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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 +. diff --git a/AudioProcessing/lib/jl1.0.1.jar b/AudioProcessing/lib/jl1.0.1.jar new file mode 100644 index 0000000..bd5fb8b Binary files /dev/null and b/AudioProcessing/lib/jl1.0.1.jar differ diff --git a/AudioProcessing/lib/mp3spi1.9.5.jar b/AudioProcessing/lib/mp3spi1.9.5.jar new file mode 100644 index 0000000..0c74dae Binary files /dev/null and b/AudioProcessing/lib/mp3spi1.9.5.jar differ diff --git a/AudioProcessing/lib/tritonus_aos-0.3.6.jar b/AudioProcessing/lib/tritonus_aos-0.3.6.jar new file mode 100644 index 0000000..2287391 Binary files /dev/null and b/AudioProcessing/lib/tritonus_aos-0.3.6.jar differ diff --git a/AudioProcessing/lib/tritonus_share-0.3.6.jar b/AudioProcessing/lib/tritonus_share-0.3.6.jar new file mode 100644 index 0000000..b9b7ac0 Binary files /dev/null and b/AudioProcessing/lib/tritonus_share-0.3.6.jar differ diff --git a/AudioProcessing/pom.xml b/AudioProcessing/pom.xml new file mode 100644 index 0000000..a04339e --- /dev/null +++ b/AudioProcessing/pom.xml @@ -0,0 +1,52 @@ + + 4.0.0 + org.wyrez + AudioProcessing + 1.0 + jar + AudioProcessing + + + + org.apache.maven.plugins + maven-compiler-plugin + 2.3.2 + + 1.7 + 1.7 + + + + + + + javazoom.jl + jlayer + 1.0.1 + system + ${basedir}/lib/jl1.0.1.jar + + + javazoom.jl + mp3spi + 1.9.5 + system + ${basedir}/lib/mp3spi1.9.5.jar + + + org.tritonus + share + 0.3.6 + system + ${basedir}/lib/tritonus_share-0.3.6.jar + + + org.tritonus + aos + 0.3.6 + system + ${basedir}/lib/tritonus_aos-0.3.6.jar + + + diff --git a/AudioProcessing/src/main/java/org/wyrez/audio/AudioDevice.java b/AudioProcessing/src/main/java/org/wyrez/audio/AudioDevice.java new file mode 100644 index 0000000..e057c4f --- /dev/null +++ b/AudioProcessing/src/main/java/org/wyrez/audio/AudioDevice.java @@ -0,0 +1,97 @@ +/* + * Copyright (C) 2013 Darth Affe 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 . + * + * + * 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); + } + } +} diff --git a/AudioProcessing/src/main/java/org/wyrez/audio/AudioProcessor.java b/AudioProcessing/src/main/java/org/wyrez/audio/AudioProcessor.java new file mode 100644 index 0000000..9438532 --- /dev/null +++ b/AudioProcessing/src/main/java/org/wyrez/audio/AudioProcessor.java @@ -0,0 +1,418 @@ +/* + * Copyright (C) 2013 Darth Affe 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 . + * + * 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 spectralFlux = new ArrayList(); + + 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 peakDiff = new HashMap(); + 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; + } +} diff --git a/AudioProcessing/src/main/java/org/wyrez/audio/analysis/DFT.java b/AudioProcessing/src/main/java/org/wyrez/audio/analysis/DFT.java new file mode 100644 index 0000000..f33dd1d --- /dev/null +++ b/AudioProcessing/src/main/java/org/wyrez/audio/analysis/DFT.java @@ -0,0 +1,135 @@ +/* + * 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. + * + * 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 + * timeSize and will throw and IllegalArgumentException if this is + * not the case. + * + * @author Damien Di Fede + * + * @see FourierTransform + * @see FFT + * @see The Discrete Fourier + * Transform + * + */ +public class DFT extends FourierTransform { + + /** + * Constructs a DFT that expects audio buffers of length + * timeSize that have been recorded with a sample rate of + * sampleRate. Will throw an IllegalArgumentException if + * timeSize 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]; + } +} diff --git a/AudioProcessing/src/main/java/org/wyrez/audio/analysis/FFT.java b/AudioProcessing/src/main/java/org/wyrez/audio/analysis/FFT.java new file mode 100644 index 0000000..a1ebc9e --- /dev/null +++ b/AudioProcessing/src/main/java/org/wyrez/audio/analysis/FFT.java @@ -0,0 +1,238 @@ +/* + * 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. + * + * 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 + * timeSize that is not a power of two, an IllegalArgumentException + * will be thrown. + * + * @see FourierTransform + * @see The Fast Fourier + * Transform + * + * @author Damien Di Fede + * + */ +public class FFT extends FourierTransform { + + /** + * Constructs an FFT that will accept sample buffers that are + * timeSize long and have been recorded with a sample rate of + * sampleRate. + * timeSize must 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); + } + } +} diff --git a/AudioProcessing/src/main/java/org/wyrez/audio/analysis/FourierTransform.java b/AudioProcessing/src/main/java/org/wyrez/audio/analysis/FourierTransform.java new file mode 100644 index 0000000..ceca7bb --- /dev/null +++ b/AudioProcessing/src/main/java/org/wyrez/audio/analysis/FourierTransform.java @@ -0,0 +1,662 @@ +/* + * 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. + * + * 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 Nyquist frequency. + * In other words, given a signal of length + * N, there will be + * N/2 frequency bands in the spectrum. + *

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

+ * Usage + *

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

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

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

+ * Averages + *

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

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

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

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

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

+ * Inverse Transform + *

+ * FourierTransform also supports taking the inverse transform of a spectrum. + * This means that a frequency spectrum will be transformed into a time domain + * signal and placed in a provided sample buffer. The length of the time domain + * signal will be + * timeSize() long. The + * set and + * scale functions allow you the ability to shape the spectrum + * already stored in the object before taking the inverse transform. You might + * use these to filter frequencies in a spectrum or modify it in some other way. + * + * @author Damien Di Fede + * @see The Discrete Fourier + * Transform + */ +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 + * 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, 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 + * specSize() / numAvg 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, + * 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) { + 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 + * 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 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 + * 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, float 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, 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 ith 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 ith 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 + * a. + * + * @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 + * a. + * + * @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 + * ith 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 + * 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 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 + * buffer. + * + * @param buffer the buffer to analyze + */ + public abstract void forward(float[] 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(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 + * buffer. + * + * @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 spec = new ArrayList(); + 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; + } +} diff --git a/AudioProcessing/src/main/java/org/wyrez/audio/decoder/Decoder.java b/AudioProcessing/src/main/java/org/wyrez/audio/decoder/Decoder.java new file mode 100644 index 0000000..a6e4705 --- /dev/null +++ b/AudioProcessing/src/main/java/org/wyrez/audio/decoder/Decoder.java @@ -0,0 +1,67 @@ +/* + * Copyright (C) 2013 Darth Affe 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 . + * + * + * 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(); +} diff --git a/AudioProcessing/src/main/java/org/wyrez/audio/decoder/DecoderFactory.java b/AudioProcessing/src/main/java/org/wyrez/audio/decoder/DecoderFactory.java new file mode 100644 index 0000000..e401c60 --- /dev/null +++ b/AudioProcessing/src/main/java/org/wyrez/audio/decoder/DecoderFactory.java @@ -0,0 +1,73 @@ +/* + * Copyright (C) 2013 Darth Affe 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 . + */ +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."); + } + } +} diff --git a/AudioProcessing/src/main/java/org/wyrez/audio/decoder/MP3Decoder.java b/AudioProcessing/src/main/java/org/wyrez/audio/decoder/MP3Decoder.java new file mode 100644 index 0000000..76367e0 --- /dev/null +++ b/AudioProcessing/src/main/java/org/wyrez/audio/decoder/MP3Decoder.java @@ -0,0 +1,365 @@ +/* + * Copyright (C) 2013 Darth Affe 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 . + * + * + * 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); + } + } +} diff --git a/AudioProcessing/src/main/java/org/wyrez/audio/decoder/WaveDecoder.java b/AudioProcessing/src/main/java/org/wyrez/audio/decoder/WaveDecoder.java new file mode 100644 index 0000000..e5462c4 --- /dev/null +++ b/AudioProcessing/src/main/java/org/wyrez/audio/decoder/WaveDecoder.java @@ -0,0 +1,198 @@ +/* + * Copyright (C) 2013 Darth Affe 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 . + * + * + * 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; + } +} diff --git a/AudioProcessing/src/main/java/org/wyrez/audio/util/Band.java b/AudioProcessing/src/main/java/org/wyrez/audio/util/Band.java new file mode 100644 index 0000000..515709d --- /dev/null +++ b/AudioProcessing/src/main/java/org/wyrez/audio/util/Band.java @@ -0,0 +1,56 @@ +/* + * Copyright (C) 2013 Darth Affe 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 . + */ +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; + } +} diff --git a/AudioProcessing/src/main/java/org/wyrez/audio/util/EndianDataInputStream.java b/AudioProcessing/src/main/java/org/wyrez/audio/util/EndianDataInputStream.java new file mode 100644 index 0000000..0975fda --- /dev/null +++ b/AudioProcessing/src/main/java/org/wyrez/audio/util/EndianDataInputStream.java @@ -0,0 +1,54 @@ +/* + * 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; + } +} diff --git a/AudioProcessing/src/main/java/org/wyrez/audio/util/SampleHelper.java b/AudioProcessing/src/main/java/org/wyrez/audio/util/SampleHelper.java new file mode 100644 index 0000000..852514e --- /dev/null +++ b/AudioProcessing/src/main/java/org/wyrez/audio/util/SampleHelper.java @@ -0,0 +1,100 @@ +/* + * Copyright (C) 2013 Darth Affe 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 . + */ +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 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 samples = new ArrayList(); + 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); + } +} diff --git a/AudioProcessing/tritonus.LICENSE b/AudioProcessing/tritonus.LICENSE new file mode 100644 index 0000000..5bc8fb2 --- /dev/null +++ b/AudioProcessing/tritonus.LICENSE @@ -0,0 +1,481 @@ + 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. 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