Alright, the docs are online, but let me know if something is unclear.

@plugintutor
You don't need to add any code for looping to work.

@Christoph-Hart
Related to this issue of not being able to grab notes by their midi channel in the NoteOn Cb, I tried a different approach using no arp or midi players at all. Instead I've built 4 arpeggiators manually and now to my question:

I use the TransportHandler and made the "NoteOn callback inside the GridChange function set to 18, 1/96 (1/64T)

const TH = Engine.createTransportHandler(); TH.setEnableGrid(true, 18); TH.setSyncMode(TH.PreferExternal); if (MPsActivate[0].getValue()) { local nv = NVLookUpNum[MPsSpeed[0].getValue()-1]; if (gridcount % (nv*96) == 0) { slider1 == NumSliders.getValue()-1 ? slider1 = 0 : slider1++; //for (sp in S) local nn = MPSPs[0][0].getSliderValueAt(slider1); local v = MPSPs[0][1].getSliderValueAt(slider1); local l = MPSPs[0][2].getSliderValueAt(slider1); local len = Engine.getSamplesForQuarterBeats(nv*l*4); for (n in PKs) { local ev = Synth.addNoteOn(1, Math.range(nn + n, 0, 126), v, 0); Synth.noteOffDelayedByEventId(ev, len); } } }

I have not tested this carefully yet.

Is there a drawback using this approach?
It feels like it will not be as accurate as using the built in NoteOn cb

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

alt text

@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.