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
   */
  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;
  };
  
  if (typeof module !== 'undefined' && module && module.exports) {
    module.exports = {
      DSP: DSP,
      DFT: DFT,
      FFT: FFT,
      RFFT: RFFT,
      Sampler: Sampler,
      Oscillator: Oscillator,
      ADSR: ADSR,
      IIRFilter: IIRFilter,
      IIRFilter2: IIRFilter2,
      WindowFunction: WindowFunction,
      sinh: sinh,
      Biquad: Biquad,
      GraphicalEq: GraphicalEq,
      MultiDelay: MultiDelay,
      SingleDelay: SingleDelay,
      Reverb: Reverb
    };
  }