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.
 *
 */

// Setup arrays for platforms which do not support byte arrays
Float32Array = (typeof Float32Array === 'undefined') ? Array : Float32Array;
Float64Array = (typeof Float64Array === 'undefined') ? Array : Float64Array;
Uint32Array  = (typeof Uint32Array  === 'undefined') ? Array : Uint32Array;

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

/**
 * DSP is an object which contains general purpose utility functions and constants
 */
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
};

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

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

  return buffer;
};

/**
 * Converts split-stereo (dual mono) sample buffers into a stereo interleaved sample buffer
 *
 * @param {Array} left  A sample buffer
 * @param {Array} right A sample buffer
 *
 * @returns The stereo interleaved buffer
 */
DSP.interleave = function(left, right) {
  if ( left.length !== right.length ) {
    throw "Can not interleave. Channel lengths differ.";
  }
  
  var stereoInterleaved = new Float32Array(left.length * 2);
  
  for (var i = 0, len = left.length; i < len; i++ ) {
    stereoInterleaved[2*i]   = left[i];
    stereoInterleaved[2*i+1] = right[i];
  }
  
  return stereoInterleaved;
};

/**
 * Converts a stereo-interleaved sample buffer into split-stereo (dual mono) sample buffers
 *
 * @param {Array} buffer A stereo-interleaved sample buffer
 *
 * @returns an Array containing left and right channels
 */
DSP.deinterleave = function(buffer) {
  var left  = new Float32Array(buffer.length/2);
  var right = new Float32Array(buffer.length/2);
  var mix   = new Float32Array(buffer.length/2);
  
  for (var i = 0, len = buffer.length/2; i < len; i ++ ) {
    left[i]  = buffer[2*i];
    right[i] = buffer[2*i+1];
    mix[i]   = (left[i] + right[i]) / 2;
  }
  
  return [left, right, mix];
};

/**
 * 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 = function(channel, buffer) {
  return DSP.deinterleave(buffer)[channel];
};

// 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;


/** 
 * DFT is a class for calculating the Discrete Fourier Transform of a signal.
 *
 * @param {Number} bufferSize The size of the sample buffer to be computed
 * @param {Number} sampleRate The sampleRate of the buffer (eg. 44100)
 *
 * @constructor
 */
DFT = function(bufferSize, sampleRate) {
  this.bufferSize = bufferSize;
  this.sampleRate = sampleRate;

  var N = bufferSize/2 * bufferSize;
      
  this.sinTable = new Float32Array(N);
  this.cosTable = new Float32Array(N);
  
  for ( var i = 0; i < N; i++ ) {
    this.sinTable[i] = Math.sin(i * DSP.TWO_PI / bufferSize);
    this.cosTable[i] = Math.cos(i * DSP.TWO_PI / bufferSize);
  }
  
  this.spectrum = new Float32Array(bufferSize/2);
  this.complexValues = new Float32Array(bufferSize/2);
};

/**
 * Performs a forward tranform 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, imag;

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

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

    this.complexValues[k] = {real: real, imag: imag};
  }
  
  for ( var i = 0; i < this.bufferSize/2; i++ ) {
    this.spectrum[i] = 2 * Math.sqrt(Math.pow(this.complexValues[i].real, 2) + Math.pow(this.complexValues[i].imag, 2)) / this.bufferSize;
  }

  return this.spectrum;
};


/** 
 * 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
 */
FFT = function(bufferSize, sampleRate) {
  this.bufferSize = bufferSize;
  this.sampleRate = sampleRate;
  this.spectrum         = new Float32Array(bufferSize/2);
  this.real             = new Float32Array(bufferSize);
  this.imag             = new Float32Array(bufferSize);
    
  this.reverseTable     = new Uint32Array(bufferSize);

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

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

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

  this.sinTable = new Float32Array(bufferSize);
  this.cosTable = new Float32Array(bufferSize);

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

/**
 * Performs a forward tranform 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; }

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

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

  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;
  }

  i = bufferSize/2;
  while(i--) {
    spectrum[i] = 2 * Math.sqrt(real[i] * real[i] + imag[i] * imag[i]) / bufferSize;
  }

  return spectrum;
};

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;

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

  var revReal = new Float32Array(bufferSize);
  var revImag = new Float32Array(bufferSize);
  
  for (var i = 0; i < real.length; i++) {
    revReal[i] = real[reverseTable[i]];
    revImag[i] = imag[reverseTable[i]];
  }
  
  real = revReal;
  imag = revImag;

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

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

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

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

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

        i += halfSize << 1;
      }

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

    halfSize = halfSize << 1;
  }

  var buffer = new Float32Array(bufferSize);
  for (var i = 0; i < bufferSize; i++) {
    buffer[i] = real[i] / bufferSize;
  }

  return buffer;
};

Sampler = 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 Float32Array(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 Float32Array(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) {
  this.frequency = frequency;
  this.step = this.frequency / this.rootFrequency;
};

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
 */
Oscillator = 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 Float32Array(bufferSize);
  this.envelope = null;

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

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

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

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

  this.generateWaveTable = function() {
    Oscillator.waveTable[this.func] = new Float32Array(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
};
  
ADSR = function(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;
};
  
IIRFilter = function(type, cutoff, resonance, sampleRate) {
  this.sampleRate = sampleRate;
  this.cutoff     = cutoff;
  this.resonance  = resonance;

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

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;
};



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

  this.f = Float32Array(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); 
};



WindowFunction = function(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
Biquad = function(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)
    */

    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)
	      var 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)
	      var coeff = sinw0 * Math.sqrt( (A^2 + 1)*(1/this.S - 1) + 2*A );
        this.b0 =    A*( (A+1) + (A-1)*cosw0 + coeff );
        this.b1 = -2*A*( (A-1) + (A+1)*cosw0                   );
        this.b2 =    A*( (A+1) + (A-1)*cosw0 - coeff );
        this.a0 =        (A+1) - (A-1)*cosw0 + coeff;
        this.a1 =    2*( (A-1) - (A+1)*cosw0                   );
        this.a2 =        (A+1) - (A-1)*cosw0 - coeff;
	      break;
    }
    
    this.b0a0 = this.b0/this.a0;
    this.b1a0 = this.b1/this.a0;
    this.b2a0 = this.b2/this.a0;
    this.a1a0 = this.a1/this.a0;
    this.a2a0 = this.a2/this.a0;
  }

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

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

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

      return output;
  }

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

      var len = buffer.length;
      var output = new Float32Array(len);
      
      for ( var i=0; i<len/2; i++ ) {
        output[2*i] = this.b0a0*buffer[2*i] + this.b1a0*this.x_1_l + this.b2a0*this.x_2_l - this.a1a0*this.y_1_l - this.a2a0*this.y_2_l;
	      this.y_2_l = this.y_1_l;
	      this.y_1_l = output[2*i];
	      this.x_2_l = this.x_1_l;
	      this.x_1_l = buffer[2*i];

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

      return output;
  }
};


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

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

  return result;
};

/*  
 *  Frequency response
 *  
 *  Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
 *  Copyright 2010 Ricard Marxer. All rights reserved.
 *
 *  Calculates the frequency response at the given points.
 *
 *  @b b coefficients of the filter
 *  @a a coefficients of the filter
 *  @w w points (normally between -PI and PI) where to calculate the frequency response
 * 
 *  @returns the frequency response in magnitude
 *
 */
DSP.freqz = function(b, a, w) {
  if (!w) {
    w = Float32Array(200);
    for (var i=0;i<w.length; i++) {
      w[i] = DSP.TWO_PI/w.length * i - Math.PI;
    }
  }

  var result = Float32Array(w.length);
  
  var sqrt = Math.sqrt;
  var cos = Math.cos;
  var sin = Math.sin;
  
  for (var i=0; i<w.length; i++) {
    var numerator = {real:0.0, imag:0.0};
    for (var 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 (var 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
 *  
 *  Created by Ricard Marxer <email@ricardmarxer.com> on 2010-05-23.
 *  Copyright 2010 Ricard Marxer. All rights reserved.
 *
 */
// Implementation of a graphic equalizer with a configurable bands-per-octave
// and minimum and maximum frequencies
GraphicalEq = function(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.";
      return;
    }

    if (!gain) {
      throw "A gain must be passed."
      return;
    }
    
    
    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 + "]"
      return;
    }
        
    if (!this.w) {
      this.w = Float32Array(400);
      for (var i=0; i<this.w.length; i++) {
         this.w[i] = Math.PI/this.w.length * i;
      }
    }
    
    var b = [this.filters[bandIndex].b0, this.filters[bandIndex].b1, this.filters[bandIndex].b2];
    var a = [this.filters[bandIndex].a0, this.filters[bandIndex].a1, this.filters[bandIndex].a2];

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

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

      return output;
  }

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

      return output;
  }

}



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

	for(var i=0; i<samples.length; i++){
	
		// delayBufferSamples could contain initial NULL's, return silence in that case
		var delaySample = (this.delayBufferSamples[this.delayOutputPointer]==null ? 0.0 : this.delayBufferSamples[this.delayOutputPointer]);
		
		// Mix normal audio data with delayed audio
		var sample = (delaySample * this.delayVolume) + samples[i]; 
		
		// Add audio data with the delay in the delay buffer
		this.delayBufferSamples[this.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
 */
SingleDelay = function(maxDelayInSamplesSize, delayInSamples, delayVolume){
	this.delayBufferSamples 	= new Float32Array(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 Float32Array
 * 
 * @returns A new Float32Array interleaved or mono non-interleaved as was fed to this function.
 */
SingleDelay.prototype.process = function (samples){
	// NB. Make a copy to put in the output samples to return. 
	var outputSamples = new Float32Array(samples.length);

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

		// Add audio data with the delay in the delay buffer
		this.delayBufferSamples[this.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
 */
Reverb = function(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 = [];
	for(var i = 0; i<this.NR_OF_SINGLEDELAYS; i++){
		var 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(var i = 0; i<this.NR_OF_MULTIDELAYS; i++){
		var 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;
	
	for(var i = 0; i<this.NR_OF_SINGLEDELAYS; i++){
		var delayMultiply = 1.0 + (i/7.0); // 1.0, 1.1, 1.2...
		this.singleDelays[i].setDelayInSamples( Math.round(this.delayInSamples * delayMultiply) );
	}
		
	for(var i = 0; i<this.NR_OF_MULTIDELAYS; i++){
		var 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;
	
	for(var i = 0; i<this.NR_OF_SINGLEDELAYS; i++){
		this.singleDelays[i].setDelayVolume(this.delayVolume);
	}	
	
	for(var i = 0; i<this.NR_OF_MULTIDELAYS; i++){
		this.multiDelays[i].setDelayVolume(this.delayVolume);
	}	
}

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

/**
 * Process a given interleaved float value Array and copies and adds the reverb signal.
 *
 * @param {Array} samples Array containing Float values or a Float32Array
 * 
 * @returns A new Float32Array interleaved buffer.
 */
Reverb.prototype.process = function (interleavedSamples){	
	// NB. Make a copy to put in the output samples to return. 
	var outputSamples = new Float32Array(interleavedSamples.length);
	
	// Perform low pass on the input samples to mimick damp
	var leftRightMix = DSP.deinterleave(interleavedSamples);
	this.LOWPASSL.process( leftRightMix[DSP.LEFT] );
	this.LOWPASSR.process( leftRightMix[DSP.RIGHT] );	
	var filteredSamples = DSP.interleave(leftRightMix[DSP.LEFT], leftRightMix[DSP.RIGHT]);

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

	// Apply the volume of the reverb signal
	for(var i = 0; i<singleDelaySamples.length; i++){
		singleDelaySamples[i] *= this.mixVolume;
	}
	
	// Mix the original signal with the reverb signal
	outputSamples = mixSampleBuffers(singleDelaySamples, interleavedSamples, 0, 1); 

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

/**
 * Helper method (for Reverb) to mix two (interleaved) samplebuffers. It's possible 
 * to negate the second buffer while mixing and to perform a volume correction 
 * on the final signal.
 * 
 * @param {Array} sampleBuffer1 Array containing Float values or a Float32Array
 * @param {Array} sampleBuffer2 Array containing Float values or a Float32Array
 * @param {Boolean} negate When true inverts/flips the audio signal
 * @param {Number} volumeCorrection When you add multiple sample buffers, use this to tame your signal ;)
 * 
 * @returns A new Float32Array interleaved buffer.
 */
function mixSampleBuffers(sampleBuffer1, sampleBuffer2, negate, volumeCorrection){
	var outputSamples = new Float32Array(sampleBuffer1);

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