LTspice - BigMuff sim

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@Chris and Ivan:

The Z3=0.82702295; did the trick, :clap:

Waiting for the HQ mode now :wink:

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I've translated it to code that runs in APE. Is anyone interested?

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Hi,
what is APE?

Chris

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It's an environment for writing plugin code in C. It's available as a VST and AU, and the thread is in this same forum actually: http://www.kvraudio.com/forum/viewtopic ... 3&t=402054

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FLWrd wrote:I've translated it to code that runs in APE. Is anyone interested?
I would love to see it :) Post - or perhaps create a pull request: https://bitbucket.org/Mayae/ape-snippets

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Mayae wrote:
FLWrd wrote:I've translated it to code that runs in APE. Is anyone interested?
I would love to see it :) Post - or perhaps create a pull request: https://bitbucket.org/Mayae/ape-snippets
Here it is. I've included the link to this page and the original code. Have fun.
wuff.c.zip
BTW: APE doesn't like to save presets in Logic X. When I open them, the original limiter.c template is there instead. It does offer to replace "template.c" whenever I close the project, though.

Edit: z3 is now initialized properly.
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Last edited by FLWrd on Sun Feb 07, 2016 9:25 pm, edited 1 time in total.

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FLWrd wrote:
Mayae wrote:
FLWrd wrote:I've translated it to code that runs in APE. Is anyone interested?
I would love to see it :) Post - or perhaps create a pull request: https://bitbucket.org/Mayae/ape-snippets
Here it is. I've included the link to this page and the original code. Have fun.
wuff.c.zip
BTW: APE doesn't like to save presets in Logic X. When I open them, the original limiter.c template is there instead. It does offer to replace "template.c" whenever I close the project, though.
Cool thanks, i like it :). Do you+OP mind if I include it in the repository?
I'll take a look at the preset handling in Logic, not sure how it's handled. I can see it's buggy in Live, as well.

e: as per the click when starting, you set the z7 state to 0.827 and not z3. quick fix (also, IMO, including all digits of 0.82702295 completely removes it).

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Damn, I forgot to zip the correct version. I'll reupload. Thanks for pointing it out.

I don't mind if you include it. The rest is up to Chris-S.

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Hi,
I'd like to try converting some simple audio circuits to C++ using this technique.
Are there any good tutorials on LTSpice -> C++?
Chris - can you outline the steps you took?

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Chris-S wrote:I'm not sure if it's worth the effort to include some HQ (oversampling option).

It's clear that the distortion introduces some aliasing, but is the human ear able to notice those aliased harmonics between all the non-aliased harmonics?

Chris
Guitar playing involves a lot of pitch bending and vibrato. That's where you'll hear it—the harmonics bending down as you bend the note up.

But the good news is that while greater distortion yields greater aliasing, the in increase in distortion masks it to some degree. So you don't need as high of an oversampling factor as it would seem just looking at aliasing magnitude.
My audio DSP blog: earlevel.com

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gleeom wrote: Are there any good tutorials on LTSpice -> C++?
Chris - can you outline the steps you took?
It's not a one-to-one implementation, of course.
I simply analysed the single stages and converted the responses to kubic-spline-modelled waveshapers.

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FLWrd wrote:I don't mind if you include it. The rest is up to Chris-S.
It's freeware, so it's fine to convert and enhance.

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Chris-S wrote:
gleeom wrote: Are there any good tutorials on LTSpice -> C++?
Chris - can you outline the steps you took?
It's not a one-to-one implementation, of course.
I simply analysed the single stages and converted the responses to kubic-spline-modelled waveshapers.
Do you mean similar methods shown at https://github.com/ValdemarOrn/SharpSoundPlugins ?

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Chris-S wrote:
gleeom wrote: Are there any good tutorials on LTSpice -> C++?
Chris - can you outline the steps you took?
It's not a one-to-one implementation, of course.
I simply analysed the single stages and converted the responses to kubic-spline-modelled waveshapers.
Actually, you could squiz the two wave shapers in just one.
You don't have the time dependency, so it's easier to do and only requires to compute the coefficients as if the condensators are disconnected. The sound may not be the same, but who would notice the difference?

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Miles1981 wrote:Actually, you could squiz the two wave shapers in just one.
I'm not sure, waveshaping is a non linear operation.

So I don't think that F1->WS1->F2->WS2->F3->WS3->F4 is replacable by F->WS->F.
(WS=Waveshaper, F=Filter).

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