This is completely off topic but are you mental? On Mon, Jan 7, 2013 at 3:01 PM, Ove Karlsen < ove.karl...@paradoxuncreated.com> wrote:
> On 1/7/2013 1:57 PM, Ove Karlsen wrote: > >> What KvR didn´t understand 10 yrs ago, and still don´t understand. >> >> Why does digital synths often sound so bad? Either stale, or harsh etc. >> >> Let me tell you in complete truth and honesty, is has got nothing to do >> with digital. It has something to do with the engineers making the >> algorithms. >> >> When I was a newbie DSP engineer, the first thing I tried was making a >> TB-303 filter. Which is what a lot of people do first. I talked to the >> people on #musicdsp, and they had little clue, some had tried and said it >> was difficult or impossible, some say they had succeded but their filters >> didn´t sound too good. >> >> On a few days, not having touched code, since I was 12 years old, I did a >> resonance filter, that screamed and shreaked. Some engineers in the KvR >> forum, said it was a bad thing to do, because their job now got so much >> more difficult. >> >> When in reality, it was not difficult at all. And this is typical for >> those kinds of engineers. They don´t get into the algorithm. They don´t >> understand what is going on. Instead very unecesary high-level theorems, >> they try to fit into what is simple analog feedback paths. >> >> One of the guys even worked with supposedly professor for many years, and >> they did not come up with anything good. >> >> They argue it is something to do with frequency-response, for instance, >> why the analog filters sound the way they do, and it cannot completely be >> done in digial. >> >> All this is just crazy trash. >> >> Later I actually looked at the schematics for the 303, and realized there >> was just four feedback-paths with one negative feedback-path around. It is >> as simple as that. That is all "analog vintage" synth-filters. There is >> absolutely no obscurity going on, it is as simple as it can be. >> >> Knowing that analog has a certain headroom, and that components are a bit >> inaccurate, and there is often some highpassing going on, due to the >> frequency-response of the components, you can model that, VERY SIMPLY, and >> without much cpu use. Some of the stuff released on KvR uses extreme cpu, >> and even sounds bad. >> >> Try this ok, in your synth, and you will realizing that digital can sound >> just as good as analog, and without the inaccuracies. And analog often has >> characteristics you DON´T want. So it is even better. >> >> Released under The Beneficient Open-source licence. Please google it. >> Since this licence allows for functions alone, to be released as opensource >> you can make it a function, and use it alongside whatever else you use. >> >> >> //licenced under The Beneficient Open-source Licence. >> // Osc lo-emph. >> b_lo = b_lo + ((-b_lo + b_v) * b_lfr); // for emulating the >> analog-charateristic of more saturation in the low-freq. (due to saturated >> buffers) >> b_v = b_v - b_lo; >> b_v = b_v + (b_lo * b_lgn); >> >> // there was some earlier code here that was not intended in the paste. > > if (i_ftype == 1) { // 24dB lowpass ("ladder") >> double b_rez = b_aflt5 - b_v; // sub = no attenuation with >> rez. >> b_v = b_v - (b_rez*b_fres); // negative feedback for >> resonance. >> b_v = b_v * b_off2; // gain offset >> b_v = b_v + ((fvar90-0.5)*2); // bias >> if (b_v > 1) {b_v = 1;} else if (b_v < -1) {b_v = -1;} // clip >> >> //sat/soften clip. >> double b_vr = b_v; if (b_vr < 0) {b_vr = -b_vr;} >> b_vr = 1-b_vr; >> b_vr = pow(b_vr,fvar91*10); // something I tested at the >> time, this is a filter from my synth "Abdullah", and work in progress. >> b_vr = 1-b_vr; >> if (b_v < 0) {b_vr = -b_vr;} >> >> b_v = b_vr; >> b_v = b_v - ((fvar90-0.5)*2); // bias >> b_v = b_v / b_off2; >> >> // you can also do clipping at 0.0001 for instance, and mix, and get a >> little resonance buildup, before resonance hits the audible range. A bit >> similar to how some zero-cross distortion works. >> >> b_aflt1 = b_aflt1 + ((-b_aflt1 + b_v) * b_fenva); >> b_aflt2 = b_aflt2 + ((-b_aflt2 + b_aflt1) * b_fenva); >> b_aflt3 = b_aflt3 + ((-b_aflt3 + b_aflt2) * b_fenva); >> b_aflt4 = b_aflt4 + ((-b_aflt4 + b_aflt3) * b_fenva); >> b_v = b_aflt4; >> >> b_hp = b_hp + ((-b_hp + b_v) * b_fhp); // highpass to emulate >> analog, and get nice resonance, and also remove DC. >> b_v = b_v - b_hp; >> b_aflt5 = b_v; >> } >> >> >> That is the ultimate "analog" filter, completely digital, and without >> inaccuracies, and ofcourse with perfect keytracking etc. >> >> Forget all the obfuscating arrogant atheist KvR-nerds. This is the real >> deal. >> >> And all my DSP is just as perfect, and they never did anything of that >> either. >> >> And Unix-philosophy is really close to my philosophy of "least >> obscurity". So it would be natural for this to develop and etablish itself >> on Linux. I was a "hacker" in my teens, and I guess many who have been into >> hacking, and brilliant programming, really celebrates God, and ofcourse >> comes to the same idea of least obcurity, which is also very much like >> (non-idolaterous) religion. >> >> Instead ofcourse KvR bans the brilliant, who even talks about a >> peacebringing religion, and peaceful meditation, according to Gods praises, >> and the highest of intelligence, infinite human unfolding and rights, if >> you wish. And that is the incoherent idolater/faithless. >> >> Peace Be With You. >> >> > ______________________________**_________________ > Linux-audio-dev mailing list > Linux-audio-dev@lists.**linuxaudio.org<Linux-audio-dev@lists.linuxaudio.org> > http://lists.linuxaudio.org/**listinfo/linux-audio-dev<http://lists.linuxaudio.org/listinfo/linux-audio-dev> > -- "Cheshire-Puss," she began, "would you tell me, please, which way I ought to go from here?" "That depends a good deal on where you want to get to," said the Cat. "I don't care much where--" said Alice. "Then it doesn't matter which way you go," said the Cat.
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