/* * Copyright (c) 2008, 2010, Oracle and/or its affiliates. All rights reserved. * DO NOT ALTER OR REMOVE COPYRIGHT NOTICES OR THIS FILE HEADER. * * This code is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License version 2 only, as * published by the Free Software Foundation. Oracle designates this * particular file as subject to the "Classpath" exception as provided * by Oracle in the LICENSE file that accompanied this code. * * This code 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 * version 2 for more details (a copy is included in the LICENSE file that * accompanied this code). * * You should have received a copy of the GNU General Public License version * 2 along with this work; if not, write to the Free Software Foundation, * Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA. * * Please contact Oracle, 500 Oracle Parkway, Redwood Shores, CA 94065 USA * or visit www.oracle.com if you need additional information or have any * questions. */package com.sun.media.sound; import java.io.IOException;import java.io.InputStream;import java.util.ArrayList;import java.util.Arrays; import javax.sound.sampled.AudioFormat;import javax.sound.sampled.AudioInputStream;import javax.sound.sampled.AudioSystem;import javax.sound.sampled.AudioFormat.Encoding;import javax.sound.sampled.spi.FormatConversionProvider; /** * This class is used to convert between 8,16,24,32 bit signed/unsigned * big/litle endian fixed/floating stereo/mono/multi-channel audio streams and * perform sample-rate conversion if needed. * * @author Karl Helgason */public class AudioFloatFormatConverter extends FormatConversionProvider { private static class AudioFloatFormatConverterInputStream extends InputStream { private AudioFloatConverter converter; private AudioFloatInputStream stream; private float[] readfloatbuffer; private int fsize = 0; public AudioFloatFormatConverterInputStream(AudioFormat targetFormat, AudioFloatInputStream stream) { this.stream = stream; converter = AudioFloatConverter.getConverter(targetFormat); fsize = ((targetFormat.getSampleSizeInBits() + 7) / 8); } public int read() throws IOException { byte[] b = new byte[1]; int ret = read(b); if (ret < 0) return ret; return b[0] & 0xFF; } public int read(byte[] b, int off, int len) throws IOException { int flen = len / fsize; if (readfloatbuffer == null || readfloatbuffer.length < flen) readfloatbuffer = new float[flen]; int ret = stream.read(readfloatbuffer, 0, flen); if (ret < 0) return ret; converter.toByteArray(readfloatbuffer, 0, ret, b, off); return ret * fsize; } public int available() throws IOException { int ret = stream.available(); if (ret < 0) return ret; return ret * fsize; } public void close() throws IOException { stream.close(); } public synchronized void mark(int readlimit) { stream.mark(readlimit * fsize); } public boolean markSupported() { return stream.markSupported(); } public synchronized void reset() throws IOException { stream.reset(); } public long skip(long n) throws IOException { long ret = stream.skip(n / fsize); if (ret < 0) return ret; return ret * fsize; } } private static class AudioFloatInputStreamChannelMixer extends AudioFloatInputStream { private int targetChannels; private int sourceChannels; private AudioFloatInputStream ais; private AudioFormat targetFormat; private float[] conversion_buffer; public AudioFloatInputStreamChannelMixer(AudioFloatInputStream ais, int targetChannels) { this.sourceChannels = ais.getFormat().getChannels(); this.targetChannels = targetChannels; this.ais = ais; AudioFormat format = ais.getFormat(); targetFormat = new AudioFormat(format.getEncoding(), format .getSampleRate(), format.getSampleSizeInBits(), targetChannels, (format.getFrameSize() / sourceChannels) * targetChannels, format.getFrameRate(), format .isBigEndian()); } public int available() throws IOException { return (ais.available() / sourceChannels) * targetChannels; } public void close() throws IOException { ais.close(); } public AudioFormat getFormat() { return targetFormat; } public long getFrameLength() { return ais.getFrameLength(); } public void mark(int readlimit) { ais.mark((readlimit / targetChannels) * sourceChannels); } public boolean markSupported() { return ais.markSupported(); } public int read(float[] b, int off, int len) throws IOException { int len2 = (len / targetChannels) * sourceChannels; if (conversion_buffer == null || conversion_buffer.length < len2) conversion_buffer = new float[len2]; int ret = ais.read(conversion_buffer, 0, len2); if (ret < 0) return ret; if (sourceChannels == 1) { int cs = targetChannels; for (int c = 0; c < targetChannels; c++) { for (int i = 0, ix = off + c; i < len2; i++, ix += cs) { b[ix] = conversion_buffer[i]; } } } else if (targetChannels == 1) { int cs = sourceChannels; for (int i = 0, ix = off; i < len2; i += cs, ix++) { b[ix] = conversion_buffer[i]; } for (int c = 1; c < sourceChannels; c++) { for (int i = c, ix = off; i < len2; i += cs, ix++) { b[ix] += conversion_buffer[i]; } } float vol = 1f / ((float) sourceChannels); for (int i = 0, ix = off; i < len2; i += cs, ix++) { b[ix] *= vol; } } else { int minChannels = Math.min(sourceChannels, targetChannels); int off_len = off + len; int ct = targetChannels; int cs = sourceChannels; for (int c = 0; c < minChannels; c++) { for (int i = off + c, ix = c; i < off_len; i += ct, ix += cs) { b[i] = conversion_buffer[ix]; } } for (int c = minChannels; c < targetChannels; c++) { for (int i = off + c; i < off_len; i += ct) { b[i] = 0; } } } return (ret / sourceChannels) * targetChannels; } public void reset() throws IOException { ais.reset(); } public long skip(long len) throws IOException { long ret = ais.skip((len / targetChannels) * sourceChannels); if (ret < 0) return ret; return (ret / sourceChannels) * targetChannels; } } private static class AudioFloatInputStreamResampler extends AudioFloatInputStream { private AudioFloatInputStream ais; private AudioFormat targetFormat; private float[] skipbuffer; private SoftAbstractResampler resampler; private float[] pitch = new float[1]; private float[] ibuffer2; private float[][] ibuffer; private float ibuffer_index = 0; private int ibuffer_len = 0; private int nrofchannels = 0; private float[][] cbuffer; private int buffer_len = 512; private int pad; private int pad2; private float[] ix = new float[1]; private int[] ox = new int[1]; private float[][] mark_ibuffer = null; private float mark_ibuffer_index = 0; private int mark_ibuffer_len = 0; public AudioFloatInputStreamResampler(AudioFloatInputStream ais, AudioFormat format) { this.ais = ais; AudioFormat sourceFormat = ais.getFormat(); targetFormat = new AudioFormat(sourceFormat.getEncoding(), format .getSampleRate(), sourceFormat.getSampleSizeInBits(), sourceFormat.getChannels(), sourceFormat.getFrameSize(), format.getSampleRate(), sourceFormat.isBigEndian()); nrofchannels = targetFormat.getChannels(); Object interpolation = format.getProperty("interpolation"); if (interpolation != null && (interpolation instanceof String)) { String resamplerType = (String) interpolation; if (resamplerType.equalsIgnoreCase("point")) this.resampler = new SoftPointResampler(); if (resamplerType.equalsIgnoreCase("linear")) this.resampler = new SoftLinearResampler2(); if (resamplerType.equalsIgnoreCase("linear1")) this.resampler = new SoftLinearResampler(); if (resamplerType.equalsIgnoreCase("linear2")) this.resampler = new SoftLinearResampler2(); if (resamplerType.equalsIgnoreCase("cubic")) this.resampler = new SoftCubicResampler(); if (resamplerType.equalsIgnoreCase("lanczos")) this.resampler = new SoftLanczosResampler(); if (resamplerType.equalsIgnoreCase("sinc")) this.resampler = new SoftSincResampler(); } if (resampler == null) resampler = new SoftLinearResampler2(); // new // SoftLinearResampler2(); pitch[0] = sourceFormat.getSampleRate() / format.getSampleRate(); pad = resampler.getPadding(); pad2 = pad * 2; ibuffer = new float[nrofchannels][buffer_len + pad2]; ibuffer2 = new float[nrofchannels * buffer_len]; ibuffer_index = buffer_len + pad; ibuffer_len = buffer_len; } public int available() throws IOException { return 0; } public void close() throws IOException { ais.close(); } public AudioFormat getFormat() { return targetFormat; } public long getFrameLength() { return AudioSystem.NOT_SPECIFIED; // ais.getFrameLength(); } public void mark(int readlimit) { ais.mark((int) (readlimit * pitch[0])); mark_ibuffer_index = ibuffer_index; mark_ibuffer_len = ibuffer_len; if (mark_ibuffer == null) { mark_ibuffer = new float[ibuffer.length][ibuffer[0].length]; } for (int c = 0; c < ibuffer.length; c++) { float[] from = ibuffer[c]; float[] to = mark_ibuffer[c]; for (int i = 0; i < to.length; i++) { to[i] = from[i]; } } } public boolean markSupported() { return ais.markSupported(); } private void readNextBuffer() throws IOException { if (ibuffer_len == -1) return; for (int c = 0; c < nrofchannels; c++) { float[] buff = ibuffer[c]; int buffer_len_pad = ibuffer_len + pad2; for (int i = ibuffer_len, ix = 0; i < buffer_len_pad; i++, ix++) { buff[ix] = buff[i]; } } ibuffer_index -= (ibuffer_len); ibuffer_len = ais.read(ibuffer2); if (ibuffer_len >= 0) { while (ibuffer_len < ibuffer2.length) { int ret = ais.read(ibuffer2, ibuffer_len, ibuffer2.length - ibuffer_len); if (ret == -1) break; ibuffer_len += ret; } Arrays.fill(ibuffer2, ibuffer_len, ibuffer2.length, 0); ibuffer_len /= nrofchannels; } else { Arrays.fill(ibuffer2, 0, ibuffer2.length, 0); } int ibuffer2_len = ibuffer2.length; for (int c = 0; c < nrofchannels; c++) { float[] buff = ibuffer[c]; for (int i = c, ix = pad2; i < ibuffer2_len; i += nrofchannels, ix++) { buff[ix] = ibuffer2[i]; } } } public int read(float[] b, int off, int len) throws IOException { if (cbuffer == null || cbuffer[0].length < len / nrofchannels) { cbuffer = new float[nrofchannels][len / nrofchannels]; } if (ibuffer_len == -1) return -1; if (len < 0) return 0; int offlen = off + len; int remain = len / nrofchannels; int destPos = 0; int in_end = ibuffer_len; while (remain > 0) { if (ibuffer_len >= 0) { if (ibuffer_index >= (ibuffer_len + pad)) readNextBuffer(); in_end = ibuffer_len + pad; } if (ibuffer_len < 0) { in_end = pad2; if (ibuffer_index >= in_end) break; } if (ibuffer_index < 0) break; int preDestPos = destPos; for (int c = 0; c < nrofchannels; c++) { ix[0] = ibuffer_index; ox[0] = destPos; float[] buff = ibuffer[c]; resampler.interpolate(buff, ix, in_end, pitch, 0, cbuffer[c], ox, len / nrofchannels); } ibuffer_index = ix[0]; destPos = ox[0]; remain -= destPos - preDestPos; } for (int c = 0; c < nrofchannels; c++) { int ix = 0; float[] buff = cbuffer[c]; for (int i = c + off; i < offlen; i += nrofchannels) { b[i] = buff[ix++]; } } return len - remain * nrofchannels; } public void reset() throws IOException { ais.reset(); if (mark_ibuffer == null) return; ibuffer_index = mark_ibuffer_index; ibuffer_len = mark_ibuffer_len; for (int c = 0; c < ibuffer.length; c++) { float[] from = mark_ibuffer[c]; float[] to = ibuffer[c]; for (int i = 0; i < to.length; i++) { to[i] = from[i]; } } } public long skip(long len) throws IOException { if (len < 0) return 0; if (skipbuffer == null) skipbuffer = new float[1024 * targetFormat.getFrameSize()]; float[] l_skipbuffer = skipbuffer; long remain = len; while (remain > 0) { int ret = read(l_skipbuffer, 0, (int) Math.min(remain, skipbuffer.length)); if (ret < 0) { if (remain == len) return ret; break; } remain -= ret; } return len - remain; } } private Encoding[] formats = { Encoding.PCM_SIGNED, Encoding.PCM_UNSIGNED, Encoding.PCM_FLOAT }; public AudioInputStream getAudioInputStream(Encoding targetEncoding, AudioInputStream sourceStream) { if (sourceStream.getFormat().getEncoding().equals(targetEncoding)) return sourceStream; AudioFormat format = sourceStream.getFormat(); int channels = format.getChannels(); Encoding encoding = targetEncoding; float samplerate = format.getSampleRate(); int bits = format.getSampleSizeInBits(); boolean bigendian = format.isBigEndian(); if (targetEncoding.equals(Encoding.PCM_FLOAT)) bits = 32; AudioFormat targetFormat = new AudioFormat(encoding, samplerate, bits, channels, channels * bits / 8, samplerate, bigendian); return getAudioInputStream(targetFormat, sourceStream); } public AudioInputStream getAudioInputStream(AudioFormat targetFormat, AudioInputStream sourceStream) { if (!isConversionSupported(targetFormat, sourceStream.getFormat())) throw new IllegalArgumentException("Unsupported conversion: " + sourceStream.getFormat().toString() + " to " + targetFormat.toString()); return getAudioInputStream(targetFormat, AudioFloatInputStream .getInputStream(sourceStream)); } Lines 1–500 of 616. 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/* * Copyright (c) 2008, 2010, Oracle and/or its affiliates. All rights reserved. * DO NOT ALTER OR REMOVE COPYRIGHT NOTICES OR THIS FILE HEADER. * * This code is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License version 2 only, as * published by the Free Software Foundation. Oracle designates this * particular file as subject to the "Classpath" exception as provided * by Oracle in the LICENSE file that accompanied this code. * * This code 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 * version 2 for more details (a copy is included in the LICENSE file that * accompanied this code). * * You should have received a copy of the GNU General Public License version * 2 along with this work; if not, write to the Free Software Foundation, * Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA. * * Please contact Oracle, 500 Oracle Parkway, Redwood Shores, CA 94065 USA * or visit www.oracle.com if you need additional information or have any * questions. */ package com.sun.media.sound; import java.io.IOException; import java.io.InputStream; import java.util.ArrayList; import java.util.Arrays; import javax.sound.sampled.AudioFormat; import javax.sound.sampled.AudioInputStream; import javax.sound.sampled.AudioSystem; import javax.sound.sampled.AudioFormat.Encoding; import javax.sound.sampled.spi.FormatConversionProvider; /** * This class is used to convert between 8,16,24,32 bit signed/unsigned * big/litle endian fixed/floating stereo/mono/multi-channel audio streams and * perform sample-rate conversion if needed. * * @author Karl Helgason */ public class AudioFloatFormatConverter extends FormatConversionProvider { private static class AudioFloatFormatConverterInputStream extends InputStream { private AudioFloatConverter converter; private AudioFloatInputStream stream; private float[] readfloatbuffer; private int fsize = 0; public AudioFloatFormatConverterInputStream(AudioFormat targetFormat, AudioFloatInputStream stream) { this.stream = stream; converter = AudioFloatConverter.getConverter(targetFormat); fsize = ((targetFormat.getSampleSizeInBits() + 7) / 8); } public int read() throws IOException { byte[] b = new byte[1]; int ret = read(b); if (ret < 0) return ret; return b[0] & 0xFF; } public int read(byte[] b, int off, int len) throws IOException { int flen = len / fsize; if (readfloatbuffer == null || readfloatbuffer.length < flen) readfloatbuffer = new float[flen]; int ret = stream.read(readfloatbuffer, 0, flen); if (ret < 0) return ret; converter.toByteArray(readfloatbuffer, 0, ret, b, off); return ret * fsize; } public int available() throws IOException { int ret = stream.available(); if (ret < 0) return ret; return ret * fsize; } public void close() throws IOException { stream.close(); } public synchronized void mark(int readlimit) { stream.mark(readlimit * fsize); } public boolean markSupported() { return stream.markSupported(); } public synchronized void reset() throws IOException { stream.reset(); } public long skip(long n) throws IOException { long ret = stream.skip(n / fsize); if (ret < 0) return ret; return ret * fsize; } } private static class AudioFloatInputStreamChannelMixer extends AudioFloatInputStream { private int targetChannels; private int sourceChannels; private AudioFloatInputStream ais; private AudioFormat targetFormat; private float[] conversion_buffer; public AudioFloatInputStreamChannelMixer(AudioFloatInputStream ais, int targetChannels) { this.sourceChannels = ais.getFormat().getChannels(); this.targetChannels = targetChannels; this.ais = ais; AudioFormat format = ais.getFormat(); targetFormat = new AudioFormat(format.getEncoding(), format .getSampleRate(), format.getSampleSizeInBits(), targetChannels, (format.getFrameSize() / sourceChannels) * targetChannels, format.getFrameRate(), format .isBigEndian()); } public int available() throws IOException { return (ais.available() / sourceChannels) * targetChannels; } public void close() throws IOException { ais.close(); } public AudioFormat getFormat() { return targetFormat; } public long getFrameLength() { return ais.getFrameLength(); } public void mark(int readlimit) { ais.mark((readlimit / targetChannels) * sourceChannels); } public boolean markSupported() { return ais.markSupported(); } public int read(float[] b, int off, int len) throws IOException { int len2 = (len / targetChannels) * sourceChannels; if (conversion_buffer == null || conversion_buffer.length < len2) conversion_buffer = new float[len2]; int ret = ais.read(conversion_buffer, 0, len2); if (ret < 0) return ret; if (sourceChannels == 1) { int cs = targetChannels; for (int c = 0; c < targetChannels; c++) { for (int i = 0, ix = off + c; i < len2; i++, ix += cs) { b[ix] = conversion_buffer[i]; } } } else if (targetChannels == 1) { int cs = sourceChannels; for (int i = 0, ix = off; i < len2; i += cs, ix++) { b[ix] = conversion_buffer[i]; } for (int c = 1; c < sourceChannels; c++) { for (int i = c, ix = off; i < len2; i += cs, ix++) { b[ix] += conversion_buffer[i]; } } float vol = 1f / ((float) sourceChannels); for (int i = 0, ix = off; i < len2; i += cs, ix++) { b[ix] *= vol; } } else { int minChannels = Math.min(sourceChannels, targetChannels); int off_len = off + len; int ct = targetChannels; int cs = sourceChannels; for (int c = 0; c < minChannels; c++) { for (int i = off + c, ix = c; i < off_len; i += ct, ix += cs) { b[i] = conversion_buffer[ix]; } } for (int c = minChannels; c < targetChannels; c++) { for (int i = off + c; i < off_len; i += ct) { b[i] = 0; } } } return (ret / sourceChannels) * targetChannels; } public void reset() throws IOException { ais.reset(); } public long skip(long len) throws IOException { long ret = ais.skip((len / targetChannels) * sourceChannels); if (ret < 0) return ret; return (ret / sourceChannels) * targetChannels; } } private static class AudioFloatInputStreamResampler extends AudioFloatInputStream { private AudioFloatInputStream ais; private AudioFormat targetFormat; private float[] skipbuffer; private SoftAbstractResampler resampler; private float[] pitch = new float[1]; private float[] ibuffer2; private float[][] ibuffer; private float ibuffer_index = 0; private int ibuffer_len = 0; private int nrofchannels = 0; private float[][] cbuffer; private int buffer_len = 512; private int pad; private int pad2; private float[] ix = new float[1]; private int[] ox = new int[1]; private float[][] mark_ibuffer = null; private float mark_ibuffer_index = 0; private int mark_ibuffer_len = 0; public AudioFloatInputStreamResampler(AudioFloatInputStream ais, AudioFormat format) { this.ais = ais; AudioFormat sourceFormat = ais.getFormat(); targetFormat = new AudioFormat(sourceFormat.getEncoding(), format .getSampleRate(), sourceFormat.getSampleSizeInBits(), sourceFormat.getChannels(), sourceFormat.getFrameSize(), format.getSampleRate(), sourceFormat.isBigEndian()); nrofchannels = targetFormat.getChannels(); Object interpolation = format.getProperty("interpolation"); if (interpolation != null && (interpolation instanceof String)) { String resamplerType = (String) interpolation; if (resamplerType.equalsIgnoreCase("point")) this.resampler = new SoftPointResampler(); if (resamplerType.equalsIgnoreCase("linear")) this.resampler = new SoftLinearResampler2(); if (resamplerType.equalsIgnoreCase("linear1")) this.resampler = new SoftLinearResampler(); if (resamplerType.equalsIgnoreCase("linear2")) this.resampler = new SoftLinearResampler2(); if (resamplerType.equalsIgnoreCase("cubic")) this.resampler = new SoftCubicResampler(); if (resamplerType.equalsIgnoreCase("lanczos")) this.resampler = new SoftLanczosResampler(); if (resamplerType.equalsIgnoreCase("sinc")) this.resampler = new SoftSincResampler(); } if (resampler == null) resampler = new SoftLinearResampler2(); // new // SoftLinearResampler2(); pitch[0] = sourceFormat.getSampleRate() / format.getSampleRate(); pad = resampler.getPadding(); pad2 = pad * 2; ibuffer = new float[nrofchannels][buffer_len + pad2]; ibuffer2 = new float[nrofchannels * buffer_len]; ibuffer_index = buffer_len + pad; ibuffer_len = buffer_len; } public int available() throws IOException { return 0; } public void close() throws IOException { ais.close(); } public AudioFormat getFormat() { return targetFormat; } public long getFrameLength() { return AudioSystem.NOT_SPECIFIED; // ais.getFrameLength(); } public void mark(int readlimit) { ais.mark((int) (readlimit * pitch[0])); mark_ibuffer_index = ibuffer_index; mark_ibuffer_len = ibuffer_len; if (mark_ibuffer == null) { mark_ibuffer = new float[ibuffer.length][ibuffer[0].length]; } for (int c = 0; c < ibuffer.length; c++) { float[] from = ibuffer[c]; float[] to = mark_ibuffer[c]; for (int i = 0; i < to.length; i++) { to[i] = from[i]; } } } public boolean markSupported() { return ais.markSupported(); } private void readNextBuffer() throws IOException { if (ibuffer_len == -1) return; for (int c = 0; c < nrofchannels; c++) { float[] buff = ibuffer[c]; int buffer_len_pad = ibuffer_len + pad2; for (int i = ibuffer_len, ix = 0; i < buffer_len_pad; i++, ix++) { buff[ix] = buff[i]; } } ibuffer_index -= (ibuffer_len); ibuffer_len = ais.read(ibuffer2); if (ibuffer_len >= 0) { while (ibuffer_len < ibuffer2.length) { int ret = ais.read(ibuffer2, ibuffer_len, ibuffer2.length - ibuffer_len); if (ret == -1) break; ibuffer_len += ret; } Arrays.fill(ibuffer2, ibuffer_len, ibuffer2.length, 0); ibuffer_len /= nrofchannels; } else { Arrays.fill(ibuffer2, 0, ibuffer2.length, 0); } int ibuffer2_len = ibuffer2.length; for (int c = 0; c < nrofchannels; c++) { float[] buff = ibuffer[c]; for (int i = c, ix = pad2; i < ibuffer2_len; i += nrofchannels, ix++) { buff[ix] = ibuffer2[i]; } } } public int read(float[] b, int off, int len) throws IOException { if (cbuffer == null || cbuffer[0].length < len / nrofchannels) { cbuffer = new float[nrofchannels][len / nrofchannels]; } if (ibuffer_len == -1) return -1; if (len < 0) return 0; int offlen = off + len; int remain = len / nrofchannels; int destPos = 0; int in_end = ibuffer_len; while (remain > 0) { if (ibuffer_len >= 0) { if (ibuffer_index >= (ibuffer_len + pad)) readNextBuffer(); in_end = ibuffer_len + pad; } if (ibuffer_len < 0) { in_end = pad2; if (ibuffer_index >= in_end) break; } if (ibuffer_index < 0) break; int preDestPos = destPos; for (int c = 0; c < nrofchannels; c++) { ix[0] = ibuffer_index; ox[0] = destPos; float[] buff = ibuffer[c]; resampler.interpolate(buff, ix, in_end, pitch, 0, cbuffer[c], ox, len / nrofchannels); } ibuffer_index = ix[0]; destPos = ox[0]; remain -= destPos - preDestPos; } for (int c = 0; c < nrofchannels; c++) { int ix = 0; float[] buff = cbuffer[c]; for (int i = c + off; i < offlen; i += nrofchannels) { b[i] = buff[ix++]; } } return len - remain * nrofchannels; } public void reset() throws IOException { ais.reset(); if (mark_ibuffer == null) return; ibuffer_index = mark_ibuffer_index; ibuffer_len = mark_ibuffer_len; for (int c = 0; c < ibuffer.length; c++) { float[] from = mark_ibuffer[c]; float[] to = ibuffer[c]; for (int i = 0; i < to.length; i++) { to[i] = from[i]; } } } public long skip(long len) throws IOException { if (len < 0) return 0; if (skipbuffer == null) skipbuffer = new float[1024 * targetFormat.getFrameSize()]; float[] l_skipbuffer = skipbuffer; long remain = len; while (remain > 0) { int ret = read(l_skipbuffer, 0, (int) Math.min(remain, skipbuffer.length)); if (ret < 0) { if (remain == len) return ret; break; } remain -= ret; } return len - remain; } } private Encoding[] formats = { Encoding.PCM_SIGNED, Encoding.PCM_UNSIGNED, Encoding.PCM_FLOAT }; public AudioInputStream getAudioInputStream(Encoding targetEncoding, AudioInputStream sourceStream) { if (sourceStream.getFormat().getEncoding().equals(targetEncoding)) return sourceStream; AudioFormat format = sourceStream.getFormat(); int channels = format.getChannels(); Encoding encoding = targetEncoding; float samplerate = format.getSampleRate(); int bits = format.getSampleSizeInBits(); boolean bigendian = format.isBigEndian(); if (targetEncoding.equals(Encoding.PCM_FLOAT)) bits = 32; AudioFormat targetFormat = new AudioFormat(encoding, samplerate, bits, channels, channels * bits / 8, samplerate, bigendian); return getAudioInputStream(targetFormat, sourceStream); } public AudioInputStream getAudioInputStream(AudioFormat targetFormat, AudioInputStream sourceStream) { if (!isConversionSupported(targetFormat, sourceStream.getFormat())) throw new IllegalArgumentException("Unsupported conversion: " + sourceStream.getFormat().toString() + " to " + targetFormat.toString()); return getAudioInputStream(targetFormat, AudioFloatInputStream .getInputStream(sourceStream)); }