951 resultados para Video Communication
Resumo:
For smart applications, nodes in wireless multimedia sensor networks (MWSNs) have to take decisions based on sensed scalar physical measurements. A routing protocol must provide the multimedia delivery with quality level support and be energy-efficient for large-scale networks. With this goal in mind, this paper proposes a smart Multi-hop hierarchical routing protocol for Efficient VIdeo communication (MEVI). MEVI combines an opportunistic scheme to create clusters, a cross-layer solution to select routes based on network conditions, and a smart solution to trigger multimedia transmission according to sensed data. Simulations were conducted to show the benefits of MEVI compared with the well-known Low-Energy Adaptive Clustering Hierarchy (LEACH) protocol. This paper includes an analysis of the signaling overhead, energy-efficiency, and video quality.
Resumo:
We study state-based video communication where a client simultaneously informs the server about the presence status of various packets in its buffer. In sender-driven transmission, the client periodically sends to the server a single acknowledgement packet that provides information about all packets that have arrived at the client by the time the acknowledgment is sent. In receiver-driven streaming, the client periodically sends to the server a single request packet that comprises a transmission schedule for sending missing data to the client over a horizon of time. We develop a comprehensive optimization framework that enables computing packet transmission decisions that maximize the end-to-end video quality for the given bandwidth resources, in both prospective scenarios. The core step of the optimization comprises computing the probability that a single packet will be communicated in error as a function of the expected transmission redundancy (or cost) used to communicate the packet. Through comprehensive simulation experiments, we carefully examine the performance advances that our framework enables relative to state-of-the-art scheduling systems that employ regular acknowledgement or request packets. Consistent gains in video quality of up to 2B are demonstrated across a variety of content types. We show that there is a direct analogy between the error-cost efficiency of streaming a single packet and the overall rate-distortion performance of streaming the whole content. In the case of sender-driven transmission, we develop an effective modeling approach that accurately characterizes the end-to-end performance as a function of the packet loss rate on the backward channel and the source encoding characteristics.
Resumo:
Thesis (Ph.D.)--University of Washington, 2016-06
Resumo:
The format of grant applications should be updated to incorporate multimedia video. This would help researchers to convey complex topics to grant-review panels. If time-poor research panels cannot quickly grasp the scientific ideas presented in a paper application, other factors, such as author affiliations and track records, may disproportionately influence project rankings...
Resumo:
Witnesses often experience lengthy delays prior to being interviewed, during which their memories inevitably decay. Video-communication technology - favored by intergovernmental organizations for playing larger roles in judicial processes - might circumvent some of the resourcing problems that can exacerbate such delays. However, whereas video-mediation might facilitate expeditious interviewing, it might also harm rapport-building, make witnesses uncomfortable, and thereby undermine the quality and detail of their reports. Participants viewed a crime film and were interviewed either one day later via video-link, one day later face-to-face, or 1-2 weeks later face-to-face. Video-mediation neither influenced the detail or the accuracy of participants' reports, nor their ratings of the quality of the interviews. However, participants who underwent video-mediated interviews after a short delay gave more accurate, detailed reports than participants who waited longer to be interviewed face-to-face. This study provides initial empirical evidence that video-mediated communication (VMC) could facilitate the expeditious conduct of high-quality investigative interviews. © 2013 © 2013 Taylor & Francis.
Resumo:
讨论了Windows环境下利用软件编解码器实现视频压缩的方法和技巧,结合视频捕获和视频传输,以网络环境下机器人遥操作的实际应用为背景,给出了数字视频实时通信的编程实例。
Resumo:
High-speed networks, such as ATM networks, are expected to support diverse Quality of Service (QoS) constraints, including real-time QoS guarantees. Real-time QoS is required by many applications such as those that involve voice and video communication. To support such services, routing algorithms that allow applications to reserve the needed bandwidth over a Virtual Circuit (VC) have been proposed. Commonly, these bandwidth-reservation algorithms assign VCs to routes using the least-loaded concept, and thus result in balancing the load over the set of all candidate routes. In this paper, we show that for such reservation-based protocols|which allow for the exclusive use of a preset fraction of a resource's bandwidth for an extended period of time-load balancing is not desirable as it results in resource fragmentation, which adversely affects the likelihood of accepting new reservations. In particular, we show that load-balancing VC routing algorithms are not appropriate when the main objective of the routing protocol is to increase the probability of finding routes that satisfy incoming VC requests, as opposed to equalizing the bandwidth utilization along the various routes. We present an on-line VC routing scheme that is based on the concept of "load profiling", which allows a distribution of "available" bandwidth across a set of candidate routes to match the characteristics of incoming VC QoS requests. We show the effectiveness of our load-profiling approach when compared to traditional load-balancing and load-packing VC routing schemes.
Resumo:
Ce document présente les résultats d’une étude empirique sur l’utilisation de la vidéoconférence mobile selon le contexte de l’usager afin de proposer des lignes directrices pour la conception des interfaces des dispositifs de communication vidéo mobile. Grâce à un échange riche d’informations, ce type de communication peut amener un sentiment de présence fort, mais les interfaces actuelles manquent de flexibilité qui permettrait aux usagers d’être créatifs et d’avoir des échanges plus riches lors d’une vidéoconférence. Nous avons mené une recherche avec seize participants dans trois activités où leurs conversations, leurs réactions et leurs comportements ont été observés. Deux groupes de discussion ont aussi servi à identifier les habitudes développées à partir de leur utilisation régulière de la vidéoconférence. Les résultats suggèrent une différence importante entre l’utilisation de la caméra avant et la caméra arrière de l’appareil mobile, et la nécessité de fournir des outils qui offrent plus de contrôle sur l’échange dans la conversation. L’étude propose plusieurs lignes directrices de conception pour les interfaces de communication vidéo mobiles, concernant la construction du contexte mobile de l’utilisateur.
Resumo:
Los sistemas de videoconferencia y colaboración en tiempo real para múltiples usuarios permiten a sus usuarios comunicarse por medio de vídeo, audio y datos. Históricamente estos han sido sistemas caros de obtener y de mantener. El paso de las décadas ha limado estos problemas acercado el mundo de comunicación en tiempo real a un grupo mucho más amplio, llegando a usarse en diversos ámbitos como la educación o la medicina. En este sentido, el último gran salto evolutivo al que hemos asistido ha sido la transición de este tipo de aplicaciones hacia la Web. Varias tecnologías han permitido este viaje hacia el navegador. Las Aplicaciones Ricas de Internet (RIAs), que permiten crear aplicaciones Web interactivas huyendo del clásico esquema de petición y respuesta y llevando funcionalidades propias de las aplicaciones nativas a la Web. Por otro lado, la computación en la nube o Cloud Computing, con su modelo de pago por uso de recursos virtualizados, ha llevado a la creación de servicios que se adaptan mejor a la demanda, han habilitado este viaje hacia el navegador. No obstante, como cada cambio, este salto presenta una serie de retos para los sistemas de videoconferencia establecidos. Esta tesis doctoral propone un conjunto de arquitecturas, mecanismos y algoritmos para adaptar los sistemas de multiconferencia al entorno Web, teniendo en cuenta que este es accedido desde dispositivos diferentes y mediante redes de acceso variadas. Para ello se comienza por el estudio de los requisitos que debe cumplir un sistema de videoconferencia en la Web. Como resultado se diseña, implementa y desarrolla un servicio de videoconferencia que permite la colaboración avanzada entre múltiples usuarios mediante vídeo, audio y compartición de escritorio. Posteriormente, se plantea un sistema de comunicación entre una aplicación nativa y Web, proponiendo técnicas de adaptación entre los dos entornos que permiten la conversación de manera transparente para los usuarios. Estos sistemas permiten facilitar la transición hacia tecnologías Web. Como siguiente paso, se identificaron los principales problemas que existen para la comunicación multiusuario en dispositivos de tamaño reducido (teléfonos inteligentes) utilizando redes de acceso heterogéneas. Se propone un mecanismo, combinación de transcodificación y algoritmos de adaptación de calidad para superar estas limitaciones y permitir a los usuarios de este tipo de dispositivos participar en igualdad de condiciones. La aparición de WebRTC como tecnología disruptiva en este entorno, permitiendo nuevas posibilidades de comunicación en navegadores, motiva la segunda iteración de esta tesis. Aquí se presenta un nuevo esquema de adaptación a la demanda para servidores de videoconferencia diseñado para las necesidades del entorno Web y para aprovechar las características de Cloud Computing. Finalmente, esta tesis repasa las conclusiones obtenidas como fruto del trabajo llevado a cabo, reflejando la evolución de la videoconferencia Web desde sus inicios hasta nuestros días. ABSTRACT Multiuser Videoconferencing and real-time collaboration systems allow users to communicate using video, audio and data streams. These systems have been historically expensive to obtain and maintain. Over the last few decades, technological breakthroughs have mitigated those costs and popularized real time video communication, allowing its use in environments such as education or health. The last big evolutionary leap forward has been the transition of these types of applications towards theWeb. Several technologies have allowed this journey to theWeb browser. Firstly, Rich Internet Applications (RIAs) enable the creation of dynamic Web pages that defy the classical request-response interaction and provide an experience similar to their native counterparts. On the other hand, Cloud Computing brings the leasing of virtualized hardware resources in a pay-peruse model and, with it, better scalability in resource-demanding services. However, as with every change, this evolution imposes a set of challenges on existing videoconferencing solutions. This dissertation proposes a set of architectures, mechanisms and algorithms that aim to adapt multi-conferencing systems to the Web platform, taking into account the variety of devices and access networks that come with it. To this end, this thesis starts with a study concerning the requirements that must be met by new Web videoconferencing systems. The result of this study is the design, development and implementation of a new videoconferencing services that provides advanced collaboration to its user by providing video and audio communication as well as desktop sharing. After this, a new communication system between Web and native applications is presented. This system proposes adaptation mechanisms to bridge the two worlds providing a seamless integration transparent to users who can now access the powerful native application via an easy Web interface. The next step is to identify the main challenges posed by multi-conferencing on small devices (smartphones) with heterogeneous access networks. This dissertation proposes a mechanism that combines transcoding and adaptive quality algorithms to overcome those limitations. A second iteration in this dissertation is motivated by WebRTC. WebRTC appears as a disrupting technology by enabling new real-time communication possibilities in browsers. A new mechanism for flexible videoconferencing server scalability is presented. This mechanism aims to address the strong scalability requirements in the Web environment by taking advantage of Cloud Computing. Finally, the dissertation discusses the results obtained throughout the study, capturing the evolution of Web videoconferencing systems.
Resumo:
The Enred@te initiative, created by Red Cross, the Vodafone Foundation and the TECSOS Foundation, emerged as an evolution of a previous project that developed and piloted a video-communication solution with older adults, using a system installed in their own televisions. Following the success of this first initiative, it was decided to advance toward a more flexible, robust, easy-to-use and high-quality solution, producing a social network accessible through tablets. Older adults can use the network to video-communicate with other older adults and stay informed on various topics of interest. Additionally, a new innovation incorporates the participation of virtual volunteers, a part of the network that promotes its use in an inclusive and participative manner. This solution was also piloted in 2014 with positive results and work to turn it into a service that can reach older adults through the Red Cross is currently on-going.