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Signal modification for robust speech coding

机译:信号修改以实现健壮的语音编码

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Usually, the performance of a low-bit-rate speech coder degrades seriously in the presence of various interfering signals such as the background noise, acoustic echo, co-talkers' speech and other unwanted signals. This comes from the mismatch between the input signal and the assumed speech production model on which the design of the given speech coder is based. In this paper, we present an approach to modify the input signal such that it can be coded more effectively within the generalized analysis-by-synthesis framework. Signal modification in the presented approach is performed according to a criterion which makes a compromise between the modification and coder quantization errors. The coder-decoder (CODEC) characteristic is described in terms of a transfer matrix, and an on-line method using the recursive least square (RLS) technique is proposed to estimate it. Since each part of the speech signal is differently affected by the modification, we also devise an adaptive method based on the signal-to-quantization noise ratio (SQNR). In contrast to the conventional modification techniques, our approach can be implemented as a simple front-end for any analysis-by-synthesis type coders.
机译:通常,低比特率语音编码器的性能在存在各种干扰信号(例如背景噪声,声学回声,共同讲话者的语音和其他有害信号)的情况下会严重降低。这是由于输入信号与给定语音编码器的设计所基于的假定语音产生模型之间的不匹配造成的。在本文中,我们提出了一种修改输入信号的方法,以便可以在广义的综合分析框架内对输入信号进行更有效的编码。所提出的方法中的信号修改是根据在修改和编码器量化误差之间做出折衷的准则来执行的。根据传输矩阵描述了编解码器(CODEC)特性,并提出了一种使用递归最小二乘(RLS)技术的在线方法进行估计。由于语音信号的每个部分受修改的影响不同,因此我们还设计了一种基于信噪比(SQNR)的自适应方法。与传统的修改技术相比,我们的方法可以实现为任何按合成分析类型的编码器的简单前端。

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