An Adaptive Receiver Buffer Adjust Algorithm for Voice & Video on IP Applications

Jing Liu, Z. Niu
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引用次数: 2

Abstract

In this paper, a new adaptive receiver buffer adjust algorithm is proposed for voice and video on IP (V2oIP) applications with the consideration of voice and video's characters and network conditions. This algorithm take the voice stream as the main stream, adjust the voice buffer according to the voice characters, and synchronize the video stream with it considering video's characters. This algorithm divides network status into two modes: normal mode and spike mode, according to the delay of the coming packets and the size of buffer. In normal mode, the receiver adjusts the buffer delay at the beginning of every voice's talk-spurt. In spike mode, buffer is adjusted when every packet arrives. The adjust can based on a prediction of the delay according to the delay of former packets, or based on a prearranged way. Simulations and experiments show that this strategy can well conceal the delay jitter and reduce the packet loss rate
一种语音接收机缓冲区自适应调整算法知识产权应用短片
本文结合语音视频的特点和网络条件,提出了一种适用于IP (V2oIP)语音视频应用的自适应接收机缓冲区调整算法。该算法以语音流为主流,根据语音字符调整语音缓冲区,并根据视频字符同步视频流。该算法根据传入数据包的延迟和缓冲区的大小,将网络状态分为正常模式和尖峰模式。在正常模式下,接收器在每个声音的谈话开始时调整缓冲延迟。在尖峰模式下,当每个数据包到达时,缓冲区被调整。该调整可以基于根据前数据包的延迟预测延迟,也可以基于预先安排的方式。仿真和实验表明,该策略可以很好地掩盖时延抖动,降低丢包率
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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