title: normalisation of audio signal and reverting to original matlab
body: <p>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.</p>
  
  <p>I have used the following code for normalization:</p>
  
  <pre><code>    [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);
  </code></pre>
  
  <p>can anyone guide me about the reverse process of this?</p>