Java example source code file (AudioFloatFormatConverter.java)
The AudioFloatFormatConverter.java Java example source code
/*
* Copyright (c) 2008, 2013, 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 final class AudioFloatFormatConverter extends FormatConversionProvider {
private static class AudioFloatFormatConverterInputStream extends
InputStream {
private final AudioFloatConverter converter;
private final AudioFloatInputStream stream;
private float[] readfloatbuffer;
private final int fsize;
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 final int targetChannels;
private final int sourceChannels;
private final AudioFloatInputStream ais;
private final AudioFormat targetFormat;
private float[] conversion_buffer;
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 final AudioFloatInputStream ais;
private final AudioFormat targetFormat;
private float[] skipbuffer;
private SoftAbstractResampler resampler;
private final float[] pitch = new float[1];
private final float[] ibuffer2;
private final float[][] ibuffer;
private float ibuffer_index = 0;
private int ibuffer_len = 0;
private final int nrofchannels;
private float[][] cbuffer;
private final int buffer_len = 512;
private final int pad;
private final int pad2;
private final float[] ix = new float[1];
private final int[] ox = new int[1];
private float[][] mark_ibuffer = null;
private float mark_ibuffer_index = 0;
private int mark_ibuffer_len = 0;
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 final 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));
}
public AudioInputStream getAudioInputStream(AudioFormat targetFormat,
AudioFloatInputStream sourceStream) {
if (!isConversionSupported(targetFormat, sourceStream.getFormat()))
throw new IllegalArgumentException("Unsupported conversion: "
+ sourceStream.getFormat().toString() + " to "
+ targetFormat.toString());
if (targetFormat.getChannels() != sourceStream.getFormat()
.getChannels())
sourceStream = new AudioFloatInputStreamChannelMixer(sourceStream,
targetFormat.getChannels());
if (Math.abs(targetFormat.getSampleRate()
- sourceStream.getFormat().getSampleRate()) > 0.000001)
sourceStream = new AudioFloatInputStreamResampler(sourceStream,
targetFormat);
return new AudioInputStream(new AudioFloatFormatConverterInputStream(
targetFormat, sourceStream), targetFormat, sourceStream
.getFrameLength());
}
public Encoding[] getSourceEncodings() {
return new Encoding[] { Encoding.PCM_SIGNED, Encoding.PCM_UNSIGNED,
Encoding.PCM_FLOAT };
}
public Encoding[] getTargetEncodings() {
return new Encoding[] { Encoding.PCM_SIGNED, Encoding.PCM_UNSIGNED,
Encoding.PCM_FLOAT };
}
public Encoding[] getTargetEncodings(AudioFormat sourceFormat) {
if (AudioFloatConverter.getConverter(sourceFormat) == null)
return new Encoding[0];
return new Encoding[] { Encoding.PCM_SIGNED, Encoding.PCM_UNSIGNED,
Encoding.PCM_FLOAT };
}
public AudioFormat[] getTargetFormats(Encoding targetEncoding,
AudioFormat sourceFormat) {
if (AudioFloatConverter.getConverter(sourceFormat) == null)
return new AudioFormat[0];
int channels = sourceFormat.getChannels();
ArrayList<AudioFormat> formats = new ArrayList
Other Java examples (source code examples)Here is a short list of links related to this Java AudioFloatFormatConverter.java source code file: |
The search page
Other Java source code examples at this package level
Click here to learn more about this project
