一种MPEG1/MPEG2视频流的自适应播放算法

提出了一种MPEG1/MPEG2视频流的自适应播放算法。算法根据播放缓冲区的占用情况来调整视频帧的播放持续时间，并在播放缓冲区上溢时判断到达视频帧的类型，以决定是暂时存储还是丢弃，使得不会造成帧内编码帧和前向预测编码帧的丢失，从而保证视频流平滑地播放。实验结果表明，在播放不连续性和播放失真上新算法都优于Yuang算法，并实现了视频流的平滑播放。     

支持IPTV高清码流连续平滑播放的多级缓冲区设计与实现

把流媒体系统作为一个复杂的控制系统进行研究,给出支持多级反馈控制机制的系统框图,重点分析媒体播放器从流接收到播放的全过程.同时,为了提高用户体验质量,实现音视频流在接收端的连续平滑播放,设计了多级缓冲机制以减少IP网络抖动和丢包产生的影响,实现了根据接收端的解码能力,自适应地选择解码层次和调整帧播放速率.最终采用多线程技术实现了MPEG-2 TS复用H.264高清视频流的解码播放,并给出试验结果.     

基于多步下溢概率估计的自适应媒体播放算法

自适应播放(AMP)是一种通过动态调整播放速率减少这种中断的技术。多数AMP算法都基于缓存的满溢度或者其变化调整播放速率,其难点在于如何选择合适的调整门限值。本文算法,利用滑动窗口估计到达帧率的经验分布,再利用统计计算方法估计多步缓存下溢概率,然后根据一步和多步下溢概率调整播放速率。仿真实验证明在一般的信道假设和多种片源的测试中,本算法较其他同类算法有更好的表现。     

基于状态机的视频码率自适应算法

由于链路带宽存在随机性,已有的基于超文本传输协议的动态自适应流媒体传输技术（DASH）的码率自适应算法不能很好解决播放流畅性和视频质量之间的矛盾。为解决该问题,提出一种基于状态机的DASH（SDASH）算法,将码率切换过程用状态机进行分析与控制。首先充分考虑客户端观看体验质量（QoE）的影响因素,对影响因素进行数值分析,并设定6个码率等级状态;然后将视频码率与影响因素的数值变化之间的联系作为状态转移条件;最后在保证播放缓存和码率偏移率处于一定阈值的条件下将视频码率切换至视频质量和播放流畅性整体性能相对最佳的码率等级上。实验结果表明,该算法与基于模糊逻辑控制的码率自适应算法相比能够提高客户端请求视频的平均码率,且尽量避免出现码率骤降等情况,从而较好地平衡播放流畅性和视频质量之间的关系,提升了视频观看过程的体验质量。     

基于流量自相似性的网络队列管理算法

网络流量的自相似性会导致数据突发状态持续,传统队列管理算法无法对网络流量突发状态进行预测,从而影响网络端到端时延、丢包率和吞吐性能。针对该问题,提出一种基于网络流量预测的主动队列管理算法P-ARED。基于网络流量的均值和方差给出网络流量等级的概念,讨论网络流量等级转移概率与Hurst参数之间的关系,提出基于贝叶斯估计思想的网络流量等级预测方法。在此基础上,在对自相似网络流量环境下的平均队列长度、缓存队列长度最小阈值等参数优化设置的基础上,基于Hurst参数和自相似流量等级预测结果,重新设计ARED算法中分组丢弃概率的计算方法,以提高缓存队列长度的稳定性。仿真结果表明,P-ARED算法与对比的主动队列管理算法相比,降低了网络端到端时延和丢包率,提高了端到端吞吐性能,其中平均吞吐量最高提升7.63%,平均时延最多降低17.52%。     

一种基于优先级区分的视频流传输控制算法

视频流传输控制对于提高网络服务质量具有重要意义.分析了MPEG视频流编码和传输特性,通过引入流间优先级和流内优先级机制,提出了一种基于优先级区分的视频流传输控制算法(PDVT),对不同类型的帧标记上不同的优先级,并进行优先级区分的调度.采用流间优先级机制区分MPEG流和其它网络流量,采用流内优先级机制区分MPEG视频流中不同类型帧的优先级.仿真实验表明,提出的PDVT算法能够向不同优先级流提供区分服务,大大提高了视频流传输质量,从而验证了算法的有效性.     

酷狗一直读取硬盘

Q我已经将播放列表清空,没有播放任何歌曲,同时还取消了“播放时预告缓存下一首歌曲”选项,但是酷狗只要在运行,硬盘灯就会不停地闪亮,处于读盘状态。请问这是什么原因？     

面向网络流量的缓存替换算法比较与分析

缓存替换算法对优化网络处理应用的性能起到关键作用，但目前面向网络流量的缓存替换算法研究主要集中在算法设计和领域应用方面，较少有文献对现有的缓存替换算法在网络环境下的性能进行分析比较。对此，本文针对主要的6种缓存替换算法进行分析和比较。通过分析网络流量的新近度与频度特征，为基于最近最少使用(Least Recently Used, LRU)和最近最不常使用(Least Frequently Used, LFU)的缓存替换算法给出实际依据。对仿真环境和实际系统的实验结果表明，类LRU算法较LFU算法更适用于网络流量，而缓存空间较大时，随机替换算法较LRU算法更适用于多核环境。     

具有P2P功能的嵌入式流媒体播放器的设计

基于IP流媒体技术的嵌入式流媒体终端因其使用简单、便于携带而迅速发展,但传统的嵌入式流媒体终端在成本、性能和算法易升级性等方面存在诸多问题;以S3C6410微处理器为核心,综合考虑硬件的多媒体处理能力和软件的可升级性,提出一种层次化、模块化的设计方案;同时在终端引入P2P播放模式,使得近距离终端设备之间以P2P方式共享视频流,减轻服务器负载。详细讨论了流媒体终端缓存空间管理机制、流媒体数据下载速度调整策略和播放流程设计。通过实验验证了该方案的可行性。     

静态与动态结合的流媒体缓存替换算法研究

为了在基于对等网的流媒体点播系统中高效地利用节点缓存区来提高流媒体服务质量,提出了一种静态与动态结合的缓存替换算法.采用将节点缓存区分为3个区的方法,综合考虑了数据块备份量的均衡性和节点跳播操作的命中率,使得节目数据块在各对等节点间缓存得到全局优化,从而有效提高播放质量.仿真实验结果表明了该算法在启动延迟和服务器负载等方面的优越性.     

Smooth Control of Adaptive Media Playout for Video Streaming

Client-side data buffering is a common technique to deal with media playout interruptions of streaming video caused by network jitters and packet losses of best-effort networks. However, stronger playout interruption protection inevitably amounts to larger data buffering and results in more memory requirements and longer playout delay. Adaptive media playout (AMP), also a client-side technique, can reduce the buffer requirement and avoid buffer outage but at the expense of visual quality degradation because of the fluctuation of playout speed. In this paper, we propose a novel AMP scheme to keep the video playout as smooth as possible while adapting to the channel condition. The triggering of the playout control is based on buffer variation rather than buffer fullness. Experimental results show that our AMP scheme surpasses conventional schemes in unfriendly network conditions. Unlike previous schemes that are tuned for a specific range of packet loss and network instability, the proposed AMP scheme maintains consistent performance across a wide range of network conditions.     

Dynamic Video Playout Smoothing Method for Multimedia Applications

Multimedia applications including video data require the smoothing of video playout to prevent potential discontinuity. In this paper, we propose a dynamic video playout smoothing method, called the Video Smoother, which dynamically adopts various playout rates in an attempt to compensate for high delay variance of networks. Specifically, if the number of frames in the buffer exceeds a given threshold (TH), the Smoother employs a maximum playout rate. Otherwise, the Smoother uses proportionally reduced rates in an effort to eliminate playout pauses resulting from the emptiness of the playout buffer. To determine THs under various loads, we present an analytic model assuming the Interrupted Poisson Process (IPP) arrival. Based on the analytic results, we establish a paradigm of determining THs and playout rates for achieving different playout qualities under various loads of networks. Finally, to demonstrate the viability of the Video Smoother, we have implemented a prototyping system including a multimedia teleconferencing application and the Video Smoother performing as part of the transport layer. The prototyping results show that the Video Smoother achieves smooth playout incurring only unnoticeable delays.     

Content-Aware Playout and Packet Scheduling for Video Streaming Over Wireless Links

Media streaming over wireless links is a challenging problem due to both the unreliable, time-varying nature of the wireless channel and the stringent delivery requirements of media traffic. In this paper, we use joint control of packet scheduling at the transmitter and content-aware playout at the receiver, so as to maximize the quality of media streaming over a wireless link. Our contributions are twofold. First, we formulate and study the problem of joint scheduling and playout control in the framework of Markov decision processes. Second, we propose a novel content-aware adaptive playout control, that takes into account the content of a video sequence, and in particular the motion characteristics of different scenes. We find that the joint scheduling and playout control can significantly improve the quality of the received video, at the expense of only a small amount of playout slowdown. Furthermore, the content-aware adaptive playout places the slowdown preferentially in the low-motion scenes, where its perceived effect is lower.      

Resource-aware and quality-fair video-streaming using multiple adaptive TCP connections

In this paper, we present a resource-aware and quality-fair video content sharing system. When a video sharing server has insufficient uplink bandwidth and needs to serve multiple video content sharing services via streaming or downloading to other client peers using TCP transport, each service shares the limited uplink bandwidth equitably, due to the fair sharing characteristics inherent in TCP. However this bandwidth fair sharing cannot always guarantee quality fairness among the services, due to the specific requirements for video-streaming services, such as the playout rate and the size of the playout buffer. In our system, the server uses multiple TCP connections adaptively, depending on the anticipated status of each client playout buffer, to guarantee the bandwidth of each video-streaming session. By guaranteeing the quality of each video-streaming session, without the quality loss of other service sessions, the proposed system can successfully achieve service quality fairness. Simulation results show that our proposed algorithm can dramatically enhance the quality of each streaming session and thus provide service quality fairness among simultaneous multiple heterogeneous video-streaming services and content download services.     

An efficient playout smoothing mechanism for layered streaming in P2P networks

Layered video streaming in peer-to-peer (P2P) networks has drawn great interest, since it can not only accommodate large numbers of users, but also handle peer heterogeneity. However, there’s still a lack of comprehensive studies on chunk scheduling for the smooth playout of layered streams in P2P networks. In these situations, a playout smoothing mechanism can be used to ensure the uniform delivery of the layered stream. This can be achieved by reducing the quality changes that the stream undergoes when adapting to changing network conditions. This paper complements previous efforts in throughput maximization and delay minimization for P2P streaming by considering the consequences of playout smoothing on the scheduling mechanisms for stream layer acquisition. The two main problems to be considered when designing a playout smoothing mechanism for P2P streaming are the fluctuation in available bandwidth between peers and the unreliability of user-contributed resources—particularly peer churn. Since the consideration of these two factors in the selection and scheduling of stream layers is crucial to maintain smooth stream playout, the main objective of our smoothing mechanism becomes the determination of how many layers to request from which peers, and in which order. In this work, we propose a playout smoothing mechanism for layered P2P streaming. The proposed mechanism relies on a novel scheduling algorithm that enables each peer to select appropriate stream layers, along with appropriate peers to provide them. In addition to playout smoothing, the presented mechanism also makes efficient use of network resources and provides high system throughput. An evaluation of the performance of the mechanism demonstrates that the proposed mechanism provides a significant improvement in the received video quality in terms of lowering the number of layer changes and useless chunks while improving bandwidth utilization.     

Joint optimization of continuity and quality for streaming video

This paper aims to reduce the amount of prebuffering required to ensure a maximum video continuity in streaming. Current approaches do this by slowing the playout frame rate of the decoder, this is known as adaptive media playout (AMP). However, doing this introduces playout distortion to the viewers as the video is played slower than its natural playout rate. We approach this by proposing a frame rate control scheme that jointly adjusts the encoder frame generation rate of the encoder and the playout frame rate of the decoder. In addition to using AMP to improve video continuity, we also allow the encoder to increase the encoder frame generation rate. This means the encoder will be sending more frames to the decoder to quickly increase the number of frames available at the playback buffer, thus lowering the chance of buffer underflow which causes discontinuity in video playback. At the same time, the increase in the number of frames at the playback buffer may mean that the decoder does not need to use AMP to delay the playback, thus lowering the playback distortion. However, the increase in encoder frame generation rate comes at a price because frame quality will need to decrease in order to meet the constraint on available network bandwidth. This implies that the scheme needs to find the optimal trade-off between frame quality, playout distortion and video continuity. To do that, we characterize the frame rate control problem using Lyapunov optimization. We then systematically derive the optimization policies. We also show that these policies can be decoupled into separate encoder and decoder optimization policies, thus allowing for a distributed implementation. Simulation results show significant reductions in the prebuffering requirements over a scheme that perform no frame rate control and lower playout distortions compared to the AMP schemes, while exhibiting a modest drop in frame quality.     

Arrival process-controlled adaptive media playout with multiple thresholds for video streaming

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.     

Playout in slotted CBR networks: Single and multiple nodes

We consider playout of a constant bit-rate (CBR) traffic after one or several multiplexors in a network with a playout buffer. Probabilistic characteristics of the playout process are found, depending on the traffic characteristics and parameters of the buffer. We present conditions on the buffer parameters that guarantee no jitter (complete playout).