音频信号的正常化，恢复到原来的MATLAB [英] normalisation of audio signal and reverting to original matlab

问题描述

我做音频速记的项目。我需要嵌入在音频信号（.wav文件）一些文本。
所以，我转换范围从-1到1（双）的音频信号-32767至+32767（INT16），所以我冷LSB系数的嵌入的数据。
现在的问题是，我不知道如何从INT16到各自的双等值得到的值。

I am doing a project in audio stenography. I need to embed some text in an audio signal(.wav file) . So i converted the audio signal ranging from -1 to 1(double) to -32767 to +32767(int16) so i cold embed the data in the LSB of the coefficients. The problem now is that i don't know how to get the values from int16 to their respective double equivalents.

我用下面的code正常化：

I have used the following code for normalization:

    [y, fs, nBits,opts]=wavread('one.wav');
     y2=y-(min(y));
     y2=y2/max(y2);
     y2=y2* (2^16 - 1) - 2^15;
     y2b=int16(y2);

任何人都可以指导我一下这个过程相反？

can anyone guide me about the reverse process of this?

推荐答案

elomage逆转正常化，我看不出有任何理由这样做，因为LSB的丢失。

elomage reversed the normalisation, I don't see any reason to do so because the lsb gets lost.

double(x)/intmax(class(x))

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