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Robust low-latency voice and video communication over best-effort networks.

机译:尽力而为网络上的强大的低延迟语音和视频通信。

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The quality of service limitation of today's best-effort networks poses a major challenge for low-latency multimedia communication. Excessive delay, packet loss, variations in throughput, and high delay jitter all impair the performance of the communication. In this work, these challenges are addressed at transport and application layers of real-time and on-demand streaming media systems.; On the client side, passive schemes including adaptive playout scheduling and low-latency loss-concealment have shown to significantly improve the trade-off between buffering delay and packet loss for real-time voice communication. The playout schedule of the media packet is adaptively adjusted in a highly dynamic way and the proposed underlying packet scaling technique based on time-scale modification works elegantly and preserves good sound quality.; At the transport layer, the communication performance is further improved by exploiting diversity of multiple transmission channels, where the source media are coded into multiple complementary streams that are sent over independent network paths. Experiments demonstrate further gains of reduced latency and distortion, resulting from path diversity.; In order to combat network losses for real-time and on-demand video communication, which exhibits stronger dependency across packets, a network-adaptive coding scheme is employed to dynamically manage the packet dependency using optimal reference picture selection. The selection of the reference is achieved within a rate-distortion optimization framework and is adapted to the varying network conditions.; For network-adaptive streaming of pre-encoded media, the potential mismatch error during bitstream assembly is avoided by using a layered coding structure. Based on an accurate loss-distortion model introduced in this work, a prescient scheme that optimizes the dependency of a group of packets is also proposed to achieve global optimality as well as improved rate-distortion performance. With the improved trade-off between compression efficiency and error resilience, the proposed system does not require retransmission of lost packets, which makes low-latency communication possible.; These solutions provided at the receiving client, the transport layer, and source coding significantly improve the perceptual quality and reduce the latency for media communication over best-effort networks, without any requirement to modify the current or future network infrastructure.
机译:当今尽力而为网络的服务质量限制对低延迟多媒体通信提出了重大挑战。过多的延迟,数据包丢失,吞吐量变化以及高延迟抖动都会损害通信性能。在这项工作中,这些挑战在实时和按需流媒体系统的传输和应用层得到解决。在客户端,包括自适应播出调度和低延迟丢失隐藏在内的无源方案已显示出可显着改善实时语音通信的缓冲延迟和数据包丢失之间的折衷。媒体包的播出时间表以高度动态的方式自适应地调整,并且所提出的基于时标修改的基础包缩放技术可以优雅地工作并保持良好的声音质量。在传输层,通过利用多个传输通道的多样性进一步提高通信性能,其中,源媒体被编码为在独立网络路径上发送的多个互补流。实验表明,由于路径的多样性,可以进一步减少延迟和失真。为了解决实时和点播视频通信的网络丢失问题,该消息在各个数据包之间表现出更强的依赖性,采用网络自适应编码方案,通过使用最佳参考图片选择来动态管理数据包的依赖性。参考的选择是在速率失真优化框架内实现的,并且适应于变化的网络条件。对于网络自适应的预编码媒体流,通过使用分层编码结构,可以避免在位流组装过程中可能出现的不匹配错误。基于这项工作中引入的精确的损失失真模型,还提出了一种优化一组数据包依存性的先见之明的方案,以实现全局最优性以及改善的速率失真性能。由于压缩效率和错误恢复能力之间的折衷关系得到改善,所以所提出的系统不需要重新传输丢失的分组,这使得低延迟通信成为可能。在接收客户端,传输层和源代码编码处提供的这些解决方案可显着提高感知质量并减少通过尽力而为网络进行媒体通信的延迟,而无需修改当前或将来的网络基础结构。

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