ddf.minim.analysis.FourierTransform Maven / Gradle / Ivy
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/*
* Copyright (c) 2007 - 2008 by Damien Di Fede
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU Library General Public License as published
* by the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
package ddf.minim.analysis;
// TODO: Auto-generated Javadoc
/**
* 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 22050, 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. The Fourier Transform assumes the sample buffer
* is is a repetitive signal, if a sample buffer is not truly periodic within
* the measured interval sharp discontinuities may arise that can introduce
* spectral leakage. Spectral leakage is the speading of signal energy across
* multiple FFT bins. This "spreading" can drown out narrow band signals and
* hinder detection.
*
*
* A windowing
* function attempts to reduce spectral leakage by attenuating the measured
* sample buffer at its end points to eliminate discontinuities. If you call the
* window() function with an appropriate WindowFunction, such as
* HammingWindow(), 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 leakage 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 The Discrete Fourier
* Transform
*/
public abstract class FourierTransform {
/**
* A constant indicating no window should be used on sample buffers. Also
* referred as a Rectangular window.
*/
public static final WindowFunction NONE = new RectangularWindow();
/**
* A constant indicating a Hamming
* window should be used on sample buffers.
*/
public static final WindowFunction HAMMING = new HammingWindow();
/**
* A constant indicating a Hann
* window should be used on sample buffers.
*/
public static final WindowFunction HANN = new HannWindow();
/**
* A constant indicating a Cosine
* window should be used on sample buffers.
*/
public static final WindowFunction COSINE = new CosineWindow();
/**
* A constant indicating a Triangular window should be used on sample buffers.
*/
public static final WindowFunction TRIANGULAR = new TriangularWindow();
/**
* A constant indicating a Bartlett window should be used on sample buffers.
*/
public static final WindowFunction BARTLETT = new BartlettWindow();
/**
* A constant indicating a Bartlett-Hann window should be used on sample buffers.
*/
public static final WindowFunction BARTLETTHANN = new BartlettHannWindow();
/**
* A constant indicating a Lanczos
* window should be used on sample buffers.
*/
public static final WindowFunction LANCZOS = new LanczosWindow();
/**
* A constant indicating a Blackman window with a default value should be used on sample buffers.
*/
public static final WindowFunction BLACKMAN = new BlackmanWindow();
/**
* A constant indicating a Gauss
* window with a default value should be used on sample buffers.
*/
public static final WindowFunction GAUSS = new GaussWindow();
/** The Constant LINAVG. */
protected static final int LINAVG = 1;
/** The Constant LOGAVG. */
protected static final int LOGAVG = 2;
/** The Constant NOAVG. */
protected static final int NOAVG = 3;
/** The Constant TWO_PI. */
protected static final double TWO_PI = (double) (2 * Math.PI);
/** The time size. */
protected int timeSize;
/** The sample rate. */
protected int sampleRate;
/** The band width. */
protected double bandWidth;
/** The window function. */
protected WindowFunction windowFunction;
/** The real. */
protected double[] real;
/** The imag. */
protected double[] imag;
/** The spectrum. */
protected double[] spectrum;
/** The averages. */
protected double[] averages;
/** The which average. */
protected int whichAverage;
/** The octaves. */
protected int octaves;
/** The avg per octave. */
protected int avgPerOctave;
/**
* Construct a FourierTransform that will analyze sample buffers that are
* ts samples long and contain samples with a sr
* sample rate.
*
* @param ts
* the length of the buffers that will be analyzed
* @param sr
* the sample rate of the samples that will be analyzed
*/
FourierTransform(int ts, double sr) {
timeSize = ts;
sampleRate = (int) sr;
bandWidth = (2f / timeSize) * ((double) sampleRate / 2f);
noAverages();
allocateArrays();
windowFunction = new RectangularWindow(); // a Rectangular window is
// analogous to using no window.
}
// allocating real, imag, and spectrum are the responsibility of derived
// classes
// because the size of the arrays will depend on the implementation being used
// this enforces that responsibility
/**
* Allocate arrays.
*/
protected abstract void allocateArrays();
/**
* Sets the complex.
*
* @param r the r
* @param i the i
*/
protected void setComplex(double[] r, double[] i) {
if (real.length != r.length && imag.length != i.length) {
throw new RuntimeException(
"FourierTransform.setComplex: the two arrays must be the same length as their member counterparts.");
} else {
System.arraycopy(r, 0, real, 0, r.length);
System.arraycopy(i, 0, imag, 0, i.length);
}
}
// fill the spectrum array with the amps of the data in real and imag
// used so that this class can handle creating the average array
// and also do spectrum shaping if necessary
/**
* Fill spectrum.
*/
protected void fillSpectrum() {
for (int i = 0; i < spectrum.length; i++) {
spectrum[i] = (double) Math.sqrt(real[i] * real[i] + imag[i] * imag[i]);
}
if (whichAverage == LINAVG) {
int avgWidth = (int) spectrum.length / averages.length;
for (int i = 0; i < averages.length; i++) {
double avg = 0;
int j;
for (j = 0; j < avgWidth; j++) {
int offset = j + i * avgWidth;
if (offset < spectrum.length) {
avg += spectrum[offset];
} else {
break;
}
}
avg /= j + 1;
averages[i] = avg;
}
} else if (whichAverage == LOGAVG) {
for (int i = 0; i < octaves; i++) {
double lowFreq, hiFreq, freqStep;
if (i == 0) {
lowFreq = 0;
} else {
lowFreq = (sampleRate / 2) / (double) Math.pow(2, octaves - i);
}
hiFreq = (sampleRate / 2) / (double) Math.pow(2, octaves - i - 1);
freqStep = (hiFreq - lowFreq) / avgPerOctave;
double f = lowFreq;
for (int j = 0; j < avgPerOctave; j++) {
int offset = j + i * avgPerOctave;
averages[offset] = calcAvg(f, f + freqStep);
f += freqStep;
}
}
}
}
/**
* Sets the object to not compute averages.
*
*/
public void noAverages() {
averages = new double[0];
whichAverage = NOAVG;
}
/**
* 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.
*
* @param numAvg
* how many averages to compute
*/
public void linAverages(int numAvg) {
if (numAvg > spectrum.length / 2) {
throw new RuntimeException("The number of averages for this transform can be at most " + spectrum.length / 2
+ ".");
} else {
averages = new double[numAvg];
}
whichAverage = LINAVG;
}
/**
* 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.
*
* @param minBandwidth
* the minimum bandwidth used for an octave
* @param bandsPerOctave
* how many bands to split each octave into
*/
public void logAverages(int minBandwidth, int bandsPerOctave) {
double nyq = (double) sampleRate / 2f;
octaves = 1;
while ((nyq /= 2) > minBandwidth) {
octaves++;
}
avgPerOctave = bandsPerOctave;
averages = new double[octaves * bandsPerOctave];
whichAverage = LOGAVG;
}
/**
* 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.
*
* @param windowFunction the window function
*/
public void window(WindowFunction windowFunction) {
if (windowFunction instanceof WindowFunction) {
this.windowFunction = windowFunction;
} else {
throw new RuntimeException("Invalid window type.");
}
}
/**
* Do window.
*
* @param samples the samples
*/
protected void doWindow(double[] samples) {
windowFunction.apply(samples);
}
/**
* Returns the length of the time domain signal expected by this transform.
*
* @return the length of the time domain signal expected by this transform
*/
public int timeSize() {
return timeSize;
}
/**
* 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.
*
* @return the size of the spectrum
*/
public int specSize() {
return spectrum.length;
}
/**
* Returns the amplitude of the requested frequency band.
*
* @param i
* the index of a frequency band
* @return the amplitude of the requested frequency band
*/
public double getBand(int i) {
if (i < 0) i = 0;
if (i > spectrum.length - 1) i = spectrum.length - 1;
return spectrum[i];
}
/**
* 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.
*
* @return the width of each frequency band in Hz.
*/
public double getBandWidth() {
return bandWidth;
}
/**
* Returns the bandwidth of the requested average band. Using this information
* and the return value of getAverageCenterFrequency you can determine the
* lower and upper frequency of any average band.
*
* @param averageIndex the average index
* @return the average band width
* @see #getAverageCenterFrequency(int)
* @related getAverageCenterFrequency ( )
*/
public double getAverageBandWidth(int averageIndex) {
if (whichAverage == LINAVG) {
// an average represents a certain number of bands in the spectrum
int avgWidth = (int) spectrum.length / averages.length;
return avgWidth * getBandWidth();
} else if (whichAverage == LOGAVG) {
// which "octave" is this index in?
int octave = averageIndex / avgPerOctave;
double lowFreq, hiFreq, freqStep;
// figure out the low frequency for this octave
if (octave == 0) {
lowFreq = 0;
} else {
lowFreq = (sampleRate / 2) / (double) Math.pow(2, octaves - octave);
}
// and the high frequency for this octave
hiFreq = (sampleRate / 2) / (double) Math.pow(2, octaves - octave - 1);
// each average band within the octave will be this big
freqStep = (hiFreq - lowFreq) / avgPerOctave;
return freqStep;
}
return 0;
}
/**
* Sets the amplitude of the ith frequency band to
* a. You can use this to shape the spectrum before using
* inverse().
*
* @param i
* the frequency band to modify
* @param a
* the new amplitude
*/
public abstract void setBand(int i, double a);
/**
* Scales the amplitude of the ith frequency band by
* s. You can use this to shape the spectrum before using
* inverse().
*
* @param i
* the frequency band to modify
* @param s
* the scaling factor
*/
public abstract void scaleBand(int i, double s);
/**
* Returns the index of the frequency band that contains the requested
* frequency.
*
* @param freq
* the frequency you want the index for (in Hz)
* @return the index of the frequency band that contains freq
*/
public int freqToIndex(double freq) {
// special case: freq is lower than the bandwidth of spectrum[0]
if (freq < getBandWidth() / 2) return 0;
// special case: freq is within the bandwidth of spectrum[spectrum.length -
// 1]
if (freq > sampleRate / 2 - getBandWidth() / 2) return spectrum.length - 1;
// all other cases
double fraction = freq / (double) sampleRate;
int i = (int) Math.round(timeSize * fraction);
return i;
}
/**
* Returns the middle frequency of the ith band.
*
* @param i the index of the band you want to middle frequency of
* @return the double
*/
public double indexToFreq(int i) {
double bw = getBandWidth();
// special case: the width of the first bin is half that of the others.
// so the center frequency is a quarter of the way.
if (i == 0) return bw * 0.25f;
// special case: the width of the last bin is half that of the others.
if (i == spectrum.length - 1) {
double lastBinBeginFreq = (sampleRate / 2) - (bw / 2);
double binHalfWidth = bw * 0.25f;
return lastBinBeginFreq + binHalfWidth;
}
// the center frequency of the ith band is simply i*bw
// because the first band is half the width of all others.
// treating it as if it wasn't offsets us to the middle
// of the band.
return i * bw;
}
/**
* Returns the center frequency of the ith average band.
*
* @param i which average band you want the center frequency of.
* @return the average center frequency
*/
public double getAverageCenterFrequency(int i) {
if (whichAverage == LINAVG) {
// an average represents a certain number of bands in the spectrum
int avgWidth = (int) spectrum.length / averages.length;
// the "center" bin of the average, this is fudgy.
int centerBinIndex = i * avgWidth + avgWidth / 2;
return indexToFreq(centerBinIndex);
} else if (whichAverage == LOGAVG) {
// which "octave" is this index in?
int octave = i / avgPerOctave;
// which band within that octave is this?
int offset = i % avgPerOctave;
double lowFreq, hiFreq, freqStep;
// figure out the low frequency for this octave
if (octave == 0) {
lowFreq = 0;
} else {
lowFreq = (sampleRate / 2) / (double) Math.pow(2, octaves - octave);
}
// and the high frequency for this octave
hiFreq = (sampleRate / 2) / (double) Math.pow(2, octaves - octave - 1);
// each average band within the octave will be this big
freqStep = (hiFreq - lowFreq) / avgPerOctave;
// figure out the low frequency of the band we care about
double f = lowFreq + offset * freqStep;
// the center of the band will be the low plus half the width
return f + freqStep / 2;
}
return 0;
}
/**
* Gets the amplitude of the requested frequency in the spectrum.
*
* @param freq
* the frequency in Hz
* @return the amplitude of the frequency in the spectrum
*/
public double getFreq(double freq) {
return getBand(freqToIndex(freq));
}
/**
* Sets the amplitude of the requested frequency in the spectrum to
* a.
*
* @param freq
* the frequency in Hz
* @param a
* the new amplitude
*/
public void setFreq(double freq, double a) {
setBand(freqToIndex(freq), a);
}
/**
* Scales the amplitude of the requested frequency by a.
*
* @param freq
* the frequency in Hz
* @param s
* the scaling factor
*/
public void scaleFreq(double freq, double s) {
scaleBand(freqToIndex(freq), s);
}
/**
* Returns the number of averages currently being calculated.
*
* @return the length of the averages array
*/
public int avgSize() {
return averages.length;
}
/**
* Gets the value of the ith average.
*
* @param i
* the average you want the value of
* @return the value of the requested average band
*/
public double getAvg(int i) {
double ret;
if (averages.length > 0)
ret = averages[i];
else
ret = 0;
return ret;
}
/**
* Calculate the average amplitude of the frequency band bounded by
* lowFreq and hiFreq, inclusive.
*
* @param lowFreq
* the lower bound of the band
* @param hiFreq
* the upper bound of the band
* @return the average of all spectrum values within the bounds
*/
public double calcAvg(double lowFreq, double hiFreq) {
int lowBound = freqToIndex(lowFreq);
int hiBound = freqToIndex(hiFreq);
double avg = 0;
for (int i = lowBound; i <= hiBound; i++) {
avg += spectrum[i];
}
avg /= (hiBound - lowBound + 1);
return avg;
}
/**
* Get the Real part of the Complex representation of the spectrum.
*
* @return the spectrum real
*/
public double[] getSpectrumReal() {
return real;
}
/**
* Get the Imaginary part of the Complex representation of the spectrum.
*
* @return the spectrum imaginary
*/
public double[] getSpectrumImaginary() {
return imag;
}
/**
* Performs a forward transform on buffer.
*
* @param buffer
* the buffer to analyze
*/
public abstract void forward(double[] buffer);
/**
* Performs a forward transform on values in buffer.
*
* @param buffer
* the buffer of samples
* @param startAt
* the index to start at in the buffer. there must be at least
* timeSize() samples between the starting index and the end of the
* buffer. If there aren't, an error will be issued and the operation
* will not be performed.
*
*/
public void forward(double[] buffer, int startAt) {
if (buffer.length - startAt < timeSize) {
throw new RuntimeException("FourierTransform.forward: not enough samples in the buffer between " + startAt
+ " and " + buffer.length + " to perform a transform.");
}
// copy the section of samples we want to analyze
double[] section = new double[timeSize];
System.arraycopy(buffer, startAt, section, 0, section.length);
forward(section);
}
/**
* Performs an inverse transform of the frequency spectrum and places the
* result in buffer.
*
* @param buffer
* the buffer to place the result of the inverse transform in
*/
public abstract void inverse(double[] buffer);
/**
* Performs an inverse transform of the frequency spectrum represented by
* freqReal and freqImag and places the result in buffer.
*
* @param freqReal
* the real part of the frequency spectrum
* @param freqImag
* the imaginary part the frequency spectrum
* @param buffer
* the buffer to place the inverse transform in
*/
public void inverse(double[] freqReal, double[] freqImag, double[] buffer) {
setComplex(freqReal, freqImag);
inverse(buffer);
}
/**
* Gets the log power spectrum.
*
* @return the log power spectrum
*/
public double[] getLogPowerSpectrum() {
double[] logSpectrum = new double[spectrum.length];
for (int i = 0; i < spectrum.length; i++) {
logSpectrum[i] = Math.log10(spectrum[i]);
}
return logSpectrum;
}
}