If you fancy playing with C++ this is the function and I don't think it would be difficult at all to modify.

String ScriptingApi::Engine::getCurrentUserPresetName() { return getProcessor()->getMainController()->getUserPresetHandler().getCurrentlyLoadedFile().getFileNameWithoutExtension(); }

What I think might be better is to add a function to the user preset handler that returns the current preset as a file object.

@Dan-Korneff That's what I did too. Once you've made a basic plugin and got a feel for it get back to me and I'll help you get up and running

@Christoph-Hart Ok, I understand, thank you!

@d-healey not possible as far as i can tell.

If I attach a listener to a child panel, and later that child panel is removed, does HISE automatically remove the listener too or is it left hanging around somewhere?

If I have to do the cleanup manually could we get a removeAllListeners function?

@KimiA

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@ulrik never mind, I set the pitch bend value with the same function

setControllerValue()

@iamlamprey said in EQ curve from object.:

Could this be a use-case for Faust when it's ready?

No, not really, Faust is not really suited for spectral processing, it shines in circuit simulation and physical modeling.

Actually I also think that using 30+ bands of peak EQs is not the 100% right way to simulate AET as this amount of filters will smear the phase and is not super efficient (especially if it's per voice). The proper solution would be writing a FFT filter that applies interpolated gain reduction for each FFT bin based on the analysis of the loudest and softest sample:

Create a samplemap with two layers (ff and pp) For each note / sample, do an FFT, and compare the gain factor for each bin (should be <1.0 so it only works subtractive) Write that gain factor array into a data file. You can just use audio files to store that, then you can load the data as SFZ into the realtime processor. Ditch the pp layer. Write a FFT processor that applies the gain factor on each bin (the FFT size of the analysis and of the realtime processor must be the same, so you need to find a compromise between latency and frequency resolution).

Everything from 1-4 can be done inside HISE, but for 5. you need to write (or commission) a custom C++ module, but I would say that's rather beginner level spectral processing. Then you need to tweak the parameters to match your use case (FFT window, FFT size, smoothing between bins, etc).

I made a couple of videos all about XY pads. Currently they're only available to Patrons.

https://www.patreon.com/posts/how-to-create-xy-68847672

@d-healey Yes of course, I've already done this plenty of times, I need one of your t-shirts!!!!

@Dan-Korneff I don't think any of this is possible with HISE

It seems like this is a bug for these components.

The Sample Rate is taking a numeric value between 0.5 and 0.65 for it to function properly. 0.5 is the minimum = 44100Hz. You cannot use the frequency mode on the sliders for this. Same for the PostFilter while this one takes values between 0 and 1.0, using a Frequency Slider results in a Crash of Hise.

@Lindon @iamlamprey Thanks for the answer, I managed to get it to work. I simply added a Button and set the processorId to Simple Reverb and the parameterId to FreezeMode.

@d-healey thank you David. I think i find a way to do this with Engine.isMpeEnabled()

@lalalandsynth I'm still exploring the possibilities, but so far I've tested the midiout function and it works in Logic and Reaper, I have no success with Live, I read at Juce forum that Live doesn't have that kind of plugin (MidiFX) that is.

I also compiled Hise as a MidiFX plugin so I can load it in the MidiFX slot which makes it a lot easier to test.