What do you mean by break, crashes? If yes, what version of libpd are you using? Up until about a month ago, floating point exceptions (such as NAN) would crash libpd, whereas pd-vanilla ignores them. Peter B. recently added a commit that changed this:<div>
<br></div><div><a href="https://github.com/libpd/libpd/commit/27c848a56d3d27e62231f8b0d9ef17c7c56cd9f9">https://github.com/libpd/libpd/commit/27c848a56d3d27e62231f8b0d9ef17c7c56cd9f9</a></div><div><br></div><div>Anywho, you'll probably want to take care of those NAN's anyway. :) Not sure why pd vanilla ignores SIGFPE.</div>
<div><br></div><div>cheers,</div><div>Rich<br><div><br><div class="gmail_quote">On Tue, Mar 20, 2012 at 4:57 PM, Ed Kelly <span dir="ltr"><<a href="mailto:morph_2016@yahoo.co.uk">morph_2016@yahoo.co.uk</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div style="font-size:12pt;font-family:times new roman,new york,times,serif"><div><span>Hi,</span></div><div><br><span></span></div>
<div><span>I'm trying to design or find a resonant lowpass filter for use with libPd for Openframeworks running on an iPhone 4.</span></div><div><span><br></span></div><div><span>1. I've tried the rjlib filter attached (filtertest_list.tar.gz) and it works fine in Pd, but instantly breaks when I run it on the iPhone - all other playing audio cuts out, and it is impossible to reset. I suspect that the buffers fb1 fb2 ff1 ff2 ff3 are hitting NAN values and staying there, but I haven't tested this.</span></div>
<div><br><span></span></div><div><span>2. I've had a go at implementing the filter from <a href="http://www.musicdsp.org/archive.php?classid=3#25" target="_blank">http://www.musicdsp.org/archive.php?classid=3#25</a> as a Pd external (mvcf~.tar.gz) and it is very
efficient CPU-wise. However, higher values of f or q cause the internal filter buffers to hit NAN and stay there (tested in Pd - I haven't tried this on the iPhone yet).</span></div><div><br><span></span></div><div>
<span>I would ideally like to find out why the second attempt breaks at high q and/or f, as what I want is a 24dB/Octave moog-ish filter. I'm sure those of you with higher math skills will intuitively know why, but I'm just learning calculus now (my early education was a bit chaotic). I'm also limited by the fact that this is for a commercial project (although I'll be happy to share the code :) so I can't just take a GPL external and recompiled libpd with it.<br>
</span></div><div> <br>Ed<br><br></div><div>Gemnotes-0.1alpha: Live music notation for Pure Data<br><a href="http://sharktracks.co.uk/" target="_blank">http://sharktracks.co.uk/</a></div></div></div><br>_______________________________________________<br>
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<br></blockquote></div><br></div></div>