1762 lines
52 KiB
JavaScript
1762 lines
52 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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* https://github.com/corbanbrook/dsp.js/
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*/
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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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// 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("Float32Array", "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 Float32Array(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 Float32Array(buffer.length/2);
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right = right || new Float32Array(buffer.length/2);
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mix = mix || new Float32Array(buffer.length/2);
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if (buffer.length/2 !== left.length) {
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left = new Float32Array(buffer.length/2);
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right = new Float32Array(buffer.length/2);
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mix = new Float32Array(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 Float32Array
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* @param {Array} sampleBuffer2 Array containing Float values or a Float32Array
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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 Float32Array interleaved buffer.
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*/
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DSP.mixSampleBuffers = function(sampleBuffer1, sampleBuffer2, negate, volumeCorrection){
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var outputSamples = new Float32Array(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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// 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 Float32Array(_bufferSize/2);
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this.real = new Float32Array(_bufferSize);
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this.imag = new Float32Array(_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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};
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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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DFT = function(_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 Float32Array(N);
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this.cosTable = new Float32Array(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) {
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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 this.calculateSpectrum();
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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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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 Float32Array(_bufferSize);
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this.cosTable = new Float32Array(_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) {
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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,
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reverseTable = this.reverseTable,
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real = this.real,
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imag = this.imag,
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spectrum = this.spectrum;
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var k = Math.floor(Math.log(_bufferSize) / Math.LN2);
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if (Math.pow(2, k) !== _bufferSize) { throw "Invalid buffer size, must be a power of 2."; }
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if (_bufferSize !== buffer.length) { throw "Supplied buffer is not the same size as defined FFT. FFT Size: " + _bufferSize + " Buffer Size: " + buffer.length; }
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var halfSize = 1,
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phaseShiftStepReal,
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phaseShiftStepImag,
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currentPhaseShiftReal,
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currentPhaseShiftImag,
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off,
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tr,
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ti,
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tmpReal,
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i;
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for (i = 0; i < _bufferSize; i++) {
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real[i] = buffer[reverseTable[i]];
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imag[i] = 0;
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}
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while (halfSize < _bufferSize) {
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//phaseShiftStepReal = Math.cos(-Math.PI/halfSize);
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//phaseShiftStepImag = Math.sin(-Math.PI/halfSize);
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phaseShiftStepReal = cosTable[halfSize];
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phaseShiftStepImag = sinTable[halfSize];
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currentPhaseShiftReal = 1;
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currentPhaseShiftImag = 0;
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for (var fftStep = 0; fftStep < halfSize; fftStep++) {
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i = fftStep;
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while (i < _bufferSize) {
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off = i + halfSize;
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tr = (currentPhaseShiftReal * real[off]) - (currentPhaseShiftImag * imag[off]);
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ti = (currentPhaseShiftReal * imag[off]) + (currentPhaseShiftImag * real[off]);
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real[off] = real[i] - tr;
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imag[off] = imag[i] - ti;
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real[i] += tr;
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imag[i] += ti;
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i += halfSize << 1;
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}
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tmpReal = currentPhaseShiftReal;
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currentPhaseShiftReal = (tmpReal * phaseShiftStepReal) - (currentPhaseShiftImag * phaseShiftStepImag);
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currentPhaseShiftImag = (tmpReal * phaseShiftStepImag) + (currentPhaseShiftImag * phaseShiftStepReal);
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}
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halfSize = halfSize << 1;
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}
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return this.calculateSpectrum();
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};
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FFT.prototype.inverse = function(real, imag) {
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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,
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reverseTable = this.reverseTable,
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spectrum = this.spectrum;
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real = real || this.real;
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imag = imag || this.imag;
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var halfSize = 1,
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phaseShiftStepReal,
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phaseShiftStepImag,
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currentPhaseShiftReal,
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currentPhaseShiftImag,
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off,
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tr,
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ti,
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tmpReal,
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i;
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for (i = 0; i < _bufferSize; i++) {
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imag[i] *= -1;
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}
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var revReal = new Float32Array(_bufferSize);
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var revImag = new Float32Array(_bufferSize);
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for (i = 0; i < real.length; i++) {
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revReal[i] = real[reverseTable[i]];
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revImag[i] = imag[reverseTable[i]];
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}
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real = revReal;
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imag = revImag;
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while (halfSize < _bufferSize) {
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phaseShiftStepReal = cosTable[halfSize];
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phaseShiftStepImag = sinTable[halfSize];
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currentPhaseShiftReal = 1;
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currentPhaseShiftImag = 0;
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for (var fftStep = 0; fftStep < halfSize; fftStep++) {
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i = fftStep;
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while (i < _bufferSize) {
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off = i + halfSize;
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tr = (currentPhaseShiftReal * real[off]) - (currentPhaseShiftImag * imag[off]);
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ti = (currentPhaseShiftReal * imag[off]) + (currentPhaseShiftImag * real[off]);
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real[off] = real[i] - tr;
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imag[off] = imag[i] - ti;
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real[i] += tr;
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imag[i] += ti;
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i += halfSize << 1;
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}
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tmpReal = currentPhaseShiftReal;
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currentPhaseShiftReal = (tmpReal * phaseShiftStepReal) - (currentPhaseShiftImag * phaseShiftStepImag);
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currentPhaseShiftImag = (tmpReal * phaseShiftStepImag) + (currentPhaseShiftImag * phaseShiftStepReal);
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}
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halfSize = halfSize << 1;
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}
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var buffer = new Float32Array(_bufferSize); // this should be reused instead
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for (i = 0; i < _bufferSize; i++) {
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buffer[i] = real[i] / _bufferSize;
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}
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return buffer;
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};
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/**
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* RFFT 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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* This method currently only contains a forward transform but is highly optimized.
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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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// lookup tables don't really gain us any speed, but they do increase
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// cache footprint, so don't use them in here
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// also we don't use sepearate arrays for real/imaginary parts
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// this one a little more than twice as fast as the one in FFT
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// however I only did the forward transform
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// the rest of this was translated from C, see http://www.jjj.de/fxt/
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// this is the real split radix FFT
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function RFFT(_bufferSize, _sampleRate) {
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FourierTransform.call(this, _bufferSize, _sampleRate);
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this.trans = new Float32Array(_bufferSize);
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this.reverseTable = new Uint32Array(_bufferSize);
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// don't use a lookup table to do the permute, use this instead
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this.reverseBinPermute = function (dest, source) {
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var _bufferSize = this._bufferSize,
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halfSize = _bufferSize >>> 1,
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nm1 = _bufferSize - 1,
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i = 1, r = 0, h;
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dest[0] = source[0];
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do {
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r += halfSize;
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dest[i] = source[r];
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dest[r] = source[i];
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i++;
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h = halfSize << 1;
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while (h = h >> 1, !((r ^= h) & h));
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if (r >= i) {
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dest[i] = source[r];
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dest[r] = source[i];
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dest[nm1-i] = source[nm1-r];
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dest[nm1-r] = source[nm1-i];
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}
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i++;
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} while (i < halfSize);
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dest[nm1] = source[nm1];
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};
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this.generateReverseTable = function () {
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var _bufferSize = this._bufferSize,
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halfSize = _bufferSize >>> 1,
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nm1 = _bufferSize - 1,
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i = 1, r = 0, h;
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this.reverseTable[0] = 0;
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do {
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r += halfSize;
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this.reverseTable[i] = r;
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this.reverseTable[r] = i;
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i++;
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h = halfSize << 1;
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while (h = h >> 1, !((r ^= h) & h));
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if (r >= i) {
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this.reverseTable[i] = r;
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this.reverseTable[r] = i;
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this.reverseTable[nm1-i] = nm1-r;
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this.reverseTable[nm1-r] = nm1-i;
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}
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i++;
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} while (i < halfSize);
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this.reverseTable[nm1] = nm1;
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};
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this.generateReverseTable();
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}
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// Ordering of output:
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//
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// trans[0] = re[0] (==zero _frequency, purely real)
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// trans[1] = re[1]
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// ...
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// trans[n/2-1] = re[n/2-1]
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// trans[n/2] = re[n/2] (==nyquist _frequency, purely real)
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//
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// trans[n/2+1] = im[n/2-1]
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// trans[n/2+2] = im[n/2-2]
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// ...
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// trans[n-1] = im[1]
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RFFT.prototype.forward = function(buffer) {
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var n = this._bufferSize,
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spectrum = this.spectrum,
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x = this.trans,
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TWO_PI = 2*Math.PI,
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sqrt = Math.sqrt,
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i = n >>> 1,
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bSi = 2 / n,
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n2, n4, n8, nn,
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t1, t2, t3, t4,
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i1, i2, i3, i4, i5, i6, i7, i8,
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st1, cc1, ss1, cc3, ss3,
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e,
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a,
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rval, ival, mag;
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this.reverseBinPermute(x, buffer);
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/*
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var reverseTable = this.reverseTable;
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for (var k = 0, len = reverseTable.length; k < len; k++) {
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x[k] = buffer[reverseTable[k]];
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}
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*/
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for (var ix = 0, id = 4; ix < n; id *= 4) {
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for (var i0 = ix; i0 < n; i0 += id) {
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//sumdiff(x[i0], x[i0+1]); // {a, b} <--| {a+b, a-b}
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st1 = x[i0] - x[i0+1];
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x[i0] += x[i0+1];
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x[i0+1] = st1;
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}
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ix = 2*(id-1);
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}
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n2 = 2;
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nn = n >>> 1;
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while((nn = nn >>> 1)) {
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ix = 0;
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n2 = n2 << 1;
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id = n2 << 1;
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n4 = n2 >>> 2;
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n8 = n2 >>> 3;
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do {
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|
if (n4 !== 1) {
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for(i0 = ix; i0 < n; i0 += id) {
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i1 = i0;
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i2 = i1 + n4;
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i3 = i2 + n4;
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i4 = i3 + n4;
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//diffsum3_r(x[i3], x[i4], t1); // {a, b, s} <--| {a, b-a, a+b}
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t1 = x[i3] + x[i4];
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x[i4] -= x[i3];
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//sumdiff3(x[i1], t1, x[i3]); // {a, b, d} <--| {a+b, b, a-b}
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x[i3] = x[i1] - t1;
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x[i1] += t1;
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i1 += n8;
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i2 += n8;
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i3 += n8;
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i4 += n8;
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//sumdiff(x[i3], x[i4], t1, t2); // {s, d} <--| {a+b, a-b}
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t1 = x[i3] + x[i4];
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t2 = x[i3] - x[i4];
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t1 = -t1 * Math.SQRT1_2;
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t2 *= Math.SQRT1_2;
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// sumdiff(t1, x[i2], x[i4], x[i3]); // {s, d} <--| {a+b, a-b}
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st1 = x[i2];
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x[i4] = t1 + st1;
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x[i3] = t1 - st1;
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//sumdiff3(x[i1], t2, x[i2]); // {a, b, d} <--| {a+b, b, a-b}
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x[i2] = x[i1] - t2;
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x[i1] += t2;
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}
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} else {
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for(i0 = ix; i0 < n; i0 += id) {
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i1 = i0;
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i2 = i1 + n4;
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i3 = i2 + n4;
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i4 = i3 + n4;
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|
//diffsum3_r(x[i3], x[i4], t1); // {a, b, s} <--| {a, b-a, a+b}
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t1 = x[i3] + x[i4];
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x[i4] -= x[i3];
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//sumdiff3(x[i1], t1, x[i3]); // {a, b, d} <--| {a+b, b, a-b}
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x[i3] = x[i1] - t1;
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x[i1] += t1;
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}
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|
}
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|
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ix = (id << 1) - n2;
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|
id = id << 2;
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|
} while (ix < n);
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|
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e = TWO_PI / n2;
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|
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|
for (var j = 1; j < n8; j++) {
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|
a = j * e;
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|
ss1 = Math.sin(a);
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|
cc1 = Math.cos(a);
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|
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|
//ss3 = sin(3*a); cc3 = cos(3*a);
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|
cc3 = 4*cc1*(cc1*cc1-0.75);
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|
ss3 = 4*ss1*(0.75-ss1*ss1);
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|
|
|
ix = 0; id = n2 << 1;
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|
do {
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|
for (i0 = ix; i0 < n; i0 += id) {
|
|
i1 = i0 + j;
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|
i2 = i1 + n4;
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|
i3 = i2 + n4;
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|
i4 = i3 + n4;
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|
i5 = i0 + n4 - j;
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|
i6 = i5 + n4;
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|
i7 = i6 + n4;
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|
i8 = i7 + n4;
|
|
|
|
//cmult(c, s, x, y, &u, &v)
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|
//cmult(cc1, ss1, x[i7], x[i3], t2, t1); // {u,v} <--| {x*c-y*s, x*s+y*c}
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|
t2 = x[i7]*cc1 - x[i3]*ss1;
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|
t1 = x[i7]*ss1 + x[i3]*cc1;
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|
|
|
//cmult(cc3, ss3, x[i8], x[i4], t4, t3);
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t4 = x[i8]*cc3 - x[i4]*ss3;
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|
t3 = x[i8]*ss3 + x[i4]*cc3;
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|
|
|
//sumdiff(t2, t4); // {a, b} <--| {a+b, a-b}
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st1 = t2 - t4;
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|
t2 += t4;
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|
t4 = st1;
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|
|
|
//sumdiff(t2, x[i6], x[i8], x[i3]); // {s, d} <--| {a+b, a-b}
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//st1 = x[i6]; x[i8] = t2 + st1; x[i3] = t2 - st1;
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|
x[i8] = t2 + x[i6];
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|
x[i3] = t2 - x[i6];
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|
|
|
//sumdiff_r(t1, t3); // {a, b} <--| {a+b, b-a}
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|
st1 = t3 - t1;
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|
t1 += t3;
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|
t3 = st1;
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|
|
|
//sumdiff(t3, x[i2], x[i4], x[i7]); // {s, d} <--| {a+b, a-b}
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//st1 = x[i2]; x[i4] = t3 + st1; x[i7] = t3 - st1;
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|
x[i4] = t3 + x[i2];
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|
x[i7] = t3 - x[i2];
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|
|
|
//sumdiff3(x[i1], t1, x[i6]); // {a, b, d} <--| {a+b, b, a-b}
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|
x[i6] = x[i1] - t1;
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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);
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|
|
|
if (mag > this.peak) {
|
|
this.peakBand = i;
|
|
this.peak = mag;
|
|
}
|
|
|
|
spectrum[i] = mag;
|
|
}
|
|
|
|
spectrum[0] = bSi * x[0];
|
|
|
|
return spectrum;
|
|
};
|
|
|
|
|
|
|
|
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 Float32Array(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 Float32Array(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 = Float32Array(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 = Float32Array(200);
|
|
for (i=0;i<w.length; i++) {
|
|
w[i] = DSP.TWO_PI/w.length * i - Math.PI;
|
|
}
|
|
}
|
|
|
|
var result = Float32Array(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 = Float32Array(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 Float32Array(maxDelayInSamplesSize); // The maximum size of delay
|
|
this.delayRotationPointer = 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.delayRotationPointer = this.delayOutputPointer + delayInSamples;
|
|
|
|
if (this.delayRotationPointer >= this.delayBufferSamples.length-1) {
|
|
this.delayRotationPointer = this.delayRotationPointer - 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 Float32Array
|
|
*
|
|
* @returns A new Float32Array 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 Float32Array(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.delayRotationPointer] = sample;
|
|
|
|
// Return the audio with delay mix
|
|
outputSamples[i] = sample * this.masterVolume;
|
|
|
|
// Manage circulair delay buffer pointers
|
|
this.delayRotationPointer++;
|
|
if (this.delayRotationPointer >= this.delayBufferSamples.length-1) {
|
|
this.delayRotationPointer = 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 Float32Array(maxDelayInSamplesSize); // The maximum size of delay
|
|
this.delayRotationPointer = 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.delayRotationPointer = this.delayOutputPointer + delayInSamples;
|
|
|
|
if (this.delayRotationPointer >= this.delayBufferSamples.length-1) {
|
|
this.delayRotationPointer = this.delayRotationPointer - 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 Float32Array
|
|
*
|
|
* @returns A new Float32Array 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 Float32Array(samples.length);
|
|
|
|
for (var i=0; i<samples.length; i++) {
|
|
|
|
// Add audio data with the delay in the delay buffer
|
|
this.delayBufferSamples[this.delayRotationPointer] = 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.delayRotationPointer++;
|
|
|
|
if (this.delayRotationPointer >= this.delayBufferSamples.length-1) {
|
|
this.delayRotationPointer = 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 Float32Array
|
|
*
|
|
* @returns A new Float32Array interleaved buffer.
|
|
*/
|
|
Reverb.prototype.process = function (interleavedSamples){
|
|
// NB. Make a copy to put in the output samples to return.
|
|
var outputSamples = new Float32Array(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 Float32Array(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;
|
|
};
|