Evaluation of unidirectional background push content download services for the delivery of television programs

Este trabajo de tesis presenta los servicios de descarga de contenido en modo push como un mecanismo eficiente para el envío de contenido de televisión pre-producido sobre redes de difusión. Hoy en día, los operadores de red dedican una cantidad considerable de recursos de red a la entrega en vivo de contenido televisivo, tanto sobre redes de difusión como sobre conexiones unidireccionales. Esta oferta de servicios responde únicamente a requisitos comerciales: disponer de los contenidos televisivos en cualquier momento y lugar. Sin embargo, desde un punto de vista estrictamente académico, el envío en vivo es únicamente un requerimiento para el contenido en vivo, no para contenidos que ya han sido producidos con anterioridad a su emisión. Más aún, la difusión es solo eficiente cuando el contenido es suficientemente popular. Los servicios bajo estudio en esta tesis utilizan capacidad residual en redes de difusión para enviar contenido pre-producido para que se almacene en los equipos de usuario. La propuesta se justifica únicamente por su eficiencia. Por un lado, genera valor de recursos de red que no se aprovecharían de otra manera. Por otro lado, realiza la entrega de contenidos pre-producidos y populares de la manera más eficiente: sobre servicios de descarga de contenidos en difusión. Los resultados incluyen modelos para la popularidad y la duración de contenidos, valiosos para cualquier trabajo de investigación basados en la entrega de contenidos televisivos. Además, la tesis evalúa la capacidad residual disponible en redes de difusión, por medio de estudios empíricos. Después, estos resultados son utilizados en simulaciones que evalúan las prestaciones de los servicios propuestos en escenarios diferentes y para aplicaciones diferentes. La evaluación demuestra que este tipo de servicios son un recurso muy útil para la entrega de contenido televisivo.This thesis dissertation presents background push Content Download Services as an efficient mechanism to deliver pre-produced television content through existing broadcast networks. Nowadays, network operators dedicate a considerable amount of network resources to live streaming live, through both broadcast and unicast connections. This service offering responds solely to commercial requirements: Content must be available anytime and anywhere. However, from a strictly academic point of view, live streaming is only a requirement for live content and not for pre-produced content. Moreover, broadcasting is only efficient when the content is sufficiently popular. The services under study in this thesis use residual capacity in broadcast networks to push popular, pre-produced content to storage capacity in customer premises equipment. The proposal responds only to efficiency requirements. On one hand, it creates value from network resources otherwise unused. On the other hand, it delivers popular pre-produced content in the most efficient way: through broadcast download services. The results include models for the popularity and the duration of television content, valuable for any research work dealing with file-based delivery of television content. Later, the thesis evaluates the residual capacity available in broadcast networks through empirical studies. These results are used in simulations to evaluate the performance of background push content download services in different scenarios and for different applications. The evaluation proves that this kind of services can become a great asset for the delivery of television contentFraile Gil, F. (2013). Evaluation of unidirectional background push content download services for the delivery of television programs [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/31656TESI

MediaSync: Handbook on Multimedia Synchronization

This book provides an approachable overview of the most recent advances in the fascinating field of media synchronization (mediasync), gathering contributions from the most representative and influential experts. Understanding the challenges of this field in the current multi-sensory, multi-device, and multi-protocol world is not an easy task. The book revisits the foundations of mediasync, including theoretical frameworks and models, highlights ongoing research efforts, like hybrid broadband broadcast (HBB) delivery and users' perception modeling (i.e., Quality of Experience or QoE), and paves the way for the future (e.g., towards the deployment of multi-sensory and ultra-realistic experiences). Although many advances around mediasync have been devised and deployed, this area of research is getting renewed attention to overcome remaining challenges in the next-generation (heterogeneous and ubiquitous) media ecosystem. Given the significant advances in this research area, its current relevance and the multiple disciplines it involves, the availability of a reference book on mediasync becomes necessary. This book fills the gap in this context. In particular, it addresses key aspects and reviews the most relevant contributions within the mediasync research space, from different perspectives. Mediasync: Handbook on Multimedia Synchronization is the perfect companion for scholars and practitioners that want to acquire strong knowledge about this research area, and also approach the challenges behind ensuring the best mediated experiences, by providing the adequate synchronization between the media elements that constitute these experiences

HbbTV-compliant Platform for Hybrid Media Delivery and Synchronization on Single- and Multi-Device Scenarios

[EN] The combination of broadcast and broadband (hybrid) technologies for delivering TV related media contents can bring fascinating opportunities. It is motivated by the large amount and diversity of media contents, together with the ubiquity and multiple connectivity capabilities of modern consumption devices. This paper presents an end-to-end platform for the preparation, delivery, and synchronized consumption of related hybrid (broadcast/broadband) media contents on a single device and/or on multiple close-by devices (i.e., a multi-device scenario). It is compatible with the latest version of the Hybrid Broadcast Broadband TV (HbbTV) standard (version 2.0.1). Additionally, it provides adaptive and efficient solutions for key issues not specified in that standard, but that are necessary to successfully deploy hybrid and multidevice media services. Moreover, apart from MPEG-DASH and HTML5, which are the broadband technologies adopted by HbbTV, the platform also provides support for using HTTP Live Streaming and Real-time Transport Protocol and its companion RTP Control Protocol broadband technologies. The presented platform can provide support for many hybrid media services. In this paper, in order to evaluate it, the use case of multi-device and multi-view TV service has been selected. The results of both objective and subjective assessments have been very satisfactory, in terms of performance (stability, smooth playout, delays, and sync accuracy), usability of the platform, usefulness of its functionalities, and the awaken interest in these kinds of platforms.This work was supported in part by the "Fondo Europeo de Desarrollo Regional" and in part by the Spanish Ministry of Economy and Competitiveness through R&D&I Support Program under Grant TEC2013-45492-R.Boronat, F.; Marfil-Reguero, D.; Montagud, M.; Pastor Castillo, FJ. (2017). HbbTV-compliant Platform for Hybrid Media Delivery and Synchronization on Single- and Multi-Device Scenarios. IEEE Transactions on Broadcasting. 1-26. https://doi.org/10.1109/TBC.2017.2781124S12

Scalability of broadcast performance in wireless network-on-chip

Networks-on-Chip (NoCs) are currently the paradigm of choice to interconnect the cores of a chip multiprocessor. However, conventional NoCs may not suffice to fulfill the on-chip communication requirements of processors with hundreds or thousands of cores. The main reason is that the performance of such networks drops as the number of cores grows, especially in the presence of multicast and broadcast traffic. This not only limits the scalability of current multiprocessor architectures, but also sets a performance wall that prevents the development of architectures that generate moderate-to-high levels of multicast. In this paper, a Wireless Network-on-Chip (WNoC) where all cores share a single broadband channel is presented. Such design is conceived to provide low latency and ordered delivery for multicast/broadcast traffic, in an attempt to complement a wireline NoC that will transport the rest of communication flows. To assess the feasibility of this approach, the network performance of WNoC is analyzed as a function of the system size and the channel capacity, and then compared to that of wireline NoCs with embedded multicast support. Based on this evaluation, preliminary results on the potential performance of the proposed hybrid scheme are provided, together with guidelines for the design of MAC protocols for WNoC.Peer ReviewedPostprint (published version

A Web-Based Collaborative Multimedia Presentation Document System

With the distributed and rapidly increasing volume of data and expeditious development of modern web browsers, web browsers have become a possible legitimate vehicle for remote interactive multimedia presentation and collaboration, especially for geographically dispersed teams. To our knowledge, although there are a large number of applications developed for these purposes, there are some drawbacks in prior work including the lack of interactive controls of presentation flows, general-purpose collaboration support on multimedia, and efficient and precise replay of presentations. To fill the research gaps in prior work, in this dissertation, we propose a web-based multimedia collaborative presentation document system, which models a presentation as media resources together with a stream of media events, attached to associated media objects. It represents presentation flows and collaboration actions in events, implements temporal and spatial scheduling on multimedia objects, and supports real-time interactive control of the predefined schedules. As all events are represented by simple messages with an object-prioritized approach, our platform can also support fine-grained precise replay of presentations. Hundreds of kilobytes could be enough to store the events in a collaborative presentation session for accurate replays, compared with hundreds of megabytes in screen recording tools with a pixel-based replay mechanism

Understanding Timelines within MPEG Standards

(c) 2016 IEEE. Personal use of this material is permitted. Permission from IEEE must be obtained for all other uses, in any current or future media, including reprinting/republishing this material for advertising or promotional purposes, creating new collective works, for resale or redistribution to servers or lists, or reuse of any copyrighted component of this work in other works.Nowadays, media content can be delivered via diverse broadband and broadcast technologies. Although these different technologies have somehow become rivals, their coordinated usage and convergence, by leveraging of their strengths and complementary characteristics, can bring many benefits to both operators and customers. For example, broadcast TV content can be augmented by on-demand broadband media content to provide enriched and personalized services, such as multi-view TV, audio language selection, and inclusion of real-time web feeds. A piece of evidence is the recent Hybrid Broadcast Broadband TV (HbbTV) standard, which aims at harmonizing the delivery and consumption of (hybrid) broadcast and broadband TV content. A key challenge in these emerging scenarios is the synchronization between the involved media streams, which can be originated by the same or different sources, and delivered via the same or different technologies. To enable synchronized (hybrid) media delivery services, some mechanisms providing timelines at the source side are necessary to accurately time align the involved media streams at the receiver-side. This paper provides a comprehensive review of how clock references (timing) and timestamps (time) are conveyed and interpreted when using the most widespread delivery technologies, such as DVB, RTP/RTCP and MPEG standards (e.g., MPEG-2, MPEG-4, MPEG-DASH, and MMT). It is particularly focused on the format, resolution, frequency, and the position within the bitstream of the fields conveying timing information, as well as on the involved components and packetization aspects. Finally, it provides a survey of proofs of concepts making use of these synchronization related mechanisms. This complete and thorough source of information can be very useful for scholars and practitioners interested in media services with synchronization demands.This work has been funded, partially, by the "Fondo Europeo de Desarrollo Regional" (FEDER) and the Spanish Ministry of Economy and Competitiveness, under its R&D&i Support Program in project with ref TEC2013-45492-R.Yuste, LB.; Boronat Segui, F.; Montagut Climent, MA.; Melvin, H. (2015). Understanding Timelines within MPEG Standards. Communications Surveys and Tutorials, IEEE Communications Society. 18(1):368-400. https://doi.org/10.1109/COMST.2015.2488483S36840018

Performance Evaluation of Scalable Multi-cell On-Demand Broadcast Protocols

As mobile data service becomes popular in today's mobile network, the data traffic burden irrevocably increases. LTE 4G, as the next-generation mobile technology, provides high data rates and improved spectral efficiency for data transmission. Currently in the mobile network, mobile data service solely relies on the point-to-point unicast transmission. In the ever-evolving 4G mobile network, mobile broadcast may serve as a supplemental means of pushing mobile data content from the data server to the mobile user devices. As part of the LTE 4G specifications, the mobile broadcast technology referred to as eMBMS is designed for supporting the mobile data service. From eMBMS, SFN broadcast transmission scheme allows data broadcasting to be synchronized in all cells of a defined core network area. LTE 4G also enables single-cell broadcast scheme in which data broadcasting is taking place independently in every cell. In this thesis, besides SFN or single-cell broadcast transmission, a hybrid broadcast transmission scheme in which SFN and single-cell broadcast transmission are used interchangeably in the same network based on the network conditions is proposed. For on-demand data service, the pull-based scheduling protocols from previous work are originally designed to work in a single-cell case scenario. With slight modifications, the batching/cbd protocol can be adapted for multi-cell data service. A new combined scheduling protocol, that is cyclic/cd,fft protocol, is devised as the second candidate for multi-cell data transmission scheduling. Based on the three broadcast transmission schemes and the two broadcast scheduling protocols, six mobile broadcast protocols are proposed. The mobile broadcast models, which correspond to the six mobile broadcast protocols, are evaluated by analysis and simulation experiment. By analysis, the cost equations are derived for calculating average server bandwidth, average client delay and maximum client delay of the mobile broadcast models. In the experiment, the input parameters of broadcast test models are assessed one at a time. The experimental results show that the hybrid broadcast transmission together with cyclic/cd,fft protocol would provide the best server bandwidth performance and the SFN broadcast transmission together with batching/cbd protocol provides the best average delay performance

Video-on-Demand over Internet: a survey of existing systems and solutions

Video-on-Demand is a service where movies are delivered to distributed users with low delay and free interactivity. The traditional client/server architecture experiences scalability issues to provide video streaming services, so there have been many proposals of systems, mostly based on a peer-to-peer or on a hybrid server/peer-to-peer solution, to solve this issue. This work presents a survey of the currently existing or proposed systems and solutions, based upon a subset of representative systems, and defines selection criteria allowing to classify these systems. These criteria are based on common questions such as, for example, is it video-on-demand or live streaming, is the architecture based on content delivery network, peer-to-peer or both, is the delivery overlay tree-based or mesh-based, is the system push-based or pull-based, single-stream or multi-streams, does it use data coding, and how do the clients choose their peers. Representative systems are briefly described to give a summarized overview of the proposed solutions, and four ones are analyzed in details. Finally, it is attempted to evaluate the most promising solutions for future experiments. Résumé La vidéo à la demande est un service où des films sont fournis à distance aux utilisateurs avec u

Synchronizing sound from different devices over a TCP network

Nowadays, we can send audio on the Internet for multiples uses like telephony, broadcast audio or teleconferencing. The issue comes when you need to synchronize the sound from different sources because the network where we are going to work could lose packets and introduce delay in the delivery. This can also come because the sound cards could be work in different speeds. In this project, we will work with two computers emitting sound (one will simulate the left channel (mono) of a stereo signal, and the other the right channel) and connected with a third computer by a TCP network. The last computer must get the sound from both computers and reproduce it in a speaker properly (without delay). So, basically, the main goal of the project is to synchronize multi-track sound over a network. TCP networks introduce latency into data transfers. Streaming audio suffers from two problems: a delay and an offset between the channels. This project explores the causes of latency, investigates the affect of the inter-channel offset and proposes a solution to synchronize the received channels. In conclusion, a good synchronization of the sound is required in a time when several audio applications are being developed. When two devices are ready to send audio over a network, this multi-track sound will arrive at the third computer with an offset giving a negative effect to the listener. This project has dealt with this offset achieving a good synchronization of the multitrack sound getting a good effect on the listener. This was achieved thanks to the division of the project into several steps having constantly a good vision of the problem, a good scalability and having controlled the latency at all times. As we can see in the chapter 4 of the project, a lack of synchronization over c. 100μs is audible to the listener. RESUMEN. A día de hoy, podemos transmitir audio a través de Internet por varios motivos como pueden ser: una llamada telefónica, una emisión de audio o una teleconferencia. El problema viene cuando necesitas sincronizar ese sonido producido por los diferentes orígenes ya que la red a la que nos vamos a conectar puede perder los paquetes y/o introducir un retardo en las entregas de los mismos. Así mismo, estos retardos también pueden venir producidos por las diferentes velocidades a las que trabajan las tarjetas de sonido de cada dispositivo. En este proyecto, se ha trabajado con dos ordenadores emitiendo sonido de manera intermitente (uno se encargará de simular el canal izquierdo (mono) de la señal estéreo emitida, y el otro del canal derecho), estando conectados a través de una red TCP a un tercer ordenador, el cual debe recibir el sonido y reproducirlo en unos altavoces adecuadamente y sin retardo (deberá juntar los dos canales y reproducirlo como si de estéreo de tratara). Así, el objetivo principal de este proyecto es el de encontrar la manera de sincronizar el sonido producido por los dos ordenadores y escuchar el conjunto en unos altavoces finales. Las redes TCP introducen latencia en la transferencia de datos. El streaming de audio emitido a través de una red de este tipo puede sufrir dos grandes contratiempos: retardo y offset, los dos existentes en las comunicaciones entre ambos canales. Este proyecto se centra en las causas de ese retardo, investiga el efecto que provoca el offset entre ambos canales y propone una solución para sincronizar los canales en el dispositivo receptor. Para terminar, una buena sincronización del sonido es requerida en una época donde las aplicaciones de audio se están desarrollando continuamente. Cuando los dos dispositivos estén preparados para enviar audio a través de la red, la señal de sonido multi-canal llegará al tercer ordenador con un offset añadido, por lo que resultará en una mala experiencia en la escucha final. En este proyecto se ha tenido que lidiar con ese offset mencionado anteriormente y se ha conseguido una buena sincronización del sonido multi-canal obteniendo un buen efecto en la escucha final. Esto ha sido posible gracias a una división del proyecto en diversas etapas que proporcionaban la facilidad de poder solucionar los errores en cada paso dando una importante visión del problema y teniendo controlada la latencia en todo momento. Como se puede ver en el capítulo 4 del proyecto, la falta de sincronización sobre una diferencia de 100μs entre dos canales (offset) empieza a ser audible en la escucha final