Robust Transmission of Lossless Audio with Low Delay over IP Networks

E. Hellerud, U. Svensson
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引用次数: 2

Abstract

A system for streaming lossless audio with low delay over IP networks is presented. To achieve error robustness, the signal is divided into a base and an enhancement layer when the network is approaching congestion. The base layer is perceptually encoded using a time-varying pre- and postfilter, and this layer is transported using a high priority traffic class in a Differentiated Services (DiffServ) network. The enhancement layer is the difference between the original signal and base layer, and is transmitted using a regular Best Effort traffic class. In our experiments the system delay is just 256 samples, and it can be seen that the layering only introduces moderate amounts of redundancy, while improving the error resilience significantly.
IP网络上低延迟无损音频的鲁棒传输
提出了一种基于IP网络的低时延无损音频流传输系统。为了实现误差鲁棒性,在网络接近拥塞时将信号分为基层和增强层。基础层使用时变的预过滤器和后过滤器进行感知编码,该层在差异化服务(DiffServ)网络中使用高优先级流量类进行传输。增强层是原始信号和基础层之间的差异,并使用常规的Best Effort流量类传输。在我们的实验中,系统延迟只有256个样本,可以看出分层只引入了适量的冗余,同时显著提高了错误恢复能力。
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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