781 resultados para Video streaming
Energy-Aware Rate and Description Allocation Optimized Video Streaming for Mobile D2D Communications
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The proliferation problem of video streaming applications and mobile devices has prompted wireless network operators to put more efforts into improving quality of experience (QoE) while saving resources that are needed for high transmission rate and large size of video streaming. To deal with this problem, we propose an energy-aware rate and description allocation optimization method for video streaming in cellular network assisted device-to-device (D2D) communications. In particular, we allocate the optimal bit rate to each layer of video segments and packetize the segments into multiple descriptions with embedded forward error correction (FEC) for realtime streaming without retransmission. Simultaneously, the optimal number of descriptions is allocated to each D2D helper for transmission. The two allocation processes are done according to the access rate of segments, channel state information (CSI) of D2D requester, and remaining energy of helpers, to gain the highest optimization performance. Simulation results demonstrate that our proposed method (named OPT) significantly enhances the performance of video streaming in terms of high QoE and energy saving.
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This paper presents a new rate-control algorithm for live video streaming over wireless IP networks, which is based on selective frame discarding. In the proposed mechanism excess 'P' frames are dropped from the output queue at the sender using a congestion estimate based on packet loss statistics obtained from RTCP feedback and from the Data Link (DL) layer. The performance of the algorithm is evaluated through computer simulation. This paper also presents a characterisation of packet losses owing to transmission errors and congestion, which can help in choosing appropriate strategies to maximise the video quality experienced by the end user. Copyright © 2007 Inderscience Enterprises Ltd.
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The Internet as a video distribution medium has seen a tremendous growth in recent years. Currently, the transmission of major live events and TV channels over the Internet can easily reach hundreds or millions of users trying to receive the same content using very distinct receiver terminals, placing both scalability and heterogeneity challenges to content and network providers. In private and well-managed Internet Protocol (IP) networks these types of distributions are supported by specially designed architectures, complemented with IP Multicast protocols and Quality of Service (QoS) solutions. However, the Best-Effort and Unicast nature of the Internet requires the introduction of a new set of protocols and related architectures to support the distribution of these contents. In the field of file and non-real time content distributions this has led to the creation and development of several Peer-to-Peer protocols that have experienced great success in recent years. This chapter presents the current research and developments in Peer-to-Peer video streaming over the Internet. A special focus is made on peer protocols, associated architectures and video coding techniques. The authors also review and describe current Peer-to-Peer streaming solutions. © 2013, IGI Global.
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This work project aims to demonstrate how to design and develop an innovative concept of video streaming app. The project combines technology push and market pull theories into developing a product that is more suitable for the customer needs, with the particularity that there is no other way of seeing any place in the world, live and ondemand. An analysis on the bigger influencers in terms of design-thinking and new product development, as Tim Brown or Paul Trott, lead to a better understanding on how There App should evolve, keeping in mind the customer desires and technical features.
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Resumen en inglés
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There is a wide range of video services over complex transmission networks, and in some cases end users fail to receive an acceptable quality level. In this paper, the different factors that degrade users' quality of experience (QoE) in video streaming service that use TCP as transmission protocol are studied. In this specific service, impairment factors are: number of pauses, their duration and temporal location. In order to measure the effect that each temporal segment has in the overall video quality, subjective tests. Because current subjective test methodologies are not adequate to assess video streaming over TCP, some recommendations are provided here. At the application layer, a customized player is used to evaluate the behavior of player buffer, and consequently, the end user QoE. Video subjective test results demonstrate that there is a close correlation between application parameters and subjective scores. Based on this fact, a new metrics named VsQM is defined, which considers the importance of temporal location of pauses to assess the user QoE of video streaming service. A useful application scenario is also presented, in which the metrics proposed herein is used to improve video services(1).
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There is a wide range of telecommunications services that transmit voice, video and data through complex transmission networks and in some cases, the service has not an acceptable quality level for the end user. In this sense the study of methods for assessing video quality and voice have a very important role. This paper presents a classification scheme, based on different criteria, of the methods and metrics that are being studied in recent years. This paper presents how the video quality is affected by degradation in the transmission channel in two kinds of services: Digital TV (ISDB-TB) due the fading in the air interface and video streaming service on an IP network due packet loss. For Digital TV tests was set up a scenario where the digital TV transmitter is connected to an RF channel emulator, where are inserted different fading models and at the end, the videos are saved in a mobile device. The tests of streaming video were performed in an isolated scenario of IP network, which are scheduled several network conditions, resulting in different qualities of video reception. The video quality assessment is performed using objective assessment methods: PSNR, SSIM and VQM. The results show how the losses in the transmission channel affects the quality of end-user experience on both services studied.
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Il video streaming in peer-to-peer sta diventando sempre più popolare e utiliz- zato. Per tali applicazioni i criteri di misurazione delle performance sono: - startup delay: il tempo che intercorre tra la connessione e l’inizio della ripro- duzione dello stream (chiamato anche switching delay), - playback delay: il tempo che intercorre tra l’invio da parte della sorgente e la riproduzione dello stream da parte di un peer, - time lag: la differenza tra i playback delay di due diversi peer. Tuttavia, al giorno d’oggi i sistemi P2P per il video streaming sono interessati da considerevoli ritardi, sia nella fase di startup che in quella di riproduzione. Un recente studio su un famoso sistema P2P per lo streaming, ha mostrato che solitamente i ritardi variano tra i 10 e i 60 secondi. Gli autori hanno osservato anche che in alcuni casi i ritardi superano i 4 minuti! Si tratta quindi di gravi inconvenienti se si vuole assistere a eventi in diretta o se si vuole fruire di applicazioni interattive. Alcuni studi hanno mostrato che questi ritardi sono la conseguenza della natura non strutturata di molti sistemi P2P. Ogni stream viene suddiviso in blocchi che vengono scambiati tra i peer. A causa della diffusione non strutturata del contenuto, i peer devono continuamente scambiare informazioni con i loro vicini prima di poter inoltrare i blocchi ricevuti. Queste soluzioni sono estremamente re- sistenti ai cambiamenti della rete, ma comportano una perdita notevole in termini di prestazioni, rendendo complicato raggiungere l’obiettivo di un broadcast in realtime. In questo progetto abbiamo lavorato su un sistema P2P strutturato per il video streaming che ha mostrato di poter offrire ottimi risultati con ritardi molto vicini a quelli ottimali. In un sistema P2P strutturato ogni peer conosce esattamente quale blocchi inviare e a quali peer. Siccome il numero di peer che compongono il sistema potrebbe essere elevato, ogni peer dovrebbe operare possedendo solo una conoscenza limitata dello stato del sistema. Inoltre il sistema è in grado di gestire arrivi e partenze, anche raggruppati, richiedendo una riorganizzazione limitata della struttura. Infine, in questo progetto abbiamo progettato e implementato una soluzione personalizzata per rilevare e sostituire i peer non più in grado di cooperare. Anche per questo aspetto, l’obiettivo è stato quello di minimizzare il numero di informazioni scambiate tra peer.
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In free viewpoint applications, the images are captured by an array of cameras that acquire a scene of interest from different perspectives. Any intermediate viewpoint not included in the camera array can be virtually synthesized by the decoder, at a quality that depends on the distance between the virtual view and the camera views available at decoder. Hence, it is beneficial for any user to receive camera views that are close to each other for synthesis. This is however not always feasible in bandwidth-limited overlay networks, where every node may ask for different camera views. In this work, we propose an optimized delivery strategy for free viewpoint streaming over overlay networks. We introduce the concept of layered quality-of-experience (QoE), which describes the level of interactivity offered to clients. Based on these levels of QoE, camera views are organized into layered subsets. These subsets are then delivered to clients through a prioritized network coding streaming scheme, which accommodates for the network and clients heterogeneity and effectively exploit the resources of the overlay network. Simulation results show that, in a scenario with limited bandwidth or channel reliability, the proposed method outperforms baseline network coding approaches, where the different levels of QoE are not taken into account in the delivery strategy optimization.
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A video-aware unequal loss protection (ULP) system for protecting RTP video streaming in bursty packet loss networks is proposed. Just considering the relevance of the frame, the state of the channel and the bitrate constraints of the protection bitstream, our algorithm selects in real time the most suitable frames to be protected through forward error correction (FEC) techniques. It benefits from a wise RTP encapsulation that allows working at a frame level without requiring any further process than that of parsing RTP headers, so it is perfectly suitable to be included in commercial transmitters. The simulation results show how our proposed ULP technique outperforms non-smart schemes.