/** * Copyright 2007 DFKI GmbH. * All Rights Reserved. Use is subject to license terms. * * This file is part of MARY TTS. * * MARY TTS is free software: you can redistribute it and/or modify * it under the terms of the GNU Lesser General Public License as published by * the Free Software Foundation, version 3 of the License. * * 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 Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public License * along with this program. If not, see <http://www.gnu.org/licenses/>. * */ package marytts.signalproc.filter; import java.io.File; import java.io.IOException; import java.util.Arrays; 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 marytts.util.data.BufferedDoubleDataSource; import marytts.util.data.audio.AudioDoubleDataSource; import marytts.util.data.audio.DDSAudioInputStream; import marytts.util.signal.SignalProcUtils; /** * @author Oytun Türk * */ public class FIRBandPassFilterBankSynthesiser { public FIRBandPassFilterBankSynthesiser() { } public double[] apply(FIRBandPassFilterBankAnalyser analyser, Subband[] subbands) { return apply(analyser, subbands, true); } public double[] apply(FIRBandPassFilterBankAnalyser analyser, Subband[] subbands, boolean bNormalizeInOverlappingRegions) { double[] x = null; if (analyser != null && analyser.filters != null && subbands != null) { int i, j, maxLen; assert analyser.filters.length == subbands.length; // Add all subbands up and then apply the smooth gain normalization filter maxLen = subbands[0].waveform.length; for (i = 1; i < subbands.length; i++) maxLen = Math.max(maxLen, subbands[i].waveform.length); x = new double[maxLen]; Arrays.fill(x, 0.0); for (i = 0; i < subbands.length; i++) { for (j = 0; j < subbands[i].waveform.length; j++) x[j] += subbands[i].waveform[j]; } x = SignalProcUtils.filterfd(analyser.normalizationFilterTransformedIR, x, subbands[0].samplingRate); } return x; } public static void main(String[] args) throws UnsupportedAudioFileException, IOException { AudioInputStream inputAudio = AudioSystem.getAudioInputStream(new File(args[0])); int samplingRate = (int) inputAudio.getFormat().getSampleRate(); AudioDoubleDataSource signal = new AudioDoubleDataSource(inputAudio); double[] x = signal.getAllData(); int i; int numBands = 4; double overlapAround1000Hz = 100.0; FIRBandPassFilterBankAnalyser analyser = new FIRBandPassFilterBankAnalyser(numBands, samplingRate, overlapAround1000Hz); Subband[] subbands = analyser.apply(x); DDSAudioInputStream outputAudio; AudioFormat outputFormat; String outFileName; // Write highpass components 0 to numLevels-1 for (i = 0; i < subbands.length; i++) { outputFormat = new AudioFormat((int) (subbands[i].samplingRate), inputAudio.getFormat().getSampleSizeInBits(), inputAudio.getFormat().getChannels(), true, true); outputAudio = new DDSAudioInputStream(new BufferedDoubleDataSource(subbands[i].waveform), outputFormat); outFileName = args[0].substring(0, args[0].length() - 4) + "_band" + String.valueOf(i + 1) + ".wav"; AudioSystem.write(outputAudio, AudioFileFormat.Type.WAVE, new File(outFileName)); } FIRBandPassFilterBankSynthesiser synthesiser = new FIRBandPassFilterBankSynthesiser(); double[] y = synthesiser.apply(analyser, subbands); outputFormat = new AudioFormat((int) (subbands[0].samplingRate), inputAudio.getFormat().getSampleSizeInBits(), inputAudio .getFormat().getChannels(), true, true); outputAudio = new DDSAudioInputStream(new BufferedDoubleDataSource(y), outputFormat); outFileName = args[0].substring(0, args[0].length() - 4) + "_resynthesis" + ".wav"; AudioSystem.write(outputAudio, AudioFileFormat.Type.WAVE, new File(outFileName)); } }