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Replacing old reverb by better convolution
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@ -1,23 +1,27 @@
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if (typeof AudioContext !== 'undefined') {
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import reverbGen from './reverbGen.mjs';
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AudioContext.prototype.impulseResponse = function (duration, channels = 1) {
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const length = this.sampleRate * duration;
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const impulse = this.createBuffer(channels, length, this.sampleRate);
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const IR = impulse.getChannelData(0);
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for (let i = 0; i < length; i++) IR[i] = (2 * Math.random() - 1) * Math.pow(1 - i / length, duration);
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return impulse;
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};
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AudioContext.prototype.createReverb = function (duration) {
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if (typeof AudioContext !== 'undefined') {
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AudioContext.prototype.generateReverb = reverbGen.generateReverb;
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AudioContext.prototype.createReverb = function(duration, audioContext) {
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const convolver = this.createConvolver();
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const convolver = this.createConvolver();
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convolver.setDuration = (d) => {
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convolver.setDuration = (d) => {
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convolver.buffer = this.impulseResponse(d);
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this.generateReverb(
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convolver.duration = duration;
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{
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return convolver;
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audioContext,
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sampleRate: 44100,
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numChannels: 2,
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decayTime: d,
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fadeInTime: d,
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lpFreqStart: 2000,
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lpFreqEnd: 15000,
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},
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(buffer) => {
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convolver.buffer = buffer;
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}
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);
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convolver.duration = d;
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};
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};
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convolver.setDuration(duration);
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convolver.setDuration(duration);
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return convolver;
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return convolver;
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};
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};
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}
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}
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// TODO: make the reverb more exciting
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// check out https://blog.gskinner.com/archives/2019/02/reverb-web-audio-api.html
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209
packages/superdough/reverbGen.mjs
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209
packages/superdough/reverbGen.mjs
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@ -0,0 +1,209 @@
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// Copyright 2014 Alan deLespinasse
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//
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// Licensed under the Apache License, Version 2.0 (the "License");
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// you may not use this file except in compliance with the License.
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// You may obtain a copy of the License at
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//
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// http://www.apache.org/licenses/LICENSE-2.0
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//
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// Unless required by applicable law or agreed to in writing, software
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// distributed under the License is distributed on an "AS IS" BASIS,
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// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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// See the License for the specific language governing permissions and
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// limitations under the License.
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"use strict";
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var reverbGen = {};
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/** Generates a reverb impulse response.
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@param {!Object} params TODO: Document the properties.
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@param {!function(!AudioBuffer)} callback Function to call when
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the impulse response has been generated. The impulse response
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is passed to this function as its parameter. May be called
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immediately within the current execution context, or later. */
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reverbGen.generateReverb = function(params, callback) {
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var audioContext = params.audioContext || new AudioContext();
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var sampleRate = params.sampleRate || 44100;
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var numChannels = params.numChannels || 2;
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// params.decayTime is the -60dB fade time. We let it go 50% longer to get to -90dB.
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var totalTime = params.decayTime * 1.5;
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var decaySampleFrames = Math.round(params.decayTime * sampleRate);
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var numSampleFrames = Math.round(totalTime * sampleRate);
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var fadeInSampleFrames = Math.round((params.fadeInTime || 0) * sampleRate);
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// 60dB is a factor of 1 million in power, or 1000 in amplitude.
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var decayBase = Math.pow(1 / 1000, 1 / decaySampleFrames);
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var reverbIR = audioContext.createBuffer(numChannels, numSampleFrames, sampleRate);
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for (var i = 0; i < numChannels; i++) {
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var chan = reverbIR.getChannelData(i);
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for (var j = 0; j < numSampleFrames; j++) {
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chan[j] = randomSample() * Math.pow(decayBase, j);
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}
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for (var j = 0; j < fadeInSampleFrames; j++) {
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chan[j] *= (j / fadeInSampleFrames);
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}
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}
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applyGradualLowpass(reverbIR, params.lpFreqStart || 0, params.lpFreqEnd || 0, params.decayTime, callback);
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};
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/** Creates a canvas element showing a graph of the given data.
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@param {!Float32Array} data An array of numbers, or a Float32Array.
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@param {number} width Width in pixels of the canvas.
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@param {number} height Height in pixels of the canvas.
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@param {number} min Minimum value of data for the graph (lower edge).
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@param {number} max Maximum value of data in the graph (upper edge).
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@return {!CanvasElement} The generated canvas element. */
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reverbGen.generateGraph = function(data, width, height, min, max) {
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var canvas = document.createElement('canvas');
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canvas.width = width;
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canvas.height = height;
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var gc = canvas.getContext('2d');
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gc.fillStyle = '#000';
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gc.fillRect(0, 0, canvas.width, canvas.height);
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gc.fillStyle = '#fff';
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var xscale = width / data.length;
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var yscale = height / (max - min);
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for (var i = 0; i < data.length; i++) {
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gc.fillRect(i * xscale, height - (data[i] - min) * yscale, 1, 1);
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}
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return canvas;
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}
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/** Saves an AudioBuffer as a 16-bit WAV file on the client's host
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file system. Normalizes it to peak at +-32767, and optionally
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truncates it if there's a lot of "silence" at the end.
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@param {!AudioBuffer} buffer The buffer to save.
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@param {string} name Name of file to create.
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@param {number?} opt_minTail Defines what counts as "silence" for
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auto-truncating the buffer. If there is a point past which every
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value of every channel is less than opt_minTail, then the buffer
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is truncated at that point. This is expressed as an integer,
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applying to the post-normalized and integer-converted
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buffer. The default is 0, meaning don't truncate. */
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reverbGen.saveWavFile = function(buffer, name, opt_minTail) {
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var bitsPerSample = 16;
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var bytesPerSample = 2;
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var sampleRate = buffer.sampleRate;
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var numChannels = buffer.numberOfChannels;
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var channels = getAllChannelData(buffer);
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var numSampleFrames = channels[0].length;
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var scale = 32767;
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// Find normalization constant.
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var max = 0;
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for (var i = 0; i < numChannels; i++) {
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for (var j = 0; j < numSampleFrames; j++) {
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max = Math.max(max, Math.abs(channels[i][j]));
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}
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}
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if (max) {
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scale = 32767 / max;
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}
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// Find truncation point.
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if (opt_minTail) {
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var truncateAt = 0;
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for (var i = 0; i < numChannels; i++) {
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for (var j = 0; j < numSampleFrames; j++) {
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var absSample = Math.abs(Math.round(scale * channels[i][j]));
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if (absSample > opt_minTail) {
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truncateAt = j;
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}
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}
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}
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numSampleFrames = truncateAt + 1;
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}
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var sampleDataBytes = bytesPerSample * numChannels * numSampleFrames;
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var fileBytes = sampleDataBytes + 44;
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var arrayBuffer = new ArrayBuffer(fileBytes);
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var dataView = new DataView(arrayBuffer);
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dataView.setUint32(0, 1179011410, true); // "RIFF"
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dataView.setUint32(4, fileBytes - 8, true); // file length
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dataView.setUint32(8, 1163280727, true); // "WAVE"
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dataView.setUint32(12, 544501094, true); // "fmt "
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dataView.setUint32(16, 16, true) // fmt chunk length
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dataView.setUint16(20, 1, true); // PCM format
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dataView.setUint16(22, numChannels, true); // NumChannels
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dataView.setUint32(24, sampleRate, true); // SampleRate
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var bytesPerSampleFrame = numChannels * bytesPerSample;
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dataView.setUint32(28, sampleRate * bytesPerSampleFrame, true); // ByteRate
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dataView.setUint16(32, bytesPerSampleFrame, true); // BlockAlign
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dataView.setUint16(34, bitsPerSample, true); // BitsPerSample
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dataView.setUint32(36, 1635017060, true); // "data"
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dataView.setUint32(40, sampleDataBytes, true);
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for (var j = 0; j < numSampleFrames; j++) {
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for (var i = 0; i < numChannels; i++) {
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dataView.setInt16(44 + j * bytesPerSampleFrame + i * bytesPerSample,
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Math.round(scale * channels[i][j]), true);
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}
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}
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var blob = new Blob([arrayBuffer], { 'type': 'audio/wav' });
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var url = window.URL.createObjectURL(blob);
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var linkEl = document.createElement('a');
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linkEl.href = url;
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linkEl.download = name;
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linkEl.style.display = 'none';
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document.body.appendChild(linkEl);
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linkEl.click();
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};
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/** Applies a constantly changing lowpass filter to the given sound.
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@private
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@param {!AudioBuffer} input
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@param {number} lpFreqStart
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@param {number} lpFreqEnd
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@param {number} lpFreqEndAt
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@param {!function(!AudioBuffer)} callback May be called
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immediately within the current execution context, or later.*/
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var applyGradualLowpass = function(input, lpFreqStart, lpFreqEnd, lpFreqEndAt, callback) {
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if (lpFreqStart == 0) {
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callback(input);
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return;
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}
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var channelData = getAllChannelData(input);
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var context = new OfflineAudioContext(input.numberOfChannels, channelData[0].length, input.sampleRate);
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var player = context.createBufferSource();
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player.buffer = input;
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var filter = context.createBiquadFilter();
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lpFreqStart = Math.min(lpFreqStart, input.sampleRate / 2);
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lpFreqEnd = Math.min(lpFreqEnd, input.sampleRate / 2);
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filter.type = "lowpass";
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filter.Q.value = 0.0001;
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filter.frequency.setValueAtTime(lpFreqStart, 0);
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filter.frequency.linearRampToValueAtTime(lpFreqEnd, lpFreqEndAt);
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player.connect(filter);
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filter.connect(context.destination);
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player.start();
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context.oncomplete = function(event) {
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callback(event.renderedBuffer);
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};
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context.startRendering();
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window.filterNode = filter;
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};
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/** @private
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@param {!AudioBuffer} buffer
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@return {!Array.<!Float32Array>} An array containing the Float32Array of each channel's samples. */
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var getAllChannelData = function(buffer) {
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var channels = [];
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for (var i = 0; i < buffer.numberOfChannels; i++) {
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channels[i] = buffer.getChannelData(i);
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}
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return channels;
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};
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/** @private
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@return {number} A random number from -1 to 1. */
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var randomSample = function() {
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return Math.random() * 2 - 1;
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};
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export default reverbGen;
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@ -107,17 +107,24 @@ function getDelay(orbit, delaytime, delayfeedback, t) {
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}
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}
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let reverbs = {};
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let reverbs = {};
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function getReverb(orbit, duration = 2) {
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function getReverb(orbit, duration = 2) {
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// If no reverb has been created for a given orbit, create one
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if (!reverbs[orbit]) {
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if (!reverbs[orbit]) {
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const ac = getAudioContext();
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const ac = getAudioContext();
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const reverb = ac.createReverb(duration);
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const reverb = ac.createReverb(duration, getAudioContext());
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reverb.connect(getDestination());
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reverb.connect(getDestination());
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console.log(reverb)
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reverbs[orbit] = reverb;
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reverbs[orbit] = reverb;
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}
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}
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// Update the reverb duration if needed after instanciation
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if (reverbs[orbit].duration !== duration) {
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if (reverbs[orbit].duration !== duration) {
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reverbs[orbit] = reverbs[orbit].setDuration(duration);
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reverbs[orbit] = reverbs[orbit].setDuration(duration);
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reverbs[orbit].duration = duration;
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reverbs[orbit].duration = duration;
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}
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}
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return reverbs[orbit];
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return reverbs[orbit];
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}
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}
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