What is your favorite screamy filter? (from a vst)

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hey aciddose, im not as hopeless/helpless as you think i am. ;) Also, you shouldn't take my threads too seriously, this is mostly for fun.

I don't care to defend myself, but I can't leave the thread like this.

I have a good idea of how to do what I want to do based on experiments with VSTs, Jesusonic, general knowledge gained through years of using analog and digital synthesizers. I don't know math and theories, but that's where paying a programmer comes in. What I do know is the behaviors and the waveshapes. I'm not concerned with harmonics. Harmonics are a result of waveshape. To think of harmonics first is backwards.
aciddose wrote:Try posting in the dsp forum with a clip of your filter:
"What makes it difficult to reproduce the harmonics generated as in this recording of the ________ VCF from my _________ synthesizer?"
I'm not going to do that because I already have my answer. As a matter of fact I will soon post the said recording and details of how I intend to make it work digitally. I will post the details of my study of analog filters.

Edit: aciddose, you are too smart for your own good, and that seems to make you obsessed with telling me how incapable I am. If I am really incapable (which is a possibility!), I'll let myself know! :tu:

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What I understand from the OP is that he/she wants to code an emulation of a specific or generic analog filter but with many other options.

I don't know dsp coding but I know that analog synths have huge differences in everything including the filter. So maybe the first step is specifying which exact filter to base the design on. Then by applying different 'randomness' functions in many objects, you can emulate the non perfection and instability of some analog things.

The flexibility should come after finishing the main filter/synth basic functions. For example, Harmor has a very flexible design that you can draw your own filter.

I agree that it's a big project and the dsp forum might help the OP in this complex task more than just taking general opinions from users other than developers.
Using: Cubase Pro 15, Reason 13, Tascam US-4x4HR, MODX6, DM12D, LaunchKey 49, Yamaha guitar(Pacifica 612v) and bass (BB234) and some virtual instruments and synths.

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BTW, helpful tip.

I'd be very, very careful who you pay and what the terms of your agreement are. You could easily lose everything.

... everything as in, whatever you pay expecting to get the project finished in a month.

You're thinking in the 50k - 80k range yearly, right? Multiple years?
Free plug-ins for Windows, MacOS and Linux. Xhip Synthesizer v8.0 and Xhip Effects Bundle v6.7.
The coder's credo: We believe our work is neither clever nor difficult; it is done because we thought it would be easy.
Work less; get more done.

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EnGee you are close. I don't want to code a specific analog filter emulation, I want to code what could produce the sound of many filters by giving the user many parameters to work with. Just as a side note, I'm not interested in analog instability and other imperfections.. well... then again, a little bit of random could possibly exaggerate any chaotic effects, and it'd be good to give users options for some randomness...

...rather... (instead of analog instability and imperfections) I'm interested in recreating the sound in its unadulterated state, the core sound and the core behaviors. Then sure, I could throw in some randomness options.

aciddose I don't intend to keep details to myself because who knows, someone in the community might be interested in helping. I don't intend to make a living with this. Between paying a programmer and income, I'd like just to break even. I'll post specific details soon, just give me some time!
Last edited by Architeuthis on Tue Jul 14, 2015 11:20 pm, edited 2 times in total.

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Architeuthis wrote:... I don't know math and theories, but ... To think of harmonics first is backwards.
Who is backward? :scared:
Free plug-ins for Windows, MacOS and Linux. Xhip Synthesizer v8.0 and Xhip Effects Bundle v6.7.
The coder's credo: We believe our work is neither clever nor difficult; it is done because we thought it would be easy.
Work less; get more done.

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https://en.wikiquote.org/wiki/Ideas

See some of these, such as:
Howard H. Aiken wrote:Don't worry about people stealing an idea. If it's original, you will have to ram it down their throats.
Or
Sergey Brin wrote:It’s important not to overstate the benefits of ideas. Quite frankly, I know it’s kind of a romantic notion that you’re just going to have this one brilliant idea and then everything is going to be great. But the fact is that coming up with an idea is the least important part of creating something great. It has to be the right idea and have good taste, but the execution and delivery are what’s key.
Free plug-ins for Windows, MacOS and Linux. Xhip Synthesizer v8.0 and Xhip Effects Bundle v6.7.
The coder's credo: We believe our work is neither clever nor difficult; it is done because we thought it would be easy.
Work less; get more done.

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Architeuthis wrote:EnGee you are close. I don't want to code a specific analog filter emulation, I want to code what could produce the sound of many filters by giving the user many parameters to work with. Just as a side note, I'm not interested in analog instability and other imperfections.. well... then again, a little bit of random could possibly exaggerate any chaotic effects, and it'd be good to give users options for some randomness...
Have you looked into Harmor? What you want might be already there:

Architeuthis wrote: ...rather... (instead of analog instability and imperfections) I'm interested in recreating the sound in its unadulterated state, the core sound and the core behaviors. Then sure, I could throw in some randomness options.
I believe that the 'analog' sound is all about imperfections and instabilities :hihi:

Anyway, I just checked FilterShaper by Cableguys and it is really wonderful. I'm thinking of buying it really 8) :


I also like an 'easy' functions that hides a complex algos, like in Codex Filter which can be modulated by OSC1 (FM), so my understanding is that OSC1 is the modulator while the Filter becomes the carrier. Interesting :)

Well, in the end there are 'cheaper' options to get what you want (OP) and I mean by 'cheaper' is by not being concerned by coding/hiring a coder ..etc (cheaper in cost including time/design/money ..etc). So, maybe something like Reaktor or Max for live can be a great platform for designing and using the many components that already there ;)
Using: Cubase Pro 15, Reason 13, Tascam US-4x4HR, MODX6, DM12D, LaunchKey 49, Yamaha guitar(Pacifica 612v) and bass (BB234) and some virtual instruments and synths.

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jbuonacc wrote:off the top of my head, i'd say that Native Instruments 'Driver' is probably the "screamiest" filter (while also one of the "creamiest") i've ever used.
Agreed, Driver is my favorite distortion plug!
[Insert Signature Here]

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ohmicide.
is tops : )
and Permt8 does some whacky stuff to...
distorted but still musical...

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One of my favorites would have to be Saurus

https://soundcloud.com/examigan/a-bit-of-scream

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Saurus sounds like a linear filter + modulation into a waveshaper of some sort rather than a non-linear filter.

In other words you should be able to get exactly the same result with any similar linear filter in combination with an external waveshaping plugin.

Here is an example of a non-linear filter. This has absolutely no modulation and the signal path after the filter is completely linear. No distortion or waveshaping effects, EQ or otherwise.

https://soundcloud.com/aciddose-1/pulse-snap-high-resb
https://soundcloud.com/aciddose-1/part-four
https://soundcloud.com/aciddose-1/wahw2
https://soundcloud.com/aciddose-1/ressarawr
https://soundcloud.com/aciddose-1/monkeybass
https://soundcloud.com/aciddose-1/cant- ... with-these

You should clearly be able to hear the difference between where in your example it has the distinct effect of a waveshaper applied to the signal post, while in these examples the effect is actually applied to the rate of charge of the individual stages inside the filter. The signal itself passes without any modification of harmonics other than resulting from the filter. If the cutoff were set to maximum the effect on the input signal would be near zero.

https://soundcloud.com/aciddose-1/pulxor

These sort of effects are very difficult (impossible) to produce in software without aliasing or extreme processing cost.
In your example if you were to play two sines at different frequencies you'd definitely get some sidebands / intermodulation distortion. This tells you this effect is not due to nonlinear charge but due to direct "zero order" non-linearity of the signal path.
Free plug-ins for Windows, MacOS and Linux. Xhip Synthesizer v8.0 and Xhip Effects Bundle v6.7.
The coder's credo: We believe our work is neither clever nor difficult; it is done because we thought it would be easy.
Work less; get more done.

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very nice demos aciddose, btw, in terms of making a fool of myself, I've come to learn it's difficult not to, so it doesn't bother me anymore. :)

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I wouldn't say it's making a fool of anyone to have ideas, thoughts or questions. I think in your case you probably have some interesting ideas and perhaps if you put some effort toward implementing them somewhere such as reaktor you'd get a combination of interesting results and some lessons learned.

So, you should definitely look toward doing that.

In the examples you gave earlier it sounded to me as if you were generating wavelets (perhaps by input of pulses to a filter?) then waveshaping them. This of course (if it is the case) has the same issues as passing any signal through a filter followed by a waveshaper, it won't produce the same or even approximately the same effect.

You also need to take into account that the pulses you generate are most likely already anti-aliased by some method such as sinc, blit or blep or so on. The filter you use is most likely anti-aliased (if non-linear) by oversampling and FIR filters (like blit, blep or etc... the same thing.)

The waveshaper is also likely anti-aliased by oversampling, while the function is likely selected to generate a limited number of harmonics or adjusted to prevent excessive aliasing even before the filter. For example x^2 can produce at most a single harmonic, double the frequency of the input. This means that you need only 2x oversampling to prevent reflected aliases.

It is just very difficult to discuss any of this with someone who claims they don't know anything about math or theory or says that the discussion in the DSP forum is confusing for them. I was once there myself, in fact I still am! I have no idea what I'm doing when it comes to DSP. You admit this yourself, at least partially, so you're part way there already.

Even the so called "experts" are in the same boat we are. The thing is... once you gain knowledge it only reveals how much larger the field really is than you had ever previously anticipated.

The only foolish thing is to think that the path you need to take to succeed is really so much different from the path all the rest of us have been on for years. If you want to learn there are a lot of people out there who can help you. If you want to be the isolated genius however you may end up finding that you've wasted an awful lot of time in the end.
Free plug-ins for Windows, MacOS and Linux. Xhip Synthesizer v8.0 and Xhip Effects Bundle v6.7.
The coder's credo: We believe our work is neither clever nor difficult; it is done because we thought it would be easy.
Work less; get more done.

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The Glue, The Drop, The Scream - www.cytomic.com

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I like Philteroid for some wicked Filter stuff.

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Underground Music Production: Sound Design, Machine Funk, High Tech Soul

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