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## ddf.minim.analysis
Class FourierTransform

    java.lang.Object
      ddf.minim.analysis.FourierTransform

Direct Known Subclasses:
  DFT, FFT

    public abstract class FourierTransform

extends java.lang.Object

A Fourier Transform is an algorithm that transforms a signal in the time domain, such as a sample buffer, into a signal in the frequency domain, often called the spectrum. The spectrum does not represent individual frequencies, but actually represents frequency bands centered on particular frequencies. The center frequency of each band is usually expressed as a fraction of the sampling rate of the time domain signal and is equal to the index of the frequency band divided by the total number of bands. The total number of frequency bands is usually equal to the length of the time domain signal, but access is only provided to frequency bands with indices less than half the length, because they correspond to frequencies below the Nyquist frequency. In other words, given a signal of length `N`, there will be `N/2` frequency bands in the spectrum.

As an example, if you construct a FourierTransform with a `timeSize` of 1024 and and a `sampleRate` of 44100 Hz, then the spectrum will contain values for frequencies below 22010 Hz, which is the Nyquist frequency (half the sample rate). If you ask for the value of band number 5, this will correspond to a frequency band centered on `5/1024 * 44100 = 0.0048828125 * 44100 = 215 Hz`. The width of that frequency band is equal to `2/1024`, expressed as a fraction of the total bandwidth of the spectrum. The total bandwith of the spectrum is equal to the Nyquist frequency, which in this case is 22100, so the bandwidth is equal to about 50 Hz. It is not necessary for you to remember all of these relationships, though it is good to be aware of them. The function `getFreq()` allows you to query the spectrum with a frequency in Hz and the function `getBandWidth()` will return the bandwidth in Hz of each frequency band in the spectrum.

Usage

A typical usage of a FourierTransform is to analyze a signal so that the frequency spectrum may be represented in some way, typically with vertical lines. You could do this in Processing with the following code, where `audio` is an AudioSource and `fft` is an FFT (one of the derived classes of FourierTransform).

     fft.forward(audio.left);
     for (int i = 0; i < fft.specSize(); i++)
     {
       // draw the line for frequency band i, scaling it by 4 so we can see it a bit better
       line(i, height, i, height - fft.getBand(i) * 4);
     }

Windowing

Windowing is the process of shaping the audio samples before transforming them to the frequency domain. If you call the `window()` function with an appropriate constant, such as FourierTransform.HAMMING, the sample buffers passed to the object for analysis will be shaped by the current window before being transformed. The result of using a window is to reduce the noise in the spectrum somewhat.

Averages

FourierTransform also has functions that allow you to request the creation of an average spectrum. An average spectrum is simply a spectrum with fewer bands than the full spectrum where each average band is the average of the amplitudes of some number of contiguous frequency bands in the full spectrum.

`linAverages()` allows you to specify the number of averages that you want and will group frequency bands into groups of equal number. So if you have a spectrum with 512 frequency bands and you ask for 64 averages, each average will span 8 bands of the full spectrum.

`logAverages()` will group frequency bands by octave and allows you to specify the size of the smallest octave to use (in Hz) and also how many bands to split each octave into. So you might ask for the smallest octave to be 60 Hz and to split each octave into two bands. The result is that the bandwidth of each average is different. One frequency is an octave above another when it's frequency is twice that of the lower frequency. So, 120 Hz is an octave above 60 Hz, 240 Hz is an octave above 120 Hz, and so on. When octaves are split, they are split based on Hz, so if you split the octave 60-120 Hz in half, you will get 60-90Hz and 90-120Hz. You can see how these bandwidths increase as your octave sizes grow. For instance, the last octave will always span `sampleRate/4 - sampleRate/2`, which in the case of audio sampled at 44100 Hz is 11025-22010 Hz. These logarithmically spaced averages are usually much more useful than the full spectrum or the linearly spaced averages because they map more directly to how humans perceive sound.

`calcAvg()` allows you to specify the frequency band you want an average calculated for. You might ask for 60-500Hz and this function will group together the bands from the full spectrum that fall into that range and average their amplitudes for you.

If you don't want any averages calculated, then you can call `noAverages()`. This will not impact your ability to use `calcAvg()`, it will merely prevent the object from calculating an average array every time you use `forward()`.

Inverse Transform

FourierTransform also supports taking the inverse transform of a spectrum. This means that a frequency spectrum will be transformed into a time domain signal and placed in a provided sample buffer. The length of the time domain signal will be `timeSize()` long. The `set` and `scale` functions allow you the ability to shape the spectrum already stored in the object before taking the inverse transform. You might use these to filter frequencies in a spectrum or modify it in some other way.

Author:
  Damien Di Fede
See Also:
  The Discrete Fourier Transform

Field Summary
`protected float[]` | `averages`
`protected int` | `avgPerOctave`
`protected float` | `bandWidth`
`static int` | `HAMMING` A constant indicating a Hamming window should be used on sample buffers.
`protected float[]` | `imag`
`protected static int` | `LINAVG`
`protected static int` | `LOGAVG`
`protected static int` | `NOAVG`
`static int` | `NONE` A constant indicating no window should be used on sample buffers.
`protected int` | `octaves`
`protected float[]` | `real`
`protected int` | `sampleRate`
`protected float[]` | `spectrum`
`protected int` | `timeSize`
`protected static float` | `TWO_PI`
`protected int` | `whichAverage`
`protected int` | `whichWindow`

Method Summary
`protected abstract void` | `allocateArrays()`
`int` | `avgSize()` Returns the number of averages currently being calculated.
`float` | `calcAvg(float lowFreq, float hiFreq)` Calculate the average amplitude of the frequency band bounded by `lowFreq` and `hiFreq`, inclusive.
`protected void` | `doWindow(float[] samples)`
`protected void` | `fillSpectrum()`
`void` | `forward(AudioBuffer buffer)` Performs a forward transform on `buffer`.
`abstract void` | `forward(float[] buffer)` Performs a forward transform on `buffer`.
`int` | `freqToIndex(float freq)` Returns the index of the frequency band that contains the requested frequency.
`float` | `getAvg(int i)` Gets the value of the `ith` average.
`float` | `getBand(int i)` Returns the amplitude of the requested frequency band.
`float` | `getBandWidth()` Returns the width of each frequency band in the spectrum (in Hz).
`float` | `getFreq(float freq)` Gets the amplitude of the requested frequency in the spectrum.
`protected void` | `hamming(float[] samples)`
`void` | `inverse(AudioBuffer buffer)` Performs an inverse transform of the frequency spectrum and places the result in `buffer`.
`abstract void` | `inverse(float[] buffer)` Performs an inverse transform of the frequency spectrum and places the result in `buffer`.
`void` | `inverse(float[] freqReal, float[] freqImag, float[] buffer)` Performs an inverse transform of the frequency spectrum represented by freqReal and freqImag and places the result in buffer.
`void` | `linAverages(int numAvg)` Sets the number of averages used when computing the spectrum and spaces the averages in a linear manner.
`void` | `logAverages(int minBandwidth, int bandsPerOctave)` Sets the number of averages used when computing the spectrum based on the minimum bandwidth for an octave and the number of bands per octave.
`void` | `noAverages()` Sets the object to not compute averages.
`abstract void` | `scaleBand(int i, float s)` Scales the amplitude of the `ith` frequency band by `s`.
`void` | `scaleFreq(float freq, float s)` Scales the amplitude of the requested frequency by `a`.
`abstract void` | `setBand(int i, float a)` Sets the amplitude of the `ith` frequency band to `a`.
`protected void` | `setComplex(float[] r, float[] i)`
`void` | `setFreq(float freq, float a)` Sets the amplitude of the requested frequency in the spectrum to `a`.
`int` | `specSize()` Returns the size of the spectrum created by this transform.
`int` | `timeSize()` Returns the length of the time domain signal expected by this transform.
`void` | `window(int which)` Sets the window to use on the samples before taking the forward transform.

Methods inherited from class java.lang.Object
`clone, equals, finalize, getClass, hashCode, notify, notifyAll, toString, wait, wait, wait`

Field Detail

### NONE

    public static final int NONE

  A constant indicating no window should be used on sample buffers.

  See Also:
    Constant Field Values

### HAMMING

    public static final int HAMMING

  A constant indicating a Hamming window should be used on sample buffers.

  See Also:
    Constant Field Values

### LINAVG

    protected static final int LINAVG

See Also:
  Constant Field Values

### LOGAVG

    protected static final int LOGAVG

See Also:
  Constant Field Values

### NOAVG

    protected static final int NOAVG

See Also:
  Constant Field Values

### TWO_PI

    protected static final float TWO_PI

See Also:
  Constant Field Values

### timeSize

    protected int timeSize

### sampleRate

    protected int sampleRate

### bandWidth

    protected float bandWidth

### whichWindow

    protected int whichWindow

### real

    protected float[] real

### imag

    protected float[] imag

### spectrum

    protected float[] spectrum

### averages

    protected float[] averages

### whichAverage

    protected int whichAverage

### octaves

    protected int octaves

### avgPerOctave

    protected int avgPerOctave

Method Detail

### allocateArrays

    protected abstract void allocateArrays()

### setComplex

    protected void setComplex(float[] r,
                              float[] i)

### fillSpectrum

    protected void fillSpectrum()

### noAverages

    public void noAverages()

  Sets the object to not compute averages.

### linAverages

    public void linAverages(int numAvg)

  Sets the number of averages used when computing the spectrum and spaces the averages in a linear manner. In other words, each average band will be `specSize() / numAvg` bands wide.

  Parameters:
    `numAvg` - how many averages to compute

### logAverages

    public void logAverages(int minBandwidth,
                            int bandsPerOctave)

  Sets the number of averages used when computing the spectrum based on the minimum bandwidth for an octave and the number of bands per octave. For example, with audio that has a sample rate of 44100 Hz, `logAverages(11, 1)` will result in 12 averages, each corresponding to an octave, the first spanning 0 to 11 Hz. To ensure that each octave band is a full octave, the number of octaves is computed by dividing the Nyquist frequency by two, and then the result of that by two, and so on. This means that the actual bandwidth of the lowest octave may not be exactly the value specified.

  Parameters:
    `minBandwidth` - the minimum bandwidth used for an octave
    `bandsPerOctave` - how many bands to split each octave into

### window

    public void window(int which)

  Sets the window to use on the samples before taking the forward transform. If an invalid window is asked for, an error will be reported and the current window will not be changed.

  Parameters:
    `which` - FourierTransform.HAMMING or FourierTransform.NONE

### doWindow

    protected void doWindow(float[] samples)

### hamming

    protected void hamming(float[] samples)

### timeSize

    public int timeSize()

  Returns the length of the time domain signal expected by this transform.

  Returns:
    the length of the time domain signal expected by this transform

### specSize

    public int specSize()

  Returns the size of the spectrum created by this transform. In other words, the number of frequency bands produced by this transform. This is typically equal to `timeSize()/2 + 1`, see above for an explanation.

  Returns:
    the size of the spectrum

### getBand

    public float getBand(int i)

  Returns the amplitude of the requested frequency band.

  Parameters:
    `i` - the index of a frequency band
  Returns:
    the amplitude of the requested frequency band

### getBandWidth

    public float getBandWidth()

  Returns the width of each frequency band in the spectrum (in Hz). It should be noted that the bandwidth of the first and last frequency bands is half as large as the value returned by this function.

  Returns:
    the width of each frequency band in Hz.

### setBand

    public abstract void setBand(int i,
                                 float a)

  Sets the amplitude of the `ith` frequency band to `a`. You can use this to shape the spectrum before using `inverse()`.

  Parameters:
    `i` - the frequency band to modify
    `a` - the new amplitude

### scaleBand

    public abstract void scaleBand(int i,
                                   float s)

  Scales the amplitude of the `ith` frequency band by `s`. You can use this to shape the spectrum before using `inverse()`.

  Parameters:
    `i` - the frequency band to modify
    `s` - the scaling factor

### freqToIndex

    public int freqToIndex(float freq)

  Returns the index of the frequency band that contains the requested frequency.

  Parameters:
    `freq` - the frequency you want the index for (in Hz)
  Returns:
    the index of the frequency band that contains freq

### getFreq

    public float getFreq(float freq)

  Gets the amplitude of the requested frequency in the spectrum.

  Parameters:
    `freq` - the frequency in Hz
  Returns:
    the amplitude of the frequency in the spectrum

### setFreq

    public void setFreq(float freq,
                        float a)

  Sets the amplitude of the requested frequency in the spectrum to `a`.

  Parameters:
    `freq` - the frequency in Hz
    `a` - the new amplitude

### scaleFreq

    public void scaleFreq(float freq,
                          float s)

  Scales the amplitude of the requested frequency by `a`.

  Parameters:
    `freq` - the frequency in Hz
    `s` - the scaling factor

### avgSize

    public int avgSize()

  Returns the number of averages currently being calculated.

  Returns:
    the length of the averages array

### getAvg

    public float getAvg(int i)

  Gets the value of the `ith` average.

  Parameters:
    `i` - the average you want the value of
  Returns:
    the value of the requested average

### calcAvg

    public float calcAvg(float lowFreq,
                         float hiFreq)

  Calculate the average amplitude of the frequency band bounded by `lowFreq` and `hiFreq`, inclusive.

  Parameters:
    `lowFreq` - the lower bound of the band
    `hiFreq` - the upper bound of the band
  Returns:
    the average of all spectrum values within the bounds

### forward

    public abstract void forward(float[] buffer)

  Performs a forward transform on `buffer`.

  Parameters:
    `buffer` - the buffer to analyze

### forward

    public void forward(AudioBuffer buffer)

  Performs a forward transform on `buffer`.

  Parameters:
    `buffer` - the buffer to analyze

### inverse

    public abstract void inverse(float[] buffer)

  Performs an inverse transform of the frequency spectrum and places the result in `buffer`.

  Parameters:
    `buffer` - the buffer to place the result of the inverse transform in

### inverse

    public void inverse(AudioBuffer buffer)

  Performs an inverse transform of the frequency spectrum and places the result in `buffer`.

  Parameters:
    `buffer` - the buffer to place the result of the inverse transform in

### inverse

    public void inverse(float[] freqReal,
                        float[] freqImag,
                        float[] buffer)

  Performs an inverse transform of the frequency spectrum represented by freqReal and freqImag and places the result in buffer.

  Parameters:
    `freqReal` - the real part of the frequency spectrum
    `freqImag` - the imaginary part the frequency spectrum
    `buffer` - the buffer to place the inverse transform in

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