![]() ladspa is also coming.Īnd, i haven't studied the licensing agreements, so if you don't want me to port your plugins, let me know, and i'll remove it, no problem. some more clean-up and tweaking could be done, of course, and (at least in theory) it should be possible to just re-compile for 64bit without any changes (except compiler command line arguments), but i haven't tested this yet. ![]() things went pretty well, the porting took about an hour or so, mainly just blindly converting the code to pascal, and then restructuring it a little to fit the zenith framework. Zenith (my new framework, this time in pascal) has "easy porting of js plugins to vst" as an "important" feature, and these was/is nice for testing it in a more "real-life" scenario. Linux (32bit): fx_sonicmaximizer.so, fx_sonicenhancer.so Windows (32bit): fx_sonicmaximizer.dll, fx_sonicenhancer.dll I converted both the sonic maximizer (just the first version, yet.) and the sonic enhancer to vst, available here: Sorry to interrupt the discussion, but i thought it might be of interest to some: Instead of summing the inverted output of the LP and HP with an non-inverted output of the BP, i sum non-inverted outputs of a AP, HP, LP filters, which results in pretty much the same thing (ignoring the prewarp issues caused by the bilinear transform) and it has the same performance. I do something "different" myself in my plugin version above though. the circuit design in question does that, for which i must say it does it to a certain extent in theory. this in fact is used in sub-woofer design quite a lot as a compensation. they probably should have used your design instead, with a passive component topology (with first order filters), but i do think there are a couple of possible reasons behind this:ġ) audiophile frequency "dip" when boosting a "band" - a bit questionable, but apparently people tend to like it.Ģ) longer delays for lower frequencies on the group delay plot. The filter itself isn't really a "pseudo" in my opinion. i haven't calculated a really accurate Q value in the above plugin, while only approximating, but then again the Q calculation itself is quite complex.i can say that its a based on a ratio of R25 and R26 and ends up near ~0.23. they are using a more "passive" (or "constant") parameter state (notice all the R values), where the only goal is to provide lower Q, for less steeper curves. Overall, apart from the choice of Fc, the Q parameter is quite odd. this will cause a group delay, where low frequencies are delayed (see "manual quote" in previous post). Lets also examine the phase of the output node: same would not be present when summing the input with a first order high pass. We get a nice boost of the HP band, but there is a "dip" (marked in yellow), due to phase cancelation. Lets examine what happens if we reduce the impedance of R19: your HP filter would result in 90deg, since its only formed by one pole. Notice the 180deg phase shift cased by the HP output. i do like however how one draws the schematic itself in LTSpice - reminds me a little of ORCAD and CADSTAR for DOS. i'm mostly a QUCS user, but both programs suck in some aspects to be honest. Looks pretty darn close to 700 Hz.so lets enter the schematic in circuit simulation software and look at things a bit more: ![]() Mind that, for example, the two integrators that you mention, feedback to the summator forming a biquadratic transfer function:Ĭode: angular frequency squared for this filter topology:Ĭonvert all to "native" units and get the value: I do not know where "224Hz and 2443Hz" come from in your calculations, so please correct me if you are running some simulations, but i can certainly point out that this filter is a _biquad_ state variable filter. ![]() this is a design based on a KHN state variable filter - which is pretty much one of the most common out there. Lets start with the filter "network" itself. No intent to slate you over public discussions, such as forums, but i think you are terribly confusing a very common, active component, state variable topology (that has been around for ages) with something else.if you are not an electronic engineer - please excuse me. The delay is only caused by phase shift of two integrators. Look the complete schematic there are no delay line in the lower frequencies process. There are two cutoff frequencies at 224Hz and 2443Hz and not only one at 700Hz !!! The filter is a pseudo state variable filter composed of two integrators and a subtractor like in my filter. Look the filter on the schematic of the BBE Sonic Maximizer 482i :
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