GuidedTrack.sublime-workspace

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				"heart",
				"heartCanvas"
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				"We",
				"Windowed"
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				"for",
				"forward"
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				"Y",
				"Ylessmean"
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				"FFT",
				"FFTwindow"
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				"fre",
				"frequency_subset"
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				"Buf",
				"BufferFillTime"
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				"upate",
				"updateMessages"
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				"get",
				"getpulse"
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				"Bu",
				"BufferFillTime"
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				"bu",
				"bufferSize"
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				"bufferSize"
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				"pea",
				"peak_idx"
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				"frequency_subset"
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				"Grn",
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				"Hea",
				"HeadPos"
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				"head",
				"head_pos"
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				"can",
				"canvasOverlay"
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				"test",
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				"ForeHead_h"
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			"contents": "darkruby501.github.io\n=====================\nJavascript Photoplethysmogram Webcam Pulse Detector\n\n<Insert Copyright Info (Ruben Bloom, 2014 + others)>\n\nBased on https://github.com/thearn/webcam-pulse-detector\n\nUses:  \n- https://github.com/auduno/headtrackr/\n- https://github.com/corbanbrook/dsp.js/\n- http://g.raphaeljs.com/\n- https://github.com/joewalnes/smoothie/\n\n\nTo-Do:\n- Debug operation in Mozilla Firefox.\n- Fix hardcoding of canvas sizes.\n- Potentially use only facedetection algorithm and no headtracking.\n- Use rotated head capture for forehead.\n- Refine FFT (windowing, etc.)\n- Apply smoothing to output? WMA?\n- Tidy variables.\n\n- Insert second plot for heart rate.\n- Set up way to record and export data.\n\nPotential Ways to Boost Accuracy:\n- Windowing on FFT\n- Filtering of Pulse Wave (band pass?).\n- Smoothing of Output (discard outliers)\n- Button to flush buffer\n- Automatically flush buffer when large movement detected\n- Green background!\n",
			"file": "README.md",
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			"contents": "/* \n *  DSP.js - a comprehensive digital signal processing  library for javascript\n * \n *  Created by Corban Brook <corbanbrook@gmail.com> on 2010-01-01.\n *  Copyright 2010 Corban Brook. All rights reserved.\n *\n */\n\n////////////////////////////////////////////////////////////////////////////////\n//                                  CONSTANTS                                 //\n////////////////////////////////////////////////////////////////////////////////\n\n/**\n * DSP is an object which contains general purpose utility functions and constants\n */\nvar DSP = {\n  // Channels\n  LEFT:           0,\n  RIGHT:          1,\n  MIX:            2,\n\n  // Waveforms\n  SINE:           1,\n  TRIANGLE:       2,\n  SAW:            3,\n  SQUARE:         4,\n\n  // Filters\n  LOWPASS:        0,\n  HIGHPASS:       1,\n  BANDPASS:       2,\n  NOTCH:          3,\n\n  // Window functions\n  BARTLETT:       1,\n  BARTLETTHANN:   2,\n  BLACKMAN:       3,\n  COSINE:         4,\n  GAUSS:          5,\n  HAMMING:        6,\n  HANN:           7,\n  LANCZOS:        8,\n  RECTANGULAR:    9,\n  TRIANGULAR:     10,\n\n  // Loop modes\n  OFF:            0,\n  FW:             1,\n  BW:             2,\n  FWBW:           3,\n\n  // Math\n  TWO_PI:         2*Math.PI\n};\n\n// Setup arrays for platforms which do not support byte arrays\nfunction setupTypedArray(name, fallback) {\n  // check if TypedArray exists\n  // typeof on Minefield and Chrome return function, typeof on Webkit returns object.\n  if (typeof this[name] !== \"function\" && typeof this[name] !== \"object\") {\n    // nope.. check if WebGLArray exists\n    if (typeof this[fallback] === \"function\" && typeof this[fallback] !== \"object\") {\n      this[name] = this[fallback];\n    } else {\n      // nope.. set as Native JS array\n      this[name] = function(obj) {\n        if (obj instanceof Array) {\n          return obj;\n        } else if (typeof obj === \"number\") {\n          return new Array(obj);\n        }\n      };\n    }\n  }\n}\n\nsetupTypedArray(\"Float32Array\", \"WebGLFloatArray\");\nsetupTypedArray(\"Int32Array\",   \"WebGLIntArray\");\nsetupTypedArray(\"Uint16Array\",  \"WebGLUnsignedShortArray\");\nsetupTypedArray(\"Uint8Array\",   \"WebGLUnsignedByteArray\");\n\n\n////////////////////////////////////////////////////////////////////////////////\n//                            DSP UTILITY FUNCTIONS                           //\n////////////////////////////////////////////////////////////////////////////////\n\n/**\n * Inverts the phase of a signal\n *\n * @param {Array} buffer A sample buffer\n *\n * @returns The inverted sample buffer\n */\nDSP.invert = function(buffer) {\n  for (var i = 0, len = buffer.length; i < len; i++) {\n    buffer[i] *= -1;\n  }\n\n  return buffer;\n};\n\n/**\n * Converts split-stereo (dual mono) sample buffers into a stereo interleaved sample buffer\n *\n * @param {Array} left  A sample buffer\n * @param {Array} right A sample buffer\n *\n * @returns The stereo interleaved buffer\n */\nDSP.interleave = function(left, right) {\n  if (left.length !== right.length) {\n    throw \"Can not interleave. Channel lengths differ.\";\n  }\n \n  var stereoInterleaved = new Float32Array(left.length * 2);\n \n  for (var i = 0, len = left.length; i < len; i++) {\n    stereoInterleaved[2*i]   = left[i];\n    stereoInterleaved[2*i+1] = right[i];\n  }\n \n  return stereoInterleaved;\n};\n\n/**\n * Converts a stereo-interleaved sample buffer into split-stereo (dual mono) sample buffers\n *\n * @param {Array} buffer A stereo-interleaved sample buffer\n *\n * @returns an Array containing left and right channels\n */\nDSP.deinterleave = (function() {\n  var left, right, mix, deinterleaveChannel = []; \n\n  deinterleaveChannel[DSP.MIX] = function(buffer) {\n    for (var i = 0, len = buffer.length/2; i < len; i++) {\n      mix[i] = (buffer[2*i] + buffer[2*i+1]) / 2;\n    }\n    return mix;\n  };\n\n  deinterleaveChannel[DSP.LEFT] = function(buffer) {\n    for (var i = 0, len = buffer.length/2; i < len; i++) {\n      left[i]  = buffer[2*i];\n    }\n    return left;\n  };\n\n  deinterleaveChannel[DSP.RIGHT] = function(buffer) {\n    for (var i = 0, len = buffer.length/2; i < len; i++) {\n      right[i]  = buffer[2*i+1];\n    }\n    return right;\n  };\n\n  return function(channel, buffer) { \n    left  = left  || new Float32Array(buffer.length/2);\n    right = right || new Float32Array(buffer.length/2);\n    mix   = mix   || new Float32Array(buffer.length/2);\n\n    if (buffer.length/2 !== left.length) {\n      left  = new Float32Array(buffer.length/2);\n      right = new Float32Array(buffer.length/2);\n      mix   = new Float32Array(buffer.length/2);\n    }\n\n    return deinterleaveChannel[channel](buffer);\n  };\n}());\n\n/**\n * Separates a channel from a stereo-interleaved sample buffer\n *\n * @param {Array}  buffer A stereo-interleaved sample buffer\n * @param {Number} channel A channel constant (LEFT, RIGHT, MIX)\n *\n * @returns an Array containing a signal mono sample buffer\n */\nDSP.getChannel = DSP.deinterleave;\n\n/**\n * Helper method (for Reverb) to mix two (interleaved) samplebuffers. It's possible\n * to negate the second buffer while mixing and to perform a volume correction\n * on the final signal.\n *\n * @param {Array} sampleBuffer1 Array containing Float values or a Float32Array\n * @param {Array} sampleBuffer2 Array containing Float values or a Float32Array\n * @param {Boolean} negate When true inverts/flips the audio signal\n * @param {Number} volumeCorrection When you add multiple sample buffers, use this to tame your signal ;)\n *\n * @returns A new Float32Array interleaved buffer.\n */\nDSP.mixSampleBuffers = function(sampleBuffer1, sampleBuffer2, negate, volumeCorrection){\n  var outputSamples = new Float32Array(sampleBuffer1);\n\n  for(var i = 0; i<sampleBuffer1.length; i++){\n    outputSamples[i] += (negate ? -sampleBuffer2[i] : sampleBuffer2[i]) / volumeCorrection;\n  }\n \n  return outputSamples;\n}; \n\n// Biquad filter types\nDSP.LPF = 0;                // H(s) = 1 / (s^2 + s/Q + 1)\nDSP.HPF = 1;                // H(s) = s^2 / (s^2 + s/Q + 1)\nDSP.BPF_CONSTANT_SKIRT = 2; // H(s) = s / (s^2 + s/Q + 1)  (constant skirt gain, peak gain = Q)\nDSP.BPF_CONSTANT_PEAK = 3;  // H(s) = (s/Q) / (s^2 + s/Q + 1)      (constant 0 dB peak gain)\nDSP.NOTCH = 4;              // H(s) = (s^2 + 1) / (s^2 + s/Q + 1)\nDSP.APF = 5;                // H(s) = (s^2 - s/Q + 1) / (s^2 + s/Q + 1)\nDSP.PEAKING_EQ = 6;         // H(s) = (s^2 + s*(A/Q) + 1) / (s^2 + s/(A*Q) + 1)\nDSP.LOW_SHELF = 7;          // H(s) = A * (s^2 + (sqrt(A)/Q)*s + A)/(A*s^2 + (sqrt(A)/Q)*s + 1)\nDSP.HIGH_SHELF = 8;         // H(s) = A * (A*s^2 + (sqrt(A)/Q)*s + 1)/(s^2 + (sqrt(A)/Q)*s + A)\n\n// Biquad filter parameter types\nDSP.Q = 1;\nDSP.BW = 2; // SHARED with BACKWARDS LOOP MODE\nDSP.S = 3;\n\n// Find RMS of signal\nDSP.RMS = function(buffer) {\n  var total = 0;\n  \n  for (var i = 0, n = buffer.length; i < n; i++) {\n    total += buffer[i] * buffer[i];\n  }\n  \n  return Math.sqrt(total / n);\n};\n\n// Find Peak of signal\nDSP.Peak = function(buffer) {\n  var peak = 0;\n  \n  for (var i = 0, n = buffer.length; i < n; i++) {\n    peak = (Math.abs(buffer[i]) > peak) ? Math.abs(buffer[i]) : peak; \n  }\n  \n  return peak;\n};\n\n// Fourier Transform Module used by DFT, FFT, RFFT\nfunction FourierTransform(bufferSize, sampleRate) {\n  this.bufferSize = bufferSize;\n  this.sampleRate = sampleRate;\n  this.bandwidth  = 2 / bufferSize * sampleRate / 2;\n\n  this.spectrum   = new Float32Array(bufferSize/2);\n  this.real       = new Float32Array(bufferSize);\n  this.imag       = new Float32Array(bufferSize);\n\n  this.peakBand   = 0;\n  this.peak       = 0;\n\n  /**\n   * Calculates the *middle* frequency of an FFT band.\n   *\n   * @param {Number} index The index of the FFT band.\n   *\n   * @returns The middle frequency in Hz.\n   */\n  this.getBandFrequency = function(index) {\n    return this.bandwidth * index + this.bandwidth / 2;\n  };\n\n  this.calculateSpectrum = function() {\n    var spectrum  = this.spectrum,\n        real      = this.real,\n        imag      = this.imag,\n        bSi       = 2 / this.bufferSize,\n        sqrt      = Math.sqrt,\n        rval, \n        ival,\n        mag;\n\n    for (var i = 0, N = bufferSize/2; i < N; i++) {\n      rval = real[i];\n      ival = imag[i];\n      mag = bSi * sqrt(rval * rval + ival * ival);\n\n      if (mag > this.peak) {\n        this.peakBand = i;\n        this.peak = mag;\n      }\n\n      spectrum[i] = mag;\n    }\n  };\n}\n\n/**\n * DFT is a class for calculating the Discrete Fourier Transform of a signal.\n *\n * @param {Number} bufferSize The size of the sample buffer to be computed\n * @param {Number} sampleRate The sampleRate of the buffer (eg. 44100)\n *\n * @constructor\n */\nfunction DFT(bufferSize, sampleRate) {\n  FourierTransform.call(this, bufferSize, sampleRate);\n\n  var N = bufferSize/2 * bufferSize;\n  var TWO_PI = 2 * Math.PI;\n\n  this.sinTable = new Float32Array(N);\n  this.cosTable = new Float32Array(N);\n\n  for (var i = 0; i < N; i++) {\n    this.sinTable[i] = Math.sin(i * TWO_PI / bufferSize);\n    this.cosTable[i] = Math.cos(i * TWO_PI / bufferSize);\n  }\n}\n\n/**\n * Performs a forward transform on the sample buffer.\n * Converts a time domain signal to frequency domain spectra.\n *\n * @param {Array} buffer The sample buffer\n *\n * @returns The frequency spectrum array\n */\nDFT.prototype.forward = function(buffer) {\n  var real = this.real, \n      imag = this.imag,\n      rval,\n      ival;\n\n  for (var k = 0; k < this.bufferSize/2; k++) {\n    rval = 0.0;\n    ival = 0.0;\n\n    for (var n = 0; n < buffer.length; n++) {\n      rval += this.cosTable[k*n] * buffer[n];\n      ival += this.sinTable[k*n] * buffer[n];\n    }\n\n    real[k] = rval;\n    imag[k] = ival;\n  }\n\n  return this.calculateSpectrum();\n};\n\n\n/**\n * FFT is a class for calculating the Discrete Fourier Transform of a signal\n * with the Fast Fourier Transform algorithm.\n *\n * @param {Number} bufferSize The size of the sample buffer to be computed. Must be power of 2\n * @param {Number} sampleRate The sampleRate of the buffer (eg. 44100)\n *\n * @constructor\n */\nfunction FFT(bufferSize, sampleRate) {\n  FourierTransform.call(this, bufferSize, sampleRate);\n   \n  this.reverseTable = new Uint32Array(bufferSize);\n\n  var limit = 1;\n  var bit = bufferSize >> 1;\n\n  var i;\n\n  while (limit < bufferSize) {\n    for (i = 0; i < limit; i++) {\n      this.reverseTable[i + limit] = this.reverseTable[i] + bit;\n    }\n\n    limit = limit << 1;\n    bit = bit >> 1;\n  }\n\n  this.sinTable = new Float32Array(bufferSize);\n  this.cosTable = new Float32Array(bufferSize);\n\n  for (i = 0; i < bufferSize; i++) {\n    this.sinTable[i] = Math.sin(-Math.PI/i);\n    this.cosTable[i] = Math.cos(-Math.PI/i);\n  }\n}\n\n/**\n * Performs a forward transform on the sample buffer.\n * Converts a time domain signal to frequency domain spectra.\n *\n * @param {Array} buffer The sample buffer. Buffer Length must be power of 2\n *\n * @returns The frequency spectrum array\n */\nFFT.prototype.forward = function(buffer) {\n  // Locally scope variables for speed up\n  var bufferSize      = this.bufferSize,\n      cosTable        = this.cosTable,\n      sinTable        = this.sinTable,\n      reverseTable    = this.reverseTable,\n      real            = this.real,\n      imag            = this.imag,\n      spectrum        = this.spectrum;\n\n  var k = Math.floor(Math.log(bufferSize) / Math.LN2);\n\n  if (Math.pow(2, k) !== bufferSize) { throw \"Invalid buffer size, must be a power of 2.\"; }\n  if (bufferSize !== buffer.length)  { throw \"Supplied buffer is not the same size as defined FFT. FFT Size: \" + bufferSize + \" Buffer Size: \" + buffer.length; }\n\n  var halfSize = 1,\n      phaseShiftStepReal,\n      phaseShiftStepImag,\n      currentPhaseShiftReal,\n      currentPhaseShiftImag,\n      off,\n      tr,\n      ti,\n      tmpReal,\n      i;\n\n  for (i = 0; i < bufferSize; i++) {\n    real[i] = buffer[reverseTable[i]];\n    imag[i] = 0;\n  }\n\n  while (halfSize < bufferSize) {\n    //phaseShiftStepReal = Math.cos(-Math.PI/halfSize);\n    //phaseShiftStepImag = Math.sin(-Math.PI/halfSize);\n    phaseShiftStepReal = cosTable[halfSize];\n    phaseShiftStepImag = sinTable[halfSize];\n    \n    currentPhaseShiftReal = 1;\n    currentPhaseShiftImag = 0;\n\n    for (var fftStep = 0; fftStep < halfSize; fftStep++) {\n      i = fftStep;\n\n      while (i < bufferSize) {\n        off = i + halfSize;\n        tr = (currentPhaseShiftReal * real[off]) - (currentPhaseShiftImag * imag[off]);\n        ti = (currentPhaseShiftReal * imag[off]) + (currentPhaseShiftImag * real[off]);\n\n        real[off] = real[i] - tr;\n        imag[off] = imag[i] - ti;\n        real[i] += tr;\n        imag[i] += ti;\n\n        i += halfSize << 1;\n      }\n\n      tmpReal = currentPhaseShiftReal;\n      currentPhaseShiftReal = (tmpReal * phaseShiftStepReal) - (currentPhaseShiftImag * phaseShiftStepImag);\n      currentPhaseShiftImag = (tmpReal * phaseShiftStepImag) + (currentPhaseShiftImag * phaseShiftStepReal);\n    }\n\n    halfSize = halfSize << 1;\n  }\n\n  return this.calculateSpectrum();\n};\n\nFFT.prototype.inverse = function(real, imag) {\n  // Locally scope variables for speed up\n  var bufferSize      = this.bufferSize,\n      cosTable        = this.cosTable,\n      sinTable        = this.sinTable,\n      reverseTable    = this.reverseTable,\n      spectrum        = this.spectrum;\n     \n      real = real || this.real;\n      imag = imag || this.imag;\n\n  var halfSize = 1,\n      phaseShiftStepReal,\n      phaseShiftStepImag,\n      currentPhaseShiftReal,\n      currentPhaseShiftImag,\n      off,\n      tr,\n      ti,\n      tmpReal,\n      i;\n\n  for (i = 0; i < bufferSize; i++) {\n    imag[i] *= -1;\n  }\n\n  var revReal = new Float32Array(bufferSize);\n  var revImag = new Float32Array(bufferSize);\n \n  for (i = 0; i < real.length; i++) {\n    revReal[i] = real[reverseTable[i]];\n    revImag[i] = imag[reverseTable[i]];\n  }\n \n  real = revReal;\n  imag = revImag;\n\n  while (halfSize < bufferSize) {\n    phaseShiftStepReal = cosTable[halfSize];\n    phaseShiftStepImag = sinTable[halfSize];\n    currentPhaseShiftReal = 1;\n    currentPhaseShiftImag = 0;\n\n    for (var fftStep = 0; fftStep < halfSize; fftStep++) {\n      i = fftStep;\n\n      while (i < bufferSize) {\n        off = i + halfSize;\n        tr = (currentPhaseShiftReal * real[off]) - (currentPhaseShiftImag * imag[off]);\n        ti = (currentPhaseShiftReal * imag[off]) + (currentPhaseShiftImag * real[off]);\n\n        real[off] = real[i] - tr;\n        imag[off] = imag[i] - ti;\n        real[i] += tr;\n        imag[i] += ti;\n\n        i += halfSize << 1;\n      }\n\n      tmpReal = currentPhaseShiftReal;\n      currentPhaseShiftReal = (tmpReal * phaseShiftStepReal) - (currentPhaseShiftImag * phaseShiftStepImag);\n      currentPhaseShiftImag = (tmpReal * phaseShiftStepImag) + (currentPhaseShiftImag * phaseShiftStepReal);\n    }\n\n    halfSize = halfSize << 1;\n  }\n\n  var buffer = new Float32Array(bufferSize); // this should be reused instead\n  for (i = 0; i < bufferSize; i++) {\n    buffer[i] = real[i] / bufferSize;\n  }\n\n  return buffer;\n};\n\n/**\n * RFFT is a class for calculating the Discrete Fourier Transform of a signal\n * with the Fast Fourier Transform algorithm.\n *\n * This method currently only contains a forward transform but is highly optimized.\n *\n * @param {Number} bufferSize The size of the sample buffer to be computed. Must be power of 2\n * @param {Number} sampleRate The sampleRate of the buffer (eg. 44100)\n *\n * @constructor\n */\n\n// lookup tables don't really gain us any speed, but they do increase\n// cache footprint, so don't use them in here\n\n// also we don't use sepearate arrays for real/imaginary parts\n\n// this one a little more than twice as fast as the one in FFT\n// however I only did the forward transform\n\n// the rest of this was translated from C, see http://www.jjj.de/fxt/\n// this is the real split radix FFT\n\nfunction RFFT(bufferSize, sampleRate) {\n  FourierTransform.call(this, bufferSize, sampleRate);\n\n  this.trans = new Float32Array(bufferSize);\n\n  this.reverseTable = new Uint32Array(bufferSize);\n\n  // don't use a lookup table to do the permute, use this instead\n  this.reverseBinPermute = function (dest, source) {\n    var bufferSize  = this.bufferSize, \n        halfSize    = bufferSize >>> 1, \n        nm1         = bufferSize - 1, \n        i = 1, r = 0, h;\n\n    dest[0] = source[0];\n\n    do {\n      r += halfSize;\n      dest[i] = source[r];\n      dest[r] = source[i];\n      \n      i++;\n\n      h = halfSize << 1;\n      while (h = h >> 1, !((r ^= h) & h));\n\n      if (r >= i) { \n        dest[i]     = source[r]; \n        dest[r]     = source[i];\n\n        dest[nm1-i] = source[nm1-r]; \n        dest[nm1-r] = source[nm1-i];\n      }\n      i++;\n    } while (i < halfSize);\n    dest[nm1] = source[nm1];\n  };\n\n  this.generateReverseTable = function () {\n    var bufferSize  = this.bufferSize, \n        halfSize    = bufferSize >>> 1, \n        nm1         = bufferSize - 1, \n        i = 1, r = 0, h;\n\n    this.reverseTable[0] = 0;\n\n    do {\n      r += halfSize;\n      \n      this.reverseTable[i] = r;\n      this.reverseTable[r] = i;\n\n      i++;\n\n      h = halfSize << 1;\n      while (h = h >> 1, !((r ^= h) & h));\n\n      if (r >= i) { \n        this.reverseTable[i] = r;\n        this.reverseTable[r] = i;\n\n        this.reverseTable[nm1-i] = nm1-r;\n        this.reverseTable[nm1-r] = nm1-i;\n      }\n      i++;\n    } while (i < halfSize);\n\n    this.reverseTable[nm1] = nm1;\n  };\n\n  this.generateReverseTable();\n}\n\n\n// Ordering of output:\n//\n// trans[0]     = re[0] (==zero frequency, purely real)\n// trans[1]     = re[1]\n//             ...\n// trans[n/2-1] = re[n/2-1]\n// trans[n/2]   = re[n/2]    (==nyquist frequency, purely real)\n//\n// trans[n/2+1] = im[n/2-1]\n// trans[n/2+2] = im[n/2-2]\n//             ...\n// trans[n-1]   = im[1] \n\nRFFT.prototype.forward = function(buffer) {\n  var n         = this.bufferSize, \n      spectrum  = this.spectrum,\n      x         = this.trans, \n      TWO_PI    = 2*Math.PI,\n      sqrt      = Math.sqrt,\n      i         = n >>> 1,\n      bSi       = 2 / n,\n      n2, n4, n8, nn, \n      t1, t2, t3, t4, \n      i1, i2, i3, i4, i5, i6, i7, i8, \n      st1, cc1, ss1, cc3, ss3,\n      e, \n      a,\n      rval, ival, mag; \n\n  this.reverseBinPermute(x, buffer);\n\n  /*\n  var reverseTable = this.reverseTable;\n\n  for (var k = 0, len = reverseTable.length; k < len; k++) {\n    x[k] = buffer[reverseTable[k]];\n  }\n  */\n\n  for (var ix = 0, id = 4; ix < n; id *= 4) {\n    for (var i0 = ix; i0 < n; i0 += id) {\n      //sumdiff(x[i0], x[i0+1]); // {a, b}  <--| {a+b, a-b}\n      st1 = x[i0] - x[i0+1];\n      x[i0] += x[i0+1];\n      x[i0+1] = st1;\n    } \n    ix = 2*(id-1);\n  }\n\n  n2 = 2;\n  nn = n >>> 1;\n\n  while((nn = nn >>> 1)) {\n    ix = 0;\n    n2 = n2 << 1;\n    id = n2 << 1;\n    n4 = n2 >>> 2;\n    n8 = n2 >>> 3;\n    do {\n      if(n4 !== 1) {\n        for(i0 = ix; i0 < n; i0 += id) {\n          i1 = i0;\n          i2 = i1 + n4;\n          i3 = i2 + n4;\n          i4 = i3 + n4;\n     \n          //diffsum3_r(x[i3], x[i4], t1); // {a, b, s} <--| {a, b-a, a+b}\n          t1 = x[i3] + x[i4];\n          x[i4] -= x[i3];\n          //sumdiff3(x[i1], t1, x[i3]);   // {a, b, d} <--| {a+b, b, a-b}\n          x[i3] = x[i1] - t1; \n          x[i1] += t1;\n     \n          i1 += n8;\n          i2 += n8;\n          i3 += n8;\n          i4 += n8;\n         \n          //sumdiff(x[i3], x[i4], t1, t2); // {s, d}  <--| {a+b, a-b}\n          t1 = x[i3] + x[i4];\n          t2 = x[i3] - x[i4];\n         \n          t1 = -t1 * Math.SQRT1_2;\n          t2 *= Math.SQRT1_2;\n     \n          // sumdiff(t1, x[i2], x[i4], x[i3]); // {s, d}  <--| {a+b, a-b}\n          st1 = x[i2];\n          x[i4] = t1 + st1; \n          x[i3] = t1 - st1;\n          \n          //sumdiff3(x[i1], t2, x[i2]); // {a, b, d} <--| {a+b, b, a-b}\n          x[i2] = x[i1] - t2;\n          x[i1] += t2;\n        }\n      } else {\n        for(i0 = ix; i0 < n; i0 += id) {\n          i1 = i0;\n          i2 = i1 + n4;\n          i3 = i2 + n4;\n          i4 = i3 + n4;\n     \n          //diffsum3_r(x[i3], x[i4], t1); // {a, b, s} <--| {a, b-a, a+b}\n          t1 = x[i3] + x[i4]; \n          x[i4] -= x[i3];\n          \n          //sumdiff3(x[i1], t1, x[i3]);   // {a, b, d} <--| {a+b, b, a-b}\n          x[i3] = x[i1] - t1; \n          x[i1] += t1;\n        }\n      }\n   \n      ix = (id << 1) - n2;\n      id = id << 2;\n    } while (ix < n);\n \n    e = TWO_PI / n2;\n\n    for (var j = 1; j < n8; j++) {\n      a = j * e;\n      ss1 = Math.sin(a);\n      cc1 = Math.cos(a);\n\n      //ss3 = sin(3*a); cc3 = cos(3*a);\n      cc3 = 4*cc1*(cc1*cc1-0.75);\n      ss3 = 4*ss1*(0.75-ss1*ss1);\n   \n      ix = 0; id = n2 << 1;\n      do {\n        for (i0 = ix; i0 < n; i0 += id) {\n          i1 = i0 + j;\n          i2 = i1 + n4;\n          i3 = i2 + n4;\n          i4 = i3 + n4;\n       \n          i5 = i0 + n4 - j;\n          i6 = i5 + n4;\n          i7 = i6 + n4;\n          i8 = i7 + n4;\n       \n          //cmult(c, s, x, y, &u, &v)\n          //cmult(cc1, ss1, x[i7], x[i3], t2, t1); // {u,v} <--| {x*c-y*s, x*s+y*c}\n          t2 = x[i7]*cc1 - x[i3]*ss1; \n          t1 = x[i7]*ss1 + x[i3]*cc1;\n          \n          //cmult(cc3, ss3, x[i8], x[i4], t4, t3);\n          t4 = x[i8]*cc3 - x[i4]*ss3; \n          t3 = x[i8]*ss3 + x[i4]*cc3;\n       \n          //sumdiff(t2, t4);   // {a, b} <--| {a+b, a-b}\n          st1 = t2 - t4;\n          t2 += t4;\n          t4 = st1;\n          \n          //sumdiff(t2, x[i6], x[i8], x[i3]); // {s, d}  <--| {a+b, a-b}\n          //st1 = x[i6]; x[i8] = t2 + st1; x[i3] = t2 - st1;\n          x[i8] = t2 + x[i6]; \n          x[i3] = t2 - x[i6];\n         \n          //sumdiff_r(t1, t3); // {a, b} <--| {a+b, b-a}\n          st1 = t3 - t1;\n          t1 += t3;\n          t3 = st1;\n          \n          //sumdiff(t3, x[i2], x[i4], x[i7]); // {s, d}  <--| {a+b, a-b}\n          //st1 = x[i2]; x[i4] = t3 + st1; x[i7] = t3 - st1;\n          x[i4] = t3 + x[i2]; \n          x[i7] = t3 - x[i2];\n         \n          //sumdiff3(x[i1], t1, x[i6]);   // {a, b, d} <--| {a+b, b, a-b}\n          x[i6] = x[i1] - t1; \n          x[i1] += t1;\n          \n          //diffsum3_r(t4, x[i5], x[i2]); // {a, b, s} <--| {a, b-a, a+b}\n          x[i2] = t4 + x[i5]; \n          x[i5] -= t4;\n        }\n     \n        ix = (id << 1) - n2;\n        id = id << 2;\n   \n      } while (ix < n);\n    }\n  }\n\n  while (--i) {\n    rval = x[i];\n    ival = x[n-i-1];\n    mag = bSi * sqrt(rval * rval + ival * ival);\n\n    if (mag > this.peak) {\n      this.peakBand = i;\n      this.peak = mag;\n    }\n\n    spectrum[i] = mag;\n  }\n\n  spectrum[0] = bSi * x[0];\n\n  return spectrum;\n};\n\nfunction Sampler(file, bufferSize, sampleRate, playStart, playEnd, loopStart, loopEnd, loopMode) {\n  this.file = file;\n  this.bufferSize = bufferSize;\n  this.sampleRate = sampleRate;\n  this.playStart  = playStart || 0; // 0%\n  this.playEnd    = playEnd   || 1; // 100%\n  this.loopStart  = loopStart || 0;\n  this.loopEnd    = loopEnd   || 1;\n  this.loopMode   = loopMode  || DSP.OFF;\n  this.loaded     = false;\n  this.samples    = [];\n  this.signal     = new Float32Array(bufferSize);\n  this.frameCount = 0;\n  this.envelope   = null;\n  this.amplitude  = 1;\n  this.rootFrequency = 110; // A2 110\n  this.frequency  = 550;\n  this.step       = this.frequency / this.rootFrequency;\n  this.duration   = 0;\n  this.samplesProcessed = 0;\n  this.playhead   = 0;\n \n  var audio = /* new Audio();*/ document.createElement(\"AUDIO\");\n  var self = this;\n \n  this.loadSamples = function(event) {\n    var buffer = DSP.getChannel(DSP.MIX, event.frameBuffer);\n    for ( var i = 0; i < buffer.length; i++) {\n      self.samples.push(buffer[i]);\n    }\n  };\n \n  this.loadComplete = function() {\n    // convert flexible js array into a fast typed array\n    self.samples = new Float32Array(self.samples);\n    self.loaded = true;\n  };\n \n  this.loadMetaData = function() {\n    self.duration = audio.duration;\n  };\n \n  audio.addEventListener(\"MozAudioAvailable\", this.loadSamples, false);\n  audio.addEventListener(\"loadedmetadata\", this.loadMetaData, false);\n  audio.addEventListener(\"ended\", this.loadComplete, false);\n  audio.muted = true;\n  audio.src = file;\n  audio.play();\n}\n\nSampler.prototype.applyEnvelope = function() {\n  this.envelope.process(this.signal);\n  return this.signal;\n};\n\nSampler.prototype.generate = function() {\n  var frameOffset = this.frameCount * this.bufferSize;\n \n  var loopWidth = this.playEnd * this.samples.length - this.playStart * this.samples.length;\n  var playStartSamples = this.playStart * this.samples.length; // ie 0.5 -> 50% of the length\n  var playEndSamples = this.playEnd * this.samples.length; // ie 0.5 -> 50% of the length\n  var offset;\n\n  for ( var i = 0; i < this.bufferSize; i++ ) {\n    switch (this.loopMode) {\n      case DSP.OFF:\n        this.playhead = Math.round(this.samplesProcessed * this.step + playStartSamples);\n        if (this.playhead < (this.playEnd * this.samples.length) ) {\n          this.signal[i] = this.samples[this.playhead] * this.amplitude;\n        } else {\n          this.signal[i] = 0;\n        }\n        break;\n     \n      case DSP.FW:\n        this.playhead = Math.round((this.samplesProcessed * this.step) % loopWidth + playStartSamples);\n        if (this.playhead < (this.playEnd * this.samples.length) ) {\n          this.signal[i] = this.samples[this.playhead] * this.amplitude;\n        }\n        break;\n       \n      case DSP.BW:\n        this.playhead = playEndSamples - Math.round((this.samplesProcessed * this.step) % loopWidth);\n        if (this.playhead < (this.playEnd * this.samples.length) ) {\n          this.signal[i] = this.samples[this.playhead] * this.amplitude;\n        }\n        break;\n       \n      case DSP.FWBW:\n        if ( Math.floor(this.samplesProcessed * this.step / loopWidth) % 2 === 0 ) {\n          this.playhead = Math.round((this.samplesProcessed * this.step) % loopWidth + playStartSamples);\n        } else {\n          this.playhead = playEndSamples - Math.round((this.samplesProcessed * this.step) % loopWidth);\n        }  \n        if (this.playhead < (this.playEnd * this.samples.length) ) {\n          this.signal[i] = this.samples[this.playhead] * this.amplitude;\n        }\n        break;\n    }\n    this.samplesProcessed++;\n  }\n\n  this.frameCount++;\n\n  return this.signal;\n};\n\nSampler.prototype.setFreq = function(frequency) {\n    var totalProcessed = this.samplesProcessed * this.step;\n    this.frequency = frequency;\n    this.step = this.frequency / this.rootFrequency;\n    this.samplesProcessed = Math.round(totalProcessed/this.step);\n};\n\nSampler.prototype.reset = function() {\n  this.samplesProcessed = 0;\n  this.playhead = 0;\n};\n\n/**\n * Oscillator class for generating and modifying signals\n *\n * @param {Number} type       A waveform constant (eg. DSP.SINE)\n * @param {Number} frequency  Initial frequency of the signal\n * @param {Number} amplitude  Initial amplitude of the signal\n * @param {Number} bufferSize Size of the sample buffer to generate\n * @param {Number} sampleRate The sample rate of the signal\n *\n * @contructor\n */\nfunction Oscillator(type, frequency, amplitude, bufferSize, sampleRate) {\n  this.frequency  = frequency;\n  this.amplitude  = amplitude;\n  this.bufferSize = bufferSize;\n  this.sampleRate = sampleRate;\n  //this.pulseWidth = pulseWidth;\n  this.frameCount = 0;\n \n  this.waveTableLength = 2048;\n\n  this.cyclesPerSample = frequency / sampleRate;\n\n  this.signal = new Float32Array(bufferSize);\n  this.envelope = null;\n\n  switch(parseInt(type, 10)) {\n    case DSP.TRIANGLE:\n      this.func = Oscillator.Triangle;\n      break;\n\n    case DSP.SAW:\n      this.func = Oscillator.Saw;\n      break;\n\n    case DSP.SQUARE:\n      this.func = Oscillator.Square;\n      break;\n\n    default:\n    case DSP.SINE:\n      this.func = Oscillator.Sine;\n      break;\n  }\n\n  this.generateWaveTable = function() {\n    Oscillator.waveTable[this.func] = new Float32Array(2048);\n    var waveTableTime = this.waveTableLength / this.sampleRate;\n    var waveTableHz = 1 / waveTableTime;\n\n    for (var i = 0; i < this.waveTableLength; i++) {\n      Oscillator.waveTable[this.func][i] = this.func(i * waveTableHz/this.sampleRate);\n    }\n  };\n\n  if ( typeof Oscillator.waveTable === 'undefined' ) {\n    Oscillator.waveTable = {};\n  }\n\n  if ( typeof Oscillator.waveTable[this.func] === 'undefined' ) {\n    this.generateWaveTable();\n  }\n \n  this.waveTable = Oscillator.waveTable[this.func];\n}\n\n/**\n * Set the amplitude of the signal\n *\n * @param {Number} amplitude The amplitude of the signal (between 0 and 1)\n */\nOscillator.prototype.setAmp = function(amplitude) {\n  if (amplitude >= 0 && amplitude <= 1) {\n    this.amplitude = amplitude;\n  } else {\n    throw \"Amplitude out of range (0..1).\";\n  }\n};\n  \n/**\n * Set the frequency of the signal\n *\n * @param {Number} frequency The frequency of the signal\n */  \nOscillator.prototype.setFreq = function(frequency) {\n  this.frequency = frequency;\n  this.cyclesPerSample = frequency / this.sampleRate;\n};\n     \n// Add an oscillator\nOscillator.prototype.add = function(oscillator) {\n  for ( var i = 0; i < this.bufferSize; i++ ) {\n    //this.signal[i] += oscillator.valueAt(i);\n    this.signal[i] += oscillator.signal[i];\n  }\n \n  return this.signal;\n};\n     \n// Add a signal to the current generated osc signal\nOscillator.prototype.addSignal = function(signal) {\n  for ( var i = 0; i < signal.length; i++ ) {\n    if ( i >= this.bufferSize ) {\n      break;\n    }\n    this.signal[i] += signal[i];\n   \n    /*\n    // Constrain amplitude\n    if ( this.signal[i] > 1 ) {\n      this.signal[i] = 1;\n    } else if ( this.signal[i] < -1 ) {\n      this.signal[i] = -1;\n    }\n    */\n  }\n  return this.signal;\n};\n     \n// Add an envelope to the oscillator\nOscillator.prototype.addEnvelope = function(envelope) {\n  this.envelope = envelope;\n};\n\nOscillator.prototype.applyEnvelope = function() {\n  this.envelope.process(this.signal);\n};\n     \nOscillator.prototype.valueAt = function(offset) {\n  return this.waveTable[offset % this.waveTableLength];\n};\n     \nOscillator.prototype.generate = function() {\n  var frameOffset = this.frameCount * this.bufferSize;\n  var step = this.waveTableLength * this.frequency / this.sampleRate;\n  var offset;\n\n  for ( var i = 0; i < this.bufferSize; i++ ) {\n    //var step = (frameOffset + i) * this.cyclesPerSample % 1;\n    //this.signal[i] = this.func(step) * this.amplitude;\n    //this.signal[i] = this.valueAt(Math.round((frameOffset + i) * step)) * this.amplitude;\n    offset = Math.round((frameOffset + i) * step);\n    this.signal[i] = this.waveTable[offset % this.waveTableLength] * this.amplitude;\n  }\n\n  this.frameCount++;\n\n  return this.signal;\n};\n\nOscillator.Sine = function(step) {\n  return Math.sin(DSP.TWO_PI * step);\n};\n\nOscillator.Square = function(step) {\n  return step < 0.5 ? 1 : -1;\n};\n\nOscillator.Saw = function(step) {\n  return 2 * (step - Math.round(step));\n};\n\nOscillator.Triangle = function(step) {\n  return 1 - 4 * Math.abs(Math.round(step) - step);\n};\n\nOscillator.Pulse = function(step) {\n  // stub\n};\n \nfunction ADSR(attackLength, decayLength, sustainLevel, sustainLength, releaseLength, sampleRate) {\n  this.sampleRate = sampleRate;\n  // Length in seconds\n  this.attackLength  = attackLength;\n  this.decayLength   = decayLength;\n  this.sustainLevel  = sustainLevel;\n  this.sustainLength = sustainLength;\n  this.releaseLength = releaseLength;\n  this.sampleRate    = sampleRate;\n \n  // Length in samples\n  this.attackSamples  = attackLength  * sampleRate;\n  this.decaySamples   = decayLength   * sampleRate;\n  this.sustainSamples = sustainLength * sampleRate;\n  this.releaseSamples = releaseLength * sampleRate;\n \n  // Updates the envelope sample positions\n  this.update = function() {\n    this.attack         =                this.attackSamples;\n    this.decay          = this.attack  + this.decaySamples;\n    this.sustain        = this.decay   + this.sustainSamples;\n    this.release        = this.sustain + this.releaseSamples;\n  };\n \n  this.update();\n \n  this.samplesProcessed = 0;\n}\n\nADSR.prototype.noteOn = function() {\n  this.samplesProcessed = 0;\n  this.sustainSamples = this.sustainLength * this.sampleRate;\n  this.update();\n};\n\n// Send a note off when using a sustain of infinity to let the envelope enter the release phase\nADSR.prototype.noteOff = function() {\n  this.sustainSamples = this.samplesProcessed - this.decaySamples;\n  this.update();\n};\n\nADSR.prototype.processSample = function(sample) {\n  var amplitude = 0;\n\n  if ( this.samplesProcessed <= this.attack ) {\n    amplitude = 0 + (1 - 0) * ((this.samplesProcessed - 0) / (this.attack - 0));\n  } else if ( this.samplesProcessed > this.attack && this.samplesProcessed <= this.decay ) {\n    amplitude = 1 + (this.sustainLevel - 1) * ((this.samplesProcessed - this.attack) / (this.decay - this.attack));\n  } else if ( this.samplesProcessed > this.decay && this.samplesProcessed <= this.sustain ) {\n    amplitude = this.sustainLevel;\n  } else if ( this.samplesProcessed > this.sustain && this.samplesProcessed <= this.release ) {\n    amplitude = this.sustainLevel + (0 - this.sustainLevel) * ((this.samplesProcessed - this.sustain) / (this.release - this.sustain));\n  }\n \n  return sample * amplitude;\n};\n\nADSR.prototype.value = function() {\n  var amplitude = 0;\n\n  if ( this.samplesProcessed <= this.attack ) {\n    amplitude = 0 + (1 - 0) * ((this.samplesProcessed - 0) / (this.attack - 0));\n  } else if ( this.samplesProcessed > this.attack && this.samplesProcessed <= this.decay ) {\n    amplitude = 1 + (this.sustainLevel - 1) * ((this.samplesProcessed - this.attack) / (this.decay - this.attack));\n  } else if ( this.samplesProcessed > this.decay && this.samplesProcessed <= this.sustain ) {\n    amplitude = this.sustainLevel;\n  } else if ( this.samplesProcessed > this.sustain && this.samplesProcessed <= this.release ) {\n    amplitude = this.sustainLevel + (0 - this.sustainLevel) * ((this.samplesProcessed - this.sustain) / (this.release - this.sustain));\n  }\n \n  return amplitude;\n};\n     \nADSR.prototype.process = function(buffer) {\n  for ( var i = 0; i < buffer.length; i++ ) {\n    buffer[i] *= this.value();\n\n    this.samplesProcessed++;\n  }\n \n  return buffer;\n};\n     \n     \nADSR.prototype.isActive = function() {\n  if ( this.samplesProcessed > this.release || this.samplesProcessed === -1 ) {\n    return false;\n  } else {\n    return true;\n  }\n};\n\nADSR.prototype.disable = function() {\n  this.samplesProcessed = -1;\n};\n \nfunction IIRFilter(type, cutoff, resonance, sampleRate) {\n  this.sampleRate = sampleRate;\n\n  switch(type) {\n    case DSP.LOWPASS:\n    case DSP.LP12:\n      this.func = new IIRFilter.LP12(cutoff, resonance, sampleRate);\n      break;\n  }\n}\n\nIIRFilter.prototype.__defineGetter__('cutoff',\n  function() {\n    return this.func.cutoff;\n  }\n);\n\nIIRFilter.prototype.__defineGetter__('resonance',\n  function() {\n    return this.func.resonance;\n  }\n);\n\nIIRFilter.prototype.set = function(cutoff, resonance) {\n  this.func.calcCoeff(cutoff, resonance);\n};\n\nIIRFilter.prototype.process = function(buffer) {\n  this.func.process(buffer);\n};\n\n// Add an envelope to the filter\nIIRFilter.prototype.addEnvelope = function(envelope) {\n  if ( envelope instanceof ADSR ) {\n    this.func.addEnvelope(envelope);\n  } else {\n    throw \"Not an envelope.\";\n  }\n};\n\nIIRFilter.LP12 = function(cutoff, resonance, sampleRate) {\n  this.sampleRate = sampleRate;\n  this.vibraPos   = 0;\n  this.vibraSpeed = 0;\n  this.envelope = false;\n \n  this.calcCoeff = function(cutoff, resonance) {\n    this.w = 2.0 * Math.PI * cutoff / this.sampleRate;\n    this.q = 1.0 - this.w / (2.0 * (resonance + 0.5 / (1.0 + this.w)) + this.w - 2.0);\n    this.r = this.q * this.q;\n    this.c = this.r + 1.0 - 2.0 * Math.cos(this.w) * this.q;\n   \n    this.cutoff = cutoff;\n    this.resonance = resonance;\n  };\n\n  this.calcCoeff(cutoff, resonance);\n\n  this.process = function(buffer) {\n    for ( var i = 0; i < buffer.length; i++ ) {\n      this.vibraSpeed += (buffer[i] - this.vibraPos) * this.c;\n      this.vibraPos   += this.vibraSpeed;\n      this.vibraSpeed *= this.r;\n   \n      /*\n      var temp = this.vibraPos;\n     \n      if ( temp > 1.0 ) {\n        temp = 1.0;\n      } else if ( temp < -1.0 ) {\n        temp = -1.0;\n      } else if ( temp != temp ) {\n        temp = 1;\n      }\n     \n      buffer[i] = temp;\n      */\n\n      if (this.envelope) {\n        buffer[i] = (buffer[i] * (1 - this.envelope.value())) + (this.vibraPos * this.envelope.value());\n        this.envelope.samplesProcessed++;\n      } else {\n        buffer[i] = this.vibraPos;\n      }\n    }\n  };\n}; \n\nIIRFilter.LP12.prototype.addEnvelope = function(envelope) {\n  this.envelope = envelope;\n};\n\nfunction IIRFilter2(type, cutoff, resonance, sampleRate) {\n  this.type = type;\n  this.cutoff = cutoff;\n  this.resonance = resonance;\n  this.sampleRate = sampleRate;\n\n  this.f = Float32Array(4);\n  this.f[0] = 0.0; // lp\n  this.f[1] = 0.0; // hp\n  this.f[2] = 0.0; // bp\n  this.f[3] = 0.0; // br \n \n  this.calcCoeff = function(cutoff, resonance) {\n    this.freq = 2 * Math.sin(Math.PI * Math.min(0.25, cutoff/(this.sampleRate*2)));  \n    this.damp = Math.min(2 * (1 - Math.pow(resonance, 0.25)), Math.min(2, 2/this.freq - this.freq * 0.5));\n  };\n\n  this.calcCoeff(cutoff, resonance);\n}\n\nIIRFilter2.prototype.process = function(buffer) {\n  var input, output;\n  var f = this.f;\n\n  for ( var i = 0; i < buffer.length; i++ ) {\n    input = buffer[i];\n\n    // first pass\n    f[3] = input - this.damp * f[2];\n    f[0] = f[0] + this.freq * f[2];\n    f[1] = f[3] - f[0];\n    f[2] = this.freq * f[1] + f[2];\n    output = 0.5 * f[this.type];\n\n    // second pass\n    f[3] = input - this.damp * f[2];\n    f[0] = f[0] + this.freq * f[2];\n    f[1] = f[3] - f[0];\n    f[2] = this.freq * f[1] + f[2];\n    output += 0.5 * f[this.type];\n\n    if (this.envelope) {\n      buffer[i] = (buffer[i] * (1 - this.envelope.value())) + (output * this.envelope.value());\n      this.envelope.samplesProcessed++;\n    } else {\n      buffer[i] = output;\n    }\n  }\n};\n\nIIRFilter2.prototype.addEnvelope = function(envelope) {\n  if ( envelope instanceof ADSR ) {\n    this.envelope = envelope;\n  } else {\n    throw \"This is not an envelope.\";\n  }\n};\n\nIIRFilter2.prototype.set = function(cutoff, resonance) {\n  this.calcCoeff(cutoff, resonance);\n};\n\n\n\nfunction WindowFunction(type, alpha) {\n  this.alpha = alpha;\n \n  switch(type) {\n    case DSP.BARTLETT:\n      this.func = WindowFunction.Bartlett;\n      break;\n     \n    case DSP.BARTLETTHANN:\n      this.func = WindowFunction.BartlettHann;\n      break;\n     \n    case DSP.BLACKMAN:\n      this.func = WindowFunction.Blackman;\n      this.alpha = this.alpha || 0.16;\n      break;\n   \n    case DSP.COSINE:\n      this.func = WindowFunction.Cosine;\n      break;\n     \n    case DSP.GAUSS:\n      this.func = WindowFunction.Gauss;\n      this.alpha = this.alpha || 0.25;\n      break;\n     \n    case DSP.HAMMING:\n      this.func = WindowFunction.Hamming;\n      break;\n     \n    case DSP.HANN:\n      this.func = WindowFunction.Hann;\n      break;\n   \n    case DSP.LANCZOS:\n      this.func = WindowFunction.Lanczoz;\n      break;\n     \n    case DSP.RECTANGULAR:\n      this.func = WindowFunction.Rectangular;\n      break;\n     \n    case DSP.TRIANGULAR:\n      this.func = WindowFunction.Triangular;\n      break;\n  }\n}\n\nWindowFunction.prototype.process = function(buffer) {\n  var length = buffer.length;\n  for ( var i = 0; i < length; i++ ) {\n    buffer[i] *= this.func(length, i, this.alpha);\n  }\n  return buffer;\n};\n\nWindowFunction.Bartlett = function(length, index) {\n  return 2 / (length - 1) * ((length - 1) / 2 - Math.abs(index - (length - 1) / 2));\n};\n\nWindowFunction.BartlettHann = function(length, index) {\n  return 0.62 - 0.48 * Math.abs(index / (length - 1) - 0.5) - 0.38 * Math.cos(DSP.TWO_PI * index / (length - 1));\n};\n\nWindowFunction.Blackman = function(length, index, alpha) {\n  var a0 = (1 - alpha) / 2;\n  var a1 = 0.5;\n  var a2 = alpha / 2;\n\n  return a0 - a1 * Math.cos(DSP.TWO_PI * index / (length - 1)) + a2 * Math.cos(4 * Math.PI * index / (length - 1));\n};\n\nWindowFunction.Cosine = function(length, index) {\n  return Math.cos(Math.PI * index / (length - 1) - Math.PI / 2);\n};\n\nWindowFunction.Gauss = function(length, index, alpha) {\n  return Math.pow(Math.E, -0.5 * Math.pow((index - (length - 1) / 2) / (alpha * (length - 1) / 2), 2));\n};\n\nWindowFunction.Hamming = function(length, index) {\n  return 0.54 - 0.46 * Math.cos(DSP.TWO_PI * index / (length - 1));\n};\n\nWindowFunction.Hann = function(length, index) {\n  return 0.5 * (1 - Math.cos(DSP.TWO_PI * index / (length - 1)));\n};\n\nWindowFunction.Lanczos = function(length, index) {\n  var x = 2 * index / (length - 1) - 1;\n  return Math.sin(Math.PI * x) / (Math.PI * x);\n};\n\nWindowFunction.Rectangular = function(length, index) {\n  return 1;\n};\n\nWindowFunction.Triangular = function(length, index) {\n  return 2 / length * (length / 2 - Math.abs(index - (length - 1) / 2));\n};\n\nfunction sinh (arg) {\n  // Returns the hyperbolic sine of the number, defined as (exp(number) - exp(-number))/2 \n  //\n  // version: 1004.2314\n  // discuss at: http://phpjs.org/functions/sinh    // +   original by: Onno Marsman\n  // *     example 1: sinh(-0.9834330348825909);\n  // *     returns 1: -1.1497971402636502\n  return (Math.exp(arg) - Math.exp(-arg))/2;\n}\n\n/* \n *  Biquad filter\n * \n *  Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.\n *  Copyright 2010 Ricard Marxer. All rights reserved.\n *\n */\n// Implementation based on:\n// http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt\nfunction Biquad(type, sampleRate) {\n  this.Fs = sampleRate;\n  this.type = type;  // type of the filter\n  this.parameterType = DSP.Q; // type of the parameter\n\n  this.x_1_l = 0;\n  this.x_2_l = 0;\n  this.y_1_l = 0;\n  this.y_2_l = 0;\n\n  this.x_1_r = 0;\n  this.x_2_r = 0;\n  this.y_1_r = 0;\n  this.y_2_r = 0;\n\n  this.b0 = 1;\n  this.a0 = 1;\n\n  this.b1 = 0;\n  this.a1 = 0;\n\n  this.b2 = 0;\n  this.a2 = 0;\n\n  this.b0a0 = this.b0 / this.a0;\n  this.b1a0 = this.b1 / this.a0;\n  this.b2a0 = this.b2 / this.a0;\n  this.a1a0 = this.a1 / this.a0;\n  this.a2a0 = this.a2 / this.a0;\n\n  this.f0 = 3000;   // \"wherever it's happenin', man.\"  Center Frequency or\n                    // Corner Frequency, or shelf midpoint frequency, depending\n                    // on which filter type.  The \"significant frequency\".\n\n  this.dBgain = 12; // used only for peaking and shelving filters\n\n  this.Q = 1;       // the EE kind of definition, except for peakingEQ in which A*Q is\n                    // the classic EE Q.  That adjustment in definition was made so that\n                    // a boost of N dB followed by a cut of N dB for identical Q and\n                    // f0/Fs results in a precisely flat unity gain filter or \"wire\".\n\n  this.BW = -3;     // the bandwidth in octaves (between -3 dB frequencies for BPF\n                    // and notch or between midpoint (dBgain/2) gain frequencies for\n                    // peaking EQ\n\n  this.S = 1;       // a \"shelf slope\" parameter (for shelving EQ only).  When S = 1,\n                    // the shelf slope is as steep as it can be and remain monotonically\n                    // increasing or decreasing gain with frequency.  The shelf slope, in\n                    // dB/octave, remains proportional to S for all other values for a\n                    // fixed f0/Fs and dBgain.\n\n  this.coefficients = function() {\n    var b = [this.b0, this.b1, this.b2];\n    var a = [this.a0, this.a1, this.a2];\n    return {b: b, a:a};\n  };\n\n  this.setFilterType = function(type) {\n    this.type = type;\n    this.recalculateCoefficients();\n  };\n\n  this.setSampleRate = function(rate) {\n    this.Fs = rate;\n    this.recalculateCoefficients();\n  };\n\n  this.setQ = function(q) {\n    this.parameterType = DSP.Q;\n    this.Q = Math.max(Math.min(q, 115.0), 0.001);\n    this.recalculateCoefficients();\n  };\n\n  this.setBW = function(bw) {\n    this.parameterType = DSP.BW;\n    this.BW = bw;\n    this.recalculateCoefficients();\n  };\n\n  this.setS = function(s) {\n    this.parameterType = DSP.S;\n    this.S = Math.max(Math.min(s, 5.0), 0.0001);\n    this.recalculateCoefficients();\n  };\n\n  this.setF0 = function(freq) {\n    this.f0 = freq;\n    this.recalculateCoefficients();\n  }; \n \n  this.setDbGain = function(g) {\n    this.dBgain = g;\n    this.recalculateCoefficients();\n  };\n\n  this.recalculateCoefficients = function() {\n    var A;\n    if (type === DSP.PEAKING_EQ || type === DSP.LOW_SHELF || type === DSP.HIGH_SHELF ) {\n      A = Math.pow(10, (this.dBgain/40));  // for peaking and shelving EQ filters only\n    } else {\n      A  = Math.sqrt( Math.pow(10, (this.dBgain/20)) );   \n    }\n\n    var w0 = DSP.TWO_PI * this.f0 / this.Fs;\n\n    var cosw0 = Math.cos(w0);\n    var sinw0 = Math.sin(w0);\n\n    var alpha = 0;\n   \n    switch (this.parameterType) {\n      case DSP.Q:\n        alpha = sinw0/(2*this.Q);\n        break;\n           \n      case DSP.BW:\n        alpha = sinw0 * sinh( Math.LN2/2 * this.BW * w0/sinw0 );\n        break;\n\n      case DSP.S:\n        alpha = sinw0/2 * Math.sqrt( (A + 1/A)*(1/this.S - 1) + 2 );\n        break;\n    }\n\n    /**\n        FYI: The relationship between bandwidth and Q is\n             1/Q = 2*sinh(ln(2)/2*BW*w0/sin(w0))     (digital filter w BLT)\n        or   1/Q = 2*sinh(ln(2)/2*BW)             (analog filter prototype)\n\n        The relationship between shelf slope and Q is\n             1/Q = sqrt((A + 1/A)*(1/S - 1) + 2)\n    */\n\n    var coeff;\n\n    switch (this.type) {\n      case DSP.LPF:       // H(s) = 1 / (s^2 + s/Q + 1)\n        this.b0 =  (1 - cosw0)/2;\n        this.b1 =   1 - cosw0;\n        this.b2 =  (1 - cosw0)/2;\n        this.a0 =   1 + alpha;\n        this.a1 =  -2 * cosw0;\n        this.a2 =   1 - alpha;\n        break;\n\n      case DSP.HPF:       // H(s) = s^2 / (s^2 + s/Q + 1)\n        this.b0 =  (1 + cosw0)/2;\n        this.b1 = -(1 + cosw0);\n        this.b2 =  (1 + cosw0)/2;\n        this.a0 =   1 + alpha;\n        this.a1 =  -2 * cosw0;\n        this.a2 =   1 - alpha;\n        break;\n\n      case DSP.BPF_CONSTANT_SKIRT:       // H(s) = s / (s^2 + s/Q + 1)  (constant skirt gain, peak gain = Q)\n        this.b0 =   sinw0/2;\n        this.b1 =   0;\n        this.b2 =  -sinw0/2;\n        this.a0 =   1 + alpha;\n        this.a1 =  -2*cosw0;\n        this.a2 =   1 - alpha;\n        break;\n\n      case DSP.BPF_CONSTANT_PEAK:       // H(s) = (s/Q) / (s^2 + s/Q + 1)      (constant 0 dB peak gain)\n        this.b0 =   alpha;\n        this.b1 =   0;\n        this.b2 =  -alpha;\n        this.a0 =   1 + alpha;\n        this.a1 =  -2*cosw0;\n        this.a2 =   1 - alpha;\n        break;\n\n      case DSP.NOTCH:     // H(s) = (s^2 + 1) / (s^2 + s/Q + 1)\n        this.b0 =   1;\n        this.b1 =  -2*cosw0;\n        this.b2 =   1;\n        this.a0 =   1 + alpha;\n        this.a1 =  -2*cosw0;\n        this.a2 =   1 - alpha;\n        break;\n\n      case DSP.APF:       // H(s) = (s^2 - s/Q + 1) / (s^2 + s/Q + 1)\n        this.b0 =   1 - alpha;\n        this.b1 =  -2*cosw0;\n        this.b2 =   1 + alpha;\n        this.a0 =   1 + alpha;\n        this.a1 =  -2*cosw0;\n        this.a2 =   1 - alpha;\n        break;\n\n      case DSP.PEAKING_EQ:  // H(s) = (s^2 + s*(A/Q) + 1) / (s^2 + s/(A*Q) + 1)\n        this.b0 =   1 + alpha*A;\n        this.b1 =  -2*cosw0;\n        this.b2 =   1 - alpha*A;\n        this.a0 =   1 + alpha/A;\n        this.a1 =  -2*cosw0;\n        this.a2 =   1 - alpha/A;\n        break;\n\n      case DSP.LOW_SHELF:   // H(s) = A * (s^2 + (sqrt(A)/Q)*s + A)/(A*s^2 + (sqrt(A)/Q)*s + 1)\n        coeff = sinw0 * Math.sqrt( (A^2 + 1)*(1/this.S - 1) + 2*A );\n        this.b0 =    A*((A+1) - (A-1)*cosw0 + coeff);\n        this.b1 =  2*A*((A-1) - (A+1)*cosw0);\n        this.b2 =    A*((A+1) - (A-1)*cosw0 - coeff);\n        this.a0 =       (A+1) + (A-1)*cosw0 + coeff;\n        this.a1 =   -2*((A-1) + (A+1)*cosw0);\n        this.a2 =       (A+1) + (A-1)*cosw0 - coeff;\n        break;\n\n      case DSP.HIGH_SHELF:   // H(s) = A * (A*s^2 + (sqrt(A)/Q)*s + 1)/(s^2 + (sqrt(A)/Q)*s + A)\n        coeff = sinw0 * Math.sqrt( (A^2 + 1)*(1/this.S - 1) + 2*A );\n        this.b0 =    A*((A+1) + (A-1)*cosw0 + coeff);\n        this.b1 = -2*A*((A-1) + (A+1)*cosw0);\n        this.b2 =    A*((A+1) + (A-1)*cosw0 - coeff);\n        this.a0 =       (A+1) - (A-1)*cosw0 + coeff;\n        this.a1 =    2*((A-1) - (A+1)*cosw0);\n        this.a2 =       (A+1) - (A-1)*cosw0 - coeff;\n        break;\n    }\n   \n    this.b0a0 = this.b0/this.a0;\n    this.b1a0 = this.b1/this.a0;\n    this.b2a0 = this.b2/this.a0;\n    this.a1a0 = this.a1/this.a0;\n    this.a2a0 = this.a2/this.a0;\n  };\n\n  this.process = function(buffer) {\n      //y[n] = (b0/a0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2]\n      //       - (a1/a0)*y[n-1] - (a2/a0)*y[n-2]\n\n      var len = buffer.length;\n      var output = new Float32Array(len);\n\n      for ( var i=0; i<buffer.length; i++ ) {\n        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;\n        this.y_2_l = this.y_1_l;\n        this.y_1_l = output[i];\n        this.x_2_l = this.x_1_l;\n        this.x_1_l = buffer[i];\n      }\n\n      return output;\n  };\n\n  this.processStereo = function(buffer) {\n      //y[n] = (b0/a0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2]\n      //       - (a1/a0)*y[n-1] - (a2/a0)*y[n-2]\n\n      var len = buffer.length;\n      var output = new Float32Array(len);\n     \n      for (var i = 0; i < len/2; i++) {\n        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;\n        this.y_2_l = this.y_1_l;\n        this.y_1_l = output[2*i];\n        this.x_2_l = this.x_1_l;\n        this.x_1_l = buffer[2*i];\n\n        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;\n        this.y_2_r = this.y_1_r;\n        this.y_1_r = output[2*i+1];\n        this.x_2_r = this.x_1_r;\n        this.x_1_r = buffer[2*i+1];\n      }\n\n      return output;\n  };\n}\n\n/* \n *  Magnitude to decibels\n * \n *  Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.\n *  Copyright 2010 Ricard Marxer. All rights reserved.\n *\n *  @buffer array of magnitudes to convert to decibels\n *\n *  @returns the array in decibels\n *\n */\nDSP.mag2db = function(buffer) {\n  var minDb = -120;\n  var minMag = Math.pow(10.0, minDb / 20.0);\n\n  var log = Math.log;\n  var max = Math.max;\n \n  var result = Float32Array(buffer.length);\n  for (var i=0; i<buffer.length; i++) {\n    result[i] = 20.0*log(max(buffer[i], minMag));\n  }\n\n  return result;\n};\n\n/* \n *  Frequency response\n * \n *  Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.\n *  Copyright 2010 Ricard Marxer. All rights reserved.\n *\n *  Calculates the frequency response at the given points.\n *\n *  @b b coefficients of the filter\n *  @a a coefficients of the filter\n *  @w w points (normally between -PI and PI) where to calculate the frequency response\n *\n *  @returns the frequency response in magnitude\n *\n */\nDSP.freqz = function(b, a, w) {\n  var i, j;\n\n  if (!w) {\n    w = Float32Array(200);\n    for (i=0;i<w.length; i++) {\n      w[i] = DSP.TWO_PI/w.length * i - Math.PI;\n    }\n  }\n\n  var result = Float32Array(w.length);\n \n  var sqrt = Math.sqrt;\n  var cos = Math.cos;\n  var sin = Math.sin;\n \n  for (i=0; i<w.length; i++) {\n    var numerator = {real:0.0, imag:0.0};\n    for (j=0; j<b.length; j++) {\n      numerator.real += b[j] * cos(-j*w[i]);\n      numerator.imag += b[j] * sin(-j*w[i]);\n    }\n\n    var denominator = {real:0.0, imag:0.0};\n    for (j=0; j<a.length; j++) {\n      denominator.real += a[j] * cos(-j*w[i]);\n      denominator.imag += a[j] * sin(-j*w[i]);\n    }\n \n    result[i] =  sqrt(numerator.real*numerator.real + numerator.imag*numerator.imag) / sqrt(denominator.real*denominator.real + denominator.imag*denominator.imag);\n  }\n\n  return result;\n};\n\n/* \n *  Graphical Equalizer\n *\n *  Implementation of a graphic equalizer with a configurable bands-per-octave\n *  and minimum and maximum frequencies\n * \n *  Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.\n *  Copyright 2010 Ricard Marxer. All rights reserved.\n *\n */\nfunction GraphicalEq(sampleRate) {\n  this.FS = sampleRate;\n  this.minFreq = 40.0;\n  this.maxFreq = 16000.0;\n\n  this.bandsPerOctave = 1.0;\n\n  this.filters = [];\n  this.freqzs = [];\n\n  this.calculateFreqzs = true;\n\n  this.recalculateFilters = function() {\n    var bandCount = Math.round(Math.log(this.maxFreq/this.minFreq) * this.bandsPerOctave/ Math.LN2);\n\n    this.filters = [];\n    for (var i=0; i<bandCount; i++) {\n      var freq = this.minFreq*(Math.pow(2, i/this.bandsPerOctave));\n      var newFilter = new Biquad(DSP.PEAKING_EQ, this.FS);\n      newFilter.setDbGain(0);\n      newFilter.setBW(1/this.bandsPerOctave);\n      newFilter.setF0(freq);\n      this.filters[i] = newFilter;\n      this.recalculateFreqz(i);\n    }\n  };\n\n  this.setMinimumFrequency = function(freq) {\n    this.minFreq = freq;\n    this.recalculateFilters();\n  };\n\n  this.setMaximumFrequency = function(freq) {\n    this.maxFreq = freq;\n    this.recalculateFilters();\n  };\n\n  this.setBandsPerOctave = function(bands) {\n    this.bandsPerOctave = bands;\n    this.recalculateFilters();\n  };\n\n  this.setBandGain = function(bandIndex, gain) {\n    if (bandIndex < 0 || bandIndex > (this.filters.length-1)) {\n      throw \"The band index of the graphical equalizer is out of bounds.\";\n    }\n\n    if (!gain) {\n      throw \"A gain must be passed.\";\n    }\n   \n    this.filters[bandIndex].setDbGain(gain);\n    this.recalculateFreqz(bandIndex);\n  };\n \n  this.recalculateFreqz = function(bandIndex) {\n    if (!this.calculateFreqzs) {\n      return;\n    }\n\n    if (bandIndex < 0 || bandIndex > (this.filters.length-1)) {\n      throw \"The band index of the graphical equalizer is out of bounds. \" + bandIndex + \" is out of [\" + 0 + \", \" + this.filters.length-1 + \"]\";\n    }\n       \n    if (!this.w) {\n      this.w = Float32Array(400);\n      for (var i=0; i<this.w.length; i++) {\n         this.w[i] = Math.PI/this.w.length * i;\n      }\n    }\n   \n    var b = [this.filters[bandIndex].b0, this.filters[bandIndex].b1, this.filters[bandIndex].b2];\n    var a = [this.filters[bandIndex].a0, this.filters[bandIndex].a1, this.filters[bandIndex].a2];\n\n    this.freqzs[bandIndex] = DSP.mag2db(DSP.freqz(b, a, this.w));\n  };\n\n  this.process = function(buffer) {\n    var output = buffer;\n\n    for (var i = 0; i < this.filters.length; i++) {\n      output = this.filters[i].process(output);\n    }\n\n    return output;\n  };\n\n  this.processStereo = function(buffer) {\n    var output = buffer;\n\n    for (var i = 0; i < this.filters.length; i++) {\n      output = this.filters[i].processStereo(output);\n    }\n\n    return output;\n  };\n}\n\n/**\n * MultiDelay effect by Almer Thie (http://code.almeros.com).\n * Copyright 2010 Almer Thie. All rights reserved.\n * Example: http://code.almeros.com/code-examples/delay-firefox-audio-api/\n *\n * This is a delay that feeds it's own delayed signal back into its circular\n * buffer. Also known as a CombFilter.\n *\n * Compatible with interleaved stereo (or more channel) buffers and\n * non-interleaved mono buffers.\n *\n * @param {Number} maxDelayInSamplesSize Maximum possible delay in samples (size of circular buffer)\n * @param {Number} delayInSamples Initial delay in samples\n * @param {Number} masterVolume Initial master volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)\n * @param {Number} delayVolume Initial feedback delay volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)\n *\n * @constructor\n */\nfunction MultiDelay(maxDelayInSamplesSize, delayInSamples, masterVolume, delayVolume) {\n  this.delayBufferSamples   = new Float32Array(maxDelayInSamplesSize); // The maximum size of delay\n  this.delayInputPointer     = delayInSamples;\n  this.delayOutputPointer   = 0;\n \n  this.delayInSamples   = delayInSamples;\n  this.masterVolume     = masterVolume;\n  this.delayVolume     = delayVolume;\n}\n\n/**\n * Change the delay time in samples.\n *\n * @param {Number} delayInSamples Delay in samples\n */\nMultiDelay.prototype.setDelayInSamples = function (delayInSamples) {\n  this.delayInSamples = delayInSamples;\n \n  this.delayInputPointer = this.delayOutputPointer + delayInSamples;\n\n  if (this.delayInputPointer >= this.delayBufferSamples.length-1) {\n    this.delayInputPointer = this.delayInputPointer - this.delayBufferSamples.length; \n  }\n};\n\n/**\n * Change the master volume.\n *\n * @param {Number} masterVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)\n */\nMultiDelay.prototype.setMasterVolume = function(masterVolume) {\n  this.masterVolume = masterVolume;\n};\n\n/**\n * Change the delay feedback volume.\n *\n * @param {Number} delayVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)\n */\nMultiDelay.prototype.setDelayVolume = function(delayVolume) {\n  this.delayVolume = delayVolume;\n};\n\n/**\n * Process a given interleaved or mono non-interleaved float value Array and adds the delayed audio.\n *\n * @param {Array} samples Array containing Float values or a Float32Array\n *\n * @returns A new Float32Array interleaved or mono non-interleaved as was fed to this function.\n */\nMultiDelay.prototype.process = function(samples) {\n  // NB. Make a copy to put in the output samples to return.\n  var outputSamples = new Float32Array(samples.length);\n\n  for (var i=0; i<samples.length; i++) {\n    // delayBufferSamples could contain initial NULL's, return silence in that case\n    var delaySample = (this.delayBufferSamples[this.delayOutputPointer] === null ? 0.0 : this.delayBufferSamples[this.delayOutputPointer]);\n   \n    // Mix normal audio data with delayed audio\n    var sample = (delaySample * this.delayVolume) + samples[i];\n   \n    // Add audio data with the delay in the delay buffer\n    this.delayBufferSamples[this.delayInputPointer] = sample;\n   \n    // Return the audio with delay mix\n    outputSamples[i] = sample * this.masterVolume;\n   \n    // Manage circulair delay buffer pointers\n    this.delayInputPointer++;\n    if (this.delayInputPointer >= this.delayBufferSamples.length-1) {\n      this.delayInputPointer = 0;\n    }\n     \n    this.delayOutputPointer++;\n    if (this.delayOutputPointer >= this.delayBufferSamples.length-1) {\n      this.delayOutputPointer = 0; \n    } \n  }\n \n  return outputSamples;\n};\n\n/**\n * SingleDelay effect by Almer Thie (http://code.almeros.com).\n * Copyright 2010 Almer Thie. All rights reserved.\n * Example: See usage in Reverb class\n *\n * This is a delay that does NOT feeds it's own delayed signal back into its \n * circular buffer, neither does it return the original signal. Also known as\n * an AllPassFilter(?).\n *\n * Compatible with interleaved stereo (or more channel) buffers and\n * non-interleaved mono buffers.\n *\n * @param {Number} maxDelayInSamplesSize Maximum possible delay in samples (size of circular buffer)\n * @param {Number} delayInSamples Initial delay in samples\n * @param {Number} delayVolume Initial feedback delay volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)\n *\n * @constructor\n */\n\nfunction SingleDelay(maxDelayInSamplesSize, delayInSamples, delayVolume) {\n  this.delayBufferSamples = new Float32Array(maxDelayInSamplesSize); // The maximum size of delay\n  this.delayInputPointer  = delayInSamples;\n  this.delayOutputPointer = 0;\n \n  this.delayInSamples     = delayInSamples;\n  this.delayVolume        = delayVolume;\n}\n\n/**\n * Change the delay time in samples.\n *\n * @param {Number} delayInSamples Delay in samples\n */\nSingleDelay.prototype.setDelayInSamples = function(delayInSamples) {\n  this.delayInSamples = delayInSamples;\n  this.delayInputPointer = this.delayOutputPointer + delayInSamples;\n\n  if (this.delayInputPointer >= this.delayBufferSamples.length-1) {\n    this.delayInputPointer = this.delayInputPointer - this.delayBufferSamples.length; \n  }\n};\n\n/**\n * Change the return signal volume.\n *\n * @param {Number} delayVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)\n */\nSingleDelay.prototype.setDelayVolume = function(delayVolume) {\n  this.delayVolume = delayVolume;\n};\n\n/**\n * Process a given interleaved or mono non-interleaved float value Array and\n * returns the delayed audio.\n *\n * @param {Array} samples Array containing Float values or a Float32Array\n *\n * @returns A new Float32Array interleaved or mono non-interleaved as was fed to this function.\n */\nSingleDelay.prototype.process = function(samples) {\n  // NB. Make a copy to put in the output samples to return.\n  var outputSamples = new Float32Array(samples.length);\n\n  for (var i=0; i<samples.length; i++) {\n\n    // Add audio data with the delay in the delay buffer\n    this.delayBufferSamples[this.delayInputPointer] = samples[i];\n   \n    // delayBufferSamples could contain initial NULL's, return silence in that case\n    var delaySample = this.delayBufferSamples[this.delayOutputPointer];\n\n    // Return the audio with delay mix\n    outputSamples[i] = delaySample * this.delayVolume;\n\n    // Manage circulair delay buffer pointers\n    this.delayInputPointer++;\n\n    if (this.delayInputPointer >= this.delayBufferSamples.length-1) {\n      this.delayInputPointer = 0;\n    }\n     \n    this.delayOutputPointer++;\n\n    if (this.delayOutputPointer >= this.delayBufferSamples.length-1) {\n      this.delayOutputPointer = 0; \n    } \n  }\n \n  return outputSamples;\n};\n\n/**\n * Reverb effect by Almer Thie (http://code.almeros.com).\n * Copyright 2010 Almer Thie. All rights reserved.\n * Example: http://code.almeros.com/code-examples/reverb-firefox-audio-api/\n *\n * This reverb consists of 6 SingleDelays, 6 MultiDelays and an IIRFilter2\n * for each of the two stereo channels.\n *\n * Compatible with interleaved stereo buffers only!\n *\n * @param {Number} maxDelayInSamplesSize Maximum possible delay in samples (size of circular buffers)\n * @param {Number} delayInSamples Initial delay in samples for internal (Single/Multi)delays\n * @param {Number} masterVolume Initial master volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)\n * @param {Number} mixVolume Initial reverb signal mix volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)\n * @param {Number} delayVolume Initial feedback delay volume for internal (Single/Multi)delays. 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