<div dir="ltr"><div style="font-family:arial,sans-serif;font-size:12.727272033691406px">It's been noted that [vcf~] can be obtained with a [cpole~] object - though I didn't do it yet as I find it a bit hard to get to the coeficients from the vcf~ code.<br>
</div><div style="font-family:arial,sans-serif;font-size:12.727272033691406px"><br></div><div style="font-family:arial,sans-serif;font-size:12.727272033691406px">Anyway, if you can get it with a [cpole~] object, it means you could do it with [biquad~] coeficients, right? I suppose so, but then, how exactly? I mean, if I have the coeficients of [cpole~], how do I get to [biquad~]'s?</div>
<div style="font-family:arial,sans-serif;font-size:12.727272033691406px"><br></div><div style="font-family:arial,sans-serif;font-size:12.727272033691406px">maybe this will help getting biquad's coefficients from [vcf~]'s parameters... maybe not, I don't know, I need help :) </div>
<div style="font-family:arial,sans-serif;font-size:12.727272033691406px"><br></div><div style="font-family:arial,sans-serif;font-size:12.727272033691406px">thanks</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">
2014-07-19 0:44 GMT-03:00 Alexandre Torres Porres <span dir="ltr"><<a href="mailto:porres@gmail.com" target="_blank">porres@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">hi all, I've been working on filter patches for my courses and I'm still failing ti get biquad coeficients from the [vcf~] code. Maybe anyone out there could help?<div><br></div><div>I wanted this to plot the frequency response in realtime... </div>
<div><br></div><div>the [vcf~] filters aren't in the audio <span style="font-family:arial,sans-serif;font-size:12.727272033691406px">Audio-EQ-Cookbook, and the code looks a bit too complicated</span></div><div><span style="font-family:arial,sans-serif;font-size:12.727272033691406px"><br>
</span></div><div><span style="font-family:arial,sans-serif;font-size:12.727272033691406px">thanks</span></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2014-05-26 10:23 GMT-03:00 Joe White <span dir="ltr"><<a href="mailto:white.joe4@gmail.com" target="_blank">white.joe4@gmail.com</a>></span>:<div>
<div class="h5"><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Ahh yes of course thanks Frank!<div><br></div><div>Have you guys checked out this paper on '<a href="http://www.aes.org/e-lib/browse.cfm?elib=13397" target="_blank">High-Order Digital Parametric Equalizer Design</a>'? Apparently it reduces the need to cascade filter implementations to achieve high orders. </div>
<div><br></div><div>Cheers,</div><div>Joe</div></div><div><div><div class="gmail_extra"><br><br><div class="gmail_quote">On 24 May 2014 09:53, Frank Barknecht <span dir="ltr"><<a href="mailto:fbar@footils.org" target="_blank">fbar@footils.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Joe,<br>
<br>
versions of these calculations without [expr] are also part of the<br>
rj-library as u_lowpass, u_lowpassq etc. These have been taken straight<br>
from the Audio-EQ-Cookbook.<br>
<br>
Ciao<br>
<span><font color="#888888">--<br>
Frank<br>
</font></span><div><div><br>
On Fri, May 23, 2014 at 12:06:45PM +0100, Joe White wrote:<br>
> Thanks for the abstractions Chris. Am I correct in thinking the licensing<br>
> issues for [expr] have been resolved now?<br>
><br>
> Cheers,<br>
> Joe<br>
><br>
><br>
> On 21 May 2014 23:22, Chris Clepper <<a href="mailto:cgclepper@gmail.com" target="_blank">cgclepper@gmail.com</a>> wrote:<br>
><br>
> > On Wed, May 21, 2014 at 5:31 PM, Joe White <<a href="mailto:white.joe4@gmail.com" target="_blank">white.joe4@gmail.com</a>> wrote:<br>
> ><br>
> >><br>
> >><br>
> >> Is it intentional to not a bank of go-to filters? [biquad~] is the next<br>
> >> one I would go to, but generating your own coefficients isn't that... err..<br>
> >> efficient when you're wanting some that just 'works' :)<br>
> >><br>
> >><br>
> > Attached are a set of abstractions wrapping most of the 'Audio EQ<br>
> > Cookbook' formulae around biquad~. It would be nice for Pd to include<br>
> > something like this.<br>
> ><br>
> > The only drawback to [biquad~] is it doesn't take audio rate coefficients.<br>
> > There are of course externals that do audio rate for cutoff, Q, etc.<br>
> ><br>
> > Chris<br>
> ><br>
> ><br>
> >><br>
> >><br>
> >> On 21 May 2014 17:31, Miller Puckette <<a href="mailto:msp@ucsd.edu" target="_blank">msp@ucsd.edu</a>> wrote:<br>
> >><br>
> >>> Hi Joe -<br>
> >>><br>
> >>> That code is an approximation that works well for low cutoff<br>
> >>> frequencies but badly for high ones. (I should probably warn<br>
> >>> about this in the help window... that'll go on my dolist)<br>
> >>><br>
> >>> cheers<br>
> >>> M<br>
> >>><br>
> >>><br>
> >>> On Fri, May 16, 2014 at 12:58:31PM +0100, Joe White wrote:<br>
> >>> > Hi,<br>
> >>> ><br>
> >>> > I've been looking at the [lop~] implementation (Pd-0.45-4) and noticed<br>
> >>> > something that seem weird to me.<br>
> >>> ><br>
> >>> > In d_filter, line 176:<br>
> >>> ><br>
> >>> > static void siglop_ft1(t_siglop *x, t_floatarg f)<br>
> >>> > {<br>
> >>> > if (f < 0) f = 0;<br>
> >>> > x->x_hz = f;<br>
> >>> > x->x_ctl->c_coef = f * (2 * 3.14159) / x->x_sr;<br>
> >>> > if (x->x_ctl->c_coef > 1)<br>
> >>> > x->x_ctl->c_coef = 1;<br>
> >>> > else if (x->x_ctl->c_coef < 0)<br>
> >>> > x->x_ctl->c_coef = 0;<br>
> >>> > }<br>
> >>> ><br>
> >>> ><br>
> >>> > Is it correct that for:<br>
> >>> ><br>
> >>> > y[n] = x[n] * a + y[n-1] * b<br>
> >>> ><br>
> >>> > *a = 2π * Fc / Fs*<br>
> >>> > b = 1.0 - a<br>
> >>> ><br>
> >>> > where Fc is the cut-off frequency and Fs the sampling frequency.<br>
> >>> ><br>
> >>> > I appreciate the a coefficient is bounded afterwards but wouldn't that<br>
> >>> mean<br>
> >>> > that Fc values greater than Fs / 2π will have no impact on the sound<br>
> >>> being<br>
> >>> > processed.<br>
> >>> ><br>
> >>> > For example if Fs is 44100, then Fc values above ~7020Hz will not<br>
> >>> affect<br>
> >>> > the filter.<br>
> >>> ><br>
> >>> > Have I missed something crucial or could this a bug in the code?<br>
> >>> ><br>
> >>> > The simple IIR filter described in<br>
> >>> > <a href="http://en.wikipedia.org/wiki/Low-pass_filter" target="_blank">http://en.wikipedia.org/wiki/Low-pass_filter</a> suggests that the actual<br>
> >>> > coefficient calculation should be more like:<br>
> >>> ><br>
> >>> > a = 2π*Fc / (2π*Fc + Fs)<br>
> >>> ><br>
> >>> > Looking forward to understand this more!<br>
> >>> ><br>
> >>> > Cheers,<br>
> >>> > Joe<br>
> >>> ><br>
> >>> > --<br>
> >>> > Follow me on Twitter @diplojocus<br>
> >>><br>
> >>> > _______________________________________________<br>
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> >>><br>
> >>><br>
> >><br>
> >><br>
> >> --<br>
> >> Follow me on Twitter @diplojocus<br>
> >><br>
> >> _______________________________________________<br>
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> >><br>
> ><br>
><br>
><br>
> --<br>
> Follow me on Twitter @diplojocus<br>
<br>
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</div></div></blockquote></div><br><br clear="all"><div><br></div>-- <br>Follow me on Twitter @diplojocus
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