arXivDaily arXiv每日学术速递 周一至周五更新
arXiv 1909.06709cs.NI
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交互式流媒体的低延迟网络自适应差错控制

Low-Latency Network-Adaptive Error Control for Interactive Streaming

Salma Emara, Silas L. Fong, Baochun Li, Ashish Khisti, Wai-Tian Tan, Xiaoqing Zhu, John Apostolopoulos

摘要:

我们介绍了一种新的网络自适应算法,适用于减轻源和目的地之间的低延迟交互式通信的网络数据包丢失。我们的网络自适应算法实时估计最近提出的流编码的最佳参数,该流编码使用前向纠错(FEC)来纠正任意和突发丢失,这分别在音频中引起噼啪作响的噪声和不期望的抖动。特别地,目的地基于其观察到的分组丢失模式来估计适当的编码参数,并将它们发送回源以更新底层代码。此外,一个新的显式构造实用的低延迟流编码,实现纠正任意丢失的能力和纠正突发丢失的能力之间的最佳权衡。基于统计损失和真实世界的数据包丢失痕迹的仿真评估揭示了以下几点:(i)我们提出的网络自适应算法结合我们的最佳流编码可以实现显着更高的性能相比,未编码和非自适应FEC方案在UDP(用户数据报协议);(ii)我们的显式流编码可以显著优于传统MDS(最大距离可分离)流方案时,他们与我们的网络自适应算法一起使用。

英文摘要:

We introduce a novel network-adaptive algorithm that is suitable for alleviating network packet losses for low-latency interactive communications between a source and a destination. Our network-adaptive algorithm estimates in real-time the best parameters of a recently proposed streaming code that uses forward error correction (FEC) to correct both arbitrary and burst losses, which cause a crackling noise and undesirable jitters, respectively in audio. In particular, the destination estimates appropriate coding parameters based on its observed packet loss pattern and sends them back to the source for updating the underlying code. Besides, a new explicit construction of practical low-latency streaming codes that achieve the optimal tradeoff between the capability of correcting arbitrary losses and the capability of correcting burst losses is provided. Simulation evaluations based on statistical losses and real-world packet loss traces reveal the following: (i) Our proposed network-adaptive algorithm combined with our optimal streaming codes can achieve significantly higher performance compared to uncoded and non-adaptive FEC schemes over UDP (User Datagram Protocol); (ii) Our explicit streaming codes can significantly outperform traditional MDS (maximum-distance separable) streaming schemes when they are used along with our network-adaptive algorithm.

机构、总结、中文摘要均由AI分析而来,可能存在错误,请以论文正文为准。

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