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Joint source/channel coding of statistically multiplexed real-time services on packet networks

机译:分组网络上统计复用实时服务的联合源/信道编码

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The interaction of congestion control with the partitioning of source information into components of varying importance for variable-bit-rate packet voice and packet video is investigated. High-priority transport for the more important signal components results in substantially increased objective service quality. Using a Markov chain voice source model with simple PCM speech encoding and a priority queue, simulation results show a signal-to-noise ratio improvement of 45 dB with two priorities over an unprioritized system. Performance is sensitive to the fraction of traffic placed in each priority, and the optimal partition depends on network loss conditions. When this partition is optimized dynamically, quality degrades gracefully over a wide range of load values. Results with DCT encoded speech and video samples show similar behavior. Variations are investigated such as further partition of low-priority information into multiple priorities. A simulation with delay added to represent other network nodes shows general insensitivity to delay of network feedback information. A comparison is made between dropping packets on buffer overflow and timeout based on service requirements.
机译:研究了拥塞控制与将源信息划分为可变比特率的分组语音和分组视频的重要性各不相同的组件之间的相互作用。对更重要的信号分量进行高优先级传输会大大提高目标服务质量。使用具有简单PCM语音编码和优先级队列的马尔可夫链式语音源模型,仿真结果表明,与未优先级的系统相比,具有两个优先级的信噪比提高了45 dB。性能对放置在每个优先级中的流量所占比例敏感,并且最佳分区取决于网络丢失情况。当动态优化此分区时,质量会在很大的负载值范围内正常降低。 DCT编码的语音和视频样本的结果显示出相似的行为。研究了各种变化,例如将低优先级信息进一步划分为多个优先级。添加延迟以表示其他网络节点的模拟显示了对网络反馈信息延迟的总体敏感性。根据服务需求,在缓冲区溢出时丢弃数据包和超时之间进行比较。

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