From 3df3b88c176048c5c8cb28bc59e436cf3f5f3030 Mon Sep 17 00:00:00 2001
From: danigb <danigb@gmail.com>
Date: Thu, 22 Dec 2016 02:16:32 +0100
Subject: [PATCH] Split source in modules. Generate API documentation. Add
eslinter.
- Split the source code into modules inside `lib/`
- Generate browser ready distribution files into `dist/` (replaces the old `dsp.js` file)
- Add missing jsdoc documentation
- Generate API documentation into `docs/`
- Update the benchmarks code to use in node
- Update the examples to use the new `dsp.min.js` file
- Add eslint to make syntax coherent and prevent errors
- Add docs/README.md with the new install and development instructions
---
.eslintrc.json | 21 +
.gitignore | 2 +
Makefile | 6 +-
bench/bench.js | 6 +-
bench/deinterleave.js | 23 +-
bench/dft.js | 12 +-
bench/fft.js | 12 +-
bench/index.js | 6 +
bench/rfft.js | 12 +-
dsp.js => dist/dsp.js | 3843 ++++++++++++----------
dist/dsp.min.js | 3 +
docs/API.md | 999 ++++++
docs/README.md | 51 +
examples/biquad.html | 2 +-
examples/fft.html | 30 +-
examples/filter.html | 26 +-
examples/grapheq.html | 2 +-
examples/nowave.html | 26 +-
examples/rfft.html | 2 +-
examples/sampler.html | 2 +-
examples/squarewave.html | 40 +-
examples/synthesizer.html | 200 +-
examples/vocoder.html | 2 +-
lib/adsr.js | 134 +
lib/biquad.js | 288 ++
lib/dft.js | 57 +
lib/dsp.js | 315 ++
lib/fft.js | 198 ++
lib/fourier.js | 57 +
lib/graphical-eq.js | 117 +
lib/iir-filter.js | 120 +
lib/iir-filter2.js | 88 +
lib/index.js | 34 +
lib/multi-delay.js | 102 +
lib/oscillator.js | 195 ++
lib/reverb.js | 174 +
lib/rfft.js | 312 ++
lib/sampler.js | 133 +
lib/single-delay.js | 93 +
lib/sinh.js | 16 +
lib/window-function.js | 115 +
package.json | 23 +-
test/{ => audio-test}/adsr-test.js | 0
test/{ => audio-test}/audio-harness.js | 0
test/{ => audio-test}/beatdetect-test.js | 0
test/{ => audio-test}/beatdetektor.js | 0
test/{ => audio-test}/dft-test.js | 0
test/{ => audio-test}/fft-test.js | 0
test/{ => audio-test}/filter-test.js | 0
test/{ => audio-test}/iirfilter-test.js | 0
test/{ => audio-test}/multidelay-test.js | 0
test/{ => audio-test}/oscillator-test.js | 0
test/{ => audio-test}/reverb-test.js | 0
test/{ => audio-test}/samples.js | 0
test/test-exports.js | 16 +
55 files changed, 5929 insertions(+), 1986 deletions(-)
create mode 100644 .eslintrc.json
create mode 100644 bench/index.js
rename dsp.js => dist/dsp.js (85%)
create mode 100644 dist/dsp.min.js
create mode 100644 docs/API.md
create mode 100644 docs/README.md
create mode 100644 lib/adsr.js
create mode 100644 lib/biquad.js
create mode 100644 lib/dft.js
create mode 100644 lib/dsp.js
create mode 100644 lib/fft.js
create mode 100644 lib/fourier.js
create mode 100644 lib/graphical-eq.js
create mode 100644 lib/iir-filter.js
create mode 100644 lib/iir-filter2.js
create mode 100644 lib/index.js
create mode 100644 lib/multi-delay.js
create mode 100644 lib/oscillator.js
create mode 100644 lib/reverb.js
create mode 100644 lib/rfft.js
create mode 100644 lib/sampler.js
create mode 100644 lib/single-delay.js
create mode 100644 lib/sinh.js
create mode 100644 lib/window-function.js
rename test/{ => audio-test}/adsr-test.js (100%)
rename test/{ => audio-test}/audio-harness.js (100%)
rename test/{ => audio-test}/beatdetect-test.js (100%)
rename test/{ => audio-test}/beatdetektor.js (100%)
rename test/{ => audio-test}/dft-test.js (100%)
rename test/{ => audio-test}/fft-test.js (100%)
rename test/{ => audio-test}/filter-test.js (100%)
rename test/{ => audio-test}/iirfilter-test.js (100%)
rename test/{ => audio-test}/multidelay-test.js (100%)
rename test/{ => audio-test}/oscillator-test.js (100%)
rename test/{ => audio-test}/reverb-test.js (100%)
rename test/{ => audio-test}/samples.js (100%)
create mode 100644 test/test-exports.js
diff --git a/.eslintrc.json b/.eslintrc.json
new file mode 100644
index 0000000..4fdf69e
--- /dev/null
+++ b/.eslintrc.json
@@ -0,0 +1,21 @@
+{
+ "env": {
+ "browser": true,
+ "node": true
+ },
+ "extends": "eslint:recommended",
+ "rules": {
+ "linebreak-style": [
+ "error",
+ "unix"
+ ],
+ "quotes": [
+ "error",
+ "double"
+ ],
+ "semi": [
+ "error",
+ "always"
+ ]
+ }
+}
diff --git a/.gitignore b/.gitignore
index b0dcffa..38d0df3 100644
--- a/.gitignore
+++ b/.gitignore
@@ -3,3 +3,5 @@ examples/audio
.project
*~
.idea
+node_modules/
+npm-debug.log
diff --git a/Makefile b/Makefile
index 1ac12e9..4b47878 100644
--- a/Makefile
+++ b/Makefile
@@ -7,9 +7,9 @@ VERSION ?= $(error Specify a version for your release (e.g., VERSION=0.5))
FREQUENCY ?= 440
benchmark:
- ${JSSHELL} -m -j -p -e 'var FREQUENCY=${FREQUENCY};' -f dsp.js -f ./bench/bench.js -f ./bench/dft.js
- ${JSSHELL} -m -j -p -e 'var FREQUENCY=${FREQUENCY};' -f dsp.js -f ./bench/bench.js -f ./bench/fft.js
- ${JSSHELL} -m -j -p -e 'var FREQUENCY=${FREQUENCY};' -f dsp.js -f ./bench/bench.js -f ./bench/rfft.js
+ ${JSSHELL} -m -j -p -e 'var FREQUENCY=${FREQUENCY};' -f ./dist/dsp.js -f ./bench/bench.js -f ./bench/dft.js
+ ${JSSHELL} -m -j -p -e 'var FREQUENCY=${FREQUENCY};' -f ./dist/dsp.js -f ./bench/bench.js -f ./bench/fft.js
+ ${JSSHELL} -m -j -p -e 'var FREQUENCY=${FREQUENCY};' -f ./dist/dsp.js -f ./bench/bench.js -f ./bench/rfft.js
${JSSHELL} -m -j -p -f dsp.js -f ./bench/bench.js -f ./bench/deinterleave.js
clean:
diff --git a/bench/bench.js b/bench/bench.js
index 7894b94..b4268e4 100644
--- a/bench/bench.js
+++ b/bench/bench.js
@@ -1,12 +1,12 @@
-function benchmark(func, loopCount) {
+module.exports = function benchmark(func, loopCount) {
loopCount = loopCount || 10000;
var start = Date.now();
- for (var i = 0; i < loopCount; i++) {
+ for (var i = 0; i < loopCount; i++) {
func();
}
var end = Date.now();
return end - start;
-}
+};
diff --git a/bench/deinterleave.js b/bench/deinterleave.js
index 930d399..ff4abf1 100644
--- a/bench/deinterleave.js
+++ b/bench/deinterleave.js
@@ -1,32 +1,37 @@
+/* global Float32Array */
+/* eslint-disable no-console */
+var benchmark = require("./bench");
+var dsp = require("..");
var bufferSize = 2048;
-var sampleRate = 44100;
+// var sampleRate = 44100;
var buffer1 = new Float32Array(bufferSize);
var buffer2 = new Float32Array(bufferSize);
var buffer3 = new Float32Array(bufferSize);
var buffer4 = new Float32Array(bufferSize);
-for (var i = 0; i < bufferSize; i++) {
+var i;
+for (i = 0; i < bufferSize; i++) {
buffer1[i] = (i % 2 === 0) ? -Math.random() : Math.random();
}
-for (var i = 0; i < bufferSize; i++) {
+for (i = 0; i < bufferSize; i++) {
buffer2[i] = (i % 2 === 0) ? -Math.random() : Math.random();
}
-for (var i = 0; i < bufferSize; i++) {
+for (i = 0; i < bufferSize; i++) {
buffer3[i] = (i % 2 === 0) ? -Math.random() : Math.random();
}
-for (var i = 0; i < bufferSize; i++) {
+for (i = 0; i < bufferSize; i++) {
buffer4[i] = (i % 2 === 0) ? -Math.random() : Math.random();
}
var channel;
var temp;
-var duration = benchmark(function() {
- channel = DSP.deinterleave(DSP.MIX, buffer1);
+var duration = benchmark(function() {
+ channel = dsp.DSP.deinterleave(dsp.DSP.MIX, buffer1);
// cycle buffers
temp = buffer1;
@@ -36,5 +41,5 @@ var duration = benchmark(function() {
buffer4 = temp;
}, 100000);
-print("Channel length: " + channel.length);
-print("100000 iterations: " + (duration) + " ms (" + ((duration) / 100000) + "ms per iter)\n");
+console.log("Channel length: " + channel.length);
+console.log("100000 iterations: " + (duration) + " ms (" + ((duration) / 100000) + "ms per iter)\n");
diff --git a/bench/dft.js b/bench/dft.js
index 40ac61e..a2c0c73 100644
--- a/bench/dft.js
+++ b/bench/dft.js
@@ -1,9 +1,13 @@
+/* global FREQUENCY */
+/* eslint-disable no-console */
+var benchmark = require("./bench");
+var dsp = require("..");
var bufferSize = 2048;
var sampleRate = 44100;
var frequency = FREQUENCY || 440;
-var dft = new DFT(bufferSize, sampleRate);
-var osc = new Oscillator(DSP.SAW, frequency, 1.0, bufferSize, sampleRate);
+var dft = new dsp.DFT(bufferSize, sampleRate);
+var osc = new dsp.Oscillator(dsp.DSP.SAW, frequency, 1.0, bufferSize, sampleRate);
var signal = osc.generate();
var duration = benchmark(function() { dft.forward(signal); }, 20);
@@ -16,5 +20,5 @@ for (var i = 0; i < dft.spectrum.length; i++) {
var peakFreq = dft.getBandFrequency(dft.peakBand);
-print("Detected peak: " + peakFreq + " Hz (error " + Math.abs(peakFreq - frequency) + " Hz)");
-print("20 DFTs: " + (duration) + " ms (" + ((duration) / 20) + "ms per DFT)\n");
+console.log("Detected peak: " + peakFreq + " Hz (error " + Math.abs(peakFreq - frequency) + " Hz)");
+console.log("20 DFTs: " + (duration) + " ms (" + ((duration) / 20) + "ms per DFT)\n");
diff --git a/bench/fft.js b/bench/fft.js
index 2a0fe58..95a5fe3 100644
--- a/bench/fft.js
+++ b/bench/fft.js
@@ -1,9 +1,13 @@
+/* global FREQUENCY */
+/* eslint-disable no-console */
+var benchmark = require("./bench");
+var dsp = require("..");
var bufferSize = 2048;
var sampleRate = 44100;
var frequency = FREQUENCY || 440;
-var fft = new FFT(bufferSize, sampleRate);
-var osc = new Oscillator(DSP.SAW, frequency, 1.0, bufferSize, sampleRate);
+var fft = new dsp.FFT(bufferSize, sampleRate);
+var osc = new dsp.Oscillator(dsp.DSP.SAW, frequency, 1.0, bufferSize, sampleRate);
var signal = osc.generate();
var duration = benchmark(function() { fft.forward(signal); });
@@ -16,5 +20,5 @@ for (var i = 0; i < fft.spectrum.length; i++) {
var peakFreq = fft.getBandFrequency(fft.peakBand);
-print("Detected peak: " + peakFreq + " Hz (error " + Math.abs(peakFreq - frequency) + " Hz)");
-print("10000 FFTs: " + (duration) + " ms (" + ((duration) / 10000) + "ms per FFT)\n");
+console.log("Detected peak: " + peakFreq + " Hz (error " + Math.abs(peakFreq - frequency) + " Hz)");
+console.log("10000 FFTs: " + (duration) + " ms (" + ((duration) / 10000) + "ms per FFT)\n");
diff --git a/bench/index.js b/bench/index.js
new file mode 100644
index 0000000..ca506a5
--- /dev/null
+++ b/bench/index.js
@@ -0,0 +1,6 @@
+/* eslint-disable */
+FREQUENCY = 440;
+require("./deinterleave");
+require("./dft");
+require("./fft");
+require("./rfft");
diff --git a/bench/rfft.js b/bench/rfft.js
index bc428d4..b70ad3d 100644
--- a/bench/rfft.js
+++ b/bench/rfft.js
@@ -1,9 +1,13 @@
+/* global FREQUENCY */
+/* eslint-disable no-console */
+var benchmark = require("./bench");
+var dsp = require("..");
var bufferSize = 2048;
var sampleRate = 44100;
var frequency = FREQUENCY || 440;
-var fft = new RFFT(bufferSize, sampleRate);
-var osc = new Oscillator(DSP.SAW, frequency, 1.0, bufferSize, sampleRate);
+var fft = new dsp.RFFT(bufferSize, sampleRate);
+var osc = new dsp.Oscillator(dsp.DSP.SAW, frequency, 1.0, bufferSize, sampleRate);
var signal = osc.generate();
var duration = benchmark(function() { fft.forward(signal); });
@@ -16,5 +20,5 @@ for (var i = 0; i < fft.spectrum.length; i++) {
var peakFreq = fft.getBandFrequency(fft.peakBand);
-print("Detected peak: " + peakFreq + " Hz (error " + Math.abs(peakFreq - frequency) + " Hz)");
-print("10000 FFTs: " + (duration) + " ms (" + ((duration) / 10000) + "ms per FFT)\n");
+console.log("Detected peak: " + peakFreq + " Hz (error " + Math.abs(peakFreq - frequency) + " Hz)");
+console.log("10000 FFTs: " + (duration) + " ms (" + ((duration) / 10000) + "ms per FFT)\n");
diff --git a/dsp.js b/dist/dsp.js
similarity index 85%
rename from dsp.js
rename to dist/dsp.js
index 7188235..2fb4a3d 100644
--- a/dsp.js
+++ b/dist/dsp.js
@@ -1,950 +1,1536 @@
-/*
- * DSP.js - a comprehensive digital signal processing library for javascript
- *
- * Created by Corban Brook <corbanbrook@gmail.com> on 2010-01-01.
- * Copyright 2010 Corban Brook. All rights reserved.
+(function e(t,n,r){function s(o,u){if(!n[o]){if(!t[o]){var a=typeof require=="function"&&require;if(!u&&a)return a(o,!0);if(i)return i(o,!0);var f=new Error("Cannot find module '"+o+"'");throw f.code="MODULE_NOT_FOUND",f}var l=n[o]={exports:{}};t[o][0].call(l.exports,function(e){var n=t[o][1][e];return s(n?n:e)},l,l.exports,e,t,n,r)}return n[o].exports}var i=typeof require=="function"&&require;for(var o=0;o<r.length;o++)s(r[o]);return s})({1:[function(require,module,exports){
+/**
+ * ADSR Envelope
*
+ * @param {Number} attack The attack length in seconds
+ * @param {Number} decay The decay length in seconds
+ * @param {Number} sustain The sustain level
+ * @param {Number} release The release length in seconds
+ * @param {Number} sampleRate The the sample rate
+ * @constructor
*/
+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;
-////////////////////////////////////////////////////////////////////////////////
-// CONSTANTS //
-////////////////////////////////////////////////////////////////////////////////
-
-/**
- * DSP is an object which contains general purpose utility functions and constants
- */
-var DSP = {
- // Channels
- LEFT: 0,
- RIGHT: 1,
- MIX: 2,
+ // Length in samples
+ this.attackSamples = attackLength * sampleRate;
+ this.decaySamples = decayLength * sampleRate;
+ this.sustainSamples = sustainLength * sampleRate;
+ this.releaseSamples = releaseLength * sampleRate;
- // Waveforms
- SINE: 1,
- TRIANGLE: 2,
- SAW: 3,
- SQUARE: 4,
+ // 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;
+ };
- // Filters
- LOWPASS: 0,
- HIGHPASS: 1,
- BANDPASS: 2,
- NOTCH: 3,
+ this.update();
- // Window functions
- BARTLETT: 1,
- BARTLETTHANN: 2,
- BLACKMAN: 3,
- COSINE: 4,
- GAUSS: 5,
- HAMMING: 6,
- HANN: 7,
- LANCZOS: 8,
- RECTANGULAR: 9,
- TRIANGULAR: 10,
+ this.samplesProcessed = 0;
+}
- // Loop modes
- OFF: 0,
- FW: 1,
- BW: 2,
- FWBW: 3,
+/**
+ * Start the envelope
+ */
+ADSR.prototype.noteOn = function() {
+ this.samplesProcessed = 0;
+ this.sustainSamples = this.sustainLength * this.sampleRate;
+ this.update();
+};
- // Math
- TWO_PI: 2*Math.PI
+/**
+ * Stop the envelope
+ *
+ * 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();
};
-// Setup arrays for platforms which do not support byte arrays
-function setupTypedArray(name, fallback) {
- // check if TypedArray exists
- // typeof on Minefield and Chrome return function, typeof on Webkit returns object.
- if (typeof this[name] !== "function" && typeof this[name] !== "object") {
- // nope.. check if WebGLArray exists
- if (typeof this[fallback] === "function" && typeof this[fallback] !== "object") {
- this[name] = this[fallback];
- } else {
- // nope.. set as Native JS array
- this[name] = function(obj) {
- if (obj instanceof Array) {
- return obj;
- } else if (typeof obj === "number") {
- return new Array(obj);
- }
- };
- }
+/**
+ * Process sample
+ */
+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));
}
-}
-setupTypedArray("Float64Array", "WebGLFloatArray");
-setupTypedArray("Int32Array", "WebGLIntArray");
-setupTypedArray("Uint16Array", "WebGLUnsignedShortArray");
-setupTypedArray("Uint8Array", "WebGLUnsignedByteArray");
+ return sample * amplitude;
+};
+/**
+ * Get current value
+ * @return {Number} amplitude
+ */
+ADSR.prototype.value = function() {
+ var amplitude = 0;
-////////////////////////////////////////////////////////////////////////////////
-// DSP UTILITY FUNCTIONS //
-////////////////////////////////////////////////////////////////////////////////
+ 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;
+};
/**
- * Inverts the phase of a signal
- *
- * @param {Array} buffer A sample buffer
- *
- * @returns The inverted sample buffer
+ * Process a buffer
+ * @param {Array} buffer
*/
-DSP.invert = function(buffer) {
- for (var i = 0, len = buffer.length; i < len; i++) {
- buffer[i] *= -1;
+ADSR.prototype.process = function(buffer) {
+ for ( var i = 0; i < buffer.length; i++ ) {
+ buffer[i] *= this.value();
+
+ this.samplesProcessed++;
}
return buffer;
};
/**
- * Converts split-stereo (dual mono) sample buffers into a stereo interleaved sample buffer
- *
- * @param {Array} left A sample buffer
- * @param {Array} right A sample buffer
- *
- * @returns The stereo interleaved buffer
+ * Test if the envelope is active
+ * @return {Boolean}
*/
-DSP.interleave = function(left, right) {
- if (left.length !== right.length) {
- throw "Can not interleave. Channel lengths differ.";
- }
-
- var stereoInterleaved = new Float64Array(left.length * 2);
-
- for (var i = 0, len = left.length; i < len; i++) {
- stereoInterleaved[2*i] = left[i];
- stereoInterleaved[2*i+1] = right[i];
+ADSR.prototype.isActive = function() {
+ if ( this.samplesProcessed > this.release || this.samplesProcessed === -1 ) {
+ return false;
+ } else {
+ return true;
}
-
- return stereoInterleaved;
};
/**
- * Converts a stereo-interleaved sample buffer into split-stereo (dual mono) sample buffers
- *
- * @param {Array} buffer A stereo-interleaved sample buffer
- *
- * @returns an Array containing left and right channels
+ * Disable the envelope
*/
-DSP.deinterleave = (function() {
- var left, right, mix, deinterleaveChannel = [];
-
- deinterleaveChannel[DSP.MIX] = function(buffer) {
- for (var i = 0, len = buffer.length/2; i < len; i++) {
- mix[i] = (buffer[2*i] + buffer[2*i+1]) / 2;
- }
- return mix;
- };
-
- deinterleaveChannel[DSP.LEFT] = function(buffer) {
- for (var i = 0, len = buffer.length/2; i < len; i++) {
- left[i] = buffer[2*i];
- }
- return left;
- };
-
- deinterleaveChannel[DSP.RIGHT] = function(buffer) {
- for (var i = 0, len = buffer.length/2; i < len; i++) {
- right[i] = buffer[2*i+1];
- }
- return right;
- };
-
- return function(channel, buffer) {
- left = left || new Float64Array(buffer.length/2);
- right = right || new Float64Array(buffer.length/2);
- mix = mix || new Float64Array(buffer.length/2);
+ADSR.prototype.disable = function() {
+ this.samplesProcessed = -1;
+};
- if (buffer.length/2 !== left.length) {
- left = new Float64Array(buffer.length/2);
- right = new Float64Array(buffer.length/2);
- mix = new Float64Array(buffer.length/2);
- }
+module.exports = ADSR;
- return deinterleaveChannel[channel](buffer);
- };
-}());
+},{}],2:[function(require,module,exports){
+/* global Float64Array */
+var DSP = require("./dsp");
+var sinh = require("./sinh");
/**
- * Separates a channel from a stereo-interleaved sample buffer
- *
- * @param {Array} buffer A stereo-interleaved sample buffer
- * @param {Number} channel A channel constant (LEFT, RIGHT, MIX)
+ * Biquad filter
*
- * @returns an Array containing a signal mono sample buffer
- */
-DSP.getChannel = DSP.deinterleave;
-
-/**
- * Helper method (for Reverb) to mix two (interleaved) samplebuffers. It's possible
- * to negate the second buffer while mixing and to perform a volume correction
- * on the final signal.
+ * Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
+ * Copyright 2010 Ricard Marxer. All rights reserved.
*
- * @param {Array} sampleBuffer1 Array containing Float values or a Float64Array
- * @param {Array} sampleBuffer2 Array containing Float values or a Float64Array
- * @param {Boolean} negate When true inverts/flips the audio signal
- * @param {Number} volumeCorrection When you add multiple sample buffers, use this to tame your signal ;)
+ * Implementation based on:
+ * http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
*
- * @returns A new Float64Array interleaved buffer.
+ * @param {Number} type
+ * @param {Number} sampleRate
+ * @constructor
*/
-DSP.mixSampleBuffers = function(sampleBuffer1, sampleBuffer2, negate, volumeCorrection){
- var outputSamples = new Float64Array(sampleBuffer1);
+function Biquad(type, sampleRate) {
+ this.Fs = sampleRate;
+ this.type = type; // type of the filter
+ this.parameterType = DSP.Q; // type of the parameter
- for(var i = 0; i<sampleBuffer1.length; i++){
- outputSamples[i] += (negate ? -sampleBuffer2[i] : sampleBuffer2[i]) / volumeCorrection;
- }
-
- return outputSamples;
-};
+ this.x_1_l = 0;
+ this.x_2_l = 0;
+ this.y_1_l = 0;
+ this.y_2_l = 0;
-// Biquad filter types
-DSP.LPF = 0; // H(s) = 1 / (s^2 + s/Q + 1)
-DSP.HPF = 1; // H(s) = s^2 / (s^2 + s/Q + 1)
-DSP.BPF_CONSTANT_SKIRT = 2; // H(s) = s / (s^2 + s/Q + 1) (constant skirt gain, peak gain = Q)
-DSP.BPF_CONSTANT_PEAK = 3; // H(s) = (s/Q) / (s^2 + s/Q + 1) (constant 0 dB peak gain)
-DSP.NOTCH = 4; // H(s) = (s^2 + 1) / (s^2 + s/Q + 1)
-DSP.APF = 5; // H(s) = (s^2 - s/Q + 1) / (s^2 + s/Q + 1)
-DSP.PEAKING_EQ = 6; // H(s) = (s^2 + s*(A/Q) + 1) / (s^2 + s/(A*Q) + 1)
-DSP.LOW_SHELF = 7; // H(s) = A * (s^2 + (sqrt(A)/Q)*s + A)/(A*s^2 + (sqrt(A)/Q)*s + 1)
-DSP.HIGH_SHELF = 8; // H(s) = A * (A*s^2 + (sqrt(A)/Q)*s + 1)/(s^2 + (sqrt(A)/Q)*s + A)
+ this.x_1_r = 0;
+ this.x_2_r = 0;
+ this.y_1_r = 0;
+ this.y_2_r = 0;
-// Biquad filter parameter types
-DSP.Q = 1;
-DSP.BW = 2; // SHARED with BACKWARDS LOOP MODE
-DSP.S = 3;
+ this.b0 = 1;
+ this.a0 = 1;
-// Find RMS of signal
-DSP.RMS = function(buffer) {
- var total = 0;
-
- for (var i = 0, n = buffer.length; i < n; i++) {
- total += buffer[i] * buffer[i];
- }
-
- return Math.sqrt(total / n);
-};
+ this.b1 = 0;
+ this.a1 = 0;
-// Find Peak of signal
-DSP.Peak = function(buffer) {
- var peak = 0;
-
- for (var i = 0, n = buffer.length; i < n; i++) {
- peak = (Math.abs(buffer[i]) > peak) ? Math.abs(buffer[i]) : peak;
- }
-
- return peak;
-};
+ this.b2 = 0;
+ this.a2 = 0;
-// Fourier Transform Module used by DFT, FFT, RFFT
-function FourierTransform(bufferSize, sampleRate) {
- this.bufferSize = bufferSize;
- this.sampleRate = sampleRate;
- this.bandwidth = 2 / bufferSize * sampleRate / 2;
+ 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.spectrum = new Float64Array(bufferSize/2);
- this.real = new Float64Array(bufferSize);
- this.imag = new Float64Array(bufferSize);
+ 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.peakBand = 0;
- this.peak = 0;
+ this.dBgain = 12; // used only for peaking and shelving filters
- /**
- * Calculates the *middle* frequency of an FFT band.
- *
- * @param {Number} index The index of the FFT band.
- *
- * @returns The middle frequency in Hz.
- */
- this.getBandFrequency = function(index) {
- return this.bandwidth * index + this.bandwidth / 2;
- };
+ 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.calculateSpectrum = function() {
- var spectrum = this.spectrum,
- real = this.real,
- imag = this.imag,
- bSi = 2 / this.bufferSize,
- sqrt = Math.sqrt,
- rval,
- ival,
- mag;
+ 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
- for (var i = 0, N = bufferSize/2; i < N; i++) {
- rval = real[i];
- ival = imag[i];
- mag = bSi * sqrt(rval * rval + ival * ival);
+ 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.
- if (mag > this.peak) {
- this.peakBand = i;
- this.peak = mag;
- }
+ this.coefficients = function() {
+ var b = [this.b0, this.b1, this.b2];
+ var a = [this.a0, this.a1, this.a2];
+ return {b: b, a:a};
+ };
- spectrum[i] = mag;
- }
+ this.setFilterType = function(type) {
+ this.type = type;
+ this.recalculateCoefficients();
};
-}
-/**
- * DFT is a class for calculating the Discrete Fourier Transform of a signal.
- *
- * @param {Number} bufferSize The size of the sample buffer to be computed
- * @param {Number} sampleRate The sampleRate of the buffer (eg. 44100)
- *
- * @constructor
- */
-function DFT(bufferSize, sampleRate) {
- FourierTransform.call(this, bufferSize, sampleRate);
+ this.setSampleRate = function(rate) {
+ this.Fs = rate;
+ this.recalculateCoefficients();
+ };
- var N = bufferSize/2 * bufferSize;
- var TWO_PI = 2 * Math.PI;
+ this.setQ = function(q) {
+ this.parameterType = DSP.Q;
+ this.Q = Math.max(Math.min(q, 115.0), 0.001);
+ this.recalculateCoefficients();
+ };
- this.sinTable = new Float64Array(N);
- this.cosTable = new Float64Array(N);
+ this.setBW = function(bw) {
+ this.parameterType = DSP.BW;
+ this.BW = bw;
+ this.recalculateCoefficients();
+ };
- for (var i = 0; i < N; i++) {
- this.sinTable[i] = Math.sin(i * TWO_PI / bufferSize);
- this.cosTable[i] = Math.cos(i * TWO_PI / bufferSize);
- }
-}
+ this.setS = function(s) {
+ this.parameterType = DSP.S;
+ this.S = Math.max(Math.min(s, 5.0), 0.0001);
+ this.recalculateCoefficients();
+ };
-/**
- * Performs a forward transform on the sample buffer.
- * Converts a time domain signal to frequency domain spectra.
- *
- * @param {Array} buffer The sample buffer
- *
- * @returns The frequency spectrum array
- */
-DFT.prototype.forward = function(buffer) {
- var real = this.real,
- imag = this.imag,
- rval,
- ival;
+ this.setF0 = function(freq) {
+ this.f0 = freq;
+ this.recalculateCoefficients();
+ };
- for (var k = 0; k < this.bufferSize/2; k++) {
- rval = 0.0;
- ival = 0.0;
+ this.setDbGain = function(g) {
+ this.dBgain = g;
+ this.recalculateCoefficients();
+ };
- for (var n = 0; n < buffer.length; n++) {
- rval += this.cosTable[k*n] * buffer[n];
- ival += this.sinTable[k*n] * buffer[n];
+ 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)) );
}
- real[k] = rval;
- imag[k] = ival;
- }
-
- return this.calculateSpectrum();
-};
+ var w0 = DSP.TWO_PI * this.f0 / this.Fs;
+ var cosw0 = Math.cos(w0);
+ var sinw0 = Math.sin(w0);
-/**
- * FFT is a class for calculating the Discrete Fourier Transform of a signal
- * with the Fast Fourier Transform algorithm.
- *
- * @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
- */
-function FFT(bufferSize, sampleRate) {
- FourierTransform.call(this, bufferSize, sampleRate);
-
- this.reverseTable = new Uint32Array(bufferSize);
+ var alpha = 0;
- var limit = 1;
- var bit = bufferSize >> 1;
+ switch (this.parameterType) {
+ case DSP.Q:
+ alpha = sinw0/(2*this.Q);
+ break;
- var i;
+ case DSP.BW:
+ alpha = sinw0 * sinh( Math.LN2/2 * this.BW * w0/sinw0 );
+ break;
- while (limit < bufferSize) {
- for (i = 0; i < limit; i++) {
- this.reverseTable[i + limit] = this.reverseTable[i] + bit;
+ case DSP.S:
+ alpha = sinw0/2 * Math.sqrt( (A + 1/A)*(1/this.S - 1) + 2 );
+ break;
}
- limit = limit << 1;
- bit = bit >> 1;
- }
+ /**
+ 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)
- this.sinTable = new Float64Array(bufferSize);
- this.cosTable = new Float64Array(bufferSize);
+ The relationship between shelf slope and Q is
+ 1/Q = sqrt((A + 1/A)*(1/S - 1) + 2)
+ */
- for (i = 0; i < bufferSize; i++) {
- this.sinTable[i] = Math.sin(-Math.PI/i);
- this.cosTable[i] = Math.cos(-Math.PI/i);
- }
-}
+ var coeff;
-/**
- * Performs a forward transform on the sample buffer.
- * Converts a time domain signal to frequency domain spectra.
- *
- * @param {Array} buffer The sample buffer. Buffer Length must be power of 2
- *
- * @returns The frequency spectrum array
- */
-FFT.prototype.forward = function(buffer) {
- // Locally scope variables for speed up
- var bufferSize = this.bufferSize,
- cosTable = this.cosTable,
- sinTable = this.sinTable,
- reverseTable = this.reverseTable,
- real = this.real,
- imag = this.imag,
- spectrum = this.spectrum;
+ 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;
- var k = Math.floor(Math.log(bufferSize) / Math.LN2);
+ 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;
- 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; }
+ 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;
- 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;
+ 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;
- for (var fftStep = 0; fftStep < halfSize; fftStep++) {
- i = fftStep;
+ 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;
- while (i < bufferSize) {
- off = i + halfSize;
- tr = (currentPhaseShiftReal * real[off]) - (currentPhaseShiftImag * imag[off]);
- ti = (currentPhaseShiftReal * imag[off]) + (currentPhaseShiftImag * real[off]);
+ 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;
- real[off] = real[i] - tr;
- imag[off] = imag[i] - ti;
- real[i] += tr;
- imag[i] += ti;
+ 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;
- i += halfSize << 1;
- }
+ 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;
- tmpReal = currentPhaseShiftReal;
- currentPhaseShiftReal = (tmpReal * phaseShiftStepReal) - (currentPhaseShiftImag * phaseShiftStepImag);
- currentPhaseShiftImag = (tmpReal * phaseShiftStepImag) + (currentPhaseShiftImag * phaseShiftStepReal);
+ 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;
}
- halfSize = halfSize << 1;
- }
-
- return this.calculateSpectrum();
-};
-
-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;
+ 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;
+ };
- var halfSize = 1,
- phaseShiftStepReal,
- phaseShiftStepImag,
- currentPhaseShiftReal,
- currentPhaseShiftImag,
- off,
- tr,
- ti,
- tmpReal,
- i;
+ 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]
- for (i = 0; i < bufferSize; i++) {
- imag[i] *= -1;
- }
+ var len = buffer.length;
+ var output = new Float64Array(len);
- 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;
+ 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];
+ }
- while (halfSize < bufferSize) {
- phaseShiftStepReal = cosTable[halfSize];
- phaseShiftStepImag = sinTable[halfSize];
- currentPhaseShiftReal = 1;
- currentPhaseShiftImag = 0;
+ return output;
+ };
- for (var fftStep = 0; fftStep < halfSize; fftStep++) {
- i = fftStep;
+ 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]
- while (i < bufferSize) {
- off = i + halfSize;
- tr = (currentPhaseShiftReal * real[off]) - (currentPhaseShiftImag * imag[off]);
- ti = (currentPhaseShiftReal * imag[off]) + (currentPhaseShiftImag * real[off]);
+ var len = buffer.length;
+ var output = new Float64Array(len);
- real[off] = real[i] - tr;
- imag[off] = imag[i] - ti;
- real[i] += tr;
- imag[i] += ti;
+ 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];
- i += halfSize << 1;
+ 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];
}
- tmpReal = currentPhaseShiftReal;
- currentPhaseShiftReal = (tmpReal * phaseShiftStepReal) - (currentPhaseShiftImag * phaseShiftStepImag);
- currentPhaseShiftImag = (tmpReal * phaseShiftStepImag) + (currentPhaseShiftImag * phaseShiftStepReal);
- }
-
- halfSize = halfSize << 1;
- }
+ return output;
+ };
+}
- var buffer = new Float64Array(bufferSize); // this should be reused instead
- for (i = 0; i < bufferSize; i++) {
- buffer[i] = real[i] / bufferSize;
- }
+module.exports = Biquad;
- return buffer;
-};
+},{"./dsp":4,"./sinh":17}],3:[function(require,module,exports){
+/* global Float64Array */
+var FourierTransform = require("./fourier");
/**
- * 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.
+ * DFT is a class for calculating the Discrete Fourier Transform of a signal.
*
- * @param {Number} bufferSize The size of the sample buffer to be computed. Must be power of 2
+ * @param {Number} bufferSize The size of the sample buffer to be computed
* @param {Number} sampleRate The sampleRate of the buffer (eg. 44100)
*
* @constructor
*/
+function DFT(bufferSize, sampleRate) {
+ FourierTransform.call(this, bufferSize, sampleRate);
-// 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
+ var N = bufferSize/2 * bufferSize;
+ var TWO_PI = 2 * Math.PI;
-// this one a little more than twice as fast as the one in FFT
-// however I only did the forward transform
+ this.sinTable = new Float64Array(N);
+ this.cosTable = new Float64Array(N);
-// the rest of this was translated from C, see http://www.jjj.de/fxt/
-// this is the real split radix FFT
+ for (var i = 0; i < N; i++) {
+ this.sinTable[i] = Math.sin(i * TWO_PI / bufferSize);
+ this.cosTable[i] = Math.cos(i * TWO_PI / bufferSize);
+ }
+}
-function RFFT(bufferSize, sampleRate) {
- FourierTransform.call(this, bufferSize, sampleRate);
+/**
+ * Performs a forward transform on the sample buffer.
+ * Converts a time domain signal to frequency domain spectra.
+ *
+ * @param {Array} buffer The sample buffer
+ *
+ * @returns The frequency spectrum array
+ */
+DFT.prototype.forward = function(buffer) {
+ var real = this.real,
+ imag = this.imag,
+ rval,
+ ival;
- this.trans = new Float64Array(bufferSize);
+ for (var k = 0; k < this.bufferSize/2; k++) {
+ rval = 0.0;
+ ival = 0.0;
- this.reverseTable = new Uint32Array(bufferSize);
+ for (var n = 0; n < buffer.length; n++) {
+ rval += this.cosTable[k*n] * buffer[n];
+ ival += this.sinTable[k*n] * buffer[n];
+ }
- // 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;
+ real[k] = rval;
+ imag[k] = ival;
+ }
- dest[0] = source[0];
+ return this.calculateSpectrum();
+};
- do {
- r += halfSize;
- dest[i] = source[r];
- dest[r] = source[i];
-
- i++;
+module.exports = DFT;
- h = halfSize << 1;
- while (h = h >> 1, !((r ^= h) & h));
+},{"./fourier":6}],4:[function(require,module,exports){
+/* global Float64Array */
- if (r >= i) {
- dest[i] = source[r];
- dest[r] = source[i];
+////////////////////////////////////////////////////////////////////////////////
+// CONSTANTS //
+////////////////////////////////////////////////////////////////////////////////
- dest[nm1-i] = source[nm1-r];
- dest[nm1-r] = source[nm1-i];
- }
- i++;
- } while (i < halfSize);
- dest[nm1] = source[nm1];
- };
+/**
+ * DSP is an object which contains general purpose utility functions and constants
+ */
+var DSP = {
+ // Channels
+ LEFT: 0,
+ RIGHT: 1,
+ MIX: 2,
- this.generateReverseTable = function () {
- var bufferSize = this.bufferSize,
- halfSize = bufferSize >>> 1,
- nm1 = bufferSize - 1,
- i = 1, r = 0, h;
+ // Waveforms
+ SINE: 1,
+ TRIANGLE: 2,
+ SAW: 3,
+ SQUARE: 4,
- this.reverseTable[0] = 0;
+ // Filters
+ LOWPASS: 0,
+ HIGHPASS: 1,
+ BANDPASS: 2,
+ NOTCH: 3,
- do {
- r += halfSize;
-
- this.reverseTable[i] = r;
- this.reverseTable[r] = i;
+ // Window functions
+ BARTLETT: 1,
+ BARTLETTHANN: 2,
+ BLACKMAN: 3,
+ COSINE: 4,
+ GAUSS: 5,
+ HAMMING: 6,
+ HANN: 7,
+ LANCZOS: 8,
+ RECTANGULAR: 9,
+ TRIANGULAR: 10,
- i++;
+ // Loop modes
+ OFF: 0,
+ FW: 1,
+ BW: 2,
+ FWBW: 3,
- h = halfSize << 1;
- while (h = h >> 1, !((r ^= h) & h));
+ // Math
+ TWO_PI: 2*Math.PI
+};
- if (r >= i) {
- this.reverseTable[i] = r;
- this.reverseTable[r] = i;
+// Setup arrays for platforms which do not support byte arrays
+function setupTypedArray(name, fallback) {
+ // check if TypedArray exists
+ // typeof on Minefield and Chrome return function, typeof on Webkit returns object.
+ if (typeof this[name] !== "function" && typeof this[name] !== "object") {
+ // nope.. check if WebGLArray exists
+ if (typeof this[fallback] === "function" && typeof this[fallback] !== "object") {
+ this[name] = this[fallback];
+ } else {
+ // nope.. set as Native JS array
+ this[name] = function(obj) {
+ if (obj instanceof Array) {
+ return obj;
+ } else if (typeof obj === "number") {
+ return new Array(obj);
+ }
+ };
+ }
+ }
+}
- this.reverseTable[nm1-i] = nm1-r;
- this.reverseTable[nm1-r] = nm1-i;
- }
- i++;
- } while (i < halfSize);
+setupTypedArray("Float64Array", "WebGLFloatArray");
+setupTypedArray("Int32Array", "WebGLIntArray");
+setupTypedArray("Uint16Array", "WebGLUnsignedShortArray");
+setupTypedArray("Uint8Array", "WebGLUnsignedByteArray");
- this.reverseTable[nm1] = nm1;
+
+////////////////////////////////////////////////////////////////////////////////
+// DSP UTILITY FUNCTIONS //
+////////////////////////////////////////////////////////////////////////////////
+
+/**
+ * Inverts the phase of a signal
+ *
+ * @param {Array} buffer A sample buffer
+ *
+ * @returns The inverted sample buffer
+ */
+DSP.invert = function(buffer) {
+ for (var i = 0, len = buffer.length; i < len; i++) {
+ buffer[i] *= -1;
+ }
+
+ return buffer;
+};
+
+/**
+ * Converts split-stereo (dual mono) sample buffers into a stereo interleaved sample buffer
+ *
+ * @param {Array} left A sample buffer
+ * @param {Array} right A sample buffer
+ *
+ * @returns The stereo interleaved buffer
+ */
+DSP.interleave = function(left, right) {
+ if (left.length !== right.length) {
+ throw "Can not interleave. Channel lengths differ.";
+ }
+
+ var stereoInterleaved = new Float64Array(left.length * 2);
+
+ for (var i = 0, len = left.length; i < len; i++) {
+ stereoInterleaved[2*i] = left[i];
+ stereoInterleaved[2*i+1] = right[i];
+ }
+
+ return stereoInterleaved;
+};
+
+/**
+ * Converts a stereo-interleaved sample buffer into split-stereo (dual mono) sample buffers
+ *
+ * @param {Array} buffer A stereo-interleaved sample buffer
+ *
+ * @returns an Array containing left and right channels
+ */
+DSP.deinterleave = (function() {
+ var left, right, mix, deinterleaveChannel = [];
+
+ deinterleaveChannel[DSP.MIX] = function(buffer) {
+ for (var i = 0, len = buffer.length/2; i < len; i++) {
+ mix[i] = (buffer[2*i] + buffer[2*i+1]) / 2;
+ }
+ return mix;
};
- this.generateReverseTable();
-}
+ deinterleaveChannel[DSP.LEFT] = function(buffer) {
+ for (var i = 0, len = buffer.length/2; i < len; i++) {
+ left[i] = buffer[2*i];
+ }
+ return left;
+ };
+ deinterleaveChannel[DSP.RIGHT] = function(buffer) {
+ for (var i = 0, len = buffer.length/2; i < len; i++) {
+ right[i] = buffer[2*i+1];
+ }
+ return right;
+ };
-// 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]
+ return function(channel, buffer) {
+ left = left || new Float64Array(buffer.length/2);
+ right = right || new Float64Array(buffer.length/2);
+ mix = mix || new Float64Array(buffer.length/2);
-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;
+ if (buffer.length/2 !== left.length) {
+ left = new Float64Array(buffer.length/2);
+ right = new Float64Array(buffer.length/2);
+ mix = new Float64Array(buffer.length/2);
+ }
- this.reverseBinPermute(x, buffer);
+ return deinterleaveChannel[channel](buffer);
+ };
+}());
- /*
- var reverseTable = this.reverseTable;
+/**
+ * Separates a channel from a stereo-interleaved sample buffer
+ *
+ * @param {Array} buffer A stereo-interleaved sample buffer
+ * @param {Number} channel A channel constant (LEFT, RIGHT, MIX)
+ *
+ * @returns an Array containing a signal mono sample buffer
+ */
+DSP.getChannel = DSP.deinterleave;
- for (var k = 0, len = reverseTable.length; k < len; k++) {
- x[k] = buffer[reverseTable[k]];
+/**
+ * Helper method (for Reverb) to mix two (interleaved) samplebuffers. It's possible
+ * to negate the second buffer while mixing and to perform a volume correction
+ * on the final signal.
+ *
+ * @param {Array} sampleBuffer1 Array containing Float values or a Float64Array
+ * @param {Array} sampleBuffer2 Array containing Float values or a Float64Array
+ * @param {Boolean} negate When true inverts/flips the audio signal
+ * @param {Number} volumeCorrection When you add multiple sample buffers, use this to tame your signal ;)
+ *
+ * @returns A new Float64Array interleaved buffer.
+ */
+DSP.mixSampleBuffers = function(sampleBuffer1, sampleBuffer2, negate, volumeCorrection){
+ var outputSamples = new Float64Array(sampleBuffer1);
+
+ for(var i = 0; i<sampleBuffer1.length; i++){
+ outputSamples[i] += (negate ? -sampleBuffer2[i] : sampleBuffer2[i]) / volumeCorrection;
}
- */
- 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);
+ return outputSamples;
+};
+
+// Biquad filter types
+DSP.LPF = 0; // H(s) = 1 / (s^2 + s/Q + 1)
+DSP.HPF = 1; // H(s) = s^2 / (s^2 + s/Q + 1)
+DSP.BPF_CONSTANT_SKIRT = 2; // H(s) = s / (s^2 + s/Q + 1) (constant skirt gain, peak gain = Q)
+DSP.BPF_CONSTANT_PEAK = 3; // H(s) = (s/Q) / (s^2 + s/Q + 1) (constant 0 dB peak gain)
+DSP.NOTCH = 4; // H(s) = (s^2 + 1) / (s^2 + s/Q + 1)
+DSP.APF = 5; // H(s) = (s^2 - s/Q + 1) / (s^2 + s/Q + 1)
+DSP.PEAKING_EQ = 6; // H(s) = (s^2 + s*(A/Q) + 1) / (s^2 + s/(A*Q) + 1)
+DSP.LOW_SHELF = 7; // H(s) = A * (s^2 + (sqrt(A)/Q)*s + A)/(A*s^2 + (sqrt(A)/Q)*s + 1)
+DSP.HIGH_SHELF = 8; // H(s) = A * (A*s^2 + (sqrt(A)/Q)*s + 1)/(s^2 + (sqrt(A)/Q)*s + A)
+
+// Biquad filter parameter types
+DSP.Q = 1;
+DSP.BW = 2; // SHARED with BACKWARDS LOOP MODE
+DSP.S = 3;
+
+/**
+ * Find RMS of signal
+ * @param {Array} buffer
+ */
+DSP.RMS = function(buffer) {
+ var total = 0;
+
+ for (var i = 0, n = buffer.length; i < n; i++) {
+ total += buffer[i] * buffer[i];
}
- n2 = 2;
- nn = n >>> 1;
+ return Math.sqrt(total / n);
+};
- 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;
+/**
+ * Find Peak of signal
+ * @param {Array} buffer
+ */
+DSP.Peak = function(buffer) {
+ var peak = 0;
- for (var j = 1; j < n8; j++) {
- a = j * e;
- ss1 = Math.sin(a);
- cc1 = Math.cos(a);
+ for (var i = 0, n = buffer.length; i < n; i++) {
+ peak = (Math.abs(buffer[i]) > peak) ? Math.abs(buffer[i]) : peak;
+ }
- //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);
+ return peak;
+};
+
+/**
+ * Magnitude to decibels
+ *
+ * Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
+ * Copyright 2010 Ricard Marxer. All rights reserved.
+ *
+ * @param {Array} @buffer The 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 = new 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.
+ *
+ * @param {Number} b The coefficients of the filter
+ * @param {Number} a The coefficients of the filter
+ * @param {Number} w The 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 = new Float64Array(200);
+ for (i=0;i<w.length; i++) {
+ w[i] = DSP.TWO_PI/w.length * i - Math.PI;
}
}
- while (--i) {
- rval = x[i];
- ival = x[n-i-1];
- mag = bSi * sqrt(rval * rval + ival * ival);
+ var result = new Float64Array(w.length);
- if (mag > this.peak) {
- this.peakBand = i;
- this.peak = mag;
+ 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]);
}
- spectrum[i] = mag;
- }
+ 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]);
+ }
- spectrum[0] = bSi * x[0];
+ result[i] = sqrt(numerator.real*numerator.real + numerator.imag*numerator.imag) / sqrt(denominator.real*denominator.real + denominator.imag*denominator.imag);
+ }
- return spectrum;
+ return result;
};
-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]);
+module.exports = DSP;
+
+},{}],5:[function(require,module,exports){
+/* global Float64Array Uint32Array */
+var FourierTransform = require("./fourier");
+
+/**
+ * FFT is a class for calculating the Discrete Fourier Transform of a signal
+ * with the Fast Fourier Transform algorithm.
+ *
+ * @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
+ */
+function FFT(bufferSize, sampleRate) {
+ FourierTransform.call(this, bufferSize, sampleRate);
+
+ this.reverseTable = new Uint32Array(bufferSize);
+
+ var limit = 1;
+ var bit = bufferSize >> 1;
+
+ var i;
+
+ while (limit < bufferSize) {
+ for (i = 0; i < limit; i++) {
+ this.reverseTable[i + limit] = this.reverseTable[i] + bit;
}
- };
-
- 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();
+
+ limit = limit << 1;
+ bit = bit >> 1;
+ }
+
+ this.sinTable = new Float64Array(bufferSize);
+ this.cosTable = new Float64Array(bufferSize);
+
+ for (i = 0; i < bufferSize; i++) {
+ this.sinTable[i] = Math.sin(-Math.PI/i);
+ this.cosTable[i] = Math.cos(-Math.PI/i);
+ }
}
-Sampler.prototype.applyEnvelope = function() {
- this.envelope.process(this.signal);
- return this.signal;
+/**
+ * Performs a forward transform on the sample buffer.
+ * Converts a time domain signal to frequency domain spectra.
+ *
+ * @param {Array} buffer The sample buffer. Buffer Length must be power of 2
+ *
+ * @returns The frequency spectrum array
+ */
+FFT.prototype.forward = function(buffer) {
+ // Locally scope variables for speed up
+ var bufferSize = this.bufferSize,
+ cosTable = this.cosTable,
+ sinTable = this.sinTable,
+ reverseTable = this.reverseTable,
+ real = this.real,
+ imag = this.imag;
+
+ 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 this.calculateSpectrum();
+};
+
+/**
+ * Performs a inverse FFT transformation
+ * Converts a frequency domain spectra to a time domain signal
+ *
+ * @param {Array} real
+ * @param {Array} imag
+ *
+ * @returns The time domain signal
+ */
+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;
+
+ 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;
+};
+
+module.exports = FFT;
+
+},{"./fourier":6}],6:[function(require,module,exports){
+/* global Float64Array */
+
+/**
+ * A Mixin for every fourier module
+ * Fourier Transform Module used by DFT, FFT, RFFT
+ * @private
+ */
+module.exports = function FourierTransform(bufferSize, sampleRate) {
+ this.bufferSize = bufferSize;
+ this.sampleRate = sampleRate;
+ this.bandwidth = 2 / bufferSize * sampleRate / 2;
+
+ this.spectrum = new Float64Array(bufferSize/2);
+ this.real = new Float64Array(bufferSize);
+ this.imag = new Float64Array(bufferSize);
+
+ this.peakBand = 0;
+ this.peak = 0;
+
+ /**
+ * Calculates the *middle* frequency of an FFT band.
+ *
+ * @param {Number} index The index of the FFT band.
+ *
+ * @returns The middle frequency in Hz.
+ */
+ this.getBandFrequency = function(index) {
+ return this.bandwidth * index + this.bandwidth / 2;
+ };
+
+ /**
+ * Calculate the spectrum (amplitude magnitudes)
+ */
+ this.calculateSpectrum = function() {
+ var spectrum = this.spectrum,
+ real = this.real,
+ imag = this.imag,
+ bSi = 2 / this.bufferSize,
+ sqrt = Math.sqrt,
+ rval,
+ ival,
+ mag;
+
+ for (var i = 0, N = bufferSize/2; i < N; i++) {
+ rval = real[i];
+ ival = imag[i];
+ mag = bSi * sqrt(rval * rval + ival * ival);
+
+ if (mag > this.peak) {
+ this.peakBand = i;
+ this.peak = mag;
+ }
+
+ spectrum[i] = mag;
+ }
+ };
+};
+
+},{}],7:[function(require,module,exports){
+/* global Float64Array */
+var DSP = require("./dsp");
+var Biquad = require("./biquad");
+
+/**
+ * Create a 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.
+ *
+ * @constructor
+ * @param {SampleRate}
+ * @example
+ * var eq = new GraphicalEq(44100)
+ */
+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 = new 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;
+ };
+}
+
+module.exports = GraphicalEq;
+
+},{"./biquad":2,"./dsp":4}],8:[function(require,module,exports){
+var DSP = require("./dsp");
+var ADSR = require("./adsr");
+
+/**
+ * IIRFilter
+ * @constructor
+ */
+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;
+ }
+);
+
+/**
+ * Set filter parameters
+ * @param {Number} cutoff
+ * @param {Number} resonance
+ */
+IIRFilter.prototype.set = function(cutoff, resonance) {
+ this.func.calcCoeff(cutoff, resonance);
+};
+
+/**
+ * Process a buffer
+ * @param {Array} buffer
+ */
+IIRFilter.prototype.process = function(buffer) {
+ this.func.process(buffer);
+};
+
+/**
+ * Add an envelope to the filter
+ * @param {ADSR} envelope
+ */
+IIRFilter.prototype.addEnvelope = function(envelope) {
+ if ( envelope instanceof ADSR ) {
+ this.func.addEnvelope(envelope);
+ } else {
+ throw "Not an envelope.";
+ }
+};
+
+/**
+ * LP12 filter
+ * @constructor
+ */
+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;
+ }
+ }
+ };
+};
+
+/**
+ * Add an envelope to the filter
+ * @param {ADSR} envelope
+ */
+IIRFilter.LP12.prototype.addEnvelope = function(envelope) {
+ this.envelope = envelope;
+};
+
+module.exports = IIRFilter;
+
+},{"./adsr":1,"./dsp":4}],9:[function(require,module,exports){
+/* global Float64Array */
+var ADSR = require("./adsr");
+
+/**
+ * IIRFilter2
+ *
+ * @constructor
+ * @param {Number} type
+ * @param {Number} cutoff
+ * @param {Number} resonance
+ * @param {Number} sampleRate
+ */
+function IIRFilter2(type, cutoff, resonance, sampleRate) {
+ this.type = type;
+ this.cutoff = cutoff;
+ this.resonance = resonance;
+ this.sampleRate = sampleRate;
+
+ this.f = new 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);
+}
+
+/**
+ * Process a buffer
+ * @param {Array} buffer
+ */
+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;
+ }
+ }
+};
+
+/**
+ * Add an envelope to the filter
+ * @param {ADSR} envelope
+ */
+IIRFilter2.prototype.addEnvelope = function(envelope) {
+ if ( envelope instanceof ADSR ) {
+ this.envelope = envelope;
+ } else {
+ throw "This is not an envelope.";
+ }
+};
+
+/**
+ * Set filter parameters
+ * @param {Number} cutoff
+ * @param {Number} resonance
+ */
+IIRFilter2.prototype.set = function(cutoff, resonance) {
+ this.calcCoeff(cutoff, resonance);
+};
+
+module.exports = IIRFilter2;
+
+},{"./adsr":1}],10:[function(require,module,exports){
+"use strict";
+
+/*
+ * DSP.js - a comprehensive digital signal processing library for javascript
+ *
+ * Created by Corban Brook <corbanbrook@gmail.com> on 2010-01-01.
+ * Copyright 2010 Corban Brook. All rights reserved.
+ *
+ */
+var DSPJS = {
+ DSP: require("./dsp"),
+ DFT: require("./dft"),
+ FFT: require("./fft"),
+ RFFT: require("./rfft"),
+ Sampler: require("./sampler"),
+ Oscillator: require("./oscillator"),
+ ADSR: require("./adsr"),
+ IIRFilter: require("./iir-filter"),
+ IIRFilter2: require("./iir-filter2"),
+ WindowFunction: require("./window-function"),
+ sinh: require("./sinh"),
+ Biquad: require("./biquad"),
+ GraphicalEq: require("./graphical-eq"),
+ MultiDelay: require("./multi-delay"),
+ SingleDelay: require("./single-delay"),
+ Reverb: require("./reverb")
+};
+
+if (typeof module === "object" && module.exports) module.exports = DSPJS;
+if (typeof window !== "undefined") {
+ Object.keys(DSPJS).forEach(function (k) {
+ window[k] = DSPJS[k];
+ });
+}
+
+},{"./adsr":1,"./biquad":2,"./dft":3,"./dsp":4,"./fft":5,"./graphical-eq":7,"./iir-filter":8,"./iir-filter2":9,"./multi-delay":11,"./oscillator":12,"./reverb":13,"./rfft":14,"./sampler":15,"./single-delay":16,"./sinh":17,"./window-function":18}],11:[function(require,module,exports){
+/* global Float64Array */
+/**
+ * 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;
+ }
};
-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;
+/**
+ * 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;
+};
- 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;
+/**
+ * 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.samplesProcessed++;
- }
- this.frameCount++;
+ this.delayOutputPointer++;
+ if (this.delayOutputPointer >= this.delayBufferSamples.length-1) {
+ this.delayOutputPointer = 0;
+ }
+ }
- return this.signal;
+ return outputSamples;
};
-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);
-};
+module.exports = MultiDelay;
-Sampler.prototype.reset = function() {
- this.samplesProcessed = 0;
- this.playhead = 0;
-};
+},{}],12:[function(require,module,exports){
+/* global Float64Array */
+var DSP = require("./dsp");
/**
* Oscillator class for generating and modifying signals
@@ -955,7 +1541,7 @@ Sampler.prototype.reset = function() {
* @param {Number} bufferSize Size of the sample buffer to generate
* @param {Number} sampleRate The sample rate of the signal
*
- * @contructor
+ * @constructor
*/
function Oscillator(type, frequency, amplitude, bufferSize, sampleRate) {
this.frequency = frequency;
@@ -964,7 +1550,7 @@ function Oscillator(type, frequency, amplitude, bufferSize, sampleRate) {
this.sampleRate = sampleRate;
//this.pulseWidth = pulseWidth;
this.frameCount = 0;
-
+
this.waveTableLength = 2048;
this.cyclesPerSample = frequency / sampleRate;
@@ -1001,14 +1587,14 @@ function Oscillator(type, frequency, amplitude, bufferSize, sampleRate) {
}
};
- if ( typeof Oscillator.waveTable === 'undefined' ) {
+ if ( typeof Oscillator.waveTable === "undefined" ) {
Oscillator.waveTable = {};
}
- if ( typeof Oscillator.waveTable[this.func] === 'undefined' ) {
+ if ( typeof Oscillator.waveTable[this.func] === "undefined" ) {
this.generateWaveTable();
}
-
+
this.waveTable = Oscillator.waveTable[this.func];
}
@@ -1024,35 +1610,42 @@ Oscillator.prototype.setAmp = function(amplitude) {
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
+
+/**
+ * Add an oscillator
+ * @param {Oscillator} oscillator The oscillator to be added to
+ * @return {Array} the current oscillator signal
+ */
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
+
+/**
+ * Add a signal to the current generated osc signal
+ * @param {Array} 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 ) {
@@ -1064,20 +1657,34 @@ Oscillator.prototype.addSignal = function(signal) {
}
return this.signal;
};
-
-// Add an envelope to the oscillator
+
+/**
+ * Add an envelope to the oscillator
+ * @param {ADSR} envelope
+ */
Oscillator.prototype.addEnvelope = function(envelope) {
this.envelope = envelope;
};
+/**
+ * Apply the oscillator envelope to its signal
+ */
Oscillator.prototype.applyEnvelope = function() {
this.envelope.process(this.signal);
};
-
+
+/**
+ * Get value
+ * @param {Number} offset
+ */
Oscillator.prototype.valueAt = function(offset) {
return this.waveTable[offset % this.waveTableLength];
};
-
+
+/**
+ * Generate the oscillator signal
+ * @return {Array} the signal
+ */
Oscillator.prototype.generate = function() {
var frameOffset = this.frameCount * this.bufferSize;
var step = this.waveTableLength * this.frequency / this.sampleRate;
@@ -1087,967 +1694,671 @@ Oscillator.prototype.generate = function() {
//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;
+ offset = Math.round((frameOffset + i) * step);
+ this.signal[i] = this.waveTable[offset % this.waveTableLength] * this.amplitude;
}
-}
-IIRFilter.prototype.__defineGetter__('cutoff',
- function() {
- return this.func.cutoff;
- }
-);
+ this.frameCount++;
-IIRFilter.prototype.__defineGetter__('resonance',
- function() {
- return this.func.resonance;
- }
-);
+ return this.signal;
+};
-IIRFilter.prototype.set = function(cutoff, resonance) {
- this.func.calcCoeff(cutoff, resonance);
+Oscillator.Sine = function(step) {
+ return Math.sin(DSP.TWO_PI * step);
};
-IIRFilter.prototype.process = function(buffer) {
- this.func.process(buffer);
+Oscillator.Square = function(step) {
+ return step < 0.5 ? 1 : -1;
};
-// Add an envelope to the filter
-IIRFilter.prototype.addEnvelope = function(envelope) {
- if ( envelope instanceof ADSR ) {
- this.func.addEnvelope(envelope);
- } else {
- throw "Not an envelope.";
- }
+Oscillator.Saw = function(step) {
+ return 2 * (step - Math.round(step));
};
-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;
- };
+Oscillator.Triangle = function(step) {
+ return 1 - 4 * Math.abs(Math.round(step) - step);
+};
- this.calcCoeff(cutoff, resonance);
+Oscillator.Pulse = function() {
+ // stub
+};
- 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;
- */
+module.exports = Oscillator;
- if (this.envelope) {
- buffer[i] = (buffer[i] * (1 - this.envelope.value())) + (this.vibraPos * this.envelope.value());
- this.envelope.samplesProcessed++;
- } else {
- buffer[i] = this.vibraPos;
- }
- }
- };
-};
+},{"./dsp":4}],13:[function(require,module,exports){
+/* global Float64Array */
+var DSP = require("./dsp");
+var IIRFilter2 = require("./iir-filter2");
+var SingleDelay = require("./single-delay");
+var MultiDelay = require("./multi-delay");
-IIRFilter.LP12.prototype.addEnvelope = function(envelope) {
- this.envelope = envelope;
-};
+/**
+ * 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;
-function IIRFilter2(type, cutoff, resonance, sampleRate) {
- this.type = type;
- this.cutoff = cutoff;
- this.resonance = resonance;
- this.sampleRate = sampleRate;
+ this.NR_OF_MULTIDELAYS = 6;
+ this.NR_OF_SINGLEDELAYS = 6;
- 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.LOWPASSL = new IIRFilter2(DSP.LOWPASS, dampFrequency, 0, 44100);
+ this.LOWPASSR = new IIRFilter2(DSP.LOWPASS, dampFrequency, 0, 44100);
- this.calcCoeff(cutoff, resonance);
-}
+ this.singleDelays = [];
-IIRFilter2.prototype.process = function(buffer) {
- var input, output;
- var f = this.f;
+ var i, delayMultiply;
- for ( var i = 0; i < buffer.length; i++ ) {
- input = buffer[i];
+ 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);
+ }
- // 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];
+ this.multiDelays = [];
- // 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];
+ 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);
+ }
+}
- if (this.envelope) {
- buffer[i] = (buffer[i] * (1 - this.envelope.value())) + (output * this.envelope.value());
- this.envelope.samplesProcessed++;
- } else {
- buffer[i] = output;
- }
+/**
+ * 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) );
}
-};
-IIRFilter2.prototype.addEnvelope = function(envelope) {
- if ( envelope instanceof ADSR ) {
- this.envelope = envelope;
- } else {
- throw "This is not an envelope.";
+ 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) );
}
};
-IIRFilter2.prototype.set = function(cutoff, resonance) {
- this.calcCoeff(cutoff, resonance);
+/**
+ * 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;
-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;
+ for (i = 0; i<this.NR_OF_SINGLEDELAYS; i++) {
+ this.singleDelays[i].setDelayVolume(this.delayVolume);
}
-}
-WindowFunction.prototype.process = function(buffer) {
- var length = buffer.length;
- for ( var i = 0; i < length; i++ ) {
- buffer[i] *= this.func(length, i, this.alpha);
+ for (i = 0; i<this.NR_OF_MULTIDELAYS; i++) {
+ this.multiDelays[i].setDelayVolume(this.delayVolume);
}
- return buffer;
};
-WindowFunction.Bartlett = function(length, index) {
- return 2 / (length - 1) * ((length - 1) / 2 - Math.abs(index - (length - 1) / 2));
-};
+/**
+ * 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;
-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));
+ this.LOWPASSL.set(dampFrequency, 0);
+ this.LOWPASSR.set(dampFrequency, 0);
};
-WindowFunction.Blackman = function(length, index, alpha) {
- var a0 = (1 - alpha) / 2;
- var a1 = 0.5;
- var a2 = alpha / 2;
+/**
+ * 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);
- return a0 - a1 * Math.cos(DSP.TWO_PI * index / (length - 1)) + a2 * Math.cos(4 * Math.PI * index / (length - 1));
-};
+ // 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]);
-WindowFunction.Cosine = function(length, index) {
- return Math.cos(Math.PI * index / (length - 1) - Math.PI / 2);
-};
+ var i;
-WindowFunction.Gauss = function(length, index, alpha) {
- return Math.pow(Math.E, -0.5 * Math.pow((index - (length - 1) / 2) / (alpha * (length - 1) / 2), 2));
-};
+ // 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);
+ }
-WindowFunction.Hamming = function(length, index) {
- return 0.54 - 0.46 * Math.cos(DSP.TWO_PI * index / (length - 1));
-};
+ // 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);
+ }
-WindowFunction.Hann = function(length, index) {
- return 0.5 * (1 - Math.cos(DSP.TWO_PI * index / (length - 1)));
-};
+ // Apply the volume of the reverb signal
+ for (i = 0; i<singleDelaySamples.length; i++) {
+ singleDelaySamples[i] *= this.mixVolume;
+ }
-WindowFunction.Lanczos = function(length, index) {
- var x = 2 * index / (length - 1) - 1;
- return Math.sin(Math.PI * x) / (Math.PI * x);
-};
+ // Mix the original signal with the reverb signal
+ outputSamples = DSP.mixSampleBuffers(singleDelaySamples, interleavedSamples, 0, 1);
-WindowFunction.Rectangular = function(length, index) {
- return 1;
-};
+ // Apply the master volume to the complete signal
+ for (i = 0; i<outputSamples.length; i++) {
+ outputSamples[i] *= this.masterVolume;
+ }
-WindowFunction.Triangular = function(length, index) {
- return 2 / length * (length / 2 - Math.abs(index - (length - 1) / 2));
+ return outputSamples;
};
-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;
-}
+module.exports = Reverb;
-/*
- * Biquad filter
- *
- * Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
- * Copyright 2010 Ricard Marxer. All rights reserved.
+},{"./dsp":4,"./iir-filter2":9,"./multi-delay":11,"./single-delay":16}],14:[function(require,module,exports){
+/* global Float64Array Uint32Array */
+var FourierTransform = require("./fourier");
+
+/**
+ * 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
*/
-// 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;
+// lookup tables don't really gain us any speed, but they do increase
+// cache footprint, so don't use them in here
- this.x_1_r = 0;
- this.x_2_r = 0;
- this.y_1_r = 0;
- this.y_2_r = 0;
+// also we don't use sepearate arrays for real/imaginary parts
- this.b0 = 1;
- this.a0 = 1;
+// this one a little more than twice as fast as the one in FFT
+// however I only did the forward transform
- this.b1 = 0;
- this.a1 = 0;
+// the rest of this was translated from C, see http://www.jjj.de/fxt/
+// this is the real split radix FFT
- this.b2 = 0;
- this.a2 = 0;
+function RFFT(bufferSize, sampleRate) {
+ FourierTransform.call(this, bufferSize, sampleRate);
- 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.trans = new Float64Array(bufferSize);
- 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.reverseTable = new Uint32Array(bufferSize);
- this.dBgain = 12; // used only for peaking and shelving filters
+ // 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;
- 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".
+ dest[0] = source[0];
- 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
+ do {
+ r += halfSize;
+ dest[i] = source[r];
+ dest[r] = source[i];
- 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.
+ i++;
- this.coefficients = function() {
- var b = [this.b0, this.b1, this.b2];
- var a = [this.a0, this.a1, this.a2];
- return {b: b, a:a};
- };
+ h = halfSize << 1;
+ while (h = h >> 1, !((r ^= h) & h));
- this.setFilterType = function(type) {
- this.type = type;
- this.recalculateCoefficients();
- };
+ if (r >= i) {
+ dest[i] = source[r];
+ dest[r] = source[i];
- this.setSampleRate = function(rate) {
- this.Fs = rate;
- this.recalculateCoefficients();
+ dest[nm1-i] = source[nm1-r];
+ dest[nm1-r] = source[nm1-i];
+ }
+ i++;
+ } while (i < halfSize);
+ dest[nm1] = source[nm1];
};
- this.setQ = function(q) {
- this.parameterType = DSP.Q;
- this.Q = Math.max(Math.min(q, 115.0), 0.001);
- this.recalculateCoefficients();
- };
+ this.generateReverseTable = function () {
+ var bufferSize = this.bufferSize,
+ halfSize = bufferSize >>> 1,
+ nm1 = bufferSize - 1,
+ i = 1, r = 0, h;
- this.setBW = function(bw) {
- this.parameterType = DSP.BW;
- this.BW = bw;
- this.recalculateCoefficients();
- };
+ this.reverseTable[0] = 0;
- this.setS = function(s) {
- this.parameterType = DSP.S;
- this.S = Math.max(Math.min(s, 5.0), 0.0001);
- this.recalculateCoefficients();
- };
+ do {
+ r += halfSize;
- this.setF0 = function(freq) {
- this.f0 = freq;
- this.recalculateCoefficients();
- };
-
- this.setDbGain = function(g) {
- this.dBgain = g;
- this.recalculateCoefficients();
- };
+ this.reverseTable[i] = r;
+ this.reverseTable[r] = i;
- 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)) );
- }
+ i++;
- var w0 = DSP.TWO_PI * this.f0 / this.Fs;
+ h = halfSize << 1;
+ while (h = h >> 1, !((r ^= h) & h));
- var cosw0 = Math.cos(w0);
- var sinw0 = Math.sin(w0);
+ if (r >= i) {
+ this.reverseTable[i] = r;
+ this.reverseTable[r] = i;
- 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;
+ this.reverseTable[nm1-i] = nm1-r;
+ this.reverseTable[nm1-r] = nm1-i;
+ }
+ i++;
+ } while (i < halfSize);
- case DSP.S:
- alpha = sinw0/2 * Math.sqrt( (A + 1/A)*(1/this.S - 1) + 2 );
- break;
- }
+ this.reverseTable[nm1] = nm1;
+ };
- /**
- 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)
+ this.generateReverseTable();
+}
+
+
+/**
+ * Performs a forward transform on the sample buffer.
+ * Converts a time domain signal to frequency domain spectra.
+ *
+ * @param {Array} buffer The sample buffer. Buffer Length must be power of 2
+ *
+ * @returns The frequency spectrum array
+ */
+
+// 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;
- The relationship between shelf slope and Q is
- 1/Q = sqrt((A + 1/A)*(1/S - 1) + 2)
- */
+ this.reverseBinPermute(x, buffer);
- var coeff;
+ /*
+ var reverseTable = this.reverseTable;
- 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;
+ for (var k = 0, len = reverseTable.length; k < len; k++) {
+ x[k] = buffer[reverseTable[k]];
+ }
+ */
- 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;
+ 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);
+ }
- 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;
+ n2 = 2;
+ nn = n >>> 1;
- 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;
+ 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;
- 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;
+ //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;
- 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;
+ i1 += n8;
+ i2 += n8;
+ i3 += n8;
+ i4 += n8;
- 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;
+ //sumdiff(x[i3], x[i4], t1, t2); // {s, d} <--| {a+b, a-b}
+ t1 = x[i3] + x[i4];
+ t2 = x[i3] - x[i4];
- 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;
+ t1 = -t1 * Math.SQRT1_2;
+ t2 *= Math.SQRT1_2;
- 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;
- };
+ // 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;
- 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]
+ //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;
- var len = buffer.length;
- var output = new Float64Array(len);
+ //diffsum3_r(x[i3], x[i4], t1); // {a, b, s} <--| {a, b-a, a+b}
+ t1 = x[i3] + x[i4];
+ x[i4] -= x[i3];
- 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];
+ //sumdiff3(x[i1], t1, x[i3]); // {a, b, d} <--| {a+b, b, a-b}
+ x[i3] = x[i1] - t1;
+ x[i1] += t1;
+ }
}
- return output;
- };
+ ix = (id << 1) - n2;
+ id = id << 2;
+ } while (ix < n);
- 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]
+ e = TWO_PI / n2;
- 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];
+ for (var j = 1; j < n8; j++) {
+ a = j * e;
+ ss1 = Math.sin(a);
+ cc1 = Math.cos(a);
- 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];
- }
+ //ss3 = sin(3*a); cc3 = cos(3*a);
+ cc3 = 4*cc1*(cc1*cc1-0.75);
+ ss3 = 4*ss1*(0.75-ss1*ss1);
- return output;
- };
-}
+ 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;
-/*
- * 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);
+ i5 = i0 + n4 - j;
+ i6 = i5 + n4;
+ i7 = i6 + n4;
+ i8 = i7 + n4;
- 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));
- }
+ //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;
- return result;
-};
+ //cmult(cc3, ss3, x[i8], x[i4], t4, t3);
+ t4 = x[i8]*cc3 - x[i4]*ss3;
+ t3 = x[i8]*ss3 + x[i4]*cc3;
-/*
- * 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;
+ //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];
- if (!w) {
- w = Float64Array(200);
- for (i=0;i<w.length; i++) {
- w[i] = DSP.TWO_PI/w.length * i - Math.PI;
+ //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);
}
}
- 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]);
- }
+ while (--i) {
+ rval = x[i];
+ ival = x[n-i-1];
+ mag = bSi * sqrt(rval * rval + ival * ival);
- 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]);
+ if (mag > this.peak) {
+ this.peakBand = i;
+ this.peak = mag;
}
-
- result[i] = sqrt(numerator.real*numerator.real + numerator.imag*numerator.imag) / sqrt(denominator.real*denominator.real + denominator.imag*denominator.imag);
+
+ spectrum[i] = mag;
}
- return result;
-};
+ spectrum[0] = bSi * x[0];
-/*
- * 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;
+ return spectrum;
+};
- this.bandsPerOctave = 1.0;
+module.exports = RFFT;
- this.filters = [];
- this.freqzs = [];
+},{"./fourier":6}],15:[function(require,module,exports){
+/* global Float64Array */
+var DSP = require("./dsp");
- this.calculateFreqzs = true;
+/**
+ * Sampler
+ *
+ * @constructor
+ * @param {} file
+ * @param {Number} bufferSize
+ * @param {Number} sampleRate
+ * @param {Number} playStart
+ * @param {Number} playEnd
+ * @param {Number} loopStart
+ * @param {Number} loopEnd
+ * @param {Number} loopMode
+ */
+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;
- this.recalculateFilters = function() {
- var bandCount = Math.round(Math.log(this.maxFreq/this.minFreq) * this.bandsPerOctave/ Math.LN2);
+ var audio = /* new Audio();*/ document.createElement("AUDIO");
+ var self = this;
- 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.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.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.loadComplete = function() {
+ // convert flexible js array into a fast typed array
+ self.samples = new Float64Array(self.samples);
+ self.loaded = true;
};
- 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.loadMetaData = function() {
+ self.duration = audio.duration;
};
-
- 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];
+ 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();
+}
- this.freqzs[bandIndex] = DSP.mag2db(DSP.freqz(b, a, this.w));
- };
+Sampler.prototype.applyEnvelope = function() {
+ this.envelope.process(this.signal);
+ return this.signal;
+};
- this.process = function(buffer) {
- var output = buffer;
+Sampler.prototype.generate = function() {
+ 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
- for (var i = 0; i < this.filters.length; i++) {
- output = this.filters[i].process(output);
- }
+ 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;
- return output;
- };
+ 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;
- this.processStereo = function(buffer) {
- var output = buffer;
+ 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;
- for (var i = 0; i < this.filters.length; i++) {
- output = this.filters[i].processStereo(output);
+ 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;
}
-
- 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;
+ this.samplesProcessed++;
}
-};
-/**
- * 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;
+ this.frameCount++;
+
+ return this.signal;
};
-/**
- * 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;
+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);
};
-/**
- * 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;
+Sampler.prototype.reset = function() {
+ this.samplesProcessed = 0;
+ this.playhead = 0;
};
+module.exports = Sampler;
+
+},{"./dsp":4}],16:[function(require,module,exports){
+/* global Float64Array */
+
/**
* 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
+ * 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(?).
*
@@ -2060,12 +2371,11 @@ MultiDelay.prototype.process = function(samples) {
*
* @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;
}
@@ -2080,7 +2390,7 @@ SingleDelay.prototype.setDelayInSamples = function(delayInSamples) {
this.delayInputPointer = this.delayOutputPointer + delayInSamples;
if (this.delayInputPointer >= this.delayBufferSamples.length-1) {
- this.delayInputPointer = this.delayInputPointer - this.delayBufferSamples.length;
+ this.delayInputPointer = this.delayInputPointer - this.delayBufferSamples.length;
}
};
@@ -2109,7 +2419,7 @@ SingleDelay.prototype.process = function(samples) {
// 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];
@@ -2122,201 +2432,152 @@ SingleDelay.prototype.process = function(samples) {
if (this.delayInputPointer >= this.delayBufferSamples.length-1) {
this.delayInputPointer = 0;
}
-
+
this.delayOutputPointer++;
if (this.delayOutputPointer >= this.delayBufferSamples.length-1) {
- this.delayOutputPointer = 0;
- }
+ this.delayOutputPointer = 0;
+ }
}
-
+
return outputSamples;
};
+module.exports = SingleDelay;
+
+},{}],17:[function(require,module,exports){
/**
- * 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/
+ * Returns the hyperbolic sine of the number
*
- * This reverb consists of 6 SingleDelays, 6 MultiDelays and an IIRFilter2
- * for each of the two stereo channels.
- *
- * Compatible with interleaved stereo buffers only!
+ * @meta version: 1004.2314
+ * @meta discuss at: http://phpjs.org/functions/sinh
+ * @meta original by: Onno Marsman
*
- * @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)
+ * @param {Number} num
+ * @example
+ * sinh(-0.9834330348825909); // => -1.1497971402636502
+ */
+function sinh (arg) {
+ return (Math.exp(arg) - Math.exp(-arg))/2;
+}
+
+module.exports = sinh;
+
+},{}],18:[function(require,module,exports){
+var DSP = require("./dsp");
+
+/**
+ * WindowFunction
*
* @constructor
+ * @param {Number} type
+ * @param {Number} alpha
*/
-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;
+function WindowFunction(type, alpha) {
+ this.alpha = alpha;
- 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 = [];
+ switch(type) {
+ case DSP.BARTLETT:
+ this.func = WindowFunction.Bartlett;
+ break;
- 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);
+ 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;
}
}
/**
- * Change the delay time in samples as a base for all delays.
- *
- * @param {Number} delayInSamples Delay in samples
+ * Process a buffer
+ * @param {Array} buffer
*/
-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) );
+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;
};
-/**
- * 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;
+WindowFunction.Bartlett = function(length, index) {
+ return 2 / (length - 1) * ((length - 1) / 2 - Math.abs(index - (length - 1) / 2));
};
-/**
- * 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;
+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));
};
-/**
- * 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;
+WindowFunction.Blackman = function(length, index, alpha) {
+ var a0 = (1 - alpha) / 2;
+ var a1 = 0.5;
+ var a2 = alpha / 2;
- 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);
- }
+ return a0 - a1 * Math.cos(DSP.TWO_PI * index / (length - 1)) + a2 * Math.cos(4 * Math.PI * index / (length - 1));
};
-/**
- * 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);
+WindowFunction.Cosine = function(length, index) {
+ return Math.cos(Math.PI * index / (length - 1) - Math.PI / 2);
};
-/**
- * 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]);
+WindowFunction.Gauss = function(length, index, alpha) {
+ return Math.pow(Math.E, -0.5 * Math.pow((index - (length - 1) / 2) / (alpha * (length - 1) / 2), 2));
+};
- var i;
+WindowFunction.Hamming = function(length, index) {
+ return 0.54 - 0.46 * Math.cos(DSP.TWO_PI * index / (length - 1));
+};
- // 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);
- }
+WindowFunction.Hann = function(length, index) {
+ return 0.5 * (1 - Math.cos(DSP.TWO_PI * index / (length - 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);
+WindowFunction.Lanczos = function(length, index) {
+ var x = 2 * index / (length - 1) - 1;
+ return Math.sin(Math.PI * x) / (Math.PI * x);
+};
- // Apply the master volume to the complete signal
- for (i = 0; i<outputSamples.length; i++) {
- outputSamples[i] *= this.masterVolume;
- }
-
- return outputSamples;
+WindowFunction.Rectangular = function() {
+ return 1;
};
-if (module && typeof module.exports !== 'undefined') {
- module.exports = {
- 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
- };
-}
\ No newline at end of file
+WindowFunction.Triangular = function(length, index) {
+ return 2 / length * (length / 2 - Math.abs(index - (length - 1) / 2));
+};
+
+module.exports = WindowFunction;
+
+},{"./dsp":4}]},{},[10]);
diff --git a/dist/dsp.min.js b/dist/dsp.min.js
new file mode 100644
index 0000000..f8c950b
--- /dev/null
+++ b/dist/dsp.min.js
@@ -0,0 +1,3 @@
+(function e(t,n,r){function s(o,u){if(!n[o]){if(!t[o]){var a=typeof require=="function"&&require;if(!u&&a)return a(o,!0);if(i)return i(o,!0);var f=new Error("Cannot find module '"+o+"'");throw f.code="MODULE_NOT_FOUND",f}var l=n[o]={exports:{}};t[o][0].call(l.exports,function(e){var n=t[o][1][e];return s(n?n:e)},l,l.exports,e,t,n,r)}return n[o].exports}var i=typeof require=="function"&&require;for(var o=0;o<r.length;o++)s(r[o]);return s})({1:[function(require,module,exports){function ADSR(attackLength,decayLength,sustainLevel,sustainLength,releaseLength,sampleRate){this.sampleRate=sampleRate;this.attackLength=attackLength;this.decayLength=decayLength;this.sustainLevel=sustainLevel;this.sustainLength=sustainLength;this.releaseLength=releaseLength;this.sampleRate=sampleRate;this.attackSamples=attackLength*sampleRate;this.decaySamples=decayLength*sampleRate;this.sustainSamples=sustainLength*sampleRate;this.releaseSamples=releaseLength*sampleRate;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()};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};module.exports=ADSR},{}],2:[function(require,module,exports){var DSP=require("./dsp");var sinh=require("./sinh");function Biquad(type,sampleRate){this.Fs=sampleRate;this.type=type;this.parameterType=DSP.Q;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=3e3;this.dBgain=12;this.Q=1;this.BW=-3;this.S=1;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),.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),1e-4);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)}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}var coeff;switch(this.type){case DSP.LPF: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: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: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: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: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: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: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: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: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){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){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}}module.exports=Biquad},{"./dsp":4,"./sinh":17}],3:[function(require,module,exports){var FourierTransform=require("./fourier");function DFT(bufferSize,sampleRate){FourierTransform.call(this,bufferSize,sampleRate);var N=bufferSize/2*bufferSize;var TWO_PI=2*Math.PI;this.sinTable=new Float64Array(N);this.cosTable=new Float64Array(N);for(var i=0;i<N;i++){this.sinTable[i]=Math.sin(i*TWO_PI/bufferSize);this.cosTable[i]=Math.cos(i*TWO_PI/bufferSize)}}DFT.prototype.forward=function(buffer){var real=this.real,imag=this.imag,rval,ival;for(var k=0;k<this.bufferSize/2;k++){rval=0;ival=0;for(var n=0;n<buffer.length;n++){rval+=this.cosTable[k*n]*buffer[n];ival+=this.sinTable[k*n]*buffer[n]}real[k]=rval;imag[k]=ival}return this.calculateSpectrum()};module.exports=DFT},{"./fourier":6}],4:[function(require,module,exports){var DSP={LEFT:0,RIGHT:1,MIX:2,SINE:1,TRIANGLE:2,SAW:3,SQUARE:4,LOWPASS:0,HIGHPASS:1,BANDPASS:2,NOTCH:3,BARTLETT:1,BARTLETTHANN:2,BLACKMAN:3,COSINE:4,GAUSS:5,HAMMING:6,HANN:7,LANCZOS:8,RECTANGULAR:9,TRIANGULAR:10,OFF:0,FW:1,BW:2,FWBW:3,TWO_PI:2*Math.PI};function setupTypedArray(name,fallback){if(typeof this[name]!=="function"&&typeof this[name]!=="object"){if(typeof this[fallback]==="function"&&typeof this[fallback]!=="object"){this[name]=this[fallback]}else{this[name]=function(obj){if(obj instanceof Array){return obj}else if(typeof obj==="number"){return new Array(obj)}}}}}setupTypedArray("Float64Array","WebGLFloatArray");setupTypedArray("Int32Array","WebGLIntArray");setupTypedArray("Uint16Array","WebGLUnsignedShortArray");setupTypedArray("Uint8Array","WebGLUnsignedByteArray");DSP.invert=function(buffer){for(var i=0,len=buffer.length;i<len;i++){buffer[i]*=-1}return buffer};DSP.interleave=function(left,right){if(left.length!==right.length){throw"Can not interleave. Channel lengths differ."}var stereoInterleaved=new Float64Array(left.length*2);for(var i=0,len=left.length;i<len;i++){stereoInterleaved[2*i]=left[i];stereoInterleaved[2*i+1]=right[i]}return stereoInterleaved};DSP.deinterleave=function(){var left,right,mix,deinterleaveChannel=[];deinterleaveChannel[DSP.MIX]=function(buffer){for(var i=0,len=buffer.length/2;i<len;i++){mix[i]=(buffer[2*i]+buffer[2*i+1])/2}return mix};deinterleaveChannel[DSP.LEFT]=function(buffer){for(var i=0,len=buffer.length/2;i<len;i++){left[i]=buffer[2*i]}return left};deinterleaveChannel[DSP.RIGHT]=function(buffer){for(var i=0,len=buffer.length/2;i<len;i++){right[i]=buffer[2*i+1]}return right};return function(channel,buffer){left=left||new Float64Array(buffer.length/2);right=right||new Float64Array(buffer.length/2);mix=mix||new Float64Array(buffer.length/2);if(buffer.length/2!==left.length){left=new Float64Array(buffer.length/2);right=new Float64Array(buffer.length/2);mix=new Float64Array(buffer.length/2)}return deinterleaveChannel[channel](buffer)}}();DSP.getChannel=DSP.deinterleave;DSP.mixSampleBuffers=function(sampleBuffer1,sampleBuffer2,negate,volumeCorrection){var outputSamples=new Float64Array(sampleBuffer1);for(var i=0;i<sampleBuffer1.length;i++){outputSamples[i]+=(negate?-sampleBuffer2[i]:sampleBuffer2[i])/volumeCorrection}return outputSamples};DSP.LPF=0;DSP.HPF=1;DSP.BPF_CONSTANT_SKIRT=2;DSP.BPF_CONSTANT_PEAK=3;DSP.NOTCH=4;DSP.APF=5;DSP.PEAKING_EQ=6;DSP.LOW_SHELF=7;DSP.HIGH_SHELF=8;DSP.Q=1;DSP.BW=2;DSP.S=3;DSP.RMS=function(buffer){var total=0;for(var i=0,n=buffer.length;i<n;i++){total+=buffer[i]*buffer[i]}return Math.sqrt(total/n)};DSP.Peak=function(buffer){var peak=0;for(var i=0,n=buffer.length;i<n;i++){peak=Math.abs(buffer[i])>peak?Math.abs(buffer[i]):peak}return peak};DSP.mag2db=function(buffer){var minDb=-120;var minMag=Math.pow(10,minDb/20);var log=Math.log;var max=Math.max;var result=new Float64Array(buffer.length);for(var i=0;i<buffer.length;i++){result[i]=20*log(max(buffer[i],minMag))}return result};DSP.freqz=function(b,a,w){var i,j;if(!w){w=new Float64Array(200);for(i=0;i<w.length;i++){w[i]=DSP.TWO_PI/w.length*i-Math.PI}}var result=new 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,imag: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,imag: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};module.exports=DSP},{}],5:[function(require,module,exports){var FourierTransform=require("./fourier");function FFT(bufferSize,sampleRate){FourierTransform.call(this,bufferSize,sampleRate);this.reverseTable=new Uint32Array(bufferSize);var limit=1;var bit=bufferSize>>1;var i;while(limit<bufferSize){for(i=0;i<limit;i++){this.reverseTable[i+limit]=this.reverseTable[i]+bit}limit=limit<<1;bit=bit>>1}this.sinTable=new Float64Array(bufferSize);this.cosTable=new Float64Array(bufferSize);for(i=0;i<bufferSize;i++){this.sinTable[i]=Math.sin(-Math.PI/i);this.cosTable[i]=Math.cos(-Math.PI/i)}}FFT.prototype.forward=function(buffer){var bufferSize=this.bufferSize,cosTable=this.cosTable,sinTable=this.sinTable,reverseTable=this.reverseTable,real=this.real,imag=this.imag;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=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 this.calculateSpectrum()};FFT.prototype.inverse=function(real,imag){var bufferSize=this.bufferSize,cosTable=this.cosTable,sinTable=this.sinTable,reverseTable=this.reverseTable;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);for(i=0;i<bufferSize;i++){buffer[i]=real[i]/bufferSize}return buffer};module.exports=FFT},{"./fourier":6}],6:[function(require,module,exports){module.exports=function FourierTransform(bufferSize,sampleRate){this.bufferSize=bufferSize;this.sampleRate=sampleRate;this.bandwidth=2/bufferSize*sampleRate/2;this.spectrum=new Float64Array(bufferSize/2);this.real=new Float64Array(bufferSize);this.imag=new Float64Array(bufferSize);this.peakBand=0;this.peak=0;this.getBandFrequency=function(index){return this.bandwidth*index+this.bandwidth/2};this.calculateSpectrum=function(){var spectrum=this.spectrum,real=this.real,imag=this.imag,bSi=2/this.bufferSize,sqrt=Math.sqrt,rval,ival,mag;for(var i=0,N=bufferSize/2;i<N;i++){rval=real[i];ival=imag[i];mag=bSi*sqrt(rval*rval+ival*ival);if(mag>this.peak){this.peakBand=i;this.peak=mag}spectrum[i]=mag}}}},{}],7:[function(require,module,exports){var DSP=require("./dsp");var Biquad=require("./biquad");function GraphicalEq(sampleRate){this.FS=sampleRate;this.minFreq=40;this.maxFreq=16e3;this.bandsPerOctave=1;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=new 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}}module.exports=GraphicalEq},{"./biquad":2,"./dsp":4}],8:[function(require,module,exports){var DSP=require("./dsp");var ADSR=require("./adsr");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)};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*Math.PI*cutoff/this.sampleRate;this.q=1-this.w/(2*(resonance+.5/(1+this.w))+this.w-2);this.r=this.q*this.q;this.c=this.r+1-2*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;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};module.exports=IIRFilter},{"./adsr":1,"./dsp":4}],9:[function(require,module,exports){var ADSR=require("./adsr");function IIRFilter2(type,cutoff,resonance,sampleRate){this.type=type;this.cutoff=cutoff;this.resonance=resonance;this.sampleRate=sampleRate;this.f=new Float64Array(4);this.f[0]=0;this.f[1]=0;this.f[2]=0;this.f[3]=0;this.calcCoeff=function(cutoff,resonance){this.freq=2*Math.sin(Math.PI*Math.min(.25,cutoff/(this.sampleRate*2)));this.damp=Math.min(2*(1-Math.pow(resonance,.25)),Math.min(2,2/this.freq-this.freq*.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];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=.5*f[this.type];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+=.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)};module.exports=IIRFilter2},{"./adsr":1}],10:[function(require,module,exports){"use strict";var DSPJS={DSP:require("./dsp"),DFT:require("./dft"),FFT:require("./fft"),RFFT:require("./rfft"),Sampler:require("./sampler"),Oscillator:require("./oscillator"),ADSR:require("./adsr"),IIRFilter:require("./iir-filter"),IIRFilter2:require("./iir-filter2"),WindowFunction:require("./window-function"),sinh:require("./sinh"),Biquad:require("./biquad"),GraphicalEq:require("./graphical-eq"),MultiDelay:require("./multi-delay"),SingleDelay:require("./single-delay"),Reverb:require("./reverb")};if(typeof module==="object"&&module.exports)module.exports=DSPJS;if(typeof window!=="undefined"){Object.keys(DSPJS).forEach(function(k){window[k]=DSPJS[k]})}},{"./adsr":1,"./biquad":2,"./dft":3,"./dsp":4,"./fft":5,"./graphical-eq":7,"./iir-filter":8,"./iir-filter2":9,"./multi-delay":11,"./oscillator":12,"./reverb":13,"./rfft":14,"./sampler":15,"./single-delay":16,"./sinh":17,"./window-function":18}],11:[function(require,module,exports){function MultiDelay(maxDelayInSamplesSize,delayInSamples,masterVolume,delayVolume){this.delayBufferSamples=new Float64Array(maxDelayInSamplesSize);this.delayInputPointer=delayInSamples;this.delayOutputPointer=0;this.delayInSamples=delayInSamples;this.masterVolume=masterVolume;this.delayVolume=delayVolume}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}};MultiDelay.prototype.setMasterVolume=function(masterVolume){this.masterVolume=masterVolume};MultiDelay.prototype.setDelayVolume=function(delayVolume){this.delayVolume=delayVolume};MultiDelay.prototype.process=function(samples){var outputSamples=new Float64Array(samples.length);for(var i=0;i<samples.length;i++){var delaySample=this.delayBufferSamples[this.delayOutputPointer]===null?0:this.delayBufferSamples[this.delayOutputPointer];var sample=delaySample*this.delayVolume+samples[i];this.delayBufferSamples[this.delayInputPointer]=sample;outputSamples[i]=sample*this.masterVolume;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};module.exports=MultiDelay},{}],12:[function(require,module,exports){var DSP=require("./dsp");function Oscillator(type,frequency,amplitude,bufferSize,sampleRate){this.frequency=frequency;this.amplitude=amplitude;this.bufferSize=bufferSize;this.sampleRate=sampleRate;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]}Oscillator.prototype.setAmp=function(amplitude){if(amplitude>=0&&litude<=1){this.amplitude=amplitude}else{throw"Amplitude out of range (0..1)."}};Oscillator.prototype.setFreq=function(frequency){this.frequency=frequency;this.cyclesPerSample=frequency/this.sampleRate};Oscillator.prototype.add=function(oscillator){for(var i=0;i<this.bufferSize;i++){this.signal[i]+=oscillator.signal[i]}return this.signal};Oscillator.prototype.addSignal=function(signal){for(var i=0;i<signal.length;i++){if(i>=this.bufferSize){break}this.signal[i]+=signal[i]}return this.signal};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++){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<.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(){};module.exports=Oscillator},{"./dsp":4}],13:[function(require,module,exports){var DSP=require("./dsp");var IIRFilter2=require("./iir-filter2");var SingleDelay=require("./single-delay");var MultiDelay=require("./multi-delay");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+i/7;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+i/10;this.multiDelays[i]=new MultiDelay(maxDelayInSamplesSize,Math.round(this.delayInSamples*delayMultiply),this.masterVolume,this.delayVolume)}}Reverb.prototype.setDelayInSamples=function(delayInSamples){this.delayInSamples=delayInSamples;var i,delayMultiply;for(i=0;i<this.NR_OF_SINGLEDELAYS;i++){delayMultiply=1+i/7;this.singleDelays[i].setDelayInSamples(Math.round(this.delayInSamples*delayMultiply))}for(i=0;i<this.NR_OF_MULTIDELAYS;i++){delayMultiply=1+i/10;this.multiDelays[i].setDelayInSamples(Math.round(this.delayInSamples*delayMultiply))}};Reverb.prototype.setMasterVolume=function(masterVolume){this.masterVolume=masterVolume};Reverb.prototype.setMixVolume=function(mixVolume){this.mixVolume=mixVolume};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)}};Reverb.prototype.setDampFrequency=function(dampFrequency){this.dampFrequency=dampFrequency;this.LOWPASSL.set(dampFrequency,0);this.LOWPASSR.set(dampFrequency,0)};Reverb.prototype.process=function(interleavedSamples){var outputSamples=new Float64Array(interleavedSamples.length);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;for(i=0;i<this.NR_OF_MULTIDELAYS;i++){outputSamples=DSP.mixSampleBuffers(outputSamples,this.multiDelays[i].process(filteredSamples),2%i===0,this.NR_OF_MULTIDELAYS)}var singleDelaySamples=new Float64Array(outputSamples.length);for(i=0;i<this.NR_OF_SINGLEDELAYS;i++){singleDelaySamples=DSP.mixSampleBuffers(singleDelaySamples,this.singleDelays[i].process(outputSamples),2%i===0,1)}for(i=0;i<singleDelaySamples.length;i++){singleDelaySamples[i]*=this.mixVolume}outputSamples=DSP.mixSampleBuffers(singleDelaySamples,interleavedSamples,0,1);for(i=0;i<outputSamples.length;i++){outputSamples[i]*=this.masterVolume}return outputSamples};module.exports=Reverb},{"./dsp":4,"./iir-filter2":9,"./multi-delay":11,"./single-delay":16}],14:[function(require,module,exports){var FourierTransform=require("./fourier");function RFFT(bufferSize,sampleRate){FourierTransform.call(this,bufferSize,sampleRate);this.trans=new Float64Array(bufferSize);this.reverseTable=new Uint32Array(bufferSize);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()}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);for(var ix=0,id=4;ix<n;id*=4){for(var i0=ix;i0<n;i0+=id){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;t1=x[i3]+x[i4];x[i4]-=x[i3];x[i3]=x[i1]-t1;x[i1]+=t1;i1+=n8;i2+=n8;i3+=n8;i4+=n8;t1=x[i3]+x[i4];t2=x[i3]-x[i4];t1=-t1*Math.SQRT1_2;t2*=Math.SQRT1_2;st1=x[i2];x[i4]=t1+st1;x[i3]=t1-st1;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;t1=x[i3]+x[i4];x[i4]-=x[i3];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);cc3=4*cc1*(cc1*cc1-.75);ss3=4*ss1*(.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;t2=x[i7]*cc1-x[i3]*ss1;t1=x[i7]*ss1+x[i3]*cc1;t4=x[i8]*cc3-x[i4]*ss3;t3=x[i8]*ss3+x[i4]*cc3;st1=t2-t4;t2+=t4;t4=st1;x[i8]=t2+x[i6];x[i3]=t2-x[i6];st1=t3-t1;t1+=t3;t3=st1;x[i4]=t3+x[i2];x[i7]=t3-x[i2];x[i6]=x[i1]-t1;x[i1]+=t1;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};module.exports=RFFT},{"./fourier":6}],15:[function(require,module,exports){var DSP=require("./dsp");function Sampler(file,bufferSize,sampleRate,playStart,playEnd,loopStart,loopEnd,loopMode){this.file=file;this.bufferSize=bufferSize;this.sampleRate=sampleRate;this.playStart=playStart||0;this.playEnd=playEnd||1;
+
+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;this.frequency=550;this.step=this.frequency/this.rootFrequency;this.duration=0;this.samplesProcessed=0;this.playhead=0;var 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(){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 loopWidth=this.playEnd*this.samples.length-this.playStart*this.samples.length;var playStartSamples=this.playStart*this.samples.length;var playEndSamples=this.playEnd*this.samples.length;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};module.exports=Sampler},{"./dsp":4}],16:[function(require,module,exports){function SingleDelay(maxDelayInSamplesSize,delayInSamples,delayVolume){this.delayBufferSamples=new Float64Array(maxDelayInSamplesSize);this.delayInputPointer=delayInSamples;this.delayOutputPointer=0;this.delayInSamples=delayInSamples;this.delayVolume=delayVolume}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}};SingleDelay.prototype.setDelayVolume=function(delayVolume){this.delayVolume=delayVolume};SingleDelay.prototype.process=function(samples){var outputSamples=new Float64Array(samples.length);for(var i=0;i<samples.length;i++){this.delayBufferSamples[this.delayInputPointer]=samples[i];var delaySample=this.delayBufferSamples[this.delayOutputPointer];outputSamples[i]=delaySample*this.delayVolume;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};module.exports=SingleDelay},{}],17:[function(require,module,exports){function sinh(arg){return(Math.exp(arg)-Math.exp(-arg))/2}module.exports=sinh},{}],18:[function(require,module,exports){var DSP=require("./dsp");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||.16;break;case DSP.COSINE:this.func=WindowFunction.Cosine;break;case DSP.GAUSS:this.func=WindowFunction.Gauss;this.alpha=this.alpha||.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.62-.48*Math.abs(index/(length-1)-.5)-.38*Math.cos(DSP.TWO_PI*index/(length-1))};WindowFunction.Blackman=function(length,index,alpha){var a0=(1-alpha)/2;var a1=.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,-.5*Math.pow((index-(length-1)/2)/(alpha*(length-1)/2),2))};WindowFunction.Hamming=function(length,index){return.54-.46*Math.cos(DSP.TWO_PI*index/(length-1))};WindowFunction.Hann=function(length,index){return.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(){return 1};WindowFunction.Triangular=function(length,index){return 2/length*(length/2-Math.abs(index-(length-1)/2))};module.exports=WindowFunction},{"./dsp":4}]},{},[10]);
\ No newline at end of file
diff --git a/docs/API.md b/docs/API.md
new file mode 100644
index 0000000..63f0940
--- /dev/null
+++ b/docs/API.md
@@ -0,0 +1,999 @@
+## Classes
+
+<dl>
+<dt><a href="#ADSR">ADSR</a></dt>
+<dd></dd>
+<dt><a href="#Biquad">Biquad</a></dt>
+<dd></dd>
+<dt><a href="#DFT">DFT</a></dt>
+<dd></dd>
+<dt><a href="#FFT">FFT</a></dt>
+<dd></dd>
+<dt><a href="#GraphicalEq">GraphicalEq</a></dt>
+<dd></dd>
+<dt><a href="#IIRFilter">IIRFilter</a></dt>
+<dd></dd>
+<dt><a href="#IIRFilter2">IIRFilter2</a></dt>
+<dd></dd>
+<dt><a href="#MultiDelay">MultiDelay</a></dt>
+<dd></dd>
+<dt><a href="#Oscillator">Oscillator</a></dt>
+<dd></dd>
+<dt><a href="#Reverb">Reverb</a></dt>
+<dd></dd>
+<dt><a href="#RFFT">RFFT</a></dt>
+<dd></dd>
+<dt><a href="#Sampler">Sampler</a></dt>
+<dd></dd>
+<dt><a href="#SingleDelay">SingleDelay</a></dt>
+<dd></dd>
+<dt><a href="#WindowFunction">WindowFunction</a></dt>
+<dd></dd>
+</dl>
+
+## Members
+
+<dl>
+<dt><a href="#DSP">DSP</a></dt>
+<dd><p>DSP is an object which contains general purpose utility functions and constants</p>
+</dd>
+</dl>
+
+## Functions
+
+<dl>
+<dt><a href="#sinh">sinh(num)</a></dt>
+<dd><p>Returns the hyperbolic sine of the number</p>
+</dd>
+</dl>
+
+<a name="ADSR"></a>
+
+## ADSR
+**Kind**: global class
+
+* [ADSR](#ADSR)
+ * [new ADSR(attack, decay, sustain, release, sampleRate)](#new_ADSR_new)
+ * [.noteOn()](#ADSR+noteOn)
+ * [.noteOff()](#ADSR+noteOff)
+ * [.processSample()](#ADSR+processSample)
+ * [.value()](#ADSR+value) ⇒ <code>Number</code>
+ * [.process(buffer)](#ADSR+process)
+ * [.isActive()](#ADSR+isActive) ⇒ <code>Boolean</code>
+ * [.disable()](#ADSR+disable)
+
+<a name="new_ADSR_new"></a>
+
+### new ADSR(attack, decay, sustain, release, sampleRate)
+ADSR Envelope
+
+
+| Param | Type | Description |
+| --- | --- | --- |
+| attack | <code>Number</code> | The attack length in seconds |
+| decay | <code>Number</code> | The decay length in seconds |
+| sustain | <code>Number</code> | The sustain level |
+| release | <code>Number</code> | The release length in seconds |
+| sampleRate | <code>Number</code> | The the sample rate |
+
+<a name="ADSR+noteOn"></a>
+
+### adsR.noteOn()
+Start the envelope
+
+**Kind**: instance method of <code>[ADSR](#ADSR)</code>
+<a name="ADSR+noteOff"></a>
+
+### adsR.noteOff()
+Stop the envelope
+
+Send a note off when using a sustain of infinity to let the envelope enter
+the release phase
+
+**Kind**: instance method of <code>[ADSR](#ADSR)</code>
+<a name="ADSR+processSample"></a>
+
+### adsR.processSample()
+Process sample
+
+**Kind**: instance method of <code>[ADSR](#ADSR)</code>
+<a name="ADSR+value"></a>
+
+### adsR.value() ⇒ <code>Number</code>
+Get current value
+
+**Kind**: instance method of <code>[ADSR](#ADSR)</code>
+**Returns**: <code>Number</code> - amplitude
+<a name="ADSR+process"></a>
+
+### adsR.process(buffer)
+Process a buffer
+
+**Kind**: instance method of <code>[ADSR](#ADSR)</code>
+
+| Param | Type |
+| --- | --- |
+| buffer | <code>Array</code> |
+
+<a name="ADSR+isActive"></a>
+
+### adsR.isActive() ⇒ <code>Boolean</code>
+Test if the envelope is active
+
+**Kind**: instance method of <code>[ADSR](#ADSR)</code>
+<a name="ADSR+disable"></a>
+
+### adsR.disable()
+Disable the envelope
+
+**Kind**: instance method of <code>[ADSR](#ADSR)</code>
+<a name="Biquad"></a>
+
+## Biquad
+**Kind**: global class
+<a name="new_Biquad_new"></a>
+
+### new Biquad(type, sampleRate)
+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
+
+
+| Param | Type |
+| --- | --- |
+| type | <code>Number</code> |
+| sampleRate | <code>Number</code> |
+
+<a name="DFT"></a>
+
+## DFT
+**Kind**: global class
+
+* [DFT](#DFT)
+ * [new DFT(bufferSize, sampleRate)](#new_DFT_new)
+ * [.forward(buffer)](#DFT+forward) ⇒
+
+<a name="new_DFT_new"></a>
+
+### new DFT(bufferSize, sampleRate)
+DFT is a class for calculating the Discrete Fourier Transform of a signal.
+
+
+| Param | Type | Description |
+| --- | --- | --- |
+| bufferSize | <code>Number</code> | The size of the sample buffer to be computed |
+| sampleRate | <code>Number</code> | The sampleRate of the buffer (eg. 44100) |
+
+<a name="DFT+forward"></a>
+
+### dfT.forward(buffer) ⇒
+Performs a forward transform on the sample buffer.
+Converts a time domain signal to frequency domain spectra.
+
+**Kind**: instance method of <code>[DFT](#DFT)</code>
+**Returns**: The frequency spectrum array
+
+| Param | Type | Description |
+| --- | --- | --- |
+| buffer | <code>Array</code> | The sample buffer |
+
+<a name="FFT"></a>
+
+## FFT
+**Kind**: global class
+
+* [FFT](#FFT)
+ * [new FFT(bufferSize, sampleRate)](#new_FFT_new)
+ * [.forward(buffer)](#FFT+forward) ⇒
+ * [.inverse(real, imag)](#FFT+inverse) ⇒
+
+<a name="new_FFT_new"></a>
+
+### new FFT(bufferSize, sampleRate)
+FFT is a class for calculating the Discrete Fourier Transform of a signal
+with the Fast Fourier Transform algorithm.
+
+
+| Param | Type | Description |
+| --- | --- | --- |
+| bufferSize | <code>Number</code> | The size of the sample buffer to be computed. Must be power of 2 |
+| sampleRate | <code>Number</code> | The sampleRate of the buffer (eg. 44100) |
+
+<a name="FFT+forward"></a>
+
+### ffT.forward(buffer) ⇒
+Performs a forward transform on the sample buffer.
+Converts a time domain signal to frequency domain spectra.
+
+**Kind**: instance method of <code>[FFT](#FFT)</code>
+**Returns**: The frequency spectrum array
+
+| Param | Type | Description |
+| --- | --- | --- |
+| buffer | <code>Array</code> | The sample buffer. Buffer Length must be power of 2 |
+
+<a name="FFT+inverse"></a>
+
+### ffT.inverse(real, imag) ⇒
+Performs a inverse FFT transformation
+Converts a frequency domain spectra to a time domain signal
+
+**Kind**: instance method of <code>[FFT](#FFT)</code>
+**Returns**: The time domain signal
+
+| Param | Type |
+| --- | --- |
+| real | <code>Array</code> |
+| imag | <code>Array</code> |
+
+<a name="GraphicalEq"></a>
+
+## GraphicalEq
+**Kind**: global class
+<a name="new_GraphicalEq_new"></a>
+
+### new GraphicalEq(sampleRate)
+Create a 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.
+
+
+| Param | Type |
+| --- | --- |
+| sampleRate | <code>SampleRate</code> |
+
+**Example**
+```js
+var eq = new GraphicalEq(44100)
+```
+<a name="IIRFilter"></a>
+
+## IIRFilter
+**Kind**: global class
+
+* [IIRFilter](#IIRFilter)
+ * [new IIRFilter()](#new_IIRFilter_new)
+ * _instance_
+ * [.set(cutoff, resonance)](#IIRFilter+set)
+ * [.process(buffer)](#IIRFilter+process)
+ * [.addEnvelope(envelope)](#IIRFilter+addEnvelope)
+ * _static_
+ * [.LP12](#IIRFilter.LP12)
+ * [new IIRFilter.LP12()](#new_IIRFilter.LP12_new)
+ * [.addEnvelope(envelope)](#IIRFilter.LP12+addEnvelope)
+
+<a name="new_IIRFilter_new"></a>
+
+### new IIRFilter()
+IIRFilter
+
+<a name="IIRFilter+set"></a>
+
+### iirFilter.set(cutoff, resonance)
+Set filter parameters
+
+**Kind**: instance method of <code>[IIRFilter](#IIRFilter)</code>
+
+| Param | Type |
+| --- | --- |
+| cutoff | <code>Number</code> |
+| resonance | <code>Number</code> |
+
+<a name="IIRFilter+process"></a>
+
+### iirFilter.process(buffer)
+Process a buffer
+
+**Kind**: instance method of <code>[IIRFilter](#IIRFilter)</code>
+
+| Param | Type |
+| --- | --- |
+| buffer | <code>Array</code> |
+
+<a name="IIRFilter+addEnvelope"></a>
+
+### iirFilter.addEnvelope(envelope)
+Add an envelope to the filter
+
+**Kind**: instance method of <code>[IIRFilter](#IIRFilter)</code>
+
+| Param | Type |
+| --- | --- |
+| envelope | <code>[ADSR](#ADSR)</code> |
+
+<a name="IIRFilter.LP12"></a>
+
+### IIRFilter.LP12
+**Kind**: static class of <code>[IIRFilter](#IIRFilter)</code>
+
+* [.LP12](#IIRFilter.LP12)
+ * [new IIRFilter.LP12()](#new_IIRFilter.LP12_new)
+ * [.addEnvelope(envelope)](#IIRFilter.LP12+addEnvelope)
+
+<a name="new_IIRFilter.LP12_new"></a>
+
+#### new IIRFilter.LP12()
+LP12 filter
+
+<a name="IIRFilter.LP12+addEnvelope"></a>
+
+#### lP12.addEnvelope(envelope)
+Add an envelope to the filter
+
+**Kind**: instance method of <code>[LP12](#IIRFilter.LP12)</code>
+
+| Param | Type |
+| --- | --- |
+| envelope | <code>[ADSR](#ADSR)</code> |
+
+<a name="IIRFilter2"></a>
+
+## IIRFilter2
+**Kind**: global class
+
+* [IIRFilter2](#IIRFilter2)
+ * [new IIRFilter2(type, cutoff, resonance, sampleRate)](#new_IIRFilter2_new)
+ * [.process(buffer)](#IIRFilter2+process)
+ * [.addEnvelope(envelope)](#IIRFilter2+addEnvelope)
+ * [.set(cutoff, resonance)](#IIRFilter2+set)
+
+<a name="new_IIRFilter2_new"></a>
+
+### new IIRFilter2(type, cutoff, resonance, sampleRate)
+IIRFilter2
+
+
+| Param | Type |
+| --- | --- |
+| type | <code>Number</code> |
+| cutoff | <code>Number</code> |
+| resonance | <code>Number</code> |
+| sampleRate | <code>Number</code> |
+
+<a name="IIRFilter2+process"></a>
+
+### iirFilter2.process(buffer)
+Process a buffer
+
+**Kind**: instance method of <code>[IIRFilter2](#IIRFilter2)</code>
+
+| Param | Type |
+| --- | --- |
+| buffer | <code>Array</code> |
+
+<a name="IIRFilter2+addEnvelope"></a>
+
+### iirFilter2.addEnvelope(envelope)
+Add an envelope to the filter
+
+**Kind**: instance method of <code>[IIRFilter2](#IIRFilter2)</code>
+
+| Param | Type |
+| --- | --- |
+| envelope | <code>[ADSR](#ADSR)</code> |
+
+<a name="IIRFilter2+set"></a>
+
+### iirFilter2.set(cutoff, resonance)
+Set filter parameters
+
+**Kind**: instance method of <code>[IIRFilter2](#IIRFilter2)</code>
+
+| Param | Type |
+| --- | --- |
+| cutoff | <code>Number</code> |
+| resonance | <code>Number</code> |
+
+<a name="MultiDelay"></a>
+
+## MultiDelay
+**Kind**: global class
+
+* [MultiDelay](#MultiDelay)
+ * [new MultiDelay(maxDelayInSamplesSize, delayInSamples, masterVolume, delayVolume)](#new_MultiDelay_new)
+ * [.setDelayInSamples(delayInSamples)](#MultiDelay+setDelayInSamples)
+ * [.setMasterVolume(masterVolume)](#MultiDelay+setMasterVolume)
+ * [.setDelayVolume(delayVolume)](#MultiDelay+setDelayVolume)
+ * [.process(samples)](#MultiDelay+process) ⇒
+
+<a name="new_MultiDelay_new"></a>
+
+### new MultiDelay(maxDelayInSamplesSize, delayInSamples, masterVolume, delayVolume)
+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 | Type | Description |
+| --- | --- | --- |
+| maxDelayInSamplesSize | <code>Number</code> | Maximum possible delay in samples (size of circular buffer) |
+| delayInSamples | <code>Number</code> | Initial delay in samples |
+| masterVolume | <code>Number</code> | Initial master volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify) |
+| delayVolume | <code>Number</code> | Initial feedback delay volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify) |
+
+<a name="MultiDelay+setDelayInSamples"></a>
+
+### multiDelay.setDelayInSamples(delayInSamples)
+Change the delay time in samples.
+
+**Kind**: instance method of <code>[MultiDelay](#MultiDelay)</code>
+
+| Param | Type | Description |
+| --- | --- | --- |
+| delayInSamples | <code>Number</code> | Delay in samples |
+
+<a name="MultiDelay+setMasterVolume"></a>
+
+### multiDelay.setMasterVolume(masterVolume)
+Change the master volume.
+
+**Kind**: instance method of <code>[MultiDelay](#MultiDelay)</code>
+
+| Param | Type | Description |
+| --- | --- | --- |
+| masterVolume | <code>Number</code> | Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify) |
+
+<a name="MultiDelay+setDelayVolume"></a>
+
+### multiDelay.setDelayVolume(delayVolume)
+Change the delay feedback volume.
+
+**Kind**: instance method of <code>[MultiDelay](#MultiDelay)</code>
+
+| Param | Type | Description |
+| --- | --- | --- |
+| delayVolume | <code>Number</code> | Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify) |
+
+<a name="MultiDelay+process"></a>
+
+### multiDelay.process(samples) ⇒
+Process a given interleaved or mono non-interleaved float value Array and adds the delayed audio.
+
+**Kind**: instance method of <code>[MultiDelay](#MultiDelay)</code>
+**Returns**: A new Float64Array interleaved or mono non-interleaved as was fed to this function.
+
+| Param | Type | Description |
+| --- | --- | --- |
+| samples | <code>Array</code> | Array containing Float values or a Float64Array |
+
+<a name="Oscillator"></a>
+
+## Oscillator
+**Kind**: global class
+
+* [Oscillator](#Oscillator)
+ * [new Oscillator(type, frequency, amplitude, bufferSize, sampleRate)](#new_Oscillator_new)
+ * [.setAmp(amplitude)](#Oscillator+setAmp)
+ * [.setFreq(frequency)](#Oscillator+setFreq)
+ * [.add(oscillator)](#Oscillator+add) ⇒ <code>Array</code>
+ * [.addSignal(signal)](#Oscillator+addSignal)
+ * [.addEnvelope(envelope)](#Oscillator+addEnvelope)
+ * [.applyEnvelope()](#Oscillator+applyEnvelope)
+ * [.valueAt(offset)](#Oscillator+valueAt)
+ * [.generate()](#Oscillator+generate) ⇒ <code>Array</code>
+
+<a name="new_Oscillator_new"></a>
+
+### new Oscillator(type, frequency, amplitude, bufferSize, sampleRate)
+Oscillator class for generating and modifying signals
+
+
+| Param | Type | Description |
+| --- | --- | --- |
+| type | <code>Number</code> | A waveform constant (eg. DSP.SINE) |
+| frequency | <code>Number</code> | Initial frequency of the signal |
+| amplitude | <code>Number</code> | Initial amplitude of the signal |
+| bufferSize | <code>Number</code> | Size of the sample buffer to generate |
+| sampleRate | <code>Number</code> | The sample rate of the signal |
+
+<a name="Oscillator+setAmp"></a>
+
+### oscillator.setAmp(amplitude)
+Set the amplitude of the signal
+
+**Kind**: instance method of <code>[Oscillator](#Oscillator)</code>
+
+| Param | Type | Description |
+| --- | --- | --- |
+| amplitude | <code>Number</code> | The amplitude of the signal (between 0 and 1) |
+
+<a name="Oscillator+setFreq"></a>
+
+### oscillator.setFreq(frequency)
+Set the frequency of the signal
+
+**Kind**: instance method of <code>[Oscillator](#Oscillator)</code>
+
+| Param | Type | Description |
+| --- | --- | --- |
+| frequency | <code>Number</code> | The frequency of the signal |
+
+<a name="Oscillator+add"></a>
+
+### oscillator.add(oscillator) ⇒ <code>Array</code>
+Add an oscillator
+
+**Kind**: instance method of <code>[Oscillator](#Oscillator)</code>
+**Returns**: <code>Array</code> - the current oscillator signal
+
+| Param | Type | Description |
+| --- | --- | --- |
+| oscillator | <code>[Oscillator](#Oscillator)</code> | The oscillator to be added to |
+
+<a name="Oscillator+addSignal"></a>
+
+### oscillator.addSignal(signal)
+Add a signal to the current generated osc signal
+
+**Kind**: instance method of <code>[Oscillator](#Oscillator)</code>
+
+| Param | Type |
+| --- | --- |
+| signal | <code>Array</code> |
+
+<a name="Oscillator+addEnvelope"></a>
+
+### oscillator.addEnvelope(envelope)
+Add an envelope to the oscillator
+
+**Kind**: instance method of <code>[Oscillator](#Oscillator)</code>
+
+| Param | Type |
+| --- | --- |
+| envelope | <code>[ADSR](#ADSR)</code> |
+
+<a name="Oscillator+applyEnvelope"></a>
+
+### oscillator.applyEnvelope()
+Apply the oscillator envelope to its signal
+
+**Kind**: instance method of <code>[Oscillator](#Oscillator)</code>
+<a name="Oscillator+valueAt"></a>
+
+### oscillator.valueAt(offset)
+Get value
+
+**Kind**: instance method of <code>[Oscillator](#Oscillator)</code>
+
+| Param | Type |
+| --- | --- |
+| offset | <code>Number</code> |
+
+<a name="Oscillator+generate"></a>
+
+### oscillator.generate() ⇒ <code>Array</code>
+Generate the oscillator signal
+
+**Kind**: instance method of <code>[Oscillator](#Oscillator)</code>
+**Returns**: <code>Array</code> - the signal
+<a name="Reverb"></a>
+
+## Reverb
+**Kind**: global class
+
+* [Reverb](#Reverb)
+ * [new Reverb(maxDelayInSamplesSize, delayInSamples, masterVolume, mixVolume, delayVolume, dampFrequency)](#new_Reverb_new)
+ * [.setDelayInSamples(delayInSamples)](#Reverb+setDelayInSamples)
+ * [.setMasterVolume(masterVolume)](#Reverb+setMasterVolume)
+ * [.setMixVolume(mixVolume)](#Reverb+setMixVolume)
+ * [.setDelayVolume(delayVolume)](#Reverb+setDelayVolume)
+ * [.setDampFrequency(dampFrequency)](#Reverb+setDampFrequency)
+ * [.process(samples)](#Reverb+process) ⇒
+
+<a name="new_Reverb_new"></a>
+
+### new Reverb(maxDelayInSamplesSize, delayInSamples, masterVolume, mixVolume, delayVolume, dampFrequency)
+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 | Type | Description |
+| --- | --- | --- |
+| maxDelayInSamplesSize | <code>Number</code> | Maximum possible delay in samples (size of circular buffers) |
+| delayInSamples | <code>Number</code> | Initial delay in samples for internal (Single/Multi)delays |
+| masterVolume | <code>Number</code> | Initial master volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify) |
+| mixVolume | <code>Number</code> | Initial reverb signal mix volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify) |
+| delayVolume | <code>Number</code> | Initial feedback delay volume for internal (Single/Multi)delays. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify) |
+| dampFrequency | <code>Number</code> | Initial low pass filter frequency. 0 to 44100 (depending on your maximum sampling frequency) |
+
+<a name="Reverb+setDelayInSamples"></a>
+
+### reverb.setDelayInSamples(delayInSamples)
+Change the delay time in samples as a base for all delays.
+
+**Kind**: instance method of <code>[Reverb](#Reverb)</code>
+
+| Param | Type | Description |
+| --- | --- | --- |
+| delayInSamples | <code>Number</code> | Delay in samples |
+
+<a name="Reverb+setMasterVolume"></a>
+
+### reverb.setMasterVolume(masterVolume)
+Change the master volume.
+
+**Kind**: instance method of <code>[Reverb](#Reverb)</code>
+
+| Param | Type | Description |
+| --- | --- | --- |
+| masterVolume | <code>Number</code> | Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify) |
+
+<a name="Reverb+setMixVolume"></a>
+
+### reverb.setMixVolume(mixVolume)
+Change the reverb signal mix level.
+
+**Kind**: instance method of <code>[Reverb](#Reverb)</code>
+
+| Param | Type | Description |
+| --- | --- | --- |
+| mixVolume | <code>Number</code> | Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify) |
+
+<a name="Reverb+setDelayVolume"></a>
+
+### reverb.setDelayVolume(delayVolume)
+Change all delays feedback volume.
+
+**Kind**: instance method of <code>[Reverb](#Reverb)</code>
+
+| Param | Type | Description |
+| --- | --- | --- |
+| delayVolume | <code>Number</code> | Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify) |
+
+<a name="Reverb+setDampFrequency"></a>
+
+### reverb.setDampFrequency(dampFrequency)
+Change the Low Pass filter frequency.
+
+**Kind**: instance method of <code>[Reverb](#Reverb)</code>
+
+| Param | Type | Description |
+| --- | --- | --- |
+| dampFrequency | <code>Number</code> | low pass filter frequency. 0 to 44100 (depending on your maximum sampling frequency) |
+
+<a name="Reverb+process"></a>
+
+### reverb.process(samples) ⇒
+Process a given interleaved float value Array and copies and adds the reverb signal.
+
+**Kind**: instance method of <code>[Reverb](#Reverb)</code>
+**Returns**: A new Float64Array interleaved buffer.
+
+| Param | Type | Description |
+| --- | --- | --- |
+| samples | <code>Array</code> | Array containing Float values or a Float64Array |
+
+<a name="RFFT"></a>
+
+## RFFT
+**Kind**: global class
+
+* [RFFT](#RFFT)
+ * [new RFFT(bufferSize, sampleRate)](#new_RFFT_new)
+ * [.forward(buffer)](#RFFT+forward) ⇒
+
+<a name="new_RFFT_new"></a>
+
+### new RFFT(bufferSize, sampleRate)
+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 | Type | Description |
+| --- | --- | --- |
+| bufferSize | <code>Number</code> | The size of the sample buffer to be computed. Must be power of 2 |
+| sampleRate | <code>Number</code> | The sampleRate of the buffer (eg. 44100) |
+
+<a name="RFFT+forward"></a>
+
+### rffT.forward(buffer) ⇒
+Performs a forward transform on the sample buffer.
+Converts a time domain signal to frequency domain spectra.
+
+**Kind**: instance method of <code>[RFFT](#RFFT)</code>
+**Returns**: The frequency spectrum array
+
+| Param | Type | Description |
+| --- | --- | --- |
+| buffer | <code>Array</code> | The sample buffer. Buffer Length must be power of 2 |
+
+<a name="Sampler"></a>
+
+## Sampler
+**Kind**: global class
+<a name="new_Sampler_new"></a>
+
+### new Sampler(file, bufferSize, sampleRate, playStart, playEnd, loopStart, loopEnd, loopMode)
+Sampler
+
+
+| Param | Type |
+| --- | --- |
+| file | |
+| bufferSize | <code>Number</code> |
+| sampleRate | <code>Number</code> |
+| playStart | <code>Number</code> |
+| playEnd | <code>Number</code> |
+| loopStart | <code>Number</code> |
+| loopEnd | <code>Number</code> |
+| loopMode | <code>Number</code> |
+
+<a name="SingleDelay"></a>
+
+## SingleDelay
+**Kind**: global class
+
+* [SingleDelay](#SingleDelay)
+ * [new SingleDelay(maxDelayInSamplesSize, delayInSamples, delayVolume)](#new_SingleDelay_new)
+ * [.setDelayInSamples(delayInSamples)](#SingleDelay+setDelayInSamples)
+ * [.setDelayVolume(delayVolume)](#SingleDelay+setDelayVolume)
+ * [.process(samples)](#SingleDelay+process) ⇒
+
+<a name="new_SingleDelay_new"></a>
+
+### new SingleDelay(maxDelayInSamplesSize, delayInSamples, delayVolume)
+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 | Type | Description |
+| --- | --- | --- |
+| maxDelayInSamplesSize | <code>Number</code> | Maximum possible delay in samples (size of circular buffer) |
+| delayInSamples | <code>Number</code> | Initial delay in samples |
+| delayVolume | <code>Number</code> | Initial feedback delay volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify) |
+
+<a name="SingleDelay+setDelayInSamples"></a>
+
+### singleDelay.setDelayInSamples(delayInSamples)
+Change the delay time in samples.
+
+**Kind**: instance method of <code>[SingleDelay](#SingleDelay)</code>
+
+| Param | Type | Description |
+| --- | --- | --- |
+| delayInSamples | <code>Number</code> | Delay in samples |
+
+<a name="SingleDelay+setDelayVolume"></a>
+
+### singleDelay.setDelayVolume(delayVolume)
+Change the return signal volume.
+
+**Kind**: instance method of <code>[SingleDelay](#SingleDelay)</code>
+
+| Param | Type | Description |
+| --- | --- | --- |
+| delayVolume | <code>Number</code> | Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify) |
+
+<a name="SingleDelay+process"></a>
+
+### singleDelay.process(samples) ⇒
+Process a given interleaved or mono non-interleaved float value Array and
+returns the delayed audio.
+
+**Kind**: instance method of <code>[SingleDelay](#SingleDelay)</code>
+**Returns**: A new Float64Array interleaved or mono non-interleaved as was fed to this function.
+
+| Param | Type | Description |
+| --- | --- | --- |
+| samples | <code>Array</code> | Array containing Float values or a Float64Array |
+
+<a name="WindowFunction"></a>
+
+## WindowFunction
+**Kind**: global class
+
+* [WindowFunction](#WindowFunction)
+ * [new WindowFunction(type, alpha)](#new_WindowFunction_new)
+ * [.process(buffer)](#WindowFunction+process)
+
+<a name="new_WindowFunction_new"></a>
+
+### new WindowFunction(type, alpha)
+WindowFunction
+
+
+| Param | Type |
+| --- | --- |
+| type | <code>Number</code> |
+| alpha | <code>Number</code> |
+
+<a name="WindowFunction+process"></a>
+
+### windowFunction.process(buffer)
+Process a buffer
+
+**Kind**: instance method of <code>[WindowFunction](#WindowFunction)</code>
+
+| Param | Type |
+| --- | --- |
+| buffer | <code>Array</code> |
+
+<a name="DSP"></a>
+
+## DSP
+DSP is an object which contains general purpose utility functions and constants
+
+**Kind**: global variable
+
+* [DSP](#DSP)
+ * [.deinterleave](#DSP.deinterleave) ⇒
+ * [.getChannel](#DSP.getChannel) ⇒
+ * [.invert(buffer)](#DSP.invert) ⇒
+ * [.interleave(left, right)](#DSP.interleave) ⇒
+ * [.mixSampleBuffers(sampleBuffer1, sampleBuffer2, negate, volumeCorrection)](#DSP.mixSampleBuffers) ⇒
+ * [.RMS(buffer)](#DSP.RMS)
+ * [.Peak(buffer)](#DSP.Peak)
+ * [.mag2db(@buffer)](#DSP.mag2db) ⇒
+ * [.freqz(b, a, w)](#DSP.freqz) ⇒
+
+<a name="DSP.deinterleave"></a>
+
+### DSP.deinterleave ⇒
+Converts a stereo-interleaved sample buffer into split-stereo (dual mono) sample buffers
+
+**Kind**: static property of <code>[DSP](#DSP)</code>
+**Returns**: an Array containing left and right channels
+
+| Param | Type | Description |
+| --- | --- | --- |
+| buffer | <code>Array</code> | A stereo-interleaved sample buffer |
+
+<a name="DSP.getChannel"></a>
+
+### DSP.getChannel ⇒
+Separates a channel from a stereo-interleaved sample buffer
+
+**Kind**: static property of <code>[DSP](#DSP)</code>
+**Returns**: an Array containing a signal mono sample buffer
+
+| Param | Type | Description |
+| --- | --- | --- |
+| buffer | <code>Array</code> | A stereo-interleaved sample buffer |
+| channel | <code>Number</code> | A channel constant (LEFT, RIGHT, MIX) |
+
+<a name="DSP.invert"></a>
+
+### DSP.invert(buffer) ⇒
+Inverts the phase of a signal
+
+**Kind**: static method of <code>[DSP](#DSP)</code>
+**Returns**: The inverted sample buffer
+
+| Param | Type | Description |
+| --- | --- | --- |
+| buffer | <code>Array</code> | A sample buffer |
+
+<a name="DSP.interleave"></a>
+
+### DSP.interleave(left, right) ⇒
+Converts split-stereo (dual mono) sample buffers into a stereo interleaved sample buffer
+
+**Kind**: static method of <code>[DSP](#DSP)</code>
+**Returns**: The stereo interleaved buffer
+
+| Param | Type | Description |
+| --- | --- | --- |
+| left | <code>Array</code> | A sample buffer |
+| right | <code>Array</code> | A sample buffer |
+
+<a name="DSP.mixSampleBuffers"></a>
+
+### DSP.mixSampleBuffers(sampleBuffer1, sampleBuffer2, negate, volumeCorrection) ⇒
+Helper method (for Reverb) to mix two (interleaved) samplebuffers. It's possible
+to negate the second buffer while mixing and to perform a volume correction
+on the final signal.
+
+**Kind**: static method of <code>[DSP](#DSP)</code>
+**Returns**: A new Float64Array interleaved buffer.
+
+| Param | Type | Description |
+| --- | --- | --- |
+| sampleBuffer1 | <code>Array</code> | Array containing Float values or a Float64Array |
+| sampleBuffer2 | <code>Array</code> | Array containing Float values or a Float64Array |
+| negate | <code>Boolean</code> | When true inverts/flips the audio signal |
+| volumeCorrection | <code>Number</code> | When you add multiple sample buffers, use this to tame your signal ;) |
+
+<a name="DSP.RMS"></a>
+
+### DSP.RMS(buffer)
+Find RMS of signal
+
+**Kind**: static method of <code>[DSP](#DSP)</code>
+
+| Param | Type |
+| --- | --- |
+| buffer | <code>Array</code> |
+
+<a name="DSP.Peak"></a>
+
+### DSP.Peak(buffer)
+Find Peak of signal
+
+**Kind**: static method of <code>[DSP](#DSP)</code>
+
+| Param | Type |
+| --- | --- |
+| buffer | <code>Array</code> |
+
+<a name="DSP.mag2db"></a>
+
+### DSP.mag2db(@buffer) ⇒
+Magnitude to decibels
+
+Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
+Copyright 2010 Ricard Marxer. All rights reserved.
+
+**Kind**: static method of <code>[DSP](#DSP)</code>
+**Returns**: the array in decibels
+
+| Param | Type | Description |
+| --- | --- | --- |
+| @buffer | <code>Array</code> | The array of magnitudes to convert to decibels |
+
+<a name="DSP.freqz"></a>
+
+### DSP.freqz(b, a, w) ⇒
+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.
+
+**Kind**: static method of <code>[DSP](#DSP)</code>
+**Returns**: the frequency response in magnitude
+
+| Param | Type | Description |
+| --- | --- | --- |
+| b | <code>Number</code> | The coefficients of the filter |
+| a | <code>Number</code> | The coefficients of the filter |
+| w | <code>Number</code> | The points (normally between -PI and PI) where to calculate the frequency response |
+
+<a name="sinh"></a>
+
+## sinh(num)
+Returns the hyperbolic sine of the number
+
+**Kind**: global function
+**Meta**: version: 1004.2314
+**Meta**: discuss at: http://phpjs.org/functions/sinh
+**Meta**: original by: Onno Marsman
+
+| Param | Type |
+| --- | --- |
+| num | <code>Number</code> |
+
+**Example**
+```js
+sinh(-0.9834330348825909); // => -1.1497971402636502
+```
diff --git a/docs/README.md b/docs/README.md
new file mode 100644
index 0000000..e3a60ce
--- /dev/null
+++ b/docs/README.md
@@ -0,0 +1,51 @@
+## DSP.js [](https://www.npmjs.com/package/dsp.js)
+
+DSP.js is a comprehensive digital signal processing library for javascript.
+It includes many functions for signal analysis and generation, including
+Oscillators(sine, saw, square, triangle), Window functions (Hann, Hamming, etc),
+Envelopes(ADSR), IIR Filters(lowpass, highpass, bandpass, notch), FFT and DFT
+transforms, Delays, Reverb.
+
+## Install
+
+__Node__
+
+Via npm: `npm install --save dsp.js` and require the modules:
+
+```js
+const { DSP, WindowFunction, Oscillator } = require('dsp.js')
+```
+
+You can require individual modules:
+
+```js
+const DSP = require('dsp.js/lib/dsp')
+const WindowFunction = require('dsp.js/lib/window-function')
+```
+
+__Browser__
+
+Or grab the [dsp.js](https://github.com/corbanbrook/dsp.js/blob/master/dist/dsp.min.js) and include it in your html:
+
+```html
+<script src="dsp.min.js"></script>
+```
+
+## Usage
+
+[Read the API documentation](https://github.com/corbanbrook/dsp.js/blob/master/docs/API.md)
+
+## Tests and scripts
+
+To setup you local machine to work with the code, first you have to clone this repository and install dependencies (`node` and `npm` are assumed to be installed): `npm install`
+
+- Run the tests: `npm test`
+- Generate API reference and distribution files: `npm run dist`
+- Run benchmarks: `npm run bench`
+
+
+## License
+
+MIT License
+
+Copyright (c) 2010 Corban Brook @corban
diff --git a/examples/biquad.html b/examples/biquad.html
index 754f017..f827564 100644
--- a/examples/biquad.html
+++ b/examples/biquad.html
@@ -11,7 +11,7 @@
<script language="javascript" src="js/init.js"></script>
<!-- Load DSP.js -->
- <script src="../dsp.js"></script>
+ <script src="../dist/dsp.js"></script>
<script>
$(function() {
$('#freq').slider({ orientation: 'vertical', range: 'min', min: 60, max: 10000, step: 1, value: 880, slide: changeF0 });
diff --git a/examples/fft.html b/examples/fft.html
index 5a35922..a6074cd 100644
--- a/examples/fft.html
+++ b/examples/fft.html
@@ -1,21 +1,21 @@
<html>
- <head>
+ <head>
<script language="javascript" src="js/processing.js"></script>
<script language="javascript" src="js/init.js"></script>
- <script language="javascript" src="../dsp.js"></script>
+ <script language="javascript" src="../dist/dsp.js"></script>
</head>
<body>
<style>
body { background: black; margin:0; padding:0;}
</style>
-
+
<script>
var frameBufferSize = 4096;
var bufferSize = frameBufferSize/2;
-
+
var signal = new Float32Array(bufferSize);
var peak = new Float32Array(bufferSize);
-
+
var fft = new FFT(bufferSize, 44100);
function loadedMetaData(event) {
@@ -30,11 +30,11 @@
// perform forward transform
fft.forward(signal);
-
+
// calculate peak values
for ( var i = 0; i < bufferSize; i++ ) {
fft.spectrum[i] *= -1 * Math.log((fft.bufferSize/2 - i) * (0.5/fft.bufferSize/2)) * fft.bufferSize; // equalize, attenuates low freqs and boosts highs
-
+
if ( peak[i] < fft.spectrum[i] ) {
peak[i] = fft.spectrum[i];
} else {
@@ -43,7 +43,7 @@
}
}
</script>
-
+
<script target="#fft" type="application/processing">
void setup() {
size(1400, 600);
@@ -52,19 +52,19 @@
frameRate(60);
//strokeCap(SQUARE);
}
-
+
void draw() {
background(0);
//fill(0, 60);
//rect(0, 0, width, height);
- for ( int i = 0; i < fft.spectrum.length/2; i += 3 ) {
+ for ( int i = 0; i < fft.spectrum.length/2; i += 3 ) {
if (3 * i > width) { break; }
var magnitude = fft.spectrum[i];
-
+
// draw magnitudes
stroke((i) % 360, 60, constrain(magnitude * 6, 20, 100));
line(3*i, height, 3*i, height - magnitude * 16);
-
+
// draw peak indicators
stroke((i) % 360, 60, constrain(magnitude * 100, 50, 100));
line(3*i, height - peak[i] * 16 - 1, 3*i, height - peak[i] * 16);
@@ -74,11 +74,11 @@
println(FRAME_RATE);
}
*/
- }
+ }
</script>
-
+
<audio id="input" src="audio/corban-peddle.ogg" controls="true" onloadedmetadata="loadedMetaData(event);"></audio><br />
-
+
<div><canvas id="fft"></canvas></div>
</body>
</html>
diff --git a/examples/filter.html b/examples/filter.html
index 9b175c1..334d142 100644
--- a/examples/filter.html
+++ b/examples/filter.html
@@ -3,13 +3,13 @@
<head>
<!-- Load JQuery and JQuery-UI -->
<script type="text/javascript" src="js/jquery-1.4.2.min.js"></script>
-
+
<!-- Load Processing.js -->
<script language="javascript" src="js/processing.js"></script>
<script language="javascript" src="js/init.js"></script>
<!-- Load DSP.js -->
- <script src="../dsp.js"></script>
+ <script src="../dist/dsp.js"></script>
<style type="text/css">
body, * {
@@ -45,7 +45,7 @@
margin-bottom: 16px;
width: 8px;
}
-
+
.ui-slider .ui-slider-handle {
width: 8px;
margin-left: 3px;
@@ -57,7 +57,7 @@
// Setup shared variables
var sampleRate = 44100;
var frameSize = 4096;
-
+
var signal = new Float32Array(frameSize/2);
var lp12;
var buffer = [];
@@ -65,7 +65,7 @@
// Setup experimental audio out
var output = new Audio();
-
+
if ( typeof output.mozSetup === 'function' ) {
output.mozSetup(1, sampleRate);
}
@@ -86,7 +86,7 @@
output.mozWriteAudio(signal);
}
</script>
-
+
<script type="application/processing" target="#signal">
Knob cut;
Knob res;
@@ -94,8 +94,8 @@
void setup() {
size(512, 200);
- cut = new Knob("", 60, 6000, 2500, 70, 20, 50, 50);
- res = new Knob("", 1, 20, 7, 0.3, 20, 100, 50);
+ cut = new Knob("", 60, 6000, 2500, 70, 20, 50, 50);
+ res = new Knob("", 1, 20, 7, 0.3, 20, 100, 50);
lp12 = new IIRFilter(DSP.LOWPASS, 22050, 0, sampleRate);
@@ -104,7 +104,7 @@
stroke(255, 0, 0);
}
-
+
void draw() {
background(255);
lp12.set(cut.value, res.value);
@@ -165,7 +165,7 @@
this.active = false;
Knob.active = false;
}
-
+
this.percent = (this.value - this.min) / (this.max - this.min);
this.angle = map(this.value, this.min, this.max, radians(-130), radians(130));
pushMatrix();
@@ -182,7 +182,7 @@
}
boolean isOver() {
- if (mouseX > this.x - this.radius &&
+ if (mouseX > this.x - this.radius &&
mouseX < this.x + this.radius &&
mouseY > this.y - this.radius &&
mouseY < this.y + this.radius) {
@@ -193,10 +193,10 @@
}
</script>
-
+
<h1>Audio Filter</h1>
<p>Applies a Low pass filter to an audio stream. Remember to <b>Mute</b> the audio to hear the pure filtered sound.</p>
- <p>The low pass filter acts as a gate, only letting frequencies below the cut off value through.</p>
+ <p>The low pass filter acts as a gate, only letting frequencies below the cut off value through.</p>
<audio id='input' tabindex="0" src="audio/megaman.ogg" controls="true" onloadedmetadata="loadedMetaData(event);" style="width: 512px;"></audio><br>
<div><canvas id="signal" width="512px" height="200px"></canvas></div>
diff --git a/examples/grapheq.html b/examples/grapheq.html
index ecc7e92..694ff79 100644
--- a/examples/grapheq.html
+++ b/examples/grapheq.html
@@ -11,7 +11,7 @@
<script language="javascript" src="js/init.js"></script>
<!-- Load DSP.js -->
- <script src="../dsp.js"></script>
+ <script src="../dist/dsp.js"></script>
<script>
$(function() {
$('#minFreq').slider({ orientation: 'vertical', range: 'min', min: 10, max: 22000, step: 1, value: 40, slide: changeMinFreq });
diff --git a/examples/nowave.html b/examples/nowave.html
index 48f3dca..da4c8d7 100644
--- a/examples/nowave.html
+++ b/examples/nowave.html
@@ -1,8 +1,8 @@
<html>
- <head>
+ <head>
<script language="javascript" src="js/processing.js"></script>
<script language="javascript" src="js/init.js"></script>
- <script language="javascript" src="../dsp.js"></script>
+ <script language="javascript" src="../dist/dsp.js"></script>
</head>
<body>
<script>
@@ -21,13 +21,13 @@
output.mozSetup(1, sampleRate, 1);
}
- var audioWriter = function(s) {
+ var audioWriter = function(s) {
if ( typeof output.mozWriteAudio === 'function' ) {
output.mozWriteAudio(s);
}
}
</script>
-
+
<script target="#signal" type="application/processing">
int nthHarmonic = 1;
float frequency = 200;
@@ -35,7 +35,7 @@
void setup() {
size(1024, 600);
frameRate(60);
-
+
sine = new Oscillator(Oscillator.Sine, frequency, 1, bufferSize, sampleRate);
sine.generate();
lp12 = new IIRFilter(DSP.LOWPASS, 400, 1, sampleRate);
@@ -52,7 +52,7 @@
rect(-10, -10, width + 20, height + 20);
- if (nthHarmonic > 40 )
+ if (nthHarmonic > 40 )
{
sizes = [2048, 1024, 512, 256, 128, 512, 256, 128];
frequency = constrain(random(200), 10, 200);
@@ -65,19 +65,19 @@
frequency = constrain((mouseX/50) * (mouseX/50), 10, 1000);
-
+
// Add harmonic
if ( nthHarmonic > 1 ) {
- harmonic = new Oscillator(Oscillator.Sine, frequency*nthHarmonic, 1/nthHarmonic, bufferSize, sampleRate);
+ harmonic = new Oscillator(Oscillator.Sine, frequency*nthHarmonic, 1/nthHarmonic, bufferSize, sampleRate);
harmonic.generate();
- sine.add(harmonic);
+ sine.add(harmonic);
}
-
+
nthHarmonic += 2; // 3rd, 5th, 7th, 9th, etc
-
+
// Draw additive signal
-
+
for ( int i = 0; i < width - 1; i+=4 ) {
stroke((i + frameCount + 50) % 360, 100, 60);
line(i, height/2 - sine.signal[i % bufferSize] * 100, i+1, height/2 - sine.signal[(i+1) % bufferSize] * 300);
@@ -96,7 +96,7 @@
audioWriter(sine.signal);
}
</script>
-
+
<div><canvas id="signal" width="200px" height="200px"></canvas></div>
</body>
</html>
diff --git a/examples/rfft.html b/examples/rfft.html
index 9a83091..0a0811b 100644
--- a/examples/rfft.html
+++ b/examples/rfft.html
@@ -2,7 +2,7 @@
<head>
<script language="javascript" src="js/processing.js"></script>
<script language="javascript" src="js/init.js"></script>
- <script language="javascript" src="../dsp.js"></script>
+ <script language="javascript" src="../dist/dsp.js"></script>
</head>
<body>
<style>
diff --git a/examples/sampler.html b/examples/sampler.html
index 6efeb3f..d8b8780 100644
--- a/examples/sampler.html
+++ b/examples/sampler.html
@@ -11,7 +11,7 @@
<script language="javascript" src="js/init.js"></script>
<!-- Load DSP.js -->
- <script src="../dsp.js"></script>
+ <script src="../dist/dsp.js"></script>
<script>
</script>
diff --git a/examples/squarewave.html b/examples/squarewave.html
index f6be448..d5fae69 100644
--- a/examples/squarewave.html
+++ b/examples/squarewave.html
@@ -1,8 +1,8 @@
<html>
- <head>
+ <head>
<script language="javascript" src="js/processing.js"></script>
<script language="javascript" src="js/init.js"></script>
- <script language="javascript" src="../dsp.js"></script>
+ <script language="javascript" src="../dist/dsp.min.js"></script>
</head>
<body>
<script>
@@ -20,11 +20,11 @@
output.mozSetup(1, sampleRate, 1);
}
- var audioWriter = function(s) {
+ var audioWriter = function(s) {
output.mozWriteAudio(s);
}
</script>
-
+
<script target="#signal" type="application/processing">
int nthHarmonic = 1;
float frequency = 344.53;
@@ -33,7 +33,7 @@
void setup() {
size(1024, 100);
frameRate(60);
-
+
fft = new FFT(bufferSize, sampleRate);
sine = new Oscillator(Oscillator.Sine, frequency, 1, bufferSize, sampleRate);
sine.generate();
@@ -42,28 +42,28 @@
void draw() {
background(0);
- if (nthHarmonic > 40 )
+ if (nthHarmonic > 40 )
{
nthHarmonic = 1;
sine.generate();
}
-
+
// Add harmonic
if ( nthHarmonic > 1 ) {
- harmonic = new Oscillator(Oscillator.Sine, frequency*nthHarmonic, 1/nthHarmonic, bufferSize, sampleRate);
+ harmonic = new Oscillator(Oscillator.Sine, frequency*nthHarmonic, 1/nthHarmonic, bufferSize, sampleRate);
harmonic.generate();
- sine.add(harmonic);
+ sine.add(harmonic);
}
-
+
nthHarmonic += 2; // 3rd, 5th, 7th, 9th, etc
// Calculate forward transform
fft.forward(sine.signal);
-
+
// Draw additive signal
stroke(255);
strokeWeight(1.5);
-
+
for ( int i = 0; i < bufferSize - 1; i++ ) {
line(i, scale + 10 - sine.signal[i] * scale, i+1, scale + 10 - sine.signal[i+1] * scale);
}
@@ -72,30 +72,30 @@
audioWriter(sine.signal);
}
</script>
-
+
<script target="#fft" type="application/processing">
void setup() {
size(1024, 300);
}
-
+
void draw() {
background(0);
-
+
// Draw spectrum
stroke(255);
strokeWeight(1.5);
-
+
for ( int i = 0; i < fft.spectrum.length - 1; i+=4 ) {
line(2*i/4, height - 10 - fft.spectrum[i] * 512, 2*i/4+1, height - 10 - fft.spectrum[i+1] * 512);
}
- }
+ }
</script>
-
+
<h1>Building a Square wave</h1>
<p>Any complex waveform can be made up of simple sinusoids.</p>
<p>A square wave is made by first starting with a sine wave then adding every other nth harmonic at frequency nth * the base frequency and amplitude of 1/n</p>
- <p>FFT breaks down a waveform into its sinusoidal components to measure the amplitude of the frequencies.</p>
-
+ <p>FFT breaks down a waveform into its sinusoidal components to measure the amplitude of the frequencies.</p>
+
<b>1.</b> Additive sine waves combine to form a complex square wave.
<div><canvas id="signal" width="200px" height="200px"></canvas></div>
<p></p>
diff --git a/examples/synthesizer.html b/examples/synthesizer.html
index a888029..fc63f91 100644
--- a/examples/synthesizer.html
+++ b/examples/synthesizer.html
@@ -2,19 +2,19 @@
<html>
<head>
<!-- Load JQuery and JQuery-UI -->
- <link type="text/css" href="css/hot-sneaks/jquery-ui-1.8.custom.css" rel="stylesheet" />
+ <link type="text/css" href="css/hot-sneaks/jquery-ui-1.8.custom.css" rel="stylesheet" />
<script type="text/javascript" src="js/jquery-1.4.2.min.js"></script>
<script type="text/javascript" src="js/jquery-ui-1.8.custom.min.js"></script>
-
+
<!-- Load Processing.js -->
<script language="javascript" src="js/processing.js"></script>
<script language="javascript" src="js/init.js"></script>
<!-- Load DSP.js -->
- <script src="../dsp.js"></script>
+ <script src="../dist/dsp.js"></script>
<script>
$(function() {
- // Amplitude
+ // Amplitude
$('#amp' ).slider({ orientation: 'vertical', range: 'min', value: 0.7, min: 0, max: 1, step: 0.01, slide: changeAmplitude });
// Amp. Envelope
@@ -35,7 +35,7 @@
// Osc Mix
$('#osc_mix').slider({ orientation: 'vertical', range: 'min', value: 0.5, min: -0.1, max: 1.1, step: 0.01, slide: changeAmplitude });
-
+
changeAmplitude(); // reset amp and mix
// Debug toggle
@@ -61,27 +61,27 @@
$("input[name='filter_type']").change(changeFilter);
changeFilter(); // reset filter to slider defaults
- // Oscillator waveform select
+ // Oscillator waveform select
$("#osc1_waveform").change(changeOsc1);
$("#osc2_waveform").change(changeOsc2);
- // Oscillator osctave select
+ // Oscillator osctave select
$("#osc1_octave").change(changeOsc1);
$("#osc2_octave").change(changeOsc2);
-
- // Oscillator osctave select
+
+ // Oscillator osctave select
$("#osc1_semi").change(changeOsc1);
$("#osc2_semi").change(changeOsc2);
changeOsc1(); // reset oscillator to select defaults
- changeOsc2();
+ changeOsc2();
});
</script>
<style type="text/css">
body, * {
font-family: Arial, sans-serif;
- font-size: 12px;
+ font-size: 12px;
}
.control {
@@ -111,7 +111,7 @@
margin-bottom: 16px;
width: 8px;
}
-
+
.ui-slider .ui-slider-handle {
width: 8px;
margin-left: 3px;
@@ -122,27 +122,27 @@
<script>
var SHOW_SEQUENCER = true;
var CURVE_EDITOR = false;
-
+
// Setup shared variables
var sampleRate = 44100;
var bufferSize = 1024;
var bufferTime = Math.floor(1000 / (sampleRate / bufferSize));
-
+
var frequency = 440;
var amplitude = 0.7; // Default amplitude at 70%
-
+
var changeAmplitude = function() {
- amplitude = $('#amp').slider('option', 'value');
+ amplitude = $('#amp').slider('option', 'value');
$('#ampLevel').html(Math.round(amplitude * 100) + "%");
-
+
var mix = $('#osc_mix').slider('option', 'value');
-
+
mix = mix > 1 ? 1 : mix;
mix = mix < 0 ? 0 : mix;
-
+
osc2_amplitude = 1 - (osc1_amplitude = mix);
-
-
+
+
osc1_amplitude *= amplitude;
osc2_amplitude *= amplitude;
};
@@ -156,18 +156,18 @@
var changeFilterEnvelope = function() {
};
-
+
var osc;
var osc1;
var osc2;
-
+
var osc1_waveform;
var osc2_waveform;
var osc1_octave;
var osc2_octave;
-
+
var osc1_amplitude;
var osc2_amplitude;
@@ -191,13 +191,13 @@
osc2_octave = parseInt($("#osc2_octave").val());
osc2_semi = parseInt($("#osc2_semi").val());
};
-
+
var bpm = 60; // Beats per minute
-
+
var sequencerStep = 0; // Sequencer step postion
-
+
var sequencerTicksPerMillisecond = bpm * 8 / 60000; // Sequencer Ticks each millisecond. A Tick is a partial step.
-
+
var polyphony = 8; // Number of oscillators that can be played at the same time
var oscPool = new Array(); // Pool of active oscillators
@@ -216,14 +216,14 @@
sequencerNoteOn[i][j] = false;
}
}
-
+
var curveOver = 0;
var curveCutoff = new Array(17);
var curveRes = new Array(17);
for ( var i = 0; i < 17; i++ ) {
curveCutoff[i] = sampleRate / 4;
curveRes[i] = 0.1;
- }
+ }
var oct = 3; // Root Octave
@@ -242,8 +242,8 @@
var writeCount = 0;
- var timeStamp = (new Date()).getTime();
-
+ var timeStamp = (new Date()).getTime();
+
var silence = new Float32Array(bufferSize);
// Setup audio channel
@@ -251,40 +251,40 @@
if ( typeof a.mozSetup == 'function' ) {
a.mozSetup(1, sampleRate);
}
-
+
var timePerWrite = 0;
var programStart;
-
+
var audioWriter = function() {
- var startTime = (new Date()).getTime();
+ var startTime = (new Date()).getTime();
var currentTime = (new Date()).getTime();
var mergedOsc; // undefined
var filterOn = $('#filter_toggle').attr('checked');
-
+
if ( programStart === undefined ) {
programStart = new Date();
}
-
+
// keep same pace as a real time
//while ( (new Date() - programStart) / 1000 * sampleRate / bufferSize >= writeCount ) {
for ( var i = 0; i < oscPool.length; i++ ) {
// Generate new frame of the signal
if ( oscPool[i].envelope.isActive() ) {
oscPool[i].osc1.generate(); // This generates a new frameBuffer of signal and applies the envelope as it goes.
- oscPool[i].osc2.generate();
-
+ oscPool[i].osc2.generate();
+
// combine osc1 and osc2
oscPool[i].osc1.addSignal(oscPool[i].osc2.signal);
-
+
// apply the filter
if ( filterOn ) {
filter.addEnvelope(oscPool[i].filterEnvelope);
filter.process(oscPool[i].osc1.signal);
}
-
+
// apply the envelope
oscPool[i].envelope.process(oscPool[i].osc1.signal);
-
+
// Merge down oscPool
if ( typeof mergedOsc === 'undefined' ) {
mergedOsc = oscPool[i].osc1;
@@ -295,7 +295,7 @@
oscPool[i] = null; // Mark for deletion
}
}
-
+
// Delete marked oscillators from the pool
var tmpPool = new Array();
for ( var i = 0; i < oscPool.length; i++ ) {
@@ -304,7 +304,7 @@
}
}
oscPool = tmpPool;
-
+
// Set the global osc object
if ( typeof mergedOsc !== 'undefined' ) {
osc = mergedOsc;
@@ -313,87 +313,87 @@
//osc.signal = undefined;
}
- // Flush buffer
+ // Flush buffer
a.mozWriteAudio([]);
-
+
// Write next audio frame
a.mozWriteAudio(osc.signal);
writeCount++;
sequencerStep += (currentTime - timeStamp) * sequencerTicksPerMillisecond;
- //}
+ //}
timeStamp = (new Date()).getTime();
var endTime = (new Date()).getTime();
-
+
timePerWrite = endTime - startTime;
}
</script>
-
+
<script target="#signal" type="application/processing">
float scale = 90.0;
void setup() {
size(512, 200);
-
+
// setup oscillator to hold the merged signal of osc1 and osc2
osc = new Oscillator(osc1_waveform, 440, 0, bufferSize, sampleRate);
-
- setInterval(audioWriter, bufferTime); // start audioWriter timer
+
+ setInterval(audioWriter, bufferTime); // start audioWriter timer
frameRate(10);
}
void draw() {
var startTime = (new Date()).getTime();
-
+
int sequencerCol = Math.floor(sequencerStep) % 32;
background(0, 0, 30);
-
+
var curCol = Math.floor(sequencerCol/2);
var nextCol = (curCol + 1) % 17;
var alpha = ((sequencerStep % 32) / 2) - curCol; // should be a value between 0 and 0.99
-
+
if ( $("#curve_cutoff_toggle").attr("checked") ) {
// interpolate value
var curveVal = curveCutoff[curCol] + (curveCutoff[nextCol] - curveCutoff[curCol]) * alpha;
-
+
$('#cutoff').slider('option', 'value', curveVal);
changeFilter();
}
-
+
if ( $("#curve_res_toggle").attr("checked") ) {
var curveVal = curveRes[curCol] + (curveRes[nextCol] - curveRes[curCol]) * alpha;
-
+
$('#res').slider('option', 'value', curveVal);
changeFilter();
}
-
+
// Check sequencer column for notes to trigger
- for ( int i = 0; i < 24; i ++ ) {
+ for ( int i = 0; i < 24; i ++ ) {
if ( sequencer[sequencerCol][i] == true && !sequencerNoteOn[sequencerCol][i] ) {
sequencerNoteOn[sequencerCol][i] = true;
-
+
frequency = midiNoteFreq[12*oct+(24-i)];
-
+
osc1_frequency = midiNoteFreq[12 * osc1_octave + (24 - i) + osc1_semi] + osc1_cent;
osc2_frequency = midiNoteFreq[12 * osc2_octave + (24 - i) + osc2_semi] + osc2_cent;
-
+
// Add oscillator to the pool
var index = oscPool.length;
-
+
var osc1 = new Oscillator(osc1_waveform, osc1_frequency, osc1_amplitude, bufferSize, sampleRate);
var osc2 = new Oscillator(osc2_waveform, osc2_frequency, osc2_amplitude, bufferSize, sampleRate);
- var adsr = new ADSR($('#A').slider('option', 'value'), $('#D').slider('option', 'value'), $('#S').slider('option', 'value'), bufferTime / 1000, $('#R').slider('option', 'value'), sampleRate);
+ var adsr = new ADSR($('#A').slider('option', 'value'), $('#D').slider('option', 'value'), $('#S').slider('option', 'value'), bufferTime / 1000, $('#R').slider('option', 'value'), sampleRate);
var fadsr = new ADSR($('#FA').slider('option', 'value'), $('#FD').slider('option', 'value'), $('#FS').slider('option', 'value'), bufferTime / 1000, $('#FR').slider('option', 'value'), sampleRate);
-
+
oscPool[index] = {};
oscPool[index].osc1 = osc1;
oscPool[index].osc2 = osc2;
oscPool[index].envelope = adsr;
oscPool[index].filterEnvelope = fadsr;
}
-
+
if ( sequencer[sequencerCol][i] == false && sequencerCol > 0 ) {
sequencerNoteOn[sequencerCol-1][i] = false;
} else if ( sequencer[sequencerCol][i] == false && sequencerCol == 0 ) {
@@ -408,12 +408,12 @@
* Show step sequencer
*
*/
- stroke(200, 255, 255, 20);
+ stroke(200, 255, 255, 20);
// Render horozontal lines
for (int i = 0; i < 24; i++ ) {
line(0, i * (height/24), width, i * (height/24));
}
-
+
// Render vertical lines
for (int i = 0; i < 32; i++ ) {
if ( i % 8 == 0 ) {
@@ -424,7 +424,7 @@
strokeWeight(1);
}
line(i * (width/32), 0, i * (width/32), height);
-
+
// Render sequencer step position
if ( i == Math.floor(sequencerStep % 32) ) {
noStroke();
@@ -452,74 +452,74 @@
}
}
}
-
+
if ( CURVE_EDITOR ) {
strokeWeight(2);
-
+
var cutoffScale = 50;
var resScale = 100;
-
+
for ( int i = 0; i < 17; i++ ) {
fill(0, 0, 30);
-
-
+
+
// Cut off curve
if ( $("#curve_cutoff_toggle").attr("checked") ) {
stroke(200, 255, 255, 100);
} else {
stroke(200, 255, 255, 40);
}
-
+
var xPos = i * (width/16);
var yPos = height/2 - curveCutoff[i] / cutoffScale;
-
+
if (i < 16 ) {
var xPosNext = (i+1) * (width/16);
var yPosNext = height/2 - curveCutoff[i+1] / cutoffScale;
-
+
line(xPos, yPos, xPosNext, yPosNext);
}
-
- if ( mouseX > xPos - 7 && mouseX < xPos + 7 && mouseY > yPos -7 && mouseY < yPos + 7 ) {
+
+ if ( mouseX > xPos - 7 && mouseX < xPos + 7 && mouseY > yPos -7 && mouseY < yPos + 7 ) {
ellipse(xPos, yPos, 7, 7);
curveOver = i;
} else {
ellipse(xPos, yPos, 5, 5);
}
-
-
+
+
// Resonance Curve
if ( $("#curve_res_toggle").attr("checked") ) {
stroke(200, 255, 255, 100);
} else {
stroke(200, 255, 255, 40);
}
-
+
var xPos = i * (width/16);
var yPos = height - curveRes[i] * resScale;
-
+
if (i < 16 ) {
var xPosNext = (i+1) * (width/16);
var yPosNext = height - curveRes[i+1] * resScale;
-
+
line(xPos, yPos, xPosNext, yPosNext);
}
-
- if ( mouseX > xPos - 7 && mouseX < xPos + 7 && mouseY > yPos -7 && mouseY < yPos + 7 ) {
+
+ if ( mouseX > xPos - 7 && mouseX < xPos + 7 && mouseY > yPos -7 && mouseY < yPos + 7 ) {
ellipse(xPos, yPos, 7, 7);
curveOver = i;
} else {
ellipse(xPos, yPos, 5, 5);
}
-
+
// draw center line
stroke(200, 255, 255, 60);
line(0, height/2, width, height/2);
-
-
+
+
}
strokeWeight(1);
-
+
if ( DRAGGING ) {
if ( mouseY < height / 2 ) {
curveCutoff[curveOver] = (height/2 - mouseY) * cutoffScale;
@@ -528,7 +528,7 @@
}
}
}
-
+
} else {
/*
* Signal View
@@ -538,15 +538,15 @@
*/
strokeWeight(1.5);
stroke(155, 200, 220);
- for ( int i = 0; i < width/4 - 1; i++ ) {
+ for ( int i = 0; i < width/4 - 1; i++ ) {
line(4*i, scale + 10 - osc.signal[4*i] * scale, 4*(i+1), scale + 10 - osc.signal[4*(i+1)] * scale);
}
}
var endTime = (new Date()).getTime();
-
+
if ( $('#debug_toggle').attr('checked') ) {
- $('#debug').html(
+ $('#debug').html(
"Time/frame (milliseconds): " + ( endTime - startTime ) + "<br>" +
"Time/write (milliseconds): " + ( timePerWrite ) + "<br>" +
"Audio frame size: " + bufferSize + "<br>" +
@@ -559,9 +559,9 @@
);
}
}
-
+
boolean DRAGGING = false;
-
+
void mousePressed() {
if ( SHOW_SEQUENCER && CURVE_EDITOR ) {
DRAGGING = true;
@@ -588,19 +588,19 @@
}
}
</script>
-
+
<h1>Sequencer</h1>
<p>Sequencer controlled oscillator with envelope and filter.</p>
<p><b>Click in the sequencer grid to plot notes. Switch to the signal view and adjust the envelope and filter to see the resulting waveforms.</b></p>
<p>Keyboard Commands:</p>
-
+
<ul>
<li><b>S</b> - toggle between sequencer/signal view</li>
<li><b>C</b> - toggle curve editor</li>
<li><b>W</b> - change waveform</li>
</ul>
-
+
<div style="float:left;margin-right: 5px;"><canvas id="signal" width="200px" height="200px"></canvas></div>
<div class="control">
diff --git a/examples/vocoder.html b/examples/vocoder.html
index f343212..0dfb3c5 100644
--- a/examples/vocoder.html
+++ b/examples/vocoder.html
@@ -11,7 +11,7 @@
<script language="javascript" src="js/init.js"></script>
<!-- Load DSP.js -->
- <script src="../dsp.js"></script>
+ <script src="../dist/dsp.js"></script>
<script>
</script>
diff --git a/lib/adsr.js b/lib/adsr.js
new file mode 100644
index 0000000..2384d0f
--- /dev/null
+++ b/lib/adsr.js
@@ -0,0 +1,134 @@
+/**
+ * ADSR Envelope
+ *
+ * @name ADSR
+ * @param {Number} attack The attack length in seconds
+ * @param {Number} decay The decay length in seconds
+ * @param {Number} sustain The sustain level
+ * @param {Number} release The release length in seconds
+ * @param {Number} sampleRate The the sample rate
+ *
+ * @constructor
+ */
+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;
+}
+
+/**
+ * Start the envelope
+ */
+ADSR.prototype.noteOn = function() {
+ this.samplesProcessed = 0;
+ this.sustainSamples = this.sustainLength * this.sampleRate;
+ this.update();
+};
+
+/**
+ * Stop the envelope
+ *
+ * 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();
+};
+
+/**
+ * Process sample
+ */
+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;
+};
+
+/**
+ * Get current value
+ * @return {Number} 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;
+};
+
+/**
+ * Process a buffer
+ * @param {Array} buffer
+ */
+ADSR.prototype.process = function(buffer) {
+ for ( var i = 0; i < buffer.length; i++ ) {
+ buffer[i] *= this.value();
+
+ this.samplesProcessed++;
+ }
+
+ return buffer;
+};
+
+/**
+ * Test if the envelope is active
+ * @return {Boolean}
+ */
+ADSR.prototype.isActive = function() {
+ if ( this.samplesProcessed > this.release || this.samplesProcessed === -1 ) {
+ return false;
+ } else {
+ return true;
+ }
+};
+
+/**
+ * Disable the envelope
+ */
+ADSR.prototype.disable = function() {
+ this.samplesProcessed = -1;
+};
+
+module.exports = ADSR;
diff --git a/lib/biquad.js b/lib/biquad.js
new file mode 100644
index 0000000..81c6e04
--- /dev/null
+++ b/lib/biquad.js
@@ -0,0 +1,288 @@
+/* global Float64Array */
+var DSP = require("./dsp");
+var sinh = require("./sinh");
+
+/**
+ * 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
+ *
+ * @param {Number} type
+ * @param {Number} sampleRate
+ * @constructor
+ */
+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;
+ };
+}
+
+module.exports = Biquad;
diff --git a/lib/dft.js b/lib/dft.js
new file mode 100644
index 0000000..d00eaae
--- /dev/null
+++ b/lib/dft.js
@@ -0,0 +1,57 @@
+/* global Float64Array */
+var FourierTransform = require("./fourier");
+
+/**
+ * DFT is a class for calculating the Discrete Fourier Transform of a signal.
+ *
+ * @param {Number} bufferSize The size of the sample buffer to be computed
+ * @param {Number} sampleRate The sampleRate of the buffer (eg. 44100)
+ *
+ * @constructor
+ */
+function DFT(bufferSize, sampleRate) {
+ FourierTransform.call(this, bufferSize, sampleRate);
+
+ var N = bufferSize/2 * bufferSize;
+ var TWO_PI = 2 * Math.PI;
+
+ this.sinTable = new Float64Array(N);
+ this.cosTable = new Float64Array(N);
+
+ for (var i = 0; i < N; i++) {
+ this.sinTable[i] = Math.sin(i * TWO_PI / bufferSize);
+ this.cosTable[i] = Math.cos(i * TWO_PI / bufferSize);
+ }
+}
+
+/**
+ * Performs a forward transform on the sample buffer.
+ * Converts a time domain signal to frequency domain spectra.
+ *
+ * @param {Array} buffer The sample buffer
+ *
+ * @returns The frequency spectrum array
+ */
+DFT.prototype.forward = function(buffer) {
+ var real = this.real,
+ imag = this.imag,
+ rval,
+ ival;
+
+ for (var k = 0; k < this.bufferSize/2; k++) {
+ rval = 0.0;
+ ival = 0.0;
+
+ for (var n = 0; n < buffer.length; n++) {
+ rval += this.cosTable[k*n] * buffer[n];
+ ival += this.sinTable[k*n] * buffer[n];
+ }
+
+ real[k] = rval;
+ imag[k] = ival;
+ }
+
+ return this.calculateSpectrum();
+};
+
+module.exports = DFT;
diff --git a/lib/dsp.js b/lib/dsp.js
new file mode 100644
index 0000000..44e05ac
--- /dev/null
+++ b/lib/dsp.js
@@ -0,0 +1,315 @@
+/* global Float64Array */
+
+////////////////////////////////////////////////////////////////////////////////
+// CONSTANTS //
+////////////////////////////////////////////////////////////////////////////////
+
+/**
+ * DSP is an object which contains general purpose utility functions and constants
+ */
+var DSP = {
+ // Channels
+ LEFT: 0,
+ RIGHT: 1,
+ MIX: 2,
+
+ // Waveforms
+ SINE: 1,
+ TRIANGLE: 2,
+ SAW: 3,
+ SQUARE: 4,
+
+ // Filters
+ LOWPASS: 0,
+ HIGHPASS: 1,
+ BANDPASS: 2,
+ NOTCH: 3,
+
+ // Window functions
+ BARTLETT: 1,
+ BARTLETTHANN: 2,
+ BLACKMAN: 3,
+ COSINE: 4,
+ GAUSS: 5,
+ HAMMING: 6,
+ HANN: 7,
+ LANCZOS: 8,
+ RECTANGULAR: 9,
+ TRIANGULAR: 10,
+
+ // Loop modes
+ OFF: 0,
+ FW: 1,
+ BW: 2,
+ FWBW: 3,
+
+ // Math
+ TWO_PI: 2*Math.PI
+};
+
+// Setup arrays for platforms which do not support byte arrays
+function setupTypedArray(name, fallback) {
+ // check if TypedArray exists
+ // typeof on Minefield and Chrome return function, typeof on Webkit returns object.
+ if (typeof this[name] !== "function" && typeof this[name] !== "object") {
+ // nope.. check if WebGLArray exists
+ if (typeof this[fallback] === "function" && typeof this[fallback] !== "object") {
+ this[name] = this[fallback];
+ } else {
+ // nope.. set as Native JS array
+ this[name] = function(obj) {
+ if (obj instanceof Array) {
+ return obj;
+ } else if (typeof obj === "number") {
+ return new Array(obj);
+ }
+ };
+ }
+ }
+}
+
+setupTypedArray("Float64Array", "WebGLFloatArray");
+setupTypedArray("Int32Array", "WebGLIntArray");
+setupTypedArray("Uint16Array", "WebGLUnsignedShortArray");
+setupTypedArray("Uint8Array", "WebGLUnsignedByteArray");
+
+
+////////////////////////////////////////////////////////////////////////////////
+// DSP UTILITY FUNCTIONS //
+////////////////////////////////////////////////////////////////////////////////
+
+/**
+ * Inverts the phase of a signal
+ *
+ * @param {Array} buffer A sample buffer
+ *
+ * @returns The inverted sample buffer
+ */
+DSP.invert = function(buffer) {
+ for (var i = 0, len = buffer.length; i < len; i++) {
+ buffer[i] *= -1;
+ }
+
+ return buffer;
+};
+
+/**
+ * Converts split-stereo (dual mono) sample buffers into a stereo interleaved sample buffer
+ *
+ * @param {Array} left A sample buffer
+ * @param {Array} right A sample buffer
+ *
+ * @returns The stereo interleaved buffer
+ */
+DSP.interleave = function(left, right) {
+ if (left.length !== right.length) {
+ throw "Can not interleave. Channel lengths differ.";
+ }
+
+ var stereoInterleaved = new Float64Array(left.length * 2);
+
+ for (var i = 0, len = left.length; i < len; i++) {
+ stereoInterleaved[2*i] = left[i];
+ stereoInterleaved[2*i+1] = right[i];
+ }
+
+ return stereoInterleaved;
+};
+
+/**
+ * Converts a stereo-interleaved sample buffer into split-stereo (dual mono) sample buffers
+ *
+ * @param {Array} buffer A stereo-interleaved sample buffer
+ *
+ * @returns an Array containing left and right channels
+ */
+DSP.deinterleave = (function() {
+ var left, right, mix, deinterleaveChannel = [];
+
+ deinterleaveChannel[DSP.MIX] = function(buffer) {
+ for (var i = 0, len = buffer.length/2; i < len; i++) {
+ mix[i] = (buffer[2*i] + buffer[2*i+1]) / 2;
+ }
+ return mix;
+ };
+
+ deinterleaveChannel[DSP.LEFT] = function(buffer) {
+ for (var i = 0, len = buffer.length/2; i < len; i++) {
+ left[i] = buffer[2*i];
+ }
+ return left;
+ };
+
+ deinterleaveChannel[DSP.RIGHT] = function(buffer) {
+ for (var i = 0, len = buffer.length/2; i < len; i++) {
+ right[i] = buffer[2*i+1];
+ }
+ return right;
+ };
+
+ return function(channel, buffer) {
+ left = left || new Float64Array(buffer.length/2);
+ right = right || new Float64Array(buffer.length/2);
+ mix = mix || new Float64Array(buffer.length/2);
+
+ if (buffer.length/2 !== left.length) {
+ left = new Float64Array(buffer.length/2);
+ right = new Float64Array(buffer.length/2);
+ mix = new Float64Array(buffer.length/2);
+ }
+
+ return deinterleaveChannel[channel](buffer);
+ };
+}());
+
+/**
+ * Separates a channel from a stereo-interleaved sample buffer
+ *
+ * @param {Array} buffer A stereo-interleaved sample buffer
+ * @param {Number} channel A channel constant (LEFT, RIGHT, MIX)
+ *
+ * @returns an Array containing a signal mono sample buffer
+ */
+DSP.getChannel = DSP.deinterleave;
+
+/**
+ * Helper method (for Reverb) to mix two (interleaved) samplebuffers. It's possible
+ * to negate the second buffer while mixing and to perform a volume correction
+ * on the final signal.
+ *
+ * @param {Array} sampleBuffer1 Array containing Float values or a Float64Array
+ * @param {Array} sampleBuffer2 Array containing Float values or a Float64Array
+ * @param {Boolean} negate When true inverts/flips the audio signal
+ * @param {Number} volumeCorrection When you add multiple sample buffers, use this to tame your signal ;)
+ *
+ * @returns A new Float64Array interleaved buffer.
+ */
+DSP.mixSampleBuffers = function(sampleBuffer1, sampleBuffer2, negate, volumeCorrection){
+ var outputSamples = new Float64Array(sampleBuffer1);
+
+ for(var i = 0; i<sampleBuffer1.length; i++){
+ outputSamples[i] += (negate ? -sampleBuffer2[i] : sampleBuffer2[i]) / volumeCorrection;
+ }
+
+ return outputSamples;
+};
+
+// Biquad filter types
+DSP.LPF = 0; // H(s) = 1 / (s^2 + s/Q + 1)
+DSP.HPF = 1; // H(s) = s^2 / (s^2 + s/Q + 1)
+DSP.BPF_CONSTANT_SKIRT = 2; // H(s) = s / (s^2 + s/Q + 1) (constant skirt gain, peak gain = Q)
+DSP.BPF_CONSTANT_PEAK = 3; // H(s) = (s/Q) / (s^2 + s/Q + 1) (constant 0 dB peak gain)
+DSP.NOTCH = 4; // H(s) = (s^2 + 1) / (s^2 + s/Q + 1)
+DSP.APF = 5; // H(s) = (s^2 - s/Q + 1) / (s^2 + s/Q + 1)
+DSP.PEAKING_EQ = 6; // H(s) = (s^2 + s*(A/Q) + 1) / (s^2 + s/(A*Q) + 1)
+DSP.LOW_SHELF = 7; // H(s) = A * (s^2 + (sqrt(A)/Q)*s + A)/(A*s^2 + (sqrt(A)/Q)*s + 1)
+DSP.HIGH_SHELF = 8; // H(s) = A * (A*s^2 + (sqrt(A)/Q)*s + 1)/(s^2 + (sqrt(A)/Q)*s + A)
+
+// Biquad filter parameter types
+DSP.Q = 1;
+DSP.BW = 2; // SHARED with BACKWARDS LOOP MODE
+DSP.S = 3;
+
+/**
+ * Find RMS of signal
+ * @param {Array} buffer
+ */
+DSP.RMS = function(buffer) {
+ var total = 0;
+
+ for (var i = 0, n = buffer.length; i < n; i++) {
+ total += buffer[i] * buffer[i];
+ }
+
+ return Math.sqrt(total / n);
+};
+
+/**
+ * Find Peak of signal
+ * @param {Array} buffer
+ */
+DSP.Peak = function(buffer) {
+ var peak = 0;
+
+ for (var i = 0, n = buffer.length; i < n; i++) {
+ peak = (Math.abs(buffer[i]) > peak) ? Math.abs(buffer[i]) : peak;
+ }
+
+ return peak;
+};
+
+/**
+ * Magnitude to decibels
+ *
+ * Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
+ * Copyright 2010 Ricard Marxer. All rights reserved.
+ *
+ * @param {Array} @buffer The 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 = new 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.
+ *
+ * @param {Number} b The coefficients of the filter
+ * @param {Number} a The coefficients of the filter
+ * @param {Number} w The 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 = new Float64Array(200);
+ for (i=0;i<w.length; i++) {
+ w[i] = DSP.TWO_PI/w.length * i - Math.PI;
+ }
+ }
+
+ var result = new 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;
+};
+
+module.exports = DSP;
diff --git a/lib/fft.js b/lib/fft.js
new file mode 100644
index 0000000..97ec669
--- /dev/null
+++ b/lib/fft.js
@@ -0,0 +1,198 @@
+/* global Float64Array Uint32Array */
+var FourierTransform = require("./fourier");
+
+/**
+ * FFT is a class for calculating the Discrete Fourier Transform of a signal
+ * with the Fast Fourier Transform algorithm.
+ *
+ * @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
+ */
+function FFT(bufferSize, sampleRate) {
+ FourierTransform.call(this, bufferSize, sampleRate);
+
+ this.reverseTable = new Uint32Array(bufferSize);
+
+ var limit = 1;
+ var bit = bufferSize >> 1;
+
+ var i;
+
+ while (limit < bufferSize) {
+ for (i = 0; i < limit; i++) {
+ this.reverseTable[i + limit] = this.reverseTable[i] + bit;
+ }
+
+ limit = limit << 1;
+ bit = bit >> 1;
+ }
+
+ this.sinTable = new Float64Array(bufferSize);
+ this.cosTable = new Float64Array(bufferSize);
+
+ for (i = 0; i < bufferSize; i++) {
+ this.sinTable[i] = Math.sin(-Math.PI/i);
+ this.cosTable[i] = Math.cos(-Math.PI/i);
+ }
+}
+
+/**
+ * Performs a forward transform on the sample buffer.
+ * Converts a time domain signal to frequency domain spectra.
+ *
+ * @param {Array} buffer The sample buffer. Buffer Length must be power of 2
+ *
+ * @returns The frequency spectrum array
+ */
+FFT.prototype.forward = function(buffer) {
+ // Locally scope variables for speed up
+ var bufferSize = this.bufferSize,
+ cosTable = this.cosTable,
+ sinTable = this.sinTable,
+ reverseTable = this.reverseTable,
+ real = this.real,
+ imag = this.imag;
+
+ 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 this.calculateSpectrum();
+};
+
+/**
+ * Performs a inverse FFT transformation
+ * Converts a frequency domain spectra to a time domain signal
+ *
+ * @param {Array} real
+ * @param {Array} imag
+ *
+ * @returns The time domain signal
+ */
+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;
+
+ 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;
+};
+
+module.exports = FFT;
diff --git a/lib/fourier.js b/lib/fourier.js
new file mode 100644
index 0000000..b47419c
--- /dev/null
+++ b/lib/fourier.js
@@ -0,0 +1,57 @@
+/* global Float64Array */
+
+/**
+ * A Mixin for every fourier module
+ * Fourier Transform Module used by DFT, FFT, RFFT
+ * @private
+ */
+module.exports = function FourierTransform(bufferSize, sampleRate) {
+ this.bufferSize = bufferSize;
+ this.sampleRate = sampleRate;
+ this.bandwidth = 2 / bufferSize * sampleRate / 2;
+
+ this.spectrum = new Float64Array(bufferSize/2);
+ this.real = new Float64Array(bufferSize);
+ this.imag = new Float64Array(bufferSize);
+
+ this.peakBand = 0;
+ this.peak = 0;
+
+ /**
+ * Calculates the *middle* frequency of an FFT band.
+ *
+ * @param {Number} index The index of the FFT band.
+ *
+ * @returns The middle frequency in Hz.
+ */
+ this.getBandFrequency = function(index) {
+ return this.bandwidth * index + this.bandwidth / 2;
+ };
+
+ /**
+ * Calculate the spectrum (amplitude magnitudes)
+ */
+ this.calculateSpectrum = function() {
+ var spectrum = this.spectrum,
+ real = this.real,
+ imag = this.imag,
+ bSi = 2 / this.bufferSize,
+ sqrt = Math.sqrt,
+ rval,
+ ival,
+ mag;
+
+ for (var i = 0, N = bufferSize/2; i < N; i++) {
+ rval = real[i];
+ ival = imag[i];
+ mag = bSi * sqrt(rval * rval + ival * ival);
+
+ if (mag > this.peak) {
+ this.peakBand = i;
+ this.peak = mag;
+ }
+
+ spectrum[i] = mag;
+ }
+ };
+};
diff --git a/lib/graphical-eq.js b/lib/graphical-eq.js
new file mode 100644
index 0000000..61d3c0e
--- /dev/null
+++ b/lib/graphical-eq.js
@@ -0,0 +1,117 @@
+/* global Float64Array */
+var DSP = require("./dsp");
+var Biquad = require("./biquad");
+
+/**
+ * Create a 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.
+ *
+ * @constructor
+ * @param {SampleRate}
+ * @example
+ * var eq = new GraphicalEq(44100)
+ */
+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 = new 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;
+ };
+}
+
+module.exports = GraphicalEq;
diff --git a/lib/iir-filter.js b/lib/iir-filter.js
new file mode 100644
index 0000000..ba0e010
--- /dev/null
+++ b/lib/iir-filter.js
@@ -0,0 +1,120 @@
+var DSP = require("./dsp");
+var ADSR = require("./adsr");
+
+/**
+ * IIRFilter
+ * @constructor
+ */
+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;
+ }
+);
+
+/**
+ * Set filter parameters
+ * @param {Number} cutoff
+ * @param {Number} resonance
+ */
+IIRFilter.prototype.set = function(cutoff, resonance) {
+ this.func.calcCoeff(cutoff, resonance);
+};
+
+/**
+ * Process a buffer
+ * @param {Array} buffer
+ */
+IIRFilter.prototype.process = function(buffer) {
+ this.func.process(buffer);
+};
+
+/**
+ * Add an envelope to the filter
+ * @param {ADSR} envelope
+ */
+IIRFilter.prototype.addEnvelope = function(envelope) {
+ if ( envelope instanceof ADSR ) {
+ this.func.addEnvelope(envelope);
+ } else {
+ throw "Not an envelope.";
+ }
+};
+
+/**
+ * LP12 filter
+ * @constructor
+ */
+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;
+ }
+ }
+ };
+};
+
+/**
+ * Add an envelope to the filter
+ * @param {ADSR} envelope
+ */
+IIRFilter.LP12.prototype.addEnvelope = function(envelope) {
+ this.envelope = envelope;
+};
+
+module.exports = IIRFilter;
diff --git a/lib/iir-filter2.js b/lib/iir-filter2.js
new file mode 100644
index 0000000..410ba92
--- /dev/null
+++ b/lib/iir-filter2.js
@@ -0,0 +1,88 @@
+/* global Float64Array */
+var ADSR = require("./adsr");
+
+/**
+ * IIRFilter2
+ *
+ * @constructor
+ * @param {Number} type
+ * @param {Number} cutoff
+ * @param {Number} resonance
+ * @param {Number} sampleRate
+ */
+function IIRFilter2(type, cutoff, resonance, sampleRate) {
+ this.type = type;
+ this.cutoff = cutoff;
+ this.resonance = resonance;
+ this.sampleRate = sampleRate;
+
+ this.f = new 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);
+}
+
+/**
+ * Process a buffer
+ * @param {Array} buffer
+ */
+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;
+ }
+ }
+};
+
+/**
+ * Add an envelope to the filter
+ * @param {ADSR} envelope
+ */
+IIRFilter2.prototype.addEnvelope = function(envelope) {
+ if ( envelope instanceof ADSR ) {
+ this.envelope = envelope;
+ } else {
+ throw "This is not an envelope.";
+ }
+};
+
+/**
+ * Set filter parameters
+ * @param {Number} cutoff
+ * @param {Number} resonance
+ */
+IIRFilter2.prototype.set = function(cutoff, resonance) {
+ this.calcCoeff(cutoff, resonance);
+};
+
+module.exports = IIRFilter2;
diff --git a/lib/index.js b/lib/index.js
new file mode 100644
index 0000000..762475d
--- /dev/null
+++ b/lib/index.js
@@ -0,0 +1,34 @@
+"use strict";
+
+/*
+ * DSP.js - a comprehensive digital signal processing library for javascript
+ *
+ * Created by Corban Brook <corbanbrook@gmail.com> on 2010-01-01.
+ * Copyright 2010 Corban Brook. All rights reserved.
+ *
+ */
+var DSPJS = {
+ DSP: require("./dsp"),
+ DFT: require("./dft"),
+ FFT: require("./fft"),
+ RFFT: require("./rfft"),
+ Sampler: require("./sampler"),
+ Oscillator: require("./oscillator"),
+ ADSR: require("./adsr"),
+ IIRFilter: require("./iir-filter"),
+ IIRFilter2: require("./iir-filter2"),
+ WindowFunction: require("./window-function"),
+ sinh: require("./sinh"),
+ Biquad: require("./biquad"),
+ GraphicalEq: require("./graphical-eq"),
+ MultiDelay: require("./multi-delay"),
+ SingleDelay: require("./single-delay"),
+ Reverb: require("./reverb")
+};
+
+if (typeof module === "object" && module.exports) module.exports = DSPJS;
+if (typeof window !== "undefined") {
+ Object.keys(DSPJS).forEach(function (k) {
+ window[k] = DSPJS[k];
+ });
+}
diff --git a/lib/multi-delay.js b/lib/multi-delay.js
new file mode 100644
index 0000000..33755de
--- /dev/null
+++ b/lib/multi-delay.js
@@ -0,0 +1,102 @@
+/* global Float64Array */
+/**
+ * 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;
+};
+
+module.exports = MultiDelay;
diff --git a/lib/oscillator.js b/lib/oscillator.js
new file mode 100644
index 0000000..e193932
--- /dev/null
+++ b/lib/oscillator.js
@@ -0,0 +1,195 @@
+/* global Float64Array */
+var DSP = require("./dsp");
+
+/**
+ * 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
+ *
+ * @constructor
+ */
+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
+ * @param {Oscillator} oscillator The oscillator to be added to
+ * @return {Array} the current oscillator signal
+ */
+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
+ * @param {Array} 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
+ * @param {ADSR} envelope
+ */
+Oscillator.prototype.addEnvelope = function(envelope) {
+ this.envelope = envelope;
+};
+
+/**
+ * Apply the oscillator envelope to its signal
+ */
+Oscillator.prototype.applyEnvelope = function() {
+ this.envelope.process(this.signal);
+};
+
+/**
+ * Get value
+ * @param {Number} offset
+ */
+Oscillator.prototype.valueAt = function(offset) {
+ return this.waveTable[offset % this.waveTableLength];
+};
+
+/**
+ * Generate the oscillator signal
+ * @return {Array} the signal
+ */
+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() {
+ // stub
+};
+
+module.exports = Oscillator;
diff --git a/lib/reverb.js b/lib/reverb.js
new file mode 100644
index 0000000..8fc958c
--- /dev/null
+++ b/lib/reverb.js
@@ -0,0 +1,174 @@
+/* global Float64Array */
+var DSP = require("./dsp");
+var IIRFilter2 = require("./iir-filter2");
+var SingleDelay = require("./single-delay");
+var MultiDelay = require("./multi-delay");
+
+/**
+ * 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 = Reverb;
diff --git a/lib/rfft.js b/lib/rfft.js
new file mode 100644
index 0000000..52ebbdd
--- /dev/null
+++ b/lib/rfft.js
@@ -0,0 +1,312 @@
+/* global Float64Array Uint32Array */
+var FourierTransform = require("./fourier");
+
+/**
+ * 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();
+}
+
+
+/**
+ * Performs a forward transform on the sample buffer.
+ * Converts a time domain signal to frequency domain spectra.
+ *
+ * @param {Array} buffer The sample buffer. Buffer Length must be power of 2
+ *
+ * @returns The frequency spectrum array
+ */
+
+// 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;
+};
+
+module.exports = RFFT;
diff --git a/lib/sampler.js b/lib/sampler.js
new file mode 100644
index 0000000..791d657
--- /dev/null
+++ b/lib/sampler.js
@@ -0,0 +1,133 @@
+/* global Float64Array */
+var DSP = require("./dsp");
+
+/**
+ * Sampler
+ *
+ * @constructor
+ * @param {} file
+ * @param {Number} bufferSize
+ * @param {Number} sampleRate
+ * @param {Number} playStart
+ * @param {Number} playEnd
+ * @param {Number} loopStart
+ * @param {Number} loopEnd
+ * @param {Number} loopMode
+ */
+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 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
+
+ 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;
+};
+
+module.exports = Sampler;
diff --git a/lib/single-delay.js b/lib/single-delay.js
new file mode 100644
index 0000000..bd3febd
--- /dev/null
+++ b/lib/single-delay.js
@@ -0,0 +1,93 @@
+/* global Float64Array */
+
+/**
+ * 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;
+};
+
+module.exports = SingleDelay;
diff --git a/lib/sinh.js b/lib/sinh.js
new file mode 100644
index 0000000..d2706f3
--- /dev/null
+++ b/lib/sinh.js
@@ -0,0 +1,16 @@
+/**
+ * Returns the hyperbolic sine of the number
+ *
+ * @meta version: 1004.2314
+ * @meta discuss at: http://phpjs.org/functions/sinh
+ * @meta original by: Onno Marsman
+ *
+ * @param {Number} num
+ * @example
+ * sinh(-0.9834330348825909); // => -1.1497971402636502
+ */
+function sinh (arg) {
+ return (Math.exp(arg) - Math.exp(-arg))/2;
+}
+
+module.exports = sinh;
diff --git a/lib/window-function.js b/lib/window-function.js
new file mode 100644
index 0000000..0e7b383
--- /dev/null
+++ b/lib/window-function.js
@@ -0,0 +1,115 @@
+var DSP = require("./dsp");
+
+/**
+ * WindowFunction
+ *
+ * @constructor
+ * @param {Number} type
+ * @param {Number} alpha
+ */
+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;
+ }
+}
+
+/**
+ * Process a buffer
+ * @param {Array} buffer
+ */
+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() {
+ return 1;
+};
+
+WindowFunction.Triangular = function(length, index) {
+ return 2 / length * (length / 2 - Math.abs(index - (length - 1) / 2));
+};
+
+module.exports = WindowFunction;
diff --git a/package.json b/package.json
index decf3e3..00a2e89 100644
--- a/package.json
+++ b/package.json
@@ -2,7 +2,16 @@
"name": "dsp.js",
"version": "1.0.0",
"description": "Digital Signal Processing for Javascript",
- "main": "dsp.js",
+ "main": "lib/index.js",
+ "scripts": {
+ "prepublish": "npm run dist",
+ "predist": "npm run test && npm run lint && npm run docs",
+ "dist": "rm dist/* && browserify lib/index.js > dist/dsp.js && uglifyjs dist/dsp.js > dist/dsp.min.js",
+ "docs": "jsdoc2md \"lib/*.js\" > docs/API.md",
+ "test": "mocha test/*.js",
+ "lint": "eslint lib/*.js",
+ "bench": "node bench/index.js"
+ },
"directories": {
"example": "examples",
"test": "test"
@@ -21,5 +30,15 @@
"bugs": {
"url": "https://github.com/corbanbrook/dsp.js/issues"
},
- "homepage": "https://github.com/corbanbrook/dsp.js#readme"
+ "homepage": "https://github.com/corbanbrook/dsp.js#readme",
+ "standard": {
+ "ignore": "**/*"
+ },
+ "devDependencies": {
+ "browserify": "^13.1.1",
+ "eslint": "^3.12.2",
+ "jsdoc-to-markdown": "^2.0.1",
+ "mocha": "^3",
+ "uglifyjs": "^2.4.10"
+ }
}
diff --git a/test/adsr-test.js b/test/audio-test/adsr-test.js
similarity index 100%
rename from test/adsr-test.js
rename to test/audio-test/adsr-test.js
diff --git a/test/audio-harness.js b/test/audio-test/audio-harness.js
similarity index 100%
rename from test/audio-harness.js
rename to test/audio-test/audio-harness.js
diff --git a/test/beatdetect-test.js b/test/audio-test/beatdetect-test.js
similarity index 100%
rename from test/beatdetect-test.js
rename to test/audio-test/beatdetect-test.js
diff --git a/test/beatdetektor.js b/test/audio-test/beatdetektor.js
similarity index 100%
rename from test/beatdetektor.js
rename to test/audio-test/beatdetektor.js
diff --git a/test/dft-test.js b/test/audio-test/dft-test.js
similarity index 100%
rename from test/dft-test.js
rename to test/audio-test/dft-test.js
diff --git a/test/fft-test.js b/test/audio-test/fft-test.js
similarity index 100%
rename from test/fft-test.js
rename to test/audio-test/fft-test.js
diff --git a/test/filter-test.js b/test/audio-test/filter-test.js
similarity index 100%
rename from test/filter-test.js
rename to test/audio-test/filter-test.js
diff --git a/test/iirfilter-test.js b/test/audio-test/iirfilter-test.js
similarity index 100%
rename from test/iirfilter-test.js
rename to test/audio-test/iirfilter-test.js
diff --git a/test/multidelay-test.js b/test/audio-test/multidelay-test.js
similarity index 100%
rename from test/multidelay-test.js
rename to test/audio-test/multidelay-test.js
diff --git a/test/oscillator-test.js b/test/audio-test/oscillator-test.js
similarity index 100%
rename from test/oscillator-test.js
rename to test/audio-test/oscillator-test.js
diff --git a/test/reverb-test.js b/test/audio-test/reverb-test.js
similarity index 100%
rename from test/reverb-test.js
rename to test/audio-test/reverb-test.js
diff --git a/test/samples.js b/test/audio-test/samples.js
similarity index 100%
rename from test/samples.js
rename to test/audio-test/samples.js
diff --git a/test/test-exports.js b/test/test-exports.js
new file mode 100644
index 0000000..5904aba
--- /dev/null
+++ b/test/test-exports.js
@@ -0,0 +1,16 @@
+/* global describe it */
+var assert = require("assert");
+var dsp = require("..");
+
+describe("dsp.js library", function () {
+ it("Export all modules", function () {
+ var keys = Object.keys(dsp);
+ assert.equal(16, keys.length, "must export 16 items");
+ assert(typeof dsp["DSP"] === "object", "DSP must be an object");
+ keys.forEach(function (key) {
+ if (key !== "DSP") {
+ assert(typeof dsp[key] === "function", key + " must be a function.");
+ }
+ });
+ });
+});