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

Free mankini with every commercial license???

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 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.

They're all crap quite honestly.

@griffinboy said in c++ function optimization using vectorization or SIMD???:

It's more popular nowadays to store waveforms in frequency domain using FFT, and to silence bins above Nyquist before inverse FFT. Either that or use filters to make mipmaps (multiple copies of your waveform at different pitches, with antialiasing filters applied, baked into the copies, play back the appropriate pre-antialiased file for the pitch) optionally doing so at 2x oversampling and using additional interpolation to remove aliasing that happens from extra processes that happen in Realtime.

Cheers dude! I was aware of this, but I wanted to see how far I could get with sinc. Turns out, quite far! I've got 22% CPU usage for about 30 voices now. Which isn't really super optimal, but it was a fun project.

That paper you linked me a while back - https://www.mp3-tech.org/programmer/docs/resampler.pdf - was what got me interested.

I think I understand the process you mean though, for the mipmapping approach. Something like:

• Oversample original audio x2 (juce::dsp::oversampling can handle this)
• Set up a root note
• For mip-maps below the root note - lowpass and downsample (dsp::FilterDesign::designFIRLowpassWindowMethod then keep every 2nd sample)
• For mip-maps above the root note - upsample and then lowpass (use the same oversampling approach here for the upsampling and then the same kind of FIR filter???)
• Store each level, and then move on to the playback engine

I think that'd be the approach??

Playback engine-wise, I'd still need to have an interpolation method to playback notes in between the mipmap levels I would guess. Can Hermite cover this, or do I need to go polyphase still?

Hey apologies for the bump. But I got this to work by putting:
JUCE_LOG_ASSERTIONS=1

into the preprocessor definitions and then rebuilding. I get data from my custom c++ node printing to the visual studio log:

It wasn't working until I did that.

@Orvillain

Lesson 4 - SNEX Node layout.

I'm still wrapping my head around how the SNEX node works.

The first thing to note is, SNEX code does not support strings. The documentation for HISE does make this clear, but if you haven't seen it yet... then I've told you again! Here's the docs link:
https://docs.hise.audio/scriptnode/manual/snex.html#getting-started

As the docs say:
The Scriptnode Expression Language (SNEX ) is a simplified subset of the C language family and is used throughout scriptnode for customization behaviour.

Which means that most of the syntax you're used to when writing interface scripts, is just not going to be the same. There are some overlaps however - Console.print() is still used in SNEX scripts. However, print messages only get logged to the console when you put the SNEX node into debug mode. Which you can do by clicking this button:

From what I can tell, by default we have the following methods:

• prepare
• reset
• process
• processFrame
• handleHiseEvent
• setExternalData
• setParameter

Each one of these methods has a purpose. I'm still experimenting to figure out what those are, but here's what I've come up with so far:

• prepare
  This is called when you compile or initialise your SNEX node, and it seems to run for each audio channel. I would guess this is meant to setup global parameters like sample rate and block size. Things that do not change from voice to voice.
• reset
  This is called when you trigger a voice, in my case from midi. When using a ScriptNode Synthesiser, the midi passes into the node automatically. This is where you would initialise variables that can hold different values from voice to voice, but that must start out with the same default value each time.
• process
  Haven't quite figured this one out yet.
• processFrame
  Haven't quite figured this one out yet.
• handleHiseEvent
  This is called when you trigger a HiseEvent - typically a midi event. This is where you would parse out your midi notes, velocities, controllers, and program changes; any midi data really.
• setExternalData
  This is called whenever there is a change to the external data. In our case, that would be the AudioFile we added in previous steps. So for example if you went to the complex data editor for the External AudioFile Slot (in the node editor) and loaded a new file, this method would get called. This is where you would resize any arrays that you're using to store the sample data, for example.
• setParameter
  This is called whenever a parameter inside the SNEX node is adjusted. You can parse the parameters out by using if statements and checking P against 0, 1, 2, 3, etc, depending on how many parameters you actually have.

SNEX has some hard-coded variable names, most of which I don't know yet. But a valuable one is "ExternalData". Consider this code:

template <int NV> struct audio_loader
{
	SNEX_NODE(audio_loader);
	
	ExternalData data;
	// 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)
	{
		
	}
	
	// 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)
	{
		
	}
};

Most of it doesn't do anything. But we have established that ExternalData is linked to a variable called data. We can also see this in the data table view:

Notice how ExternalData is a Data Type, and it is named data. Also notice how it has a variety of sub attributes - dataType, numSamples, numChannels, etc.

Let's swap out the file loaded in the AudioFile editor:

Notice how numSamples has updated, and also numChannels.

Back to the code:

// 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)
	{
		
	}

The data variable we established as ExternalData at the top of the script, is now actually having the data pushed into it by the setExternalData method - which has two inputs; "d" and "index".

This shows the very very basics of getting sample data into a SNEX script. But we're still not doing anything with it yet.

@d-healey I don't mean to be rude, but please don't distract from the purpose of this thread. Beautiful code and efficient code isn't the point here.

The point is to demonstrate how the API works, and for there to be a resource for people who come along in the future looking to do sample loading from their scripts, and looking to do advanced things in ScriptNode or SNEX.

I know full well that in a real world scenario, you wouldn't specify a bunch of files each as an individual const, and you'd put them in a key value pair inside of an array, or perhaps a function acting as a meta-object.

@Orvillain

Lesson 3 - using the SNEX node with sample content.

This one is something I'm still getting my head around. @Christoph-Hart kindly provided a one shot SNEX node demo, which you can find by going to the example snippet browser in the Help menu:

This will open a whole new window where you can experiment with snippets. Maybe I'll go over the specific snippet in another post, but for this one... we're starting fresh, and we're going to just do the basics.

So.... here we have a basic interface script that gives us some file const references, an AudioSampleProcessor retreiving the AudioSampleProcessor from a Scriptnode Synthesiser in our module tree. That synthesiser has a DspNetwork assigned to it:

Right now, if we run the code... it will fail to find the AudioSampleProcessor as explained above. Let's add a SNEX node:

When you do this, it will be blank. You will need to click the three dot menu icon and choose "create new file" - strangely enough, you have to do this even when creating an embedded network. But fine. Let's do it:

We need to give it a name:

At this point, the node becomes active, indicated by the green highlight text:

Now if you open the same menu, you get more options:

We're going to select 'Add AudioFile':

You can see that now there is an extra icon in the SNEX node, which opens the AudioFile "Complex Data Editor" panel.

We can add an External AudioFile Slot using the icon on the right hand side:

And now you can see that the data editor will display whatever sample you assign to that AudioSampleProcessor from your script:

So here we see that file_r has been loaded into the buffer, and if we wanted to do file_f instead we could change the code to do that:

Note - you will not be able to playback the audio at this stage, as your SNEX code will be completely empty. If you click the icon on the right side, it will open the code editor for this SNEX node:

So whilst the data is loaded, our code isn't doing anything with it.

I get this with a custom hardcoded script envelope modulator. You can see the display at the top is transparent, and I can see whatever window is below it through the postbox.

But I know the display buffer is working, because when I connect a floating tile to it I get this:

@Dan-Korneff Works here Dan. BTW, you can reduce your CPU usage by doing this:

In case you weren't aware. Trick from @aaronventure from another thread.

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:

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.

See this post.

@Christoph-Hart You legend! Thanks very much! Will test here and report back if any problems. But I've got quite a cool LFO thing here now!

@Christoph-Hart I'm a bit confused here.

I put a regular ramp into my network. I put it inside a no_midi container. When I play and hold some low notes, and crucially then I play and hold some high notes afterwards, I get the same speed ramp out of this.

But if I try the same with a clock_ramp, the high-notes change the speed of the ramp I get from it, for all of the voices.

In fact, it doesn't really matter if I play low and high notes. If I play any notes, then while those are held, I play some others... the speed of the clock ramp will change. Why is this??

This only happens in polyphonic mode. Does not happen in monophonic mode.

@Christoph-Hart Gotcha. No, I'm fine using that function. Just didn't know if there was a reason to do it another way.

@Christoph-Hart

@Christoph-Hart said in Scriptnode clock-sync - is there a preferred solution???:

Why don't you want to use the inbuilt function?

You mean the built in LFO block??? Coz I'm adding a bunch of waveshaping and warping and fun things to the LFO shapes.

Sorry, two more things.

For retriggering... is it better to just use the monophonic/polyphonic button for the modulator in the module tree....

Or should I attempt to build the retrigger functionality into the custom envelope modulator scriptnode that I am building?

I'm intending on compiling it once the network is done.

Also, am I right in thinking for the clock-ramp it just syncs to the incoming clock no matter what, and that there is no concept of retriggering or free-running to think about??

Yeah as I said, the multipler feature is great for those super slow LFO's. Awesome when you start combining different expressions to shape the ramp too, to create some organic modulation shapes beyond the basics.

@Christoph-Hart said in Scriptnode clock-sync - is there a preferred solution???:

@Orvillain you can container.branch between a normal ramp and the clock sync one for switching between unsynced & synced time.

Right gotcha, and just sack off the tempo_sync node altogether when using the regular ramp?