Pro Audio Asylum

• You are aware of "fast convolution" aren't you? - real_jj 12:54:26 05/25/06 (0)
  In Reply to: posted by on December , at :

  Consider.

  If I do f&y (& meaning convolution), f a filter, y the signal in the time domain...

  If I then calculate F (fourier transform of the filter) and Y (fourier transform of the signal), then F*Y is the exact dual of f&y.
  (* being point by point multiply, complex)

  If I then do IFFT(F*Y), I get exactly f&y. You need some guard band to prevent wraparound, but that's easy to calculate.

  Perhaps you're mistaking this for something else?
  
  

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• I haven't been inside any DAW code so I can't say what's used - jcox 10:23:41 05/16/06 (0)
  In Reply to: posted by on December , at :

  but software to do synchronous conversion certainly exists

  http://www.soundslogical.com/support/resample/documentation/english/documentparts/resamplehelp-30.html

  cruise the site, matlab files, textbook references

  http://www.analog.com/UploadedFiles/Application_Notes/154260303EE183Rev5.pdf

  just from the first few hits from

  google: 320 147 sample rate conversion software

  The polyphase filter implementations can be thought of as the “fancy indexing” I referred to previously

  I have designed/implemented fir filters using polyphase concepts to deskew/time align data from a multiplexed multichannel adc system, so I am familiar with those big word too
  
  
  

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