An algorithm for playout of packet voice based on adaptive adjustment of talkspurt silence periods

Jesus Pinto, Kenneth J. Christensen
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引用次数: 75

Abstract

In a typical real-time voice application, voice packets are produced at deterministically-spaced time intervals. In the network they encounter a variable amount of delay that changes the deterministic time intervals. A receiving host can employ a buffer to delay the playout of the voice packets in order to reconstruct the original timing. Adaptive techniques can perform continuous estimations of the network delays and dynamically adjust the buffering delay at the beginning of each talkspurt. Such adjustments are usually undetectable by the human listener. This research develops a new, adaptive "gap-based" algorithm that can be tuned for both end-to-end delay and packet loss to satisfy a user-desired tolerance. This new gap based algorithm adapts the buffering delay based on historical information of arrival and playout times of received voice packers in the previous talkspurt. A simulation study shows that the new gap based algorithm can reduce delay by 10% when compared with existing methods.
一种基于自适应调整话音爆发沉默周期的分组语音播放算法
在典型的实时语音应用中,语音包是在确定间隔的时间间隔内产生的。在网络中,它们会遇到可变量的延迟,这会改变确定性的时间间隔。接收主机可以使用缓冲器来延迟语音包的播放,以便重建原始时间。自适应技术可以对网络延迟进行连续估计,并在每次对话爆发开始时动态调整缓冲延迟。这种调整通常是人类听众察觉不到的。本研究开发了一种新的自适应“基于间隙”的算法,该算法可以针对端到端延迟和数据包丢失进行调整,以满足用户期望的容忍度。基于间隙的缓冲延迟算法是根据上次话音爆发中接收到的语音包到达和播放时间的历史信息来调整缓冲延迟的。仿真研究表明,与现有算法相比,该算法可将时延降低10%。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
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