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Congestion control based dynamic encoding algorithms for voice and video streams in integrated services communications networks.

机译:针对集成服务通信网络中语音和视频流的基于拥塞控制的动态编码算法。

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摘要

The dynamic encoding algorithms for real-time services, including voice and video, have been known to be effective for congestion-control in packet networks. Under such an algorithm, the voice and video information is first separated into multiple streams marked with different priorities according to the significance in the reconstructing process carried out at the decoder. At a multiplexer or a switch within a network, a bit/cell-discarding algorithm is used so that when high congestion conditions are observed, less significant information is discarded. In this study, the system performance under such a mechanism is extensively studied. In the first part of this study, we consider an application of the voice dynamic encoding algorithm in a satellite network. A load-adaptive/TDMA satellite communications link which serves to interconnect broadband multi-media stations is considered. Each station supports packetized voice and data message streams and employs a dynamic voice encoding algorithm. Two distinct backbone channel allocation algorithms are introduced. We develop analytical methods for the analysis and design of such integrated networks. The effects of the propagation delay across the backbone link are especially demonstrated. Also illustrated are the performance improvements attained due to the use of the load-adaptive/TDMA scheme. In the second part of this Dissertation, the voice/video quality control under dynamic encoding algorithms is studied. In previous studies done by others, the quality of the voice/video streams has been examined for a single stream, without incorporating the impact of the multiplexing operation. In addition, only long-term quality has been considered. We take into account the high burstiness embedded in the voice and video traffic, and show that, due to this high burstiness, if not carefully designed, a dynamic encoding algorithm can cause severe short-term quality degradations. To correct this situation, we introduce various bits/cell-discarding algorithms which take into account the features of the burstiness. New voice and video degradation indicators are introduced to measure the short-term voice/video quality degradation levels. Through numerical examples, we identify and compare the system performance under the introduced bit/cell-discarding algorithms. It is shown that, under a prescribed long-term voice/video quality level, the introduction of a probabilistic component in the bit/cell-discarding process significantly improves the system performance. Moreover, it is observed for video streams that the short-term video quality performance is further improved by dynamically calculating the value of the probabilistic component according to the frame-level burstiness information.
机译:已知用于语音和视频的实时服务的动态编码算法对于分组网络中的拥塞控制有效。在这种算法下,语音和视频信息首先根据在解码器执行的重构过程中的重要性,被分离为标记有不同优先级的多个流。在网络中的多路复用器或交换机处,使用位/单元丢弃算法,以便在观察到高拥塞情况时,丢弃不太重要的信息。在这项研究中,对这种机制下的系统性能进行了广泛的研究。在本研究的第一部分中,我们考虑了语音动态编码算法在卫星网络中的应用。考虑了用于互连宽带多媒体站的负载自适应/ TDMA卫星通信链路。每个站都支持打包的语音和数据消息流,并采用动态语音编码算法。介绍了两种不同的骨干信道分配算法。我们开发用于分析和设计此类集成网络的分析方法。特别说明了骨干链路上传播延迟的影响。还说明了由于使用负载自适应/ TDMA方案而获得的性能改进。在本文的第二部分,研究了动态编码算法下的语音/视频质量控制。在其他人进行的先前研究中,已经针对单个流检查了语音/视频流的质量,而没有考虑多路复用操作的影响。另外,仅考虑了长期质量。我们考虑到嵌入在语音和视频流量中的高突发性,并表明由于这种高突发性,如果不精心设计,动态编码算法会导致严重的短期质量下降。为了纠正这种情况,我们引入了多种考虑了突发性特征的比特/信元丢弃算法。引入了新的语音和视频质量下降指标,以测量短期语音/视频质量下降水平。通过数值示例,我们在引入的位/单元丢弃算法下识别并比较了系统性能。结果表明,在规定的长期语音/视频质量水平下,在比特/单元丢弃过程中引入概率性分量可以显着提高系统性能。此外,对于视频流观察到,通过根据帧级突发性信息动态地计算概率分量的值,可以进一步改善短期视频质量性能。

著录项

  • 作者

    Chang, Kuo-Kuang.;

  • 作者单位

    University of California, Los Angeles.;

  • 授予单位 University of California, Los Angeles.;
  • 学科 Engineering Electronics and Electrical.
  • 学位 Ph.D.
  • 年度 1995
  • 页码 122 p.
  • 总页数 122
  • 原文格式 PDF
  • 正文语种 eng
  • 中图分类 无线电电子学、电信技术;
  • 关键词

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