That's the right direction, but I would also try to stay within the HISE user preset system so you get the intended loading prodecure (kill voices, load the preset on a background thread, then unsuspend the audio processing).

With the custom data model you can define the layout of the user preset file as you wish. Note that it will be converted from JSON to XML at some point though (which is a bit weird).

They also want to be able to export a preset, and include any sample content with the preset - ie; bundle the custom loaded .wav file, or the samplemap, with the preset. For personal sharing purposes.

I would detach this from the user preset system. A preset with sample mapping data and samples should be

The preset file The JSON (or SFZ or whatever) mapping file The audio samples

all in one folder.

@HISEnberg If in doubt, stalk @griffinboy 🤣

So following Chris's comment in that thread, it sounds like the node needs to inherit from:

public hise::TempoListener

And then you write a custom callback. Something like:

void tempoChanged(double newTempo) override { // yo bro, do a thing here innit fam bruv dudemeister for (auto& t : data) { t.bpm = newTempo; t.refresh(); } }

But I haven't tried it yet.

@Orvillain I was gonna suggest to try that but like....nah that cant be it. haha.

Interestingly, that last solution works for the node. But when I try to compile the network, the network won't compile. Not sure why yet. Need to look into it.

@Orvillain Agreed so far I am really happy about this! Very simple to use. Need to test it out in some real world examples still!

@Orvillain

Voices in Hise are managed 'automatically'.
Take a read of Polydata.

I don't remember where it can be found. But the Hise source has all the .h and .cpp files which have the implementations for voice handling. You can see what's currently going on, and perhaps there will be some useful api that you're not yet making use of.

Christoph is the person to ask though!

Is there a way to update the scriptnode UI with the relevant new parameter

Not without hacks. The best way of thinking about this is a black box communication of parameters into the node. Now if you want to update a UI state that you display on the plugin interface, global cables (and their data callback) are the way to go, there you can easily pack everything up into a nice JSON and send it back to HISE Script (on a deferred thread!), but I wouldn't recommend going the extra mile of updating the internal scriptnode parameters only so that you can look at them in the network with the right value.

@Orvillain

Oh yeah I tried a moog. It was arguably not very good haha.
It becomes very dark in a very musty way.

Unless you're willing to stack a few to get a high order it's going to sound quite blurry. Whether or not that's a bad thing is up to you.

For refrence the real antialising filters from pedals like the memory man are steep. They are almost like low order elliptic lowpasses. That's the closest 'standard' filter response I found to the real thing. This allows them to have a high and crisp feeling cut-off (3.5k) while still killing aliasing.
It's different to a synth filter.

But that's what the hardware BBD effects do anyway.

@Orvillain apparently, the problem is in the Reaper itself, everything works fine for me on version 7.23.

@Christoph-Hart said in Force mono node?:

Not sure I understand, where do you do the switch? And why do you want to switch if you know that a node is only processing a single channel?

If you're node is known to be mono, then just process the first channel in the implementation, or if you want to force a generic node to only process the first channel, you can use the wrap::fix template to force mono processing.

// instead of this member declaration MyClass obj; // use wrap::fix<1, MyClass> obj;

Your code is probably throwing an assertion at auto* right = block.getChannelPointer(1); if you have created the audio block from a single channel process data object.

I was thinking in situations where I want to do a tubescreamer emulation, for example. I don't want true stereo processing.

But maybe I'm overthinking it, and I just do a stereo node... collapse the channels... process the result... write the result back to output left and output right, with some gain compensation?

@Orvillain
There is an audio render snippet in the snippet browser. Maybe you can build your own exporter based on that?

@Christoph-Hart Thank you!

@griffinboy Oh nice!! Some night time reading material right there!

@Christoph-Hart Ah nice! Yeah that gives me a similar result to what I posted above, just with less faffing around I suppose!

@Orvillain yeah that‘s just a glitch of the popup interface of the editor and I never looked at one long enough to bother fixing it.

Just to report back, it worked perfectly. It even now picks up the lastest value properly, whereas I'm pretty sure it didn't before. Very nice!

So here's my take on how to do switchable LFO's:

You primarily rely on the built in modules monophonic/polyphonic flag for reset/retrigger behaviour. Don't bother even building it for your own network.

Do your sync this way:
0bc0eeef-5d7a-4c85-822a-a9b901fc0519-image.png

A branch container, containing two chains. Each one has its own ramp source in it. Make one of them the regular ramp source, the other one the clock_ramp source.

You can even write a simple math expression to adjust the phase.

@Christoph-Hart Yep, that has fixed it, thanks!!

@Christoph-Hart Yep, this is fixed now. Thanks Christoph!