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added library for audio-processing

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Also add information on how to contact you by electronic and paper mail.
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The hypothetical commands `show w' and `show c' should show the appropriate
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For more information on this, and how to apply and follow the GNU GPL, see
<http://www.gnu.org/licenses/>.
The GNU General Public License does not permit incorporating your program
into proprietary programs. If your program is a subroutine library, you
may consider it more useful to permit linking proprietary applications with
the library. If this is what you want to do, use the GNU Lesser General
Public License instead of this License. But first, please read
<http://www.gnu.org/philosophy/why-not-lgpl.html>.

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<project xmlns="http://maven.apache.org/POM/4.0.0" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
xsi:schemaLocation="http://maven.apache.org/POM/4.0.0 http://maven.apache.org/xsd/maven-4.0.0.xsd">
<modelVersion>4.0.0</modelVersion>
<groupId>org.wyrez</groupId>
<artifactId>AudioProcessing</artifactId>
<version>1.0</version>
<packaging>jar</packaging>
<name>AudioProcessing</name>
<build>
<plugins>
<plugin>
<groupId>org.apache.maven.plugins</groupId>
<artifactId>maven-compiler-plugin</artifactId>
<version>2.3.2</version>
<configuration>
<source>1.7</source>
<target>1.7</target>
</configuration>
</plugin>
</plugins>
</build>
<dependencies>
<dependency>
<groupId>javazoom.jl</groupId>
<artifactId>jlayer</artifactId>
<version>1.0.1</version>
<scope>system</scope>
<systemPath>${basedir}/lib/jl1.0.1.jar</systemPath>
</dependency>
<dependency>
<groupId>javazoom.jl</groupId>
<artifactId>mp3spi</artifactId>
<version>1.9.5</version>
<scope>system</scope>
<systemPath>${basedir}/lib/mp3spi1.9.5.jar</systemPath>
</dependency>
<dependency>
<groupId>org.tritonus</groupId>
<artifactId>share</artifactId>
<version>0.3.6</version>
<scope>system</scope>
<systemPath>${basedir}/lib/tritonus_share-0.3.6.jar</systemPath>
</dependency>
<dependency>
<groupId>org.tritonus</groupId>
<artifactId>aos</artifactId>
<version>0.3.6</version>
<scope>system</scope>
<systemPath>${basedir}/lib/tritonus_aos-0.3.6.jar</systemPath>
</dependency>
</dependencies>
</project>

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/*
* Copyright (C) 2013 Darth Affe <http://wyrez.org> and contributors
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*
*
* This Code is based on the source provided with a tutorial by
* http://www.badlogicgames.com (as of 16.05.2013).
* The original source can be found here:
* http://code.google.com/p/audio-analysis/source/browse/trunk/src/com/badlogic/audio/io/AudioDevice.java
*/
package org.wyrez.audio;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.SourceDataLine;
/**
* Class that allows directly passing PCM float data to the sound card for
* playback.
*
* @author Darth Affe
*/
public class AudioDevice {
private final SourceDataLine out;
private byte[] buffer;
/**
* Constructor, initializes the audio system for 16-bit signed stereo output
* with a sampling rate of 44100 and a buffer size of 2048.
*
* @throws Exception in case the audio system could not be initialized
*/
public AudioDevice() throws Exception {
this(44100, 2);
}
/**
* Constructor, initializes the audio system for 16-bit signed output. Sets
* the buffer size to 1024*channels
*
* @param samplingRate The sampling rate
* @param channels The number of channels
* @throws Exception in case the audio system could not be initialized
*/
public AudioDevice(float samplingRate, int channels) throws Exception {
this(samplingRate, channels, 1024 * channels);
}
/**
* Constructor, initializes the audio system for 16-bit signed output.
*
* @param samplingRate The sampling rate
* @param channels The number of channels
* @param bufferSize The buffer size (have to be a even number)
* @throws Exception in case the audio system could not be initialized
*/
public AudioDevice(float samplingRate, int channels, int bufferSize) throws Exception {
AudioFormat format = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED,
samplingRate, 16, channels, channels * 2, samplingRate, false);
buffer = new byte[bufferSize * 2];
out = AudioSystem.getSourceDataLine(format);
out.open(format);
out.start();
}
/**
* Writes the given samples to the audio device. The samples have to be in
* the range [-1,1].
*
* @param samples The samples.
*/
public void writeSamples(float[] samples) {
fillBuffer(samples);
out.write(buffer, 0, buffer.length);
}
private void fillBuffer(float[] samples) {
for (int i = 0, j = 0; i < samples.length; i++, j += 2) {
short value = (short) (samples[i] * Short.MAX_VALUE);
buffer[j] = (byte) (value | 0xff);
buffer[j + 1] = (byte) (value >> 8);
}
}
}

418
AudioProcessing/src/main/java/org/wyrez/audio/AudioProcessor.java Normal file
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@ -0,0 +1,418 @@
/*
* Copyright (C) 2013 Darth Affe <http://wyrez.org> and contributors
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*
* This code is based on a tutorial by http://www.badlogicgames.com (as of 16.05.2013).
* The whole tutorial can be found here:
* http://www.badlogicgames.com/wordpress/?cat=18&paged=3
*/
package org.wyrez.audio;
import java.util.ArrayList;
import java.util.HashMap;
import java.util.List;
import org.wyrez.audio.analysis.FFT;
import org.wyrez.audio.decoder.Decoder;
import org.wyrez.audio.util.Band;
import org.wyrez.audio.util.SampleHelper;
/**
* Processor to create analysis-data of audio samples
*
* @author Darth Affe
*/
public class AudioProcessor {
private Band band;
private int bufferSize;
private final float[] samples;
private float[] spectralFlux;
private float[] threshold;
private float[] prunedSpectralFlux;
private float[] peaks;
private float peakAverage;
private final float samplingRate;
private float bpm;
/**
* Constructor, initializes an AudioProcessor for the given Decoder.
*
* @param decoder The decoder to process
*/
public AudioProcessor(Decoder decoder) {
this.samples = SampleHelper.readAllSamples(decoder);
this.samplingRate = decoder.getSamplingRate();
}
/**
* Constructor, initializes an AudioProcessor for the given Decoder and
* band.
*
* @param decoder The decoder to process
* @param band The band to process
*/
public AudioProcessor(Decoder decoder, Band band) {
this.samples = SampleHelper.readAllSamples(decoder);
this.samplingRate = decoder.getSamplingRate();
this.band = band;
}
/**
* Removes cached objects to save some memory.
*/
public void clean() {
spectralFlux = null;
threshold = null;
prunedSpectralFlux = null;
peaks = null;
bpm = 0f;
bufferSize = 0;
}
/**
* Calculates a full analysis of the samples data using default values:
* BufferSize = 1024 ThresholdWindowSize = 10 ThresholdMultiplier = 1.5
*/
public void calculate() {
calculate(1024, 10, 1.5f);
}
/**
* Calculates a full analysis of the samples data using a given threshold.
*
* @param thresholdWindowSize The window -size used to calculate the
* threshold
* @param thresholdMultiplier The multiplier used to calculate the threshold
*/
public void calculate(int thresholdWindowSize, float thresholdMultiplier) {
calculate(1024, thresholdWindowSize, thresholdMultiplier);
}
/**
* Calculates a full analysis of the samples data using a given threshold
* and a custom buffer size.
*
* @param bufferSize The size of the buffer
* @param thresholdWindowSize The window -size used to calculate the
* threshold
* @param thresholdMultiplier The multiplier used to calculate the threshold
*/
public void calculate(int bufferSize, int thresholdWindowSize, float thresholdMultiplier) {
calculate(bufferSize, thresholdWindowSize, thresholdMultiplier, -1f, -1);
}
/**
* Calculates a full analysis of the samples data using a given
* threshold-multiplier and cuts fast peaks.
*
* @param thresholdWindowSize The window -size used to calculate the
* threshold
* @param minPeakDiff The minimum difference in samples between to peaks
*/
public void calculate(int thresholdWindowSize, float thresholdMultiplier, int minPeakDiff) {
calculate(bufferSize, thresholdWindowSize, thresholdMultiplier, -1f, -1);
}
/**
* Calculates a full analysis of the samples data using a given
* threshold-multiplier and cuts low peaks.
*
* @param thresholdWindowSize The window -size used to calculate the
* threshold
* @param percentageToKeep The minimum amplitude in percent of the avarage
* amplitude
*/
public void calculate(int thresholdWindowSize, float thresholdMultiplier, float percentageToKeep) {
calculate(bufferSize, thresholdWindowSize, thresholdMultiplier, percentageToKeep, -1);
}
/**
* Calculates a full analysis of the samples data.
*
* @param bufferSize The size of the buffer
* @param thresholdWindowSize The window -size used to calculate the
* threshold
* @param thresholdMultiplier The multiplier used to calculate the threshold
* @param percentageToKeep The minimum amplitude in percent of the avarage
* amplitude
* @param minPeakDiff The minimum difference in samples between to peaks
*/
public void calculate(int bufferSize, int thresholdWindowSize, float thresholdMultiplier, float percentageToKeep, int minPeakDiff) {
if ((bufferSize & (bufferSize - 1)) != 0) {
throw new IllegalArgumentException("FFT: bufferSize must be a power of two.");
}
this.bufferSize = bufferSize;
calculateSpectralFlux();
calculateThreshold(thresholdWindowSize, thresholdMultiplier);
calculatePrunedSpectralFlux();
calculatePeaks();
if (percentageToKeep > 0f) {
cutLowPeaks(percentageToKeep);
}
if (minPeakDiff > 0) {
cutFastPeaks(minPeakDiff);
}
}
private void calculateSpectralFlux() {
FFT fft = new FFT(bufferSize, samplingRate);
fft.window(FFT.HAMMING);
float[] buffer;
float[] spectrum = new float[fft.getBandSize(band)];
float[] lastSpectrum = new float[spectrum.length];
List<Float> spectralFlux = new ArrayList<Float>();
for (int i = 0; i < samples.length; i += bufferSize) {
buffer = SampleHelper.splitSampleArray(samples, i, bufferSize);
fft.forward(buffer);
System.arraycopy(spectrum, 0, lastSpectrum, 0, spectrum.length);
System.arraycopy(fft.getSpectrum(band), 0, spectrum, 0, spectrum.length);
float flux = 0;
for (int j = 0; j < spectrum.length; j++) {
float value = (spectrum[j] - lastSpectrum[j]);
flux += value < 0 ? 0 : value;
}
spectralFlux.add(flux);
}
this.spectralFlux = SampleHelper.convertToFloatArray(spectralFlux);
}
private void calculateThreshold(int thresholdWindowSize, float multiplier) {
threshold = new float[spectralFlux.length];
for (int i = 0; i < spectralFlux.length; i++) {
int start = Math.max(0, i - thresholdWindowSize);
int end = Math.min(spectralFlux.length - 1, i + thresholdWindowSize);
float mean = 0;
for (int j = start; j <= end; j++) {
mean += spectralFlux[j];
}
mean /= (end - start);
threshold[i] = mean * multiplier;
}
}
private void calculatePrunedSpectralFlux() {
prunedSpectralFlux = new float[threshold.length];
for (int i = 0; i < threshold.length; i++) {
prunedSpectralFlux[i] = (threshold[i] <= spectralFlux[i]) ? spectralFlux[i] - threshold[i] : 0f;
}
}
private void calculatePeaks() {
peaks = new float[prunedSpectralFlux.length - 1];
for (int i = 0; i < prunedSpectralFlux.length - 1; i++) {
peaks[i] = (prunedSpectralFlux[i] > prunedSpectralFlux[i + 1]) ? prunedSpectralFlux[i] : 0f;
}
calculatePeakData();
}
private void calculatePeakData() {
calculatePeakAverage();
calculateBPM();
}
private void calculatePeakAverage() {
int counter = 0;
float sum = 0f;
for (int i = 0; i < peaks.length; i++) {
if (peaks[i] > 0f) {
sum += peaks[i];
counter++;
}
}
peakAverage = sum / counter;
}
private void calculateBPM() {
HashMap<Integer, Integer> peakDiff = new HashMap<Integer, Integer>();
int counter = 0;
boolean first = true;
for (int i = 0; i < peaks.length; i++) {
if (peaks[i] > 0f) {
if (!first) {
Integer value = peakDiff.get(counter);
if (value == null) {
value = 1;
} else {
value += 1;
}
peakDiff.put(counter, value);
} else {
first = false;
}
counter = 1;
} else {
counter++;
}
}
int most = -1;
int highestCount = -1;
for (Integer i : peakDiff.keySet()) {
if (peakDiff.get(i) > highestCount) {
highestCount = peakDiff.get(i);
most = i;
}
}
float timePerPeak = (1 / samplingRate) * bufferSize;
bpm = 1f / (timePerPeak * most) * 60f;
}
/**
* Cuts the last calculated peaks. Removes every peak with a amplitude lower
* than peakAverage*percentageToKeep.
*/
public void cutLowPeaks(float percentageToKeep) {
if (peaks == null) {
return;
}
float valueToKeep = peakAverage * percentageToKeep;
for (int i = 0; i < peaks.length; i++) {
if (peaks[i] < valueToKeep) {
peaks[i] = 0f;
}
}
calculatePeakData();
}
/**
* Cuts the last calculated peaks. Removes peaks witch a sample difference
* lower than minPeakDiff. (This algorithm does NOT analyse the whole data
* therfore the result may be a bit inaccurately)
*/
public void cutFastPeaks(int minPeakDiff) {
if (peaks == null) {
return;
}
int counter = 1;
boolean first = true;
for (int i = 0; i < peaks.length; i++) {
if (peaks[i] > 0f) {
if (!first) {
if (counter < minPeakDiff) {
peaks[i] = 0f;
} else {
counter = 1;
}
} else {
first = false;
}
} else {
counter++;
}
}
calculatePeakData();
}
/**
* Returns the samples which are analysed.
*
* @returns A float array containing the samples
*/
public float[] getSamples() {
return samples;
}
/**
* Returns the last calculated spectral flux.
*
* @returns A float array containing the spectral flux data
*/
public float[] getSpectralFlux() {
return spectralFlux;
}
/**
* Returns the last calculated threshold.
*
* @returns A float array containing the threshold data
*/
public float[] getThreshold() {
return threshold;
}
/**
* Returns the last calculated pruned spectral flux.
*
* @returns A float array containing the pruned spectral flux data
*/
public float[] getPrunedSpectralFlux() {
return prunedSpectralFlux;
}
/**
* Returns the last calculated peaks.
*
* @returns A float array containing the peaks
*/
public float[] getPeaks() {
return peaks;
}
/**
* Returns the last calculated avarage peak amplitude.
*
* @returns The avarage peak amplitude as float
*/
public float getPeakAverage() {
return peakAverage;
}
/**
* Returns the sampling rate of the used samples.
*
* @return The sampling rate as float
*/
public float getSamplingRate() {
return samplingRate;
}
/**
* Returns the last calculated bpm.
*
* @returns The bpm as float
*/
public float getBpm() {
return bpm;
}
/**
* Returns the buffer size used for the last calculation.
*
* @returns The buffer size as int
*/
public int getBufferSize() {
return bufferSize;
}
/**
* Returns the band used for the last calculation.
*
* @returns The band
*/
public Band getBand() {
return band;
}
/**
* Sets the band.
*
* @param band Band to use
*/
public void setBand(Band band) {
this.band = band;
}
}

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/*
* Copyright (c) 2007 - 2008 by Damien Di Fede <ddf@compartmental.net>
*
* 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.
*
* This Code is taken from the source provided with a tutorial by
* http://www.badlogicgames.com (as of 16.05.2013).
* The original source can be found here:
* http://code.google.com/p/audio-analysis/source/browse/trunk/src/com/badlogic/audio/analysis/DFT.java
*/
package org.wyrez.audio.analysis;
/**
* DFT stands for Discrete Fourier Transform and is the most widely used Fourier
* Transform. You will never want to use this class due to the fact that it is a
* brute force implementation of the DFT and as such is quite slow. Use an FFT
* instead. This exists primarily as a way to ensure that other implementations
* of the DFT are working properly. This implementation expects an even
* <code>timeSize</code> and will throw and IllegalArgumentException if this is
* not the case.
*
* @author Damien Di Fede
*
* @see FourierTransform
* @see FFT
* @see <a href="http://www.dspguide.com/ch8.htm">The Discrete Fourier
* Transform</a>
*
*/
public class DFT extends FourierTransform {
/**
* Constructs a DFT that expects audio buffers of length
* <code>timeSize</code> that have been recorded with a sample rate of
* <code>sampleRate</code>. Will throw an IllegalArgumentException if
* <code>timeSize</code> is not even.
*
* @param timeSize the length of the audio buffers you plan to analyze
* @param sampleRate the sample rate of the audio samples you plan to
* analyze
*/
public DFT(int timeSize, float sampleRate) {
super(timeSize, sampleRate);
if (timeSize % 2 != 0) {
throw new IllegalArgumentException("DFT: timeSize must be even.");
}
buildTrigTables();
}
protected void allocateArrays() {
spectrum = new float[timeSize / 2 + 1];
real = new float[timeSize / 2 + 1];
imag = new float[timeSize / 2 + 1];
}
/**
* Not currently implemented.
*/
public void scaleBand(int i, float s) {
}
/**
* Not currently implemented.
*/
public void setBand(int i, float a) {
}
public void forward(float[] samples) {
if (samples.length != timeSize) {
throw new IllegalArgumentException("DFT.forward: The length of the passed sample buffer must be equal to DFT.timeSize().");
}
doWindow(samples);
int N = samples.length;
for (int f = 0; f <= N / 2; f++) {
real[f] = 0.0f;
imag[f] = 0.0f;
for (int t = 0; t < N; t++) {
real[f] += samples[t] * cos(t * f);
imag[f] += samples[t] * -sin(t * f);
}
}
fillSpectrum();
}
public void inverse(float[] buffer) {
int N = buffer.length;
real[0] /= N;
imag[0] = -imag[0] / (N / 2);
real[N / 2] /= N;
imag[N / 2] = -imag[0] / (N / 2);
for (int i = 0; i < N / 2; i++) {
real[i] /= (N / 2);
imag[i] = -imag[i] / (N / 2);
}
for (int t = 0; t < N; t++) {
buffer[t] = 0.0f;
for (int f = 0; f < N / 2; f++) {
buffer[t] += real[f] * cos(t * f) + imag[f] * sin(t * f);
}
}
}
// lookup table data and functions
private float[] sinlookup;
private float[] coslookup;
private void buildTrigTables() {
int N = spectrum.length * timeSize;
sinlookup = new float[N];
coslookup = new float[N];
for (int i = 0; i < N; i++) {
sinlookup[i] = (float) Math.sin(i * TWO_PI / timeSize);
coslookup[i] = (float) Math.cos(i * TWO_PI / timeSize);
}
}
private float sin(int i) {
return sinlookup[i];
}
private float cos(int i) {
return coslookup[i];
}
}

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/*
* Copyright (c) 2007 - 2008 by Damien Di Fede <ddf@compartmental.net>
*
* 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.
*
* This Code is taken from the source provided with a tutorial by
* http://www.badlogicgames.com (as of 16.05.2013).
* The original source can be found here:
* http://code.google.com/p/audio-analysis/source/browse/trunk/src/com/badlogic/audio/analysis/FFT.java
*/
package org.wyrez.audio.analysis;
/**
* FFT stands for Fast Fourier Transform. It is an efficient way to calculate
* the Complex Discrete Fourier Transform. There is not much to say about this
* class other than the fact that when you want to analyze the spectrum of an
* audio buffer you will almost always use this class. One restriction of this
* class is that the audio buffers you want to analyze must have a length that
* is a power of two. If you try to construct an FFT with a
* <code>timeSize</code> that is not a power of two, an IllegalArgumentException
* will be thrown.
*
* @see FourierTransform
* @see <a href="http://www.dspguide.com/ch12.htm">The Fast Fourier
* Transform</a>
*
* @author Damien Di Fede
*
*/
public class FFT extends FourierTransform {
/**
* Constructs an FFT that will accept sample buffers that are
* <code>timeSize</code> long and have been recorded with a sample rate of
* <code>sampleRate</code>.
* <code>timeSize</code> <em>must</em> be a power of two. This will throw an
* exception if it is not.
*
* @param timeSize the length of the sample buffers you will be analyzing
* @param sampleRate the sample rate of the audio you will be analyzing
*/
public FFT(int timeSize, float sampleRate) {
super(timeSize, sampleRate);
if ((timeSize & (timeSize - 1)) != 0) {
throw new IllegalArgumentException(
"FFT: timeSize must be a power of two.");
}
buildReverseTable();
buildTrigTables();
}
protected void allocateArrays() {
spectrum = new float[timeSize / 2 + 1];
real = new float[timeSize];
imag = new float[timeSize];
}
public void scaleBand(int i, float s) {
if (s < 0) {
throw new IllegalArgumentException("Can't scale a frequency band by a negative value.");
}
if (spectrum[i] != 0) {
real[i] /= spectrum[i];
imag[i] /= spectrum[i];
spectrum[i] *= s;
real[i] *= spectrum[i];
imag[i] *= spectrum[i];
}
if (i != 0 && i != timeSize / 2) {
real[timeSize - i] = real[i];
imag[timeSize - i] = -imag[i];
}
}
public void setBand(int i, float a) {
if (a < 0) {
throw new IllegalArgumentException("Can't set a frequency band to a negative value.");
}
if (real[i] == 0 && imag[i] == 0) {
real[i] = a;
spectrum[i] = a;
} else {
real[i] /= spectrum[i];
imag[i] /= spectrum[i];
spectrum[i] = a;
real[i] *= spectrum[i];
imag[i] *= spectrum[i];
}
if (i != 0 && i != timeSize / 2) {
real[timeSize - i] = real[i];
imag[timeSize - i] = -imag[i];
}
}
// performs an in-place fft on the data in the real and imag arrays
// bit reversing is not necessary as the data will already be bit reversed
private void fft() {
for (int halfSize = 1; halfSize < real.length; halfSize *= 2) {
// float k = -(float)Math.PI/halfSize;
// phase shift step
// float phaseShiftStepR = (float)Math.cos(k);
// float phaseShiftStepI = (float)Math.sin(k);
// using lookup table
float phaseShiftStepR = cos(halfSize);
float phaseShiftStepI = sin(halfSize);
// current phase shift
float currentPhaseShiftR = 1.0f;
float currentPhaseShiftI = 0.0f;
for (int fftStep = 0; fftStep < halfSize; fftStep++) {
for (int i = fftStep; i < real.length; i += 2 * halfSize) {
int off = i + halfSize;
float tr = (currentPhaseShiftR * real[off]) - (currentPhaseShiftI * imag[off]);
float ti = (currentPhaseShiftR * imag[off]) + (currentPhaseShiftI * real[off]);
real[off] = real[i] - tr;
imag[off] = imag[i] - ti;
real[i] += tr;
imag[i] += ti;
}
float tmpR = currentPhaseShiftR;
currentPhaseShiftR = (tmpR * phaseShiftStepR) - (currentPhaseShiftI * phaseShiftStepI);
currentPhaseShiftI = (tmpR * phaseShiftStepI) + (currentPhaseShiftI * phaseShiftStepR);
}
}
}
public void forward(float[] buffer) {
if (buffer.length != timeSize) {
throw new IllegalArgumentException("FFT.forward: The length of the passed sample buffer must be equal to timeSize().");
}
doWindow(buffer);
// copy samples to real/imag in bit-reversed order
bitReverseSamples(buffer);
// perform the fft
fft();
// fill the spectrum buffer with amplitudes
fillSpectrum();
}
/**
* Performs a forward transform on the passed buffers.
*
* @param buffReal the real part of the time domain signal to transform
* @param buffImag the imaginary part of the time domain signal to transform
*/
public void forward(float[] buffReal, float[] buffImag) {
if (buffReal.length != timeSize || buffImag.length != timeSize) {
throw new IllegalArgumentException("FFT.forward: The length of the passed buffers must be equal to timeSize().");
}
setComplex(buffReal, buffImag);
bitReverseComplex();
fft();
fillSpectrum();
}
public void inverse(float[] buffer) {
if (buffer.length > real.length) {
throw new IllegalArgumentException("FFT.inverse: the passed array's length must equal FFT.timeSize().");
}
// conjugate
for (int i = 0; i < timeSize; i++) {
imag[i] *= -1;
}
bitReverseComplex();
fft();
// copy the result in real into buffer, scaling as we do
for (int i = 0; i < buffer.length; i++) {
buffer[i] = real[i] / real.length;
}
}
private int[] reverse;
private void buildReverseTable() {
int N = timeSize;
reverse = new int[N];
// set up the bit reversing table
reverse[0] = 0;
for (int limit = 1, bit = N / 2; limit < N; limit <<= 1, bit >>= 1) {
for (int i = 0; i < limit; i++) {
reverse[i + limit] = reverse[i] + bit;
}
}
}
// copies the values in the samples array into the real array
// in bit reversed order. the imag array is filled with zeros.
private void bitReverseSamples(float[] samples) {
for (int i = 0; i < samples.length; i++) {
real[i] = samples[reverse[i]];
imag[i] = 0.0f;
}
}
// bit reverse real[] and imag[]
private void bitReverseComplex() {
float[] revReal = new float[real.length];
float[] revImag = new float[imag.length];
for (int i = 0; i < real.length; i++) {
revReal[i] = real[reverse[i]];
revImag[i] = imag[reverse[i]];
}
real = revReal;
imag = revImag;
}
// lookup tables
private float[] sinlookup;
private float[] coslookup;
private float sin(int i) {
return sinlookup[i];
}
private float cos(int i) {
return coslookup[i];
}
private void buildTrigTables() {
int N = timeSize;
sinlookup = new float[N];
coslookup = new float[N];
for (int i = 0; i < N; i++) {
sinlookup[i] = (float) Math.sin(-(float) Math.PI / i);
coslookup[i] = (float) Math.cos(-(float) Math.PI / i);
}
}
}

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/*
* Copyright (c) 2007 - 2008 by Damien Di Fede <ddf@compartmental.net>
*
* 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.
*
* This Code is taken from the source provided with a tutorial by
* http://www.badlogicgames.com (as of 16.05.2013).
* The original source can be found here:
* http://code.google.com/p/audio-analysis/source/browse/trunk/src/com/badlogic/audio/analysis/FourierTransform.java
*/
//########## Added Code ##########
package org.wyrez.audio.analysis;
import java.util.ArrayList;
import org.wyrez.audio.util.Band;
import org.wyrez.audio.util.SampleHelper;
//#######End of Added Code #######
/**
* 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 <a
* href="http://en.wikipedia.org/wiki/Nyquist_frequency">Nyquist frequency</a>.
* In other words, given a signal of length
* <code>N</code>, there will be
* <code>N/2</code> frequency bands in the spectrum.
* <p>
* As an example, if you construct a FourierTransform with a
* <code>timeSize</code> of 1024 and and a
* <code>sampleRate</code> 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
* <code>5/1024 * 44100 = 0.0048828125 * 44100 = 215 Hz</code>. The width of
* that frequency band is equal to
* <code>2/1024</code>, 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
* <code>getFreq()</code> allows you to query the spectrum with a frequency in
* Hz and the function
* <code>getBandWidth()</code> will return the bandwidth in Hz of each frequency
* band in the spectrum.
* <p>
* <b>Usage</b>
* <p>
* 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
* <code>audio</code> is an AudioSource and
* <code>fft</code> is an FFT (one of the derived classes of FourierTransform).
*
* <pre>
* fft.forward(audio.left);
* for (int i = 0; i &lt; 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);
* }
* </pre>
*
* <b>Windowing</b>
* <p>
* Windowing is the process of shaping the audio samples before transforming
* them to the frequency domain. If you call the
* <code>window()</code> 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.
* <p>
* <b>Averages</b>
* <p>
* 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.
* <p>
* <code>linAverages()</code> 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.
* <p>
* <code>logAverages()</code> 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
* <code>sampleRate/4 - sampleRate/2</code>, 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.
* <p>
* <code>calcAvg()</code> 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.
* <p>
* If you don't want any averages calculated, then you can call
* <code>noAverages()</code>. This will not impact your ability to use
* <code>calcAvg()</code>, it will merely prevent the object from calculating an
* average array every time you use
* <code>forward()</code>.
* <p>
* <b>Inverse Transform</b>
* <p>
* 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
* <code>timeSize()</code> long. The
* <code>set</code> and
* <code>scale</code> 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 <a href="http://www.dspguide.com/ch8.htm">The Discrete Fourier
* Transform</a>
*/
public abstract class FourierTransform {
/**
* A constant indicating no window should be used on sample buffers.
*/
public static final int NONE = 0;
/**
* A constant indicating a Hamming window should be used on sample buffers.
*/
public static final int HAMMING = 1;
protected static final int LINAVG = 2;
protected static final int LOGAVG = 3;
protected static final int NOAVG = 4;
protected static final float TWO_PI = (float) (2 * Math.PI);
protected int timeSize;
protected int sampleRate;
protected float bandWidth;
protected int whichWindow;
protected float[] real;
protected float[] imag;
protected float[] spectrum;
protected float[] averages;
protected int whichAverage;
protected int octaves;
protected int avgPerOctave;
/**
* Construct a FourierTransform that will analyze sample buffers that are
* <code>ts</code> samples long and contain samples with a
* <code>sr</code> 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, float sr) {
timeSize = ts;
sampleRate = (int) sr;
bandWidth = (2f / timeSize) * ((float) sampleRate / 2f);
noAverages();
allocateArrays();
whichWindow = NONE;
}
// 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
protected abstract void allocateArrays();
protected void setComplex(float[] r, float[] i) {
if (real.length != r.length && imag.length != i.length) {
throw new IllegalArgumentException("This won't work");
} 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
protected void fillSpectrum() {
for (int i = 0; i < spectrum.length; i++) {
spectrum[i] = (float) 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++) {
float 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++) {
float lowFreq, hiFreq, freqStep;
if (i == 0) {
lowFreq = 0;
} else {
lowFreq = (sampleRate / 2) / (float) Math.pow(2, octaves - i);
}
hiFreq = (sampleRate / 2) / (float) Math.pow(2, octaves - i - 1);
freqStep = (hiFreq - lowFreq) / avgPerOctave;
float 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 float[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
* <code>specSize() / numAvg</code> bands wide.
*
* @param numAvg how many averages to compute
*/
public void linAverages(int numAvg) {
if (numAvg > spectrum.length / 2) {
throw new IllegalArgumentException("The number of averages for this transform can be at most " + spectrum.length / 2 + ".");
} else {
averages = new float[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,
* <code>logAverages(11, 1)</code> 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) {
float nyq = (float) sampleRate / 2f;
octaves = 1;
while ((nyq /= 2) > minBandwidth) {
octaves++;
}
avgPerOctave = bandsPerOctave;
averages = new float[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 which FourierTransform.HAMMING or FourierTransform.NONE
*/
public void window(int which) {
if (which < 0 || which > 1) {
throw new IllegalArgumentException("Invalid window type.");
} else {
whichWindow = which;
}
}
protected void doWindow(float[] samples) {
switch (whichWindow) {
case HAMMING:
hamming(samples);
break;
}
}
// windows the data in samples with a Hamming window
protected void hamming(float[] samples) {
for (int i = 0; i < samples.length; i++) {
samples[i] *= (0.54f - 0.46f * Math.cos(TWO_PI * i / (samples.length - 1)));
}
}
/**
* 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
* <code>timeSize()/2 + 1</code>, 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 float 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 float getBandWidth() {
return bandWidth;
}
/**
* Sets the amplitude of the
* <code>i<sup>th</sup></code> frequency band to
* <code>a</code>. You can use this to shape the spectrum before using
* <code>inverse()</code>.
*
* @param i the frequency band to modify
* @param a the new amplitude
*/
public abstract void setBand(int i, float a);
/**
* Scales the amplitude of the
* <code>i<sup>th</sup></code> frequency band by
* <code>s</code>. You can use this to shape the spectrum before using
* <code>inverse()</code>.
*
* @param i the frequency band to modify
* @param s the scaling factor
*/
public abstract void scaleBand(int i, float 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(float 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
float fraction = freq / (float) sampleRate;
int i = Math.round(timeSize * fraction);
return i;
}
/**
* Returns the middle frequency of the i<sup>th</sup> band.
*
* @param i the index of the band you want to middle frequency of
*/
public float indexToFreq(int i) {
float 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) {
float lastBinBeginFreq = (sampleRate / 2) - (bw / 2);
float 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 i<sup>th</sup> average band.
*
* @param i which average band you want the center frequency of.
*/
public float 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;
float lowFreq, hiFreq, freqStep;
// figure out the low frequency for this octave
if (octave == 0) {
lowFreq = 0;
} else {
lowFreq = (sampleRate / 2) / (float) Math.pow(2, octaves - octave);
}
// and the high frequency for this octave
hiFreq = (sampleRate / 2) / (float) 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
float 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 float getFreq(float freq) {
return getBand(freqToIndex(freq));
}
/**
* Sets the amplitude of the requested frequency in the spectrum to
* <code>a</code>.
*
* @param freq the frequency in Hz
* @param a the new amplitude
*/
public void setFreq(float freq, float a) {
setBand(freqToIndex(freq), a);
}
/**
* Scales the amplitude of the requested frequency by
* <code>a</code>.
*
* @param freq the frequency in Hz
* @param s the scaling factor
*/
public void scaleFreq(float freq, float 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
* <code>i<sup>th</sup></code> average.
*
* @param i the average you want the value of
* @return the value of the requested average band
*/
public float getAvg(int i) {
float ret;
if (averages.length > 0) {
ret = averages[i];
} else {
ret = 0;
}
return ret;
}
/**
* Calculate the average amplitude of the frequency band bounded by
* <code>lowFreq</code> and
* <code>hiFreq</code>, 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 float calcAvg(float lowFreq, float hiFreq) {
int lowBound = freqToIndex(lowFreq);
int hiBound = freqToIndex(hiFreq);
float avg = 0;
for (int i = lowBound; i <= hiBound; i++) {
avg += spectrum[i];
}
avg /= (hiBound - lowBound + 1);
return avg;
}
/**
* Performs a forward transform on
* <code>buffer</code>.
*
* @param buffer the buffer to analyze
*/
public abstract void forward(float[] buffer);
/**
* Performs a forward transform on values in
* <code>buffer</code>.
*
* @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(float[] buffer, int startAt) {
if (buffer.length - startAt < timeSize) {
throw new IllegalArgumentException("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
float[] section = new float[timeSize];
System.arraycopy(buffer, startAt, section, 0, section.length);
forward(section);
}
/**
* Performs an inverse transform of the frequency spectrum and places the
* result in
* <code>buffer</code>.
*
* @param buffer the buffer to place the result of the inverse transform in
*/
public abstract void inverse(float[] 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(float[] freqReal, float[] freqImag, float[] buffer) {
setComplex(freqReal, freqImag);
inverse(buffer);
}
/**
* @return the spectrum of the last FourierTransform.forward() call.
*/
public float[] getSpectrum() {
return spectrum;
}
//########## Added Code ##########
/**
* @return the spectrum of the last FourierTransform.forward() call for the
* specified range.
*/
public float[] getSpectrum(Band band) {
if (band == null) {
return spectrum;
}
ArrayList<Float> spec = new ArrayList<Float>();
for (float f = band.getLowFrequenzy(); f < band.getHighFrequenzy() + getBandWidth(); f += getBandWidth()) {
spec.add(getFreq(f));
}
return SampleHelper.convertToFloatArray(spec);
}
public int getBandSize(Band band) {
if (band == null) {
return spectrum.length;
}
int counter = 0;
for (float f = band.getLowFrequenzy(); f < band.getHighFrequenzy() + getBandWidth(); f += getBandWidth()) {
counter++;
}
return counter;
}
//#######End of Added Code #######
/**
* @return the real part of the last FourierTransform.forward() call.
*/
public float[] getRealPart() {
return real;
}
/**
* @return the imaginary part of the last FourierTransform.forward() call.
*/
public float[] getImaginaryPart() {
return imag;
}
}

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/*
* Copyright (C) 2013 Darth Affe <http://wyrez.org> and contributors
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*
*
* This Code is based on the source provided with a tutorial by
* http://www.badlogicgames.com (as of 16.05.2013).
* The original source can be found here:
* http://code.google.com/p/audio-analysis/source/browse/trunk/src/com/badlogic/audio/io/Decoder.java
*/
package org.wyrez.audio.decoder;
/**
* Interface for audio decoders that return successive amplitude frames.
*
* @author Darth Affe
*/
public interface Decoder {
/**
* Reads in samples.length stereo samples from the decoder. Returns the
* actual number read in. If this number is smaller than samples.length then
* the end of stream has been reached.
*
* @param samples The samples array to write the samples to
* @returns The number of samples actually read
*/
public int readSamplesStereo(float[] samples);
/**
* Reads in samples.length mono samples from the decoder. If the audio
* source isn't mono they channels are merged. Returns the actual number
* read in. If this number is smaller than samples.length then the end of
* stream has been reached. All Channels are merged by calculating the
* average.
*
* @param samples The samples array to write the samples to
* @returns The number of samples actually read
*/
public int readSamplesMergedMono(float[] samples);
/**
* Returns the number of channels in the audio source.
*
* @returns The channel count
*/
public int getChannelCount();
/**
* Returns the sampling rate in Hz of the audio source.
*
* @returns the sampling rate as float
*/
public float getSamplingRate();
}

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/*
* Copyright (C) 2013 Darth Affe <http://wyrez.org> and contributors
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
package org.wyrez.audio.decoder;
import java.io.FileInputStream;
import java.io.InputStream;
/**
* Factory used to create Audio-Decoder.
*
* @author Darth Affe
*/
public enum DecoderFactory {
WAVE, MP3;
/**
* Creates a decoder for the selected codec.
*
* @returns The new decoder
* @throws Exception in case the Decoder could not be created
*/
public Decoder create(InputStream is) throws Exception {
switch (this) {
case WAVE:
return new WaveDecoder(is);
case MP3:
return new MP3Decoder(is);
default: //impossible
return null;
}
}
/**
* Creates a decoder for the given file, if there isn't one an exception is
* thrown.
*
* @returns The new decoder
* @throws Exception in case the Decoder could not be created
* @throws IllegalArgumentException in case there is now decoder for the
* given file
*/
public static Decoder create(String file) throws Exception, IllegalArgumentException {
int i = file.lastIndexOf('.');
if (i == 0) {
throw new IllegalArgumentException("Can't detect a filetype");
}
String extension = file.substring(i + 1);
switch (extension.toUpperCase()) {
case "MP3":
return MP3.create(new FileInputStream(file));
case "WAV":
case "WAVE":
return WAVE.create(new FileInputStream(file));
default:
throw new IllegalArgumentException("Filetype \"" + extension + "\" is not supported.");
}
}
}

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/*
* Copyright (C) 2013 Darth Affe <http://wyrez.org> and contributors
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*
*
* This Code is based on the source provided with a tutorial by
* http://www.badlogicgames.com (as of 16.05.2013).
* The original source can be found here:
* http://code.google.com/p/audio-analysis/source/browse/trunk/src/com/badlogic/audio/io/MP3Decoder.java
*/
package org.wyrez.audio.decoder;
import java.io.BufferedInputStream;
import java.io.IOException;
import java.io.InputStream;
import java.io.PushbackInputStream;
import java.util.HashMap;
import javax.sound.sampled.AudioFileFormat;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioInputStream;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.UnsupportedAudioFileException;
import javazoom.jl.decoder.Bitstream;
import javazoom.jl.decoder.Header;
import javazoom.spi.mpeg.sampled.file.MpegAudioFileFormat;
import javazoom.spi.mpeg.sampled.file.MpegAudioFormat;
import javazoom.spi.mpeg.sampled.file.MpegEncoding;
import javazoom.spi.mpeg.sampled.file.MpegFileFormatType;
import org.tritonus.share.TDebug;
import org.tritonus.share.sampled.FloatSampleBuffer;
import org.tritonus.share.sampled.file.TAudioFileReader;
/**
* A decoder for MP3 files with a maximum of 2 channels.
*
* @author Darth Affe
*/
public class MP3Decoder implements Decoder {
private AudioInputStream in;
private FloatSampleBuffer buffer;
private byte[] bytes;
private int channels;
private float samplingRate;
/**
* Constructor, sets the input stream to read the MP3 file from. Audio files
* with more than 2 channels aren't supported!
*
* @param stream The input stream.
* @throws Exception in case the input stream couldn't be read properly
*/
public MP3Decoder(InputStream stream) throws Exception {
InputStream in = new BufferedInputStream(stream, 1024 * 1024);
this.in = new MP3AudioFileReader().getAudioInputStream(in);
AudioFormat baseFormat = this.in.getFormat();
this.channels = baseFormat.getChannels();
if (channels > 2) {
throw new IllegalArgumentException("more than 2 channels aren't supported");
}
this.samplingRate = baseFormat.getSampleRate();
AudioFormat format = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED,
baseFormat.getSampleRate(), 16,
baseFormat.getChannels(),
baseFormat.getChannels() * 2,
baseFormat.getSampleRate(), false);
this.in = AudioSystem.getAudioInputStream(format, this.in);
}
/**
* Tries to read in samples.length stereo samples. Returns the number of
* samples actually read. Guarantees that samples.length samples are read in
* if there was enough data in the stream.
*
* @param samples The samples array to write the samples to
* @return The number of samples actually read
*/
@Override
public int readSamplesStereo(float[] samples) {
if (buffer == null || buffer.getSampleCount() < samples.length / 2) {
buffer = new FloatSampleBuffer(in.getFormat().getChannels(), samples.length / 2, in.getFormat().getSampleRate());
bytes = new byte[buffer.getByteArrayBufferSize(in.getFormat())];
}
int read = 0;
int readBytes = 0;
try {
readBytes = in.read(bytes, read, bytes.length - read);
} catch (IOException e) {
return 0;
}
if (readBytes == -1) {
return 0;
}
read += readBytes;
while (readBytes != -1 && read != bytes.length) {
try {
readBytes = in.read(bytes, read, bytes.length - read);
} catch (IOException e) {
return 0;
}
read += readBytes;
}
int frameCount = bytes.length / in.getFormat().getFrameSize();
buffer.setSamplesFromBytes(bytes, 0, in.getFormat(), 0, frameCount);
for (int i = 0, j = 0; i < buffer.getSampleCount(); i++, j += 2) {
samples[j] = buffer.getChannel(0)[i];
samples[j + 1] = buffer.getChannel(buffer.getChannelCount() == 2 ? 1 : 0)[i];
}
return buffer.getSampleCount() * 2;
}
/**
* Tries to read in samples.length samples, merging stereo to a mono channel
* by averaging. Returns the number of samples actually read. Guarantees
* that samples.length samples are read in if there was enough data in the
* stream.
*
* @param samples The samples array to write the samples to
* @return The number of samples actually read
*/
@Override
public int readSamplesMergedMono(float[] samples) {
if (buffer == null || buffer.getSampleCount() < samples.length) {
buffer = new FloatSampleBuffer(in.getFormat().getChannels(), samples.length, in.getFormat().getSampleRate());
bytes = new byte[buffer.getByteArrayBufferSize(in.getFormat())];
}
int read = 0;
int readBytes = 0;
try {
readBytes = in.read(bytes, read, bytes.length - read);
} catch (IOException e) {
return 0;
}
if (readBytes == -1) {
return 0;
}
read += readBytes;
while (readBytes != -1 && read != bytes.length) {
try {
readBytes = in.read(bytes, read, bytes.length - read);
} catch (IOException e) {
return 0;
}
read += readBytes;
}
int frameCount = bytes.length / in.getFormat().getFrameSize();
buffer.setSamplesFromBytes(bytes, 0, in.getFormat(), 0, frameCount);
for (int i = 0; i < buffer.getSampleCount(); i++) {
if (buffer.getChannelCount() == 2) {
samples[i] = (buffer.getChannel(0)[i] + buffer.getChannel(1)[i]) / 2;
} else {
samples[i] = buffer.getChannel(0)[i];
}
}
return buffer.getSampleCount();
}
/**
* Returns the number of channels in the audio source. (This value can't be
* greater than 2)
*
* @returns The channel count
*/
@Override
public int getChannelCount() {
return channels;
}
/**
* Returns the sampling rate in Hz of the audio source.
*
* @returns the sampling rate as float
*/
@Override
public float getSamplingRate() {
return samplingRate;
}
class MP3AudioFileReader extends TAudioFileReader {
public static final int INITAL_READ_LENGTH = 128000;
private static final int MARK_LIMIT = INITAL_READ_LENGTH + 1;
private final AudioFormat.Encoding[][] sm_aEncodings = {
{MpegEncoding.MPEG2L1, MpegEncoding.MPEG2L2, MpegEncoding.MPEG2L3},
{MpegEncoding.MPEG1L1, MpegEncoding.MPEG1L2, MpegEncoding.MPEG1L3},
{MpegEncoding.MPEG2DOT5L1, MpegEncoding.MPEG2DOT5L2,
MpegEncoding.MPEG2DOT5L3},};
protected MP3AudioFileReader() {
super(MARK_LIMIT, true);
}
@Override
protected AudioFileFormat getAudioFileFormat(InputStream inputStream, long mediaLength)
throws UnsupportedAudioFileException, IOException {
HashMap aff_properties = new HashMap();
HashMap af_properties = new HashMap();
int mLength = (int) mediaLength;
int size = inputStream.available();
PushbackInputStream pis = new PushbackInputStream(inputStream, MARK_LIMIT);
byte head[] = new byte[22];
pis.read(head);
// Check for WAV, AU, and AIFF, Ogg Vorbis, Flac, MAC file formats.
// Next check for Shoutcast (supported) and OGG (unsupported) streams.
if ((head[0] == 'R') && (head[1] == 'I') && (head[2] == 'F')
&& (head[3] == 'F') && (head[8] == 'W') && (head[9] == 'A')
&& (head[10] == 'V') && (head[11] == 'E')) {
int isPCM = ((head[21] << 8) & 0x0000FF00) | ((head[20]) & 0x00000FF);
throw new UnsupportedAudioFileException("WAV PCM stream found");
} else if ((head[0] == '.') && (head[1] == 's') && (head[2] == 'n')
&& (head[3] == 'd')) {
throw new UnsupportedAudioFileException("AU stream found");
} else if ((head[0] == 'F') && (head[1] == 'O') && (head[2] == 'R')
&& (head[3] == 'M') && (head[8] == 'A') && (head[9] == 'I')
&& (head[10] == 'F') && (head[11] == 'F')) {
throw new UnsupportedAudioFileException("AIFF stream found");
} else if (((head[0] == 'M') | (head[0] == 'm'))
&& ((head[1] == 'A') | (head[1] == 'a'))
&& ((head[2] == 'C') | (head[2] == 'c'))) {
throw new UnsupportedAudioFileException("APE stream found");
} else if (((head[0] == 'F') | (head[0] == 'f'))
&& ((head[1] == 'L') | (head[1] == 'l'))
&& ((head[2] == 'A') | (head[2] == 'a'))
&& ((head[3] == 'C') | (head[3] == 'c'))) {
throw new UnsupportedAudioFileException("FLAC stream found");
} // Shoutcast stream ?
else if (((head[0] == 'I') | (head[0] == 'i'))
&& ((head[1] == 'C') | (head[1] == 'c'))
&& ((head[2] == 'Y') | (head[2] == 'y'))) {
pis.unread(head);
// Load shoutcast meta data.
} // Ogg stream ?
else if (((head[0] == 'O') | (head[0] == 'o'))
&& ((head[1] == 'G') | (head[1] == 'g'))
&& ((head[2] == 'G') | (head[2] == 'g'))) {
throw new UnsupportedAudioFileException("Ogg stream found");
} // No, so pushback.
else {
pis.unread(head);
}
// MPEG header info.
int nVersion = AudioSystem.NOT_SPECIFIED;
int nLayer = AudioSystem.NOT_SPECIFIED;
// int nSFIndex = AudioSystem.NOT_SPECIFIED;
int nMode = AudioSystem.NOT_SPECIFIED;
int FrameSize = AudioSystem.NOT_SPECIFIED;
// int nFrameSize = AudioSystem.NOT_SPECIFIED;
int nFrequency = AudioSystem.NOT_SPECIFIED;
int nTotalFrames = AudioSystem.NOT_SPECIFIED;
float FrameRate = AudioSystem.NOT_SPECIFIED;
int BitRate = AudioSystem.NOT_SPECIFIED;
int nChannels = AudioSystem.NOT_SPECIFIED;
int nHeader = AudioSystem.NOT_SPECIFIED;
int nTotalMS = AudioSystem.NOT_SPECIFIED;
boolean nVBR = false;
AudioFormat.Encoding encoding = null;
try {
Bitstream m_bitstream = new Bitstream(pis);
aff_properties.put("mp3.header.pos",
new Integer(m_bitstream.header_pos()));
Header m_header = m_bitstream.readFrame();
// nVersion = 0 => MPEG2-LSF (Including MPEG2.5), nVersion = 1 => MPEG1
nVersion = m_header.version();
if (nVersion == 2) {
aff_properties.put("mp3.version.mpeg", Float.toString(2.5f));
} else {
aff_properties.put("mp3.version.mpeg",
Integer.toString(2 - nVersion));
}
// nLayer = 1,2,3
nLayer = m_header.layer();
aff_properties.put("mp3.version.layer", Integer.toString(nLayer));
// nSFIndex = m_header.sample_frequency();
nMode = m_header.mode();
aff_properties.put("mp3.mode", new Integer(nMode));
nChannels = nMode == 3 ? 1 : 2;
aff_properties.put("mp3.channels", new Integer(nChannels));
nVBR = m_header.vbr();
af_properties.put("vbr", new Boolean(nVBR));
aff_properties.put("mp3.vbr", new Boolean(nVBR));
aff_properties.put("mp3.vbr.scale", new Integer(m_header.vbr_scale()));
FrameSize = m_header.calculate_framesize();
aff_properties.put("mp3.framesize.bytes", new Integer(FrameSize));
if (FrameSize < 0) {
throw new UnsupportedAudioFileException("Invalid FrameSize : " + FrameSize);
}
nFrequency = m_header.frequency();
aff_properties.put("mp3.frequency.hz", new Integer(nFrequency));
FrameRate = (float) ((1.0 / (m_header.ms_per_frame())) * 1000.0);
aff_properties.put("mp3.framerate.fps", new Float(FrameRate));
if (FrameRate < 0) {
throw new UnsupportedAudioFileException("Invalid FrameRate : " + FrameRate);
}
if (mLength != AudioSystem.NOT_SPECIFIED) {
aff_properties.put("mp3.length.bytes", new Integer(mLength));
nTotalFrames = m_header.max_number_of_frames(mLength);
aff_properties.put("mp3.length.frames", new Integer(nTotalFrames));
}
BitRate = m_header.bitrate();
af_properties.put("bitrate", new Integer(BitRate));
aff_properties.put("mp3.bitrate.nominal.bps", new Integer(BitRate));
nHeader = m_header.getSyncHeader();
encoding = sm_aEncodings[nVersion][nLayer - 1];
aff_properties.put("mp3.version.encoding", encoding.toString());
if (mLength != AudioSystem.NOT_SPECIFIED) {
nTotalMS = Math.round(m_header.total_ms(mLength));
aff_properties.put("duration", new Long((long) nTotalMS * 1000L));
}
aff_properties.put("mp3.copyright", new Boolean(m_header.copyright()));
aff_properties.put("mp3.original", new Boolean(m_header.original()));
aff_properties.put("mp3.crc", new Boolean(m_header.checksums()));
aff_properties.put("mp3.padding", new Boolean(m_header.padding()));
InputStream id3v2 = m_bitstream.getRawID3v2();
if (id3v2 != null) {
aff_properties.put("mp3.id3tag.v2", id3v2);
}
if (TDebug.TraceAudioFileReader) {
TDebug.out(m_header.toString());
}
} catch (Exception e) {
throw new UnsupportedAudioFileException("not a MPEG stream:"
+ e.getMessage());
}
// Deeper checks ?
int cVersion = (nHeader >> 19) & 0x3;
if (cVersion == 1) {
throw new UnsupportedAudioFileException(
"not a MPEG stream: wrong version");
}
int cSFIndex = (nHeader >> 10) & 0x3;
if (cSFIndex == 3) {
throw new UnsupportedAudioFileException(
"not a MPEG stream: wrong sampling rate");
}
AudioFormat format = new MpegAudioFormat(encoding, (float) nFrequency,
AudioSystem.NOT_SPECIFIED, nChannels, -1, FrameRate, true, af_properties);
return new MpegAudioFileFormat(MpegFileFormatType.MP3, format,
nTotalFrames, mLength, aff_properties);
}
}
}

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/*
* Copyright (C) 2013 Darth Affe <http://wyrez.org> and contributors
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*
*
* This Code is based on the source provided with a tutorial by
* http://www.badlogicgames.com (as of 16.05.2013).
* The original source can be found here:
* http://code.google.com/p/audio-analysis/source/browse/trunk/src/com/badlogic/audio/io/WaveDecoder.java
*/
package org.wyrez.audio.decoder;
import java.io.BufferedInputStream;
import java.io.InputStream;
import org.wyrez.audio.util.EndianDataInputStream;
/**
* A simple converter that can read in the PCM data from a Wav file, converting
* the data to signed 32-bit floats in the range [-1,1]. This only supports
* 16-bit Wav files with a maximum of 2 channels.
*
* @author Darth Affe
*/
public class WaveDecoder implements Decoder {
private final float MAX_VALUE = 1.0f / Short.MAX_VALUE;//inverse max short value as float
private final EndianDataInputStream in;
private final int channels;
private final float samplingRate;
/**
* Constructor, sets the input stream to read the Wav file from. Audio files
* with more than 2 channels aren't supported!
*
* @param stream The input stream.
* @throws Exception in case the input stream couldn't be read properly
*/
public WaveDecoder(InputStream stream) throws Exception {
if (stream == null) {
throw new IllegalArgumentException("Input stream must not be null");
}
in = new EndianDataInputStream(new BufferedInputStream(stream, 1024 * 1024));
if (!in.read4ByteString().equals("RIFF")) {
throw new IllegalArgumentException("not a wav");
}
in.readIntLittleEndian();
if (!in.read4ByteString().equals("WAVE")) {
throw new IllegalArgumentException("expected WAVE tag");
}
if (!in.read4ByteString().equals("fmt ")) {
throw new IllegalArgumentException("expected fmt tag");
}
if (in.readIntLittleEndian() != 16) {
throw new IllegalArgumentException("expected wave chunk size to be 16");
}
if (in.readShortLittleEndian() != 1) {
throw new IllegalArgumentException("expected format to be 1");
}
channels = in.readShortLittleEndian();
if (channels > 2) {
throw new IllegalArgumentException("more than 2 channels aren't supported");
}
samplingRate = in.readIntLittleEndian();
in.readIntLittleEndian();
in.readShortLittleEndian();
int fmt = in.readShortLittleEndian();
if (fmt != 16) {
throw new IllegalArgumentException("Only 16-bit signed format supported");
}
if (!in.read4ByteString().equals("data")) {
throw new RuntimeException("expected data tag");
}
in.readIntLittleEndian();
}
/**
* Tries to read in samples.length stereo samples, converting non float
* formats to float 32-bit. Returns the number of samples actually read.
* Guarantees that samples.length samples are read in if there was enough
* data in the stream.
*
* @param samples The samples array to write the samples to
* @return The number of samples actually read
*/
@Override
public int readSamplesStereo(float[] samples) {
switch (channels) {
case 1:
return readMonoAsStereo(samples);
case 2:
return readRealStereo(samples);
default:
return 0;
}
}
private int readRealStereo(float[] samples) {
int readSamples = 0;
for (int i = 0; i < samples.length - 1; i += 2) {
try {
samples[i] = in.readShortLittleEndian() * MAX_VALUE;
samples[i + 1] = in.readShortLittleEndian() * MAX_VALUE;
readSamples += 2;
} catch (Exception ex) {
break;
}
}
return readSamples;
}
private int readMonoAsStereo(float[] samples) {
int readSamples = 0;
for (int i = 0; i < samples.length - 1; i += 2) {
try {
float sample = in.readShortLittleEndian() * MAX_VALUE;
samples[i] = sample;
samples[i + 1] = sample;
readSamples += 2;
} catch (Exception ex) {
break;
}
}
return readSamples;
}
/**
* Tries to read in samples.length samples, merging stereo to a mono channel
* by averaging and converting non float formats to float 32-bit. Returns
* the number of samples actually read. Guarantees that samples.length
* samples are read in if there was enough data in the stream.
*
* @param samples The samples array to write the samples to
* @return The number of samples actually read
*/
@Override
public int readSamplesMergedMono(float[] samples) {
int readSamples = 0;
for (int i = 0; i < samples.length; i++) {
float sample = 0;
try {
for (int j = 0; j < channels; j++) {
int shortValue = in.readShortLittleEndian();
sample += (shortValue * MAX_VALUE);
}
sample /= channels;
samples[i] = sample;
readSamples++;
} catch (Exception ex) {
break;
}
}
return readSamples;
}
/**
* Returns the number of channels in the audio source. (This value can't be
* greater than 2)
*
* @returns The channel count
*/
@Override
public int getChannelCount() {
return channels;
}
/**
* Returns the sampling rate in Hz of the audio source.
*
* @returns the sampling rate as floatF
*/
@Override
public float getSamplingRate() {
return samplingRate;
}
}

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/*
* Copyright (C) 2013 Darth Affe <http://wyrez.org> and contributors
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
package org.wyrez.audio.util;
/**
* Defines a band between of two frequencies.
*
* @author Darth Affe
*/
public class Band {
private float lowFrequenzy;
private float highFrequenzy;
public Band(float lowFrequenzy, float highFrequenzy) {
if (lowFrequenzy <= highFrequenzy) {
this.lowFrequenzy = lowFrequenzy;
this.highFrequenzy = highFrequenzy;
} else {
this.lowFrequenzy = highFrequenzy;
this.highFrequenzy = lowFrequenzy;
}
}
/**
* Returns the lower frequency of the band.
*
* @return The frequency as float
*/
public float getLowFrequenzy() {
return lowFrequenzy;
}
/**
* Returns the higher frequency of the band.
*
* @return The frequency as float
*/
public float getHighFrequenzy() {
return highFrequenzy;
}
}

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/*
* This Code is taken from the source provided with a tutorial by
* http://www.badlogicgames.com (as of 16.05.2013).
* The original source can be found here:
* http://code.google.com/p/audio-analysis/source/browse/trunk/src/com/badlogic/audio/io/EndianDataInputStream.java
*/
package org.wyrez.audio.util;
import java.io.DataInputStream;
import java.io.InputStream;
public class EndianDataInputStream extends DataInputStream {
public EndianDataInputStream(InputStream in) {
super(in);
}
public String read4ByteString() throws Exception {
byte[] bytes = new byte[4];
readFully(bytes);
return new String(bytes, "US-ASCII");
}
public short readShortLittleEndian() throws Exception {
int result = readUnsignedByte();
result |= readUnsignedByte() << 8;
return (short) result;
}
public int readIntLittleEndian() throws Exception {
int result = readUnsignedByte();
result |= readUnsignedByte() << 8;
result |= readUnsignedByte() << 16;
result |= readUnsignedByte() << 24;
return result;
}
public int readInt24BitLittleEndian() throws Exception {
int result = readUnsignedByte();
result |= readUnsignedByte() << 8;
result |= readUnsignedByte() << 16;
if ((result & (1 << 23)) == 8388608) {
result |= 0xff000000;
}
return result;
}
public int readInt24Bit() throws Exception {
int result = readUnsignedByte() << 16;
result |= readUnsignedByte() << 8;
result |= readUnsignedByte();
return result;
}
}

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/*
* Copyright (C) 2013 Darth Affe <http://wyrez.org> and contributors
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, either version 3 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
package org.wyrez.audio.util;
import java.util.ArrayList;
import java.util.List;
import org.wyrez.audio.decoder.Decoder;
/**
* Helper class for sample-processing.
*
* @author Darth Affe
*/
public class SampleHelper {
/**
* Converts a list of floats to an array of floats.
*
* @param samples The List to convert
* @returns The new float array
*/
public static float[] convertToFloatArray(List<Float> samples) {
float[] array = new float[samples.size()];
for (int i = 0; i < array.length; i++) {
array[i] = samples.get(i);
}
return array;
}
/**
* Creates an float array with the given length containing only the one
* given value.
*
* @param f The value to fill the array with
* @param length The length of the array
* @returns The new array
*/
public static float[] createFloatArray(float f, int length) {
float[] result = new float[length];
for (int i = 0; i < length; i++) {
result[i] = f;
}
return result;
}
/**
* Splits an array of floats.
*
* @param samples The source array to split
* @param offset The offset to split from
* @param length The length of the new array
*
* @retrn The new array
*
* @throws IllegalArgumentException Thrown if there aren't enough samples to
* proceed
*/
public static float[] splitSampleArray(float[] samples, int offset, int length)
throws IllegalArgumentException {
if (offset + length > samples.length) {
throw new IllegalArgumentException("There aren't enough samples in order to do that");
}
float[] result = new float[length];
for (int i = 0, j = offset; i < length; i++, j++) {
result[i] = samples[j];
}
return result;
}
/**
* Reads all samples mono merged from the given decoder.
*
* @param deocder The decoder to read from
* @returns The float array containing the samples
*/
public static float[] readAllSamples(Decoder decoder) {
List<Float> samples = new ArrayList<Float>();
float[] buffer = new float[1024];
while (decoder.readSamplesMergedMono(buffer) > 0) {
for (int i = 0; i < buffer.length; i++) {
samples.add(buffer[i]);
}
}
return convertToFloatArray(samples);
}
}

481
AudioProcessing/tritonus.LICENSE Normal file
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GNU LIBRARY GENERAL PUBLIC LICENSE
Version 2, June 1991
Copyright (C) 1991 Free Software Foundation, Inc.
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.
[This is the first released version of the library GPL. It is
numbered 2 because it goes with version 2 of the ordinary GPL.]
Preamble
The licenses for most software are designed to take away your
freedom to share and change it. By contrast, the GNU General Public
Licenses are intended to guarantee your freedom to share and change
free software--to make sure the software is free for all its users.
This license, the Library General Public License, applies to some
specially designated Free Software Foundation software, and to any
other libraries whose authors decide to use it. You can use it for
your libraries, too.
When we speak of free software, we are referring to freedom, not
price. Our General Public Licenses are designed to make sure that you
have the freedom to distribute copies of free software (and charge for
this service if you wish), that you receive source code or can get it
if you want it, that you can change the software or use pieces of it
in new free programs; and that you know you can do these things.
To protect your rights, we need to make restrictions that forbid
anyone to deny you these rights or to ask you to surrender the rights.
These restrictions translate to certain responsibilities for you if
you distribute copies of the library, or if you modify it.
For example, if you distribute copies of the library, whether gratis
or for a fee, you must give the recipients all the rights that we gave
you. You must make sure that they, too, receive or can get the source
code. If you link a program with the library, you must provide
complete object files to the recipients so that they can relink them
with the library, after making changes to the library and recompiling
it. And you must show them these terms so they know their rights.
Our method of protecting your rights has two steps: (1) copyright
the library, and (2) offer you this license which gives you legal
permission to copy, distribute and/or modify the library.
Also, for each distributor's protection, we want to make certain
that everyone understands that there is no warranty for this free
library. If the library is modified by someone else and passed on, we
want its recipients to know that what they have is not the original
version, so that any problems introduced by others will not reflect on
the original authors' reputations.
Finally, any free program is threatened constantly by software
patents. We wish to avoid the danger that companies distributing free
software will individually obtain patent licenses, thus in effect
transforming the program into proprietary software. To prevent this,
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That's all there is to it!