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基于CELP的常用语音编码器间的参数直接转换

Smart Transcoding for Celp Based Speech Coders

【作者】 桂苹

【导师】 吴镇扬;

【作者基本信息】 东南大学 , 信号与信息处理, 2004, 硕士

【摘要】 随着无线通信网络和互联网络技术的发展,综合语音网络的互通性变的越来越重要。综合语音网络支持多种语音编码器,需要对各种码流之间进行转换。传统的码流转换方法是直通级联转换(tandem)方式。直通级联方式通过解码前一个编码器的码流重构语音信号,然后再通过后一个编码器将重构语音信号编码为目标码流,然而,该方法的缺点是:运算量大,额外的延时。本文在深入研究了目前应用广泛的三种基于CELP技术的语音编码标准G.723.1、G.729和AMR的基础上,提出了一种G.723.1与G.729码流间参数直接转换(Smart Transcoding)的算法。该算法利用了G.723.1和G.729编码器的共性,对两个语音编码器的线谱对(LSP)参数、自适应码本(adaptive codebook)参数、固定码本(fixed codebook)参数和增益参数进行了直接转换。在不产生中间合成语音的情况下,实现了码流间的直接转换。本文并给出了非正式的主观听音测试结果和算法的复杂度分析。最后,在简单介绍了已有的AMR与G.729码流间参数直接转换算法的基础上,本文给出了该算法的改进,包括固定码本的快速搜索和增益码本的预搜索。

【Abstract】 With the development of wireless network and Internet, for a successful integration of the speech networks, the interoperability becomes more and more important. It is necessary for integrated speech networks, which support multiple speech coders, to translate bit streams between different speech coders seamlessly.Connecting two coders in tandem is the traditional way to realize bit streams translation. Tandem is to reconstruct speech signals by decoding bit streams of one codec and then to encode the speech re-constructed by another coder, however, tandem coding is associated with several problems such as high computational load and additional transmission delay.In this paper, after thorough study of the three popular CELP based speech coders including G.723.1, G.729 and AMR, a smart transcoding algorithm between G.723.1 and G.729 speech coders is proposed. This algorithm utilizes the commonness between these two speech coders to make a direct translation of LSP (Linear Spectral Pair) parameters, adaptive codebook parameters, fixed codebook parameters and codebook gain parameters. Therefore, the bit streams conversion is accomplished without reconstructing the speech signals. Informal subjective speech quality evaluations and analysis of the complexity of the transcoding algorithm are also given.Finally, a review of the already existed smart transcoding algorithm between G.729 and AMR speech coders is presented. Some further improvements on this algorithm are made, including fast fixed codebook search and codebook gain pre-search.

  • 【网络出版投稿人】 东南大学
  • 【网络出版年期】2005年 02期
  • 【分类号】TN761
  • 【被引频次】10
  • 【下载频次】212
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