/*
 *  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.
 *
 *  https://github.com/corbanbrook/dsp.js/
 */

////////////////////////////////////////////////////////////////////////////////
//                                  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("Float32Array", "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 Float32Array(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 Float32Array(buffer.length/2);
    right = right || new Float32Array(buffer.length/2);
    mix   = mix   || new Float32Array(buffer.length/2);

    if (buffer.length/2 !== left.length) {
      left  = new Float32Array(buffer.length/2);
      right = new Float32Array(buffer.length/2);
      mix   = new Float32Array(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 Float32Array
 * @param {Array} sampleBuffer2 Array containing Float values or a Float32Array
 * @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 Float32Array interleaved buffer.
 */
DSP.mixSampleBuffers = function(sampleBuffer1, sampleBuffer2, negate, volumeCorrection){
  var outputSamples = new Float32Array(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
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
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;
};

// 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.spectrum   = new Float32Array(_bufferSize/2);
  this.real       = new Float32Array(_bufferSize);
  this.imag       = new Float32Array(_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;
  };

  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;
    }
  };
}

/**
 * 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
 */
DFT = function(_bufferSize, _sampleRate) {
  FourierTransform.call(this, _bufferSize, _sampleRate);

  var N = _bufferSize/2 * _bufferSize;
  var TWO_PI = 2 * Math.PI;

  this.sinTable = new Float32Array(N);
  this.cosTable = new Float32Array(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();
};


/**
 * 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 Float32Array(_bufferSize);
  this.cosTable = new Float32Array(_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,
      spectrum        = this.spectrum;

  var k = Math.floor(Math.log(_bufferSize) / Math.LN2);

  if (Math.pow(2, k) !== _bufferSize) { throw "Invalid buffer size, must be a power of 2."; }
  if (_bufferSize !== buffer.length)  { throw "Supplied buffer is not the same size as defined FFT. FFT Size: " + _bufferSize + " Buffer Size: " + buffer.length; }

  var halfSize = 1,
      phaseShiftStepReal,
      phaseShiftStepImag,
      currentPhaseShiftReal,
      currentPhaseShiftImag,
      off,
      tr,
      ti,
      tmpReal,
      i;

  for (i = 0; i < _bufferSize; i++) {
    real[i] = buffer[reverseTable[i]];
    imag[i] = 0;
  }

  while (halfSize < _bufferSize) {
    //phaseShiftStepReal = Math.cos(-Math.PI/halfSize);
    //phaseShiftStepImag = Math.sin(-Math.PI/halfSize);
    phaseShiftStepReal = cosTable[halfSize];
    phaseShiftStepImag = sinTable[halfSize];

    currentPhaseShiftReal = 1;
    currentPhaseShiftImag = 0;

    for (var fftStep = 0; fftStep < halfSize; fftStep++) {
      i = fftStep;

      while (i < _bufferSize) {
        off = i + halfSize;
        tr = (currentPhaseShiftReal * real[off]) - (currentPhaseShiftImag * imag[off]);
        ti = (currentPhaseShiftReal * imag[off]) + (currentPhaseShiftImag * real[off]);

        real[off] = real[i] - tr;
        imag[off] = imag[i] - ti;
        real[i] += tr;
        imag[i] += ti;

        i += halfSize << 1;
      }

      tmpReal = currentPhaseShiftReal;
      currentPhaseShiftReal = (tmpReal * phaseShiftStepReal) - (currentPhaseShiftImag * phaseShiftStepImag);
      currentPhaseShiftImag = (tmpReal * phaseShiftStepImag) + (currentPhaseShiftImag * phaseShiftStepReal);
    }

    halfSize = halfSize << 1;
  }

  return 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;

  var halfSize = 1,
      phaseShiftStepReal,
      phaseShiftStepImag,
      currentPhaseShiftReal,
      currentPhaseShiftImag,
      off,
      tr,
      ti,
      tmpReal,
      i;

  for (i = 0; i < _bufferSize; i++) {
    imag[i] *= -1;
  }

  var revReal = new Float32Array(_bufferSize);
  var revImag = new Float32Array(_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 Float32Array(_bufferSize); // this should be reused instead
  for (i = 0; i < _bufferSize; i++) {
    buffer[i] = real[i] / _bufferSize;
  }

  return buffer;
};

/**
 * RFFT is a class for calculating the Discrete Fourier Transform of a signal
 * with the Fast Fourier Transform algorithm.
 *
 * This method currently only contains a forward transform but is highly optimized.
 *
 * @param {Number} _bufferSize The size of the sample buffer to be computed. Must be power of 2
 * @param {Number} _sampleRate The _sampleRate of the buffer (eg. 44100)
 *
 * @constructor
 */

// lookup tables don't really gain us any speed, but they do increase
// cache footprint, so don't use them in here

// also we don't use sepearate arrays for real/imaginary parts

// this one a little more than twice as fast as the one in FFT
// however I only did the forward transform

// the rest of this was translated from C, see http://www.jjj.de/fxt/
// this is the real split radix FFT

function RFFT(_bufferSize, _sampleRate) {
  FourierTransform.call(this, _bufferSize, _sampleRate);

  this.trans = new Float32Array(_bufferSize);

  this.reverseTable = new Uint32Array(_bufferSize);

  // don't use a lookup table to do the permute, use this instead
  this.reverseBinPermute = function (dest, source) {
    var _bufferSize  = this._bufferSize,
        halfSize    = _bufferSize >>> 1,
        nm1         = _bufferSize - 1,
        i = 1, r = 0, h;

    dest[0] = source[0];

    do {
      r += halfSize;
      dest[i] = source[r];
      dest[r] = source[i];

      i++;

      h = halfSize << 1;
      while (h = h >> 1, !((r ^= h) & h));

      if (r >= i) {
        dest[i]     = source[r];
        dest[r]     = source[i];

        dest[nm1-i] = source[nm1-r];
        dest[nm1-r] = source[nm1-i];
      }
      i++;
    } while (i < halfSize);
    dest[nm1] = source[nm1];
  };

  this.generateReverseTable = function () {
    var _bufferSize  = this._bufferSize,
        halfSize    = _bufferSize >>> 1,
        nm1         = _bufferSize - 1,
        i = 1, r = 0, h;

    this.reverseTable[0] = 0;

    do {
      r += halfSize;

      this.reverseTable[i] = r;
      this.reverseTable[r] = i;

      i++;

      h = halfSize << 1;
      while (h = h >> 1, !((r ^= h) & h));

      if (r >= i) {
        this.reverseTable[i] = r;
        this.reverseTable[r] = i;

        this.reverseTable[nm1-i] = nm1-r;
        this.reverseTable[nm1-r] = nm1-i;
      }
      i++;
    } while (i < halfSize);

    this.reverseTable[nm1] = nm1;
  };

  this.generateReverseTable();
}


// Ordering of output:
//
// trans[0]     = re[0] (==zero _frequency, purely real)
// trans[1]     = re[1]
//             ...
// trans[n/2-1] = re[n/2-1]
// trans[n/2]   = re[n/2]    (==nyquist _frequency, purely real)
//
// trans[n/2+1] = im[n/2-1]
// trans[n/2+2] = im[n/2-2]
//             ...
// trans[n-1]   = im[1]

RFFT.prototype.forward = function(buffer) {
  var n         = this._bufferSize,
      spectrum  = this.spectrum,
      x         = this.trans,
      TWO_PI    = 2*Math.PI,
      sqrt      = Math.sqrt,
      i         = n >>> 1,
      bSi       = 2 / n,
      n2, n4, n8, nn,
      t1, t2, t3, t4,
      i1, i2, i3, i4, i5, i6, i7, i8,
      st1, cc1, ss1, cc3, ss3,
      e,
      a,
      rval, ival, mag;

  this.reverseBinPermute(x, buffer);

  /*
  var reverseTable = this.reverseTable;

  for (var k = 0, len = reverseTable.length; k < len; k++) {
    x[k] = buffer[reverseTable[k]];
  }
  */

  for (var ix = 0, id = 4; ix < n; id *= 4) {
    for (var i0 = ix; i0 < n; i0 += id) {
      //sumdiff(x[i0], x[i0+1]); // {a, b}  <--| {a+b, a-b}
      st1 = x[i0] - x[i0+1];
      x[i0] += x[i0+1];
      x[i0+1] = st1;
    }
    ix = 2*(id-1);
  }

  n2 = 2;
  nn = n >>> 1;

  while((nn = nn >>> 1)) {
    ix = 0;
    n2 = n2 << 1;
    id = n2 << 1;
    n4 = n2 >>> 2;
    n8 = n2 >>> 3;
    do {
      if (n4 !== 1) {
        for(i0 = ix; i0 < n; i0 += id) {
          i1 = i0;
          i2 = i1 + n4;
          i3 = i2 + n4;
          i4 = i3 + n4;

          //diffsum3_r(x[i3], x[i4], t1); // {a, b, s} <--| {a, b-a, a+b}
          t1 = x[i3] + x[i4];
          x[i4] -= x[i3];
          //sumdiff3(x[i1], t1, x[i3]);   // {a, b, d} <--| {a+b, b, a-b}
          x[i3] = x[i1] - t1;
          x[i1] += t1;

          i1 += n8;
          i2 += n8;
          i3 += n8;
          i4 += n8;

          //sumdiff(x[i3], x[i4], t1, t2); // {s, d}  <--| {a+b, a-b}
          t1 = x[i3] + x[i4];
          t2 = x[i3] - x[i4];

          t1 = -t1 * Math.SQRT1_2;
          t2 *= Math.SQRT1_2;

          // sumdiff(t1, x[i2], x[i4], x[i3]); // {s, d}  <--| {a+b, a-b}
          st1 = x[i2];
          x[i4] = t1 + st1;
          x[i3] = t1 - st1;

          //sumdiff3(x[i1], t2, x[i2]); // {a, b, d} <--| {a+b, b, a-b}
          x[i2] = x[i1] - t2;
          x[i1] += t2;
        }
      } else {
        for(i0 = ix; i0 < n; i0 += id) {
          i1 = i0;
          i2 = i1 + n4;
          i3 = i2 + n4;
          i4 = i3 + n4;

          //diffsum3_r(x[i3], x[i4], t1); // {a, b, s} <--| {a, b-a, a+b}
          t1 = x[i3] + x[i4];
          x[i4] -= x[i3];

          //sumdiff3(x[i1], t1, x[i3]);   // {a, b, d} <--| {a+b, b, a-b}
          x[i3] = x[i1] - t1;
          x[i1] += t1;
        }
      }

      ix = (id << 1) - n2;
      id = id << 2;
    } while (ix < n);

    e = TWO_PI / n2;

    for (var j = 1; j < n8; j++) {
      a = j * e;
      ss1 = Math.sin(a);
      cc1 = Math.cos(a);

      //ss3 = sin(3*a); cc3 = cos(3*a);
      cc3 = 4*cc1*(cc1*cc1-0.75);
      ss3 = 4*ss1*(0.75-ss1*ss1);

      ix = 0; id = n2 << 1;
      do {
        for (i0 = ix; i0 < n; i0 += id) {
          i1 = i0 + j;
          i2 = i1 + n4;
          i3 = i2 + n4;
          i4 = i3 + n4;

          i5 = i0 + n4 - j;
          i6 = i5 + n4;
          i7 = i6 + n4;
          i8 = i7 + n4;

          //cmult(c, s, x, y, &u, &v)
          //cmult(cc1, ss1, x[i7], x[i3], t2, t1); // {u,v} <--| {x*c-y*s, x*s+y*c}
          t2 = x[i7]*cc1 - x[i3]*ss1;
          t1 = x[i7]*ss1 + x[i3]*cc1;

          //cmult(cc3, ss3, x[i8], x[i4], t4, t3);
          t4 = x[i8]*cc3 - x[i4]*ss3;
          t3 = x[i8]*ss3 + x[i4]*cc3;

          //sumdiff(t2, t4);   // {a, b} <--| {a+b, a-b}
          st1 = t2 - t4;
          t2 += t4;
          t4 = st1;

          //sumdiff(t2, x[i6], x[i8], x[i3]); // {s, d}  <--| {a+b, a-b}
          //st1 = x[i6]; x[i8] = t2 + st1; x[i3] = t2 - st1;
          x[i8] = t2 + x[i6];
          x[i3] = t2 - x[i6];

          //sumdiff_r(t1, t3); // {a, b} <--| {a+b, b-a}
          st1 = t3 - t1;
          t1 += t3;
          t3 = st1;

          //sumdiff(t3, x[i2], x[i4], x[i7]); // {s, d}  <--| {a+b, a-b}
          //st1 = x[i2]; x[i4] = t3 + st1; x[i7] = t3 - st1;
          x[i4] = t3 + x[i2];
          x[i7] = t3 - x[i2];

          //sumdiff3(x[i1], t1, x[i6]);   // {a, b, d} <--| {a+b, b, a-b}
          x[i6] = x[i1] - t1;
          x[i1] += t1;

          //diffsum3_r(t4, x[i5], x[i2]); // {a, b, s} <--| {a, b-a, a+b}
          x[i2] = t4 + x[i5];
          x[i5] -= t4;
        }

        ix = (id << 1) - n2;
        id = id << 2;

      } while (ix < n);
    }
  }

  while (--i) {
    rval = x[i];
    ival = x[n-i-1];
    mag = bSi * sqrt(rval * rval + ival * ival);

    if (mag > this.peak) {
      this.peakBand = i;
      this.peak = mag;
    }

    spectrum[i] = mag;
  }

  spectrum[0] = bSi * x[0];

  return spectrum;
};



function WindowFunction(type, alpha) {
  this.alpha = alpha;

  switch(type) {
    case DSP.BARTLETT:
      this.func = WindowFunction.Bartlett;
      break;

    case DSP.BARTLETTHANN:
      this.func = WindowFunction.BartlettHann;
      break;

    case DSP.BLACKMAN:
      this.func = WindowFunction.Blackman;
      this.alpha = this.alpha || 0.16;
      break;

    case DSP.COSINE:
      this.func = WindowFunction.Cosine;
      break;

    case DSP.GAUSS:
      this.func = WindowFunction.Gauss;
      this.alpha = this.alpha || 0.25;
      break;

    case DSP.HAMMING:
      this.func = WindowFunction.Hamming;
      break;

    case DSP.HANN:
      this.func = WindowFunction.Hann;
      break;

    case DSP.LANCZOS:
      this.func = WindowFunction.Lanczoz;
      break;

    case DSP.RECTANGULAR:
      this.func = WindowFunction.Rectangular;
      break;

    case DSP.TRIANGULAR:
      this.func = WindowFunction.Triangular;
      break;
  }
}

WindowFunction.prototype.process = function(buffer) {
  var length = buffer.length;
  for ( var i = 0; i < length; i++ ) {
    buffer[i] *= this.func(length, i, this.alpha);
  }
  return buffer;
};

WindowFunction.Bartlett = function(length, index) {
  return 2 / (length - 1) * ((length - 1) / 2 - Math.abs(index - (length - 1) / 2));
};

WindowFunction.BartlettHann = function(length, index) {
  return 0.62 - 0.48 * Math.abs(index / (length - 1) - 0.5) - 0.38 * Math.cos(DSP.TWO_PI * index / (length - 1));
};

WindowFunction.Blackman = function(length, index, alpha) {
  var a0 = (1 - alpha) / 2;
  var a1 = 0.5;
  var a2 = alpha / 2;

  return a0 - a1 * Math.cos(DSP.TWO_PI * index / (length - 1)) + a2 * Math.cos(4 * Math.PI * index / (length - 1));
};

WindowFunction.Cosine = function(length, index) {
  return Math.cos(Math.PI * index / (length - 1) - Math.PI / 2);
};

WindowFunction.Gauss = function(length, index, alpha) {
  return Math.pow(Math.E, -0.5 * Math.pow((index - (length - 1) / 2) / (alpha * (length - 1) / 2), 2));
};

WindowFunction.Hamming = function(length, index) {
  return 0.54 - 0.46 * Math.cos(DSP.TWO_PI * index / (length - 1));
};

WindowFunction.Hann = function(length, index) {
  return 0.5 * (1 - Math.cos(DSP.TWO_PI * index / (length - 1)));
};

WindowFunction.Lanczos = function(length, index) {
  var x = 2 * index / (length - 1) - 1;
  return Math.sin(Math.PI * x) / (Math.PI * x);
};

WindowFunction.Rectangular = function(length, index) {
  return 1;
};

WindowFunction.Triangular = function(length, index) {
  return 2 / length * (length / 2 - Math.abs(index - (length - 1) / 2));
};

function sinh (arg) {
  // Returns the hyperbolic sine of the number, defined as (exp(number) - exp(-number))/2
  //
  // version: 1004.2314
  // discuss at: http://phpjs.org/functions/sinh    // +   original by: Onno Marsman
  // *     example 1: sinh(-0.9834330348825909);
  // *     returns 1: -1.1497971402636502
  return (Math.exp(arg) - Math.exp(-arg))/2;
}

/*
 *  Biquad filter
 *
 *  Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
 *  Copyright 2010 Ricard Marxer. All rights reserved.
 *
 */
// Implementation based on:
// http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
function Biquad(type, _sampleRate) {
  this.Fs = _sampleRate;
  this.type = type;  // type of the filter
  this.parameterType = DSP.Q; // type of the parameter

  this.x_1_l = 0;
  this.x_2_l = 0;
  this.y_1_l = 0;
  this.y_2_l = 0;

  this.x_1_r = 0;
  this.x_2_r = 0;
  this.y_1_r = 0;
  this.y_2_r = 0;

  this.b0 = 1;
  this.a0 = 1;

  this.b1 = 0;
  this.a1 = 0;

  this.b2 = 0;
  this.a2 = 0;

  this.b0a0 = this.b0 / this.a0;
  this.b1a0 = this.b1 / this.a0;
  this.b2a0 = this.b2 / this.a0;
  this.a1a0 = this.a1 / this.a0;
  this.a2a0 = this.a2 / this.a0;

  this.f0 = 3000;   // "wherever it's happenin', man."  Center Frequency or
                    // Corner Frequency, or shelf midpoint _frequency, depending
                    // on which filter type.  The "significant _frequency".

  this.dBgain = 12; // used only for peaking and shelving filters

  this.Q = 1;       // the EE kind of definition, except for peakingEQ in which A*Q is
                    // the classic EE Q.  That adjustment in definition was made so that
                    // a boost of N dB followed by a cut of N dB for identical Q and
                    // f0/Fs results in a precisely flat unity gain filter or "wire".

  this.BW = -3;     // the bandwidth in octaves (between -3 dB frequencies for BPF
                    // and notch or between midpoint (dBgain/2) gain frequencies for
                    // peaking EQ

  this.S = 1;       // a "shelf slope" parameter (for shelving EQ only).  When S = 1,
                    // the shelf slope is as steep as it can be and remain monotonically
                    // increasing or decreasing gain with _frequency.  The shelf slope, in
                    // dB/octave, remains proportional to S for all other values for a
                    // fixed f0/Fs and dBgain.

  this.coefficients = function() {
    var b = [this.b0, this.b1, this.b2];
    var a = [this.a0, this.a1, this.a2];
    return {b: b, a:a};
  };

  this.setFilterType = function(type) {
    this.type = type;
    this.recalculateCoefficients();
  };

  this.setSampleRate = function(rate) {
    this.Fs = rate;
    this.recalculateCoefficients();
  };

  this.setQ = function(q) {
    this.parameterType = DSP.Q;
    this.Q = Math.max(Math.min(q, 115.0), 0.001);
    this.recalculateCoefficients();
  };

  this.setBW = function(bw) {
    this.parameterType = DSP.BW;
    this.BW = bw;
    this.recalculateCoefficients();
  };

  this.setS = function(s) {
    this.parameterType = DSP.S;
    this.S = Math.max(Math.min(s, 5.0), 0.0001);
    this.recalculateCoefficients();
  };

  this.setF0 = function(freq) {
    this.f0 = freq;
    this.recalculateCoefficients();
  };

  this.setDbGain = function(g) {
    this.dBgain = g;
    this.recalculateCoefficients();
  };

  this.recalculateCoefficients = function() {
    var A;
    if (type === DSP.PEAKING_EQ || type === DSP.LOW_SHELF || type === DSP.HIGH_SHELF ) {
      A = Math.pow(10, (this.dBgain/40));  // for peaking and shelving EQ filters only
    } else {
      A  = Math.sqrt( Math.pow(10, (this.dBgain/20)) );
    }

    var w0 = DSP.TWO_PI * this.f0 / this.Fs;

    var cosw0 = Math.cos(w0);
    var sinw0 = Math.sin(w0);

    var alpha = 0;

    switch (this.parameterType) {
      case DSP.Q:
        alpha = sinw0/(2*this.Q);
        break;

      case DSP.BW:
        alpha = sinw0 * sinh( Math.LN2/2 * this.BW * w0/sinw0 );
        break;

      case DSP.S:
        alpha = sinw0/2 * Math.sqrt( (A + 1/A)*(1/this.S - 1) + 2 );
        break;
    }

    /**
        FYI: The relationship between bandwidth and Q is
             1/Q = 2*sinh(ln(2)/2*BW*w0/sin(w0))     (digital filter w BLT)
        or   1/Q = 2*sinh(ln(2)/2*BW)             (analog filter prototype)

        The relationship between shelf slope and Q is
             1/Q = sqrt((A + 1/A)*(1/S - 1) + 2)
    */

    var coeff;

    switch (this.type) {
      case DSP.LPF:       // H(s) = 1 / (s^2 + s/Q + 1)
        this.b0 =  (1 - cosw0)/2;
        this.b1 =   1 - cosw0;
        this.b2 =  (1 - cosw0)/2;
        this.a0 =   1 + alpha;
        this.a1 =  -2 * cosw0;
        this.a2 =   1 - alpha;
        break;

      case DSP.HPF:       // H(s) = s^2 / (s^2 + s/Q + 1)
        this.b0 =  (1 + cosw0)/2;
        this.b1 = -(1 + cosw0);
        this.b2 =  (1 + cosw0)/2;
        this.a0 =   1 + alpha;
        this.a1 =  -2 * cosw0;
        this.a2 =   1 - alpha;
        break;

      case DSP.BPF_CONSTANT_SKIRT:       // H(s) = s / (s^2 + s/Q + 1)  (constant skirt gain, peak gain = Q)
        this.b0 =   sinw0/2;
        this.b1 =   0;
        this.b2 =  -sinw0/2;
        this.a0 =   1 + alpha;
        this.a1 =  -2*cosw0;
        this.a2 =   1 - alpha;
        break;

      case DSP.BPF_CONSTANT_PEAK:       // H(s) = (s/Q) / (s^2 + s/Q + 1)      (constant 0 dB peak gain)
        this.b0 =   alpha;
        this.b1 =   0;
        this.b2 =  -alpha;
        this.a0 =   1 + alpha;
        this.a1 =  -2*cosw0;
        this.a2 =   1 - alpha;
        break;

      case DSP.NOTCH:     // H(s) = (s^2 + 1) / (s^2 + s/Q + 1)
        this.b0 =   1;
        this.b1 =  -2*cosw0;
        this.b2 =   1;
        this.a0 =   1 + alpha;
        this.a1 =  -2*cosw0;
        this.a2 =   1 - alpha;
        break;

      case DSP.APF:       // H(s) = (s^2 - s/Q + 1) / (s^2 + s/Q + 1)
        this.b0 =   1 - alpha;
        this.b1 =  -2*cosw0;
        this.b2 =   1 + alpha;
        this.a0 =   1 + alpha;
        this.a1 =  -2*cosw0;
        this.a2 =   1 - alpha;
        break;

      case DSP.PEAKING_EQ:  // H(s) = (s^2 + s*(A/Q) + 1) / (s^2 + s/(A*Q) + 1)
        this.b0 =   1 + alpha*A;
        this.b1 =  -2*cosw0;
        this.b2 =   1 - alpha*A;
        this.a0 =   1 + alpha/A;
        this.a1 =  -2*cosw0;
        this.a2 =   1 - alpha/A;
        break;

      case DSP.LOW_SHELF:   // H(s) = A * (s^2 + (sqrt(A)/Q)*s + A)/(A*s^2 + (sqrt(A)/Q)*s + 1)
        coeff = sinw0 * Math.sqrt( (A^2 + 1)*(1/this.S - 1) + 2*A );
        this.b0 =    A*((A+1) - (A-1)*cosw0 + coeff);
        this.b1 =  2*A*((A-1) - (A+1)*cosw0);
        this.b2 =    A*((A+1) - (A-1)*cosw0 - coeff);
        this.a0 =       (A+1) + (A-1)*cosw0 + coeff;
        this.a1 =   -2*((A-1) + (A+1)*cosw0);
        this.a2 =       (A+1) + (A-1)*cosw0 - coeff;
        break;

      case DSP.HIGH_SHELF:   // H(s) = A * (A*s^2 + (sqrt(A)/Q)*s + 1)/(s^2 + (sqrt(A)/Q)*s + A)
        coeff = sinw0 * Math.sqrt( (A^2 + 1)*(1/this.S - 1) + 2*A );
        this.b0 =    A*((A+1) + (A-1)*cosw0 + coeff);
        this.b1 = -2*A*((A-1) + (A+1)*cosw0);
        this.b2 =    A*((A+1) + (A-1)*cosw0 - coeff);
        this.a0 =       (A+1) - (A-1)*cosw0 + coeff;
        this.a1 =    2*((A-1) - (A+1)*cosw0);
        this.a2 =       (A+1) - (A-1)*cosw0 - coeff;
        break;
    }

    this.b0a0 = this.b0/this.a0;
    this.b1a0 = this.b1/this.a0;
    this.b2a0 = this.b2/this.a0;
    this.a1a0 = this.a1/this.a0;
    this.a2a0 = this.a2/this.a0;
  };

  this.process = function(buffer) {
      //y[n] = (b0/a0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2]
      //       - (a1/a0)*y[n-1] - (a2/a0)*y[n-2]

      var len = buffer.length;
      var output = new Float32Array(len);

      for ( var i=0; i<buffer.length; i++ ) {
        output[i] = this.b0a0*buffer[i] + this.b1a0*this.x_1_l + this.b2a0*this.x_2_l - this.a1a0*this.y_1_l - this.a2a0*this.y_2_l;
        this.y_2_l = this.y_1_l;
        this.y_1_l = output[i];
        this.x_2_l = this.x_1_l;
        this.x_1_l = buffer[i];
      }

      return output;
  };

  this.processStereo = function(buffer) {
      //y[n] = (b0/a0)*x[n] + (b1/a0)*x[n-1] + (b2/a0)*x[n-2]
      //       - (a1/a0)*y[n-1] - (a2/a0)*y[n-2]

      var len = buffer.length;
      var output = new Float32Array(len);

      for (var i = 0; i < len/2; i++) {
        output[2*i] = this.b0a0*buffer[2*i] + this.b1a0*this.x_1_l + this.b2a0*this.x_2_l - this.a1a0*this.y_1_l - this.a2a0*this.y_2_l;
        this.y_2_l = this.y_1_l;
        this.y_1_l = output[2*i];
        this.x_2_l = this.x_1_l;
        this.x_1_l = buffer[2*i];

        output[2*i+1] = this.b0a0*buffer[2*i+1] + this.b1a0*this.x_1_r + this.b2a0*this.x_2_r - this.a1a0*this.y_1_r - this.a2a0*this.y_2_r;
        this.y_2_r = this.y_1_r;
        this.y_1_r = output[2*i+1];
        this.x_2_r = this.x_1_r;
        this.x_1_r = buffer[2*i+1];
      }

      return output;
  };
}

/*
 *  Magnitude to decibels
 *
 *  Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
 *  Copyright 2010 Ricard Marxer. All rights reserved.
 *
 *  @buffer array of magnitudes to convert to decibels
 *
 *  @returns the array in decibels
 *
 */
DSP.mag2db = function(buffer) {
  var minDb = -120;
  var minMag = Math.pow(10.0, minDb / 20.0);

  var log = Math.log;
  var max = Math.max;

  var result = Float32Array(buffer.length);
  for (var i=0; i<buffer.length; i++) {
    result[i] = 20.0*log(max(buffer[i], minMag));
  }

  return result;
};

/*
 *  Frequency response
 *
 *  Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
 *  Copyright 2010 Ricard Marxer. All rights reserved.
 *
 *  Calculates the _frequency response at the given points.
 *
 *  @b b coefficients of the filter
 *  @a a coefficients of the filter
 *  @w w points (normally between -PI and PI) where to calculate the _frequency response
 *
 *  @returns the _frequency response in magnitude
 *
 */
DSP.freqz = function(b, a, w) {
  var i, j;

  if (!w) {
    w = Float32Array(200);
    for (i=0;i<w.length; i++) {
      w[i] = DSP.TWO_PI/w.length * i - Math.PI;
    }
  }

  var result = Float32Array(w.length);

  var sqrt = Math.sqrt;
  var cos = Math.cos;
  var sin = Math.sin;

  for (i=0; i<w.length; i++) {
    var numerator = {real:0.0, imag:0.0};
    for (j=0; j<b.length; j++) {
      numerator.real += b[j] * cos(-j*w[i]);
      numerator.imag += b[j] * sin(-j*w[i]);
    }

    var denominator = {real:0.0, imag:0.0};
    for (j=0; j<a.length; j++) {
      denominator.real += a[j] * cos(-j*w[i]);
      denominator.imag += a[j] * sin(-j*w[i]);
    }

    result[i] =  sqrt(numerator.real*numerator.real + numerator.imag*numerator.imag) / sqrt(denominator.real*denominator.real + denominator.imag*denominator.imag);
  }

  return result;
};

/*
 *  Graphical Equalizer
 *
 *  Implementation of a graphic equalizer with a configurable bands-per-octave
 *  and minimum and maximum frequencies
 *
 *  Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
 *  Copyright 2010 Ricard Marxer. All rights reserved.
 *
 */
function GraphicalEq(_sampleRate) {
  this.FS = _sampleRate;
  this.minFreq = 40.0;
  this.maxFreq = 16000.0;

  this.bandsPerOctave = 1.0;

  this.filters = [];
  this.freqzs = [];

  this.calculateFreqzs = true;

  this.recalculateFilters = function() {
    var bandCount = Math.round(Math.log(this.maxFreq/this.minFreq) * this.bandsPerOctave/ Math.LN2);

    this.filters = [];
    for (var i=0; i<bandCount; i++) {
      var freq = this.minFreq*(Math.pow(2, i/this.bandsPerOctave));
      var newFilter = new Biquad(DSP.PEAKING_EQ, this.FS);
      newFilter.setDbGain(0);
      newFilter.setBW(1/this.bandsPerOctave);
      newFilter.setF0(freq);
      this.filters[i] = newFilter;
      this.recalculateFreqz(i);
    }
  };

  this.setMinimumFrequency = function(freq) {
    this.minFreq = freq;
    this.recalculateFilters();
  };

  this.setMaximumFrequency = function(freq) {
    this.maxFreq = freq;
    this.recalculateFilters();
  };

  this.setBandsPerOctave = function(bands) {
    this.bandsPerOctave = bands;
    this.recalculateFilters();
  };

  this.setBandGain = function(bandIndex, gain) {
    if (bandIndex < 0 || bandIndex > (this.filters.length-1)) {
      throw "The band index of the graphical equalizer is out of bounds.";
    }

    if (!gain) {
      throw "A gain must be passed.";
    }

    this.filters[bandIndex].setDbGain(gain);
    this.recalculateFreqz(bandIndex);
  };

  this.recalculateFreqz = function(bandIndex) {
    if (!this.calculateFreqzs) {
      return;
    }

    if (bandIndex < 0 || bandIndex > (this.filters.length-1)) {
      throw "The band index of the graphical equalizer is out of bounds. " + bandIndex + " is out of [" + 0 + ", " + this.filters.length-1 + "]";
    }

    if (!this.w) {
      this.w = Float32Array(400);
      for (var i=0; i<this.w.length; i++) {
         this.w[i] = Math.PI/this.w.length * i;
      }
    }

    var b = [this.filters[bandIndex].b0, this.filters[bandIndex].b1, this.filters[bandIndex].b2];
    var a = [this.filters[bandIndex].a0, this.filters[bandIndex].a1, this.filters[bandIndex].a2];

    this.freqzs[bandIndex] = DSP.mag2db(DSP.freqz(b, a, this.w));
  };

  this.process = function(buffer) {
    var output = buffer;

    for (var i = 0; i < this.filters.length; i++) {
      output = this.filters[i].process(output);
    }

    return output;
  };

  this.processStereo = function(buffer) {
    var output = buffer;

    for (var i = 0; i < this.filters.length; i++) {
      output = this.filters[i].processStereo(output);
    }

    return output;
  };
}

/**
 * MultiDelay effect by Almer Thie (http://code.almeros.com).
 * Copyright 2010 Almer Thie. All rights reserved.
 * Example: http://code.almeros.com/code-examples/delay-firefox-audio-api/
 *
 * This is a delay that feeds it's own delayed signal back into its circular
 * buffer. Also known as a CombFilter.
 *
 * Compatible with interleaved stereo (or more channel) buffers and
 * non-interleaved mono buffers.
 *
 * @param {Number} maxDelayInSamplesSize Maximum possible delay in samples (size of circular buffer)
 * @param {Number} delayInSamples Initial delay in samples
 * @param {Number} masterVolume Initial master volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 * @param {Number} delayVolume Initial feedback delay volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 *
 * @constructor
 */
function MultiDelay(maxDelayInSamplesSize, delayInSamples, masterVolume, delayVolume) {
  this.delayBufferSamples   = new Float32Array(maxDelayInSamplesSize); // The maximum size of delay
  this.delayRotationPointer     = delayInSamples;
  this.delayOutputPointer   = 0;

  this.delayInSamples   = delayInSamples;
  this.masterVolume     = masterVolume;
  this.delayVolume     = delayVolume;
}

/**
 * Change the delay time in samples.
 *
 * @param {Number} delayInSamples Delay in samples
 */
MultiDelay.prototype.setDelayInSamples = function (delayInSamples) {
  this.delayInSamples = delayInSamples;

  this.delayRotationPointer = this.delayOutputPointer + delayInSamples;

  if (this.delayRotationPointer >= this.delayBufferSamples.length-1) {
    this.delayRotationPointer = this.delayRotationPointer - this.delayBufferSamples.length;
  }
};

/**
 * Change the master volume.
 *
 * @param {Number} masterVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 */
MultiDelay.prototype.setMasterVolume = function(masterVolume) {
  this.masterVolume = masterVolume;
};

/**
 * Change the delay feedback volume.
 *
 * @param {Number} delayVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 */
MultiDelay.prototype.setDelayVolume = function(delayVolume) {
  this.delayVolume = delayVolume;
};

/**
 * Process a given interleaved or mono non-interleaved float value Array and adds the delayed audio.
 *
 * @param {Array} samples Array containing Float values or a Float32Array
 *
 * @returns A new Float32Array interleaved or mono non-interleaved as was fed to this function.
 */
MultiDelay.prototype.process = function(samples) {
  // NB. Make a copy to put in the output samples to return.
  var outputSamples = new Float32Array(samples.length);

  for (var i=0; i<samples.length; i++) {
    // delayBufferSamples could contain initial NULL's, return silence in that case
    var delaySample = (this.delayBufferSamples[this.delayOutputPointer] === null ? 0.0 : this.delayBufferSamples[this.delayOutputPointer]);

    // Mix normal audio data with delayed audio
    var sample = (delaySample * this.delayVolume) + samples[i];

    // Add audio data with the delay in the delay buffer
    this.delayBufferSamples[this.delayRotationPointer] = sample;

    // Return the audio with delay mix
    outputSamples[i] = sample * this.masterVolume;

    // Manage circulair delay buffer pointers
    this.delayRotationPointer++;
    if (this.delayRotationPointer >= this.delayBufferSamples.length-1) {
      this.delayRotationPointer = 0;
    }

    this.delayOutputPointer++;
    if (this.delayOutputPointer >= this.delayBufferSamples.length-1) {
      this.delayOutputPointer = 0;
    }
  }

  return outputSamples;
};

/**
 * SingleDelay effect by Almer Thie (http://code.almeros.com).
 * Copyright 2010 Almer Thie. All rights reserved.
 * Example: See usage in Reverb class
 *
 * This is a delay that does NOT feeds it's own delayed signal back into its
 * circular buffer, neither does it return the original signal. Also known as
 * an AllPassFilter(?).
 *
 * Compatible with interleaved stereo (or more channel) buffers and
 * non-interleaved mono buffers.
 *
 * @param {Number} maxDelayInSamplesSize Maximum possible delay in samples (size of circular buffer)
 * @param {Number} delayInSamples Initial delay in samples
 * @param {Number} delayVolume Initial feedback delay volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 *
 * @constructor
 */

function SingleDelay(maxDelayInSamplesSize, delayInSamples, delayVolume) {
  this.delayBufferSamples = new Float32Array(maxDelayInSamplesSize); // The maximum size of delay
  this.delayRotationPointer  = delayInSamples;
  this.delayOutputPointer = 0;

  this.delayInSamples     = delayInSamples;
  this.delayVolume        = delayVolume;
}

/**
 * Change the delay time in samples.
 *
 * @param {Number} delayInSamples Delay in samples
 */
SingleDelay.prototype.setDelayInSamples = function(delayInSamples) {
  this.delayInSamples = delayInSamples;
  this.delayRotationPointer = this.delayOutputPointer + delayInSamples;

  if (this.delayRotationPointer >= this.delayBufferSamples.length-1) {
    this.delayRotationPointer = this.delayRotationPointer - this.delayBufferSamples.length;
  }
};

/**
 * Change the return signal volume.
 *
 * @param {Number} delayVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 */
SingleDelay.prototype.setDelayVolume = function(delayVolume) {
  this.delayVolume = delayVolume;
};

/**
 * Process a given interleaved or mono non-interleaved float value Array and
 * returns the delayed audio.
 *
 * @param {Array} samples Array containing Float values or a Float32Array
 *
 * @returns A new Float32Array interleaved or mono non-interleaved as was fed to this function.
 */
SingleDelay.prototype.process = function(samples) {
  // NB. Make a copy to put in the output samples to return.
  var outputSamples = new Float32Array(samples.length);

  for (var i=0; i<samples.length; i++) {

    // Add audio data with the delay in the delay buffer
    this.delayBufferSamples[this.delayRotationPointer] = samples[i];

    // delayBufferSamples could contain initial NULL's, return silence in that case
    var delaySample = this.delayBufferSamples[this.delayOutputPointer];

    // Return the audio with delay mix
    outputSamples[i] = delaySample * this.delayVolume;

    // Manage circulair delay buffer pointers
    this.delayRotationPointer++;

    if (this.delayRotationPointer >= this.delayBufferSamples.length-1) {
      this.delayRotationPointer = 0;
    }

    this.delayOutputPointer++;

    if (this.delayOutputPointer >= this.delayBufferSamples.length-1) {
      this.delayOutputPointer = 0;
    }
  }

  return outputSamples;
};

/**
 * Reverb effect by Almer Thie (http://code.almeros.com).
 * Copyright 2010 Almer Thie. All rights reserved.
 * Example: http://code.almeros.com/code-examples/reverb-firefox-audio-api/
 *
 * This reverb consists of 6 SingleDelays, 6 MultiDelays and an IIRFilter2
 * for each of the two stereo channels.
 *
 * Compatible with interleaved stereo buffers only!
 *
 * @param {Number} maxDelayInSamplesSize Maximum possible delay in samples (size of circular buffers)
 * @param {Number} delayInSamples Initial delay in samples for internal (Single/Multi)delays
 * @param {Number} masterVolume Initial master volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 * @param {Number} mixVolume Initial reverb signal mix volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 * @param {Number} delayVolume Initial feedback delay volume for internal (Single/Multi)delays. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 * @param {Number} dampFrequency Initial low pass filter _frequency. 0 to 44100 (depending on your maximum sampling _frequency)
 *
 * @constructor
 */
function Reverb(maxDelayInSamplesSize, delayInSamples, masterVolume, mixVolume, delayVolume, dampFrequency) {
  this.delayInSamples   = delayInSamples;
  this.masterVolume     = masterVolume;
  this.mixVolume       = mixVolume;
  this.delayVolume     = delayVolume;
  this.dampFrequency     = dampFrequency;

  this.NR_OF_MULTIDELAYS = 6;
  this.NR_OF_SINGLEDELAYS = 6;

  this.LOWPASSL = new IIRFilter2(DSP.LOWPASS, dampFrequency, 0, 44100);
  this.LOWPASSR = new IIRFilter2(DSP.LOWPASS, dampFrequency, 0, 44100);

  this.singleDelays = [];

  var i, delayMultiply;

  for (i = 0; i < this.NR_OF_SINGLEDELAYS; i++) {
    delayMultiply = 1.0 + (i/7.0); // 1.0, 1.1, 1.2...
    this.singleDelays[i] = new SingleDelay(maxDelayInSamplesSize, Math.round(this.delayInSamples * delayMultiply), this.delayVolume);
  }

  this.multiDelays = [];

  for (i = 0; i < this.NR_OF_MULTIDELAYS; i++) {
    delayMultiply = 1.0 + (i/10.0); // 1.0, 1.1, 1.2...
    this.multiDelays[i] = new MultiDelay(maxDelayInSamplesSize, Math.round(this.delayInSamples * delayMultiply), this.masterVolume, this.delayVolume);
  }
}

/**
 * Change the delay time in samples as a base for all delays.
 *
 * @param {Number} delayInSamples Delay in samples
 */
Reverb.prototype.setDelayInSamples = function (delayInSamples){
  this.delayInSamples = delayInSamples;

  var i, delayMultiply;

  for (i = 0; i < this.NR_OF_SINGLEDELAYS; i++) {
    delayMultiply = 1.0 + (i/7.0); // 1.0, 1.1, 1.2...
    this.singleDelays[i].setDelayInSamples( Math.round(this.delayInSamples * delayMultiply) );
  }

  for (i = 0; i < this.NR_OF_MULTIDELAYS; i++) {
    delayMultiply = 1.0 + (i/10.0); // 1.0, 1.1, 1.2...
    this.multiDelays[i].setDelayInSamples( Math.round(this.delayInSamples * delayMultiply) );
  }
};

/**
 * Change the master volume.
 *
 * @param {Number} masterVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 */
Reverb.prototype.setMasterVolume = function (masterVolume){
  this.masterVolume = masterVolume;
};

/**
 * Change the reverb signal mix level.
 *
 * @param {Number} mixVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 */
Reverb.prototype.setMixVolume = function (mixVolume){
  this.mixVolume = mixVolume;
};

/**
 * Change all delays feedback volume.
 *
 * @param {Number} delayVolume Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 */
Reverb.prototype.setDelayVolume = function (delayVolume){
  this.delayVolume = delayVolume;

  var i;

  for (i = 0; i<this.NR_OF_SINGLEDELAYS; i++) {
    this.singleDelays[i].setDelayVolume(this.delayVolume);
  }

  for (i = 0; i<this.NR_OF_MULTIDELAYS; i++) {
    this.multiDelays[i].setDelayVolume(this.delayVolume);
  }
};

/**
 * Change the Low Pass filter _frequency.
 *
 * @param {Number} dampFrequency low pass filter _frequency. 0 to 44100 (depending on your maximum sampling _frequency)
 */
Reverb.prototype.setDampFrequency = function (dampFrequency){
  this.dampFrequency = dampFrequency;

  this.LOWPASSL.set(dampFrequency, 0);
  this.LOWPASSR.set(dampFrequency, 0);
};

/**
 * Process a given interleaved float value Array and copies and adds the reverb signal.
 *
 * @param {Array} samples Array containing Float values or a Float32Array
 *
 * @returns A new Float32Array interleaved buffer.
 */
Reverb.prototype.process = function (interleavedSamples){
  // NB. Make a copy to put in the output samples to return.
  var outputSamples = new Float32Array(interleavedSamples.length);

  // Perform low pass on the input samples to mimick damp
  var leftRightMix = DSP.deinterleave(interleavedSamples);
  this.LOWPASSL.process( leftRightMix[DSP.LEFT] );
  this.LOWPASSR.process( leftRightMix[DSP.RIGHT] );
  var filteredSamples = DSP.interleave(leftRightMix[DSP.LEFT], leftRightMix[DSP.RIGHT]);

  var i;

  // Process MultiDelays in parallel
  for (i = 0; i<this.NR_OF_MULTIDELAYS; i++) {
    // Invert the signal of every even multiDelay
    outputSamples = DSP.mixSampleBuffers(outputSamples, this.multiDelays[i].process(filteredSamples), 2%i === 0, this.NR_OF_MULTIDELAYS);
  }

  // Process SingleDelays in series
  var singleDelaySamples = new Float32Array(outputSamples.length);
  for (i = 0; i<this.NR_OF_SINGLEDELAYS; i++) {
    // Invert the signal of every even singleDelay
    singleDelaySamples = DSP.mixSampleBuffers(singleDelaySamples, this.singleDelays[i].process(outputSamples), 2%i === 0, 1);
  }

  // Apply the volume of the reverb signal
  for (i = 0; i<singleDelaySamples.length; i++) {
    singleDelaySamples[i] *= this.mixVolume;
  }

  // Mix the original signal with the reverb signal
  outputSamples = DSP.mixSampleBuffers(singleDelaySamples, interleavedSamples, 0, 1);

  // Apply the master volume to the complete signal
  for (i = 0; i<outputSamples.length; i++) {
    outputSamples[i] *= this.masterVolume;
  }

  return outputSamples;
};