I made this sketch in Faust to allow us to test different types of saturation

import("stdfaust.lib");

//  Parameters
drive = hslider("Drive", 1, 1, 10, 0.1);
saturationType = nentry("Saturation Type", 0, 0, 4, 1);
dryWet = hslider("Dry/Wet", 1, 0, 1, 0.01);
outGain = hslider("Output Gain", 1, 0, 2, 0.1);

// Saturation Functions
saturate(x) = 
    (saturationType == 0) * aa.tanh1(x) +  
    (saturationType == 1) * aa.arctan(x) + 
    (saturationType == 2) * aa.hardclip(x) + 
    (saturationType == 3) * aa.cubic1(x) + 
    (saturationType == 4) * aa.hyperbolic(x); 

// Main Process
process = _ <: (_, (*(drive) : saturate)) : dry_wet(dryWet) : *(outGain)
with {
    dry_wet(mix, dry, wet) = (1-mix)*dry + mix*wet;
};

Thank you for the tutorial. As someone new to this, I'd like to understand a few things. Apart from when we need to export script nodes to other projects, when exactly is it necessary to compile script nodes? Sometimes, I find that I need to compile the script node to get the sound output, while other times everything works smoothly without needing to compile

Ideas:

• Adjust gain curve to be more "exponential" for natural decay
• Add smoothing to crossover frequencies to prevent zipper noise
• Increase filter orders for steeper cutoffs
• Use fractional delays for smoother modulation (?)

import("stdfaust.lib");

// Early Reflections
earlyReflections = par(i, 4, de.delay(ma.SR, delayTime(i)) * gain(i))
with {
    delayTime(i) = ba.take(i+1, (13, 17, 19, 23)) * 1.5;
    gain(i) = ba.take(i+1, (0.7, 0.6, 0.5, 0.4));
};

FDN = si.bus(2) : (preDelay, preDelay) <: (earlyReflections, earlyReflections, MIMO_Network ~ MATRIX) :> par(i, 2, (+:applyFilters))
with {    
    // User-controllable parameters
    decayMs = hslider("Decay[unit:ms][style:knob]", 1000, 100, 10000, 1);
    preDelayMs = hslider("Pre-Delay[unit:ms][style:knob]", 20, 0, 200, 1);
    modDepth = hslider("Modulation Depth[unit:%][style:knob]", 0.5, 0, 2, 0.01) : *(0.01);
    modRate = hslider("Modulation Rate[unit:Hz][style:knob]", 0.5, 0.1, 2, 0.1);

    // 
    // High crossover scales with decay time - longer reverbs need lower high crossover
    highCrossover = min(12000, max(2000, 8000 - (decayMs - 1000) * 0.4));
    
    // Low crossover is related to high crossover but stays within reasonable bounds
    lowCrossover = min(500, max(100, highCrossover * 0.06));
    
    // Decay ratios adapt based on decay time
    // Longer reverbs typically have more low-end build-up, so we reduce low decay ratio
    lowDecayRatio = max(0.4, min(1.2, 1.0 - (decayMs - 1000) * 0.0001));
    
    // High frequencies decay faster in longer reverbs
    highDecayRatio = max(0.3, min(1.0, 0.8 - (decayMs - 1000) * 0.0002));

    // Pre-delay
    maxPreDelay = 200;
    preDelay = de.sdelay(ma.SR * maxPreDelay / 1000, 512, preDelayMs * ma.SR / 1000);
    
    // Calculate decay factor based on milliseconds and frequency band
    calcDecayFactor(delayLengthSamples, decayMs, ratio) = pow(0.001, delayLengthSamples / (decayMs * ratio * ma.SR / 1000.0));
    
    // Matrix
    MATRIX = ro.hadamard(16) : par(i, 16, *(1/4));
    
    // Network
    MIMO_Network(fb1, fb2, fb3, fb4, fb5, fb6, fb7, fb8, fb9, fb10, fb11, fb12, fb13, fb14, fb15, fb16, in1, in2) = 
        par(i, 16, DEL(i))
    with {
        baseDelays = (3797, 3541, 3323, 3109, 2939, 2801, 2693, 2593, 2503, 2423, 2351, 2287, 2237, 2179, 2131, 2083);
        DEL(i) = de.fdelay(5000, modulatedDelay(i), fb(i) + input(i)) : multiband_decay
        with {
            delay = baseDelays : ba.selectn(16, i);
            modulatedDelay(i) = delay + os.osc(modRate + 0.01*i) * delay * modDepth;
            fb(i) = ba.selectn(16, i, fb1, fb2, fb3, fb4, fb5, fb6, fb7, fb8, fb9, fb10, fb11, fb12, fb13, fb14, fb15, fb16);
            input(i) = select2(i < 2, 0, ba.selectn(2, i, in1, in2));
            
            // Multi-band decay using automatic parameters
            multiband_decay = _ <: (
                fi.lowpass(2, lowCrossover) * calcDecayFactor(delay, decayMs, lowDecayRatio),
                fi.bandpass(2, lowCrossover, highCrossover) * calcDecayFactor(delay, decayMs, 1.0),
                fi.highpass(2, highCrossover) * calcDecayFactor(delay, decayMs, highDecayRatio)
            ) :> _;
        };
    };
    
    // Apply filters to each channel (oh! I see colours, that's so trippy :P)
    applyFilters = fi.lowpass(1, highCrossover * 1.5) : fi.highpass(1, lowCrossover * 0.5);
};    

process = FDN;

@Lindon Yes, I use it as a support tool to explore different solutions in Faust, but not primarily to generate code directly. In my experience, using it to generate code from scratch can often be much more time-consuming. Instead, I focus on:

• Analyzing and better understanding the documentation
• Discussing various approaches and strategies
• Reviewing the logic of my code
• Getting suggestions for potential optimizations

@DanH
Ok, it was easier than expected, There are some graphical glitches but the time in ms is correct.

@dannytaurus
have you tried adding this line in extra definition, in project settings?

ENABLE_ALL_PEAK_METERS=1

@aaronventure
the full code is in the snippet in the main post. inside Script FX1 there is the faust node.

    // Add gain reduction meters to the UI
    gainReductionLeft = gain_reduction(l) : vbargraph("Gain Reduction Left", -60, 0);   // Left channel
    gainReductionRight = gain_reduction(r) : vbargraph("Gain Reduction Right", -60, 0);  // Right channel
};

@clevername27 Thanks for the explanation. Your comment on the post helped me solve the memory management. Thanks again.

@aaronventure said in softbypass_switch crashes:

@Mighty23 first, build the latest commit, except if that breaks your project for whatever reason.

I'd rather not do that, I have some projects with different FAUST versions and often in the forum it is recommended not to have different HISE versions. With these two assumptions, the 4+ version still has to wait for me.

Hi,
I would like to know if there are any "tips & tricks" to be able to save the samples loaded on Audio Loop Player in the presets. I searched a bit in the documentation but I don't think I found anything to do this.

I would mainly like to understand if it is possible to save a sample loaded inside Audio Loop Player to be able to distribute it in the presets, not only" factory presets" but also to be able to let end users do it too.

@Straticah I would never have been able to find this solution alone, Wow. Using a panel is the most immediate, brilliant solution, thanks

@Matt_SF This version also seems to work reasonably well. Thanks for your time, it's time to add breakpoints and understand better. Thanks again for your time

@ulrik This solves the problem completely, thanks. By commenting that line I can get the desired result

@HISEnberg Thanks for this. I've compiled and tested in DAW, No problems here

Update / Solved

Add: (I Don't know why)

ENABLE_ALL_PEAK_METERS=1

In 'Extra Definitions' for all OS. Do it in 'Settings' menù or in 'autogenerated.projucer' file.

@ulrik said in Failing at Music 101 , tempo_sync values. 3/16, 1/3, 3/4:

@Mighty23 so your questions
3/16 = tempo 1/16, multiplier 3, same with 3/4, tempo 1/4, multiplier 3

as I show in this video
video

1/3 meaning 1 third of 1 bar (4/4) = 1/2T

Thank you for the time spent trying to understand together. I am on Windows and have not yet tested on Mac if tempo_sync follows the values of the DAW. The title of the post is ironic and means that I have failed the basic concepts of music theory. Thank you again for your time. I'll try to run some tests myself

@Mighty23 said in Missing Binaries (vst3) but XCode notifies "Build Succeeded":

@d-healey said in Missing Binaries (vst3) but XCode notifies "Build Succeeded":

@Mighty23 said in Missing Binaries (vst3) but XCode notifies "Build Succeeded":

HISE-master.

I thought you were using the latest develop branch?

I'm using the the develop branch

You've found the issue, I feel so silly. Again, thank you for the always prompt assistance.

@udalilprofile
I tried using this project in a VST and tested it in my DAW, I had no problems.

There is also a scriptnode example called "transient designer" here, in the forum.
Both examples are good starting points.
https://forum.hise.audio/post/45656

Without making any further changes, I think this FAUST version introduces too much distortion.
(opinion of a newbie).

Something like this?
items: knbValue & lblValue

const var lblValue = Content.getComponent("lblValue");  // your label

inline function onknbValueControl(component, value)
{
	lblValue.set("text",  value); 
	StereoDelay.setAttribute(StereoDelay.DelayTimeLeft, value);   // i've used the stereo delay, use your fx or instrument
};

Content.getComponent("knbValue").setControlCallback(onknbValueControl);

If your knob has a 101 steps this should work.

@pgaudioworks try someting like this, where "cmbSampleMap" is my Combo Box

// My Sample Map on Combobox
const var sampleMaps = Sampler.getSampleMapList(); // get a Sample Map List
const var cmbSampleMap = Content.getComponent("cmbSampleMap"); //combo box
const var Sampler1 = Synth.getChildSynth("Sampler1"); // Sampler Reference

// populate combo box with sample maps
cmbSampleMap.set("items", sampleMaps.join("\n"));

inline function oncmbSampleMapControl(component, value)
{
	//Console.print(value);
	Sampler1.asSampler().loadSampleMap(sampleMaps[value-1]);
};

Content.getComponent("cmbSampleMap").setControlCallback(oncmbSampleMapControl);