Streaming Media

The reliance on Network-on-Chip (NoC) based Multiprocessor Systems-on-Chips (MPSoCs) is proliferating in modern embedded systems to satisfy the higher performance requirement of multimedia streaming applications. Task level coarse grained software pipeling also called re-timing when combined with Dynamic Voltage and Frequency Scaling (DVFS) has shown to be an effective approach in significantly reducing energy consumption of the multiprocessor systems at the expense of additional delay. In this paper we develop a novel energy-aware scheduler considering tasks with conditional constraints on Voltage Frequency Island (VFI) based heterogeneous NoC-MPSoCs deploying re-timing integrated with DVFS for real-time streaming applications. We propose a novel task level re-timing approach called R-CTG and integrate it with non linear programming based scheduling and voltage scaling approach referred to as ALI-EBAD. The R-CTG approach aims to minimize the latency Utilizing features such as video-streams transmitted via the Internet, these systems are able to monitor the healthcare needs for anyone. This is considered most beneficial to people with increases support needs such as elderly, inform or those with alternative abilities. The video and data streams are stored securely in the cloud but available for access by the right people when needed. An abstract IoT based healthcare system architecture is demonstrated in Fig. (1) [1], [2]. Remote Monitoring caused by re-timing without compromising on energy-efficiency. Compared to R-DAG, the state-of-the-art approach designed for traditional Directed Acyclic Graph (DAG) based task graphs, R-CTG significantly reduces the re-timing latency because it only re-times tasks that free up the wasted slack. To validate our claims we performed experiments on using 12 real benchmarks, the results demonstrate that ALI-EBAD out performs CA-TMES-Visualization GUI Search and CA-TMES-Quick task schedulers in terms of energyefficiency.

The modern expansion of public streaming has played a considerable part in the growth of distributed multime- dia, with special reference to video. However, since the Internet is an inter-connected web of different network speeds, streaming video is still a difficult service to implement to all locations. This paper illustrates two simple simulated network streaming environments. The first models a broadcast network and the second model represents a multicast network. A simple cache system is then implemented into these models to investigate whether a significant difference exists between the cache size and the different network bandwidths of these two optimization systems. This caching system will then be used to compare the impact of videos (characterised by different resolutions and degrees of picture motion) on the cache performance. Index Terms—Network simulation, Cache, Multicasting, Video streaming, NS-2, Distributed Multimedia, Evalvid, I. INTRODUCTION Distributed media stre...

A new network architecture for the Internet needs ingredients from three approaches: information-centric networking, cloud computing integrated with networking, and open connectivity. Information-centric networking considers pieces of information as first-class entities of a networking architecture, rather than only indirectly identifying and manipulating them via a node hosting that information; this way, information becomes independent from the devices they are stored in, enabling efficient and application-independent information caching in the network. Cloud networking offers a combination and integration of cloud computing and virtual networking. It is a solution that distributes the benefits of cloud computing more deeply into the network, and provides a tighter integration of virtualisation features at computing and networking levels. To support these concepts, open connectivity services need to provide advanced transport and networking mechanisms, making use of network and path diversity (even leveraging direct optical paths) and encoding techniques, and dealing with ubiquitous mobility of user, content and information objects in a unified way.

Most applications and network protocols are designed under the assumption of a stable and consistent physical network infrastructure. However, the communications infrastructures of most military tactical networks are dynamic and dependent on mission CONOPS and network topology. Further, they are highly variable in their capabilities from one data link to the next. Consequently, it is difficult for standard applications and services to maintain reliable quality of service when transiting to a dynamic tactical network, or crossing the borders between different classes of military networks. Ultimately, this lack of network awareness impacts the quality and timeliness of actionable intelligence delivered to the warfighter. have collaborated on efforts to improve the reliable delivery of live video data across these disadvantaged tactical networks. We address the development of a network-aware video server, which senses network characteristics and dynamically transcodes video to meet the capacity of network between the server and clients. End-to-end network characteristics are required to drive the video transcoder. These network attributes can either be derived from software methods by analyzing the network protocol, or using specialized network hardware (e.g. tactical radios, routers, or QoS devices). For this effort end-to-end network awareness was collected from the NACK-Oriented Reliable Multicast (NORM) [1] protocol operation. Hooks in the NORM protocol provide a network information service with access to path bandwidth, delay, and lost packets. The video server uses the network information service to adapt the video by adjusting frame size, frame rate, and compression according to current network conditions to deliver the video with minimal delay. A comparison of how video is streamed over enterprise networks and the improved delivery methods made by General Dynamics and NRL for tactical networks is presented. I.

A novel experimental approach for investigating the performances of context-aware content delivery schemes is presented in this paper. An innovative testbed, capable of remotely controlling multiple terminals, injecting a wide range of traffic loads in real networks and monitoring different performance measures has been developed and utilized for quantifying both the energy costs and user perceived service quality associated with different context-aware content pre-fetching schemes. In implementation proposed in this paper, the context information required for performing content pre-fetching is extracted and utilized by individual user terminals and does not require any support from mobile operators. The performances of pre-fetching are compared to those of an on-demand content delivery scheme, for both video streaming and file downloading services. The results show that not only pre-fetching can increase user service appreciation by reducing the time needed to access the information, but it can also significantly lower the amount of energy consumed in user terminals for retrieving the content. Our experiments further indicate that in order to achieve these additional energy gains only limited content prediction capabilities are required, thus making pre-fetching a solid candidate for the provision of a wide range of content types and services in both wide and local area networks.

Resumo: O aumento da demanda de serviços multimídia sobre redes de computadores tem despertado a atenção da comunidade de pesquisa nos últimos anos. As transmissões de conteúdo utilizando áudio e vídeo vêm ganhando espaço nas universidades, empresas, sistemas de ensino a distância, governo, além de alcançar a todos que possuem acesso a Internet. A grande aceitação e, muitas vezes, a preferência dos usuários, evidencia o crescimento do consumo deste tipo de conteúdo na busca por entretenimento e educação. No entanto, as transmissões multimídia acrescentam maior grau de sobrecarga em uma rede, principalmente quando se trata das populares redes WLAN (Wireless Local Area Network), devido à maior probabilidade de interferências. Esta realidade justifica as diversas iniciativas de estudos e pesquisas que procuram desenvolver e propor melhorias em termos de qualidade de serviço e qualidade de vídeo nas transmissões de tráfego multimídia em redes IP ( Internet Protocol). Geralmente, as pesquisas possuem características experimentais, onde os pesquisadores desenvolvem rotinas próprias a fim de otimizar as coletas de dados que possam ser usados em suas propostas. Porém, muito do que se faz em um ambiente de transmissão multimídia, caracteriza-se por tarefas repetitivas e manuais, principalmente nas etapas de transmissão de conteúdo, coleta, sintetização e análise dos dados. Este trabalho apresenta uma proposta de testbed reutilizável, que automatiza as tarefas repetitivas para testes desta natureza, proporcionando ao pesquisador maior dedicação a proposta da pesquisa e a análise dos dados coletados durante as seqüências de transmissão.

The remarkable success of deep learning in various domains relies on the availability of large-scale annotated datasets. However, obtaining annotations is expensive and requires great effort, which is especially challenging for videos. Moreover, the use of human-generated annotations leads to models with biased learning and poor domain generalization and robustness. As an alternative, self-supervised learning provides a way for representation learning that does not require annotations and has shown promise in both image and video domains. In contrast to the image domain, learning video representations are more challenging due to the temporal dimension, bringing in motion and other environmental dynamics. This also provides opportunities for video-exclusive ideas that advance self-supervised learning in the video and multimodal domains. In this survey, we provide a review of existing approaches on self-supervised learning focusing on the video domain. We summarize these methods into ...

Na verdade, o cinema sempre foi múltiplo, mas esta multiplicidade foi, por assim dizer, encoberta e/ou recalcada por sua forma dominante. Ao longo da história do cinema temos não apenas experiências esparsas, mas cinco momentos fortes (cinema do dispositivo, cinema experimental, arte do vídeo, cinema expandido, cinema interativo) que se notabilizam por grandes transformações e experimentações quanto ao dispositivo cinematográfico, sobretudo depois do pós-guerra. Antes, porém, gostaríamos de situar uma série de questões conceituais. Manuela Penafria & Índia Mara Martins típica dos fenômenos imaginários, que nos dão uma imagem em troca de um mundo. É assim que, para Baudry, o dispositivo cinematográfico -a projeção, a sala escura, a imobilidade do espectador -re-encena a fase do espelho tal como descrita por Lacan, na qual uma criança entre oito e dezoito meses se constitui em sujeito, pela identificação com a imagem de seu corpo unificado, visto no espelho. Tal situação é a mesma daquela do espectador de cinema, ambos são vítimas de uma submotricidade e de uma super percepção. O dispositivo do cinema tem a ver com o dispositivo psíquico na medida em que, neles, o sujeito é uma ilusão produzida a partir de um lugar. Por me encontrar no centro da cena, "o espectador se identifica menos com o que é representado no espetáculo, do que com que produz o espetáculo: com o que não é visível, mas torna visível". Trata-se, tanto no cinema, como na fase do espelho, de um sujeito transcendental que se constitui por se encontrar no centro, e, estando no centro, se sentir como condição de possibilidade do que existe. Esta relação traz consigo o processo genético das relações estruturais, onde a oposição e a falta são determinantes. Como no dispositivo de representação conhecido como campo/contracampo, o dispositivo cinematográfico é ao mesmo tempo, um conjunto de relações aonde cada elemento se define por oposição aos outros (presente/ausente), e aonde o espaço do ausente (imaginário) se torna o lugar (vimos que é ele que torna visível) onde uma não-presença se mistura, ou melhor, se sobrepõe a uma presença. A análise de Baudry apresenta duas questões principais. A primeira diz respeito ao fato de que, para ele, o dispositivo é fruto de uma visada sincrônica que, portanto, despreza as suas transformações históricas. Em segundo lugar, Baudry em sua crítica do cinema dominante, e talvez por uma certa preocupação com a questão da especificidade, deixa de lado a questão da organização discursiva do cinema, o que na verdade era um dos pontos centrais da crítica que se vinha tecendo ao modelo de representação do cinema comercial. Jean-Louis Comolli, em ensaio intitulado Técnica e Ideologia -Câmera, Perspectiva, Profundidade de Campo (Cahiers du Cinéma, 1971Cinéma, -1972)), transfere para a organização discursiva o que, antes, em Baudry, parecia ser um efeito específico do aparelho de base, pela câmera/projeção. Hoje, está claro que o dispositivo cinematográfico apresenta, ao lado das dimensões arquitetônicas e técnicas, uma dimensão discursivo-formal ou estético-formal que é

Parallel computing is currently the dominating architecture in embedded systems. Concurrency improves the performance of the system rather without increasing the clock speed which affects the power consumption of the system. However, concurrency needs to be exploited in order to improve the system performance in different applications environments. Multimedia applications (real-time conversational services such as video conferencing, video phone, etc.) have many new features that require complex computations compared to previous video coding standards. These applications have a challenging workload for future multiprocessors. Exploiting parallelism in multimedia applications can be done at data and functional levels or using different instruction sets and architectures. In this research, we design new parallel algorithms and mapping methodologies in order to exploit the natural existence of parallelism in multimedia applications, specifically the H.264/AVC video decoder. We mainly target symmetric shared-memory multiprocessors (SMPs) for embedded devices such as ARM Cortex-A9 multicore chips. We evaluate our novel parallel algorithms of the H.264/AVC video decoder on different levels : memory load, energy consumption, and execution time.

Parallel computing is currently the dominating architecture in embedded systems. Concurrency improves the performance of the system rather without increasing the clock speed which affects the power consumption of the system. However, concurrency needs to be exploited in order to improve the system performance in different applications environments. Multimedia applications (real-time conversational services such as video conferencing, video phone, etc.) have many new features that require complex computations compared to previous video coding standards. These applications have a challenging workload for future multiprocessors. Exploiting parallelism in multimedia applications can be done at data and functional levels or using different instruction sets and architectures. In this research, we design new parallel algorithms and mapping methodologies in order to exploit the natural existence of parallelism in multimedia applications, specifically the H.264/AVC video decoder. We mainly target symmetric shared-memory multiprocessors (SMPs) for embedded devices such as ARM Cortex-A9 multicore chips. We evaluate our novel parallel algorithms of the H.264/AVC video decoder on different levels : memory load, energy consumption, and execution time.

Distributed media streaming has been driven by the combination of improved media compression techniques and an increase in the availability of bandwidth. This increase has lead to the development of various streaming frameworks, which currently provide the majority of the streaming media available throughout the Internet. This study seeks to analyse a range of existing commercial and open-source streaming media distribu- tion engines, and will attempt to classify them in such a way as to define a Common Analysis Framework. The common framework will be used as the basis for a simulation tool intended to predict the performance impacts of network configuration changes, new features, and general scaling. The intention is to validate the theoretical simulation tool against empirical evidence collected from configuration and engineering extensions to existing systems. The research will be undertaken using an experimental Computer Science approach, in which a system is developed or extend...

The modern expansion of public streaming has played a considerable part in the growth of distributed multime- dia, with special reference to video. However, since the Internet is an inter-connected web of different network speeds, streaming video is still a difficult service to implement to all locations. This paper illustrates two simple simulated network streaming environments. The first models a broadcast network and the second model represents a multicast network. A simple cache system is then implemented into these models to investigate whether a significant difference exists between the cache size and the different network bandwidths of these two optimization systems. This caching system will then be used to compare the impact of videos (characterised by different resolutions and degrees of picture motion) on the cache performance. Index Terms—Network simulation, Cache, Multicasting, Video streaming, NS-2, Distributed Multimedia, Evalvid, I. INTRODUCTION Distributed media stre...

With the standardization of SVC, the scalable extension of H.264/AVC, layered peer-to-peer streaming has attracted more and more attention as it offers adaptability to network fluctuations and heterogeneous end users. Although overlay construction is important for system performance, not much effort has been spent on unstructured overlay construction for layered peer-to-peer streaming. Related work concentrates on layered streaming algorithms, and assumes that a list of peers for data exchange, called neighbors, is provided by traditional membership management protocols, e.g., SCAMP. Our previous studies have demonstrated that a random overlay is not good enough for layered peer-to-peer streaming. In this paper, we propose a new membership management protocol, based on peer sampling services. The protocol is quality-aware as it constructs the overlay so that (1) high capacity peers will be located at good positions in the overlay, e.g., close to the server, and (2) peers having similar capacity are likely to connect to each other. Both features are necessary to maximize bandwidth utilization of peers and to mitigate layer bottlenecks. With implementation of the protocol in PeerSim, we evaluate important graph properties of the overlay, constructed by the proposed protocol, to understand how it evolves during the streaming session with peer churn. Evaluation results show that the overlay is (1) scalable: it is stable with different sizes, from hundreds up to 10000 peers; and (2) robust: the good features are maintained or recovered fast under a high peer churn rate, and it only becomes disconnected when more than 86% of the peers are removed from the network.

Multiparty networked applications, which are gaining interest in areas as diverse as online games, e-learning, e-health, and collaborative work, tend to depend on a set of common functions. In conventional IP networks, these shared functions require application-specific servers, leading to scale diseconomies and complicating interoperability. The IP Multimedia Subsystem addresses this by incorporating basic services called service enablers, which can be used as building blocks to facilitate application development. This article discusses our experience creating innovative enablers for IMS networks, and combining them into new multimedia applications. An enabler-centric process is described, and the configuration of a development IMS network is explained, with suggestions on how to replicate this architecture with freely available components. Following these principles, we designed and implemented a series of novel group-centric enablers exploiting the advantages of group management, which we describe in detail. A use case made possible by these enablers is shown, consisting of an e-health scenario for remote medical visits. We reflect on the advantages and challenges introduced by this development methodology, and provide insight on how a project of this nature should be approached.

Virtual drama is based on the use of a shareable virtual world as a stage setting, with avatars controlled by actors and audience members. The Caracol Time Travel Project was an experiment in the use of virtual drama for learning about archaeology. Eighteen undergraduate students at the University of Central Florida used a locally developed Java‐based system for sharing VRML worlds. They designed and constructed a virtual drama to teach basic concepts of Mesoamerican archaeology and the cultural history of the ancient Maya for middle schools. This paper presents their story design and details of the system we developed to support interaction in this shared virtual world. We then discuss performance issues, lessons learned and newer features that we did not have available at the time. Copyright © 2001 John Wiley & Sons, Ltd.

Distributed media streaming has been driven by the combination of improved media compression techniques and an increase in the availability of bandwidth. This increase has lead to the development of various streaming frameworks, which currently provide the majority of the streaming media available throughout the Internet. This study seeks to analyse a range of existing commercial and open-source streaming media distribu- tion engines, and will attempt to classify them in such a way as to define a Common Analysis Framework. The common framework will be used as the basis for a simulation tool intended to predict the performance impacts of network configuration changes, new features, and general scaling. The intention is to validate the theoretical simulation tool against empirical evidence collected from configuration and engineering extensions to existing systems. The research will be undertaken using an experimental Computer Science approach, in which a system is developed or extend...

This paper is on a microcontroller-based camera controller with sound source localization (SSL). With the rising frequency of highway accidents in Malaysia, there is a pressing need for a reliable detection system. The current approach, involving fixed-angled cameras, necessitates constant human monitoring, proving inefficient. To address this, the study introduces a hybrid camera system incorporating a camera for image capture and a microphone to detect collision sounds. By integrating a pan-tilt (PT) camera controller driven by time difference of arrival (TDOA) inputs, the system can swiftly move toward accident locations. The TDOA method is employed to convert sound arrival time differences into camera angles. The accuracy of the PT camera's rotation angle was analyzed based on the original sound source angle. As a result, this project produced an automated highway monitoring camera system that uses sound SSL to detect car crash sounds on highways. Its PT feature will help cover a large highway area and eliminate blind spots to capture possible accident scenes. The average inaccuracy of the experimental test of the pan and tilt angle of the camera is 19 and 23%, respectively. The accuracy of the pan tilt angle can be increased by adding more analog acoustic sensors.

S ession Initiation Protocol (S IP) has been adopted by telecom Next Generation Networks (NGN) and 3rd Generation Partnership Project's IP Multimedia S ubsystem (IMS). Charging is one of the most fundamental components of any commercial S IP service. S ervice providers have to employ an online charging mechanism to support the prepaid payment model and to create flexible pricing policies that interwork with customers in real-time, to achieve marketing targets while improving customer experience. In this paper, we discuss the requirements of online charging mechanisms for S IP services, investigate the existing online charging mechanisms for S IP services namely AAA-based and S IP-based, and compare between them according to their security, performance, supported charging services, and supported service access types. Our contributions are: (1) comparing between AAA-based and S IP-based online charging mechanisms, (2) checking the vulnerability of the AAA-based online charging mechanism to charging attacks implemented by R. Zhang et al. on Vonage and AT&T VoIP service providers, (3) identifying new security attacks against the latest S IP-based online charging mechanismS IPCoin-. Index Term-AAA, charging, micro-payment, S IP 1.

SIP (The Session Initiation Protocol) is widely used as the signaling protocol for various services in the omnipresent environment. SIP is textual based protocol so the ability to process SIP messages quickly is critical for the performance of consumer electronic devices, such as SIP phone and the various Gateway. Our work enables homogeneous communication between heterogeneous distributed devices. Using SIP protocol communication bus to control widely varying entities, including ZigBEE, serial based (RS232) X10 and other based devices.

Video oriented services are taking the lead in terms of revenue in the current Internet environment. In order to tap this revenue, Telcos are resorting to serious measures such as moving to Next Generation Network (NGN) where they can provide QoS for services that demand this feature, such as, IPTV. IPTV is more than just what its name implies and

This paper introduces the experience of using videoconferencing and recording as a mechanism to support courses which need to be promoted or discontinued within the framework of the European convergence process. Our objective is to make these courses accessible as live streaming during the lessons as well as recorded lectures and associated documents available to the students as soon as the lesson has finished. The technology used has been developed in our university and it is all open source. Although this is a technical project the key is the human factor involved. The people managing the virtual sessions are students of the courses being recorded. However, they lack technical knowledge, so we had to train them in audiovisuals and enhance the usability of the videoconferencing tool and platform. The validation process is being carried out in five real scenarios at our university. During the whole period we are evaluating technical and pedagogical issues of this experience for both students and teachers to guide the future development of the service. Depending on the final results, the service of lectures recording will be available as educational resource for all of the teaching staff of our university.

With the proliferation of multimedia content on the Internet, there is an increasing demand for video streams with high perceptual quality. The capability of present-day Internet links in delivering high-perceptual-quality streaming services, however, is not completely understood. Link-level degradations caused by intradomain routing policies and inter-ISP peering policies are hard to obtain, as Internet service providers often consider such information proprietary. Understanding link-level degradations will enable us in designing future protocols, policies, and architectures to meet the rising multimedia demands. This paper presents a trace-driven study to understand quality-of-experience (QoE) capabilities of present-day Internet links using 51 diverse ISPs with a major presence in the US, Europe, and Asia-Pacific. We study their links from 38 vantage points in the Internet using both passive tracing and active probing for six days. We provide the first measurements of link-level degradations and case studies of intra-ISP and inter-ISP peering links from a multimedia standpoint. Our study offers surprising insights into intradomain traffic engineering, peering link loading, BGP, and the inefficiencies of using autonomous system (AS)-path lengths as a routing metric. Though our results indicate that Internet routing policies are not optimized for delivering high-perceptual-quality streaming services, we argue that alternative strategies such as overlay networks can help meet QoE demands over the Internet. Streaming services apart, our Internet measurement results can be used as an input to a variety of research problems. Index Terms-Digital video broadcasting, Internet, quality of experience (QoE). I. INTRODUCTION A N ESTIMATED $4.2 billion from revenue generation is predicted for Internet video by 2011, with an annual growth rate of 36% [6], [10]. About 30% of the Internet traffic already carries multimedia content. As customers spend more and more time watching videos online, they are increasingly becoming unsatisfied by low-bit-rate videos available on YouTube and are embracing high-definition (HD) and switched-digital (SD) streaming services. To attract customers and prevent subscriber churn, network service providers need to deliver streaming content with high perceptual quality, or quality of Manuscript

Netflix is the leading provider of on-demand Internet video streaming in the US and Canada, accounting for 29.7% of the peak downstream traffic in US. Understanding the Netflix architecture and its performance can shed light on how to best optimize its design as well as on the design of similar on-demand streaming services. In this paper, we perform a measurement study of Netflix to uncover its architecture and service strategy. We find that Netflix employs a blend of data centers and Content Delivery Networks (CDNs) for content distribution. We also perform active measurements of the three CDNs employed by Netflix to quantify the video delivery bandwidth available to users across the US. Finally, as improvements to Netflix's current CDN assignment strategy, we propose a measurement-based adaptive CDN selection strategy and a multiple-CDN-based video delivery strategy, and demonstrate their potentials in significantly increasing user's average bandwidth.

The survey conducted recently reckons that by the year 2010 more than 10 billion wireless sensors have been deployed for various applications, as diverse as environmental monitoring, agricultural monitoring, machine health monitoring, surveillance, and medical monitoring. These networks connect the physical world with the digital world and provide us a better understanding of controlling our surroundings. Advances in wireless sensor network technology has lead to the availability of small and less cost sensor nodes which have the capability of sensing different types of physical and earthly conditions, data processing and wireless communications. Various sensing capabilities results in profusion of application area. Many routing, power management, and data dissemination protocols have been specifically designed for WSNs. We would like to highlight the strengths and performance issues of each routing technique and conclude the paper by bringing to your notice the possible future rese...

We develop scheduling strategies for carrying multimedia traffic over a polled multiple access wireless network with fading. We consider a slotted system with three classes of traffic (voice, streaming media and file transfers). A Markov model is used for the fading and also for modeling voice packet arrivals and streaming arrivals. The performance objectives are a loss probability for voice, mean network delay for streaming media, and time average throughput for file transfers. A central scheduler (e.g., the access point in a single cell IEEE 802.11 wireless local area network (WLAN)) is assumed to be able to keep track of all the available state information and make the scheduling decision in each slot (e.g., as would be the case for PCF mode operation of the IEEE 802.11 WLAN). The problem is modeled as a constrained Markov decision problem. By using constraint relaxations (a linear relaxation and Whittle type relaxations) an index based policy is obtained. For the file transfers the decision problem turns out to be one with partial state information. Numerical comparisons are provided with the performance obtained from some simple policies.

We propose a tracking-aware system that removes video components of low tracking interest and optimizes the quantization during compression of frequency coefficients, particularly those that most influence trackers, significantly reducing bitrate while maintaining comparable tracking accuracy. We utilize tracking accuracy as our compression criterion in lieu of mean squared error metrics. The process of optimizing quantization tables suitable for automated tracking can be executed online or offline. The online implementation initializes the encoding procedure for a specific scene, but introduces delay. On the other hand, the offline procedure produces globally optimum quantization tables where the optimization occurs for a collection of video sequences. Our proposed system is designed with low processing power and memory requirements in mind, and as such can be deployed on remote nodes. Using H.264/AVC video coding and a commonly used state-of-the-art tracker we show that while maintaining comparable tracking accuracy our system allows for over 50% bitrate savings on top of existing savings from previous work.

Distribution of streaming media content, including live news, music and videos, is becoming increasingly popular in today's Internet. Traditional client/server architectures are inefficient for distributing streaming media objects because of the high demands for system resources, especially server and network bandwidth, which severely limit the total number of simultaneous users the system can support. One proposal for improving the scalability of media distribution systems is the use of P2P overlay networks. Although a number of previous work has evaluated different aspects of P2P systems, mainly through simulation, there is a lack of a thorough quantitative analysis of the requirements for server and network resources (i.e., CPU, server and network bandwidth) in actual P2P systems, compared to traditional client/server systems. This work aims at filling this gap by providing experimental results that quantify the savings in server and network resources if a P2P approach is used for distributing live streaming media instead of the traditional client/server approach. Towards this goal, we build an experimental testbed, in a controlled environment, to evaluate actual systems with varying number of clients during periods when the distribution tree is static. A key component of this experimental testbed is a new efficient and scalable application called streaming servent, which can act both as a client and a server, forwarding packets to other clients. We also use simple analytical formulas to evaluate the scalability of our servent application. The experimental results quantify the intuitive better scalability of the P2P architecture. As an example, the total server bandwidth decreases from 15 Mbits/s to 9 Mbits/s (a 40% reduction) if a P2P architecture is used instead of a client/server architecture for live delivery of a given file to 24 clients.

Abstract: Community identification algorithms have been used to enhance the quality of the services perceived by its users. Although algorithms for community have a widespread use in the Web, their application to portals or specific subsets of the Web has not been much studied. In this paper, we propose a technique for local community identification that takes into account user access behavior derived from access logs of servers in the Web. The technique takes a departure from the existing community algorithms since it changes the ...

We propose a tracking-aware system that removes video components of low tracking interest and optimizes the quantization during compression of frequency coefficients, particularly those that most influence trackers, significantly reducing bitrate while maintaining comparable tracking accuracy. We utilize tracking accuracy as our compression criterion in lieu of mean squared error metrics. The process of optimizing quantization tables suitable for automated tracking can be executed online or offline. The online implementation initializes the encoding procedure for a specific scene, but introduces delay. On the other hand, the offline procedure produces globally optimum quantization tables where the optimization occurs for a collection of video sequences. Our proposed system is designed with low processing power and memory requirements in mind, and as such can be deployed on remote nodes. Using H.264/AVC video coding and a commonly used state-of-the-art tracker we show that while maintaining comparable tracking accuracy our system allows for over 50% bitrate savings on top of existing savings from previous work.

Cycle based vs Event Based simulation Cycle based Evaluates all logic gates during every simulation cycle Handles synchronous designs Suitable for circuits with high activity rate Performs unnecessary simulations (extra computation) Event based Evaluates only logic gates with a change on their inputs Handles both synchronous and asynchronous designs Suitable for circuits with low activity rate Requires a centralized scheduler that may cause large amount of overhead Maintaining queue for the list of events is challenging SIMD simulation

With the standardization of SVC, the scalable extension of H.264/AVC, layered peer-to-peer streaming has attracted more and more attention as it offers adaptability to network fluctuations and heterogeneous end users. Although overlay construction is important for system performance, not much effort has been spent on unstructured overlay construction for layered peer-to-peer streaming. Related work concentrates on layered streaming algorithms, and assumes that a list of peers for data exchange, called neighbors, is provided by traditional membership management protocols, e.g., SCAMP. Our previous studies have demonstrated that a random overlay is not good enough for layered peer-to-peer streaming. In this paper, we propose a new membership management protocol, based on peer sampling services. The protocol is quality-aware as it constructs the overlay so that (1) high capacity peers will be located at good positions in the overlay, e.g., close to the server, and (2) peers having similar capacity are likely to connect to each other. Both features are necessary to maximize bandwidth utilization of peers and to mitigate layer bottlenecks. With implementation of the protocol in PeerSim, we evaluate important graph properties of the overlay, constructed by the proposed protocol, to understand how it evolves during the streaming session with peer churn. Evaluation results show that the overlay is (1) scalable: it is stable with different sizes, from hundreds up to 10000 peers; and (2) robust: the good features are maintained or recovered fast under a high peer churn rate, and it only becomes disconnected when more than 86% of the peers are removed from the network.

Looking critically at the current landscape of home media, even as different forms of media converge, the living room remains the entertainment hub for most families. Televisions now offer far more types of entertainment than just TV shows, and the large breadth of content is changing the way stories are told at all levels. This paper explores the history of home media from broadcast to cable, profiling the successes and failures in the current landscape, including streaming and video games. This is approached from a critical view of the companies controlling production and access to content, exemplifying an understanding of the need to think ethically when creating messages. By looking into the origins of the medium, it evaluates media in historical and global context. Research includes studying popular examples from many genres and target demographics from all of the largest distribution platforms over key time periods. It also involves a content analysis research study of the pickups and cancellations for over-the-top streaming platforms with over 30 million subscribers to look for patterns in how modern networks differ from traditional broadcast media, and examines key case studies where viewership data is available to do so and compare the budgets of the biggest streaming shows to those of broadcast television. Finally, it makes educated predictions for what the postconvergence future might entail.

Video services are being adopted widely in both mobile and fixed networks. For their successful deployment, the content providers are increasingly becoming interested in evaluating the performance of such traffic from the final users' perspective, that is, their Quality of Experience (QoE). For this purpose, subjective quality assessment methods are costly and can not be used in real time. Therefore, automatic estimation of QoE is highly desired. In this paper, we propose a noreference QoE monitoring module for adaptive HTTP streaming using TCP and the H.264 video codec. HTTP streaming using TCP is the popular choice of many web based and IPTV applications due to the intrinsic advantages of the protocol. Moreover, these applications do not suffer from video data loss due to the reliable nature of the transport layer. However, there can be playout interruptions and if adaptive bitrate video streaming is used then the quality of video can vary due to lossy compression. Our QoE estimation module, based on Random Neural Networks, models the impact of both factors. The results presented in this paper show that our model accurately captures the relation between them and QoE.

ISPs and enterprises usually overprovision their networks as a means of supporting QoS. In spite of that, the service quality of QoS-sensitive applications such as VoIP, video conferencing and streaming media may not be up to expectations. We believe this is due to sporadic but non-negligible packet losses due to traffic bursts even in underutilized links. Our earlier work attempted to detect and analyze packet losses in underutilized links in an enterprise network environment using SNMP. This paper presents an extension of our earlier work by attempting to detect and analyze packet loss at finer granularity of time scale. We have developed a passive traffic capturing system, which can provide smaller time scale analysis of packet loss. We have analyzed the data and identified parts that are representative of packet loss across various time scales: 10 milliseconds, one second, 10 seconds and one minute. Analysis reveals that packet losses on underutilized link do occur due to bursty traffic packets in a small time scale. We also present analysis of other traffic properties such as packet size distribution and flows for the packet loss.

This paper presents the first 4Kp30 end-to-end video streaming demonstration based on the upcoming Scalable High efficiency Video Coding (SHVC) standard. The optimized and parallel SHVC decoder is used under the GPAC player to decode and display in real time the received SHVC layers. The SHVC reference software model (SHM) is used to encode the 4K original video in two spatial scalability layers: the base layer at 1080p resolution and the enhancement layer at 2160p resolution. The SHVC bitstream is encapsulated with the GPAC multimedia library into MP4 file format. The GPAC player at the server side broadcasts the MP4 content in MPEG-2 TS. At the client side, the GPAC player receives the SHVC video packets which are decoded by the SHVC decoder and then rendered in real time by the player. The GPAC player provides an interactive interface enabling to switch between displaying the base and the enhancement layers.

In retail stores, cashier non-compliance activities at the Point of Sale (POS) are one of the prevalent sources of retail loss. In this paper, we propose a novel approach to reliably rank the list of detected non-compliance activities of a given retail surveillance system, thereby provide a means of significantly reducing the false alarms and improving the precision in non-compliance detection. Our approach represents each detected non-compliance activity using multimodal features coming from video data, transaction logs (TLog) data and intermediate results of the video analytics. We then learn a binary classifier that successfully separate true positives and false positives in a labeled training set. A confidence score for each detection can then be computed using the decision value of the trained classifier, and a ranked list of detections can be formed based on this score. The benefit from having this ranked list is twofold. First, a large number of false alarms can be avoided by simply keeping the top part of the list and discarding the rest. Second, a trade off between precision and recall can easily be performed by sliding the discarding threshold along this ranked list. Experimental results on a large scale dataset captured from real stores demonstrate that our approach achieves better precision than a state-of-the-art system at the same recall. Our approach can also reach an operating point that exceeds the retailers' expectation in terms of precision, while retaining an acceptable recall of more than 60%.

Video streaming is one of the most important applications of Internet Protocol (IP) networks nowadays. The growth in demand for video streaming services with higher resolution and quality pushes the development of techniques to deliver the best possible quality to users. The Stream Control Transmission Protocol (SCTP), with its ability to support multi-homing systems, has been proposed as an alternative to the transport layer protocols in use. In this article, we propose a new handover strategy using partial reliability extension of SCTP and Moving Average Convergence Divergence (MACD). This strategy is designed to preserve the quality of video streaming over IP networks. The proposed method is based on forecasting the end-to-end delay and switching the primary path before an increase in delay impairs the quality of the video. The performance of the proposed method is evaluated through computer simulations which considers handover between ADSL, WiFi, and 3G networks. The results show that the handover scheme provides better quality of experience for low-and medium-motion videos and no worse performance for high-motion videos.

In this paper, error resilience is achieved by adaptive, application-layer rateless channel coding, which is used to protect H.264/Advanced Video Coding (AVC) codec data-partitioned videos. A packetization strategy is an effective tool to control error rates and, in the paper, source-coded data partitioning serves to allocate smaller packets to more important compressed video data. The scheme for doing this is applied to real-time streaming across a broadband wireless link. The advantages of rateless code rate adaptivity are then demonstrated in the paper. Because the data partitions of a video slice are each assigned to different network packets, in congestion-prone wireless networks the increased number of packets per slice and their size disparity may increase the packet loss rate from buffer overflows. As a form of congestion resilience, this paper recommends packet-size dependent scheduling as a relatively simple way of alleviating the buffer-overflow problem arising from data-partitioned packets. The paper also contributes an analysis of data partitioning and packet sizes as a prelude to considering scheduling regimes. The combination of adaptive channel coding and prioritized packetization for error resilience with packet-size dependent packet scheduling results in a robust streaming scheme specialized for broadband wireless and real-time streaming applications such as video conferencing, video telephony, and telemedicine.

Free-viewpoint television (FTV) is a visionary application that provides immersive experience to the audience with the freedom of changing viewpoint during the video playout. However, live broadcasting and user interaction do not coexist in existing FTV systems. In this paper, we propose F.Live, a framework of FTV content dissemination that supports userinitiated viewpoint changing for live broadcasting. Simulation result of a large-scale experiment, based on camera array settings of existing Nagoya systems and EyeVision System, shows that F.Live is capable of supporting 100,000 concurrent audiences with free-viewpoint low user interaction latency and feasible bandwidth requirements.

Existing media playout scheduling (MPS) schemes usually focus on selecting and scheduling packets according to optimized Internet media metrics, which are only partially relevant to the subjective human perception in the interactive system. The MPS design challenges are twofold. First, human preferences are concurrently dominated by multiple quality attributes of the streaming media whose perceptual tradeoffs were not well understood, so they were not used as an integral part of an efficient MPS design. Second, people's perceptions can be impacted by the flicker effect caused by Internet dynamics and the resulting MPS adaptations. In this paper, we propose a new and adaptive perception-based MPS scheme to deliver highquality real-time interactive multimedia. We first investigate the perceptual tradeoffs among the multi-modal bundle streaming qualities in a real Internet environment. We then present our MPS design that finds the bundle quality tradeoffs, while minimizing flicker degradations. Evaluation results show the performance of our MPS scheme.

In the context of communication networks with no QoS guarantees, we describe the problem of streaming layer encoded multimedia presentation data as a two-phase decision problem. We propose a measure of presentation quality called the refined maxmin metric that yields a computationally inexpensive algorithm for determining an optimal transmission policy needed within the second decision making phase. We compare this metric with a more traditional one, called the total quality metric, which yields an optimization problem that can be solved with dynamic programming. Using a slide show presentation with a randomly generated sequence of layer-encoded JPEG images, we compare the refined max-min criterion with the total quality criterion.

A transient spark (TS) pulsed discharge of streamerto-spark transition type is studied using fast iCCD camera and fast photomultiplier tube. Emission profiles and images of single TS pulse at different frequencies help to understand the influence of repetition frequency on the streamer-to-spark transition process.

The main topic of this paper is the quality of the WWW service evaluation. The authors present well-known measurement methods, and present the new “Power” metric for quality, advocating it as a method of assessing the quality of such service. This metric is based on the most important network parameters that affect any assessment of the WWW service, i.e. Web page opening time and download data transfer rate. The new method is easy to implement, fast in operation, and provides stable and repeatable result