基于PID控制的自适应媒体播放算法

针对视频播放中因缓冲区下溢带来的抖动问题,提出一种基于PID控制的自适应播放算法。算法结合PID控制和自适应播放算法,根据网络信道状态和缓冲区状态控制视频播放速率,并对播放速率的范围和相邻帧播放速率的突变进行控制,在减少缓冲区下溢概率的同时实现视频的平滑播放。该算法在减小缓冲区下溢概率、实现视频的平滑播放以及减少播放延迟方面均优于固定因子自适应播放算法。     

基于概率估计的自适应媒体播放算法

针对缓冲区下溢造成的视频播放抖动和中断问题,提出一种基于概率估计的自适应媒体播放算法。根据网络信道状态和估计的下溢概率和上溢概率控制视频帧的持续播放时间,适当控制播放速率的变化范围和变化量,减少缓冲区下溢概率和播放时延,实现视频平滑播放。仿真结果证明,该算法性能优于传统的自适应媒体播放算法。     

实时流媒体传输中客户终端系统缓冲区设计

网络时延抖动以及时钟偏移等问题,会对媒体流播放是否流畅产生重要影响。从客户终端的角度来讨论如何优化设置和管理缓冲区,以及平滑媒体流的播放,并提出了以SLOW-START为启动模型的动态缓冲控制算法。该算法可以有效地减小起始状态的播放延迟,并有效防止缓冲区上溢造成的播放跳跃以及缓冲区下溢造成的播放停顿。     

双重控制策略在流媒体连续播放中的应用

传统的发送速率控制和播放速率控制由于各自的局限性,很难克服网络时延扰动对流媒体连续播放的影响.为了改善播放系统的性能,将双重控制策略应用到流媒体播放中,在发送端发送速率控制采用了内模控制器以克服传输延迟的影响,在接收端播放速率控制采用了简单的PID控制器,两种控制共同调节缓冲区长度.实验表明该方法能够处理传输时延和网络扰动引起的失步对播放的影响,缓冲区长度能够稳定在合理的区间,有效地防止缓冲区下溢造成的播放停顿以及缓冲区上溢造成的播放跳跃.发送速率变化率较小,有利于避免网络拥塞的发生.尤其是在网络出现大的扰动时,与其他方法相比,控制效果更加理想,播放更加流畅.     

基于G/G/1排队系统的播放缓冲控制机制

提出基于下溢概率估计的AMP算法(简称AMP-UPE算法),以下溢概率作为播放速率调整的指示,同时考虑了当前缓存队列长度和缓存队列长度的变化。基于G/G/1排队系统提出下溢概率估计模型,并通过滑动窗口法估计帧到达时间间隔的均值和方差,从而实现下溢概率的在线估计。仿真实验结果表明,AMP-UPE算法性能优于AMP-RB算法和AMP-SC算法;且AMP-UPE算法的性能受下溢概率阈值影响,下溢概率阈值越小,播放中断的频率越低,帧间隔的短期标准差越大。     

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。     

一种考虑节点拥塞情况的DTN概率路由算法*

容迟网络是一种新型网络,其概率路由算法根据历史相遇频率对相遇概率进行计算与更新,通过相遇概率判断是否转发报文。当节点缓存受限时,在网络中采用概率路由算法使得节点很容易发生拥塞,对报文的传送产生影响。为了减小拥塞对概率路由算法的影响,提出了一种考虑节点拥塞情况的概率路由算法,将节点相遇的概率和节点拥塞的情况综合起来,得到一个报文的递交概率,降低了由于拥塞对网络性能的影响,提高了报文的递交率,减小了报文在缓存中排队等候的时间。仿真结果表明,与传统的概率路由算法相比,在改进后的概率路由算法中报文递交率显著提高,平均延迟也在降低。     

无线传感器网络节点的公平缓冲队列管理

在无线传感器网络中,存在拥塞的现象,而拥塞造成的延迟或报文丢弃在某些关键应用中是不允许发生的。从节点缓冲管理的角度,分析了基于优先级的VPRED缓冲管理算法,经过计算简化,应用到无线传感器节点的缓冲队列管理中,保证了关键数据较少丢包的传输。     

基于客户端下溢概率估计的可伸缩视频传输算法

针对无线网络中可伸缩视频传输存在的问题,提出一种基于客户端下溢概率估计的传输算法.算法通过统计当前播放缓冲区容量的变化情况,计算出缓冲区下溢的概率,以此为根据选择合适的视频发送层数,从而在充分利用有效带宽的同时尽可能减少视频下溢频率.仿真结果表明,该算法能够保证在无线网络中视频传输具有较低的下溢概率和较高的视频质量.     

网络环境下视频连续播放控制算法

从信宿端的角度来解决视频媒体的同步连续播放,提出了一种自适应的动态媒体播放算法.分析了马尔可夫调制的泊松到达情况下的排队模型,给出了缓冲区门限的选取原则,最后提出了通过用不连续性和播放失真的方差来衡量同步性能方法,实验结果表明,该算法的同步性能优于Yuang的算法.通过选择合适的参数,可有效地防止缓冲区下溢造成的播放停顿以及缓冲区上溢造成的播放跳跃,从而实现同步平滑播放.     

实时语音的自适应回放算法

由于丢包和延时抖动的引入而使网络传输的实时语音质量让人难以接受，目前对丢包和延时抖动提出了很多的解决方案．但是却很少把这两者结合在一起进行研究。本文提出了一种新的自适应回放算法，通过监测接收和回放队列，结合丢包的自适应恢复技术，达到语音高质量的连续回放。实验证明，该算法能在严格的平均回放延时条件下努力减小由于超时而引起的丢包，获得较好的重建语音质量。     

Playout buffering of speech packets based on a quality maximization approach

To combat jitter problems in voice streaming over packet networks, playout buffering algorithms are used at the receiver side. Most of the proposed solutions rely on two main operations: prediction of delay statistics for future packets; setting of the end-to-end delay so as to limit or avoid packet losses. In recent years, a new approach has been presented, which is based on using a quality model to evaluate the impact of both packet loss and delay on the voice quality. Such a model is used to find the buffer setting that maximizes the expected quality. In this paper, we present a playout buffering algorithm whose main contribution is the extension of the new quality-based approach to the case of voice communications affected by bursty packet losses. This work is motivated by two main considerations: most of IP telephony applications are characterized by bursty losses instead of random ones; the human perception of the speech quality is significantly affected by the temporal correlation of losses. To this purpose, we make use of the extensions proposed in the ETSI Tiphon for the ITU-T E-Model so as to incorporate the effects of loss burstiness on the perceived quality. The resulting playout algorithm estimates the characteristics of the loss process varying the end-to-end delay, weights the loss and the delay effects on the perceived quality, and maximizes the overall quality to find the optimal setting for the playout buffer. The experimental results prove the effectiveness of the proposed technique.     

Adaptive VoIP playout scheduling: assessing user satisfaction

Delay and packet loss dramatically affect the quality of voice-over-IP (VoIP) calls and depend on the playout buffer scheme implemented at the receiver. The choice of playout algorithm can't be based on statistical metrics without considering the perceived end-to-end conversational speech quality. The authors present a method for evaluating various playout algorithms that extends the E-model concept by estimating user satisfaction from time-varying transmission impairments. This article evaluates several playout algorithms and shows a correspondence between the authors' results and those obtained via statistical loss and delay metrics.     

HMM delay prediction technique for VoIP

This paper proposes a new algorithm for predicting audio packet playout delay for voice conferencing applications that use silence suppression. The proposed algorithm uses a hidden Markov model (HMM) to predict the playout delay. Several existing algorithms are reviewed to show that the HMM technique is based on a combination of various desirable features of other algorithms. Voice over Internet protocol (VoIP) applications produce packets at a deterministic rate but various queuing delays are added to the packets by the network causing packet interarrival jitter. Playout delay prediction techniques schedule audio packets for playout and attempt to make a reasonable compromise between the number of lost packets, the one-way delay and the delay variation since these criteria cannot be optimized simultaneously. In particular, this paper will show that the proposed HMM technique makes a good compromise between the mean end-to-end delay, end-to-end delay standard deviation and average packet loss rate.     

Novel approaches for online playout delay prediction in VoIP applications using time series models

Voice over IP (VoIP) applications requires a buffer at the receiver to minimize the packet loss due to late arrival. Several algorithms are available in the literature to estimate the playout buffer delay. Classic estimation algorithms are non-adaptive, i.e. they differ from more recent approaches basically due to the absence of learning mechanisms. This paper introduces two new formulations of adaptive algorithms for online learning and prediction of the playout buffer delay, the first one being based on the standard Box-Jenkins autoregressive model, while the second one being based on the feedforward and recurrent neural networks. The obtained results indicate that the proposed algorithms present better overall performance than the classic ones.     

A Garch-based adaptive playout delay algorithm for VoIP

Network delay, packet loss and network delay variability (jitter) are important factors that impact on perceived voice quality in VoIP networks. An adaptive playout buffer is used in a VoIP terminal to overcome jitter. Such a buffer-control must operate a trade-off between the buffer-induced delay and any additional packet loss rate. In this paper, a Garch-based adaptive playout algorithm is proposed which is capable of operating in both inter-talkspurt and intra-talkspurt modes. The proposed new model is based on a Garch model approach; an ARMA model is used to model changes in the mean and the variance. In addition, a parameter estimation procedure is proposed, termed Direct Garch whose cost function is designed to implement a desired packet loss rate whilst minimising the probability of consecutive packet losses occurring. Simulations were carried out to evaluate the performance of the proposed algorithm using recorded VoIP traces. The main result is as follows; given a target Packet Loss Rate (PLR) the Direct Garch algorithm produces parameter estimates which result in a PLR closer than other algorithms. In addition, the proposed Direct Garch algorithm offers the best trade-off between additional buffering delay and Packet Loss Rate (PLR) compared with other traditional algorithms.     

Packet audio playout delay adjustment: performance bounds and algorithms

In packet audio applications, packets are buffered at a receiving site and their playout delayed in order to compensate for variable network delays. In this paper, we consider the problem of adaptively adjusting the playout delay in order to keep this delay as small as possible, while at the same time avoiding excessive “loss” due to the arrival of packets at the receiver after their playout time has already passed. The contributions of this paper are twofold. First, given a trace of packet audio receptions at a receiver, we present efficient algorithms for computing a bound on the achievable performance of any playout delay adjustment algorithm. More precisely, we compute upper and lower bounds (which are shown to be tight for the range of loss and delay values of interest) on the optimum (minimum) average playout delay for a given number of packet losses (due to late arrivals) at the receiver for that trace. Second, we present a new adaptive delay adjustment algorithm that tracks the network delay of recently received packets and efficiently maintains delay percentile information. This information, together with a “delay spike” detection algorithm based on (but extending) our earlier work, is used to dynamically adjust talkspurt playout delay. We show that this algorithm outperforms existing delay adjustment algorithms over a number of measured audio delay traces and performs close to the theoretical optimum over a range of parameter values of interest.     

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.     

Impact of Network Dynamics on User's Video Quality: Analytical Framework and QoS Provision

We develop an analytical framework to investigate the impacts of network dynamics on the user perceived video quality. Our investigation stands from the end user's perspective by analyzing the receiver playout buffer. In specific, we model the playback buffer at the receiver by a G/G/1/? and G/G/1/N queue, respectively, with arbitrary patterns of packet arrival and playback. We then examine the transient queue length of the buffer using the diffusion approximation. We obtain the closed-form expressions of the video quality in terms of the start-up delay, fluency of video playback and packet loss, and represent them by the network statistics, i.e., the average network throughput and delay jitter. Based on the analytical framework, we propose adaptive playout buffer management schemes to optimally manage the threshold of video playback towards the maximal user utility, according to different quality-of-service requirements of end users. The proposed framework is validated by extensive simulations.     

Adaptive playout scheduling algorithm tailored for real-time packet-based voice conversations over wireless ad-hoc networks

The effective provision of real-time, packet-based voice conversations over multi-hop wireless ad-hoc networks faces several stringent constraints not found in conventional packet-based networks. Indeed, MANETs (mobile ad-hoc networks) are characterized by mobility of all nodes, bandwidth-limited channel, unreliable wireless transmission medium, etc. This environment will surely induce a high delay variation and packet loss rate impairing dramatically the user experienced quality of conversational services such as VoIP. Indeed, such services require the reception of each media unit before its deadline to guarantee a synchronous playback process. This requirement is typically achieved by artificially delaying received packets inside a de-jitter buffer. To enhance the perceptual quality the buffering delay should be adjusted dynamically throughout the vocal conversation.In this work, we describe the design of a playout algorithm tailored for real-time, packet-based voice conversations delivered over multi-hop wireless ad-hoc networks. The designed playout algorithm, which is denoted MAPA (mobility aware playout algorithm), adjusts the playout delay according to node mobility, which characterizes mobile ad-hoc networks, and talk-spurt, which is an intrinsic feature of voice signals. The detection of mobility is done in service passively at the receiver using several metrics gathered at the application layer. The perceptual quality is estimated using an augmented assessment approach relying on the ITU-T E-Model paradigm while including the time varying impairments observed by users throughout a packet-based voice conversation. Simulation results show that the tailored playout algorithm significantly outperforms conventional playout algorithms, specifically over a MANET with a high degree of mobility.