/*
* 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
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package com.sun.media.sound;
import java.io.IOException;
import java.util.ArrayList;
import java.util.Arrays;
import java.util.List;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.BooleanControl;
import javax.sound.sampled.Control;
import javax.sound.sampled.Control.Type;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.FloatControl;
import javax.sound.sampled.LineEvent;
import javax.sound.sampled.LineListener;
/**
* General software mixing line.
*
* @author Karl Helgason
*/
public abstract class SoftMixingDataLine implements DataLine {
public static final FloatControl.Type CHORUS_SEND = new FloatControl.Type(
"Chorus Send") {
};
protected static final 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 int nrofchannels = 0;
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;
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;
}
@Override
public int available() throws IOException {
return 0;
}
@Override
public void close() throws IOException {
ais.close();
}
@Override
public AudioFormat getFormat() {
return targetFormat;
}
@Override
public long getFrameLength() {
return AudioSystem.NOT_SPECIFIED; // ais.getFrameLength();
}
@Override
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];
}
}
}
@Override
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];
}
}
}
@Override
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 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; i < b.length; i += nrofchannels) {
b[i] = buff[ix++];
}
}
return len - remain * nrofchannels;
}
@Override
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];
}
}
}
@Override
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 class Gain extends FloatControl {
private Gain() {
super(FloatControl.Type.MASTER_GAIN, -80f, 6.0206f, 80f / 128.0f,
-1, 0.0f, "dB", "Minimum", "", "Maximum");
}
@Override
public void setValue(float newValue) {
super.setValue(newValue);
calcVolume();
}
}
private final class Mute extends BooleanControl {
private Mute() {
super(BooleanControl.Type.MUTE, false, "True", "False");
}
@Override
public void setValue(boolean newValue) {
super.setValue(newValue);
calcVolume();
}
}
private final class ApplyReverb extends BooleanControl {
private ApplyReverb() {
super(BooleanControl.Type.APPLY_REVERB, false, "True", "False");
}
@Override
public void setValue(boolean newValue) {
super.setValue(newValue);
calcVolume();
}
}
private final class Balance extends FloatControl {
private Balance() {
super(FloatControl.Type.BALANCE, -1.0f, 1.0f, (1.0f / 128.0f), -1,
0.0f, "", "Left", "Center", "Right");
}
@Override
public void setValue(float newValue) {
super.setValue(newValue);
calcVolume();
}
}
private final class Pan extends FloatControl {
private Pan() {
super(FloatControl.Type.PAN, -1.0f, 1.0f, (1.0f / 128.0f), -1,
0.0f, "", "Left", "Center", "Right");
}
@Override
public void setValue(float newValue) {
super.setValue(newValue);
balance_control.setValue(newValue);
}
@Override
public float getValue() {
return balance_control.getValue();
}
}
private final class ReverbSend extends FloatControl {
private ReverbSend() {
super(FloatControl.Type.REVERB_SEND, -80f, 6.0206f, 80f / 128.0f,
-1, -80f, "dB", "Minimum", "", "Maximum");
}
@Override
public void setValue(float newValue) {
super.setValue(newValue);
balance_control.setValue(newValue);
}
}
private final class ChorusSend extends FloatControl {
private ChorusSend() {
super(CHORUS_SEND, -80f, 6.0206f, 80f / 128.0f, -1, -80f, "dB",
"Minimum", "", "Maximum");
}
@Override
public void setValue(float newValue) {
super.setValue(newValue);
balance_control.setValue(newValue);
}
}
private final Gain gain_control = new Gain();
private final Mute mute_control = new Mute();
private final Balance balance_control = new Balance();
private final Pan pan_control = new Pan();
private final ReverbSend reverbsend_control = new ReverbSend();
private final ChorusSend chorussend_control = new ChorusSend();
private final ApplyReverb apply_reverb = new ApplyReverb();
private final Control[] controls;
float leftgain = 1;
float rightgain = 1;
float eff1gain = 0;
float eff2gain = 0;
List<LineListener> listeners = new ArrayList<>();
final Object control_mutex;
SoftMixingMixer mixer;
DataLine.Info info;
protected abstract void processControlLogic();
protected abstract void processAudioLogic(SoftAudioBuffer[] buffers);
SoftMixingDataLine(SoftMixingMixer mixer, DataLine.Info info) {
this.mixer = mixer;
this.info = info;
this.control_mutex = mixer.control_mutex;
controls = new Control[] { gain_control, mute_control, balance_control,
pan_control, reverbsend_control, chorussend_control,
apply_reverb };
calcVolume();
}
final void calcVolume() {
synchronized (control_mutex) {
double gain = Math.pow(10.0, gain_control.getValue() / 20.0);
if (mute_control.getValue())
gain = 0;
leftgain = (float) gain;
rightgain = (float) gain;
if (mixer.getFormat().getChannels() > 1) {
// -1 = Left, 0 Center, 1 = Right
double balance = balance_control.getValue();
if (balance > 0)
leftgain *= (1 - balance);
else
rightgain *= (1 + balance);
}
}
eff1gain = (float) Math.pow(10.0, reverbsend_control.getValue() / 20.0);
eff2gain = (float) Math.pow(10.0, chorussend_control.getValue() / 20.0);
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