Don't know if this is the done thing really, but I wanted to show off:
https://youtu.be/1kMHloRQLcM

Again, wasn't sure where to put this. But I created my own node.

https://youtu.be/-siB1UJfrD0

Modelled analog bucket brigade delay. I'm starting to build up quite a nice collection of delay and reverb utilities now!

@Chazrox I might do a video or two on everything I've learned!

I'm actually moving into more JUCE C++ based projects now. But I've spent 18 years in music tech. Used to work for FXpansion, then ROLI, then inMusic until mid last year when I made the transition to developer. I've got a couple of HISE projects on the go right now - more synths and effects than sample libraries at the moment, but I've done sample libraries throughout my career, with a specific focus in drum libraries. I have a few Kontakt ports in the pipeline for this year too.

Eventually will launch my own plugin company as well. That is the goal.

More showoff stuff from me. I've built my own standalone JUCE app that allows me to build effects processors on the fly:
https://youtu.be/ogaE-hkUcsU

This is primarily a dev tool for my own projects. The eventual plan is to have it export c++ code, and maybe even have it spit out a custom HISE node that I can just drop into projects too!!

Lesson 5 - SNEX code in a bit more detail.

So I'm by no means an expert in C or C++ - in fact I only just recently started learning it. But here's what I've sussed out in regards to the HISE template.... and template is exactly the right word, because the first line is:

template <int NV> struct audio_loader

Somewhere under the hood, HISE must be setup to send in an integer into any SNEX node, that integer corresponding to a voice. NV = new voice perhaps, or number of voices ????

The line above declares a template that takes this NV integer in, and creates a struct called audio_loader for each instance of NV. Indeed we can prove this by running the following code:

template <int NV> struct audio_loader
{
	SNEX_NODE(audio_loader);
	
	ExternalData data;
	double note = 0.0;
	
	// Initialise the processing specs here
	void prepare(PrepareSpecs ps)
	{

	}
	
	// Reset the processing pipeline here
	void reset()
	{
		
	}
	
	// Process the signal here
	template <typename ProcessDataType> void process(ProcessDataType& data)
	{

	}
	
	// Process the signal as frame here
	template <int C> void processFrame(span<float, C>& data)
	{

	}
	
	// Process the MIDI events here
	void handleHiseEvent(HiseEvent& e)
	{
		double note = e.getNoteNumber();
		Console.print(note);

	}
	
	// Use this function to setup the external data
	void setExternalData(const ExternalData& d, int index)
	{
		data = d;
	}
	
	// Set the parameters here
	template <int P> void setParameter(double v)
	{
		
	}
};

There are only three things happening here:

1. We set the ExternalData as in a previous post.
2. We establish a variable with the datatype of double called 'note' and we initialise it as 0.0. But this value will never hold because....
3. In the handleHiseEvent() method, we use e.getNoteNumber() and we assign this to the note variable. We then print the note variable out inside of the handleHiseEvent() method.

Now when we run this script, any time we play a midi note, the console will show us the note number that we pressed. This is even true if you play chords, or in a scenario where no note off events occur.

That's a long winded way of saying that a SNEX node is run for each active voice; at least when it is within a ScriptNode Synthesiser dsp network.

The next line in the script after the template is established is:

SNEX_NODE(audio_loader);

This is pretty straight forward. The text you pass here has to match the name of the script loaded inside your SNEX node - not the name of the SNEX node itself.

Here you can see my SNEX node is just called: snex_node.

But the script loaded into it is called audio_loader, and so the reference to SNEX_NODE inside the script has to also reference audio_loader.

@d-healey I really like that UI. Very simple, accessible, and smooth looking - for lack of a better word!

I've been posting on KvR on and off since I was a teenager. FFS, I'm sooooooo olllldddddd.

But yes ... it has always been a bit of a Wild West type of place. Not for the feint of heart. I had a guy stalking me around the forum recently, commenting on all my posts, telling me how he was going to make sure everyone knew what a POS I was - for the crime of having and then leaving a job I did my best at for 18 odd years!

There's not enough asylum's in the world apparently!

Having said that, there are some easy wins you can do to head it all off at the pass:

• Just have Claude build you a stupid installer. Use Inno on Windows, and MacOS packages; make sure MacOS performs a postscript process to remove the quarantine flag that Apple helpfully decide to add to almost anything you download through a browser these days.
• Have a proper list of features, known issues, known omissions, and "keep it simple, stupid" - don't give them any more ammunition!
• There's a big difference between "I had a crash", "there's a bug" and "I don't like this" - a lot of the time, you can simply ignore the "I don't like this" stuff.

@HISEnberg said in Third party HISE developers:

A bit abashed posting here with so much talent but here it goes!

Wouldn't worry about that, you've been very helpful in getting me up and running in various threads! So thanks! (and to everyone else also!)

@HISEnberg
Yes I wrote a custom transient detector in a c++ node, and made sure it utilised an audio file, which I can load in my UI using the audio waveform floating tile.

I implemented spectral flux extraction:

• Take your audio.
• Perform an FFT on it.
• Extract the spectral flux envelope from the FFT.
• Downsample the spectral flux envelope (optional but can help accuracy)
• Perform peak picking on the spectral flux envelope.

I used the stock JUCE FFT processor.

Free mankini with every commercial license???

@Chazrox said in I wrote a reverb:

@Orvillain Please. I've been waiting for some dsp videos! I've been watching ADC's everyday on baby topics just to familiarize myself with the lingo and what nots. I think im ready to start diving in! There are some pretty wicked dsp guys in here for sure and I'd love to get some tutuorials for writing c++ nodes.

There's two guys who got me started in this. One is a dude called Geraint Luff aka SignalSmith. This is probably his most accessible video:
https://youtu.be/6ZK2Goiyotk

Then the other guy of course is Sean Costello of ValhallaDSP fame:
https://valhalladsp.com/2021/09/22/getting-started-with-reverb-design-part-2-the-foundations/
https://valhalladsp.com/2021/09/23/getting-started-with-reverb-design-part-3-online-resources/

In essence, here's the journey; assuming you know at least a little bit of C++

1. Learn how to create a ring buffer (aka my Ring Delay thread)
2. Learn how to create an all-pass filter using a ring buffer.
3. Understand how fractional delays work, and the various types of interpolation.
4. Learn how to manage feedback loops.

Loads of resources out there for sure!

@d-healey said in scriptAudioWaveForm and updating contents:

@Orvillain Did you try, AudioWaveform.set("processorId", value); ?

Yeah I did, and it does update it based on a follow up AudioWaveform.get('processorId') call - but the UI component doesn't seem to update, and still shows data from the previous processorId. When I compile the script, then the UI updates one time... but not on subsequent calls to the set method.

I figured I needed to call some kind of update() function after setting the processorId, but no such luck so far.

woohoo! I've now got a custom FX chain file format too. At least in prototype form.

I think I'll write a tutorial on all of this once I've properly sussed it all out. Because it is quite powerful, and no-one else really seems to have dove into this.

Start off with a tanh function.

Then put a high-pass filter before it. Set the range on the tanh to be quite high. 1-10 would do. Then tweak the high-pass cutoff and notice how it impacts the distortion.

Now add a low-pass filter after the tanh. Tweak the cutoff and notice how it cleans up the high frequencies.

Start from there.

Hmmmm I started using it for the modulation matrix controller, but there were things I wanted to do that I couldn't, so I wrote my own script panel matrix controller using LAF instead.

LAF is - I guess - a subset of the JUCE graphics API, so I'm more comfortable with LAF. CSS hasn't really fully clicked for me yet, if I'm honest. I've never needed it for anything else I've ever done, because none of it was web based and it was all Python, c++, JUCE, Lua, and C#.

Don't forget to add this to HISE's preprocessor definitions too:
JUCE_LOG_ASSERTIONS=1

Oh for n00b sake... I just call .changed() it would seem.

Could we get this extended to cover 20hz-24kHz ??? Unless I'm missing something, there doesn't seem to be a way to do it. I've been writing some custom Butterworth filters and evaluating them and diagnosing issues with cascading has proven a bit tricky in a few instances.

They're all crap quite honestly.