dsp.js

/* 
 *  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.
 *
 */

////////////////////////////////////////////////////////////////////////////////
//                                  CONSTANTS                                 //
////////////////////////////////////////////////////////////////////////////////

/**
 * DSP is an object which contains general purpose utility functions and constants
 */
var DSP = {
  // Channels
  LEFT: 0,
  RIGHT: 1,
  MIX: 2,

  // Waveforms
  SINE: 1,
  TRIANGLE: 2,
  SAW: 3,
  SQUARE: 4,

  // Filters
  LOWPASS: 0,
  HIGHPASS: 1,
  BANDPASS: 2,
  NOTCH: 3,

  // Window functions
  BARTLETT: 1,
  BARTLETTHANN: 2,
  BLACKMAN: 3,
  COSINE: 4,
  GAUSS: 5,
  HAMMING: 6,
  HANN: 7,
  LANCZOS: 8,
  RECTANGULAR: 9,
  TRIANGULAR: 10,

  // Loop modes
  OFF: 0,
  FW: 1,
  BW: 2,
  FWBW: 3,

  // Math
  TWO_PI: 2 * Math.PI
};

// // Setup arrays for platforms which do not support byte arrays
// function setupTypedArray(name, fallback) {
//   // check if TypedArray exists
//   // typeof on Minefield and Chrome return function, typeof on Webkit returns object.
//   if (typeof this[name] !== "function" && typeof this[name] !== "object") {
//     // nope.. check if WebGLArray exists
//     if (typeof this[fallback] === "function" && typeof this[fallback] !== "object") {
//       this[name] = this[fallback];
//     } else {
//       // nope.. set as Native JS array
//       this[name] = function (obj) {
//         if (obj instanceof Array) {
//           return obj;
//         } else if (typeof obj === "number") {
//           return new Array(obj);
//         }
//       };
//     }
//   }
// }

// setupTypedArray("Float64Array", "WebGLFloatArray");
// setupTypedArray("Int32Array", "WebGLIntArray");
// setupTypedArray("Uint16Array", "WebGLUnsignedShortArray");
// setupTypedArray("Uint8Array", "WebGLUnsignedByteArray");


////////////////////////////////////////////////////////////////////////////////
//                            DSP UTILITY FUNCTIONS                           //
////////////////////////////////////////////////////////////////////////////////

/**
 * Inverts the phase of a signal
 *
 * @param {Array} buffer A sample buffer
 *
 * @returns The inverted sample buffer
 */
DSP.invert = function (buffer) {
  for (var i = 0, len = buffer.length; i < len; i++) {
    buffer[i] *= -1;
  }

  return buffer;
};

/**
 * Converts split-stereo (dual mono) sample buffers into a stereo interleaved sample buffer
 *
 * @param {Array} left  A sample buffer
 * @param {Array} right A sample buffer
 *
 * @returns The stereo interleaved buffer
 */
DSP.interleave = function (left, right) {
  if (left.length !== right.length) {
    throw "Can not interleave. Channel lengths differ.";
  }

  var stereoInterleaved = new Float64Array(left.length * 2);

  for (var i = 0, len = left.length; i < len; i++) {
    stereoInterleaved[2 * i] = left[i];
    stereoInterleaved[2 * i + 1] = right[i];
  }

  return stereoInterleaved;
};

/**
 * Converts a stereo-interleaved sample buffer into split-stereo (dual mono) sample buffers
 *
 * @param {Array} buffer A stereo-interleaved sample buffer
 *
 * @returns an Array containing left and right channels
 */
DSP.deinterleave = (function () {
  var left, right, mix, deinterleaveChannel = [];

  deinterleaveChannel[DSP.MIX] = function (buffer) {
    for (var i = 0, len = buffer.length / 2; i < len; i++) {
      mix[i] = (buffer[2 * i] + buffer[2 * i + 1]) / 2;
    }
    return mix;
  };

  deinterleaveChannel[DSP.LEFT] = function (buffer) {
    for (var i = 0, len = buffer.length / 2; i < len; i++) {
      left[i] = buffer[2 * i];
    }
    return left;
  };

  deinterleaveChannel[DSP.RIGHT] = function (buffer) {
    for (var i = 0, len = buffer.length / 2; i < len; i++) {
      right[i] = buffer[2 * i + 1];
    }
    return right;
  };

  return function (channel, buffer) {
    left = left || new Float64Array(buffer.length / 2);
    right = right || new Float64Array(buffer.length / 2);
    mix = mix || new Float64Array(buffer.length / 2);

    if (buffer.length / 2 !== left.length) {
      left = new Float64Array(buffer.length / 2);
      right = new Float64Array(buffer.length / 2);
      mix = new Float64Array(buffer.length / 2);
    }

    return deinterleaveChannel[channel](buffer);
  };
}());

/**
 * Separates a channel from a stereo-interleaved sample buffer
 *
 * @param {Array}  buffer A stereo-interleaved sample buffer
 * @param {Number} channel A channel constant (LEFT, RIGHT, MIX)
 *
 * @returns an Array containing a signal mono sample buffer
 */
DSP.getChannel = DSP.deinterleave;

/**
 * Helper method (for Reverb) to mix two (interleaved) samplebuffers. It's possible
 * to negate the second buffer while mixing and to perform a volume correction
 * on the final signal.
 *
 * @param {Array} sampleBuffer1 Array containing Float values or a Float64Array
 * @param {Array} sampleBuffer2 Array containing Float values or a Float64Array
 * @param {Boolean} negate When true inverts/flips the audio signal
 * @param {Number} volumeCorrection When you add multiple sample buffers, use this to tame your signal ;)
 *
 * @returns A new Float64Array interleaved buffer.
 */
DSP.mixSampleBuffers = function (sampleBuffer1, sampleBuffer2, negate, volumeCorrection) {
  var outputSamples = new Float64Array(sampleBuffer1);

  for (var i = 0; i < sampleBuffer1.length; i++) {
    outputSamples[i] += (negate ? -sampleBuffer2[i] : sampleBuffer2[i]) / volumeCorrection;
  }

  return outputSamples;
};

// Biquad filter types
DSP.LPF = 0;                // H(s) = 1 / (s^2 + s/Q + 1)
DSP.HPF = 1;                // H(s) = s^2 / (s^2 + s/Q + 1)
DSP.BPF_CONSTANT_SKIRT = 2; // H(s) = s / (s^2 + s/Q + 1)  (constant skirt gain, peak gain = Q)
DSP.BPF_CONSTANT_PEAK = 3;  // H(s) = (s/Q) / (s^2 + s/Q + 1)      (constant 0 dB peak gain)
DSP.NOTCH = 4;              // H(s) = (s^2 + 1) / (s^2 + s/Q + 1)
DSP.APF = 5;                // H(s) = (s^2 - s/Q + 1) / (s^2 + s/Q + 1)
DSP.PEAKING_EQ = 6;         // H(s) = (s^2 + s*(A/Q) + 1) / (s^2 + s/(A*Q) + 1)
DSP.LOW_SHELF = 7;          // H(s) = A * (s^2 + (sqrt(A)/Q)*s + A)/(A*s^2 + (sqrt(A)/Q)*s + 1)
DSP.HIGH_SHELF = 8;         // H(s) = A * (A*s^2 + (sqrt(A)/Q)*s + 1)/(s^2 + (sqrt(A)/Q)*s + A)

// Biquad filter parameter types
DSP.Q = 1;
DSP.BW = 2; // SHARED with BACKWARDS LOOP MODE
DSP.S = 3;

// Find RMS of signal
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 Float64Array(bufferSize / 2);
  this.real = new Float64Array(bufferSize);
  this.imag = new Float64Array(bufferSize);

  this.peakBand = 0;
  this.peak = 0;

  /**
   * Calculates the *middle* frequency of an FFT band.
   *
   * @param {Number} index The index of the FFT band.
   *
   * @returns The middle frequency in Hz.
   */
  this.getBandFrequency = function (index) {
    return this.bandwidth * index + this.bandwidth / 2;
  };

  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
 */
function DFT(bufferSize, sampleRate) {
  FourierTransform.call(this, bufferSize, sampleRate);

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

  this.sinTable = new Float64Array(N);
  this.cosTable = new Float64Array(N);

  for (var i = 0; i < N; i++) {
    this.sinTable[i] = Math.sin(i * TWO_PI / bufferSize);
    this.cosTable[i] = Math.cos(i * TWO_PI / bufferSize);
  }
}

/**
 * Performs a forward transform on the sample buffer.
 * Converts a time domain signal to frequency domain spectra.
 *
 * @param {Array} buffer The sample buffer
 *
 * @returns The frequency spectrum array
 */
DFT.prototype.forward = function (buffer) {
  var real = this.real,
    imag = this.imag,
    rval,
    ival;

  for (var k = 0; k < this.bufferSize / 2; k++) {
    rval = 0.0;
    ival = 0.0;

    for (var n = 0; n < buffer.length; n++) {
      rval += this.cosTable[k * n] * buffer[n];
      ival += this.sinTable[k * n] * buffer[n];
    }

    real[k] = rval;
    imag[k] = ival;
  }

  return this.calculateSpectrum();
};


/**
 * 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
 */
export function FFT(bufferSize, sampleRate) {
  FourierTransform.call(this, bufferSize, sampleRate);

  this.reverseTable = new Uint32Array(bufferSize);

  var limit = 1;
  var bit = bufferSize >> 1;

  var i;

  while (limit < bufferSize) {
    for (i = 0; i < limit; i++) {
      this.reverseTable[i + limit] = this.reverseTable[i] + bit;
    }

    limit = limit << 1;
    bit = bit >> 1;
  }

  this.sinTable = new Float64Array(bufferSize);
  this.cosTable = new Float64Array(bufferSize);

  for (i = 0; i < bufferSize; i++) {
    this.sinTable[i] = Math.sin(-Math.PI / i);
    this.cosTable[i] = Math.cos(-Math.PI / i);
  }
}

/**
 * Performs a forward transform on the sample buffer.
 * Converts a time domain signal to frequency domain spectra.
 *
 * @param {Array} buffer The sample buffer. Buffer Length must be power of 2
 *
 * @returns The frequency spectrum array
 */
FFT.prototype.forward = function (buffer) {
  // Locally scope variables for speed up
  var bufferSize = this.bufferSize,
    cosTable = this.cosTable,
    sinTable = this.sinTable,
    reverseTable = this.reverseTable,
    real = this.real,
    imag = this.imag,
    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 Float64Array(bufferSize);
  var revImag = new Float64Array(bufferSize);

  for (i = 0; i < real.length; i++) {
    revReal[i] = real[reverseTable[i]];
    revImag[i] = imag[reverseTable[i]];
  }

  real = revReal;
  imag = revImag;

  while (halfSize < bufferSize) {
    phaseShiftStepReal = cosTable[halfSize];
    phaseShiftStepImag = sinTable[halfSize];
    currentPhaseShiftReal = 1;
    currentPhaseShiftImag = 0;

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

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

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

        i += halfSize << 1;
      }

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

    halfSize = halfSize << 1;
  }

  var buffer = new Float64Array(bufferSize); // this should be reused instead
  for (i = 0; i < bufferSize; i++) {
    buffer[i] = real[i] / bufferSize;
  }

  return buffer;
};

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

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

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

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

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

function RFFT(bufferSize, sampleRate) {
  FourierTransform.call(this, bufferSize, sampleRate);

  this.trans = new Float64Array(bufferSize);

  this.reverseTable = new Uint32Array(bufferSize);

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

    dest[0] = source[0];

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

      i++;

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

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

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

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

    this.reverseTable[0] = 0;

    do {
      r += halfSize;

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

      i++;

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

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

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

    this.reverseTable[nm1] = nm1;
  };

  this.generateReverseTable();
}


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

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

  this.reverseBinPermute(x, buffer);

  /*
  var reverseTable = this.reverseTable;

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

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

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

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

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

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

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

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

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

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

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

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

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

    e = TWO_PI / n2;

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

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

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

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

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

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

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

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

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

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

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

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

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

      } while (ix < n);
    }
  }

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

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

    spectrum[i] = mag;
  }

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

  return spectrum;
};

function Sampler(file, bufferSize, sampleRate, playStart, playEnd, loopStart, loopEnd, loopMode) {
  this.file = file;
  this.bufferSize = bufferSize;
  this.sampleRate = sampleRate;
  this.playStart = playStart || 0; // 0%
  this.playEnd = playEnd || 1; // 100%
  this.loopStart = loopStart || 0;
  this.loopEnd = loopEnd || 1;
  this.loopMode = loopMode || DSP.OFF;
  this.loaded = false;
  this.samples = [];
  this.signal = new Float64Array(bufferSize);
  this.frameCount = 0;
  this.envelope = null;
  this.amplitude = 1;
  this.rootFrequency = 110; // A2 110
  this.frequency = 550;
  this.step = this.frequency / this.rootFrequency;
  this.duration = 0;
  this.samplesProcessed = 0;
  this.playhead = 0;

  var audio = /* new Audio();*/ document.createElement("AUDIO");
  var self = this;

  this.loadSamples = function (event) {
    var buffer = DSP.getChannel(DSP.MIX, event.frameBuffer);
    for (var i = 0; i < buffer.length; i++) {
      self.samples.push(buffer[i]);
    }
  };

  this.loadComplete = function () {
    // convert flexible js array into a fast typed array
    self.samples = new Float64Array(self.samples);
    self.loaded = true;
  };

  this.loadMetaData = function () {
    self.duration = audio.duration;
  };

  audio.addEventListener("MozAudioAvailable", this.loadSamples, false);
  audio.addEventListener("loadedmetadata", this.loadMetaData, false);
  audio.addEventListener("ended", this.loadComplete, false);
  audio.muted = true;
  audio.src = file;
  audio.play();
}

Sampler.prototype.applyEnvelope = function () {
  this.envelope.process(this.signal);
  return this.signal;
};

Sampler.prototype.generate = function () {
  var frameOffset = this.frameCount * this.bufferSize;

  var loopWidth = this.playEnd * this.samples.length - this.playStart * this.samples.length;
  var playStartSamples = this.playStart * this.samples.length; // ie 0.5 -> 50% of the length
  var playEndSamples = this.playEnd * this.samples.length; // ie 0.5 -> 50% of the length
  var offset;

  for (var i = 0; i < this.bufferSize; i++) {
    switch (this.loopMode) {
      case DSP.OFF:
        this.playhead = Math.round(this.samplesProcessed * this.step + playStartSamples);
        if (this.playhead < (this.playEnd * this.samples.length)) {
          this.signal[i] = this.samples[this.playhead] * this.amplitude;
        } else {
          this.signal[i] = 0;
        }
        break;

      case DSP.FW:
        this.playhead = Math.round((this.samplesProcessed * this.step) % loopWidth + playStartSamples);
        if (this.playhead < (this.playEnd * this.samples.length)) {
          this.signal[i] = this.samples[this.playhead] * this.amplitude;
        }
        break;

      case DSP.BW:
        this.playhead = playEndSamples - Math.round((this.samplesProcessed * this.step) % loopWidth);
        if (this.playhead < (this.playEnd * this.samples.length)) {
          this.signal[i] = this.samples[this.playhead] * this.amplitude;
        }
        break;

      case DSP.FWBW:
        if (Math.floor(this.samplesProcessed * this.step / loopWidth) % 2 === 0) {
          this.playhead = Math.round((this.samplesProcessed * this.step) % loopWidth + playStartSamples);
        } else {
          this.playhead = playEndSamples - Math.round((this.samplesProcessed * this.step) % loopWidth);
        }
        if (this.playhead < (this.playEnd * this.samples.length)) {
          this.signal[i] = this.samples[this.playhead] * this.amplitude;
        }
        break;
    }
    this.samplesProcessed++;
  }

  this.frameCount++;

  return this.signal;
};

Sampler.prototype.setFreq = function (frequency) {
  var totalProcessed = this.samplesProcessed * this.step;
  this.frequency = frequency;
  this.step = this.frequency / this.rootFrequency;
  this.samplesProcessed = Math.round(totalProcessed / this.step);
};

Sampler.prototype.reset = function () {
  this.samplesProcessed = 0;
  this.playhead = 0;
};

/**
 * Oscillator class for generating and modifying signals
 *
 * @param {Number} type       A waveform constant (eg. DSP.SINE)
 * @param {Number} frequency  Initial frequency of the signal
 * @param {Number} amplitude  Initial amplitude of the signal
 * @param {Number} bufferSize Size of the sample buffer to generate
 * @param {Number} sampleRate The sample rate of the signal
 *
 * @contructor
 */
function Oscillator(type, frequency, amplitude, bufferSize, sampleRate) {
  this.frequency = frequency;
  this.amplitude = amplitude;
  this.bufferSize = bufferSize;
  this.sampleRate = sampleRate;
  //this.pulseWidth = pulseWidth;
  this.frameCount = 0;

  this.waveTableLength = 2048;

  this.cyclesPerSample = frequency / sampleRate;

  this.signal = new Float64Array(bufferSize);
  this.envelope = null;

  switch (parseInt(type, 10)) {
    case DSP.TRIANGLE:
      this.func = Oscillator.Triangle;
      break;

    case DSP.SAW:
      this.func = Oscillator.Saw;
      break;

    case DSP.SQUARE:
      this.func = Oscillator.Square;
      break;

    default:
    case DSP.SINE:
      this.func = Oscillator.Sine;
      break;
  }

  this.generateWaveTable = function () {
    Oscillator.waveTable[this.func] = new Float64Array(2048);
    var waveTableTime = this.waveTableLength / this.sampleRate;
    var waveTableHz = 1 / waveTableTime;

    for (var i = 0; i < this.waveTableLength; i++) {
      Oscillator.waveTable[this.func][i] = this.func(i * waveTableHz / this.sampleRate);
    }
  };

  if (typeof Oscillator.waveTable === 'undefined') {
    Oscillator.waveTable = {};
  }

  if (typeof Oscillator.waveTable[this.func] === 'undefined') {
    this.generateWaveTable();
  }

  this.waveTable = Oscillator.waveTable[this.func];
}

/**
 * Set the amplitude of the signal
 *
 * @param {Number} amplitude The amplitude of the signal (between 0 and 1)
 */
Oscillator.prototype.setAmp = function (amplitude) {
  if (amplitude >= 0 && amplitude <= 1) {
    this.amplitude = amplitude;
  } else {
    throw "Amplitude out of range (0..1).";
  }
};

/**
 * Set the frequency of the signal
 *
 * @param {Number} frequency The frequency of the signal
 */
Oscillator.prototype.setFreq = function (frequency) {
  this.frequency = frequency;
  this.cyclesPerSample = frequency / this.sampleRate;
};

// Add an oscillator
Oscillator.prototype.add = function (oscillator) {
  for (var i = 0; i < this.bufferSize; i++) {
    //this.signal[i] += oscillator.valueAt(i);
    this.signal[i] += oscillator.signal[i];
  }

  return this.signal;
};

// Add a signal to the current generated osc signal
Oscillator.prototype.addSignal = function (signal) {
  for (var i = 0; i < signal.length; i++) {
    if (i >= this.bufferSize) {
      break;
    }
    this.signal[i] += signal[i];

    /*
    // Constrain amplitude
    if ( this.signal[i] > 1 ) {
      this.signal[i] = 1;
    } else if ( this.signal[i] < -1 ) {
      this.signal[i] = -1;
    }
    */
  }
  return this.signal;
};

// Add an envelope to the oscillator
Oscillator.prototype.addEnvelope = function (envelope) {
  this.envelope = envelope;
};

Oscillator.prototype.applyEnvelope = function () {
  this.envelope.process(this.signal);
};

Oscillator.prototype.valueAt = function (offset) {
  return this.waveTable[offset % this.waveTableLength];
};

Oscillator.prototype.generate = function () {
  var frameOffset = this.frameCount * this.bufferSize;
  var step = this.waveTableLength * this.frequency / this.sampleRate;
  var offset;

  for (var i = 0; i < this.bufferSize; i++) {
    //var step = (frameOffset + i) * this.cyclesPerSample % 1;
    //this.signal[i] = this.func(step) * this.amplitude;
    //this.signal[i] = this.valueAt(Math.round((frameOffset + i) * step)) * this.amplitude;
    offset = Math.round((frameOffset + i) * step);
    this.signal[i] = this.waveTable[offset % this.waveTableLength] * this.amplitude;
  }

  this.frameCount++;

  return this.signal;
};

Oscillator.Sine = function (step) {
  return Math.sin(DSP.TWO_PI * step);
};

Oscillator.Square = function (step) {
  return step < 0.5 ? 1 : -1;
};

Oscillator.Saw = function (step) {
  return 2 * (step - Math.round(step));
};

Oscillator.Triangle = function (step) {
  return 1 - 4 * Math.abs(Math.round(step) - step);
};

Oscillator.Pulse = function (step) {
  // stub
};

function ADSR(attackLength, decayLength, sustainLevel, sustainLength, releaseLength, sampleRate) {
  this.sampleRate = sampleRate;
  // Length in seconds
  this.attackLength = attackLength;
  this.decayLength = decayLength;
  this.sustainLevel = sustainLevel;
  this.sustainLength = sustainLength;
  this.releaseLength = releaseLength;
  this.sampleRate = sampleRate;

  // Length in samples
  this.attackSamples = attackLength * sampleRate;
  this.decaySamples = decayLength * sampleRate;
  this.sustainSamples = sustainLength * sampleRate;
  this.releaseSamples = releaseLength * sampleRate;

  // Updates the envelope sample positions
  this.update = function () {
    this.attack = this.attackSamples;
    this.decay = this.attack + this.decaySamples;
    this.sustain = this.decay + this.sustainSamples;
    this.release = this.sustain + this.releaseSamples;
  };

  this.update();

  this.samplesProcessed = 0;
}

ADSR.prototype.noteOn = function () {
  this.samplesProcessed = 0;
  this.sustainSamples = this.sustainLength * this.sampleRate;
  this.update();
};

// Send a note off when using a sustain of infinity to let the envelope enter the release phase
ADSR.prototype.noteOff = function () {
  this.sustainSamples = this.samplesProcessed - this.decaySamples;
  this.update();
};

ADSR.prototype.processSample = function (sample) {
  var amplitude = 0;

  if (this.samplesProcessed <= this.attack) {
    amplitude = 0 + (1 - 0) * ((this.samplesProcessed - 0) / (this.attack - 0));
  } else if (this.samplesProcessed > this.attack && this.samplesProcessed <= this.decay) {
    amplitude = 1 + (this.sustainLevel - 1) * ((this.samplesProcessed - this.attack) / (this.decay - this.attack));
  } else if (this.samplesProcessed > this.decay && this.samplesProcessed <= this.sustain) {
    amplitude = this.sustainLevel;
  } else if (this.samplesProcessed > this.sustain && this.samplesProcessed <= this.release) {
    amplitude = this.sustainLevel + (0 - this.sustainLevel) * ((this.samplesProcessed - this.sustain) / (this.release - this.sustain));
  }

  return sample * amplitude;
};

ADSR.prototype.value = function () {
  var amplitude = 0;

  if (this.samplesProcessed <= this.attack) {
    amplitude = 0 + (1 - 0) * ((this.samplesProcessed - 0) / (this.attack - 0));
  } else if (this.samplesProcessed > this.attack && this.samplesProcessed <= this.decay) {
    amplitude = 1 + (this.sustainLevel - 1) * ((this.samplesProcessed - this.attack) / (this.decay - this.attack));
  } else if (this.samplesProcessed > this.decay && this.samplesProcessed <= this.sustain) {
    amplitude = this.sustainLevel;
  } else if (this.samplesProcessed > this.sustain && this.samplesProcessed <= this.release) {
    amplitude = this.sustainLevel + (0 - this.sustainLevel) * ((this.samplesProcessed - this.sustain) / (this.release - this.sustain));
  }

  return amplitude;
};

ADSR.prototype.process = function (buffer) {
  for (var i = 0; i < buffer.length; i++) {
    buffer[i] *= this.value();

    this.samplesProcessed++;
  }

  return buffer;
};


ADSR.prototype.isActive = function () {
  if (this.samplesProcessed > this.release || this.samplesProcessed === -1) {
    return false;
  } else {
    return true;
  }
};

ADSR.prototype.disable = function () {
  this.samplesProcessed = -1;
};

function IIRFilter(type, cutoff, resonance, sampleRate) {
  this.sampleRate = sampleRate;

  switch (type) {
    case DSP.LOWPASS:
    case DSP.LP12:
      this.func = new IIRFilter.LP12(cutoff, resonance, sampleRate);
      break;
  }
}

IIRFilter.prototype.__defineGetter__('cutoff',
  function () {
    return this.func.cutoff;
  }
);

IIRFilter.prototype.__defineGetter__('resonance',
  function () {
    return this.func.resonance;
  }
);

IIRFilter.prototype.set = function (cutoff, resonance) {
  this.func.calcCoeff(cutoff, resonance);
};

IIRFilter.prototype.process = function (buffer) {
  this.func.process(buffer);
};

// Add an envelope to the filter
IIRFilter.prototype.addEnvelope = function (envelope) {
  if (envelope instanceof ADSR) {
    this.func.addEnvelope(envelope);
  } else {
    throw "Not an envelope.";
  }
};

IIRFilter.LP12 = function (cutoff, resonance, sampleRate) {
  this.sampleRate = sampleRate;
  this.vibraPos = 0;
  this.vibraSpeed = 0;
  this.envelope = false;

  this.calcCoeff = function (cutoff, resonance) {
    this.w = 2.0 * Math.PI * cutoff / this.sampleRate;
    this.q = 1.0 - this.w / (2.0 * (resonance + 0.5 / (1.0 + this.w)) + this.w - 2.0);
    this.r = this.q * this.q;
    this.c = this.r + 1.0 - 2.0 * Math.cos(this.w) * this.q;

    this.cutoff = cutoff;
    this.resonance = resonance;
  };

  this.calcCoeff(cutoff, resonance);

  this.process = function (buffer) {
    for (var i = 0; i < buffer.length; i++) {
      this.vibraSpeed += (buffer[i] - this.vibraPos) * this.c;
      this.vibraPos += this.vibraSpeed;
      this.vibraSpeed *= this.r;

      /*
      var temp = this.vibraPos;
     
      if ( temp > 1.0 ) {
        temp = 1.0;
      } else if ( temp < -1.0 ) {
        temp = -1.0;
      } else if ( temp != temp ) {
        temp = 1;
      }
     
      buffer[i] = temp;
      */

      if (this.envelope) {
        buffer[i] = (buffer[i] * (1 - this.envelope.value())) + (this.vibraPos * this.envelope.value());
        this.envelope.samplesProcessed++;
      } else {
        buffer[i] = this.vibraPos;
      }
    }
  };
};

IIRFilter.LP12.prototype.addEnvelope = function (envelope) {
  this.envelope = envelope;
};

function IIRFilter2(type, cutoff, resonance, sampleRate) {
  this.type = type;
  this.cutoff = cutoff;
  this.resonance = resonance;
  this.sampleRate = sampleRate;

  this.f = Float64Array(4);
  this.f[0] = 0.0; // lp
  this.f[1] = 0.0; // hp
  this.f[2] = 0.0; // bp
  this.f[3] = 0.0; // br 

  this.calcCoeff = function (cutoff, resonance) {
    this.freq = 2 * Math.sin(Math.PI * Math.min(0.25, cutoff / (this.sampleRate * 2)));
    this.damp = Math.min(2 * (1 - Math.pow(resonance, 0.25)), Math.min(2, 2 / this.freq - this.freq * 0.5));
  };

  this.calcCoeff(cutoff, resonance);
}

IIRFilter2.prototype.process = function (buffer) {
  var input, output;
  var f = this.f;

  for (var i = 0; i < buffer.length; i++) {
    input = buffer[i];

    // first pass
    f[3] = input - this.damp * f[2];
    f[0] = f[0] + this.freq * f[2];
    f[1] = f[3] - f[0];
    f[2] = this.freq * f[1] + f[2];
    output = 0.5 * f[this.type];

    // second pass
    f[3] = input - this.damp * f[2];
    f[0] = f[0] + this.freq * f[2];
    f[1] = f[3] - f[0];
    f[2] = this.freq * f[1] + f[2];
    output += 0.5 * f[this.type];

    if (this.envelope) {
      buffer[i] = (buffer[i] * (1 - this.envelope.value())) + (output * this.envelope.value());
      this.envelope.samplesProcessed++;
    } else {
      buffer[i] = output;
    }
  }
};

IIRFilter2.prototype.addEnvelope = function (envelope) {
  if (envelope instanceof ADSR) {
    this.envelope = envelope;
  } else {
    throw "This is not an envelope.";
  }
};

IIRFilter2.prototype.set = function (cutoff, resonance) {
  this.calcCoeff(cutoff, resonance);
};



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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

    var alpha = 0;

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

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

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

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

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

    var coeff;

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

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

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

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

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

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

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

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

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

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

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

    var len = buffer.length;
    var output = new Float64Array(len);

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

    return output;
  };

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

    var len = buffer.length;
    var output = new Float64Array(len);

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

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

    return output;
  };
}

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

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

  var result = Float64Array(buffer.length);
  for (var i = 0; i < buffer.length; i++) {
    result[i] = 20.0 * log(max(buffer[i], minMag));
  }

  return result;
};

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

  if (!w) {
    w = Float64Array(200);
    for (i = 0; i < w.length; i++) {
      w[i] = DSP.TWO_PI / w.length * i - Math.PI;
    }
  }

  var result = Float64Array(w.length);

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

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

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

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

  return result;
};

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

  this.bandsPerOctave = 1.0;

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

  this.calculateFreqzs = true;

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

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

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

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

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

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

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

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

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

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

    if (!this.w) {
      this.w = Float64Array(400);
      for (var i = 0; i < this.w.length; i++) {
        this.w[i] = Math.PI / this.w.length * i;
      }
    }

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

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

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

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

    return output;
  };

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

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

    return output;
  };
}

/**
 * MultiDelay effect by Almer Thie (http://code.almeros.com).
 * Copyright 2010 Almer Thie. All rights reserved.
 * Example: http://code.almeros.com/code-examples/delay-firefox-audio-api/
 *
 * This is a delay that feeds it's own delayed signal back into its circular
 * buffer. Also known as a CombFilter.
 *
 * Compatible with interleaved stereo (or more channel) buffers and
 * non-interleaved mono buffers.
 *
 * @param {Number} maxDelayInSamplesSize Maximum possible delay in samples (size of circular buffer)
 * @param {Number} delayInSamples Initial delay in samples
 * @param {Number} masterVolume Initial master volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 * @param {Number} delayVolume Initial feedback delay volume. Float value: 0.0 (silence), 1.0 (normal), >1.0 (amplify)
 *
 * @constructor
 */
function MultiDelay(maxDelayInSamplesSize, delayInSamples, masterVolume, delayVolume) {
  this.delayBufferSamples = new Float64Array(maxDelayInSamplesSize); // The maximum size of delay
  this.delayInputPointer = delayInSamples;
  this.delayOutputPointer = 0;

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

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

  this.delayInputPointer = this.delayOutputPointer + delayInSamples;

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

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

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

/**
 * Process a given interleaved or mono non-interleaved float value Array and adds the delayed audio.
 *
 * @param {Array} samples Array containing Float values or a Float64Array
 *
 * @returns A new Float64Array interleaved or mono non-interleaved as was fed to this function.
 */
MultiDelay.prototype.process = function (samples) {
  // NB. Make a copy to put in the output samples to return.
  var outputSamples = new Float64Array(samples.length);

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

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

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

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

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

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

  return outputSamples;
};

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

function SingleDelay(maxDelayInSamplesSize, delayInSamples, delayVolume) {
  this.delayBufferSamples = new Float64Array(maxDelayInSamplesSize); // The maximum size of delay
  this.delayInputPointer = delayInSamples;
  this.delayOutputPointer = 0;

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

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

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

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

/**
 * Process a given interleaved or mono non-interleaved float value Array and
 * returns the delayed audio.
 *
 * @param {Array} samples Array containing Float values or a Float64Array
 *
 * @returns A new Float64Array interleaved or mono non-interleaved as was fed to this function.
 */
SingleDelay.prototype.process = function (samples) {
  // NB. Make a copy to put in the output samples to return.
  var outputSamples = new Float64Array(samples.length);

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

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

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

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

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

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

    this.delayOutputPointer++;

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

  return outputSamples;
};

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

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

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

  this.singleDelays = [];

  var i, delayMultiply;

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

  this.multiDelays = [];

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

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

  var i, delayMultiply;

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

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

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

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

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

  var i;

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

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

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

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

/**
 * Process a given interleaved float value Array and copies and adds the reverb signal.
 *
 * @param {Array} samples Array containing Float values or a Float64Array
 *
 * @returns A new Float64Array interleaved buffer.
 */
Reverb.prototype.process = function (interleavedSamples) {
  // NB. Make a copy to put in the output samples to return.
  var outputSamples = new Float64Array(interleavedSamples.length);

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

  var i;

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

  // Process SingleDelays in series
  var singleDelaySamples = new Float64Array(outputSamples.length);
  for (i = 0; i < this.NR_OF_SINGLEDELAYS; i++) {
    // Invert the signal of every even singleDelay
    singleDelaySamples = DSP.mixSampleBuffers(singleDelaySamples, this.singleDelays[i].process(outputSamples), 2 % i === 0, 1);
  }

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

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

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

  return outputSamples;
};

// export FFT

// export default {
//   DSP: DSP,
//   DFT: DFT,
//   FFT: FFT,
//   RFFT: RFFT,
//   Sampler: Sampler,
//   Oscillator: Oscillator,
//   ADSR: ADSR,
//   IIRFilter: IIRFilter,
//   IIRFilter2: IIRFilter2,
//   WindowFunction: WindowFunction,
//   sinh: sinh,
//   Biquad: Biquad,
//   GraphicalEq: GraphicalEq,
//   MultiDelay: MultiDelay,
//   SingleDelay: SingleDelay,
//   Reverb: Reverb
// };