formula: Transpose for EQ / Other devices

ffx · April 12, 2015, 1:16pm

wow cool doofa. sounds nice.

EDIT: Man, that sound really nice, just like a hard-synced osc!

EDIT: btw. why do you use an additional hydra to split freq to freq and range? You can do it directly in the knob binding, there is also scaling available…?

EDIT: Repaired some stuff
How can I use this inline hidden code display?

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true

true

[/details]

gova · April 12, 2015, 2:49pm

use the [spoiler] [/spoiler] tags

the additionalhydra is remaining from the filter with the formula device, it is used there to set a min and max value to increase the range of the frequency rotary encoder

relatedly, the more paragraphs you have in the formula device the faster It loads (which would seem counterintuitive as every paragraph has its own

gova · April 19, 2015, 3:54pm

Well I had some time to work on that tool, but I seemed to have messed up the cubic interpolation code, table.insert gets the value as NIL therefore, thinks the 3rd passed parameter is the value.

http://pastebin.com/P2baGpRE

gova · April 20, 2015, 5:50am

Found some of the issuesI had with the paste above, but I also found a copy of inkscape so maybe fitting a custom path would be a better solution instead of a(for plugins ~13 000 value) lookup table that still won’t really be precise enough. I don’t yet know how to use the path data though. I think I can get that data directly into inkscape or maybe just get the vector data out of a excel graph at a very high resolution

really don’t know yet

edit: hm… maybe not, I tried to fit a spline to the renoise filter curve, and I got it reasonable at8 points, so ~40 values to perform math on, the precision I’m looking for is 8 digits, that of many vst parameters, if I understand correctly, from the rule in inscape, my ‘precision’ is5.5 decimal points at the moment? this need more testing

EDIT2: with those 8 points, looking at the rule, I managed to graphically get a 'precision 'of 6.5 digits, still no idea how to use this data, haha

ffx · May 7, 2015, 6:56pm

Gova your algo looks interesting. Is it a general translation table generator? Sopar.value_string is the translated / displayed value?

Maybe then interpolate in realtime…

gova · May 7, 2015, 8:11pm

The script is designed to find the renoise.song().selected_device:parameter( parameter ).valueevery time the parameter’s displayedvalue changes, therefore renoise.song().selected_device:parameter( parameter ).value_string changes

It creates 2 tables with the same index, one for the parameters and the other for the strings.

Often the string table is not necessaryas the index will match the strings.

We use these tables to create a lookup table, where frequency=index and the parameter.value of that frequency as the value.

The actual translation is happening entirely in the formula device:

The frequency input sets the frequency of the base note, and that is the note received by the key input in the formula. When the key changes, the frequency of the base note is the same, so now the formula inside simply finds the frequency of this new note we received.

This frequency is looked up directly as the index of the lookup table in the formula device.

Here’s the bare version, no interpolation (the usage was completely incorrect anyway)
http://pastebin.com/2eAg9NQL

Be my guest to implement interpolation, I really don’t have the time, you may think, why bother, we already have ~4 thousand lookup values, but that’s far from what’s available, the non-renoise effectsI tested seemed to have a resolution ofat least 6decimal places of parameter().value (we’re not just talking filters here), so there’s definitely a point in that.

It would be cool if it would make a lfo device with a representation of that input curve so that linear parts if any would be clearly visible, and would allow maximum precision without increasing the lookup table size.

ffx · May 17, 2016, 7:47pm

Ok, because of a forum accident, some important contents by OopslFly were deleted. I hope u don’t mind if I repost your text from google cache. Also have a look here, Taktik posted the proper formulas:https://forum.renoise.com/t/kbd-to-filter-play-the-filter-resoance-not-quit-there-yet/44409

------- OopsIFly wrote:

This is presenting some Work on and inspired by the Slider/Knob range translation formula given by Taktik. Also I would like to have this thread as a place for discussing the formulas and keep an collection of others similliar stuff, to have it all together at one place, and to try to keep it some kind of up to date. I’ve also put some obvious (=linear) scalings here that can be useful to keytrack, and this also has the reason to have a collection of knowledge so you can see right away which parameters could be script-mapped and wich are uncertain atm.

One of the most obvious uses of knowing this formula is keytracking all sorts of frequencies. I.e. to “play” a comb filter or autooscillating resonant filter with the keyboard, to shift filter cutoffs relative to the note they filter, transpose eqs by musical intervals, you name it. Great stuff, very powerful.

I will present each formula with some approximated form that wolfram alpha spat out after setting variables and solving for slider 0…1 and one simple copy&paste formula device in a spoiler that can translate from a keytracker to cutoff/frequency of the named devices. Keep in mind this is different to the LFO table solution given here before for the old devices, for example I’ve added a pitch bend knob to the devices, that can bend the pitch by a number of semitones without steps. You could also detune, add an offset and lfo it, whatever you like, be it in Frequency or Note scaling. The device is split in two functions, one to map key and bend to frequency in herz (“kt2frq”) and another to map frequency to cutoff/freq parameter value (“frq2flt”) to allow for easier modifications of the code, it could maybe be optimized by baking a single function out of it. You can also swap out the formula in ftq2flt easily with the ones given in the listings below.

Also the fomula is configurable in the top rows of the lua code. The “fromnote” and “tonote” numbers are for defining the range of the keytracker that is fed into the device. 0-119 is the full range from c-0 to b-9, you can change both to reflect another range. The “bendrange” is the range in semitones that can be bent up or down with the second slider when it is different from 0.5 . The “a4” and “a4f” are for defining a reference note (a-4 by default, number 57, hence the name of the vars) and its frequency (440 hz by default). You can tune with these values, i.e. to pitch up or down one octave quickly try 880.0 or 220.0 for “a4f”.

Please bear in mind that I’m no experienced code, and the functions can have errors and won’t check for bad configurations. Also the formula devices can only trigger frequencies that the device allows within its range. The formula device seems to clamp out of range values automatically. But for example the mixer eq peak is pretty limited in its note range, and this can’t be done better.

Here is a little lua function to translate two strings (like “a-3” or “f#6” or “Bb5”) to their note numbers so you can configure the formula device note ranges with a little more convenience. You can load it in the renoise scripting window/terminal for example, or make a console app out of it:

Click to view contents

function NoteRange(minnote,maxnote)
local NoteVal = { [“c”]=0, [“d”]=2, [“e”]=4, [“f”]=5, [“g”]=7, [“a”]=9, [“b”]=11 }
local SignVal = { [“b”]=-1, ["-"]=0, ["#"]=1 }
local result1 = -1;
local result2 = -1;
if minnote ~= nil then
local MyNote = string.lower(string.sub(minnote,1,1))
if NoteVal[MyNote] ~= nil then
result1 = NoteVal[MyNote]
local MyNote = string.lower(string.sub(minnote,2,2))
if SignVal[MyNote] ~= nil then result1 = result1 + SignVal[MyNote]; end
if tonumber(string.sub(minnote,3,3)) ~= nil then result1 = result1 + 12.0 * tonumber(string.sub(minnote,3,3)) else result1 = -1; end;
end
end
if maxnote ~= nil then
local MyNote = string.lower(string.sub(maxnote,1,1))
if NoteVal[MyNote] ~= nil then
result2 = NoteVal[MyNote]
local MyNote = string.lower(string.sub(maxnote,2,2))
if SignVal[MyNote] ~= nil then result2 = result2 + SignVal[MyNote]; end
if tonumber(string.sub(maxnote,3,3)) ~= nil then result2 = result2 + 12.0 * tonumber(string.sub(maxnote,3,3)) else result2 = -1; end;
end
end
print ( "From: ", result1, " to ", result2 )
end

Of course there can be other stuff done With these formulas. I just thought It’d be nice to present this with something that people without coding background can use straight away and have fun with.

The main formula from taktik and then modified a bit (the maxf and minf part, it’s maximum and minimum freqeuency of the device) by me is “y = ((pow(10, log10(s)x) - 1) / (s - 1))(maxf-minf)+minf” - where y is the frequency in hz and x the slider value. This is the other way round as we would find interesting to use, I guess this is because the formula was for translating a slider value to the number of hz for the parameter value display. What I’ve done was putting in “s” and “maxf/minf” values in, put it into wolfram alpha and let it solve for “x”, then using the real result and the approximation of it. This seems to work quite nice for s=100.

Ok, here we go, the formulas and the devices that use them

1.)

s = 100, min = 40 (?) max = 18000 (?)

AnalogFilterDevice cutoff (Analog Filter in Renoise 3.1, but the diode filter seems off, with cutoff rising when reso increases)

Cutoff/Freq of Moog, Clean, K35, the Diode, Band Pass/Stop, Decimator - filters in Chorus2Device (new chorus in Renoise 3.1)

Most Probably also the Filters in Flanger2Device, (3.1) same as the Chorus (untested).

Also the new 3.1 instrument modulation filters seem to respond by this formula, they are also the same as the Chorus. No real use, because in polyphonic modulation there is no formula device, so frequencies can only be tracked from an effect slot via a macro - only monophonic, no polyphonic action.

Map Hz to Slider: return 0.217147log(0.0000556793(99*inp+14000))

Click to view contents

<?xml version="1.0" encoding="UTF-8"?>

true

true

Init

Bundled Content

true

1.0

Device only

frq2flt(kt2frq(A, B))

– keytracker range:

local fromnote = 0.0

local tonote = 119.0

– bend range in semitones:

local bendrange = 12.0

– reference note and its frequency:

local a4 = 57.0

local a4f = 440.0

local trt = 1.0594630943592952645

function frq2flt (inp)

– put Hz to slider translation formula here

return 0.217147log(0.0000556793(99inp+14000))

end

function kt2frq (inp, bnd)

– maps 0…1 keytracker value with bend to frequency in hz

local nnbr = inp(tonote-fromnote) + fromnote + (bnd-0.5)bendrange2.0

return a4f * pow(trt, nnbr-a4)

end

Key

Bend

-

true

1.0

Device only

0.5

Device only

0.0

Device only

-1

Device only

-1

Device only

-1

Device only

2.)

s = 100, min = 20 (?), max = 20000 (?)

DigitalFilterDevice (Digital Filter in Renoise 3.1)

Theoretically: Phaser2Device (Phase), Floor and Ceiling display correct values, but resonant freq will be very different depending on number of stages, so the mapping is rather useless.

Map Hz to Slider: return 0.217147log(0.00045045(11*inp+2000))

Click to view contents

<?xml version="1.0" encoding="UTF-8"?>

true

true

Init

Bundled Content

true

1.0

Device only

frq2flt(kt2frq(A, B))

– keytracker range:

local fromnote = 0.0

local tonote = 119.0

– bend range in semitones:

local bendrange = 12.0

– reference note and its frequency:

local a4 = 57.0

local a4f = 440.0

local trt = 1.0594630943592952645

function frq2flt (inp)

– put Hz to slider translation formula here

return 0.217147log(0.00045045(11inp+2000))

end

function kt2frq (inp, bnd)

– maps 0…1 keytracker value with bend to frequency in hz

local nnbr = inp(tonote-fromnote) + fromnote + (bnd-0.5)bendrange2.0

return a4f * pow(trt, nnbr-a4)

end

Key

Bend

-

true

1.0

Device only

0.5

Device only

0.0

Device only

-1

Device only

-1

Device only

-1

Device only

Linear Scaling, min = 20, max = 20000

Eq5Device, Eq10Device EQ Points Frequency (EQ5 and EQ10)

Map Hz to Slider: return (inp - 20.0) / (20000.0 - 20.0)

Map Hz to Slider Nodivision: return (inp - 20.0) * 0.00005005

Click to view contents

<?xml version="1.0" encoding="UTF-8"?>

true

true

Init

Bundled Content

true

1.0

Device only

frq2flt(kt2frq(A, B))

– keytracker range:

local fromnote = 0.0

local tonote = 119.0

– bend range in semitones:

local bendrange = 12.0

– reference note and its frequency:

local a4 = 57.0

local a4f = 440.0

local trt = 1.0594630943592952645

function frq2flt (inp)

– put Hz to slider translation formula here

return (inp - 20.0) * 0.00005005

end

function kt2frq (inp, bnd)

– maps 0…1 keytracker value with bend to frequency in hz

local nnbr = inp*(tonote-fromnote) + fromnote + (bnd-0.5)bendrange2.0

return a4f * pow(trt, nnbr-a4)

end

Key

Bend

-

true

1.0

Device only

0.5

Device only

0.0

Device only

-1

Device only

-1

Device only

-1

Device only

Linear Scaling, min = 400, max = 3400

MixerEqDevice “Mid Freq.” (Mixer EQ)

Map Hz to Slider: return (inp - 400.0) / (3400.0 - 400.0)

Click to view contents

<?xml version="1.0" encoding="UTF-8"?>

true

true

Init

Bundled Content

true

1.0

Device only

frq2flt(kt2frq(A, B))

– keytracker range:

local fromnote = 0.0

local tonote = 119.0

– bend range in semitones:

local bendrange = 12.0

– reference note and its frequency:

local a4 = 57.0

local a4f = 440.0

local trt = 1.0594630943592952645

function frq2flt (inp)

– put Hz to slider translation formula here

return (inp - 400.0) / (3400.0 - 400.0)

end

function kt2frq (inp, bnd)

– maps 0…1 keytracker value with bend to frequency in hz

local nnbr = inp*(tonote-fromnote) + fromnote + (bnd-0.5)bendrange2.0

return a4f * pow(trt, nnbr-a4)

end

Key

Bend

-

true

1.0

Device only

0.5

Device only

0.0

Device only

-1

Device only

-1

Device only

-1

Device only

Linear Scaling, min = 10, max = 4999

Reverb3Device “Low Cut” (mpReverb)

Map Hz to Slider: return (inp - 10.0) / (4999.0 - 10.0)

Click to view contents

<?xml version="1.0" encoding="UTF-8"?>

true

true

Init

Bundled Content

true

1.0

Device only

frq2flt(kt2frq(A, B))

– keytracker range:

local fromnote = 0.0

local tonote = 119.0

– bend range in semitones:

local bendrange = 12.0

– reference note and its frequency:

local a4 = 57.0

local a4f = 440.0

local trt = 1.0594630943592952645

function frq2flt (inp)

– put Hz to slider translation formula here

return (inp - 10.0) / (4999.0 - 10.0)

end

function kt2frq (inp, bnd)

– maps 0…1 keytracker value with bend to frequency in hz

local nnbr = inp*(tonote-fromnote) + fromnote + (bnd-0.5)bendrange2.0

return a4f * pow(trt, nnbr-a4)

end

Key

Bend

-

true

1.0

Device only

0.5

Device only

0.0

Device only

-1

Device only

-1

Device only

-1

Device only

Linear Scaling, min = 500(=Off), max = 3000

Reverb3Device “Color” (mpReverb, Will be “off” if the lowest possible value of 0 / 500hz is hit)

Map Hz to Slider: return (inp - 500.0) / (3000.0 - 500.0)

Click to view contents

<?xml version="1.0" encoding="UTF-8"?>

true

true

Init

Bundled Content

true

1.0

Device only

frq2flt(kt2frq(A, B))

– keytracker range:

local fromnote = 0.0

local tonote = 119.0

– bend range in semitones:

local bendrange = 12.0

– reference note and its frequency:

local a4 = 57.0

local a4f = 440.0

local trt = 1.0594630943592952645

function frq2flt (inp)

– put Hz to slider translation formula here

return (inp - 500.0) / (3000.0 - 500.0)

end

function kt2frq (inp, bnd)

– maps 0…1 keytracker value with bend to frequency in hz

local nnbr = inp*(tonote-fromnote) + fromnote + (bnd-0.5)bendrange2.0

return a4f * pow(trt, nnbr-a4)

end

Key

Bend

-

true

1.0

Device only

0.5

Device only

0.0

Device only

-1

Device only

-1

Device only

-1

Device only

7.)

s = 100, min = 0(?), max = 22050(?):

Filter3Device (Filter up to renoise 3, but 4pole shelf will be off and drift when shelf is changed)

FilterDistortionDevice (Scream Filter up to renoise 3)

CrossoverDevice Low & High (Multiband Send)

ExciterDevice Low & High (Exciter)

Map Hz to Slider: return 0.217147log(0.000408163(11*inp+2450))

Click to view contents

<?xml version="1.0" encoding="UTF-8"?>

true

true

Init

Bundled Content

true

1.0

Device only

frq2flt(kt2frq(A, B))

– keytracker range:

local fromnote = 0.0

local tonote = 119.0

– bend range in semitones:

local bendrange = 12.0

– reference note and its frequency:

local a4 = 57.0

local a4f = 440.0

local trt = 1.0594630943592952645

function frq2flt (inp)

– put Hz to slider translation formula here

return 0.217147log(0.000408163(11inp+2450))

end

function kt2frq (inp, bnd)

– maps 0…1 keytracker value with bend to frequency in hz

local nnbr = inp(tonote-fromnote) + fromnote + (bnd-0.5)bendrange2.0

return a4f * pow(trt, nnbr-a4)

end

Key

Bend

-

true

1.0

Device only

0.5

Device only

0.0

Device only

-1

Device only

-1

Device only

-1

Device only

8.)

s = 100, min = 20(?), max = 11020(?)

CombDevice (old comb filter)

Map Hz to Slider: return 0.217147log(9inp+820)-1.5

Click to view contents

<?xml version="1.0" encoding="UTF-8"?>

true

true

Init

Bundled Content

true

1.0

Device only

frq2flt(kt2frq(A, B))

– keytracker range:

local fromnote = 0.0

local tonote = 119.0

– bend range in semitones:

local bendrange = 12.0

– reference note and its frequency:

local a4 = 57.0

local a4f = 440.0

local trt = 1.0594630943592952645

function frq2flt (inp)

– put Hz to slider translation formula here

return 0.217147log(9inp+820)-1.5

end

function kt2frq (inp, bnd)

– maps 0…1 keytracker value with bend to frequency in hz

local nnbr = inp*(tonote-fromnote) + fromnote + (bnd-0.5)bendrange2.0

return a4f * pow(trt, nnbr-a4)

end

Key

Bend

-

true

1.0

Device only

0.5

Device only

0.0

Device only

-1

Device only

-1

Device only

-1

Device only

A Todo List for parameter I (or anyone of you) would like to see in this collection:
Old Chorus filters (like Filter3Device?)
Multitapdelay Filters (seems not the same formula as Filter3Device by the reported slider values…)
flanger + chorus + delay + multitapdelay + mpreverb & convolver pre-delay - delay values could be mapped to freq in hz for comb filter like FX
LoFiDevice + Old Ringmod
test instrument filters (could be mapped via a macro)
Other than direct frequency: value to delay in modulation devices delays, for perfect automated syncing to song speed.
Delay sliders in general for automated syncing purposes
Wishlist for other functions we would like to know from Taktik

(sorry big and red, so meeebie he will have his eyes on this sometime)

Old Chorus filters (like Filter3Device?)
Multitapdelay Filters (seems not the same formula as Filter3Device by the reported slider values…)
Old and New Chorus and Flanger Delay parameter scalings
LoFiDevice real frequency formula (given seems wrong)
old Ringmod real frequency formula (given seems wrong)
The real Phaser reso peaks Frequencies?
Correction Formula for the 3.1 diode filter?

LoFiDevice and the old Ringmod were given by Taktik with the log formula with “s” being 16 for the Lofidevice and 20000 for the old Ringmod - I’ve tested both configurations, and some other I just guessed, but couldn’t find any configuration that hit the frequencies just right.

Treat this as a gift to the community (= public domain), but don’t demand anything from me.

I have taken Time to research and test this, but errors could still be here, be it copy & paste or bad code, please comment here if you find something fishy…

Original Thread Opening Message:

Hi, Taktikmade a nice postclearing up the actual formula involved in translating a frequency to some filter devices cutoffs. Keep in mind the formula given there seems to translate the other way round, and needs to be solved for “x” to be useful for i.e. translating from a keytracker with a formula device. No need to break your head over this, stuff like wolfram alpha can do it automatically for you.

I thought in this thread we could collect the basic and warping formulas for all kinds of sliders of devices, and maybe share formula device setups for warping something useful with them.

Or nag @taktik to give out more formulas, before I make a post with the formulas and setups already known after some testings, I’d like to make a request regarding the formula for EQ5 and EQ10 frequency parameter warpings, the currently known formula won’t fit in.

Edited by OopsIFly, 09 January 2016 - 15:09.

danoise and ffx like this

ffx · May 17, 2016, 8:04pm

------- OopsIFly wrote:

Modulation filtering with the eq devices

Ok, I bragged in excitement in the status stuff about using the native eq devices for modulation filtering. So here comes a first, rather plain example in the form of a doofer.

This is not really the faint of the heart, but is an example of using the equalizers to do modulation filterings. It’s not so very complex - it’s basically some form of multimode filter, allowing setups like lowpass, bandpass, hipass, peaking, and notch to be setup or morphed between each other. It works around using 2 eq peaks and the hi and low shelves for creating these filter types. It somehow has its special sound - not so much like analog simulations, but those eq biquads can do some pretty whacky stuff, too. This version stacks two indentically setup EQ-5s to make a stronger effect and allow greater reductions in the stopbands.

Normally when trying to make music, I’d have used such eq techniques in a kind of “one-purpose-setup”, but for sharing I’ve put a little work into making it more generic configurable. Well…some work went into keeping the gain around the cutoff constant despite of the state of low and hi shelves, but you’ll see it is not so very consistent in terms of ease-of-use when automating it a lot and using the skew and Q parameters dynamically. Also as some diagnostic EQ instead of real filters were used, you’ll have to expect that the high frequencies will pierce a little, there’s no compensation to make it more ear-pleasing at hi cutoffs as you might expect from a true va filter. Today it is very hot here, and I don’t think it’d be such a great thing to make the “perfect allround doofer tuned to max” as it’d suck cpu like mad from all the formula devices, I’d say.

Ok, here the 8 parameter’s descriptions for use:

“Cutoff” - Yay, automate this like mad, please…

“Q” - The shape of the Peaks/Notches around cutoff. Low is wide, hi is narrow, high enough can make resonance-style action, but not so much self-oscillation but rather massive ringing with the input.

“Peak” - centered around 50%. The strength of the peak around the cutoff frequency. Higher than 50 will boost as allowed by the “TakeCare!” parameter, lower will create a notch relative to the level of low and high gains. Note this parameter tries to keep the cutoff-stuff at vaguely the same gain regardless of the settings of the “Low” and “High” parameters

“Low” and “High” - these will control the gain below and above the cutoff. No boosting possible, but gain reduction to up to -40db. I don’t need to say, if “Low” is up to 100, and “High” down to zero, you’ll have a lowpass with some cutoff attenuation or boosting depending on the “Peak” parameter.

“Skew” - this is a little speciality of this filter. It can shift the frequency response around the cutoff a little, and the effects depend on the other parameters. For example in a “bandpass” setup, low skew will put extra pronounciation on the freqs below cutoff, high skew will let high freqs shine through stronger. Note this might produce extra (maybe unwanted) gain when off the neutral position of 50%.

“Lag” - simple linear interpolation of cutoff changes. The number in percent should be equal to the number in ticks for the filter to reach the next value. Don’t beat me if the algorithm isn’t perfect, the form.device sucks in some stuff, but that is another story.

“TakeCare!” - this is named deliberatly this way, it will enable boosting the cutoff-peak from zero to up to +40db at 100%. Yea, this will be loud and shrill. I use such to overload distortions and such, and with heavy limiters. Take care.

Now I’ve added a limiter and a mild distortion to the end of the chain, to make sure noone blasts ears or speakers too much before understanding what this baby does. Remove if you know what you’re doing. In the doofer you can also see the second stage EQ graph as frequency response, but mind the response is double, there’s an identical EQ in the chain going on doubling the effect.

Copy&Paste doofer:

Click to view contents

<?xml version="1.0" encoding="UTF-8"?>

EQMulti

true

true

EQPB3

false

1.0

Device only

50

Device only

Cutoff

3

1

0.0

1.0

Linear

50

Device only

Q

5

12

1.0

0.0

Linear

5

13

1.0

0.0

Linear

6

12

1.0

0.0

Linear

6

13

1.0

0.0

Linear

50

Device only

Peak

1

1

0.0

1.0

Linear

100

Device only

Low

5

1

0.0

0.5

Linear

0

1

0.0

1.0

Linear

6

1

0.0

0.5

Linear

100

Device only

High

5

5

0.0

0.5

Linear

0

2

0.0

1.0

Linear

6

5

0.0

0.5

Linear

50

Device only

Skew

5

11

0.0

1.0

Linear

5

15

1.0

0.0

Linear

6

11

0.0

1.0

Linear

6

15

1.0

0.0

Linear

12

Device only

Lag

3

2

0.0

1.0

Linear

0.0

Device only

TakeCare!

1

3

0.0

1.0

Linear

8

true

Init

true

LHFac

false

false

#Init

true

1.0

Device only

lhfac(A, B)

function lhfac(Ls,Hs)

return 1.0-(Ls+Hs)0.5

end

L

H

-

true

1

1.0

Device only

1.0

Device only

0.0

Device only

-1

Device only

1.0

Device only

2

Device only

PeakComp1

false

false

#Init

true

1.0

Device only

peak_comp(A,B,C)

function peak_comp (Pk,LHf,Max)

local Maxdb = Max20.0

local LHfdb = LHf*20.0

if(Pk>0.5) then

Pk = (Pk-0.5)2.0

local Pkdb = PkMaxdb + LHfdb

Pk = (Pkdb/40.0)*0.5 + 0.5

else

Pk = (0.5-Pk)2.0

local Pkdb = Pk20.0 - LHfdb

Pk = 0.5-(Pkdb/40.0)*0.5

end

return Pk

end

P

LH

Max

true

1

0.5

Device only

0.0

Device only

0.0

Device only

-1

Device only

2

Device only

1.0

Device only

Peak

false

false

Init

true

1.0

Device only

1

0.5

Mixer and Device

-1

Device only

5

Device only

2

Device only

0.0

Device only

1.0

Device only

Linear

-1

Device only

5

Device only

3

Device only

0.0

Device only

1.0

Device only

Linear

-1

Device only

6

Device only

2

Device only

0.0

Device only

1.0

Device only

Linear

-1

Device only

6

Device only

3

Device only

0.0

Device only

1.0

Device only

Linear

-1

Device only

-1

Device only

-1

Device only

0.0

Device only

1.0

Device only

Linear

-1

Device only

-1

Device only

-1

Device only

0.0

Device only

1.0

Device only

Linear

-1

Device only

-1

Device only

-1

Device only

0.0

Device only

1.0

Device only

Linear

-1

Device only

-1

Device only

-1

Device only

0.0

Device only

1.0

Device only

Linear

-1

Device only

-1

Device only

-1

Device only

0.0

Device only

1.0

Device only

Linear

FreqSmth

false

true

linsmooth

true

1.0

Device only

inertia_lin(A, B)

glideticksscaler = 100.0

from = 0.5

target = 0.5

current = 0.5

last = 0.5

addstp = 0.0

curticks = 1.0

ticksdone = 0.0

function inertia_lin (input, ticks)

if (ticks ~= curticks) then

curticks = ticks * glideticksscaler

end

if (input ~= last) then

target = input

last = input

from = current

addstp = (target - current) / curticks

end

if (addstp ~= 0.0) then

current = current + addstp

if (addstp > 0.0) then

if (current > target) then

addstp = 0.0

current = target

end

end

if (addstp < 0.0) then

if (current < target) then

addstp = 0.0

current = target

end

end

end

return current

end

Input

Inertia

_

true

1

0.5

Device only

0.119999997

Device only

0.0

Device only

-1

Device only

4

Device only

1.0

Device only

Freq

false

false

Init

true

1.0

Device only

2

0.5

Mixer and Device

-1

Device only

5

Device only

7

Device only

0.0

Device only

0.4994995

Device only

Exp Fast

-1

Device only

5

Device only

8

Device only

0.0

Device only

0.4994995

Device only

Exp Fast

-1

Device only

5

Device only

6

Device only

0.0

Device only

0.4994995

Device only

Exp Fast

-1

Device only

5

Device only

10

Device only

0.0

Device only

0.4994995

Device only

Exp Fast

-1

Device only

6

Device only

7

Device only

0.0

Device only

0.4994995

Device only

Exp Fast

-1

Device only

6

Device only

8

Device only

0.0

Device only

0.4994995

Device only

Exp Fast

-1

Device only

6

Device only

6

Device only

0.0

Device only

0.4994995

Device only

Exp Fast

-1

Device only

6

Device only

10

Device only

0.0

Device only

0.4994995

Device only

Exp Fast

-1

Device only

-1

Device only

-1

Device only

0.0

Device only

1.0

Device only

Linear

EQst1

false

false

Init

true

1.0

Device only

L+R

20

0.0

Device only

0.0

Device only

0.0

Device only

0.0

Device only

0.0

Device only

685.333374

Device only

685.333374

Device only

685.333374

Device only

20000

Device only

685.333374

Device only

0.500050008

Device only

2.00005007

Device only

2.00005007

Device only

4

Device only

0.500050008

Device only

Graph Only

EQst2

true

false

Init

true

1.0

Device only

L+R

20

0.0

Device only

0.0

Device only

0.0

Device only

0.0

Device only

0.0

Device only

685.333374

Device only

685.333374

Device only

685.333374

Device only

20000

Device only

685.333374

Device only

0.500050008

Device only

2.00005007

Device only

2.00005007

Device only

4

Device only

0.500050008

Device only

Graph Only

true

false

Init

true

1.0

Device only

13.9280224

Device only

16

Device only

0.199990004

Device only

100

Device only

1.23253775

Mixer and Device

true

false

Init

true

1.0

Device only

1.0

Device only

0.0599999949

Device only

0.5

Device only

1.0

Device only

0.0

Device only

I don’t take any responsibility of damages to whatever caused by using it - use at your own risk.

Next time I’m bored and find no approaches to actual music, I might make another doofer with more complex stuff. This one is way simple. You can do vowel like stuff, or other mad shit with automating EQs. Happy experimenting!

ffx · August 14, 2021, 7:40pm

Sadly all code contents were ripped off by the forum filters… Any way to get the original posts?