On Wed, 16 Jan 2013 06:07:51 -0500, robert bristow-johnson wrote: >>if i were to try to re-calculate the coefficients, i would first factor >>out the constant gain, then factor both numerator and denominator into >>discrete-time poles and zeros. then map those poles and zeros back to >>analog poles and zeros using, i suppose the inverse bilinear transform >>(with warping). then re-transform back with the bilinear transform with >>the new sampling rate. >> >>i dunno. that's how i might approach it.
I just caught the tail end of this thread, so forgive me if this has been mentioned before, but Frequency Domain Least Squares (FDLS) is perfect for this application. Original IEEE article is available at http://ieeexplore.ieee.org/xpl/login.jsp?tp=&arnumber=4049924&url=http%3A%2F%2Fieeexplore.ieee.org%2Fiel5%2F79%2F4049870%2F04049924, or directly from me. MATLAB code is available from IEEE; your choice of MATLAB or C++ code is available directly from me. Greg Berchin gjberchin (at) charter (dot) net (note that Reply-To: field is corrupted) ========================= Everybody has their moment of great opportunity in life. If you happen to miss the one you care about, then everything else in life becomes eerily easy. -- Douglas Adams -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
