping C based applications to CAL. We also propose solutions for some specific dependencies that were revealed in the AMR-WB encoder analysis and suggest further investigation of possible modifications to the encoder to enable more efficient implementation on a multi-core target system.</p>

Continuous improvements to the state of the art have made it easier to create, send and receive vast quantities of video over the Internet. Catalysed by these developments, video is now the largest, and fastest growing type of traffic on modern IP networks. In 2015, video was responsible for 70% of all traffic on the Internet, with an compound annual growth rate of 27%. On the other hand, concerns about the growing energy consumption of ICT in general, continue to rise. It is not surprising that there is a significant energy cost associated with these extensive video usage patterns. In this thesis, I examine the energy consumption of typical video configurations during decoding (playback) and encoding through empirical measurements on an experimental test-bed. I then make extrapolations to a global scale to show the opportunity for significant energy savings, achievable by simple modifications to these video configurations. Based on insights gained from these measurements, I propose a novel, energy-aware Quality of Experience (QoE) metric for digital video - the Energy - Video Quality Index (EnVI). Then, I present and evaluate vEQ-benchmark, a benchmarking and measurement tool for the purpose of generating EnVI scores. The tool enables fine-grained resource-usage analyses on video playback systems, and facilitates the creation of statistical models of power usage for these systems. I propose GreenDASH, an energy-aware extension of the existing Dynamic Adaptive Streaming over HTTP standard (DASH). GreenDASH incorporates relevant energy-usage and video quality information into the existing standard. It could enable dynamic, energy-aware adaptation for video in response to energy-usage and user ‘green' preferences. I also evaluate the subjective perception of such energy-aware, adaptive video streaming by means of a user study featuring 36 participants. I examine how video may be adapted to save energy without a significant impact on the Quality of Experience of these users. In summary, this thesis highlights the significant opportunities for energy savings if Internet users gain an awareness about their energy usage, and presents a technical discussion how this can be achieved by straightforward extensions to the current state of the art.

Over the last three decades, computer architects have been able to achieve an increase in performance for single processors by, e.g., increasing clock speed, introducing cache memories and using instruction level parallelism. However, because of power consumption and heat dissipation constraints, this trend is going to cease. In recent times, hardware engineers have instead moved to new chip architectures with multiple processor cores on a single chip. With multi-core processors, applications can complete more total work than with one core alone. To take advantage of multi-core processors, we have to develop parallel applications that assign tasks to different cores. On each core, pipeline, data and task parallelization can be used to achieve higher performance. Dataflow programming languages are attractive for achieving parallelism because of their high-level, machine-independent, implicitly parallel notation and because of their fine-grain parallelism. These features are essential for obtaining effective, scalable utilization of multi-core processors. In this thesis work we have parallelized an existing audio codec - Adaptive Multi-Rate Wide Band (AMR-WB) - written in the C language for single core processor. The target platform is a multi-core AMR11 MP developer board. The final result of the efforts is a working AMR-WB encoder implemented in CAL and running in the OpenDF simulator. The C specification of the AMR-WB encoder was analysed with respect to dataflow and parallelism. The final implementation was developed in the CAL Actor Language, with the goal of exposing available parallelism - different dataflows - as well as removing unwanted data dependencies. Our thesis work discusses mapping techniques and guidelines that we followed and which can be used in any future work regarding mapping C based applications to CAL. We also propose solutions for some specific dependencies that were revealed in the AMR-WB encoder analysis and suggest further investigation of possible modifications to the encoder to enable more efficient implementation on a multi-core target system.

This diploma thesis focus on transmiting multimedia over computer network. There are detailed informations about protocols RTP and RTPC in a first part because a transmition over a network is realized by using these protocols. Some basic multimedia terms, FFmpeg codecs and SDL library are described in next chapters. A multimedia player using FFmpeg and SDL is implemented in a second part of thesis. The player is console application and it has basic user interface. The player reproduces video and audio from a given file. RTP communication is described in next chapters of the second part. RTP server and client are implemented there too. They are console aplications and they use data coded by Theora or Vorbis. There are summarized results in a conclusion.

This master's thesis is about making draft and implementing land-line phone call encryption using FITkit. The ultimate goal is to find suitable compression and encryption methods, implement or adapt them for FITkit board and create functional solution.

Submitted by Johnny Rodrigues (johnnyrodrigues@ufcg.edu.br) on 2018-05-07T15:46:17Z No. of bitstreams: 1 LUIZ TEIXEIRA MACHADO NETO - DISSERTAÇÃO PPGCC 2015..pdf: 936539 bytes, checksum: 75e4ab7774a9d4c965e23ec09254fc86 (MD5) / Made available in DSpace on 2018-05-07T15:46:17Z (GMT). No. of bitstreams: 1 LUIZ TEIXEIRA MACHADO NETO - DISSERTAÇÃO PPGCC 2015..pdf: 936539 bytes, checksum: 75e4ab7774a9d4c965e23ec09254fc86 (MD5) Previous issue date: 2015-03-01 / A utilização de serviços de streaming cresceu bastante nos últimos anos, por meio de sistemas como Youtube, Hulu, Netflix, Vimeo, etc. Utilizando o stream, os vídeos são transmitidos e exibidos em tempo real e em qualquer lugar do mundo por meio da internet. Muitosusuáriosdestessistemasosutilizamemsuaresidênciaondeébastante comumencontrarumaredesemfio(devidoàmobilidadequesepodealcançarpormeio deste tipo de rede). Apesar de mais mobilidade, uma rede sem fio está mais suscetível a interferências do que a rede cabeada e, por isso, um vídeo pode ter sua imagem degradada com mais facilidade. Neste trabalho, é avaliada a transmissão de vídeos via stream para descobrir se o vídeo realmente é degradado pela transmissão; se tipos de conteúdos diferentes afetam a qualidade do vídeo recebido; e se de acordo com o padrão de compressão utilizado, é possível observar melhorias na qualidade do vídeo recebido. Por meio de uma abordagem experimental com um design de experimentos fatorial completo, foram feitas transmissões de vídeos utilizando o H.264, o HEVC e o MPEG-4; padrões mais utilizados atualmente. Além de definir os padrões, foram definidas outras variáveis: porcentagem de ocupação do canal de transmissão (com o objetivo de avaliar a degradação dos vídeos de acordo com a competição que a rede está sofrendo); potência do sinal de transmissão (com o objetivo de avaliar o impacto da qualidade do sinal da rede no vídeo recebido); quantidade de movimento no vídeo (para avaliar se a quantidade de movimento que o vídeo exibe impacta na sua qualidade). Nos experimentos, foi utilizado um ambiente residencial que conta as interferências de outras redes, exatamente como pode acontecer em um ambiente real. Para definir as porcentagens de ocupação, foram realizados experimentos para medição da capacidade máxima de transmissão da rede de testes. Em se tratando da quantidade de movimento, foi necessário fazer uma classificação prévia dos vídeos de acordo com características espaciais e temporais de cada vídeo. Os vídeos foram separados em três categorias e dentro dessas categorias, três vídeos foram escolhidos aleatoriamente para participar dos experimentos. Os resultados mostram que o HEVC obteveamelhormédiaparaasmétricasdequalidadedevídeoescolhidas, sendoocodec que menos perde qualidade numa transmissão sem fio. Também foi possível observar que a quantidade de movimento foi o parâmetro que mais influenciou na qualidade do vídeos nos experimentos realizados. / The use of streaming services has grown significantly in recent years, through systems such as Youtube, Hulu, Netflix, Vimeo, etc. Using the stream, videos are transmitted and displayed in real time and from anywhere in the world via the Internet. Many users of these systems use the same in their homes where it is quite common to find a wireless network (due to the mobility we can achieve through this type of computer network). In spite of having more mobility, a wireless network is more susceptible to interference than the wired network so a video can have its picture degraded more easily just because it is transmitted over a wireless network. We evaluate the transmission of videos via stream to find out whether the video is actually degraded by transmission; if different types of content affect the quality of the received video; and if there are compression standarts (H.264 and MPEG-4 HEVC, the most currently used) which ensure a better received video quality. Through an experimental approach with a design of full factorial experiments, several transmissions of videos were made in the three chosen standarts. In addition to defining the standarts, other variables were defined as: transmission channel occupancy percentage (in order to evaluate the degradation of videos according to occupation); power transmission signal (in order to assess the impact of network signal quality in the video received); amount of motion in the video (to evaluate whether the amount of motion the video displays impacts on its quality). For the experiments we used a residential environment that has all the interference from other networks, just as it can happen in a real environment. To set the occupancy percentages, experiments were performed to measure the maximum transmission capacity of the test network. Concerning the quantity of movement, it was necessary to make a preliminary classification of videos according to spatial and temporal characteristics of each one. The videos were separated into three categories and within these categories, three videos were chosen at random to participate in the experiments. The results show that the HEVC achieved the highest average for quality metricsofthechosenvideos,andthecodecistheonethatloseslessqualityinawireless transmission. The amount of movement affects the quality of the received video, and the greater the amount of motion, the bigger the loss of image quality.