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| // Copyright 2014 Alan deLespinasse | ||
| // | ||
| // Licensed under the Apache License, Version 2.0 (the "License"); | ||
| // you may not use this file except in compliance with the License. | ||
| // You may obtain a copy of the License at | ||
| // | ||
| // http://www.apache.org/licenses/LICENSE-2.0 | ||
| // | ||
| // Unless required by applicable law or agreed to in writing, software | ||
| // distributed under the License is distributed on an "AS IS" BASIS, | ||
| // WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. | ||
| // See the License for the specific language governing permissions and | ||
| // limitations under the License. | ||
|
|
||
| 'use strict'; | ||
|
|
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| var reverbGen = {}; | ||
|
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| /** Generates a reverb impulse response. | ||
| @param {!Object} params TODO: Document the properties. | ||
| @param {!function(!AudioBuffer)} callback Function to call when | ||
| the impulse response has been generated. The impulse response | ||
| is passed to this function as its parameter. May be called | ||
| immediately within the current execution context, or later. */ | ||
| reverbGen.generateReverb = function (params, callback) { | ||
| var audioContext = params.audioContext || new AudioContext(); | ||
| var sampleRate = params.sampleRate || 44100; | ||
| var numChannels = params.numChannels || 2; | ||
| // params.decayTime is the -60dB fade time. We let it go 50% longer to get to -90dB. | ||
| var totalTime = params.decayTime * 1.5; | ||
| var decaySampleFrames = Math.round(params.decayTime * sampleRate); | ||
| var numSampleFrames = Math.round(totalTime * sampleRate); | ||
| var fadeInSampleFrames = Math.round((params.fadeInTime || 0) * sampleRate); | ||
| // 60dB is a factor of 1 million in power, or 1000 in amplitude. | ||
| var decayBase = Math.pow(1 / 1000, 1 / decaySampleFrames); | ||
| var reverbIR = audioContext.createBuffer(numChannels, numSampleFrames, sampleRate); | ||
| for (var i = 0; i < numChannels; i++) { | ||
| var chan = reverbIR.getChannelData(i); | ||
| for (var j = 0; j < numSampleFrames; j++) { | ||
| chan[j] = randomSample() * Math.pow(decayBase, j); | ||
| } | ||
| for (var j = 0; j < fadeInSampleFrames; j++) { | ||
| chan[j] *= j / fadeInSampleFrames; | ||
| } | ||
| } | ||
|
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| applyGradualLowpass(reverbIR, params.lpFreqStart || 0, params.lpFreqEnd || 0, params.decayTime, callback); | ||
| }; | ||
|
|
||
| /** Creates a canvas element showing a graph of the given data. | ||
| @param {!Float32Array} data An array of numbers, or a Float32Array. | ||
| @param {number} width Width in pixels of the canvas. | ||
| @param {number} height Height in pixels of the canvas. | ||
| @param {number} min Minimum value of data for the graph (lower edge). | ||
| @param {number} max Maximum value of data in the graph (upper edge). | ||
| @return {!CanvasElement} The generated canvas element. */ | ||
| reverbGen.generateGraph = function (data, width, height, min, max) { | ||
| var canvas = document.createElement('canvas'); | ||
| canvas.width = width; | ||
| canvas.height = height; | ||
| var gc = canvas.getContext('2d'); | ||
| gc.fillStyle = '#000'; | ||
| gc.fillRect(0, 0, canvas.width, canvas.height); | ||
| gc.fillStyle = '#fff'; | ||
| var xscale = width / data.length; | ||
| var yscale = height / (max - min); | ||
| for (var i = 0; i < data.length; i++) { | ||
| gc.fillRect(i * xscale, height - (data[i] - min) * yscale, 1, 1); | ||
| } | ||
| return canvas; | ||
| }; | ||
|
|
||
| /** Saves an AudioBuffer as a 16-bit WAV file on the client's host | ||
| file system. Normalizes it to peak at +-32767, and optionally | ||
| truncates it if there's a lot of "silence" at the end. | ||
| @param {!AudioBuffer} buffer The buffer to save. | ||
| @param {string} name Name of file to create. | ||
| @param {number?} opt_minTail Defines what counts as "silence" for | ||
| auto-truncating the buffer. If there is a point past which every | ||
| value of every channel is less than opt_minTail, then the buffer | ||
| is truncated at that point. This is expressed as an integer, | ||
| applying to the post-normalized and integer-converted | ||
| buffer. The default is 0, meaning don't truncate. */ | ||
| reverbGen.saveWavFile = function (buffer, name, opt_minTail) { | ||
| var bitsPerSample = 16; | ||
| var bytesPerSample = 2; | ||
| var sampleRate = buffer.sampleRate; | ||
| var numChannels = buffer.numberOfChannels; | ||
| var channels = getAllChannelData(buffer); | ||
| var numSampleFrames = channels[0].length; | ||
| var scale = 32767; | ||
| // Find normalization constant. | ||
| var max = 0; | ||
| for (var i = 0; i < numChannels; i++) { | ||
| for (var j = 0; j < numSampleFrames; j++) { | ||
| max = Math.max(max, Math.abs(channels[i][j])); | ||
| } | ||
| } | ||
| if (max) { | ||
| scale = 32767 / max; | ||
| } | ||
| // Find truncation point. | ||
| if (opt_minTail) { | ||
| var truncateAt = 0; | ||
| for (var i = 0; i < numChannels; i++) { | ||
| for (var j = 0; j < numSampleFrames; j++) { | ||
| var absSample = Math.abs(Math.round(scale * channels[i][j])); | ||
| if (absSample > opt_minTail) { | ||
| truncateAt = j; | ||
| } | ||
| } | ||
| } | ||
| numSampleFrames = truncateAt + 1; | ||
| } | ||
| var sampleDataBytes = bytesPerSample * numChannels * numSampleFrames; | ||
| var fileBytes = sampleDataBytes + 44; | ||
| var arrayBuffer = new ArrayBuffer(fileBytes); | ||
| var dataView = new DataView(arrayBuffer); | ||
| dataView.setUint32(0, 1179011410, true); // "RIFF" | ||
| dataView.setUint32(4, fileBytes - 8, true); // file length | ||
| dataView.setUint32(8, 1163280727, true); // "WAVE" | ||
| dataView.setUint32(12, 544501094, true); // "fmt " | ||
| dataView.setUint32(16, 16, true); // fmt chunk length | ||
| dataView.setUint16(20, 1, true); // PCM format | ||
| dataView.setUint16(22, numChannels, true); // NumChannels | ||
| dataView.setUint32(24, sampleRate, true); // SampleRate | ||
| var bytesPerSampleFrame = numChannels * bytesPerSample; | ||
| dataView.setUint32(28, sampleRate * bytesPerSampleFrame, true); // ByteRate | ||
| dataView.setUint16(32, bytesPerSampleFrame, true); // BlockAlign | ||
| dataView.setUint16(34, bitsPerSample, true); // BitsPerSample | ||
| dataView.setUint32(36, 1635017060, true); // "data" | ||
| dataView.setUint32(40, sampleDataBytes, true); | ||
| for (var j = 0; j < numSampleFrames; j++) { | ||
| for (var i = 0; i < numChannels; i++) { | ||
| dataView.setInt16(44 + j * bytesPerSampleFrame + i * bytesPerSample, Math.round(scale * channels[i][j]), true); | ||
| } | ||
| } | ||
| var blob = new Blob([arrayBuffer], { type: 'audio/wav' }); | ||
| var url = window.URL.createObjectURL(blob); | ||
| var linkEl = document.createElement('a'); | ||
| linkEl.href = url; | ||
| linkEl.download = name; | ||
| linkEl.style.display = 'none'; | ||
| document.body.appendChild(linkEl); | ||
| linkEl.click(); | ||
| }; | ||
|
|
||
| /** Applies a constantly changing lowpass filter to the given sound. | ||
| @private | ||
| @param {!AudioBuffer} input | ||
| @param {number} lpFreqStart | ||
| @param {number} lpFreqEnd | ||
| @param {number} lpFreqEndAt | ||
| @param {!function(!AudioBuffer)} callback May be called | ||
| immediately within the current execution context, or later.*/ | ||
| var applyGradualLowpass = function (input, lpFreqStart, lpFreqEnd, lpFreqEndAt, callback) { | ||
| if (lpFreqStart == 0) { | ||
| callback(input); | ||
| return; | ||
| } | ||
| var channelData = getAllChannelData(input); | ||
| var context = new OfflineAudioContext(input.numberOfChannels, channelData[0].length, input.sampleRate); | ||
| var player = context.createBufferSource(); | ||
| player.buffer = input; | ||
| var filter = context.createBiquadFilter(); | ||
|
|
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| lpFreqStart = Math.min(lpFreqStart, input.sampleRate / 2); | ||
| lpFreqEnd = Math.min(lpFreqEnd, input.sampleRate / 2); | ||
|
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| filter.type = 'lowpass'; | ||
| filter.Q.value = 0.0001; | ||
| filter.frequency.setValueAtTime(lpFreqStart, 0); | ||
| filter.frequency.linearRampToValueAtTime(lpFreqEnd, lpFreqEndAt); | ||
|
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| player.connect(filter); | ||
| filter.connect(context.destination); | ||
| player.start(); | ||
| context.oncomplete = function (event) { | ||
| callback(event.renderedBuffer); | ||
| }; | ||
| context.startRendering(); | ||
|
|
||
| window.filterNode = filter; | ||
| }; | ||
|
|
||
| /** @private | ||
| @param {!AudioBuffer} buffer | ||
| @return {!Array.<!Float32Array>} An array containing the Float32Array of each channel's samples. */ | ||
| var getAllChannelData = function (buffer) { | ||
| var channels = []; | ||
| for (var i = 0; i < buffer.numberOfChannels; i++) { | ||
| channels[i] = buffer.getChannelData(i); | ||
| } | ||
| return channels; | ||
| }; | ||
|
|
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| /** @private | ||
| @return {number} A random number from -1 to 1. */ | ||
| var randomSample = function () { | ||
| return Math.random() * 2 - 1; | ||
| }; | ||
|
|
||
| export default reverbGen; |
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