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1.
The MPEG has recently Querydeveloped a new standard, MPEG media transport (MMT), for the next-generation hybrid media delivery service over IP networks considering the emerging convergence of digital broadcast and broadband services. On account of the heterogeneous characteristics of broadcast and broadband networks, MMT provides an efficient delivery timing model to enable inter-network synchronization, measure various kinds of transmission delays and jitters caused by the transmission delay, and re-adjust the timing relationship between the MMT packets to ensure synchronized playback. By exploiting the delivery timing model, it is possible to accurately estimate the round-trip time (RTT) experienced during MMT packet transmission. Based on the measured RTT, we propose an efficient delay-constrained automatic repeat request (ARQ) scheme, which is applicable to MMT packet-based real-time video streaming service over IP networks. In the proposed ARQ scheme, the receiver buffer fullness at the time of packet loss detection is used to compute the arrival deadline, which is the maximum allowed time for completing the requesting and retransmitting of the lost MMT packet. Simulation results demonstrate that the proposed delay-constrained ARQ scheme can not only provide reliable error recovery, but it also achieves significant bandwidth savings by reducing the number of wastefully retransmitted packets that arrive at the receiver side and exceed the allowed arrival deadline.  相似文献   

2.
Coolstreaming is a mesh based peer-to-peer (P2P) video streaming system in which single video stream is decomposed into multiple sub-streams. A client-peer node retrieves the sub-streams from multiple parent-peer nodes, combining them into the original video stream. Each client-peer node has two buffers, a synchronization buffer and a cache buffer, and arriving data blocks are synchronized at the synchronization buffer and then forwarded to the cache buffer. In this buffering system, data-block synchronization is important to guarantee high video quality. In this paper, we consider the effect of churn on the performance of data-block synchronization scheme with which data blocks are simultaneously forwarded just after all the data blocks composing a macro data block arrive at the synchronization buffer. It is assumed that data blocks belonging to a sub-stream arrive at the client-peer node according to an interrupted Poisson process. The synchronization buffer is modeled as a multiple-buffer queueing system with homogeneous interrupted Poisson processes, and the mean forwarding interval is derived. Numerical examples show that the average forwarding interval increases as parent-peer nodes leave more frequently.  相似文献   

3.
P2P video streaming networks are found as a scalable solution and an alternative for traditional client–server based video streaming over the Internet. One of the significant issues affecting the success of any P2P streaming network is cooperation between peers. Practical observations have proved the prevalence of free riders in P2P networks that degrade their performance. To solve this problem, using incentive mechanisms, which encourage peers to contribute more in the network, is necessary. In this paper, we designed and proposed a distributed and scalable incentive mechanism for mesh based P2P video streaming networks. In the proposed approach the contribution of the peers is measured and maintained in a distributed fashion. Furthermore, we proposed an incentive sending side scheduler in which peers are served based on their contribution in the network. Our simulation evaluations show the efficiency of the proposed approach in improving the overall perceived video quality by the non-free rider nodes and consequently in the whole network.  相似文献   

4.
The Journal of Supercomputing - We consider the problem of live video streaming in peer-to-peer networks consisting of selfish peers aiming at maximizing peers’ individual utilities. We use...  相似文献   

5.
Free-riding is one of the main challenges of Peer-to-Peer (P2P) streaming systems which results in reduction in video streaming quality. Therefore, providing an incentive mechanism for stimulating cooperation is one of the essential requirements to maintain video Quality of Experience (QoE) in such systems. Among the existing mechanisms, payment-based schemes are most suitable for streaming applications due to their low overhead. However, to date, no dynamic payment mechanism has been proposed which can take the stochastic dynamics of the video streaming ecosystem (e.g., the request arrival, demand submission, bandwidth availability, etc.) into account. In this paper, we propose a dynamic token-based payment mechanism in which each peer earns tokens by admitting other peers’ requests and spends tokens for submitting its demands to the others. This system allows the peers to dynamically adjust their income level in adaptation to changes in the system state. We propose a Constrained Markov Decision Process (CMDP) formulation in which the goal of each peer is to obtain a request admission policy which minimizes the expected cumulative cost of consumed bandwidth, while satisfying a long-term constraint on the Mean Opinion Score (MOS) of the users as the measure of QoE. The proposed admission policy is adaptive to the request arrival process, bandwidth state and the token bucket length of each peer. To make up for the lack of design-time knowledge of the system’s statistics, each individual peer is equipped with a model-free algorithm to learn its optimal admission policy over the course of real-time interaction with the system. Simulation results are presented to compare the performance of the proposed algorithm against baseline schemes such as: random, token-threshold, bandwidth-threshold and myopic algorithms.  相似文献   

6.

With the recent emergence of cloud computing, growing numbers of clients are using online cloud services through the Internet such as video streaming service. The rent costs of cloud service providers increase when the resource utilizations of the cloud-servers are not well. Therefore, resource allocation is a crucial problem for cloud data centers. The resource allocation problem is an NP-hard problem. This paper proposes a novel cloud resource allocation mechanism based on a winning strategy for a Nim game. This mechanism offers all clients an effective number of running cloud servers, and allocates cloud resources rapidly and effectively by using a pre-pairing approach. The proposed mechanism does not require searching for remaining resources of the running cloud server; hence, it can reduce the time taken to arrange resources. The experimental results show that the proposed mechanism can improve utilization of cloud servers and reduce the rent costs of the cloud service providers. The proposed mechanism can reach the utilization of cloud servers by as much as 99.96 %. The proposed mechanism is approximately 9 % more efficient than the market-based grid resource allocation algorithm, and 19 % more efficient than the modified best fit decreasing algorithm.

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7.

Based on the three rotational degrees (video in three dimensions, on the X, Y and Z axes) of freedom provided by VR, the viewer is free to control the viewing point and has six degrees of freedom (6DoF). When watching a sports game, the audience is no longer limited by the position of the camera, and can freely choose the viewing angle and position just like watching in the real world, which can greatly improve the immersion of viewing. However, the major barrier that prevents 6DoF video live from being industrialized lies in the extremely high computational complexity, of which multi-view depth estimation and Depth Image Based Rendering (DIBR) is difficult to realize. And existing devices do not have hardware interfaces that support multi-views coding technology. Therefore, we need new technologies for depth estimation and virtual view synthesis, and we need to use existing hardware coding/decoding interfaces to reduce power consumption. In this paper, we provide a 6DoF live video system, which includes multi-view depth estimation technique based on unsupervised learning, virtual viewpoint real-time rendering technology and 6DoF video coding. Experimental results demonstrate that our proposed acceleration method can speed up the original depth estimation algorithm by more than 34x, and can speed up the original DIBR algorithm by more than 168x. With our 6DoF video coding method, experimental results show that the bit rate achieves an average of 70%, 64%, 33%, 60% and 66% bitrate saving for AVC, HEVC, AV1, AVS3, VVC codec standard respectively.

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8.
无线网络动态的信道特性和带宽有限等特点,使得在无线环境下为流媒体应用提供QoS保证面临更大的挑战。提出一种用于无线实时流媒体传输的增强型自适应前向纠错控制策略,以提高接收方的播放质量。该策略采用跨层设计的方法,根据当前的网络状态,自适应地调整MPEG视频帧的发送速率,在视频源数据和冗余数据之间动态分配网络带宽。仿真结果表明,该策略能使接收方获得最大的可播放帧率,有效提高流媒体传输的可靠性和实时性。  相似文献   

9.
一种适用于无线网络的流媒体传输机制   总被引:4,自引:0,他引:4  
孙伟  温涛  郭权 《计算机应用》2009,29(1):12-15
为保证无线网络中多媒体数据的传输质量,提出了一种适用于无线网络的流媒体传输机制(WMTCC)。该机制通过发送探测报文区分网络拥塞丢包和链路误码随机丢包,准确判断网络的拥塞状况,实施发送速率调节,保证了流媒体服务质量(QoS)。由于准确区分出无线链路误码丢包,该机制在链路误码率较高时能维持较高的网络吞吐量。仿真实验结果显示在高误码率无线网络中,该机制可以获得更高的吞吐量和更大的拥塞窗口,并且发送速率的变化更加平滑。  相似文献   

10.
为了应对H.264可伸缩视频编码(SVC)应用中网络特性的波动,提出了一种预测播放中断与缓冲区溢出风险进行及早调节的自适应媒体播放(AMP)算法。该算法估算网络流量与视频图像组(GOP)结构中各帧长度用于风险预测,通过K步调节过程实现良好的调节平滑性与速度,并利用SVC的可伸缩性尽量减少溢出带来的质量损失。仿真结果表明,该算法在抑制播放中断、处理缓冲区溢出与抖动性能等方面,优于现行的平滑AMP与常规AMP算法。  相似文献   

11.
To improve the playout quality of video streaming services, an arrival process-controlled adaptive media playout (AMP) mechanism is designed in this study. The proposed AMP scheme sets three threshold values, denoted by P n , L and H, for the playout controller to start playback and dynamically adjust the playout rate based on the buffer fullness. In the preroll period, the playout can start only when the buffer fullness n is not less than the dynamic playback threshold P n ,?which is determined by the jitters of incoming video frames. In the playback period, if the buffer fullness is below L or over H,?the playout rate will slow down or speed up in a quadratic manner. Otherwise, the playback speed depends on the instantaneous frame arrival rate, which is estimated by the proposed arrival process tracking algorithm. We employ computer simulations to demonstrate the performance of the proposed AMP scheme, and compare it with several conventional AMP mechanisms. Numerical results show that our AMP design can shorten the playout delay and reduce both buffer underflow and overflow probabilities. In addition, our proposed AMP also outperforms traditional AMP schemes in terms of the variance of distortion of playout and the playout curve. Hence, the proposed arrival process-controlled AMP is really an outstanding design.  相似文献   

12.
流媒体视频存储服务器设计与研究   总被引:1,自引:0,他引:1  
详细分析了在视频监控系统中流媒体视频存储服务器的作用和功能,针对传统的流媒体定时长存储方式的不足提出了固定空间存储的新策略.该存储策略将总空间初始化为多个固定大小的数据文件,每个数据文件中存储的流媒体时长可能是不相同的.针对新存储策略下出现的1秒尾巴,事件尾巴等新问题,提出了解决方案.固定空间存储策略虽然增加了索引的复杂度,但是有效避免了磁盘碎片的产生,对于改善流媒体视频存储服务器性能具有重要作用.  相似文献   

13.
In this paper, we propose a real-time panoramic video streaming system with overlaid interface concept for social media. The proposed system provides real-time panorama images for smart displays such as smart TVs, smart phones, and tablet PCs. Panorama images are collected at sporting events via panorama cameras. Contents thus collected are sent to servers and then provided to smart devices via live sports video streams. Users select a panorama camera and choose their viewing angle and zooming factor for the selected panorama camera. The proposed system provides immersive and realistic views of live sporting events on users’ displays. Furthermore, an overlaid panoramic interface concept is proposed for immersive live baseball watching combined with tightly integrated social media experience.  相似文献   

14.
Multimedia Tools and Applications - Adaptive Media Playout (AMP) controls adapt playout rate to prevent buffer outage and to reduce delay in playout. Most AMP techniques use buffer fullness or its...  相似文献   

15.
Traditional Forward Error Correction (FEC) mechanisms can be divided into Packet level FEC (PFEC) mechanisms and Byte level FEC (BFEC) mechanisms. The PFEC mechanism of recovering from errors in a source packet requires an entire FEC redundant packet even though the error involves a few bit errors. The recovery capability of the BFEC mechanism is only half of the FEC redundancy. Accordingly, an adaptive Sub-Packet FEC (SPFEC) mechanism is proposed in this paper to improve the quality of video streaming data over wireless networks, simultaneously enhancing the recovery performance and reducing the end-to-end delay jitter. The SPFEC mechanism divides a packet into n sub-packets by means of the concept of a virtual packet. The SPFEC mechanism uses a checksum in each sub-packet to identify the position of the error sub-packet. Simulation experiments show the adaptive SPFEC mechanism achieves high recovery performance and low end-to-end delay jitter. The SPFEC mechanism outperforms traditional FEC mechanism in terms of packet loss rate and video Peak Signal-to-Noise Ratio (PSNR). SPFEC offers an alternative for improved efficiency video streaming that will be of interest to the designers of the next generation environments.  相似文献   

16.
Multimedia streaming allows consumers to view multimedia content anywhere. However, quality of service is a major concern amid heightened levels of network traffic caused by increasing user demand. Accordingly, media streaming technology is adopting a new paradigm: adaptive HTTP streaming (AHS). AHS is widely used for real-time streaming content delivery in the Internet environment. In streaming, selection of appropriate bitrate is crucial for adapting media rate to network variations and client processing capabilities while ensuring optimal service for the consumer. We evaluate a proposed client-driven three-level optimized rate adaptation algorithm for adaptive HTTP media streaming. In the first stage, the algorithm chooses a suitable starting bitrate close to the available channel capacity. Next, the algorithm monitors the client parameters in real time, precisely detecting network variations and choosing a likely available bit representation for the next download segment. The algorithm controls and minimizes the effects of buffer stalls and overflow resulting from the brief network variations occurring between consecutive segments. The proposed algorithm is implemented in Dynamic Adaptive Streaming over HTTP (DASH) player and its performance compared to that of commercially available Gstreamer-based HTTP Live Streaming (HLS) and DASH players which use conventional segment fetch time–based adaptation and throughput-based adaptation algorithms respectively. This evaluation uses a real-time cloud server client and test bed streaming setup. The resulting analysis shows that the client-driven three-level rate adaptation (TLRA) approach allows adaptive streaming clients to maximize use of end-to-end network capacity, delivering an ideal user experience by precisely predicting network variations and rapidly adapting to available bandwidth, minimizing rebuffering events and bitrate level changes.  相似文献   

17.
基于Directshow的流媒体视频混合及网络传输系统   总被引:3,自引:0,他引:3  
本文基于Directshow和Windows Socket技术,建立了一种流媒体视频混合及网络传输系统,实现了两路视频流媒体的混合,给出了混合视频流网络传输的解决方案。即在一台视频流媒体服务器、一台终端和一条低带宽线路的条件下同时传输两路/多路动态视频流,并且可以随时调整叠加视频的位置和Alpha值;具有客户端简单,节约带宽,传输效率高,传输信号准确清晰的优点。  相似文献   

18.
Urban traffic control systems have based their technological infrastructure both on advanced analog close-circuit television systems (CCTV) and point-to-point links, providing difficult-to-scale and very expensive systems. The main goal of an urban traffic monitoring system is to capture, send, play and distribute video information from the streets of a certain city to a management centre where it is processed by different services. Current digitalization process of video networks, and the research carried out in the field of streaming media, has led vendors to present proprietary hardware and software solutions resulting in a strong dependency by their customers. This work presents an open urban traffic control system based in commercial off-the-shelf (COTS) philosophy for hardware and software components, as well as open source and standardized protocols. The existence of open standards for video encoding and protocols for streaming media transmission over IP networks has led to the proposal of such a system. The proposed system is a suitable solution in terms of scalability, cost, interoperability and performance for traffic control systems. Furthermore, its architecture can be easily adapted to other video applications and tools like command and control, surveillance or security systems for military and civilian applications.  相似文献   

19.
Kalman滤波的自适应链路层FEC控制策略   总被引:2,自引:0,他引:2       下载免费PDF全文
无线网络动态的信道特性、高误码率和带宽有限等特点,使得在无线环境下为实时流媒体传输提供QoS保证面临更大的挑战。提出一种用于无线实时流媒体传输的自适应链路层FEC控制策略,以显著提高接收方的播放质量。该策略采用跨层设计的方法,基于Kalman滤波器预测当前的网络状态,考虑物理带宽限制和GOP可解码帧数的特性自适应地调整FEC参数N;另一方面,在应用层采用自适应FEC策略,在视频源数据和冗余数据之间动态分配网络带宽。数学分析和仿真验证均表明,该策略能使接收方获得最大的可播放帧率,有效地提高了流媒体传输的可靠性和实时性。  相似文献   

20.
Streaming media from the Internet is a successful application for end-users. With the upcoming success of mobile devices and home networking environments, cooperation among users will become more important in the future. To achieve such cooperation, explicit middleware standards have been defined. On the other hand, Internet conferencing applications do not handle collaborative streaming sessions with individual control for each user. We propose a new concept for cooperation exemplary for collaborative media streaming using IETF multimedia session control protocols together with a proxy architecture. This concept enables both synchronization among clients and flexible control to individual users.  相似文献   

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