Neural and fuzzy computation techniques for playout delay adaptation in VoIP networks

Playout delay adaptation algorithms are often used in real time voice communication over packet-switched networks to counteract the effects of network jitter at the receiver. Whilst the conventional algorithms developed for silence-suppressed speech transmission focused on preserving the relative temporal structure of speech frames/packets within a talkspurt (intertalkspurt adaptation), more recently developed algorithms strive to achieve better quality by allowing for playout delay adaptation within a talkspurt (intratalkspurt adaptation). The adaptation algorithms, both intertalkspurt and intratalkspurt based, rely on short term estimations of the characteristics of network delay that would be experienced by up-coming voice packets. The use of novel neural networks and fuzzy systems as estimators of network delay characteristics are presented in this paper. Their performance is analyzed in comparison with a number of traditional techniques for both inter and intratalkspurt adaptation paradigms. The design of a novel fuzzy trend analyzer system (FTAS) for network delay trend analysis and its usage in intratalkspurt playout delay adaptation are presented in greater detail. The performance of the proposed mechanism is analyzed based on measured Internet delays.     

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.     

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.     

Switching between fixed and call-adaptive playout: a per-call playout algorithm

Adaptive playout algorithms provide a popular way to calculate voice-over-IP (VoIP) packets' playout delay - the difference between the playout time at the receiver and the packet-generation time at the sender. The authors' proposed per-call adaptive algorithm uses network delays received from the VoIP gatekeeper to switch between fixed and call-adaptive playout. Their approach also reduces loss rates while increasing playout delay only slightly.     

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.     

大时滞网络自适应主动队列管理新算法

针对PID控制器无法严格处理主动队列管理（AQM）中的大时滞情况,且不能随着变化的网络环境在线调节参数,提出了一种基于增益自适应Smith预估控制和模糊控制的大时滞网络的自适应PID主动队列管理（GAS-FPID）算法。引入增益自适应Smith预估控制器实现滞后补偿,模糊控制器来实现PID参数动态网络环境的在线调整;NS2仿真表明,所提出算法能克服滞后的影响,能快速的适应动态网络环境,具有很好的稳定性和鲁棒性。     

Optimize adaptive media playout using dynamic fuzzy logic control for video streaming

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...     

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.     

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.     

EPA跨网段延迟自适应实时调度算法研究

针对EPA网络只支持网段内的实时数据传输而不支持跨网段的实时数据传输问题,提出了一种EPA跨网段延迟自适应实时调度算法;分析了EPA跨网段数据传输过程,给出了EPA跨网段数据传输延迟模型;通过对EPA跨网传输报文的过滤规则的改变,以及对非周期数据声明报文的改造,实现了跨网数据的实时传输。     

基于H.264可伸缩视频流的自适应媒体播放算法

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。     

一种均衡时延和精确性的自适应融合算法研究

针对无线传感器网络中时延小和精确性高不能兼得这一状况,提出了一种建立在博弈模型基础上的均衡时延和精确性的自适应数据融合方法。该方法将所有网络节点根据能耗最优进行分簇,簇头与监控中心通过博弈来自适应地选择不同融合因子的融合算法使整个网络的总效益最大。实验仿真表明,在丢包率不同时,自适应融合算法可以得到最佳的融合因子,有效实现了时延和精确性的均衡。该方法为无线传感器网络中各个指标的折中提供了参考方向。     

A new algorithm for decentralized adaptive control

A new algorithm for decentralized adaptive control is proposed in this paper. This algorithm consists of an ordinary local adaptive controller and a variable structure adaptive controller. The adaptive variable structure component of this algorithm is used to compensate for uncertain interconnections among the subsystems and to ensure global stability of the overall system. Simulation results are also presented to demonstrate the performance of the closed-loop control system     

时滞系统关于时滞参数的自适应输出反馈控制

对存在状态时滞的线性时滞系统，给出符合分离性原理的动态输出反馈控制器形式，当时滞参数不能精确已知时，给出基于观测器的关于时滞参数的自适应动态输出反馈控制器设计方案，通过求解两个相应的Riccati型矩阵不等式即可求得满足设计要求的动态输出反馈控制器及关于时滞参数的自适应律，且控制器的存在性与时滞参数精确已知时相同．最后给出了一个应用仿真示例．     

A simplified algorithm for model reference adaptive control

Generation of one of the auxiliary inputs in model reference adaptive control systems requires a positive definite quadratic function of2nsystem variables for annth order system. To generate such a function can require as many as2nsignal multiplications. In this paper, it is shown how this input can be replaced by one that is much simpler to generate and reduces the number of signal multiplications required to one. The property of global asymptotic stability of the closed-loop adaptive system is preserved with the simpler auxiliary input.     

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.     

A time-varying Newton algorithm for adaptive subspace tracking

We propose a general framework for tracking the zeros of a time-varying gradient vector field on Riemannian manifolds. Thus, a differential equation, called the time-varying Newton flow, is introduced, whose solutions asymptotically converge to a time-varying family of critical points of the corresponding cost function. A discretization of the differential equation leads to a recursive update scheme for the time-varying critical point.     

A hybrid MPSO-BP structure adaptive algorithm for RBFNs

This paper introduces a novel hybrid algorithm to determine the parameters of radial basis function neural networks (number of neurons, centers, width and weights) automatically. The hybrid algorithm combines the mix encoding particle swarm optimization algorithm with the back propagation (BP) algorithm to form a hybrid learning algorithm (MPSO-BP) for training Radial Basis Function Networks (RBFNs), which adapts to the network structure and updates its weights by choosing a special fitness function. The proposed method is used to deal with three nonlinear problems, and the results obtained are compared with existent bibliography, showing an improvement over the published methods.     

A pattern adaptive thinning algorithm

A simple sequential thinning algorithm for peeling off pixels along contours is described. An adaptive algorithm obtained by incorporating shape adaptivity into this sequential process is also given. The distortions in the skeleton at the right-angle and acute-angle corners are minimized in the adaptive algorithm. The asymmetry of the skeleton, which is a characteristic of sequential algorithm, and is due to the presence of T-corners in some of the even-thickness pattern is eliminated. The performance (in terms of time requirements and shape preservation) is compared with that of a modern thinning algorithm.