2475 lines
70 KiB
JavaScript
2475 lines
70 KiB
JavaScript
/*
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* DSP.js - a comprehensive digital signal processing library for javascript
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*
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* Created by Corban Brook <corbanbrook@gmail.com> on 2010-01-01.
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* Copyright 2010 Corban Brook. All rights reserved.
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*
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* Modified by Stefan Bosse <sbosse@uni-bremen.de> on 2018
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*
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*/
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var current=none;
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var Aios=none;
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////////////////////////////////////////////////////////////////////////////////
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// CONSTANTS //
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////////////////////////////////////////////////////////////////////////////////
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/**
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* DSP is an object which contains general purpose utility functions and constants
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*/
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var DSP = {
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// Channels
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LEFT: 0,
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RIGHT: 1,
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MIX: 2,
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// Waveforms
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SINE: 1,
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TRIANGLE: 2,
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SAW: 3,
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SQUARE: 4,
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// Filters
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LOWPASS: 0,
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HIGHPASS: 1,
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BANDPASS: 2,
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NOTCH: 3,
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// Window functions
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BARTLETT: 1,
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BARTLETTHANN: 2,
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BLACKMAN: 3,
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COSINE: 4,
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GAUSS: 5,
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HAMMING: 6,
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HANN: 7,
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LANCZOS: 8,
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RECTANGULAR: 9,
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TRIANGULAR: 10,
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// Mean Filter
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ARITH: 0,
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EXP: 1,
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PEAK: 2,
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// Loop modes
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OFF: 0,
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FW: 1,
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BW: 2,
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FWBW: 3,
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// Math
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TWO_PI: 2*Math.PI
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};
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// Setup arrays for platforms which do not support byte arrays
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function setupTypedArray(name, fallback) {
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// check if TypedArray exists
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// typeof on Minefield and Chrome return function, typeof on Webkit returns object.
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if (typeof this[name] !== "function" && typeof this[name] !== "object") {
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// nope.. check if WebGLArray exists
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if (typeof this[fallback] === "function" && typeof this[fallback] !== "object") {
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this[name] = this[fallback];
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} else {
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// nope.. set as Native JS array
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this[name] = function(obj) {
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if (obj instanceof Array) {
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return obj;
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} else if (typeof obj === "number") {
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return new Array(obj);
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}
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};
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}
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}
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}
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setupTypedArray("Float64Array", "WebGLFloatArray");
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setupTypedArray("Int32Array", "WebGLIntArray");
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setupTypedArray("Uint16Array", "WebGLUnsignedShortArray");
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setupTypedArray("Uint8Array", "WebGLUnsignedByteArray");
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////////////////////////////////////////////////////////////////////////////////
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// DSP UTILITY FUNCTIONS //
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////////////////////////////////////////////////////////////////////////////////
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/**
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* Inverts the phase of a signal
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*
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* @param {Array} buffer A sample buffer
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*
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* @returns The inverted sample buffer
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*/
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DSP.invert = function(buffer) {
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for (var i = 0, len = buffer.length; i < len; i++) {
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buffer[i] *= -1;
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}
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return buffer;
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};
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/**
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* Converts split-stereo (dual mono) sample buffers into a stereo interleaved sample buffer
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*
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* @param {Array} left A sample buffer
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* @param {Array} right A sample buffer
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*
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* @returns The stereo interleaved buffer
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*/
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DSP.interleave = function(left, right) {
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if (left.length !== right.length) {
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throw "Can not interleave. Channel lengths differ.";
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}
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var stereoInterleaved = new Float64Array(left.length * 2);
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for (var i = 0, len = left.length; i < len; i++) {
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stereoInterleaved[2*i] = left[i];
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stereoInterleaved[2*i+1] = right[i];
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}
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return stereoInterleaved;
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};
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/**
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* Converts a stereo-interleaved sample buffer into split-stereo (dual mono) sample buffers
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*
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* @param {Array} buffer A stereo-interleaved sample buffer
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*
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* @returns an Array containing left and right channels
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*/
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DSP.deinterleave = (function() {
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var left, right, mix, deinterleaveChannel = [];
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deinterleaveChannel[DSP.MIX] = function(buffer) {
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for (var i = 0, len = buffer.length/2; i < len; i++) {
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mix[i] = (buffer[2*i] + buffer[2*i+1]) / 2;
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}
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return mix;
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};
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deinterleaveChannel[DSP.LEFT] = function(buffer) {
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for (var i = 0, len = buffer.length/2; i < len; i++) {
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left[i] = buffer[2*i];
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}
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return left;
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};
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deinterleaveChannel[DSP.RIGHT] = function(buffer) {
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for (var i = 0, len = buffer.length/2; i < len; i++) {
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right[i] = buffer[2*i+1];
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}
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return right;
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};
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return function(channel, buffer) {
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left = left || new Float64Array(buffer.length/2);
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right = right || new Float64Array(buffer.length/2);
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mix = mix || new Float64Array(buffer.length/2);
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if (buffer.length/2 !== left.length) {
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left = new Float64Array(buffer.length/2);
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right = new Float64Array(buffer.length/2);
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mix = new Float64Array(buffer.length/2);
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}
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return deinterleaveChannel[channel](buffer);
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};
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}());
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/**
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* Separates a channel from a stereo-interleaved sample buffer
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*
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* @param {Array} buffer A stereo-interleaved sample buffer
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* @param {Number} channel A channel constant (LEFT, RIGHT, MIX)
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*
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* @returns an Array containing a signal mono sample buffer
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*/
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DSP.getChannel = DSP.deinterleave;
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/**
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* Helper method (for Reverb) to mix two (interleaved) samplebuffers. It's possible
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* to negate the second buffer while mixing and to perform a volume correction
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* on the final signal.
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*
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* @param {Array} sampleBuffer1 Array containing Float values or a Float64Array
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* @param {Array} sampleBuffer2 Array containing Float values or a Float64Array
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* @param {Boolean} negate When true inverts/flips the audio signal
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* @param {Number} volumeCorrection When you add multiple sample buffers, use this to tame your signal ;)
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*
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* @returns A new Float64Array interleaved buffer.
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*/
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DSP.mixSampleBuffers = function(sampleBuffer1, sampleBuffer2, negate, volumeCorrection){
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var outputSamples = new Float64Array(sampleBuffer1);
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for(var i = 0; i<sampleBuffer1.length; i++){
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outputSamples[i] += (negate ? -sampleBuffer2[i] : sampleBuffer2[i]) / volumeCorrection;
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}
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return outputSamples;
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};
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// Biquad filter types
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DSP.LPF = 0; // H(s) = 1 / (s^2 + s/Q + 1)
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DSP.HPF = 1; // H(s) = s^2 / (s^2 + s/Q + 1)
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DSP.BPF_CONSTANT_SKIRT = 2; // H(s) = s / (s^2 + s/Q + 1) (constant skirt gain, peak gain = Q)
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DSP.BPF_CONSTANT_PEAK = 3; // H(s) = (s/Q) / (s^2 + s/Q + 1) (constant 0 dB peak gain)
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DSP.NOTCH = 4; // H(s) = (s^2 + 1) / (s^2 + s/Q + 1)
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DSP.APF = 5; // H(s) = (s^2 - s/Q + 1) / (s^2 + s/Q + 1)
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DSP.PEAKING_EQ = 6; // H(s) = (s^2 + s*(A/Q) + 1) / (s^2 + s/(A*Q) + 1)
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DSP.LOW_SHELF = 7; // H(s) = A * (s^2 + (sqrt(A)/Q)*s + A)/(A*s^2 + (sqrt(A)/Q)*s + 1)
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DSP.HIGH_SHELF = 8; // H(s) = A * (A*s^2 + (sqrt(A)/Q)*s + 1)/(s^2 + (sqrt(A)/Q)*s + A)
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// Biquad filter parameter types
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DSP.Q = 1;
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DSP.BW = 2; // SHARED with BACKWARDS LOOP MODE
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DSP.S = 3;
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// Find RMS of signal
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DSP.RMS = function(buffer) {
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var total = 0;
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for (var i = 0, n = buffer.length; i < n; i++) {
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total += buffer[i] * buffer[i];
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}
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return Math.sqrt(total / n);
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};
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// Find Peak of signal
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DSP.Peak = function(buffer) {
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var peak = 0;
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for (var i = 0, n = buffer.length; i < n; i++) {
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peak = (Math.abs(buffer[i]) > peak) ? Math.abs(buffer[i]) : peak;
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}
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return peak;
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};
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// Get features of a signal [min,max,mean,deviation}
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DSP.Features = function(buffer) {
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var i,x,
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ft = { min:Number.MAX_VALUE, max:Number.MIN_VALUE, mean:0.0, deviation:0.0 };
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for(i=0;i<buffer.length;i++) {
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x=buffer[i];
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ft.min=Math.min(ft.min,x);
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ft.max=Math.max(ft.max,x);
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ft.mean += x;
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};
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ft.mean /= buffer.length;
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for(i=0;i<buffer.length;i++) {
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x=buffer[i];
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ft.deviation += Math.pow(x-ft.mean,2);
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};
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ft.deviation /= (buffer.length-1);
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return ft;
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}
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DSP.Histogram = function (buffer,size) {
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var i,item,bin_width,histogram,
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length = buffer.length,
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min = buffer[0],
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max = buffer[1];
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for (i = 0; i < length; i++) {
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item = buffer[i];
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min = Math.min(item,min);
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max = Math.max(item,max);
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}
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bin_width = (max - min) / (size - 1);
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histogram = new Array(size);
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for (i = 0; i < size; i++) histogram[i] = 0;
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for (i = 0; i < length; i++)
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histogram[Math.floor((buffer[i] - min) / bin_width)]++;
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return histogram;
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}
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DSP.Normalize = function(buffer) {
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var i,
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output,
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length = buffer.length,
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min = buffer[0],
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max = buffer[1];
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if(buffer instanceof Array) output=new Array(buffer.length);
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else if(buffer instanceof Float32Array) output=new Float32Array(buffer.length);
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else if(buffer instanceof Float64Array) output=new Float64Array(buffer.length);
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for (i = 0; i < length; i++) {
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item = buffer[i];
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min = Math.min(item,min);
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max = Math.max(item,max);
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}
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for (i = 0; i < length; i++) {
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item = buffer[i];
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output[i]=item/max;
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}
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return output;
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}
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DSP.Reduce = function(buffer, segments, filter) {
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var i,j=0,k=Math.floor(buffer.length/segments),
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first=true,
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n=1,y=0,
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output;
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if (buffer instanceof Array) output=new Array(segments);
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else if (buffer instanceof Float32Array) output=new Float32Array(segments);
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else if (buffer instanceof Float64Array) output=new Float64Array(segments);
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if (!filter) filter=DSP.ARITH;
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for(i in buffer) {
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switch (filter) {
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case DSP.EXP:
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y=(y+buffer[i])/2; break;
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if (i>0 && (i%k)==0) {
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output[j]=y; j++;
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y=0;n=1;
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}
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case DSP.ARITH:
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y=y+buffer[i]; n++;
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if (i>0 && (i%k)==0) {
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output[j]=(y/n); j++;
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y=0;n=1;
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}
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break;
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case DSP.PEAK:
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if (first) y=buffer[i];
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y=Math.max(y,buffer[i]);
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if (i>0 && (i%k)==0) {
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output[j]=y; j++;
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y=0;n=1;first=true;
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}
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break;
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}
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first=false;
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}
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if (filter==DSP.ARITH) {output[j]=(y/n); j++}
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return output;
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}
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DSP.Resample = function(buffer, divider, filter) {
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return DSP.Reduce(buffer,(buffer.length/divider)|0,filter);
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}
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// Add noise (sample + [-noise,+noise] random interval)
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DSP.Noise = function(buffer,noise,random,inplace) {
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var ouput;
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if (inplace) output=buffer;
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else if(buffer instanceof Array) output=new Array(buffer.length);
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else if(buffer instanceof Float32Array) output=new Float32Array(buffer.length);
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else if(buffer instanceof Float64Array) output=new Float64Array(buffer.length);
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if (!random) random=function(e) {return -e+2*e*Math.random()};
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for(var i=0;i < buffer.length;i++) output[i]=buffer[i]+random(noise);
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}
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// Fourier Transform Module used by DFT, FFT, RFFT
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function FourierTransform(bufferSize, sampleRate) {
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this.bufferSize = bufferSize;
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this.sampleRate = sampleRate;
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this.bandwidth = 2 / bufferSize * sampleRate / 2;
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this.spectrum = new Float64Array(bufferSize/2);
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this.real = new Float64Array(bufferSize);
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this.imag = new Float64Array(bufferSize);
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this.peakBand = 0;
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this.peak = 0;
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/**
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* Calculates the *middle* frequency of an FFT band.
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*
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* @param {Number} index The index of the FFT band.
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*
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* @returns The middle frequency in Hz.
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*/
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this.getBandFrequency = function(index) {
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return this.bandwidth * index + this.bandwidth / 2;
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};
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this.calculateSpectrum = function() {
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var spectrum = this.spectrum,
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real = this.real,
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imag = this.imag,
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bSi = 2 / this.bufferSize,
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sqrt = Math.sqrt,
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rval,
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ival,
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mag;
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for (var i = 0, N = bufferSize/2; i < N; i++) {
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rval = real[i];
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ival = imag[i];
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mag = bSi * sqrt(rval * rval + ival * ival);
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if (mag > this.peak) {
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this.peakBand = i;
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this.peak = mag;
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}
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spectrum[i] = mag;
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}
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return spectrum;
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};
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}
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/**
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* DFT is a class for calculating the Discrete Fourier Transform of a signal.
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*
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* @param {Number} bufferSize The size of the sample buffer to be computed
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* @param {Number} sampleRate The sampleRate of the buffer (eg. 44100)
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*
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* @constructor
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*/
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function DFT(bufferSize, sampleRate) {
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FourierTransform.call(this, bufferSize, sampleRate);
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var N = bufferSize/2 * bufferSize;
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var TWO_PI = 2 * Math.PI;
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this.sinTable = new Float64Array(N);
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this.cosTable = new Float64Array(N);
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for (var i = 0; i < N; i++) {
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this.sinTable[i] = Math.sin(i * TWO_PI / bufferSize);
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this.cosTable[i] = Math.cos(i * TWO_PI / bufferSize);
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}
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}
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/**
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* Performs a forward transform on the sample buffer.
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* Converts a time domain signal to frequency domain spectra.
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*
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* @param {Array} buffer The sample buffer
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*
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* @returns The frequency spectrum array
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*/
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DFT.prototype.forward = function(buffer,spectrum) {
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var real = this.real,
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imag = this.imag,
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rval,
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ival;
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for (var k = 0; k < this.bufferSize/2; k++) {
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rval = 0.0;
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ival = 0.0;
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for (var n = 0; n < buffer.length; n++) {
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rval += this.cosTable[k*n] * buffer[n];
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ival += this.sinTable[k*n] * buffer[n];
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}
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real[k] = rval;
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imag[k] = ival;
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}
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return spectrum?this.calculateSpectrum():{real:this.real,imag:this.imag};
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};
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/**
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* FFT is a class for calculating the Discrete Fourier Transform of a signal
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* with the Fast Fourier Transform algorithm.
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*
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* @param {Number} bufferSize The size of the sample buffer to be computed. Must be power of 2
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* @param {Number} sampleRate The sampleRate of the buffer (eg. 44100)
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*
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* @constructor
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*/
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function FFT(bufferSize, sampleRate) {
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if (!(this instanceof FFT)) return new FFT(bufferSize, sampleRate);
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FourierTransform.call(this, bufferSize, sampleRate);
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this.reverseTable = new Uint32Array(bufferSize);
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var limit = 1;
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var bit = bufferSize >> 1;
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var i;
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while (limit < bufferSize) {
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for (i = 0; i < limit; i++) {
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this.reverseTable[i + limit] = this.reverseTable[i] + bit;
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}
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limit = limit << 1;
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bit = bit >> 1;
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}
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this.sinTable = new Float64Array(bufferSize);
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this.cosTable = new Float64Array(bufferSize);
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for (i = 0; i < bufferSize; i++) {
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this.sinTable[i] = Math.sin(-Math.PI/i);
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this.cosTable[i] = Math.cos(-Math.PI/i);
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}
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}
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/**
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* Performs a forward transform on the sample buffer.
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* Converts a time domain signal to frequency domain spectra.
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*
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* @param {Array} buffer The sample buffer. Buffer Length must be power of 2
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*
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* @returns The frequency spectrum array
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*/
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FFT.prototype.forward = function(buffer,spectrum) {
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// Locally scope variables for speed up
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var bufferSize = this.bufferSize,
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cosTable = this.cosTable,
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sinTable = this.sinTable,
|
|
reverseTable = this.reverseTable,
|
|
real = this.real,
|
|
imag = this.imag,
|
|
spectrum = this.spectrum;
|
|
|
|
var k = Math.floor(Math.log(bufferSize) / Math.LN2);
|
|
|
|
if (Math.pow(2, k) !== bufferSize) { throw "Invalid buffer size, must be a power of 2."; }
|
|
if (bufferSize !== buffer.length) { throw "Supplied buffer is not the same size as defined FFT. FFT Size: " + bufferSize + " Buffer Size: " + buffer.length; }
|
|
|
|
var halfSize = 1,
|
|
phaseShiftStepReal,
|
|
phaseShiftStepImag,
|
|
currentPhaseShiftReal,
|
|
currentPhaseShiftImag,
|
|
off,
|
|
tr,
|
|
ti,
|
|
tmpReal,
|
|
i;
|
|
|
|
for (i = 0; i < bufferSize; i++) {
|
|
real[i] = buffer[reverseTable[i]];
|
|
imag[i] = 0;
|
|
}
|
|
|
|
while (halfSize < bufferSize) {
|
|
//phaseShiftStepReal = Math.cos(-Math.PI/halfSize);
|
|
//phaseShiftStepImag = Math.sin(-Math.PI/halfSize);
|
|
phaseShiftStepReal = cosTable[halfSize];
|
|
phaseShiftStepImag = sinTable[halfSize];
|
|
|
|
currentPhaseShiftReal = 1;
|
|
currentPhaseShiftImag = 0;
|
|
|
|
for (var fftStep = 0; fftStep < halfSize; fftStep++) {
|
|
i = fftStep;
|
|
|
|
while (i < bufferSize) {
|
|
off = i + halfSize;
|
|
tr = (currentPhaseShiftReal * real[off]) - (currentPhaseShiftImag * imag[off]);
|
|
ti = (currentPhaseShiftReal * imag[off]) + (currentPhaseShiftImag * real[off]);
|
|
|
|
real[off] = real[i] - tr;
|
|
imag[off] = imag[i] - ti;
|
|
real[i] += tr;
|
|
imag[i] += ti;
|
|
|
|
i += halfSize << 1;
|
|
}
|
|
|
|
tmpReal = currentPhaseShiftReal;
|
|
currentPhaseShiftReal = (tmpReal * phaseShiftStepReal) - (currentPhaseShiftImag * phaseShiftStepImag);
|
|
currentPhaseShiftImag = (tmpReal * phaseShiftStepImag) + (currentPhaseShiftImag * phaseShiftStepReal);
|
|
}
|
|
|
|
halfSize = halfSize << 1;
|
|
}
|
|
|
|
return spectrum?this.calculateSpectrum():{real:this.real,imag:this.imag};
|
|
};
|
|
|
|
FFT.prototype.inverse = function(real, imag) {
|
|
// Locally scope variables for speed up
|
|
var bufferSize = this.bufferSize,
|
|
cosTable = this.cosTable,
|
|
sinTable = this.sinTable,
|
|
reverseTable = this.reverseTable,
|
|
spectrum = this.spectrum;
|
|
|
|
real = real || this.real;
|
|
imag = imag || this.imag;
|
|
|
|
var halfSize = 1,
|
|
phaseShiftStepReal,
|
|
phaseShiftStepImag,
|
|
currentPhaseShiftReal,
|
|
currentPhaseShiftImag,
|
|
off,
|
|
tr,
|
|
ti,
|
|
tmpReal,
|
|
i;
|
|
|
|
for (i = 0; i < bufferSize; i++) {
|
|
imag[i] *= -1;
|
|
}
|
|
|
|
var revReal = new Float64Array(bufferSize);
|
|
var revImag = new Float64Array(bufferSize);
|
|
|
|
for (i = 0; i < real.length; i++) {
|
|
revReal[i] = real[reverseTable[i]];
|
|
revImag[i] = imag[reverseTable[i]];
|
|
}
|
|
|
|
real = revReal;
|
|
imag = revImag;
|
|
|
|
while (halfSize < bufferSize) {
|
|
phaseShiftStepReal = cosTable[halfSize];
|
|
phaseShiftStepImag = sinTable[halfSize];
|
|
currentPhaseShiftReal = 1;
|
|
currentPhaseShiftImag = 0;
|
|
|
|
for (var fftStep = 0; fftStep < halfSize; fftStep++) {
|
|
i = fftStep;
|
|
|
|
while (i < bufferSize) {
|
|
off = i + halfSize;
|
|
tr = (currentPhaseShiftReal * real[off]) - (currentPhaseShiftImag * imag[off]);
|
|
ti = (currentPhaseShiftReal * imag[off]) + (currentPhaseShiftImag * real[off]);
|
|
|
|
real[off] = real[i] - tr;
|
|
imag[off] = imag[i] - ti;
|
|
real[i] += tr;
|
|
imag[i] += ti;
|
|
|
|
i += halfSize << 1;
|
|
}
|
|
|
|
tmpReal = currentPhaseShiftReal;
|
|
currentPhaseShiftReal = (tmpReal * phaseShiftStepReal) - (currentPhaseShiftImag * phaseShiftStepImag);
|
|
currentPhaseShiftImag = (tmpReal * phaseShiftStepImag) + (currentPhaseShiftImag * phaseShiftStepReal);
|
|
}
|
|
|
|
halfSize = halfSize << 1;
|
|
}
|
|
|
|
var buffer = new Float64Array(bufferSize); // this should be reused instead
|
|
for (i = 0; i < bufferSize; i++) {
|
|
buffer[i] = real[i] / bufferSize;
|
|
}
|
|
|
|
return buffer;
|
|
};
|
|
|
|
/**
|
|
* RFFT is a class for calculating the Discrete Fourier Transform of a signal
|
|
* with the Fast Fourier Transform algorithm.
|
|
*
|
|
* This method currently only contains a forward transform but is highly optimized.
|
|
*
|
|
* @param {Number} bufferSize The size of the sample buffer to be computed. Must be power of 2
|
|
* @param {Number} sampleRate The sampleRate of the buffer (eg. 44100)
|
|
*
|
|
* @constructor
|
|
*/
|
|
|
|
// lookup tables don't really gain us any speed, but they do increase
|
|
// cache footprint, so don't use them in here
|
|
|
|
// also we don't use sepearate arrays for real/imaginary parts
|
|
|
|
// this one a little more than twice as fast as the one in FFT
|
|
// however I only did the forward transform
|
|
|
|
// the rest of this was translated from C, see http://www.jjj.de/fxt/
|
|
// this is the real split radix FFT
|
|
|
|
function RFFT(bufferSize, sampleRate) {
|
|
FourierTransform.call(this, bufferSize, sampleRate);
|
|
|
|
this.trans = new Float64Array(bufferSize);
|
|
|
|
this.reverseTable = new Uint32Array(bufferSize);
|
|
|
|
// don't use a lookup table to do the permute, use this instead
|
|
this.reverseBinPermute = function (dest, source) {
|
|
var bufferSize = this.bufferSize,
|
|
halfSize = bufferSize >>> 1,
|
|
nm1 = bufferSize - 1,
|
|
i = 1, r = 0, h;
|
|
|
|
dest[0] = source[0];
|
|
|
|
do {
|
|
r += halfSize;
|
|
dest[i] = source[r];
|
|
dest[r] = source[i];
|
|
|
|
i++;
|
|
|
|
h = halfSize << 1;
|
|
while (h = h >> 1, !((r ^= h) & h));
|
|
|
|
if (r >= i) {
|
|
dest[i] = source[r];
|
|
dest[r] = source[i];
|
|
|
|
dest[nm1-i] = source[nm1-r];
|
|
dest[nm1-r] = source[nm1-i];
|
|
}
|
|
i++;
|
|
} while (i < halfSize);
|
|
dest[nm1] = source[nm1];
|
|
};
|
|
|
|
this.generateReverseTable = function () {
|
|
var bufferSize = this.bufferSize,
|
|
halfSize = bufferSize >>> 1,
|
|
nm1 = bufferSize - 1,
|
|
i = 1, r = 0, h;
|
|
|
|
this.reverseTable[0] = 0;
|
|
|
|
do {
|
|
r += halfSize;
|
|
|
|
this.reverseTable[i] = r;
|
|
this.reverseTable[r] = i;
|
|
|
|
i++;
|
|
|
|
h = halfSize << 1;
|
|
while (h = h >> 1, !((r ^= h) & h));
|
|
|
|
if (r >= i) {
|
|
this.reverseTable[i] = r;
|
|
this.reverseTable[r] = i;
|
|
|
|
this.reverseTable[nm1-i] = nm1-r;
|
|
this.reverseTable[nm1-r] = nm1-i;
|
|
}
|
|
i++;
|
|
} while (i < halfSize);
|
|
|
|
this.reverseTable[nm1] = nm1;
|
|
};
|
|
|
|
this.generateReverseTable();
|
|
}
|
|
|
|
|
|
// Ordering of output:
|
|
//
|
|
// trans[0] = re[0] (==zero frequency, purely real)
|
|
// trans[1] = re[1]
|
|
// ...
|
|
// trans[n/2-1] = re[n/2-1]
|
|
// trans[n/2] = re[n/2] (==nyquist frequency, purely real)
|
|
//
|
|
// trans[n/2+1] = im[n/2-1]
|
|
// trans[n/2+2] = im[n/2-2]
|
|
// ...
|
|
// trans[n-1] = im[1]
|
|
|
|
RFFT.prototype.forward = function(buffer) {
|
|
var n = this.bufferSize,
|
|
spectrum = this.spectrum,
|
|
x = this.trans,
|
|
TWO_PI = 2*Math.PI,
|
|
sqrt = Math.sqrt,
|
|
i = n >>> 1,
|
|
bSi = 2 / n,
|
|
n2, n4, n8, nn,
|
|
t1, t2, t3, t4,
|
|
i1, i2, i3, i4, i5, i6, i7, i8,
|
|
st1, cc1, ss1, cc3, ss3,
|
|
e,
|
|
a,
|
|
rval, ival, mag;
|
|
|
|
this.reverseBinPermute(x, buffer);
|
|
|
|
/*
|
|
var reverseTable = this.reverseTable;
|
|
|
|
for (var k = 0, len = reverseTable.length; k < len; k++) {
|
|
x[k] = buffer[reverseTable[k]];
|
|
}
|
|
*/
|
|
|
|
for (var ix = 0, id = 4; ix < n; id *= 4) {
|
|
for (var i0 = ix; i0 < n; i0 += id) {
|
|
//sumdiff(x[i0], x[i0+1]); // {a, b} <--| {a+b, a-b}
|
|
st1 = x[i0] - x[i0+1];
|
|
x[i0] += x[i0+1];
|
|
x[i0+1] = st1;
|
|
}
|
|
ix = 2*(id-1);
|
|
}
|
|
|
|
n2 = 2;
|
|
nn = n >>> 1;
|
|
|
|
while((nn = nn >>> 1)) {
|
|
ix = 0;
|
|
n2 = n2 << 1;
|
|
id = n2 << 1;
|
|
n4 = n2 >>> 2;
|
|
n8 = n2 >>> 3;
|
|
do {
|
|
if(n4 !== 1) {
|
|
for(i0 = ix; i0 < n; i0 += id) {
|
|
i1 = i0;
|
|
i2 = i1 + n4;
|
|
i3 = i2 + n4;
|
|
i4 = i3 + n4;
|
|
|
|
//diffsum3_r(x[i3], x[i4], t1); // {a, b, s} <--| {a, b-a, a+b}
|
|
t1 = x[i3] + x[i4];
|
|
x[i4] -= x[i3];
|
|
//sumdiff3(x[i1], t1, x[i3]); // {a, b, d} <--| {a+b, b, a-b}
|
|
x[i3] = x[i1] - t1;
|
|
x[i1] += t1;
|
|
|
|
i1 += n8;
|
|
i2 += n8;
|
|
i3 += n8;
|
|
i4 += n8;
|
|
|
|
//sumdiff(x[i3], x[i4], t1, t2); // {s, d} <--| {a+b, a-b}
|
|
t1 = x[i3] + x[i4];
|
|
t2 = x[i3] - x[i4];
|
|
|
|
t1 = -t1 * Math.SQRT1_2;
|
|
t2 *= Math.SQRT1_2;
|
|
|
|
// sumdiff(t1, x[i2], x[i4], x[i3]); // {s, d} <--| {a+b, a-b}
|
|
st1 = x[i2];
|
|
x[i4] = t1 + st1;
|
|
x[i3] = t1 - st1;
|
|
|
|
//sumdiff3(x[i1], t2, x[i2]); // {a, b, d} <--| {a+b, b, a-b}
|
|
x[i2] = x[i1] - t2;
|
|
x[i1] += t2;
|
|
}
|
|
} else {
|
|
for(i0 = ix; i0 < n; i0 += id) {
|
|
i1 = i0;
|
|
i2 = i1 + n4;
|
|
i3 = i2 + n4;
|
|
i4 = i3 + n4;
|
|
|
|
//diffsum3_r(x[i3], x[i4], t1); // {a, b, s} <--| {a, b-a, a+b}
|
|
t1 = x[i3] + x[i4];
|
|
x[i4] -= x[i3];
|
|
|
|
//sumdiff3(x[i1], t1, x[i3]); // {a, b, d} <--| {a+b, b, a-b}
|
|
x[i3] = x[i1] - t1;
|
|
x[i1] += t1;
|
|
}
|
|
}
|
|
|
|
ix = (id << 1) - n2;
|
|
id = id << 2;
|
|
} while (ix < n);
|
|
|
|
e = TWO_PI / n2;
|
|
|
|
for (var j = 1; j < n8; j++) {
|
|
a = j * e;
|
|
ss1 = Math.sin(a);
|
|
cc1 = Math.cos(a);
|
|
|
|
//ss3 = sin(3*a); cc3 = cos(3*a);
|
|
cc3 = 4*cc1*(cc1*cc1-0.75);
|
|
ss3 = 4*ss1*(0.75-ss1*ss1);
|
|
|
|
ix = 0; id = n2 << 1;
|
|
do {
|
|
for (i0 = ix; i0 < n; i0 += id) {
|
|
i1 = i0 + j;
|
|
i2 = i1 + n4;
|
|
i3 = i2 + n4;
|
|
i4 = i3 + n4;
|
|
|
|
i5 = i0 + n4 - j;
|
|
i6 = i5 + n4;
|
|
i7 = i6 + n4;
|
|
i8 = i7 + n4;
|
|
|
|
//cmult(c, s, x, y, &u, &v)
|
|
//cmult(cc1, ss1, x[i7], x[i3], t2, t1); // {u,v} <--| {x*c-y*s, x*s+y*c}
|
|
t2 = x[i7]*cc1 - x[i3]*ss1;
|
|
t1 = x[i7]*ss1 + x[i3]*cc1;
|
|
|
|
//cmult(cc3, ss3, x[i8], x[i4], t4, t3);
|
|
t4 = x[i8]*cc3 - x[i4]*ss3;
|
|
t3 = x[i8]*ss3 + x[i4]*cc3;
|
|
|
|
//sumdiff(t2, t4); // {a, b} <--| {a+b, a-b}
|
|
st1 = t2 - t4;
|
|
t2 += t4;
|
|
t4 = st1;
|
|
|
|
//sumdiff(t2, x[i6], x[i8], x[i3]); // {s, d} <--| {a+b, a-b}
|
|
//st1 = x[i6]; x[i8] = t2 + st1; x[i3] = t2 - st1;
|
|
x[i8] = t2 + x[i6];
|
|
x[i3] = t2 - x[i6];
|
|
|
|
//sumdiff_r(t1, t3); // {a, b} <--| {a+b, b-a}
|
|
st1 = t3 - t1;
|
|
t1 += t3;
|
|
t3 = st1;
|
|
|
|
//sumdiff(t3, x[i2], x[i4], x[i7]); // {s, d} <--| {a+b, a-b}
|
|
//st1 = x[i2]; x[i4] = t3 + st1; x[i7] = t3 - st1;
|
|
x[i4] = t3 + x[i2];
|
|
x[i7] = t3 - x[i2];
|
|
|
|
//sumdiff3(x[i1], t1, x[i6]); // {a, b, d} <--| {a+b, b, a-b}
|
|
x[i6] = x[i1] - t1;
|
|
x[i1] += t1;
|
|
|
|
//diffsum3_r(t4, x[i5], x[i2]); // {a, b, s} <--| {a, b-a, a+b}
|
|
x[i2] = t4 + x[i5];
|
|
x[i5] -= t4;
|
|
}
|
|
|
|
ix = (id << 1) - n2;
|
|
id = id << 2;
|
|
|
|
} while (ix < n);
|
|
}
|
|
}
|
|
|
|
while (--i) {
|
|
rval = x[i];
|
|
ival = x[n-i-1];
|
|
mag = bSi * sqrt(rval * rval + ival * ival);
|
|
|
|
if (mag > this.peak) {
|
|
this.peakBand = i;
|
|
this.peak = mag;
|
|
}
|
|
|
|
spectrum[i] = mag;
|
|
}
|
|
|
|
spectrum[0] = bSi * x[0];
|
|
|
|
return spectrum;
|
|
};
|
|
|
|
function Sampler(file, bufferSize, sampleRate, playStart, playEnd, loopStart, loopEnd, loopMode) {
|
|
this.file = file;
|
|
this.bufferSize = bufferSize;
|
|
this.sampleRate = sampleRate;
|
|
this.playStart = playStart || 0; // 0%
|
|
this.playEnd = playEnd || 1; // 100%
|
|
this.loopStart = loopStart || 0;
|
|
this.loopEnd = loopEnd || 1;
|
|
this.loopMode = loopMode || DSP.OFF;
|
|
this.loaded = false;
|
|
this.samples = [];
|
|
this.signal = new Float64Array(bufferSize);
|
|
this.frameCount = 0;
|
|
this.envelope = null;
|
|
this.amplitude = 1;
|
|
this.rootFrequency = 110; // A2 110
|
|
this.frequency = 550;
|
|
this.step = this.frequency / this.rootFrequency;
|
|
this.duration = 0;
|
|
this.samplesProcessed = 0;
|
|
this.playhead = 0;
|
|
|
|
var audio = /* new Audio();*/ document.createElement("AUDIO");
|
|
var self = this;
|
|
|
|
this.loadSamples = function(event) {
|
|
var buffer = DSP.getChannel(DSP.MIX, event.frameBuffer);
|
|
for ( var i = 0; i < buffer.length; i++) {
|
|
self.samples.push(buffer[i]);
|
|
}
|
|
};
|
|
|
|
this.loadComplete = function() {
|
|
// convert flexible js array into a fast typed array
|
|
self.samples = new Float64Array(self.samples);
|
|
self.loaded = true;
|
|
};
|
|
|
|
this.loadMetaData = function() {
|
|
self.duration = audio.duration;
|
|
};
|
|
|
|
audio.addEventListener("MozAudioAvailable", this.loadSamples, false);
|
|
audio.addEventListener("loadedmetadata", this.loadMetaData, false);
|
|
audio.addEventListener("ended", this.loadComplete, false);
|
|
audio.muted = true;
|
|
audio.src = file;
|
|
audio.play();
|
|
}
|
|
|
|
Sampler.prototype.applyEnvelope = function() {
|
|
this.envelope.process(this.signal);
|
|
return this.signal;
|
|
};
|
|
|
|
Sampler.prototype.generate = function() {
|
|
var frameOffset = this.frameCount * this.bufferSize;
|
|
|
|
var loopWidth = this.playEnd * this.samples.length - this.playStart * this.samples.length;
|
|
var playStartSamples = this.playStart * this.samples.length; // ie 0.5 -> 50% of the length
|
|
var playEndSamples = this.playEnd * this.samples.length; // ie 0.5 -> 50% of the length
|
|
var offset;
|
|
|
|
for ( var i = 0; i < this.bufferSize; i++ ) {
|
|
switch (this.loopMode) {
|
|
case DSP.OFF:
|
|
this.playhead = Math.round(this.samplesProcessed * this.step + playStartSamples);
|
|
if (this.playhead < (this.playEnd * this.samples.length) ) {
|
|
this.signal[i] = this.samples[this.playhead] * this.amplitude;
|
|
} else {
|
|
this.signal[i] = 0;
|
|
}
|
|
break;
|
|
|
|
case DSP.FW:
|
|
this.playhead = Math.round((this.samplesProcessed * this.step) % loopWidth + playStartSamples);
|
|
if (this.playhead < (this.playEnd * this.samples.length) ) {
|
|
this.signal[i] = this.samples[this.playhead] * this.amplitude;
|
|
}
|
|
break;
|
|
|
|
case DSP.BW:
|
|
this.playhead = playEndSamples - Math.round((this.samplesProcessed * this.step) % loopWidth);
|
|
if (this.playhead < (this.playEnd * this.samples.length) ) {
|
|
this.signal[i] = this.samples[this.playhead] * this.amplitude;
|
|
}
|
|
break;
|
|
|
|
case DSP.FWBW:
|
|
if ( Math.floor(this.samplesProcessed * this.step / loopWidth) % 2 === 0 ) {
|
|
this.playhead = Math.round((this.samplesProcessed * this.step) % loopWidth + playStartSamples);
|
|
} else {
|
|
this.playhead = playEndSamples - Math.round((this.samplesProcessed * this.step) % loopWidth);
|
|
}
|
|
if (this.playhead < (this.playEnd * this.samples.length) ) {
|
|
this.signal[i] = this.samples[this.playhead] * this.amplitude;
|
|
}
|
|
break;
|
|
}
|
|
this.samplesProcessed++;
|
|
}
|
|
|
|
this.frameCount++;
|
|
|
|
return this.signal;
|
|
};
|
|
|
|
Sampler.prototype.setFreq = function(frequency) {
|
|
var totalProcessed = this.samplesProcessed * this.step;
|
|
this.frequency = frequency;
|
|
this.step = this.frequency / this.rootFrequency;
|
|
this.samplesProcessed = Math.round(totalProcessed/this.step);
|
|
};
|
|
|
|
Sampler.prototype.reset = function() {
|
|
this.samplesProcessed = 0;
|
|
this.playhead = 0;
|
|
};
|
|
|
|
/**
|
|
* Oscillator class for generating and modifying signals
|
|
*
|
|
* @param {Number} type A waveform constant (eg. DSP.SINE)
|
|
* @param {Number} frequency Initial frequency of the signal
|
|
* @param {Number} amplitude Initial amplitude of the signal
|
|
* @param {Number} bufferSize Size of the sample buffer to generate
|
|
* @param {Number} sampleRate The sample rate of the signal
|
|
*
|
|
* @contructor
|
|
*/
|
|
function Oscillator(type, frequency, amplitude, bufferSize, sampleRate) {
|
|
this.frequency = frequency;
|
|
this.amplitude = amplitude;
|
|
this.bufferSize = bufferSize;
|
|
this.sampleRate = sampleRate;
|
|
//this.pulseWidth = pulseWidth;
|
|
this.frameCount = 0;
|
|
|
|
this.waveTableLength = 2048;
|
|
|
|
this.cyclesPerSample = frequency / sampleRate;
|
|
|
|
this.signal = new Float64Array(bufferSize);
|
|
this.envelope = null;
|
|
|
|
switch(parseInt(type, 10)) {
|
|
case DSP.TRIANGLE:
|
|
this.func = Oscillator.Triangle;
|
|
break;
|
|
|
|
case DSP.SAW:
|
|
this.func = Oscillator.Saw;
|
|
break;
|
|
|
|
case DSP.SQUARE:
|
|
this.func = Oscillator.Square;
|
|
break;
|
|
|
|
default:
|
|
case DSP.SINE:
|
|
this.func = Oscillator.Sine;
|
|
break;
|
|
}
|
|
|
|
this.generateWaveTable = function() {
|
|
Oscillator.waveTable[this.func] = new Float64Array(2048);
|
|
var waveTableTime = this.waveTableLength / this.sampleRate;
|
|
var waveTableHz = 1 / waveTableTime;
|
|
|
|
for (var i = 0; i < this.waveTableLength; i++) {
|
|
Oscillator.waveTable[this.func][i] = this.func(i * waveTableHz/this.sampleRate);
|
|
}
|
|
};
|
|
|
|
if ( typeof Oscillator.waveTable === 'undefined' ) {
|
|
Oscillator.waveTable = {};
|
|
}
|
|
|
|
if ( typeof Oscillator.waveTable[this.func] === 'undefined' ) {
|
|
this.generateWaveTable();
|
|
}
|
|
|
|
this.waveTable = Oscillator.waveTable[this.func];
|
|
}
|
|
|
|
/**
|
|
* Set the amplitude of the signal
|
|
*
|
|
* @param {Number} amplitude The amplitude of the signal (between 0 and 1)
|
|
*/
|
|
Oscillator.prototype.setAmp = function(amplitude) {
|
|
if (amplitude >= 0 && amplitude <= 1) {
|
|
this.amplitude = amplitude;
|
|
} else {
|
|
throw "Amplitude out of range (0..1).";
|
|
}
|
|
};
|
|
|
|
/**
|
|
* Set the frequency of the signal
|
|
*
|
|
* @param {Number} frequency The frequency of the signal
|
|
*/
|
|
Oscillator.prototype.setFreq = function(frequency) {
|
|
this.frequency = frequency;
|
|
this.cyclesPerSample = frequency / this.sampleRate;
|
|
};
|
|
|
|
// Add an oscillator
|
|
Oscillator.prototype.add = function(oscillator) {
|
|
for ( var i = 0; i < this.bufferSize; i++ ) {
|
|
//this.signal[i] += oscillator.valueAt(i);
|
|
this.signal[i] += oscillator.signal[i];
|
|
}
|
|
|
|
return this.signal;
|
|
};
|
|
|
|
// Add a signal to the current generated osc signal
|
|
Oscillator.prototype.addSignal = function(signal) {
|
|
for ( var i = 0; i < signal.length; i++ ) {
|
|
if ( i >= this.bufferSize ) {
|
|
break;
|
|
}
|
|
this.signal[i] += signal[i];
|
|
|
|
/*
|
|
// Constrain amplitude
|
|
if ( this.signal[i] > 1 ) {
|
|
this.signal[i] = 1;
|
|
} else if ( this.signal[i] < -1 ) {
|
|
this.signal[i] = -1;
|
|
}
|
|
*/
|
|
}
|
|
return this.signal;
|
|
};
|
|
|
|
// Add an envelope to the oscillator
|
|
Oscillator.prototype.addEnvelope = function(envelope) {
|
|
this.envelope = envelope;
|
|
};
|
|
|
|
Oscillator.prototype.applyEnvelope = function() {
|
|
this.envelope.process(this.signal);
|
|
};
|
|
|
|
Oscillator.prototype.valueAt = function(offset) {
|
|
return this.waveTable[offset % this.waveTableLength];
|
|
};
|
|
|
|
Oscillator.prototype.generate = function() {
|
|
var frameOffset = this.frameCount * this.bufferSize;
|
|
var step = this.waveTableLength * this.frequency / this.sampleRate;
|
|
var offset;
|
|
|
|
for ( var i = 0; i < this.bufferSize; i++ ) {
|
|
//var step = (frameOffset + i) * this.cyclesPerSample % 1;
|
|
//this.signal[i] = this.func(step) * this.amplitude;
|
|
//this.signal[i] = this.valueAt(Math.round((frameOffset + i) * step)) * this.amplitude;
|
|
offset = Math.round((frameOffset + i) * step);
|
|
this.signal[i] = this.waveTable[offset % this.waveTableLength] * this.amplitude;
|
|
}
|
|
|
|
this.frameCount++;
|
|
|
|
return this.signal;
|
|
};
|
|
|
|
Oscillator.Sine = function(step) {
|
|
return Math.sin(DSP.TWO_PI * step);
|
|
};
|
|
|
|
Oscillator.Square = function(step) {
|
|
return step < 0.5 ? 1 : -1;
|
|
};
|
|
|
|
Oscillator.Saw = function(step) {
|
|
return 2 * (step - Math.round(step));
|
|
};
|
|
|
|
Oscillator.Triangle = function(step) {
|
|
return 1 - 4 * Math.abs(Math.round(step) - step);
|
|
};
|
|
|
|
Oscillator.Pulse = function(step) {
|
|
// stub
|
|
};
|
|
|
|
function ADSR(attackLength, decayLength, sustainLevel, sustainLength, releaseLength, sampleRate) {
|
|
this.sampleRate = sampleRate;
|
|
// Length in seconds
|
|
this.attackLength = attackLength;
|
|
this.decayLength = decayLength;
|
|
this.sustainLevel = sustainLevel;
|
|
this.sustainLength = sustainLength;
|
|
this.releaseLength = releaseLength;
|
|
this.sampleRate = sampleRate;
|
|
|
|
// Length in samples
|
|
this.attackSamples = attackLength * sampleRate;
|
|
this.decaySamples = decayLength * sampleRate;
|
|
this.sustainSamples = sustainLength * sampleRate;
|
|
this.releaseSamples = releaseLength * sampleRate;
|
|
|
|
// Updates the envelope sample positions
|
|
this.update = function() {
|
|
this.attack = this.attackSamples;
|
|
this.decay = this.attack + this.decaySamples;
|
|
this.sustain = this.decay + this.sustainSamples;
|
|
this.release = this.sustain + this.releaseSamples;
|
|
};
|
|
|
|
this.update();
|
|
|
|
this.samplesProcessed = 0;
|
|
}
|
|
|
|
ADSR.prototype.noteOn = function() {
|
|
this.samplesProcessed = 0;
|
|
this.sustainSamples = this.sustainLength * this.sampleRate;
|
|
this.update();
|
|
};
|
|
|
|
// Send a note off when using a sustain of infinity to let the envelope enter the release phase
|
|
ADSR.prototype.noteOff = function() {
|
|
this.sustainSamples = this.samplesProcessed - this.decaySamples;
|
|
this.update();
|
|
};
|
|
|
|
ADSR.prototype.processSample = function(sample) {
|
|
var amplitude = 0;
|
|
|
|
if ( this.samplesProcessed <= this.attack ) {
|
|
amplitude = 0 + (1 - 0) * ((this.samplesProcessed - 0) / (this.attack - 0));
|
|
} else if ( this.samplesProcessed > this.attack && this.samplesProcessed <= this.decay ) {
|
|
amplitude = 1 + (this.sustainLevel - 1) * ((this.samplesProcessed - this.attack) / (this.decay - this.attack));
|
|
} else if ( this.samplesProcessed > this.decay && this.samplesProcessed <= this.sustain ) {
|
|
amplitude = this.sustainLevel;
|
|
} else if ( this.samplesProcessed > this.sustain && this.samplesProcessed <= this.release ) {
|
|
amplitude = this.sustainLevel + (0 - this.sustainLevel) * ((this.samplesProcessed - this.sustain) / (this.release - this.sustain));
|
|
}
|
|
|
|
return sample * amplitude;
|
|
};
|
|
|
|
ADSR.prototype.value = function() {
|
|
var amplitude = 0;
|
|
|
|
if ( this.samplesProcessed <= this.attack ) {
|
|
amplitude = 0 + (1 - 0) * ((this.samplesProcessed - 0) / (this.attack - 0));
|
|
} else if ( this.samplesProcessed > this.attack && this.samplesProcessed <= this.decay ) {
|
|
amplitude = 1 + (this.sustainLevel - 1) * ((this.samplesProcessed - this.attack) / (this.decay - this.attack));
|
|
} else if ( this.samplesProcessed > this.decay && this.samplesProcessed <= this.sustain ) {
|
|
amplitude = this.sustainLevel;
|
|
} else if ( this.samplesProcessed > this.sustain && this.samplesProcessed <= this.release ) {
|
|
amplitude = this.sustainLevel + (0 - this.sustainLevel) * ((this.samplesProcessed - this.sustain) / (this.release - this.sustain));
|
|
}
|
|
|
|
return amplitude;
|
|
};
|
|
|
|
ADSR.prototype.process = function(buffer) {
|
|
for ( var i = 0; i < buffer.length; i++ ) {
|
|
buffer[i] *= this.value();
|
|
|
|
this.samplesProcessed++;
|
|
}
|
|
|
|
return buffer;
|
|
};
|
|
|
|
|
|
ADSR.prototype.isActive = function() {
|
|
if ( this.samplesProcessed > this.release || this.samplesProcessed === -1 ) {
|
|
return false;
|
|
} else {
|
|
return true;
|
|
}
|
|
};
|
|
|
|
ADSR.prototype.disable = function() {
|
|
this.samplesProcessed = -1;
|
|
};
|
|
|
|
function IIRFilter(type, cutoff, resonance, sampleRate) {
|
|
this.sampleRate = sampleRate;
|
|
|
|
switch(type) {
|
|
case DSP.LOWPASS:
|
|
case DSP.LP12:
|
|
this.func = new IIRFilter.LP12(cutoff, resonance, sampleRate);
|
|
break;
|
|
}
|
|
}
|
|
|
|
IIRFilter.prototype.__defineGetter__('cutoff',
|
|
function() {
|
|
return this.func.cutoff;
|
|
}
|
|
);
|
|
|
|
IIRFilter.prototype.__defineGetter__('resonance',
|
|
function() {
|
|
return this.func.resonance;
|
|
}
|
|
);
|
|
|
|
IIRFilter.prototype.set = function(cutoff, resonance) {
|
|
this.func.calcCoeff(cutoff, resonance);
|
|
};
|
|
|
|
IIRFilter.prototype.process = function(buffer) {
|
|
this.func.process(buffer);
|
|
};
|
|
|
|
// Add an envelope to the filter
|
|
IIRFilter.prototype.addEnvelope = function(envelope) {
|
|
if ( envelope instanceof ADSR ) {
|
|
this.func.addEnvelope(envelope);
|
|
} else {
|
|
throw "Not an envelope.";
|
|
}
|
|
};
|
|
|
|
IIRFilter.LP12 = function(cutoff, resonance, sampleRate) {
|
|
this.sampleRate = sampleRate;
|
|
this.vibraPos = 0;
|
|
this.vibraSpeed = 0;
|
|
this.envelope = false;
|
|
|
|
this.calcCoeff = function(cutoff, resonance) {
|
|
this.w = 2.0 * Math.PI * cutoff / this.sampleRate;
|
|
this.q = 1.0 - this.w / (2.0 * (resonance + 0.5 / (1.0 + this.w)) + this.w - 2.0);
|
|
this.r = this.q * this.q;
|
|
this.c = this.r + 1.0 - 2.0 * Math.cos(this.w) * this.q;
|
|
|
|
this.cutoff = cutoff;
|
|
this.resonance = resonance;
|
|
};
|
|
|
|
this.calcCoeff(cutoff, resonance);
|
|
|
|
this.process = function(buffer) {
|
|
for ( var i = 0; i < buffer.length; i++ ) {
|
|
this.vibraSpeed += (buffer[i] - this.vibraPos) * this.c;
|
|
this.vibraPos += this.vibraSpeed;
|
|
this.vibraSpeed *= this.r;
|
|
|
|
/*
|
|
var temp = this.vibraPos;
|
|
|
|
if ( temp > 1.0 ) {
|
|
temp = 1.0;
|
|
} else if ( temp < -1.0 ) {
|
|
temp = -1.0;
|
|
} else if ( temp != temp ) {
|
|
temp = 1;
|
|
}
|
|
|
|
buffer[i] = temp;
|
|
*/
|
|
|
|
if (this.envelope) {
|
|
buffer[i] = (buffer[i] * (1 - this.envelope.value())) + (this.vibraPos * this.envelope.value());
|
|
this.envelope.samplesProcessed++;
|
|
} else {
|
|
buffer[i] = this.vibraPos;
|
|
}
|
|
}
|
|
};
|
|
};
|
|
|
|
IIRFilter.LP12.prototype.addEnvelope = function(envelope) {
|
|
this.envelope = envelope;
|
|
};
|
|
|
|
function IIRFilter2(type, cutoff, resonance, sampleRate) {
|
|
this.type = type;
|
|
this.cutoff = cutoff;
|
|
this.resonance = resonance;
|
|
this.sampleRate = sampleRate;
|
|
|
|
this.f = Float64Array(4);
|
|
this.f[0] = 0.0; // lp
|
|
this.f[1] = 0.0; // hp
|
|
this.f[2] = 0.0; // bp
|
|
this.f[3] = 0.0; // br
|
|
|
|
this.calcCoeff = function(cutoff, resonance) {
|
|
this.freq = 2 * Math.sin(Math.PI * Math.min(0.25, cutoff/(this.sampleRate*2)));
|
|
this.damp = Math.min(2 * (1 - Math.pow(resonance, 0.25)), Math.min(2, 2/this.freq - this.freq * 0.5));
|
|
};
|
|
|
|
this.calcCoeff(cutoff, resonance);
|
|
}
|
|
|
|
IIRFilter2.prototype.process = function(buffer) {
|
|
var input, output;
|
|
var f = this.f;
|
|
|
|
for ( var i = 0; i < buffer.length; i++ ) {
|
|
input = buffer[i];
|
|
|
|
// first pass
|
|
f[3] = input - this.damp * f[2];
|
|
f[0] = f[0] + this.freq * f[2];
|
|
f[1] = f[3] - f[0];
|
|
f[2] = this.freq * f[1] + f[2];
|
|
output = 0.5 * f[this.type];
|
|
|
|
// second pass
|
|
f[3] = input - this.damp * f[2];
|
|
f[0] = f[0] + this.freq * f[2];
|
|
f[1] = f[3] - f[0];
|
|
f[2] = this.freq * f[1] + f[2];
|
|
output += 0.5 * f[this.type];
|
|
|
|
if (this.envelope) {
|
|
buffer[i] = (buffer[i] * (1 - this.envelope.value())) + (output * this.envelope.value());
|
|
this.envelope.samplesProcessed++;
|
|
} else {
|
|
buffer[i] = output;
|
|
}
|
|
}
|
|
};
|
|
|
|
IIRFilter2.prototype.addEnvelope = function(envelope) {
|
|
if ( envelope instanceof ADSR ) {
|
|
this.envelope = envelope;
|
|
} else {
|
|
throw "This is not an envelope.";
|
|
}
|
|
};
|
|
|
|
IIRFilter2.prototype.set = function(cutoff, resonance) {
|
|
this.calcCoeff(cutoff, resonance);
|
|
};
|
|
|
|
|
|
|
|
function WindowFunction(type, alpha) {
|
|
this.alpha = alpha;
|
|
|
|
switch(type) {
|
|
case DSP.BARTLETT:
|
|
this.func = WindowFunction.Bartlett;
|
|
break;
|
|
|
|
case DSP.BARTLETTHANN:
|
|
this.func = WindowFunction.BartlettHann;
|
|
break;
|
|
|
|
case DSP.BLACKMAN:
|
|
this.func = WindowFunction.Blackman;
|
|
this.alpha = this.alpha || 0.16;
|
|
break;
|
|
|
|
case DSP.COSINE:
|
|
this.func = WindowFunction.Cosine;
|
|
break;
|
|
|
|
case DSP.GAUSS:
|
|
this.func = WindowFunction.Gauss;
|
|
this.alpha = this.alpha || 0.25;
|
|
break;
|
|
|
|
case DSP.HAMMING:
|
|
this.func = WindowFunction.Hamming;
|
|
break;
|
|
|
|
case DSP.HANN:
|
|
this.func = WindowFunction.Hann;
|
|
break;
|
|
|
|
case DSP.LANCZOS:
|
|
this.func = WindowFunction.Lanczoz;
|
|
break;
|
|
|
|
case DSP.RECTANGULAR:
|
|
this.func = WindowFunction.Rectangular;
|
|
break;
|
|
|
|
case DSP.TRIANGULAR:
|
|
this.func = WindowFunction.Triangular;
|
|
break;
|
|
}
|
|
}
|
|
|
|
WindowFunction.prototype.process = function(buffer) {
|
|
var length = buffer.length;
|
|
for ( var i = 0; i < length; i++ ) {
|
|
buffer[i] *= this.func(length, i, this.alpha);
|
|
}
|
|
return buffer;
|
|
};
|
|
|
|
WindowFunction.Bartlett = function(length, index) {
|
|
return 2 / (length - 1) * ((length - 1) / 2 - Math.abs(index - (length - 1) / 2));
|
|
};
|
|
|
|
WindowFunction.BartlettHann = function(length, index) {
|
|
return 0.62 - 0.48 * Math.abs(index / (length - 1) - 0.5) - 0.38 * Math.cos(DSP.TWO_PI * index / (length - 1));
|
|
};
|
|
|
|
WindowFunction.Blackman = function(length, index, alpha) {
|
|
var a0 = (1 - alpha) / 2;
|
|
var a1 = 0.5;
|
|
var a2 = alpha / 2;
|
|
|
|
return a0 - a1 * Math.cos(DSP.TWO_PI * index / (length - 1)) + a2 * Math.cos(4 * Math.PI * index / (length - 1));
|
|
};
|
|
|
|
WindowFunction.Cosine = function(length, index) {
|
|
return Math.cos(Math.PI * index / (length - 1) - Math.PI / 2);
|
|
};
|
|
|
|
WindowFunction.Gauss = function(length, index, alpha) {
|
|
return Math.pow(Math.E, -0.5 * Math.pow((index - (length - 1) / 2) / (alpha * (length - 1) / 2), 2));
|
|
};
|
|
|
|
WindowFunction.Hamming = function(length, index) {
|
|
return 0.54 - 0.46 * Math.cos(DSP.TWO_PI * index / (length - 1));
|
|
};
|
|
|
|
WindowFunction.Hann = function(length, index) {
|
|
return 0.5 * (1 - Math.cos(DSP.TWO_PI * index / (length - 1)));
|
|
};
|
|
|
|
WindowFunction.Lanczos = function(length, index) {
|
|
var x = 2 * index / (length - 1) - 1;
|
|
return Math.sin(Math.PI * x) / (Math.PI * x);
|
|
};
|
|
|
|
WindowFunction.Rectangular = function(length, index) {
|
|
return 1;
|
|
};
|
|
|
|
WindowFunction.Triangular = function(length, index) {
|
|
return 2 / length * (length / 2 - Math.abs(index - (length - 1) / 2));
|
|
};
|
|
|
|
function sinh (arg) {
|
|
// Returns the hyperbolic sine of the number, defined as (exp(number) - exp(-number))/2
|
|
//
|
|
// version: 1004.2314
|
|
// discuss at: http://phpjs.org/functions/sinh // + original by: Onno Marsman
|
|
// * example 1: sinh(-0.9834330348825909);
|
|
// * returns 1: -1.1497971402636502
|
|
return (Math.exp(arg) - Math.exp(-arg))/2;
|
|
}
|
|
|
|
/*
|
|
* Biquad filter
|
|
*
|
|
* Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
|
|
* Copyright 2010 Ricard Marxer. All rights reserved.
|
|
*
|
|
*/
|
|
// Implementation based on:
|
|
// http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
|
|
function Biquad(type, sampleRate) {
|
|
this.Fs = sampleRate;
|
|
this.type = type; // type of the filter
|
|
this.parameterType = DSP.Q; // type of the parameter
|
|
|
|
this.x_1_l = 0;
|
|
this.x_2_l = 0;
|
|
this.y_1_l = 0;
|
|
this.y_2_l = 0;
|
|
|
|
this.x_1_r = 0;
|
|
this.x_2_r = 0;
|
|
this.y_1_r = 0;
|
|
this.y_2_r = 0;
|
|
|
|
this.b0 = 1;
|
|
this.a0 = 1;
|
|
|
|
this.b1 = 0;
|
|
this.a1 = 0;
|
|
|
|
this.b2 = 0;
|
|
this.a2 = 0;
|
|
|
|
this.b0a0 = this.b0 / this.a0;
|
|
this.b1a0 = this.b1 / this.a0;
|
|
this.b2a0 = this.b2 / this.a0;
|
|
this.a1a0 = this.a1 / this.a0;
|
|
this.a2a0 = this.a2 / this.a0;
|
|
|
|
this.f0 = 3000; // "wherever it's happenin', man." Center Frequency or
|
|
// Corner Frequency, or shelf midpoint frequency, depending
|
|
// on which filter type. The "significant frequency".
|
|
|
|
this.dBgain = 12; // used only for peaking and shelving filters
|
|
|
|
this.Q = 1; // the EE kind of definition, except for peakingEQ in which A*Q is
|
|
// the classic EE Q. That adjustment in definition was made so that
|
|
// a boost of N dB followed by a cut of N dB for identical Q and
|
|
// f0/Fs results in a precisely flat unity gain filter or "wire".
|
|
|
|
this.BW = -3; // the bandwidth in octaves (between -3 dB frequencies for BPF
|
|
// and notch or between midpoint (dBgain/2) gain frequencies for
|
|
// peaking EQ
|
|
|
|
this.S = 1; // a "shelf slope" parameter (for shelving EQ only). When S = 1,
|
|
// the shelf slope is as steep as it can be and remain monotonically
|
|
// increasing or decreasing gain with frequency. The shelf slope, in
|
|
// dB/octave, remains proportional to S for all other values for a
|
|
// fixed f0/Fs and dBgain.
|
|
|
|
this.coefficients = function() {
|
|
var b = [this.b0, this.b1, this.b2];
|
|
var a = [this.a0, this.a1, this.a2];
|
|
return {b: b, a:a};
|
|
};
|
|
|
|
this.setFilterType = function(type) {
|
|
this.type = type;
|
|
this.recalculateCoefficients();
|
|
};
|
|
|
|
this.setSampleRate = function(rate) {
|
|
this.Fs = rate;
|
|
this.recalculateCoefficients();
|
|
};
|
|
|
|
this.setQ = function(q) {
|
|
this.parameterType = DSP.Q;
|
|
this.Q = Math.max(Math.min(q, 115.0), 0.001);
|
|
this.recalculateCoefficients();
|
|
};
|
|
|
|
this.setBW = function(bw) {
|
|
this.parameterType = DSP.BW;
|
|
this.BW = bw;
|
|
this.recalculateCoefficients();
|
|
};
|
|
|
|
this.setS = function(s) {
|
|
this.parameterType = DSP.S;
|
|
this.S = Math.max(Math.min(s, 5.0), 0.0001);
|
|
this.recalculateCoefficients();
|
|
};
|
|
|
|
this.setF0 = function(freq) {
|
|
this.f0 = freq;
|
|
this.recalculateCoefficients();
|
|
};
|
|
|
|
this.setDbGain = function(g) {
|
|
this.dBgain = g;
|
|
this.recalculateCoefficients();
|
|
};
|
|
|
|
this.recalculateCoefficients = function() {
|
|
var A;
|
|
if (type === DSP.PEAKING_EQ || type === DSP.LOW_SHELF || type === DSP.HIGH_SHELF ) {
|
|
A = Math.pow(10, (this.dBgain/40)); // for peaking and shelving EQ filters only
|
|
} else {
|
|
A = Math.sqrt( Math.pow(10, (this.dBgain/20)) );
|
|
}
|
|
|
|
var w0 = DSP.TWO_PI * this.f0 / this.Fs;
|
|
|
|
var cosw0 = Math.cos(w0);
|
|
var sinw0 = Math.sin(w0);
|
|
|
|
var alpha = 0;
|
|
|
|
switch (this.parameterType) {
|
|
case DSP.Q:
|
|
alpha = sinw0/(2*this.Q);
|
|
break;
|
|
|
|
case DSP.BW:
|
|
alpha = sinw0 * sinh( Math.LN2/2 * this.BW * w0/sinw0 );
|
|
break;
|
|
|
|
case DSP.S:
|
|
alpha = sinw0/2 * Math.sqrt( (A + 1/A)*(1/this.S - 1) + 2 );
|
|
break;
|
|
}
|
|
|
|
/**
|
|
FYI: The relationship between bandwidth and Q is
|
|
1/Q = 2*sinh(ln(2)/2*BW*w0/sin(w0)) (digital filter w BLT)
|
|
or 1/Q = 2*sinh(ln(2)/2*BW) (analog filter prototype)
|
|
|
|
The relationship between shelf slope and Q is
|
|
1/Q = sqrt((A + 1/A)*(1/S - 1) + 2)
|
|
*/
|
|
|
|
var coeff;
|
|
|
|
switch (this.type) {
|
|
case DSP.LPF: // H(s) = 1 / (s^2 + s/Q + 1)
|
|
this.b0 = (1 - cosw0)/2;
|
|
this.b1 = 1 - cosw0;
|
|
this.b2 = (1 - cosw0)/2;
|
|
this.a0 = 1 + alpha;
|
|
this.a1 = -2 * cosw0;
|
|
this.a2 = 1 - alpha;
|
|
break;
|
|
|
|
case DSP.HPF: // H(s) = s^2 / (s^2 + s/Q + 1)
|
|
this.b0 = (1 + cosw0)/2;
|
|
this.b1 = -(1 + cosw0);
|
|
this.b2 = (1 + cosw0)/2;
|
|
this.a0 = 1 + alpha;
|
|
this.a1 = -2 * cosw0;
|
|
this.a2 = 1 - alpha;
|
|
break;
|
|
|
|
case DSP.BPF_CONSTANT_SKIRT: // H(s) = s / (s^2 + s/Q + 1) (constant skirt gain, peak gain = Q)
|
|
this.b0 = sinw0/2;
|
|
this.b1 = 0;
|
|
this.b2 = -sinw0/2;
|
|
this.a0 = 1 + alpha;
|
|
this.a1 = -2*cosw0;
|
|
this.a2 = 1 - alpha;
|
|
break;
|
|
|
|
case DSP.BPF_CONSTANT_PEAK: // H(s) = (s/Q) / (s^2 + s/Q + 1) (constant 0 dB peak gain)
|
|
this.b0 = alpha;
|
|
this.b1 = 0;
|
|
this.b2 = -alpha;
|
|
this.a0 = 1 + alpha;
|
|
this.a1 = -2*cosw0;
|
|
this.a2 = 1 - alpha;
|
|
break;
|
|
|
|
case DSP.NOTCH: // H(s) = (s^2 + 1) / (s^2 + s/Q + 1)
|
|
this.b0 = 1;
|
|
this.b1 = -2*cosw0;
|
|
this.b2 = 1;
|
|
this.a0 = 1 + alpha;
|
|
this.a1 = -2*cosw0;
|
|
this.a2 = 1 - alpha;
|
|
break;
|
|
|
|
case DSP.APF: // H(s) = (s^2 - s/Q + 1) / (s^2 + s/Q + 1)
|
|
this.b0 = 1 - alpha;
|
|
this.b1 = -2*cosw0;
|
|
this.b2 = 1 + alpha;
|
|
this.a0 = 1 + alpha;
|
|
this.a1 = -2*cosw0;
|
|
this.a2 = 1 - alpha;
|
|
break;
|
|
|
|
case DSP.PEAKING_EQ: // H(s) = (s^2 + s*(A/Q) + 1) / (s^2 + s/(A*Q) + 1)
|
|
this.b0 = 1 + alpha*A;
|
|
this.b1 = -2*cosw0;
|
|
this.b2 = 1 - alpha*A;
|
|
this.a0 = 1 + alpha/A;
|
|
this.a1 = -2*cosw0;
|
|
this.a2 = 1 - alpha/A;
|
|
break;
|
|
|
|
case DSP.LOW_SHELF: // H(s) = A * (s^2 + (sqrt(A)/Q)*s + A)/(A*s^2 + (sqrt(A)/Q)*s + 1)
|
|
coeff = sinw0 * Math.sqrt( (A^2 + 1)*(1/this.S - 1) + 2*A );
|
|
this.b0 = A*((A+1) - (A-1)*cosw0 + coeff);
|
|
this.b1 = 2*A*((A-1) - (A+1)*cosw0);
|
|
this.b2 = A*((A+1) - (A-1)*cosw0 - coeff);
|
|
this.a0 = (A+1) + (A-1)*cosw0 + coeff;
|
|
this.a1 = -2*((A-1) + (A+1)*cosw0);
|
|
this.a2 = (A+1) + (A-1)*cosw0 - coeff;
|
|
break;
|
|
|
|
case DSP.HIGH_SHELF: // H(s) = A * (A*s^2 + (sqrt(A)/Q)*s + 1)/(s^2 + (sqrt(A)/Q)*s + A)
|
|
coeff = sinw0 * Math.sqrt( (A^2 + 1)*(1/this.S - 1) + 2*A );
|
|
this.b0 = A*((A+1) + (A-1)*cosw0 + coeff);
|
|
this.b1 = -2*A*((A-1) + (A+1)*cosw0);
|
|
this.b2 = A*((A+1) + (A-1)*cosw0 - coeff);
|
|
this.a0 = (A+1) - (A-1)*cosw0 + coeff;
|
|
this.a1 = 2*((A-1) - (A+1)*cosw0);
|
|
this.a2 = (A+1) - (A-1)*cosw0 - coeff;
|
|
break;
|
|
}
|
|
|
|
this.b0a0 = this.b0/this.a0;
|
|
this.b1a0 = this.b1/this.a0;
|
|
this.b2a0 = this.b2/this.a0;
|
|
this.a1a0 = this.a1/this.a0;
|
|
this.a2a0 = this.a2/this.a0;
|
|
};
|
|
|
|
this.process = function(buffer) {
|
|
//y[n] = (b0/a0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2]
|
|
// - (a1/a0)*y[n-1] - (a2/a0)*y[n-2]
|
|
|
|
var len = buffer.length;
|
|
var output = new Float64Array(len);
|
|
|
|
for ( var i=0; i<buffer.length; i++ ) {
|
|
output[i] = this.b0a0*buffer[i] + this.b1a0*this.x_1_l + this.b2a0*this.x_2_l - this.a1a0*this.y_1_l - this.a2a0*this.y_2_l;
|
|
this.y_2_l = this.y_1_l;
|
|
this.y_1_l = output[i];
|
|
this.x_2_l = this.x_1_l;
|
|
this.x_1_l = buffer[i];
|
|
}
|
|
|
|
return output;
|
|
};
|
|
|
|
this.processStereo = function(buffer) {
|
|
//y[n] = (b0/a0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2]
|
|
// - (a1/a0)*y[n-1] - (a2/a0)*y[n-2]
|
|
|
|
var len = buffer.length;
|
|
var output = new Float64Array(len);
|
|
|
|
for (var i = 0; i < len/2; i++) {
|
|
output[2*i] = this.b0a0*buffer[2*i] + this.b1a0*this.x_1_l + this.b2a0*this.x_2_l - this.a1a0*this.y_1_l - this.a2a0*this.y_2_l;
|
|
this.y_2_l = this.y_1_l;
|
|
this.y_1_l = output[2*i];
|
|
this.x_2_l = this.x_1_l;
|
|
this.x_1_l = buffer[2*i];
|
|
|
|
output[2*i+1] = this.b0a0*buffer[2*i+1] + this.b1a0*this.x_1_r + this.b2a0*this.x_2_r - this.a1a0*this.y_1_r - this.a2a0*this.y_2_r;
|
|
this.y_2_r = this.y_1_r;
|
|
this.y_1_r = output[2*i+1];
|
|
this.x_2_r = this.x_1_r;
|
|
this.x_1_r = buffer[2*i+1];
|
|
}
|
|
|
|
return output;
|
|
};
|
|
}
|
|
|
|
/*
|
|
* Magnitude to decibels
|
|
*
|
|
* Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
|
|
* Copyright 2010 Ricard Marxer. All rights reserved.
|
|
*
|
|
* @buffer array of magnitudes to convert to decibels
|
|
*
|
|
* @returns the array in decibels
|
|
*
|
|
*/
|
|
DSP.mag2db = function(buffer) {
|
|
var minDb = -120;
|
|
var minMag = Math.pow(10.0, minDb / 20.0);
|
|
|
|
var log = Math.log;
|
|
var max = Math.max;
|
|
|
|
var result = Float64Array(buffer.length);
|
|
for (var i=0; i<buffer.length; i++) {
|
|
result[i] = 20.0*log(max(buffer[i], minMag));
|
|
}
|
|
|
|
return result;
|
|
};
|
|
|
|
/*
|
|
* Frequency response
|
|
*
|
|
* Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
|
|
* Copyright 2010 Ricard Marxer. All rights reserved.
|
|
*
|
|
* Calculates the frequency response at the given points.
|
|
*
|
|
* @b b coefficients of the filter
|
|
* @a a coefficients of the filter
|
|
* @w w points (normally between -PI and PI) where to calculate the frequency response
|
|
*
|
|
* @returns the frequency response in magnitude
|
|
*
|
|
*/
|
|
DSP.freqz = function(b, a, w) {
|
|
var i, j;
|
|
|
|
if (!w) {
|
|
w = Float64Array(200);
|
|
for (i=0;i<w.length; i++) {
|
|
w[i] = DSP.TWO_PI/w.length * i - Math.PI;
|
|
}
|
|
}
|
|
|
|
var result = Float64Array(w.length);
|
|
|
|
var sqrt = Math.sqrt;
|
|
var cos = Math.cos;
|
|
var sin = Math.sin;
|
|
|
|
for (i=0; i<w.length; i++) {
|
|
var numerator = {real:0.0, imag:0.0};
|
|
for (j=0; j<b.length; j++) {
|
|
numerator.real += b[j] * cos(-j*w[i]);
|
|
numerator.imag += b[j] * sin(-j*w[i]);
|
|
}
|
|
|
|
var denominator = {real:0.0, imag:0.0};
|
|
for (j=0; j<a.length; j++) {
|
|
denominator.real += a[j] * cos(-j*w[i]);
|
|
denominator.imag += a[j] * sin(-j*w[i]);
|
|
}
|
|
|
|
result[i] = sqrt(numerator.real*numerator.real + numerator.imag*numerator.imag) / sqrt(denominator.real*denominator.real + denominator.imag*denominator.imag);
|
|
}
|
|
|
|
return result;
|
|
};
|
|
|
|
/*
|
|
* Graphical Equalizer
|
|
*
|
|
* Implementation of a graphic equalizer with a configurable bands-per-octave
|
|
* and minimum and maximum frequencies
|
|
*
|
|
* Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
|
|
* Copyright 2010 Ricard Marxer. All rights reserved.
|
|
*
|
|
*/
|
|
function GraphicalEq(sampleRate) {
|
|
this.FS = sampleRate;
|
|
this.minFreq = 40.0;
|
|
this.maxFreq = 16000.0;
|
|
|
|
this.bandsPerOctave = 1.0;
|
|
|
|
this.filters = [];
|
|
this.freqzs = [];
|
|
|
|
this.calculateFreqzs = true;
|
|
|
|
this.recalculateFilters = function() {
|
|
var bandCount = Math.round(Math.log(this.maxFreq/this.minFreq) * this.bandsPerOctave/ Math.LN2);
|
|
|
|
this.filters = [];
|
|
for (var i=0; i<bandCount; i++) {
|
|
var freq = this.minFreq*(Math.pow(2, i/this.bandsPerOctave));
|
|
var newFilter = new Biquad(DSP.PEAKING_EQ, this.FS);
|
|
newFilter.setDbGain(0);
|
|
newFilter.setBW(1/this.bandsPerOctave);
|
|
newFilter.setF0(freq);
|
|
this.filters[i] = newFilter;
|
|
this.recalculateFreqz(i);
|
|
}
|
|
};
|
|
|
|
this.setMinimumFrequency = function(freq) {
|
|
this.minFreq = freq;
|
|
this.recalculateFilters();
|
|
};
|
|
|
|
this.setMaximumFrequency = function(freq) {
|
|
this.maxFreq = freq;
|
|
this.recalculateFilters();
|
|
};
|
|
|
|
this.setBandsPerOctave = function(bands) {
|
|
this.bandsPerOctave = bands;
|
|
this.recalculateFilters();
|
|
};
|
|
|
|
this.setBandGain = function(bandIndex, gain) {
|
|
if (bandIndex < 0 || bandIndex > (this.filters.length-1)) {
|
|
throw "The band index of the graphical equalizer is out of bounds.";
|
|
}
|
|
|
|
if (!gain) {
|
|
throw "A gain must be passed.";
|
|
}
|
|
|
|
this.filters[bandIndex].setDbGain(gain);
|
|
this.recalculateFreqz(bandIndex);
|
|
};
|
|
|
|
this.recalculateFreqz = function(bandIndex) {
|
|
if (!this.calculateFreqzs) {
|
|
return;
|
|
}
|
|
|
|
if (bandIndex < 0 || bandIndex > (this.filters.length-1)) {
|
|
throw "The band index of the graphical equalizer is out of bounds. " + bandIndex + " is out of [" + 0 + ", " + this.filters.length-1 + "]";
|
|
}
|
|
|
|
if (!this.w) {
|
|
this.w = Float64Array(400);
|
|
for (var i=0; i<this.w.length; i++) {
|
|
this.w[i] = Math.PI/this.w.length * i;
|
|
}
|
|
}
|
|
|
|
var b = [this.filters[bandIndex].b0, this.filters[bandIndex].b1, this.filters[bandIndex].b2];
|
|
var a = [this.filters[bandIndex].a0, this.filters[bandIndex].a1, this.filters[bandIndex].a2];
|
|
|
|
this.freqzs[bandIndex] = DSP.mag2db(DSP.freqz(b, a, this.w));
|
|
};
|
|
|
|
this.process = function(buffer) {
|
|
var output = buffer;
|
|
|
|
for (var i = 0; i < this.filters.length; i++) {
|
|
output = this.filters[i].process(output);
|
|
}
|
|
|
|
return output;
|
|
};
|
|
|
|
this.processStereo = function(buffer) {
|
|
var output = buffer;
|
|
|
|
for (var i = 0; i < this.filters.length; i++) {
|
|
output = this.filters[i].processStereo(output);
|
|
}
|
|
|
|
return output;
|
|
};
|
|
}
|
|
|
|
/**
|
|
* MultiDelay effect by Almer Thie (http://code.almeros.com).
|
|
* Copyright 2010 Almer Thie. All rights reserved.
|
|
* Example: http://code.almeros.com/code-examples/delay-firefox-audio-api/
|
|
*
|
|
* This is a delay that feeds it's own delayed signal back into its circular
|
|
* buffer. Also known as a CombFilter.
|
|
*
|
|
* Compatible with interleaved stereo (or more channel) buffers and
|
|
* non-interleaved mono buffers.
|
|
*
|
|
* @param {Number} maxDelayInSamplesSize Maximum possible delay in samples (size of circular buffer)
|
|
* @param {Number} delayInSamples Initial delay in samples
|
|
* @param {Number} masterVolume Initial master volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
|
|
* @param {Number} delayVolume Initial feedback delay volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
|
|
*
|
|
* @constructor
|
|
*/
|
|
function MultiDelay(maxDelayInSamplesSize, delayInSamples, masterVolume, delayVolume) {
|
|
this.delayBufferSamples = new Float64Array(maxDelayInSamplesSize); // The maximum size of delay
|
|
this.delayInputPointer = delayInSamples;
|
|
this.delayOutputPointer = 0;
|
|
|
|
this.delayInSamples = delayInSamples;
|
|
this.masterVolume = masterVolume;
|
|
this.delayVolume = delayVolume;
|
|
}
|
|
|
|
/**
|
|
* Change the delay time in samples.
|
|
*
|
|
* @param {Number} delayInSamples Delay in samples
|
|
*/
|
|
MultiDelay.prototype.setDelayInSamples = function (delayInSamples) {
|
|
this.delayInSamples = delayInSamples;
|
|
|
|
this.delayInputPointer = this.delayOutputPointer + delayInSamples;
|
|
|
|
if (this.delayInputPointer >= this.delayBufferSamples.length-1) {
|
|
this.delayInputPointer = this.delayInputPointer - this.delayBufferSamples.length;
|
|
}
|
|
};
|
|
|
|
/**
|
|
* Change the master volume.
|
|
*
|
|
* @param {Number} masterVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
|
|
*/
|
|
MultiDelay.prototype.setMasterVolume = function(masterVolume) {
|
|
this.masterVolume = masterVolume;
|
|
};
|
|
|
|
/**
|
|
* Change the delay feedback volume.
|
|
*
|
|
* @param {Number} delayVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
|
|
*/
|
|
MultiDelay.prototype.setDelayVolume = function(delayVolume) {
|
|
this.delayVolume = delayVolume;
|
|
};
|
|
|
|
/**
|
|
* Process a given interleaved or mono non-interleaved float value Array and adds the delayed audio.
|
|
*
|
|
* @param {Array} samples Array containing Float values or a Float64Array
|
|
*
|
|
* @returns A new Float64Array interleaved or mono non-interleaved as was fed to this function.
|
|
*/
|
|
MultiDelay.prototype.process = function(samples) {
|
|
// NB. Make a copy to put in the output samples to return.
|
|
var outputSamples = new Float64Array(samples.length);
|
|
|
|
for (var i=0; i<samples.length; i++) {
|
|
// delayBufferSamples could contain initial NULL's, return silence in that case
|
|
var delaySample = (this.delayBufferSamples[this.delayOutputPointer] === null ? 0.0 : this.delayBufferSamples[this.delayOutputPointer]);
|
|
|
|
// Mix normal audio data with delayed audio
|
|
var sample = (delaySample * this.delayVolume) + samples[i];
|
|
|
|
// Add audio data with the delay in the delay buffer
|
|
this.delayBufferSamples[this.delayInputPointer] = sample;
|
|
|
|
// Return the audio with delay mix
|
|
outputSamples[i] = sample * this.masterVolume;
|
|
|
|
// Manage circulair delay buffer pointers
|
|
this.delayInputPointer++;
|
|
if (this.delayInputPointer >= this.delayBufferSamples.length-1) {
|
|
this.delayInputPointer = 0;
|
|
}
|
|
|
|
this.delayOutputPointer++;
|
|
if (this.delayOutputPointer >= this.delayBufferSamples.length-1) {
|
|
this.delayOutputPointer = 0;
|
|
}
|
|
}
|
|
|
|
return outputSamples;
|
|
};
|
|
|
|
/**
|
|
* SingleDelay effect by Almer Thie (http://code.almeros.com).
|
|
* Copyright 2010 Almer Thie. All rights reserved.
|
|
* Example: See usage in Reverb class
|
|
*
|
|
* This is a delay that does NOT feeds it's own delayed signal back into its
|
|
* circular buffer, neither does it return the original signal. Also known as
|
|
* an AllPassFilter(?).
|
|
*
|
|
* Compatible with interleaved stereo (or more channel) buffers and
|
|
* non-interleaved mono buffers.
|
|
*
|
|
* @param {Number} maxDelayInSamplesSize Maximum possible delay in samples (size of circular buffer)
|
|
* @param {Number} delayInSamples Initial delay in samples
|
|
* @param {Number} delayVolume Initial feedback delay volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
|
|
*
|
|
* @constructor
|
|
*/
|
|
|
|
function SingleDelay(maxDelayInSamplesSize, delayInSamples, delayVolume) {
|
|
this.delayBufferSamples = new Float64Array(maxDelayInSamplesSize); // The maximum size of delay
|
|
this.delayInputPointer = delayInSamples;
|
|
this.delayOutputPointer = 0;
|
|
|
|
this.delayInSamples = delayInSamples;
|
|
this.delayVolume = delayVolume;
|
|
}
|
|
|
|
/**
|
|
* Change the delay time in samples.
|
|
*
|
|
* @param {Number} delayInSamples Delay in samples
|
|
*/
|
|
SingleDelay.prototype.setDelayInSamples = function(delayInSamples) {
|
|
this.delayInSamples = delayInSamples;
|
|
this.delayInputPointer = this.delayOutputPointer + delayInSamples;
|
|
|
|
if (this.delayInputPointer >= this.delayBufferSamples.length-1) {
|
|
this.delayInputPointer = this.delayInputPointer - this.delayBufferSamples.length;
|
|
}
|
|
};
|
|
|
|
/**
|
|
* Change the return signal volume.
|
|
*
|
|
* @param {Number} delayVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
|
|
*/
|
|
SingleDelay.prototype.setDelayVolume = function(delayVolume) {
|
|
this.delayVolume = delayVolume;
|
|
};
|
|
|
|
/**
|
|
* Process a given interleaved or mono non-interleaved float value Array and
|
|
* returns the delayed audio.
|
|
*
|
|
* @param {Array} samples Array containing Float values or a Float64Array
|
|
*
|
|
* @returns A new Float64Array interleaved or mono non-interleaved as was fed to this function.
|
|
*/
|
|
SingleDelay.prototype.process = function(samples) {
|
|
// NB. Make a copy to put in the output samples to return.
|
|
var outputSamples = new Float64Array(samples.length);
|
|
|
|
for (var i=0; i<samples.length; i++) {
|
|
|
|
// Add audio data with the delay in the delay buffer
|
|
this.delayBufferSamples[this.delayInputPointer] = samples[i];
|
|
|
|
// delayBufferSamples could contain initial NULL's, return silence in that case
|
|
var delaySample = this.delayBufferSamples[this.delayOutputPointer];
|
|
|
|
// Return the audio with delay mix
|
|
outputSamples[i] = delaySample * this.delayVolume;
|
|
|
|
// Manage circulair delay buffer pointers
|
|
this.delayInputPointer++;
|
|
|
|
if (this.delayInputPointer >= this.delayBufferSamples.length-1) {
|
|
this.delayInputPointer = 0;
|
|
}
|
|
|
|
this.delayOutputPointer++;
|
|
|
|
if (this.delayOutputPointer >= this.delayBufferSamples.length-1) {
|
|
this.delayOutputPointer = 0;
|
|
}
|
|
}
|
|
|
|
return outputSamples;
|
|
};
|
|
|
|
/**
|
|
* Reverb effect by Almer Thie (http://code.almeros.com).
|
|
* Copyright 2010 Almer Thie. All rights reserved.
|
|
* Example: http://code.almeros.com/code-examples/reverb-firefox-audio-api/
|
|
*
|
|
* This reverb consists of 6 SingleDelays, 6 MultiDelays and an IIRFilter2
|
|
* for each of the two stereo channels.
|
|
*
|
|
* Compatible with interleaved stereo buffers only!
|
|
*
|
|
* @param {Number} maxDelayInSamplesSize Maximum possible delay in samples (size of circular buffers)
|
|
* @param {Number} delayInSamples Initial delay in samples for internal (Single/Multi)delays
|
|
* @param {Number} masterVolume Initial master volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
|
|
* @param {Number} mixVolume Initial reverb signal mix volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
|
|
* @param {Number} delayVolume Initial feedback delay volume for internal (Single/Multi)delays. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
|
|
* @param {Number} dampFrequency Initial low pass filter frequency. 0 to 44100 (depending on your maximum sampling frequency)
|
|
*
|
|
* @constructor
|
|
*/
|
|
function Reverb(maxDelayInSamplesSize, delayInSamples, masterVolume, mixVolume, delayVolume, dampFrequency) {
|
|
this.delayInSamples = delayInSamples;
|
|
this.masterVolume = masterVolume;
|
|
this.mixVolume = mixVolume;
|
|
this.delayVolume = delayVolume;
|
|
this.dampFrequency = dampFrequency;
|
|
|
|
this.NR_OF_MULTIDELAYS = 6;
|
|
this.NR_OF_SINGLEDELAYS = 6;
|
|
|
|
this.LOWPASSL = new IIRFilter2(DSP.LOWPASS, dampFrequency, 0, 44100);
|
|
this.LOWPASSR = new IIRFilter2(DSP.LOWPASS, dampFrequency, 0, 44100);
|
|
|
|
this.singleDelays = [];
|
|
|
|
var i, delayMultiply;
|
|
|
|
for (i = 0; i < this.NR_OF_SINGLEDELAYS; i++) {
|
|
delayMultiply = 1.0 + (i/7.0); // 1.0, 1.1, 1.2...
|
|
this.singleDelays[i] = new SingleDelay(maxDelayInSamplesSize, Math.round(this.delayInSamples * delayMultiply), this.delayVolume);
|
|
}
|
|
|
|
this.multiDelays = [];
|
|
|
|
for (i = 0; i < this.NR_OF_MULTIDELAYS; i++) {
|
|
delayMultiply = 1.0 + (i/10.0); // 1.0, 1.1, 1.2...
|
|
this.multiDelays[i] = new MultiDelay(maxDelayInSamplesSize, Math.round(this.delayInSamples * delayMultiply), this.masterVolume, this.delayVolume);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Change the delay time in samples as a base for all delays.
|
|
*
|
|
* @param {Number} delayInSamples Delay in samples
|
|
*/
|
|
Reverb.prototype.setDelayInSamples = function (delayInSamples){
|
|
this.delayInSamples = delayInSamples;
|
|
|
|
var i, delayMultiply;
|
|
|
|
for (i = 0; i < this.NR_OF_SINGLEDELAYS; i++) {
|
|
delayMultiply = 1.0 + (i/7.0); // 1.0, 1.1, 1.2...
|
|
this.singleDelays[i].setDelayInSamples( Math.round(this.delayInSamples * delayMultiply) );
|
|
}
|
|
|
|
for (i = 0; i < this.NR_OF_MULTIDELAYS; i++) {
|
|
delayMultiply = 1.0 + (i/10.0); // 1.0, 1.1, 1.2...
|
|
this.multiDelays[i].setDelayInSamples( Math.round(this.delayInSamples * delayMultiply) );
|
|
}
|
|
};
|
|
|
|
/**
|
|
* Change the master volume.
|
|
*
|
|
* @param {Number} masterVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
|
|
*/
|
|
Reverb.prototype.setMasterVolume = function (masterVolume){
|
|
this.masterVolume = masterVolume;
|
|
};
|
|
|
|
/**
|
|
* Change the reverb signal mix level.
|
|
*
|
|
* @param {Number} mixVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
|
|
*/
|
|
Reverb.prototype.setMixVolume = function (mixVolume){
|
|
this.mixVolume = mixVolume;
|
|
};
|
|
|
|
/**
|
|
* Change all delays feedback volume.
|
|
*
|
|
* @param {Number} delayVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
|
|
*/
|
|
Reverb.prototype.setDelayVolume = function (delayVolume){
|
|
this.delayVolume = delayVolume;
|
|
|
|
var i;
|
|
|
|
for (i = 0; i<this.NR_OF_SINGLEDELAYS; i++) {
|
|
this.singleDelays[i].setDelayVolume(this.delayVolume);
|
|
}
|
|
|
|
for (i = 0; i<this.NR_OF_MULTIDELAYS; i++) {
|
|
this.multiDelays[i].setDelayVolume(this.delayVolume);
|
|
}
|
|
};
|
|
|
|
/**
|
|
* Change the Low Pass filter frequency.
|
|
*
|
|
* @param {Number} dampFrequency low pass filter frequency. 0 to 44100 (depending on your maximum sampling frequency)
|
|
*/
|
|
Reverb.prototype.setDampFrequency = function (dampFrequency){
|
|
this.dampFrequency = dampFrequency;
|
|
|
|
this.LOWPASSL.set(dampFrequency, 0);
|
|
this.LOWPASSR.set(dampFrequency, 0);
|
|
};
|
|
|
|
/**
|
|
* Process a given interleaved float value Array and copies and adds the reverb signal.
|
|
*
|
|
* @param {Array} samples Array containing Float values or a Float64Array
|
|
*
|
|
* @returns A new Float64Array interleaved buffer.
|
|
*/
|
|
Reverb.prototype.process = function (interleavedSamples){
|
|
// NB. Make a copy to put in the output samples to return.
|
|
var outputSamples = new Float64Array(interleavedSamples.length);
|
|
|
|
// Perform low pass on the input samples to mimick damp
|
|
var leftRightMix = DSP.deinterleave(interleavedSamples);
|
|
this.LOWPASSL.process( leftRightMix[DSP.LEFT] );
|
|
this.LOWPASSR.process( leftRightMix[DSP.RIGHT] );
|
|
var filteredSamples = DSP.interleave(leftRightMix[DSP.LEFT], leftRightMix[DSP.RIGHT]);
|
|
|
|
var i;
|
|
|
|
// Process MultiDelays in parallel
|
|
for (i = 0; i<this.NR_OF_MULTIDELAYS; i++) {
|
|
// Invert the signal of every even multiDelay
|
|
outputSamples = DSP.mixSampleBuffers(outputSamples, this.multiDelays[i].process(filteredSamples), 2%i === 0, this.NR_OF_MULTIDELAYS);
|
|
}
|
|
|
|
// Process SingleDelays in series
|
|
var singleDelaySamples = new Float64Array(outputSamples.length);
|
|
for (i = 0; i<this.NR_OF_SINGLEDELAYS; i++) {
|
|
// Invert the signal of every even singleDelay
|
|
singleDelaySamples = DSP.mixSampleBuffers(singleDelaySamples, this.singleDelays[i].process(outputSamples), 2%i === 0, 1);
|
|
}
|
|
|
|
// Apply the volume of the reverb signal
|
|
for (i = 0; i<singleDelaySamples.length; i++) {
|
|
singleDelaySamples[i] *= this.mixVolume;
|
|
}
|
|
|
|
// Mix the original signal with the reverb signal
|
|
outputSamples = DSP.mixSampleBuffers(singleDelaySamples, interleavedSamples, 0, 1);
|
|
|
|
// Apply the master volume to the complete signal
|
|
for (i = 0; i<outputSamples.length; i++) {
|
|
outputSamples[i] *= this.masterVolume;
|
|
}
|
|
|
|
return outputSamples;
|
|
};
|
|
|
|
module.exports = {
|
|
agent:{
|
|
DSP: DSP,
|
|
DFT: DFT,
|
|
FFT: FFT,
|
|
RFFT: RFFT,
|
|
Sampler: Sampler,
|
|
Oscillator: Oscillator,
|
|
ADSR: ADSR,
|
|
IIRFilter: IIRFilter,
|
|
IIRFilter2: IIRFilter2,
|
|
WindowFunction: WindowFunction,
|
|
sinh: sinh,
|
|
Biquad: Biquad,
|
|
GraphicalEq: GraphicalEq,
|
|
MultiDelay: MultiDelay,
|
|
SingleDelay: SingleDelay,
|
|
Reverb: Reverb
|
|
},
|
|
DSP: DSP,
|
|
DFT: DFT,
|
|
FFT: FFT,
|
|
RFFT: RFFT,
|
|
Sampler: Sampler,
|
|
Oscillator: Oscillator,
|
|
ADSR: ADSR,
|
|
IIRFilter: IIRFilter,
|
|
IIRFilter2: IIRFilter2,
|
|
WindowFunction: WindowFunction,
|
|
sinh: sinh,
|
|
Biquad: Biquad,
|
|
GraphicalEq: GraphicalEq,
|
|
MultiDelay: MultiDelay,
|
|
SingleDelay: SingleDelay,
|
|
Reverb: Reverb,
|
|
|
|
current:function (module) { current=module.current; Aios=module; }
|
|
};
|
|
|