mercury-engine 1.6.0 → 1.7.0

package/dist/mercury.js

@@ -15480,6 +15480,12 @@ const fxMap = {
 	'comp' : (params) => {
 		return new Compressor(params);
 	},
+	'comb' : (params) => {
+		return new CombFilter(params);
+	},
+	'karplus' : (params) => {
+		return new CombFilter(params);
+	},
 	'lfo' : (params) => {
 		return new LFO(params);
 	},
@@ -15570,6 +15576,63 @@ function disposeNodes(nodes=[]) {
 	});
 }
 
+// A Lowpass Feedback CombFiltering effect
+// Adds a short feedback delay to the sound based on a specific note
+// resulting in a tonal output, like the resonating sound of a string 
+// sometimes also called Karplus Strong String Synthesis.
+// Negative feedback is possible for generating odd harmonics
+//
+const CombFilter = function(_params) {
+	// the default parameters
+	_params = Util.mapDefaults(_params, [0, 0.8, 0.5, 0.5]);
+	this._pitch = Util.toArray(_params[0]);
+	this._fback = Util.toArray(_params[1]);
+	this._damp = Util.toArray(_params[2]);
+	this._wet = Util.toArray(_params[3]);
+
+	// ToneAudioNode has all the tone effect parameters
+	this._fx = new Tone.ToneAudioNode();
+
+	// A gain node for connecting with input and output
+	this._fx.input = new Tone.Gain(1);
+	this._fx.output = new Tone.Gain(1);
+	// the fx processor
+	this._fx.workletNode = Tone.getContext().createAudioWorkletNode('combfilter-processor');
+	// connect input, fx and output
+	this._fx.input.chain(this._fx.workletNode, this._fx.output);
+
+	this.set = (count, time, bpm) => {
+		const pitch = Util.toMidi(Util.getParam(this._pitch, count));
+		const _dt = 1000 / Util.mtof(pitch);
+		
+		// some mapping for the feedback to make it logarithmic in length
+		let _fb = Util.getParam(this._fback, count);
+		let sign = _fb < 0 ? -1 : 1;
+		_fb = Util.clip(Math.pow(Math.abs(_fb), 0.1) * sign, -0.999, 0.999);
+
+		const _dm = Util.clip(Util.getParam(this._damp, count));
+		const _dw = Util.clip(Util.getParam(this._wet, count));
+		
+		// get parameters from workletprocessor
+		const dt = this._fx.workletNode.parameters.get('time');	
+		dt.setValueAtTime(_dt, time);
+		const fb = this._fx.workletNode.parameters.get('feedback');	
+		fb.setValueAtTime(_fb, time);
+		const dm = this._fx.workletNode.parameters.get('damping');	
+		dm.setValueAtTime(_dm, time);
+		const dw = this._fx.workletNode.parameters.get('drywet');	
+		dw.setValueAtTime(_dw, time);
+	}
+
+	this.chain = () => {
+		return { 'send' : this._fx, 'return' : this._fx }
+	}
+
+	this.delete = () => {
+		disposeNodes([this._fx.input, this._fx.output, this._fx]);
+	}
+}
+
 // A formant/vowel filter. With this filter you can imitate the vowels of human 
 // speech. 
 // 
@@ -16039,8 +16102,7 @@ const DattorroReverb = function(_params){
 	}
 
 	this.delete = () => {
-		const nodes = [ this._fx, this._mix, this._mixDry, this._fx.input, this._fx.output ];
-		nodes.forEach(n => { n.disconnect(); n.dispose() });
+		disposeNodes([ this._fx, this._mix, this._mixDry, this._mixWet, this._fx.input, this._fx.output ]);
 	}
 }
 
@@ -18869,7 +18931,7 @@ const { WebMidi } = require("webmidi");
 // load extra AudioWorkletProcessors from file
 // transformed to inline with browserify brfs
 
-const fxExtensions = "\n// Various noise type processors for the MonoNoise source\n// Type 2 is Pink noise, used from Tone.Noise('pink') instead of calc\n//\nclass NoiseProcessor extends AudioWorkletProcessor {\n\tstatic get parameterDescriptors(){\n\t\treturn [{\n\t\t\tname: 'type',\n\t\t\tdefaultValue: 5,\n\t\t\tminValue: 0,\n\t\t\tmaxValue: 5\n\t\t},{\n\t\t\tname: 'density',\n\t\t\tdefaultValue: 0.125,\n\t\t\tminValue: 0,\n\t\t\tmaxValue: 1\n\t\t}];\n\t}\n\t\n\tconstructor(){\n\t\tsuper();\n\t\t// sample previous value\n\t\tthis.prev = 0;\n\t\t// latch to a sample \n\t\tthis.latch = 0;\n\t\t// phasor ramp\n\t\tthis.phasor = 0;\n\t\tthis.delta = 0;\n\t}\n\n\tprocess(inputs, outputs, parameters){\n\t\t// input is not used because this is a source\n\t\tconst input = inputs[0];\n\t\tconst output = outputs[0];\n\t\tconst HALF_PI = Math.PI/2;\n\n\t\t// for one output channel generate some noise\t\n\t\tif (input.length > 0){\n\t\t\tfor (let i = 0; i < input[0].length; i++){\n\t\t\t\tconst t = (parameters.type.length > 1) ? parameters.type[i] : parameters.type[0];\n\t\t\t\tconst d = (parameters.density.length > 1) ? parameters.density[i] : parameters.density[0];\n\t\t\t\n\t\t\t\t// some bipolar white noise -1 to 1\n\t\t\t\tconst biNoise = Math.random() * 2 - 1;\n\t\t\t\t// empty output\n\t\t\t\tlet out = 0;\n\n\t\t\t\t// White noise, Use for every other choice\n\t\t\t\tif (t < 1){\n\t\t\t\t\tout = biNoise * 0.707;\n\t\t\t\t}\n\t\t\t\t// Pink noise,  use Tone.Noise('pink') object for simplicity\n\t\t\t\telse if (t < 2){\n\t\t\t\t\tout = input[0][i] * 1.413;\n\t\t\t\t}\n\t\t\t\t// Brownian noise\n\t\t\t\t// calculate a random next value in \"step size\" and add to \n\t\t\t\t// the previous noise signal value creating a \"drunk walk\" \n\t\t\t\t// or brownian motion\n\t\t\t\telse if (t < 3){\t\t\n\t\t\t\t\tthis.prev += biNoise * d*d;\n\t\t\t\t\tthis.prev = Math.asin(Math.sin(this.prev * HALF_PI)) / HALF_PI;\n\t\t\t\t\tout = this.prev * 0.707;\n\t\t\t\t}\n\t\t\t\t// Lo-Fi (sampled) noise\n\t\t\t\t// creates random values at a specified frequency and slowly \n\t\t\t\t// ramps to that new value\n\t\t\t\telse if (t < 4){\n\t\t\t\t\t// create a ramp from 0-1 at specific frequency/density\n\t\t\t\t\tthis.phasor = (this.phasor + d * d * 0.5) % 1;\n\t\t\t\t\t// calculate the delta\n\t\t\t\t\tlet dlt = this.phasor - this.delta;\n\t\t\t\t\tthis.delta = this.phasor;\n\t\t\t\t\t// when ramp resets, latch a new noise value\n\t\t\t\t\tif (dlt < 0){\n\t\t\t\t\t\tthis.prev = this.latch;\n\t\t\t\t\t\tthis.latch = biNoise;\n\t\t\t\t\t}\n\t\t\t\t\t// linear interpolation from previous to next point\n\t\t\t\t\tout = this.prev + this.phasor * (this.latch - this.prev);\n\t\t\t\t\tout *= 0.707;\n\t\t\t\t}\n\t\t\t\t// Dust noise\n\t\t\t\t// randomly generate an impulse/click of value 1 depending \n\t\t\t\t// on the density, average amount of impulses per second\n\t\t\t\telse if (t < 5){\n\t\t\t\t\tout = Math.random() > (1 - d*d*d * 0.5);\n\t\t\t\t}\n\t\t\t\t// Crackle noise\n\t\t\t\t// Pink generator with \"wave-loss\" leaving gaps\n\t\t\t\telse {\n\t\t\t\t\tlet delta = input[0][i] - this.prev;\n\t\t\t\t\tthis.prev = input[0][i];\n\t\t\t\t\tif (delta > 0){\n\t\t\t\t\t\tthis.latch = Math.random();\n\t\t\t\t\t}\n\t\t\t\t\tout = (this.latch < (1 - d*d*d)) ? 0 : input[0][i] * 1.413;\n\t\t\t\t}\n\t\t\t\t// send to output whichever noise type was chosen\n\t\t\t\toutput[0][i] = out;\n\t\t\t}\n\t\t}\t\t\n\t\treturn true;\n\t}\n}\nregisterProcessor('noise-processor', NoiseProcessor);\n\n// A Downsampling Chiptune effect. Downsamples the signal by a specified amount\n// Resulting in a lower samplerate, making it sound more like 8bit/chiptune\n// Programmed with a custom AudioWorkletProcessor, see effects/Processors.js\n//\nclass DownSampleProcessor extends AudioWorkletProcessor {\n\tstatic get parameterDescriptors() {\n\t\treturn [{\n\t\t\tname: 'down',\n\t\t\tdefaultValue: 8,\n\t\t\tminValue: 1,\n\t\t\tmaxValue: 2048\n\t\t}];\n\t}\n\n\tconstructor(){\n\t\tsuper();\n\t\t// the frame counter\n\t\tthis.count = 0;\n\t\t// sample and hold variable array\n\t\tthis.sah = [];\n\t}\n\n\tprocess(inputs, outputs, parameters){\n\t\tconst input = inputs[0];\n\t\tconst output = outputs[0];\n\n\t\t// if there is anything to process\n\t\tif (input.length > 0){\n\t\t\t// for the length of the sample array (generally 128)\n\t\t\tfor (let i=0; i<input[0].length; i++){\n\t\t\t\tconst d = (parameters.down.length > 1) ? parameters.down[i] : parameters.down[0];\n\t\t\t\t// for every channel\n\t\t\t\tfor (let channel=0; channel<input.length; ++channel){\n\t\t\t\t\t// if counter equals 0, sample and hold\n\t\t\t\t\tif (this.count % d === 0){\n\t\t\t\t\t\tthis.sah[channel] = input[channel][i];\n\t\t\t\t\t}\n\t\t\t\t\t// output the currently held sample\n\t\t\t\t\toutput[channel][i] = this.sah[channel];\n\t\t\t\t}\n\t\t\t\t// increment sample counter\n\t\t\t\tthis.count++;\n\t\t\t}\n\t\t}\n\t\treturn true;\n\t}\n}\nregisterProcessor('downsampler-processor', DownSampleProcessor);\n\n// A distortion algorithm using the tanh (hyperbolic-tangent) as a \n// waveshaping technique. Some mapping to apply a more equal loudness \n// distortion is applied on the overdrive parameter\n//\nclass TanhDistortionProcessor extends AudioWorkletProcessor {\n\tstatic get parameterDescriptors(){\n\t\treturn [{\n\t\t\tname: 'amount',\n\t\t\tdefaultValue: 4,\n\t\t\tminValue: 1\n\t\t}, {\n\t\t\tname: 'makeup',\n\t\t\tdefaultValue: 0.5,\n\t\t\tminValue: 0,\n\t\t\tmaxValue: 2\n\t\t}]\n\t}\n\n\tconstructor(){\n\t\tsuper();\n\t}\n\n\tprocess(inputs, outputs, parameters){\n\t\tconst input = inputs[0];\n\t\tconst output = outputs[0];\n\n\t\tif (input.length > 0){\n\t\t\tfor (let channel=0; channel<input.length; ++channel){\n\t\t\t\tfor (let i=0; i<input[channel].length; i++){\n\t\t\t\t\tconst a = (parameters.amount.length > 1)? parameters.amount[i] : parameters.amount[0];\n\t\t\t\t\tconst m = (parameters.makeup.length > 1)? parameters.makeup[i] : parameters.makeup[0];\n\t\t\t\t\t// simple waveshaping with tanh\n\t\t\t\t\toutput[channel][i] = Math.tanh(input[channel][i] * a) * m;\n\t\t\t\t}\n\t\t\t}\n\t\t}\n\t\treturn true;\n\t}\n}\nregisterProcessor('tanh-distortion-processor', TanhDistortionProcessor);\n\n// A distortion algorithm using the arctan function as a \n// waveshaping technique. Some mapping to apply a more equal loudness \n// distortion is applied on the overdrive parameter\n//\nclass ArctanDistortionProcessor extends AudioWorkletProcessor {\n\tstatic get parameterDescriptors(){\n\t\treturn [{\n\t\t\tname: 'amount',\n\t\t\tdefaultValue: 5,\n\t\t\tminValue: 1\n\t\t}]\n\t}\n\n\tconstructor(){\n\t\tsuper();\n\n\t\t// quarter pi constant and inverse\n\t\tthis.Q_PI = 0.7853981633974483; // 0.25 * Math.PI;\n\t\tthis.INVQ_PI = 1.2732395447351628; //1.0 / this.Q_PI;\n\t}\n\n\tprocess(inputs, outputs, parameters){\n\t\tconst input = inputs[0];\n\t\tconst output = outputs[0];\n\n\t\tconst gain = parameters.amount[0];\n\t\tconst makeup = Math.min(1, Math.max(0, 1 - ((Math.atan(gain) - this.Q_PI) * this.INVQ_PI * 0.823)));\n\n\t\tif (input.length > 0){\n\t\t\tfor (let channel=0; channel<input.length; channel++){\n\t\t\t\tfor (let i=0; i<input[channel].length; i++){\n\t\t\t\t\toutput[channel][i] = Math.atan(input[channel][i] * gain) * makeup;\n\t\t\t\t}\n\t\t\t}\n\t\t}\n\t\treturn true;\n\t}\n}\nregisterProcessor('arctan-distortion-processor', ArctanDistortionProcessor);\n\n\n// A fuzz distortion effect in modelled after the Big Muff Pi pedal \n// by Electro Harmonics. Using three stages of distortion: \n// 1 soft-clipping stage, 2 half-wave rectifier, 3 hard-clipping stage\n// Based on: https://github.com/hazza-music/EHX-Big-Muff-Pi-Emulation/blob/main/Technical%20Essay.pdf\n// \nclass FuzzProcessor extends AudioWorkletProcessor {\n\tstatic get parameterDescriptors() {\n\t\treturn [{\n\t\t\tname: 'amount',\n\t\t\tdefaultValue: 5,\n\t\t\tminValue: 1\n\t\t}]\n\t}\n\n\tconstructor(){ \n\t\tsuper(); \n\t\t// history for onepole filter for dcblocking\n\t\tthis.history = [0, 0];\n\t}\n\n\tprocess(inputs, outputs, parameters){\n\t\tconst input = inputs[0];\n\t\tconst output = outputs[0];\n\n\t\tconst gain = parameters.amount[0];\n\t\tconst makeup = Math.max((1 - Math.pow((gain-1) / 63, 0.13)) * 0.395 + 0.605, 0.605);\n\n\t\tif (input.length > 0){\n\t\t\tfor (let channel = 0; channel < input.length; channel++){\n\t\t\t\tfor (let i = 0; i < input[channel].length; i++){\n\t\t\t\t\t// soft-clipping\n\t\t\t\t\tconst sc = Math.atan(input[channel][i] * gain * 2) * 0.6;\n\t\t\t\t\t// half-wave rectification and add for \n\t\t\t\t\t// asymmetric distortion\n\t\t\t\t\tconst hw = ((sc > 0) ? sc : 0) + input[channel][i];\n\t\t\t\t\t// hard-clipping\n\t\t\t\t\tconst hc = Math.max(-0.707, Math.min(0.707, hw));\n\t\t\t\t\t// onepole lowpass filter for dc-block\n\t\t\t\t\tthis.history[channel] = (hc - this.history[channel]) * 0.0015 + this.history[channel];\n\t\t\t\t\t// dc-block and gain compensation and output\n\t\t\t\t\toutput[channel][i] = (hc - this.history[channel]) * makeup;\n\t\t\t\t}\n\t\t\t}\n\t\t}\n\t\treturn true;\n\t}\n}\nregisterProcessor('fuzz-processor', FuzzProcessor);\n\n// A distortion/compression effect of an incoming signal\n// Based on an algorithm by Peter McCulloch\n// \nclass SquashProcessor extends AudioWorkletProcessor {\n\tstatic get parameterDescriptors(){\n\t\treturn [{\n\t\t\tname: 'amount',\n\t\t\tdefaultValue: 4,\n\t\t\tminValue: 1,\n\t\t\tmaxValue: 1024\n\t\t}, {\n\t\t\tname: 'makeup',\n\t\t\tdefaultValue: 0.5,\n\t\t\tminValue: 0,\n\t\t\tmaxValue: 2\n\t\t}];\n\t}\n\n\tconstructor(){\n\t\tsuper();\n\t}\n\n\tprocess(inputs, outputs, parameters){\n\t\tconst input = inputs[0];\n\t\tconst output = outputs[0];\n\t\t\n\t\tif (input.length > 0){\n\t\t\tfor (let channel=0; channel<input.length; ++channel){\n\t\t\t\tfor (let i=0; i<input[channel].length; i++){\n\t\t\t\t\t// (s * a) / ((s * a)^2 * 0.28 + 1) / √a\n\t\t\t\t\t// drive amount, minimum of 1\n\t\t\t\t\tconst a = (parameters.amount.length > 1)? parameters.amount[i] : parameters.amount[0];\n\t\t\t\t\t// makeup gain\n\t\t\t\t\tconst m = (parameters.makeup.length > 1)? parameters.makeup[i] : parameters.makeup[0];\n\t\t\t\t\t// set the waveshaper effect\n\t\t\t\t\tconst s = input[channel][i];\n\t\t\t\t\tconst x = s * a * 1.412;\n\t\t\t\t\toutput[channel][i] = (x / (x * x * 0.28 + 1.0)) * m * 0.708;\n\t\t\t\t}\n\t\t\t}\n\t\t}\n\t\treturn true;\n\t}\n}\nregisterProcessor('squash-processor', SquashProcessor);\n\n// Dattorro Reverberator\n// Thanks to port by khoin, taken from:\n// https://github.com/khoin/DattorroReverbNode\n// based on the paper from Jon Dattorro:\n// https://ccrma.stanford.edu/~dattorro/EffectDesignPart1.pdf\n// with small modifications to work in Mercury\n//\n// In jurisdictions that recognize copyright laws, this software is to\n// be released into the public domain.\n\n// THE SOFTWARE IS PROVIDED \"AS IS\", WITHOUT WARRANTY OF ANY KIND.\n// THE AUTHOR(S) SHALL NOT BE LIABLE FOR ANYTHING, ARISING FROM, OR IN\n// CONNECTION WITH THE SOFTWARE OR THE DISTRIBUTION OF THE SOFTWARE.\n// \nclass DattorroReverb extends AudioWorkletProcessor {\n\tstatic get parameterDescriptors() {\n\t\treturn [\n\t\t\t[\"preDelay\", 0, 0, sampleRate - 1, \"k-rate\"],\n\t\t\t// [\"bandwidth\", 0.9999, 0, 1, \"k-rate\"],\t\n\t\t\t[\"inputDiffusion1\", 0.75, 0, 1, \"k-rate\"],\n\t\t\t[\"inputDiffusion2\", 0.625, 0, 1, \"k-rate\"],\n\t\t\t[\"decay\", 0.5, 0, 1, \"k-rate\"],\n\t\t\t[\"decayDiffusion1\", 0.7, 0, 0.999999, \"k-rate\"],\n\t\t\t[\"decayDiffusion2\", 0.5, 0, 0.999999, \"k-rate\"],\n\t\t\t[\"damping\", 0.005, 0, 1, \"k-rate\"],\n\t\t\t[\"excursionRate\", 0.5, 0, 2, \"k-rate\"],\n\t\t\t[\"excursionDepth\", 0.7, 0, 2, \"k-rate\"],\n\t\t\t[\"wet\", 0.7, 0, 2, \"k-rate\"],\n\t\t\t// [\"dry\", 0.7, 0, 2, \"k-rate\"]\n\t\t].map(x => new Object({\n\t\t\tname: x[0],\n\t\t\tdefaultValue: x[1],\n\t\t\tminValue: x[2],\n\t\t\tmaxValue: x[3],\n\t\t\tautomationRate: x[4]\n\t\t}));\n\t}\n\n\tconstructor(options) {\n\t\tsuper(options);\n\n\t\tthis._Delays = [];\n\t\t// Pre-delay is always one-second long, rounded to the nearest 128-chunk\n\t\tthis._pDLength = sampleRate + (128 - sampleRate % 128);\n\t\tthis._preDelay = new Float32Array(this._pDLength);\n\t\tthis._pDWrite = 0;\n\t\tthis._lp1 = 0.0;\n\t\tthis._lp2 = 0.0;\n\t\tthis._lp3 = 0.0;\n\t\tthis._excPhase = 0.0;\n\n\t\t[\n\t\t\t0.004771345, 0.003595309, 0.012734787, 0.009307483, // pre-tank\n\t\t\t0.022579886, 0.149625349, 0.060481839, 0.1249958, // left-loop\n\t\t\t0.030509727, 0.141695508, 0.089244313, 0.106280031 // right-loop\n\t\t].forEach(x => this.makeDelay(x));\n\n\t\tthis._taps = Int16Array.from([\n\t\t\t0.008937872, 0.099929438, 0.064278754, 0.067067639, \n\t\t\t0.066866033, 0.006283391, 0.035818689, // left-output\n\t\t\t0.011861161, 0.121870905, 0.041262054, 0.08981553, \n\t\t\t0.070931756, 0.011256342, 0.004065724 // right-output\n\t\t], x => Math.round(x * sampleRate));\n\t}\n\n\tmakeDelay(length) {\n\t\t// len, array, write, read, mask\n\t\tlet len = Math.round(length * sampleRate);\n\t\tlet nextPow2 = 2 ** Math.ceil(Math.log2((len)));\n\t\tthis._Delays.push([\n\t\t\tnew Float32Array(nextPow2), len - 1, 0 | 0, nextPow2 - 1\n\t\t]);\n\t}\n\n\twriteDelay(index, data) {\n\t\treturn this._Delays[index][0][this._Delays[index][1]] = data;\n\t}\n\n\treadDelay(index) {\n\t\treturn this._Delays[index][0][this._Delays[index][2]];\n\t}\n\n\treadDelayAt(index, i) {\n\t\tlet d = this._Delays[index];\n\t\treturn d[0][(d[2] + i) & d[3]];\n\t}\n\n\t// cubic interpolation\n\t// O. Niemitalo: \n\t// https://www.musicdsp.org/en/latest/Other/49-cubic-interpollation.html\n\treadDelayCAt(index, i) {\n\t\tlet d = this._Delays[index],\n\t\t\tfrac = i - ~~i,\n\t\t\tint = ~~i + d[2] - 1,\n\t\t\tmask = d[3];\n\n\t\tlet x0 = d[0][int++ & mask],\n\t\t\tx1 = d[0][int++ & mask],\n\t\t\tx2 = d[0][int++ & mask],\n\t\t\tx3 = d[0][int & mask];\n\n\t\tlet a = (3 * (x1 - x2) - x0 + x3) / 2,\n\t\t\tb = 2 * x2 + x0 - (5 * x1 + x3) / 2,\n\t\t\tc = (x2 - x0) / 2;\n\n\t\treturn (((a * frac) + b) * frac + c) * frac + x1;\n\t}\n\n\t// First input will be downmixed to mono if number of channels is not 2\n\t// Outputs Stereo.\n\tprocess(inputs, outputs, parameters) {\n\t\tconst pd = ~~parameters.preDelay[0],\n\t\t\t// bw = parameters.bandwidth[0], // replaced by using damping\n\t\t\tfi = parameters.inputDiffusion1[0],\n\t\t\tsi = parameters.inputDiffusion2[0],\n\t\t\tdc = parameters.decay[0],\n\t\t\tft = parameters.decayDiffusion1[0],\n\t\t\tst = parameters.decayDiffusion2[0],\n\t\t\tdp = 1 - parameters.damping[0],\n\t\t\tex = parameters.excursionRate[0] / sampleRate,\n\t\t\ted = parameters.excursionDepth[0] * sampleRate / 1000,\n\t\t\twe = parameters.wet[0]; //* 0.6, // lo & ro both mult. by 0.6 anyways\n\t\t\t// dr = parameters.dry[0];\n\n\t\t// write to predelay and dry output\n\t\tif (inputs[0].length == 2) {\n\t\t\tfor (let i = 127; i >= 0; i--) {\n\t\t\t\tthis._preDelay[this._pDWrite + i] = (inputs[0][0][i] + inputs[0][1][i]) * 0.5;\n\n\t\t\t\t// removed the dry parameter, this is handled in the Tone Node\n\t\t\t\t// outputs[0][0][i] = inputs[0][0][i] * dr;\n\t\t\t\t// outputs[0][1][i] = inputs[0][1][i] * dr;\n\t\t\t}\n\t\t} else if (inputs[0].length > 0) {\n\t\t\tthis._preDelay.set(\n\t\t\t\tinputs[0][0],\n\t\t\t\tthis._pDWrite\n\t\t\t);\n\t\t\t// for (let i = 127; i >= 0; i--)\n\t\t\t// \toutputs[0][0][i] = outputs[0][1][i] = inputs[0][0][i] * dr;\n\t\t} else {\n\t\t\tthis._preDelay.set(\n\t\t\t\tnew Float32Array(128),\n\t\t\t\tthis._pDWrite\n\t\t\t);\n\t\t}\n\n\t\tlet i = 0 | 0;\n\t\twhile (i < 128) {\n\t\t\tlet lo = 0.0,\n\t\t\t\tro = 0.0;\n\n\t\t\t// input damping (formerly known as bandwidth bw, now uses dp)\n\t\t\tthis._lp1 += dp * (this._preDelay[(this._pDLength + this._pDWrite - pd + i) % this._pDLength] - this._lp1);\n\n\t\t\t// pre-tank\n\t\t\tlet pre = this.writeDelay(0, this._lp1 - fi * this.readDelay(0));\n\t\t\tpre = this.writeDelay(1, fi * (pre - this.readDelay(1)) + this.readDelay(0));\n\t\t\tpre = this.writeDelay(2, fi * pre + this.readDelay(1) - si * this.readDelay(2));\n\t\t\tpre = this.writeDelay(3, si * (pre - this.readDelay(3)) + this.readDelay(2));\n\n\t\t\tlet split = si * pre + this.readDelay(3);\n\n\t\t\t// excursions\n\t\t\t// could be optimized?\n\t\t\tlet exc = ed * (1 + Math.cos(this._excPhase * 6.2800));\n\t\t\tlet exc2 = ed * (1 + Math.sin(this._excPhase * 6.2847));\n\n\t\t\t// left loop\n\t\t\t// tank diffuse 1\n\t\t\tlet temp = this.writeDelay(4, split + dc * this.readDelay(11) + ft * this.readDelayCAt(4, exc));\n\t\t\t// long delay 1\n\t\t\tthis.writeDelay(5, this.readDelayCAt(4, exc) - ft * temp);\n\t\t\t// damp 1\n\t\t\tthis._lp2 += dp * (this.readDelay(5) - this._lp2);\n\t\t\ttemp = this.writeDelay(6, dc * this._lp2 - st * this.readDelay(6)); // tank diffuse 2\n\t\t\t// long delay 2\n\t\t\tthis.writeDelay(7, this.readDelay(6) + st * temp);\n\n\t\t\t// right loop \n\t\t\t// tank diffuse 3\n\t\t\ttemp = this.writeDelay(8, split + dc * this.readDelay(7) + ft * this.readDelayCAt(8, exc2));\n\t\t\t// long delay 3\n\t\t\tthis.writeDelay(9, this.readDelayCAt(8, exc2) - ft * temp);\n\t\t\t// damp 2\n\t\t\tthis._lp3 += dp * (this.readDelay(9) - this._lp3);\n\t\t\t// tank diffuse 4\n\t\t\ttemp = this.writeDelay(10, dc * this._lp3 - st * this.readDelay(10));\n\t\t\t// long delay 4\n\t\t\tthis.writeDelay(11, this.readDelay(10) + st * temp);\n\n\t\t\tlo = this.readDelayAt(9, this._taps[0]) +\n\t\t\t\tthis.readDelayAt(9, this._taps[1]) -\n\t\t\t\tthis.readDelayAt(10, this._taps[2]) +\n\t\t\t\tthis.readDelayAt(11, this._taps[3]) -\n\t\t\t\tthis.readDelayAt(5, this._taps[4]) -\n\t\t\t\tthis.readDelayAt(6, this._taps[5]) -\n\t\t\t\tthis.readDelayAt(7, this._taps[6]);\n\n\t\t\tro = this.readDelayAt(5, this._taps[7]) +\n\t\t\t\tthis.readDelayAt(5, this._taps[8]) -\n\t\t\t\tthis.readDelayAt(6, this._taps[9]) +\n\t\t\t\tthis.readDelayAt(7, this._taps[10]) -\n\t\t\t\tthis.readDelayAt(9, this._taps[11]) -\n\t\t\t\tthis.readDelayAt(10, this._taps[12]) -\n\t\t\t\tthis.readDelayAt(11, this._taps[13]);\n\n\t\t\toutputs[0][0][i] += lo * we;\n\t\t\toutputs[0][1][i] += ro * we;\n\n\t\t\tthis._excPhase += ex;\n\n\t\t\ti++;\n\n\t\t\tfor (let j = 0, d = this._Delays[0]; j < this._Delays.length; d = this._Delays[++j]) {\n\t\t\t\td[1] = (d[1] + 1) & d[3];\n\t\t\t\td[2] = (d[2] + 1) & d[3];\n\t\t\t}\n\t\t}\n\n\t\t// Update preDelay index\n\t\tthis._pDWrite = (this._pDWrite + 128) % this._pDLength;\n\n\t\treturn true;\n\t}\n}\nregisterProcessor('dattorro-reverb', DattorroReverb);\n";
+const fxExtensions = "\n// Various noise type processors for the MonoNoise source\n// Type 2 is Pink noise, used from Tone.Noise('pink') instead of calc\n//\nclass NoiseProcessor extends AudioWorkletProcessor {\n\tstatic get parameterDescriptors(){\n\t\treturn [{\n\t\t\tname: 'type',\n\t\t\tdefaultValue: 5,\n\t\t\tminValue: 0,\n\t\t\tmaxValue: 5\n\t\t},{\n\t\t\tname: 'density',\n\t\t\tdefaultValue: 0.125,\n\t\t\tminValue: 0,\n\t\t\tmaxValue: 1\n\t\t}];\n\t}\n\t\n\tconstructor(){\n\t\tsuper();\n\t\t// sample previous value\n\t\tthis.prev = 0;\n\t\t// latch to a sample \n\t\tthis.latch = 0;\n\t\t// phasor ramp\n\t\tthis.phasor = 0;\n\t\tthis.delta = 0;\n\t}\n\n\tprocess(inputs, outputs, parameters){\n\t\t// input is not used because this is a source\n\t\tconst input = inputs[0];\n\t\tconst output = outputs[0];\n\t\tconst HALF_PI = Math.PI/2;\n\n\t\t// for one output channel generate some noise\t\n\t\tif (input.length > 0){\n\t\t\tfor (let i = 0; i < input[0].length; i++){\n\t\t\t\tconst t = (parameters.type.length > 1) ? parameters.type[i] : parameters.type[0];\n\t\t\t\tconst d = (parameters.density.length > 1) ? parameters.density[i] : parameters.density[0];\n\t\t\t\n\t\t\t\t// some bipolar white noise -1 to 1\n\t\t\t\tconst biNoise = Math.random() * 2 - 1;\n\t\t\t\t// empty output\n\t\t\t\tlet out = 0;\n\n\t\t\t\t// White noise, Use for every other choice\n\t\t\t\tif (t < 1){\n\t\t\t\t\tout = biNoise * 0.707;\n\t\t\t\t}\n\t\t\t\t// Pink noise,  use Tone.Noise('pink') object for simplicity\n\t\t\t\telse if (t < 2){\n\t\t\t\t\tout = input[0][i] * 1.413;\n\t\t\t\t}\n\t\t\t\t// Brownian noise\n\t\t\t\t// calculate a random next value in \"step size\" and add to \n\t\t\t\t// the previous noise signal value creating a \"drunk walk\" \n\t\t\t\t// or brownian motion\n\t\t\t\telse if (t < 3){\t\t\n\t\t\t\t\tthis.prev += biNoise * d*d;\n\t\t\t\t\tthis.prev = Math.asin(Math.sin(this.prev * HALF_PI)) / HALF_PI;\n\t\t\t\t\tout = this.prev * 0.707;\n\t\t\t\t}\n\t\t\t\t// Lo-Fi (sampled) noise\n\t\t\t\t// creates random values at a specified frequency and slowly \n\t\t\t\t// ramps to that new value\n\t\t\t\telse if (t < 4){\n\t\t\t\t\t// create a ramp from 0-1 at specific frequency/density\n\t\t\t\t\tthis.phasor = (this.phasor + d * d * 0.5) % 1;\n\t\t\t\t\t// calculate the delta\n\t\t\t\t\tlet dlt = this.phasor - this.delta;\n\t\t\t\t\tthis.delta = this.phasor;\n\t\t\t\t\t// when ramp resets, latch a new noise value\n\t\t\t\t\tif (dlt < 0){\n\t\t\t\t\t\tthis.prev = this.latch;\n\t\t\t\t\t\tthis.latch = biNoise;\n\t\t\t\t\t}\n\t\t\t\t\t// linear interpolation from previous to next point\n\t\t\t\t\tout = this.prev + this.phasor * (this.latch - this.prev);\n\t\t\t\t\tout *= 0.707;\n\t\t\t\t}\n\t\t\t\t// Dust noise\n\t\t\t\t// randomly generate an impulse/click of value 1 depending \n\t\t\t\t// on the density, average amount of impulses per second\n\t\t\t\telse if (t < 5){\n\t\t\t\t\tout = Math.random() > (1 - d*d*d * 0.5);\n\t\t\t\t}\n\t\t\t\t// Crackle noise\n\t\t\t\t// Pink generator with \"wave-loss\" leaving gaps\n\t\t\t\telse {\n\t\t\t\t\tlet delta = input[0][i] - this.prev;\n\t\t\t\t\tthis.prev = input[0][i];\n\t\t\t\t\tif (delta > 0){\n\t\t\t\t\t\tthis.latch = Math.random();\n\t\t\t\t\t}\n\t\t\t\t\tout = (this.latch < (1 - d*d*d)) ? 0 : input[0][i] * 1.413;\n\t\t\t\t}\n\t\t\t\t// send to output whichever noise type was chosen\n\t\t\t\toutput[0][i] = out;\n\t\t\t}\n\t\t}\t\t\n\t\treturn true;\n\t}\n}\nregisterProcessor('noise-processor', NoiseProcessor);\n\n// A Downsampling Chiptune effect. Downsamples the signal by a specified amount\n// Resulting in a lower samplerate, making it sound more like 8bit/chiptune\n// Programmed with a custom AudioWorkletProcessor, see effects/Processors.js\n//\nclass DownSampleProcessor extends AudioWorkletProcessor {\n\tstatic get parameterDescriptors() {\n\t\treturn [{\n\t\t\tname: 'down',\n\t\t\tdefaultValue: 8,\n\t\t\tminValue: 1,\n\t\t\tmaxValue: 2048\n\t\t}];\n\t}\n\n\tconstructor(){\n\t\tsuper();\n\t\t// the frame counter\n\t\tthis.count = 0;\n\t\t// sample and hold variable array\n\t\tthis.sah = [];\n\t}\n\n\tprocess(inputs, outputs, parameters){\n\t\tconst input = inputs[0];\n\t\tconst output = outputs[0];\n\n\t\t// if there is anything to process\n\t\tif (input.length > 0){\n\t\t\t// for the length of the sample array (generally 128)\n\t\t\tfor (let i=0; i<input[0].length; i++){\n\t\t\t\tconst d = (parameters.down.length > 1) ? parameters.down[i] : parameters.down[0];\n\t\t\t\t// for every channel\n\t\t\t\tfor (let channel=0; channel<input.length; ++channel){\n\t\t\t\t\t// if counter equals 0, sample and hold\n\t\t\t\t\tif (this.count % d === 0){\n\t\t\t\t\t\tthis.sah[channel] = input[channel][i];\n\t\t\t\t\t}\n\t\t\t\t\t// output the currently held sample\n\t\t\t\t\toutput[channel][i] = this.sah[channel];\n\t\t\t\t}\n\t\t\t\t// increment sample counter\n\t\t\t\tthis.count++;\n\t\t\t}\n\t\t}\n\t\treturn true;\n\t}\n}\nregisterProcessor('downsampler-processor', DownSampleProcessor);\n\n// A distortion algorithm using the tanh (hyperbolic-tangent) as a \n// waveshaping technique. Some mapping to apply a more equal loudness \n// distortion is applied on the overdrive parameter\n//\nclass TanhDistortionProcessor extends AudioWorkletProcessor {\n\tstatic get parameterDescriptors(){\n\t\treturn [{\n\t\t\tname: 'amount',\n\t\t\tdefaultValue: 4,\n\t\t\tminValue: 1\n\t\t}, {\n\t\t\tname: 'makeup',\n\t\t\tdefaultValue: 0.5,\n\t\t\tminValue: 0,\n\t\t\tmaxValue: 2\n\t\t}]\n\t}\n\n\tconstructor(){\n\t\tsuper();\n\t}\n\n\tprocess(inputs, outputs, parameters){\n\t\tconst input = inputs[0];\n\t\tconst output = outputs[0];\n\n\t\tif (input.length > 0){\n\t\t\tfor (let channel=0; channel<input.length; ++channel){\n\t\t\t\tfor (let i=0; i<input[channel].length; i++){\n\t\t\t\t\tconst a = (parameters.amount.length > 1)? parameters.amount[i] : parameters.amount[0];\n\t\t\t\t\tconst m = (parameters.makeup.length > 1)? parameters.makeup[i] : parameters.makeup[0];\n\t\t\t\t\t// simple waveshaping with tanh\n\t\t\t\t\toutput[channel][i] = Math.tanh(input[channel][i] * a) * m;\n\t\t\t\t}\n\t\t\t}\n\t\t}\n\t\treturn true;\n\t}\n}\nregisterProcessor('tanh-distortion-processor', TanhDistortionProcessor);\n\n// A distortion algorithm using the arctan function as a \n// waveshaping technique. Some mapping to apply a more equal loudness \n// distortion is applied on the overdrive parameter\n//\nclass ArctanDistortionProcessor extends AudioWorkletProcessor {\n\tstatic get parameterDescriptors(){\n\t\treturn [{\n\t\t\tname: 'amount',\n\t\t\tdefaultValue: 5,\n\t\t\tminValue: 1\n\t\t}]\n\t}\n\n\tconstructor(){\n\t\tsuper();\n\n\t\t// quarter pi constant and inverse\n\t\tthis.Q_PI = 0.7853981633974483; // 0.25 * Math.PI;\n\t\tthis.INVQ_PI = 1.2732395447351628; //1.0 / this.Q_PI;\n\t}\n\n\tprocess(inputs, outputs, parameters){\n\t\tconst input = inputs[0];\n\t\tconst output = outputs[0];\n\n\t\tconst gain = parameters.amount[0];\n\t\tconst makeup = Math.min(1, Math.max(0, 1 - ((Math.atan(gain) - this.Q_PI) * this.INVQ_PI * 0.823)));\n\n\t\tif (input.length > 0){\n\t\t\tfor (let channel=0; channel<input.length; channel++){\n\t\t\t\tfor (let i=0; i<input[channel].length; i++){\n\t\t\t\t\toutput[channel][i] = Math.atan(input[channel][i] * gain) * makeup;\n\t\t\t\t}\n\t\t\t}\n\t\t}\n\t\treturn true;\n\t}\n}\nregisterProcessor('arctan-distortion-processor', ArctanDistortionProcessor);\n\n\n// A fuzz distortion effect in modelled after the Big Muff Pi pedal \n// by Electro Harmonics. Using three stages of distortion: \n// 1 soft-clipping stage, 2 half-wave rectifier, 3 hard-clipping stage\n// Based on: https://github.com/hazza-music/EHX-Big-Muff-Pi-Emulation/blob/main/Technical%20Essay.pdf\n// \nclass FuzzProcessor extends AudioWorkletProcessor {\n\tstatic get parameterDescriptors() {\n\t\treturn [{\n\t\t\tname: 'amount',\n\t\t\tdefaultValue: 5,\n\t\t\tminValue: 1\n\t\t}]\n\t}\n\n\tconstructor(){ \n\t\tsuper(); \n\t\t// history for onepole filter for dcblocking\n\t\tthis.history = [0, 0];\n\t}\n\n\tprocess(inputs, outputs, parameters){\n\t\tconst input = inputs[0];\n\t\tconst output = outputs[0];\n\n\t\tconst gain = parameters.amount[0];\n\t\tconst makeup = Math.max((1 - Math.pow((gain-1) / 63, 0.13)) * 0.395 + 0.605, 0.605);\n\n\t\tif (input.length > 0){\n\t\t\tfor (let channel = 0; channel < input.length; channel++){\n\t\t\t\tfor (let i = 0; i < input[channel].length; i++){\n\t\t\t\t\t// soft-clipping\n\t\t\t\t\tconst sc = Math.atan(input[channel][i] * gain * 2) * 0.6;\n\t\t\t\t\t// half-wave rectification and add for \n\t\t\t\t\t// asymmetric distortion\n\t\t\t\t\tconst hw = ((sc > 0) ? sc : 0) + input[channel][i];\n\t\t\t\t\t// hard-clipping\n\t\t\t\t\tconst hc = Math.max(-0.707, Math.min(0.707, hw));\n\t\t\t\t\t// onepole lowpass filter for dc-block\n\t\t\t\t\tthis.history[channel] = (hc - this.history[channel]) * 0.0015 + this.history[channel];\n\t\t\t\t\t// dc-block and gain compensation and output\n\t\t\t\t\toutput[channel][i] = (hc - this.history[channel]) * makeup;\n\t\t\t\t}\n\t\t\t}\n\t\t}\n\t\treturn true;\n\t}\n}\nregisterProcessor('fuzz-processor', FuzzProcessor);\n\n// A distortion/compression effect of an incoming signal\n// Based on an algorithm by Peter McCulloch\n// \nclass SquashProcessor extends AudioWorkletProcessor {\n\tstatic get parameterDescriptors(){\n\t\treturn [{\n\t\t\tname: 'amount',\n\t\t\tdefaultValue: 4,\n\t\t\tminValue: 1,\n\t\t\tmaxValue: 1024\n\t\t}, {\n\t\t\tname: 'makeup',\n\t\t\tdefaultValue: 0.5,\n\t\t\tminValue: 0,\n\t\t\tmaxValue: 2\n\t\t}];\n\t}\n\n\tconstructor(){\n\t\tsuper();\n\t}\n\n\tprocess(inputs, outputs, parameters){\n\t\tconst input = inputs[0];\n\t\tconst output = outputs[0];\n\t\t\n\t\tif (input.length > 0){\n\t\t\tfor (let channel=0; channel<input.length; ++channel){\n\t\t\t\tfor (let i=0; i<input[channel].length; i++){\n\t\t\t\t\t// (s * a) / ((s * a)^2 * 0.28 + 1) / √a\n\t\t\t\t\t// drive amount, minimum of 1\n\t\t\t\t\tconst a = (parameters.amount.length > 1)? parameters.amount[i] : parameters.amount[0];\n\t\t\t\t\t// makeup gain\n\t\t\t\t\tconst m = (parameters.makeup.length > 1)? parameters.makeup[i] : parameters.makeup[0];\n\t\t\t\t\t// set the waveshaper effect\n\t\t\t\t\tconst s = input[channel][i];\n\t\t\t\t\tconst x = s * a * 1.412;\n\t\t\t\t\toutput[channel][i] = (x / (x * x * 0.28 + 1.0)) * m * 0.708;\n\t\t\t\t}\n\t\t\t}\n\t\t}\n\t\treturn true;\n\t}\n}\nregisterProcessor('squash-processor', SquashProcessor);\n\n// Comb Filter processor\n// A LowPass FeedBack CombFilter effect (LBCF)\n// Uses a onepole lowpass filter in the feedback delay for damping\n// Feedback amount can be positive or negative \n// (negative creates odd harmonics one octave lower)\n// \nclass CombFilterProcessor extends AudioWorkletProcessor {\n\tstatic get parameterDescriptors() {\n\t\treturn [\n\t\t\t[ 'time', 5, 0, 120, \"k-rate\" ],\n\t\t\t[ 'feedback', 0.8, -0.999, 0.999, \"k-rate\" ],\n\t\t\t[ 'damping', 0.5, 0, 1, \"k-rate\" ],\n\t\t\t[ 'drywet', 0.8, 0, 1, \"k-rate\" ]\n\t\t].map(x => new Object({\n\t\t\tname: x[0],\n\t\t\tdefaultValue: x[1],\n\t\t\tminValue: x[2],\n\t\t\tmaxValue: x[3],\n\t\t\tautomationRate: x[4]\n\t\t}));\n\t}\n\t\n\tconstructor(info) {\n\t\tsuper();\n\n\t\tconst numChannels = info.channelCount;\n\t\tconst delaySize = 120;\n\t\t// make delays for amount of channels and\n\t\t// initialize history values for lowpass\n\t\tthis.delays = [];\n\t\tthis.lpf = [];\n\t\tfor (let i = 0; i < numChannels; i++){\n\t\t\tthis.delays[i] = this.makeDelay(delaySize);\n\t\t\tthis.lpf[i] = 0;\n\t\t}\n\t}\n\n\t// makeDelay code based on Dattorro Reverberator delays\n\t// Thanks to khoin: https://github.com/khoin\n\tmakeDelay(length) {\n\t\tlet size = Math.round(length * 0.001 * sampleRate);\n\t\tlet nextPow2 = 2 ** Math.ceil(Math.log2((size)));\n\t\treturn [\n\t\t\tnew Float32Array(nextPow2), nextPow2-1, 0, nextPow2 - 1\n\t\t];\n\t}\n\t// write to specific delayline at delaysize\n\twriteDelay(i, data) {\n\t\treturn this.delays[i][0][this.delays[i][1]] = data;\n\t}\n\n\t// read from delayline at specified time\n\treadDelayAt(i, ms) {\n\t\tlet s = Math.round(ms * 0.001 * sampleRate);\n\t\treturn this.delays[i][0][(this.delays[i][2] - s) & this.delays[i][3]];\n\t}\n\n\t// move the read and writeheads of the delayline\n\tupdateReadWriteHeads(i){\n\t\t// increment read and write heads in delay and wrap at delaysize\n\t\tthis.delays[i][1] = (this.delays[i][1] + 1) & this.delays[i][3];\n\t\tthis.delays[i][2] = (this.delays[i][2] + 1) & this.delays[i][3];\n\t}\n\n\tprocess(inputs, outputs, parameters){\n\t\tconst input = inputs[0];\n\t\tconst output = outputs[0];\n\n\t\tconst dt = parameters.time[0];\n\t\tconst fb = parameters.feedback[0];\n\t\tconst dm = Math.max(0, parameters.damping[0]);\n\t\tconst dw = parameters.drywet[0];\n\n\t\t// process for every channel and every sample in the channel\n\t\tif (input.length > 0){\n\t\t\tfor (let channel = 0; channel < input.length; channel++){\n\t\t\t\tfor (let i = 0; i < input[0].length; i++){\n\t\t\t\t\t// a onepole lowpass filter after delay\n\t\t\t\t\tthis.lpf[channel] = this.readDelayAt(channel, dt) * (1 - dm) + this.lpf[channel] * dm;\n\t\t\t\t\t// write to the delayline \n\t\t\t\t\tthis.writeDelay(channel, input[channel][i] + this.lpf[channel] * fb);\n\t\t\t\t\t// apply drywet and send output from the filter\n\t\t\t\t\toutput[channel][i] = this.lpf[channel] * dw + input[channel][i] * (1-dw);\n\t\t\t\t\t// update the read and write heads of the delaylines\n\t\t\t\t\tthis.updateReadWriteHeads(channel);\n\t\t\t\t}\n\t\t\t}\n\t\t}\n\t\treturn true;\n\t}\n}\nregisterProcessor('combfilter-processor', CombFilterProcessor);\n\n// Dattorro Reverberator\n// Thanks to port by khoin, taken from:\n// https://github.com/khoin/DattorroReverbNode\n// based on the paper from Jon Dattorro:\n// https://ccrma.stanford.edu/~dattorro/EffectDesignPart1.pdf\n// with small modifications to work in Mercury\n//\n// In jurisdictions that recognize copyright laws, this software is to\n// be released into the public domain.\n\n// THE SOFTWARE IS PROVIDED \"AS IS\", WITHOUT WARRANTY OF ANY KIND.\n// THE AUTHOR(S) SHALL NOT BE LIABLE FOR ANYTHING, ARISING FROM, OR IN\n// CONNECTION WITH THE SOFTWARE OR THE DISTRIBUTION OF THE SOFTWARE.\n// \nclass DattorroReverb extends AudioWorkletProcessor {\n\tstatic get parameterDescriptors() {\n\t\treturn [\n\t\t\t[\"preDelay\", 0, 0, sampleRate - 1, \"k-rate\"],\n\t\t\t// [\"bandwidth\", 0.9999, 0, 1, \"k-rate\"],\t\n\t\t\t[\"inputDiffusion1\", 0.75, 0, 1, \"k-rate\"],\n\t\t\t[\"inputDiffusion2\", 0.625, 0, 1, \"k-rate\"],\n\t\t\t[\"decay\", 0.5, 0, 1, \"k-rate\"],\n\t\t\t[\"decayDiffusion1\", 0.7, 0, 0.999999, \"k-rate\"],\n\t\t\t[\"decayDiffusion2\", 0.5, 0, 0.999999, \"k-rate\"],\n\t\t\t[\"damping\", 0.005, 0, 1, \"k-rate\"],\n\t\t\t[\"excursionRate\", 0.5, 0, 2, \"k-rate\"],\n\t\t\t[\"excursionDepth\", 0.7, 0, 2, \"k-rate\"],\n\t\t\t[\"wet\", 0.7, 0, 2, \"k-rate\"],\n\t\t\t// [\"dry\", 0.7, 0, 2, \"k-rate\"]\n\t\t].map(x => new Object({\n\t\t\tname: x[0],\n\t\t\tdefaultValue: x[1],\n\t\t\tminValue: x[2],\n\t\t\tmaxValue: x[3],\n\t\t\tautomationRate: x[4]\n\t\t}));\n\t}\n\n\tconstructor(options) {\n\t\tsuper(options);\n\n\t\tthis._Delays = [];\n\t\t// Pre-delay is always one-second long, rounded to the nearest 128-chunk\n\t\tthis._pDLength = sampleRate + (128 - sampleRate % 128);\n\t\tthis._preDelay = new Float32Array(this._pDLength);\n\t\tthis._pDWrite = 0;\n\t\tthis._lp1 = 0.0;\n\t\tthis._lp2 = 0.0;\n\t\tthis._lp3 = 0.0;\n\t\tthis._excPhase = 0.0;\n\n\t\t[\n\t\t\t0.004771345, 0.003595309, 0.012734787, 0.009307483, // pre-tank\n\t\t\t0.022579886, 0.149625349, 0.060481839, 0.1249958, // left-loop\n\t\t\t0.030509727, 0.141695508, 0.089244313, 0.106280031 // right-loop\n\t\t].forEach(x => this.makeDelay(x));\n\n\t\tthis._taps = Int16Array.from([\n\t\t\t0.008937872, 0.099929438, 0.064278754, 0.067067639, \n\t\t\t0.066866033, 0.006283391, 0.035818689, // left-output\n\t\t\t0.011861161, 0.121870905, 0.041262054, 0.08981553, \n\t\t\t0.070931756, 0.011256342, 0.004065724 // right-output\n\t\t], x => Math.round(x * sampleRate));\n\t}\n\n\tmakeDelay(length) {\n\t\t// len, array, write, read, mask\n\t\tlet len = Math.round(length * sampleRate);\n\t\tlet nextPow2 = 2 ** Math.ceil(Math.log2((len)));\n\t\tthis._Delays.push([\n\t\t\tnew Float32Array(nextPow2), len - 1, 0 | 0, nextPow2 - 1\n\t\t]);\n\t}\n\n\twriteDelay(index, data) {\n\t\treturn this._Delays[index][0][this._Delays[index][1]] = data;\n\t}\n\n\treadDelay(index) {\n\t\treturn this._Delays[index][0][this._Delays[index][2]];\n\t}\n\n\treadDelayAt(index, i) {\n\t\tlet d = this._Delays[index];\n\t\treturn d[0][(d[2] + i) & d[3]];\n\t}\n\n\t// cubic interpolation\n\t// O. Niemitalo: \n\t// https://www.musicdsp.org/en/latest/Other/49-cubic-interpollation.html\n\treadDelayCAt(index, i) {\n\t\tlet d = this._Delays[index],\n\t\t\tfrac = i - ~~i,\n\t\t\tint = ~~i + d[2] - 1,\n\t\t\tmask = d[3];\n\n\t\tlet x0 = d[0][int++ & mask],\n\t\t\tx1 = d[0][int++ & mask],\n\t\t\tx2 = d[0][int++ & mask],\n\t\t\tx3 = d[0][int & mask];\n\n\t\tlet a = (3 * (x1 - x2) - x0 + x3) / 2,\n\t\t\tb = 2 * x2 + x0 - (5 * x1 + x3) / 2,\n\t\t\tc = (x2 - x0) / 2;\n\n\t\treturn (((a * frac) + b) * frac + c) * frac + x1;\n\t}\n\n\t// First input will be downmixed to mono if number of channels is not 2\n\t// Outputs Stereo.\n\tprocess(inputs, outputs, parameters) {\n\t\tconst pd = ~~parameters.preDelay[0],\n\t\t\t// bw = parameters.bandwidth[0], // replaced by using damping\n\t\t\tfi = parameters.inputDiffusion1[0],\n\t\t\tsi = parameters.inputDiffusion2[0],\n\t\t\tdc = parameters.decay[0],\n\t\t\tft = parameters.decayDiffusion1[0],\n\t\t\tst = parameters.decayDiffusion2[0],\n\t\t\tdp = 1 - parameters.damping[0],\n\t\t\tex = parameters.excursionRate[0] / sampleRate,\n\t\t\ted = parameters.excursionDepth[0] * sampleRate / 1000,\n\t\t\twe = parameters.wet[0]; //* 0.6, // lo & ro both mult. by 0.6 anyways\n\t\t\t// dr = parameters.dry[0];\n\n\t\t// write to predelay and dry output\n\t\tif (inputs[0].length == 2) {\n\t\t\tfor (let i = 127; i >= 0; i--) {\n\t\t\t\tthis._preDelay[this._pDWrite + i] = (inputs[0][0][i] + inputs[0][1][i]) * 0.5;\n\n\t\t\t\t// removed the dry parameter, this is handled in the Tone Node\n\t\t\t\t// outputs[0][0][i] = inputs[0][0][i] * dr;\n\t\t\t\t// outputs[0][1][i] = inputs[0][1][i] * dr;\n\t\t\t}\n\t\t} else if (inputs[0].length > 0) {\n\t\t\tthis._preDelay.set(\n\t\t\t\tinputs[0][0],\n\t\t\t\tthis._pDWrite\n\t\t\t);\n\t\t\t// for (let i = 127; i >= 0; i--)\n\t\t\t// \toutputs[0][0][i] = outputs[0][1][i] = inputs[0][0][i] * dr;\n\t\t} else {\n\t\t\tthis._preDelay.set(\n\t\t\t\tnew Float32Array(128),\n\t\t\t\tthis._pDWrite\n\t\t\t);\n\t\t}\n\n\t\tlet i = 0 | 0;\n\t\twhile (i < 128) {\n\t\t\tlet lo = 0.0,\n\t\t\t\tro = 0.0;\n\n\t\t\t// input damping (formerly known as bandwidth bw, now uses dp)\n\t\t\tthis._lp1 += dp * (this._preDelay[(this._pDLength + this._pDWrite - pd + i) % this._pDLength] - this._lp1);\n\n\t\t\t// pre-tank\n\t\t\tlet pre = this.writeDelay(0, this._lp1 - fi * this.readDelay(0));\n\t\t\tpre = this.writeDelay(1, fi * (pre - this.readDelay(1)) + this.readDelay(0));\n\t\t\tpre = this.writeDelay(2, fi * pre + this.readDelay(1) - si * this.readDelay(2));\n\t\t\tpre = this.writeDelay(3, si * (pre - this.readDelay(3)) + this.readDelay(2));\n\n\t\t\tlet split = si * pre + this.readDelay(3);\n\n\t\t\t// excursions\n\t\t\t// could be optimized?\n\t\t\tlet exc = ed * (1 + Math.cos(this._excPhase * 6.2800));\n\t\t\tlet exc2 = ed * (1 + Math.sin(this._excPhase * 6.2847));\n\n\t\t\t// left loop\n\t\t\t// tank diffuse 1\n\t\t\tlet temp = this.writeDelay(4, split + dc * this.readDelay(11) + ft * this.readDelayCAt(4, exc));\n\t\t\t// long delay 1\n\t\t\tthis.writeDelay(5, this.readDelayCAt(4, exc) - ft * temp);\n\t\t\t// damp 1\n\t\t\tthis._lp2 += dp * (this.readDelay(5) - this._lp2);\n\t\t\ttemp = this.writeDelay(6, dc * this._lp2 - st * this.readDelay(6)); // tank diffuse 2\n\t\t\t// long delay 2\n\t\t\tthis.writeDelay(7, this.readDelay(6) + st * temp);\n\n\t\t\t// right loop \n\t\t\t// tank diffuse 3\n\t\t\ttemp = this.writeDelay(8, split + dc * this.readDelay(7) + ft * this.readDelayCAt(8, exc2));\n\t\t\t// long delay 3\n\t\t\tthis.writeDelay(9, this.readDelayCAt(8, exc2) - ft * temp);\n\t\t\t// damp 2\n\t\t\tthis._lp3 += dp * (this.readDelay(9) - this._lp3);\n\t\t\t// tank diffuse 4\n\t\t\ttemp = this.writeDelay(10, dc * this._lp3 - st * this.readDelay(10));\n\t\t\t// long delay 4\n\t\t\tthis.writeDelay(11, this.readDelay(10) + st * temp);\n\n\t\t\tlo = this.readDelayAt(9, this._taps[0]) +\n\t\t\t\tthis.readDelayAt(9, this._taps[1]) -\n\t\t\t\tthis.readDelayAt(10, this._taps[2]) +\n\t\t\t\tthis.readDelayAt(11, this._taps[3]) -\n\t\t\t\tthis.readDelayAt(5, this._taps[4]) -\n\t\t\t\tthis.readDelayAt(6, this._taps[5]) -\n\t\t\t\tthis.readDelayAt(7, this._taps[6]);\n\n\t\t\tro = this.readDelayAt(5, this._taps[7]) +\n\t\t\t\tthis.readDelayAt(5, this._taps[8]) -\n\t\t\t\tthis.readDelayAt(6, this._taps[9]) +\n\t\t\t\tthis.readDelayAt(7, this._taps[10]) -\n\t\t\t\tthis.readDelayAt(9, this._taps[11]) -\n\t\t\t\tthis.readDelayAt(10, this._taps[12]) -\n\t\t\t\tthis.readDelayAt(11, this._taps[13]);\n\n\t\t\toutputs[0][0][i] += lo * we;\n\t\t\toutputs[0][1][i] += ro * we;\n\n\t\t\tthis._excPhase += ex;\n\n\t\t\ti++;\n\n\t\t\tfor (let j = 0, d = this._Delays[0]; j < this._Delays.length; d = this._Delays[++j]) {\n\t\t\t\td[1] = (d[1] + 1) & d[3];\n\t\t\t\td[2] = (d[2] + 1) & d[3];\n\t\t\t}\n\t\t}\n\n\t\t// Update preDelay index\n\t\tthis._pDWrite = (this._pDWrite + 128) % this._pDLength;\n\n\t\treturn true;\n\t}\n}\nregisterProcessor('dattorro-reverb', DattorroReverb);\n";
 Tone.getContext().addAudioWorkletModule(URL.createObjectURL(new Blob([ fxExtensions ], { type: 'text/javascript' })));
 
 // Mercury main class controls Tone and loads samples