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Combined Source and Channel Coding Using a Sub-Band Coder

机译:使用子带编码器的组合源和信道编码

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Sub-band coding has been shown to provide high quality speech at rates of 16 kb/s and above. In this paper we describe how one particular subband coder has been modified to give good quality speech over a noisy channel. The coder is a dynamic bit allocation sub-band coder of the type first proposed by Ramstad in 1982. It is structured in such a way that the relative importance of all bits is established as a byproduct of the dynamic bit allocation. It is shown that there is a difference in error sensitivity of four orders of magnitude between the most and the least important bits of the bit stream on an average. A very flexible unequal error protection scheme is used to match the error sensitivity of the sub-band coder bitstream. This is accomphshed using the concept of rate compatible punctured convolutional coding. These trellis codes are decoded by means of the Viterbi algorithm using soft decisions on the noisy received signal If necessary, the decoder can also include information about the time varying channel characteristics in a rather straightforward way to improve performance. All this is accomplished without any change in the encoder and the decoder strocture during changes in the level of error protection.The resulting coder is more robust to channel errors than one in which all the error protection is spread evenly among all bits or one in which all error protection is used for just a small number of important bits. The resulting coder gives robust perfonnance over a simulated noisy channel.Its nominal bit rate is 16 kb/s, with 12 kb/s assigned for speech coding and 4 kb/s assigned for error correction. The entire encoder and decoder have been implemented on a single AT&T DSP-32 digital signal processor.
机译:子带编码已显示以16 kb / s或更高的速率提供高质量的语音。在本文中,我们描述了如何修改一个特定的子带编码器以在嘈杂的信道上提供高质量的语音。该编码器是一种动态比特分配子带编码器,它是由Ramstad于1982年首次提出的。它的结构使得将所有比特的相对重要性确定为动态比特分配的副产品。结果表明,平均而言,位流中最重要和最不重要的位之间的误码敏感性相差四个数量级。一种非常灵活的不均等错误保护方案用于匹配子带编码器比特流的错误敏感性。这是使用速率兼容删余卷积编码的概念来实现的。这些网格码是通过维特比算法对有噪声的接收信号进行软判决来解码的。如果有必要,解码器还可以以一种非常直接的方式包括有关时变信道特性的信息,以提高性能。所有这些都是在错误保护级别发生变化的情况下完成的,而编码器和解码器的结构不会发生任何变化。所得到的编码器比其中所有错误保护均等地分布在所有位或其中一个错误保护的编码器对信道错误的鲁棒性更高。所有错误保护仅用于少数重要位。最终的编码器在模拟的噪声信道上提供了强大的性能,其标称比特率为16 kb / s,语音编码分配为12 kb / s,纠错分配为4 kb / s。整个编码器和解码器已在单个AT&T DSP-32数字信号处理器上实现。

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