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Design and Analysis of Optimal Adaptive De-jitter Buffers

机译:最优自适应De-Jitter缓冲区的设计与分析

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In order to transfer voice or some other application requiring real-time presentation over a packet network we need a de-jitter buffer to eliminate delay jitters. An important design parameter is the depth of the de-jitter buffer since it influences two important parameters controlling voice quality, namely voice-path delay and packet loss probability. In this paper we propose and study several schemes for optimally adjusting the depth of the de-jitter buffer. In addition to de-jitter-buffer depth adjustments within a call, the initial value and rates of changes of the de-jitter buffer depth are allowed to depend on the class of the call and are adaptively adjusted (upwards or downwards) for every new call based on voice-path delay and packet loss probability measurements over one or more previous calls. Parameter adjustments are geared towards either (a) minimizing voice-path delay while maintaining a packet loss probability objective, or (b) maximizing R-Factor, an objective measure of voice quality that depends both on the voice-path delay and the packet loss probability. Using simulation models it is shown that adaptive schemes perform better than static ones and adaptive schemes with learning perform better than ones without learning.
机译:为了传递声音或在分组网络上的一些其他应用要求实时呈现,我们需要去抖动缓冲器,以消除时延抖动。一个重要的设计参数是去抖动缓冲器的深度,因为它影响控制语音质量,即语音路径延迟和分组丢失概率的两个重要参数。在本文中,我们提出并研究一些方案来优化调整去抖动缓冲器的深度。除了去抖动缓冲器的呼叫内的深度的调整,被允许依赖于类呼叫的初始值和去抖动缓冲器深度的变化率,并自适应地调整(向上或向下),用于每一个新的基于语音的路径时延和丢包率的测量在一个或多个以前的电话呼叫。参数调整被朝向任一齿轮的(a)最小化语音路径延迟,同时维持的分组丢失概率的目标,或(b)最大限度地提高R-因子,既依赖于语音路径延迟和分组丢失的语音质量的客观量度可能性。使用仿真模型则显示,自适应方案不是静态的,并与学习比那些没有进行学习较好的适应方案有更好的表现。

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