MTurbo Filter discussion

[Chandlerhimself]

Has anyone else played with this yet? I'm enjoying it, but I'm hoping other people have some insights on programming it and creating some interesting filters for it. I've messed around with it a little and I tried to create a filter that decreases in resonance as the frequency drops. I'm not super happy with it, but its a start.

Code: Select all

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Anyone else have any comments or observations about MTurbofilter.

[vectorwarrior]

Yeah I'm really looking forward to this plug, but without more scripting info it's super hard to dig into. I'm not one for code scripting personally so I'm not sure how much I'll actually design in it, but I'm excited by the possibility regardless.

[n9research]

Algorithm entry via keyboard (UK layout) is totally messed up when using FL Studio as the host, but not Reaper, wierdly...

Non-alphanumeric characters are all in the wrong place, and the delete key types a period.. But it only happens in FLS

I don't have any other plugins with keyboard entry problems on this host - even Melda plugins!

[jmg8]

Has anyone else played with this yet? I'm enjoying it, but I'm hoping other people have some insights on programming it and creating some interesting filters for it. I've messed around with it a little and I tried to create a filter that decreases in resonance as the frequency drops. I'm not super happy with it, but its a start.

Code: Select all

$eNqVlltz2jgUx9-5FBrxkuymgHwBZ9Z2JyGmm9kQKNC0+yjsA2gqS1SSadiH-ew7tkOgXL0vDJZ+-3PT0cX-+JpytAKlmRQBJo0WRiBimTAxD3BmZh88-DGs+f2xzETSo7GRaq3BGCbmGtUDTDCqf+JySrme0Gnx3c2UAmGGCjSYx4cAP8CMZtxgFCXMSDVm-0CASadFbtquszsaUb0OMHGIfdNp32L05+cvS03TJS+iaWH0NBovOTMGCpwEmFiu0-BsQhzP25+2AmyT2wZpe55LyK2LUZ++PvUG91zG38sgbAujl03ynmtZloNDf5xNhzybM-GWb+iXn+Nf8z5qbQwcYgNJqQhw68bFG73e-EG9jPNnmkKABzpmnFMjFSpM6u+Pokh1fxqjb8X43-nviYjqXwTTUkSCTjl0Jed0qSE5GWzojxd0CWg8iYbDUTSOJoWHFeVZvkA2boZ+81dH7wNHUplkaip7jBsocjkY3MnAPZnBHZ9LxcwiDbDF5c8l1fpq9sfVj9-j699mLXI1u77OKUG5nJeK7oIqGhtQAf7QahDXwqj+OcCthtf2TuXdHzzUe6grM2EKIw+wNItcky--qLUzo1XcCnBfJhmnhkmBfi4AOOp266QoT2Ep9LdxD6kAXvZ+nwo6B3W4IboLKhIO6l8aswRNmQE0K0t0tuLN9zY6ZA56trbTxt9OVPtYcY508Lbf9NH4dh3V-EisSOhPFKSSy2MNltsooIK1qrBWydpVWLtknSqsU7JuFdYt2XYVtl2ynSpsp2S9KqwX1vzytI3E6qJgS9b8p96AoCcQ87zP81XeowvCukjYFwnnIuFeJNoXic5FwjtPjKhIZFo0KhV6JlVabu-o+WUDekURN+BGYlWVWO8Su6rEfpc4VSV5B-8Fax1TDhWy2aJbmVVdluf0AlzGzFR2ucfvGbD+p4HdCPbR2tZUftbqTME5hJl4cQ8iOcP0ZfK1OO9PIxOWThV0nDPIvQJqFi3rDBK9LvOImRSEnMGepYHBbHZIfOIsgfIcOJi7iw1bQXn7oM2rYKIAits4f33ll0Hod6mBuVTrRwPpo5jJZui-3VyHqvJyz5fnblhazldq11PuWC3RiBrIt2AeSB8MKG2F-iAzffTA9JLTdZ++sjRL386srywxi37oj2WmYkBDBeXF+PYAzJ1skObGXM1vHnue6rD2H9vKahc=

Anyone else have any comments or observations about MTurbofilter.

I'm not sure how you made that. I really wish I was better at maths, haha.
I was hoping there would be a way to link the resonance to the frequency (so that the resonance would be less for lower frequencies) but I couldnt figure it out. I will have to analyse your syntax to work it out. Can you shed any light on this?

[werzel]

I too am not the most mathematic of persons.
Would it be possible to have a graphic region where we could draw a shape and have it converted to the correct syntax?
How would one create an organic filter that morphs between states? I'm guessing that the answer will be to use the A-D presets and do the morphing there, but I'd like to watch the shape change (sometimes).
Could we have some presets from MFilter and MWobbler imported to Turbo with the syntax revealed so that we might use them as educational tools?

[Chandlerhimself]

I was hoping someone else here was better at math than me. If I'm the one trying to impart math knowledge we're in trouble . Anyway I'll try to explain what I did in that preset.

The syntax is - 2lowpass(f;(q+c)*f01(f))

lowpass is a 12db/oct filter and by putting 2 in front I'm muliplying it by 2 making it a 24db/oct filter

f is the frequency, so the cutoff will follow the frequency exactly

(q+c)*f01(f) this is controling the q(resonance) value. I added q and c(character), so I could adjust the effect, but now I realize this is perhaps a mistake, I should just use q.
f01(f) takes the frequency and converts it to a value between 0 and 1, so it will be compatible with the q value.

If the q value knob is 50%(0.5) and the frequency is at the top (f01(f)=1) then the resonance will be equal to 50%(0.5).

If the q value knob is 50%(0.5) and the frequency is near the middle (f01(f)=0.5) then the resonance will be equal to 25%.

Here is another example. I took out the "c" value. I want to use "c" to control how fast the q gets cut as the frequency lowers, but I need to think about it more. Anyway this shows the difference between the normal algorithm and the one with the decreasing resonance. Use the ratio control to A/B them. The difference isn't subtle IMO.

Code: Select all

$eNrtmFtz4jYUgN-5FRrlZbezAVvYmExtdrIJbDMNCwWa3T4K+wCaFRKV5TTpQ397R1IAJ9y8HR77wiD5O3ddjh1-fFpy9AgqZ1Ik2K97GIFIZcbEPMGFnl228cdOLe6PZSGyHk21VM85aM3EPEcXCfYxuvjM5ZTyfEKndnxTKAVCDxXkoO9uE3wLM1pwjVE3Y1qqMfsbEuxHnv+hFQbl2S7NnxN8FfofQuJhdM+WTIOySu9H4xVnWoMF-QT7vhfUr0Jy1Wpf7TwnCSZtv+4Fkdf0vXaEUZ8+3fcGn7hMvzv7TYJRv+CaDamiS9Cg8hvJOV3lkCXYw+hhnZR2SAgJcCceF9MhL+ZMvOShE7vh+HU+9poaA4dUQ+YkEux9iPBaPl--Qb2C8y90CQke5CnjnGqpkFWZf78T1q23jzH6Zuf-ML8HPLr4XbBciq6gUw6lKA8424nHC7oCNJ50h8NRd9ydWAuPlBemcE3c6MSN14Y2E3tCmRRqKnuMa7Cx7EyWIggORnDN51IxvVgmmHD514rm+bvZz3-+NPP8d7P37w0hKJdzR-cSfEnqJDCLkzIxKHSCL8N6M2phdDFkOl30ZQYJvpEi11RojC5+S7BXj0JyKCX9we1FD93IQmhr4xZWemFkjM8jr-QkV6mX4K54RL7NlJXsxNsQhlQAd9ujTwWdg9rdMzcLKjIO6h+asgxNmQY0c9n678knGN2JVaHvRAZPns25WVg2O2RvYQaFLvG+rZT-45X6vzoHqjOimkmzKbZ18a2XdjzWQLn-sk3dAdBc-7XPSmWzml4VzE-wpV-aW9HBilnZl+Wwm9xGKYDG5sDaDXLndKyVDsxvB2zvM7jnrNyebPneBJcN1eKuePTRLaT02a6AyGtHPvG9VtTGbtoebwm+9OrE2PtFcnPk1z3SDEjYwmhc5Jpay7gTTxQsJZf7jkTjizHmbJIqLHFsswrbdGxQhQ0cG1ZhQ8e2qrAtx0ZV2Mix7Spsu1OLXd-QFY8nBbZkLb7vDXx0D2JuNrhZLW9oS5CTRPMkEZwkwpNE6yQRnSTax4kRFZlc+iaHVOQzqZZmOwvU-fKwBts2iWtwLUKqipCNSLOqSHMjElQVMSv4V3jOU8qhQjRbdCtGqouZmB6Ay5Tpyibf8G8UkB9UUPbgLVrbqjKXTl4oOIaYK-MTiOwI05fZ1wUAP4JM2HKqIAqOIJ8UUL3wyBGk+7QyHjMpfP8I9kVqGMxmu8RnzjJw58DOs+tUs0dw1zAqD9yNIdXr3nY9migAe72Z94txqgBE-23D77v3EPOnE99QDXOpnu80LO-ETDY68UsLsKvTXb+mvNdD54ypdNk547haoRHVYLawCcS+c9zL9DtyPXnm7G5eRWyLMKJiDqbTQpv5De2VJp0HdohC25yfQY93Hj3B1Xn0NM8UVzM4k57wTHpaZ9JzpjwH56q7a64369x8QrBbjnTiQaH76JblK06f+-SJLYvlyz3-lWV60e-EY1moFNBQgeuq7ccI1wmskcZaXS1u7Ps4kXdq-wI2eFdA

Hopefully some other people with better math can help us here and also people who are familar with the subtle details of analog filters will chime in.

[mevla]

Just tried it on a guitar track - sounds pretty good so far.

Any drawback at using it right now for a permanent track configuration ? Will further updates change its current use on tracks today ?

[jmg8]

OK, so does anyone know how to do this?:

I would like 3 lowpass filters in series.
First with full range 20hz - 20000hz
Second with range 200hz - 2000hz
Third with range 2000hz - 20000hz

With this technique, I am hoping to achieve a filter that has a steeper slope at higher frequencies and a lower slope at lower frequencies.

I am not sure how to limit the range of each filter.

[drewfx]

With this technique, I am hoping to achieve a filter that has a steeper slope at higher frequencies and a lower slope at lower frequencies.

It might be easier to just run them in parallel and scale them each by frequency using LevelC:

Parallel[ Serial[2lowpass(f;q); LevelC(f01(f))];
Serial[lowpass6(f;q); LevelC(1-f01(f))] ]; LevelC(2)

The LevelC(2) at the end adds +6dB of gain to match a "normal" filter's level.

[werzel]

Thanks for both of your examples and explanations Chandler. Interesting stuff indeed.
The only thing that I've tried so far was to change the (q+c) in your original filter to read as (q-c) and then modulate a few things in order to focus on what was happening to the sound.

drewfx, be careful there, answers like that will only get you more questions! Got any more ideas?

[werzel]

I caused my first MSF crash today while messing with the Filter module!
While editing the syntax of Chandler's filter 2lowpass(f;(q+c)*f01(f)) I was trying to type '2.5' to replace the 2 in front of the lowpass. I missed the decimal!
If you try to repeat this please do not do it with headphones on and do everything that you can think of to control the volume!
The crash occurred when I added a Distortion to the FX and began twisting knobs to find interesting textures. (Note: I place all Distortion parameters at 0, not their default settings, before adjusting anything.)

[drewfx]

drewfx, be careful there, answers like that will only get you more questions! Got any more ideas?

Who me?

How about a primitive formant filter that morphs vowels with character? Set Pitch mode on the filter to constant for this one, feed it some noise along with an osc and watch out for levels when adjusting character. Needs a little work, but:

Parallel[
bp(f*300/440 + (c<0 ? (c*-90) : (c*-30));q);
Serial[bp(f*870/440 + (c<0 ? (c*-1120) : (c*1420));q);Level(-16 + (c<0 ? (c*4) : (c*-5)))];
Serial[bp(f*2240/440 + (c<0 ? (c*-310) : (c*770));q);Level(-40 + (c<0 ? (c*-16) : (c*15)))]
];Level(q*20)

[EDIT: updated filter]

[drewfx]

I was hoping someone else here was better at math than me. If I'm the one trying to impart math knowledge we're in trouble . Anyway I'll try to explain what I did in that preset.

The syntax is - 2lowpass(f;(q+c)*f01(f))

lowpass is a 12db/oct filter and by putting 2 in front I'm muliplying it by 2 making it a 24db/oct filter

f is the frequency, so the cutoff will follow the frequency exactly

(q+c)*f01(f) this is controling the q(resonance) value. I added q and c(character), so I could adjust the effect, but now I realize this is perhaps a mistake, I should just use q.
f01(f) takes the frequency and converts it to a value between 0 and 1, so it will be compatible with the q value.

If the q value knob is 50%(0.5) and the frequency is at the top (f01(f)=1) then the resonance will be equal to 50%(0.5).

If the q value knob is 50%(0.5) and the frequency is near the middle (f01(f)=0.5) then the resonance will be equal to 25%.

Here is another example. I took out the "c" value. I want to use "c" to control how fast the q gets cut as the frequency lowers, but I need to think about it more. Anyway this shows the difference between the normal algorithm and the one with the decreasing resonance. Use the ratio control to A/B them. The difference isn't subtle IMO.

Code: Select all

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Hopefully some other people with better math can help us here and also people who are familar with the subtle details of analog filters will chime in.

Try this. First the simple version for c=0 (12 o'clock) to c=1 (which is easier to understand, but you probably want the corrected version for the full range of c):

2lowpass(f;q*(1-c)+q*c*f01(f))

Now the corrected version for the full range of c (-1..1):

2lowpass(f;q*(1-(c+1)/2)+q*((c+1)/2)*f01(f))

Let me know what you think.

[drewfx]

And if you want to make the resonance ramp up a little earlier:

2lowpass(f;q*(1-(c+1)/2)+q*((c+1)/2)*sqrt(f01(f)))

[jmg8]

So, does anyone know if it is possible to set min and max values for the frequency?
If so, how would it be written in the syntax?
Lowpass(fmin200max2000)

I noticed in the help text, there is a reference to min and max but not yet sure how to use them in the syntax.