Shawn> 2- Resample the input file to something ridiculously low like 40Hz, & subtract 
this from the original WAV. (Although that would probably require the above method 
anyway.)


  Assume that the original sampling frequency is f.
  Downsampling to sampling frequency f/n (n is integer) is easy, since
this is done by picking up every nth sample.
  Upsampling this data to frequency f requires filter with large order,
but since almost all input sample is zero, this process requires only
small computational power.


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Naoki Shibata   e-mail: [EMAIL PROTECTED]

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