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authorSven Gothel <[email protected]>2023-05-18 04:13:57 +0200
committerSven Gothel <[email protected]>2023-05-18 04:13:57 +0200
commitd5daaaab3544d9af49056f57a1fcf53abef17deb (patch)
tree47d664d81f6262596d148e8dda4f8052aea2872d /src/java/com/jogamp/openal/util
parent5320233de825b5e3c2131c9303ef94990a40fcb4 (diff)
ALAudioSink: Promote to public, be fully functional regarding AudioFormat and OpenAL paremeter. Can be 'plugged' into existing OpenAL logic.
Diffstat (limited to 'src/java/com/jogamp/openal/util')
-rw-r--r--src/java/com/jogamp/openal/util/ALAudioSink.java1190
-rw-r--r--src/java/com/jogamp/openal/util/ALHelpers.java242
2 files changed, 1413 insertions, 19 deletions
diff --git a/src/java/com/jogamp/openal/util/ALAudioSink.java b/src/java/com/jogamp/openal/util/ALAudioSink.java
new file mode 100644
index 0000000..e763899
--- /dev/null
+++ b/src/java/com/jogamp/openal/util/ALAudioSink.java
@@ -0,0 +1,1190 @@
+/**
+ * Copyright 2013-2023 JogAmp Community. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without modification, are
+ * permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice, this list of
+ * conditions and the following disclaimer.
+ *
+ * 2. Redistributions in binary form must reproduce the above copyright notice, this list
+ * of conditions and the following disclaimer in the documentation and/or other materials
+ * provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY JogAmp Community ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND
+ * FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL JogAmp Community OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ *
+ * The views and conclusions contained in the software and documentation are those of the
+ * authors and should not be interpreted as representing official policies, either expressed
+ * or implied, of JogAmp Community.
+ */
+package com.jogamp.openal.util;
+
+
+import java.nio.ByteBuffer;
+import java.util.Arrays;
+import java.util.concurrent.TimeUnit;
+
+import jogamp.openal.Debug;
+
+import com.jogamp.common.ExceptionUtils;
+import com.jogamp.common.av.AudioFormat;
+import com.jogamp.common.av.AudioSink;
+import com.jogamp.common.av.AudioSink.AudioFrame;
+import com.jogamp.common.os.Clock;
+import com.jogamp.common.util.LFRingbuffer;
+import com.jogamp.common.util.PropertyAccess;
+import com.jogamp.common.util.Ringbuffer;
+import com.jogamp.common.util.locks.LockFactory;
+import com.jogamp.common.util.locks.RecursiveLock;
+import com.jogamp.openal.AL;
+import com.jogamp.openal.ALC;
+import com.jogamp.openal.ALCConstants;
+import com.jogamp.openal.ALCcontext;
+import com.jogamp.openal.ALCdevice;
+import com.jogamp.openal.ALConstants;
+import com.jogamp.openal.ALExt;
+import com.jogamp.openal.ALFactory;
+
+/***
+ * OpenAL {@link AudioSink} implementation.
+ * <p>
+ * Besides given {@link AudioSink} functionality, implementation is fully functional regarding {@link AudioFormat} and all OpenAL parameter.<br/>
+ * <ul>
+ * <li>All OpenAL parameter can be queried</li>
+ * <li>Instance can be constructed with an OpenAL device and context, see {@link #ALAudioSink(ALCdevice, ALCcontext)}</li>
+ * <li>Initialization can be performed with OpenAL paramters, see {@link #init(int, int, int, int, int, float, int, int, int)}</li>
+ * </ul>
+ * </p>
+ */
+public class ALAudioSink implements AudioSink {
+ private static final boolean DEBUG_TRACE;
+ private static final ALC alc;
+ private static final AL al;
+ private static final ALExt alExt;
+ private static final boolean staticAvailable;
+
+ private String deviceSpecifier;
+ private ALCdevice device;
+ private boolean hasSOFTBufferSamples;
+ private boolean hasEXTMcFormats;
+ private boolean hasEXTFloat32;
+ private boolean hasEXTDouble;
+ private boolean hasALC_thread_local_context;
+ private int preferredSampleRate;
+ private AudioFormat preferredAudioFormat;
+ private ALCcontext context;
+ private final RecursiveLock lock = LockFactory.createRecursiveLock();
+ private volatile Thread exclusiveThread = null;
+
+ /** Playback speed, range [0.5 - 2.0], default 1.0. */
+ private float playSpeed;
+ private float volume = 1.0f;
+
+ static class ALAudioFrame extends AudioFrame {
+ private final int alBuffer;
+
+ ALAudioFrame(final int alBuffer) {
+ this.alBuffer = alBuffer;
+ }
+ public ALAudioFrame(final int alBuffer, final int pts, final int duration, final int dataSize) {
+ super(pts, duration, dataSize);
+ this.alBuffer = alBuffer;
+ }
+
+ /** Get this frame's OpenAL buffer name */
+ public final int getALBuffer() { return alBuffer; }
+
+ @Override
+ public String toString() {
+ return "ALAudioFrame[pts " + pts + " ms, l " + duration + " ms, " + byteSize + " bytes, buffer "+alBuffer+"]";
+ }
+ }
+
+ private int[] alBufferNames = null;
+ private int avgFrameDuration = 0; // [ms]
+ private int frameGrowAmount = 0;
+ private int frameLimit = 0;
+
+ private Ringbuffer<ALAudioFrame> alFramesAvail = null;
+ private Ringbuffer<ALAudioFrame> alFramesPlaying = null;
+ private volatile int alBufferBytesQueued = 0;
+ private volatile int playingPTS = AudioFrame.INVALID_PTS;
+ private volatile int enqueuedFrameCount;
+
+ private final int[] alSource = { -1 }; // actually ALuint, but JOAL expects INT_MAX limit is ok!
+ private AudioFormat chosenFormat;
+ private int alChannelLayout;
+ private int alSampleType;
+ private int alFormat;
+ private boolean available;
+
+ private volatile boolean playRequested = false;
+
+ static {
+ Debug.initSingleton();
+ DEBUG_TRACE = PropertyAccess.isPropertyDefined("joal.debug.AudioSink.trace", true);
+
+ ALC _alc = null;
+ AL _al = null;
+ ALExt _alExt = null;
+ try {
+ _alc = ALFactory.getALC();
+ _al = ALFactory.getAL();
+ _alExt = ALFactory.getALExt();
+ } catch(final Throwable t) {
+ if( DEBUG ) {
+ System.err.println("ALAudioSink: Caught "+t.getClass().getName()+": "+t.getMessage());
+ t.printStackTrace();
+ }
+ }
+ alc = _alc;
+ al = _al;
+ alExt = _alExt;
+ staticAvailable = null != alc && null != al && null != alExt;
+ }
+
+ private void clearPreALError(final String prefix) {
+ checkALError(prefix);
+ }
+ private boolean checkALError(final String prefix) {
+ final int alcErr = alc.alcGetError(device);
+ final int alErr = al.alGetError();
+ final boolean ok = ALCConstants.ALC_NO_ERROR == alcErr && ALConstants.AL_NO_ERROR == alErr;
+ if( DEBUG ) {
+ System.err.println("ALAudioSink."+prefix+": ok "+ok+", err [alc "+toHexString(alcErr)+", al "+toHexString(alErr)+"]");
+ }
+ return ok;
+ }
+
+ /**
+ * Create a new instance with all new OpenAL objects, i.e. opened default {@link ALCdevice} and new {@link ALCcontext}.
+ */
+ public ALAudioSink() {
+ this(null, null);
+ }
+
+ /**
+ * Create a new instance based on optional given OpenAL objects.
+ *
+ * @param alDevice optional OpenAL device, a default device is opened if null.
+ * @param alContext optional OpenAL context associated to the given device, a new context is created if null.
+ */
+ public ALAudioSink(final ALCdevice alDevice, final ALCcontext alContext) {
+ available = false;
+ chosenFormat = null;
+
+ if( !staticAvailable ) {
+ return;
+ }
+ synchronized(ALAudioSink.class) {
+ try {
+ if( null == alDevice ) {
+ // Get handle to default device.
+ device = alc.alcOpenDevice(null);
+ if (device == null) {
+ throw new RuntimeException(getThreadName()+": ALAudioSink: Error opening default OpenAL device");
+ }
+ clearPreALError("init.dev");
+ } else {
+ device = alDevice;
+ }
+ int checkErrIter = 1;
+
+ // Get the device specifier.
+ deviceSpecifier = alc.alcGetString(device, ALCConstants.ALC_DEVICE_SPECIFIER);
+ if (deviceSpecifier == null) {
+ throw new RuntimeException(getThreadName()+": ALAudioSink: Error getting specifier for default OpenAL device");
+ }
+ clearPreALError("init."+checkErrIter++);
+
+ if( null == alContext ) {
+ // Create audio context.
+ // final int[] attrs = new int[] { ALC.ALC_FREQUENCY, DefaultFormat.sampleRate, 0 };
+ // context = alc.alcCreateContext(device, attrs, 0);
+ context = alc.alcCreateContext(device, null);
+ if (context == null) {
+ throw new RuntimeException(getThreadName()+": ALAudioSink: Error creating OpenAL context for "+deviceSpecifier);
+ }
+ } else {
+ context = alContext;
+ }
+
+ lockContext();
+ try {
+ // Check for an error.
+ if ( alc.alcGetError(device) != ALCConstants.ALC_NO_ERROR ) {
+ throw new RuntimeException(getThreadName()+": ALAudioSink: Error making OpenAL context current");
+ }
+ hasSOFTBufferSamples = al.alIsExtensionPresent(ALHelpers.AL_SOFT_buffer_samples);
+ hasEXTMcFormats = al.alIsExtensionPresent(ALHelpers.AL_EXT_MCFORMATS);
+ hasEXTFloat32 = al.alIsExtensionPresent(ALHelpers.AL_EXT_FLOAT32);
+ hasEXTDouble = al.alIsExtensionPresent(ALHelpers.AL_EXT_DOUBLE);
+ hasALC_thread_local_context = alc.alcIsExtensionPresent(null, ALHelpers.ALC_EXT_thread_local_context) ||
+ alc.alcIsExtensionPresent(device, ALHelpers.ALC_EXT_thread_local_context) ;
+ clearPreALError("init."+checkErrIter++);
+ preferredSampleRate = querySampleRate();
+ preferredAudioFormat = new AudioFormat(preferredSampleRate, DefaultFormat.sampleSize, DefaultFormat.channelCount, DefaultFormat.signed, DefaultFormat.fixedP, DefaultFormat.planar, DefaultFormat.littleEndian);
+ if( DEBUG ) {
+ final int[] alcvers = { 0, 0 };
+ System.out.println("ALAudioSink: OpenAL Version: "+al.alGetString(ALConstants.AL_VERSION));
+ System.out.println("ALAudioSink: OpenAL Extensions: "+al.alGetString(ALConstants.AL_EXTENSIONS));
+ clearPreALError("init."+checkErrIter++);
+ System.out.println("ALAudioSink: Null device OpenALC:");
+ alc.alcGetIntegerv(null, ALCConstants.ALC_MAJOR_VERSION, 1, alcvers, 0);
+ alc.alcGetIntegerv(null, ALCConstants.ALC_MINOR_VERSION, 1, alcvers, 1);
+ System.out.println(" Version: "+alcvers[0]+"."+alcvers[1]);
+ System.out.println(" Extensions: "+alc.alcGetString(null, ALCConstants.ALC_EXTENSIONS));
+ clearPreALError("init."+checkErrIter++);
+ System.out.println("ALAudioSink: Device "+deviceSpecifier+" OpenALC:");
+ alc.alcGetIntegerv(device, ALCConstants.ALC_MAJOR_VERSION, 1, alcvers, 0);
+ alc.alcGetIntegerv(device, ALCConstants.ALC_MINOR_VERSION, 1, alcvers, 1);
+ System.out.println(" Version: "+alcvers[0]+"."+alcvers[1]);
+ System.out.println(" Extensions: "+alc.alcGetString(device, ALCConstants.ALC_EXTENSIONS));
+ System.out.println("ALAudioSink: hasSOFTBufferSamples "+hasSOFTBufferSamples);
+ System.out.println("ALAudioSink: hasEXTMcFormats "+hasEXTMcFormats);
+ System.out.println("ALAudioSink: hasEXTFloat32 "+hasEXTFloat32);
+ System.out.println("ALAudioSink: hasEXTDouble "+hasEXTDouble);
+ System.out.println("ALAudioSink: hasALC_thread_local_context "+hasALC_thread_local_context);
+ System.out.println("ALAudioSink: preferredAudioFormat "+preferredAudioFormat);
+ System.out.println("ALAudioSink: maxSupportedChannels "+getMaxSupportedChannels());
+ clearPreALError("init."+checkErrIter++);
+ }
+
+ // Create source
+ {
+ al.alGenSources(1, alSource, 0);
+ final int err = al.alGetError();
+ if( ALConstants.AL_NO_ERROR != err ) {
+ alSource[0] = -1;
+ throw new RuntimeException(getThreadName()+": ALAudioSink: Error generating Source: 0x"+Integer.toHexString(err));
+ }
+ }
+
+ if( DEBUG ) {
+ System.err.println("ALAudioSink: Using device: " + deviceSpecifier);
+ }
+ available = true;
+ } finally {
+ unlockContext();
+ }
+ return;
+ } catch ( final Exception e ) {
+ if( DEBUG ) {
+ System.err.println(e.getMessage());
+ e.printStackTrace();
+ }
+ destroy();
+ }
+ }
+ }
+
+ private final int querySampleRate() {
+ final int sampleRate;
+ final int[] value = new int[1];
+ alc.alcGetIntegerv(device, ALCConstants.ALC_FREQUENCY, 1, value, 0);
+ final int alcErr = alc.alcGetError(device);
+ final int alErr = al.alGetError();
+ if ( ALCConstants.ALC_NO_ERROR == alcErr && ALConstants.AL_NO_ERROR == alErr && 0 != value[0] ) {
+ sampleRate = value[0];
+ } else {
+ sampleRate = DefaultFormat.sampleRate;
+ }
+ if( DEBUG ) {
+ System.err.println("ALAudioSink.querySampleRate: err [alc "+toHexString(alcErr)+", al "+toHexString(alErr)+"], freq: "+value[0]+" -> "+sampleRate);
+ }
+ return sampleRate;
+ }
+
+ // Expose AudioSink OpenAL implementation specifics
+
+ /** Return OpenAL global {@link AL}. */
+ public static final AL getAL() { return al; }
+ /** Return OpenAL global {@link ALC}. */
+ public static final ALC getALC() { return alc; }
+ /** Return OpenAL global {@link ALExt}. */
+ public static final ALExt getALExt() { return alExt; }
+
+ /** Return this instance's OpenAL {@link ALCdevice}. */
+ public final ALCdevice getDevice() { return device; }
+ /** Return this instance's OpenAL {@link ALCdevice} specifier. */
+ public final String getDeviceSpec() { return deviceSpecifier; }
+ /** Return this instance's OpenAL {@link ALCcontext}. */
+ public final ALCcontext getALContext() { return context; }
+ /** Return this instance's OpenAL source ID. */
+ public final int getALSource() { return alSource[0]; }
+
+ /** Return whether OpenAL extension <code>AL_SOFT_buffer_samples</code> is available. */
+ public final boolean hasSOFTBufferSamples() { return hasSOFTBufferSamples; }
+ /** Return whether OpenAL extension <code>AL_EXT_MCFORMATS</code> is available. */
+ public final boolean hasEXTMcFormats() { return hasEXTMcFormats; }
+ /** Return whether OpenAL extension <code>AL_EXT_FLOAT32</code> is available. */
+ public final boolean hasEXTFloat32() { return hasEXTFloat32; }
+ /** Return whether OpenAL extension <code>AL_EXT_DOUBLE</code> is available. */
+ public final boolean hasEXTDouble() { return hasEXTDouble; }
+ /** Return whether OpenAL extension <code>ALC_EXT_thread_local_context</code> is available. */
+ public final boolean hasALCThreadLocalContext() { return hasALC_thread_local_context; }
+
+ /** Return this instance's OpenAL channel layout, set after {@link #init(AudioFormat, float, int, int, int)}. */
+ public final int getALChannelLayout() { return alChannelLayout; }
+ /** Return this instance's OpenAL sample type, set after {@link #init(AudioFormat, float, int, int, int)}. */
+ public final int getALSampleType() { return alSampleType; }
+ /** Return this instance's OpenAL format, set after {@link #init(AudioFormat, float, int, int, int)}. */
+ public final int getALFormat() { return alFormat; }
+
+ // AudioSink implementation ...
+
+ @Override
+ public final void lockExclusive() {
+ lockContext();
+ exclusiveThread = Thread.currentThread();
+ }
+ @Override
+ public final void unlockExclusive() {
+ exclusiveThread = null;
+ unlockContext();
+ }
+ private final void lockContext() {
+ if( null != exclusiveThread ) {
+ if( Thread.currentThread() == exclusiveThread ) {
+ return;
+ }
+ throw new IllegalStateException("Exclusive lock by "+exclusiveThread+", but current is "+Thread.currentThread());
+ }
+ lock.lock();
+ if( hasALC_thread_local_context ) {
+ alExt.alcSetThreadContext(context);
+ } else {
+ alc.alcMakeContextCurrent(context);
+ }
+ final int alcErr = alc.alcGetError(null);
+ if( ALCConstants.ALC_NO_ERROR != alcErr ) {
+ final String err = getThreadName()+": ALCError "+toHexString(alcErr)+" while makeCurrent. "+this;
+ System.err.println(err);
+ ExceptionUtils.dumpStack(System.err);
+ lock.unlock();
+ throw new RuntimeException(err);
+ }
+ final int alErr = al.alGetError();
+ if( ALCConstants.ALC_NO_ERROR != alErr ) {
+ if( DEBUG ) {
+ System.err.println(getThreadName()+": Prev - ALError "+toHexString(alErr)+" @ makeCurrent. "+this);
+ ExceptionUtils.dumpStack(System.err);
+ }
+ }
+ }
+ private final void unlockContext() {
+ if( null != exclusiveThread ) {
+ if( Thread.currentThread() == exclusiveThread ) {
+ return;
+ }
+ throw new IllegalStateException("Exclusive lock by "+exclusiveThread+", but current is "+Thread.currentThread());
+ }
+ if( hasALC_thread_local_context ) {
+ alExt.alcSetThreadContext(null);
+ } else {
+ alc.alcMakeContextCurrent(null);
+ }
+ lock.unlock();
+ }
+ private final void destroyContext() {
+ lock.lock();
+ try {
+ if( null != context ) {
+ try {
+ alc.alcDestroyContext(context);
+ } catch (final Throwable t) {
+ if( DEBUG ) {
+ ExceptionUtils.dumpThrowable("", t);
+ }
+ }
+ context = null;
+ }
+ // unroll lock !
+ while(lock.getHoldCount() > 1) {
+ lock.unlock();
+ }
+ } finally {
+ lock.unlock();
+ }
+ }
+
+ @Override
+ public final String toString() {
+ final int alSrcName = null != alSource ? alSource[0] : 0;
+ final int alBuffersLen = null != alBufferNames ? alBufferNames.length : 0;
+ final int ctxHash = context != null ? context.hashCode() : 0;
+ final int alFramesAvailSize = alFramesAvail != null ? alFramesAvail.size() : 0;
+ final int alFramesPlayingSize = alFramesPlaying != null ? alFramesPlaying.size() : 0;
+ return "ALAudioSink[avail "+available+", playRequested "+playRequested+", device "+deviceSpecifier+", ctx "+toHexString(ctxHash)+", alSource "+alSrcName+
+ ", chosen "+chosenFormat+
+ ", al[chan "+ALHelpers.alChannelLayoutName(alChannelLayout)+", type "+ALHelpers.alSampleTypeName(alSampleType)+
+ ", fmt "+toHexString(alFormat)+", tlc "+hasALC_thread_local_context+", soft "+hasSOFTBufferSamples+
+ "], playSpeed "+playSpeed+", buffers[total "+alBuffersLen+", avail "+alFramesAvailSize+", "+
+ "queued["+alFramesPlayingSize+", apts "+getPTS()+", "+getQueuedTime() + " ms, " + alBufferBytesQueued+" bytes], "+
+ "queue[g "+frameGrowAmount+", l "+frameLimit+"]";
+ }
+
+ private final String shortString() {
+ final int alSrcName = null != alSource ? alSource[0] : 0;
+ final int ctxHash = context != null ? context.hashCode() : 0;
+ return "[ctx "+toHexString(ctxHash)+", playReq "+playRequested+", alSrc "+alSrcName+
+ ", queued["+alFramesPlaying.size()+", " + alBufferBytesQueued+" bytes], "+
+ "queue[g "+frameGrowAmount+", l "+frameLimit+"]";
+ }
+
+ public final String getPerfString() {
+ final int alBuffersLen = null != alBufferNames ? alBufferNames.length : 0;
+ return "Play [buffer "+alFramesPlaying.size()+"/"+alBuffersLen+", apts "+getPTS()+", "+getQueuedTime() + " ms, " + alBufferBytesQueued+" bytes]";
+ }
+
+ @Override
+ public int getPreferredSampleRate() {
+ return preferredSampleRate;
+ }
+
+ @Override
+ public final AudioFormat getPreferredFormat() {
+ if( !staticAvailable ) {
+ return null;
+ }
+ return preferredAudioFormat;
+ }
+
+ @Override
+ public final int getMaxSupportedChannels() {
+ if( !staticAvailable ) {
+ return 0;
+ }
+ if( hasEXTMcFormats || hasSOFTBufferSamples ) {
+ return 8;
+ } else {
+ return 2;
+ }
+ }
+
+ @Override
+ public final boolean isSupported(final AudioFormat format) {
+ if( !staticAvailable ) {
+ return false;
+ }
+ if( format.planar || !format.littleEndian ) {
+ if( DEBUG ) {
+ System.err.println(getThreadName()+": ALAudioSink.isSupported: NO.0 "+format);
+ }
+ return false;
+ }
+ final int alFormat = ALHelpers.getALFormat(format, al, alExt,
+ hasSOFTBufferSamples, hasEXTMcFormats,
+ hasEXTFloat32, hasEXTDouble);
+ if( ALConstants.AL_NONE != alFormat ) {
+ if( DEBUG ) {
+ System.err.println(getThreadName()+": ALAudioSink.isSupported: OK "+format+", alFormat "+toHexString(alFormat));
+ }
+ return true;
+ } else {
+ if( DEBUG ) {
+ System.err.println(getThreadName()+": ALAudioSink.isSupported: NO.1 "+format);
+ }
+ return false;
+ }
+ }
+
+ @Override
+ public final boolean init(final AudioFormat requestedFormat, final float frameDuration, final int initialQueueSize, final int queueGrowAmount, final int queueLimit) {
+ if( !staticAvailable ) {
+ return false;
+ }
+ final int alChannelLayout = ALHelpers.getDefaultALChannelLayout(requestedFormat.channelCount);
+ final int alSampleType = ALHelpers.getALSampleType(requestedFormat.sampleSize, requestedFormat.signed, requestedFormat.fixedP);
+ final int alFormat;
+ if( ALConstants.AL_NONE != alChannelLayout && ALConstants.AL_NONE != alSampleType ) {
+ alFormat = ALHelpers.getALFormat(alChannelLayout, alSampleType, al, alExt,
+ hasSOFTBufferSamples, hasEXTMcFormats,
+ hasEXTFloat32, hasEXTDouble);
+ } else {
+ alFormat = ALConstants.AL_NONE;
+ }
+ if( ALConstants.AL_NONE == alFormat ) {
+ // not supported
+ if( DEBUG ) {
+ System.err.println(getThreadName()+": ALAudioSink.init1: Not supported: "+requestedFormat+", "+toString());
+ }
+ return false;
+ }
+ return initImpl(requestedFormat, alChannelLayout, alSampleType, alFormat,
+ frameDuration, initialQueueSize, queueGrowAmount, queueLimit);
+ }
+
+ /**
+ * Initializes the sink using the given OpenAL audio parameter and streaming details.
+ * @param alChannelLayout OpenAL channel layout
+ * @param alSampleType OpenAL sample type
+ * @param alFormat OpenAL format
+ * @param sampleRate sample rate, e.g. 44100
+ * @param sampleSize sample size in bits, e.g. 16
+ * @param frameDuration average or fixed frame duration in milliseconds
+ * helping a caching {@link AudioFrame} based implementation to determine the frame count in the queue.
+ * See {@link #DefaultFrameDuration}.
+ * @param initialQueueSize initial time in milliseconds to queue in this sink, see {@link #DefaultInitialQueueSize}.
+ * @param queueGrowAmount time in milliseconds to grow queue if full, see {@link #DefaultQueueGrowAmount}.
+ * @param queueLimit maximum time in milliseconds the queue can hold (and grow), see {@link #DefaultQueueLimitWithVideo} and {@link #DefaultQueueLimitAudioOnly}.
+ * @return true if successful, otherwise false
+ * @see ALHelpers#getAudioFormat(int, int, int, int, int)
+ * @see #init(AudioFormat, float, int, int, int)
+ */
+ public final boolean init(final int alChannelLayout, final int alSampleType, final int alFormat,
+ final int sampleRate, final int sampleSize,
+ final float frameDuration, final int initialQueueSize, final int queueGrowAmount, final int queueLimit) {
+ final AudioFormat requestedFormat = ALHelpers.getAudioFormat(alChannelLayout, alSampleType, alFormat, sampleRate, sampleSize);
+ if( null == requestedFormat ) {
+ if( DEBUG ) {
+ System.err.println(getThreadName()+": ALAudioSink.init2: Invalid AL channelLayout "+toHexString(alChannelLayout)+
+ ", sampleType "+toHexString(alSampleType)+", format "+toHexString(alFormat)+" or sample[rate "+sampleRate+", size "+sampleSize+"]; "+toString());
+ }
+ return false;
+ }
+ return initImpl(requestedFormat, alChannelLayout, alSampleType, alFormat,
+ frameDuration, initialQueueSize, queueGrowAmount, queueLimit);
+ }
+
+ private final boolean initImpl(final AudioFormat requestedFormat,
+ final int alChannelLayout, final int alSampleType, final int alFormat,
+ final float frameDuration, final int initialQueueSize, final int queueGrowAmount, final int queueLimit) {
+ this.alChannelLayout = alChannelLayout;
+ this.alSampleType = alSampleType;
+ this.alFormat = alFormat;
+
+ lockContext();
+ try {
+ // Allocate buffers
+ destroyBuffers();
+ {
+ final float useFrameDuration = frameDuration > 1f ? frameDuration : AudioSink.DefaultFrameDuration;
+ avgFrameDuration = (int) useFrameDuration;
+ final int initialFrameCount = requestedFormat.getFrameCount(
+ initialQueueSize > 0 ? initialQueueSize : AudioSink.DefaultInitialQueueSize, useFrameDuration);
+ // frameDuration, int initialQueueSize, int queueGrowAmount, int queueLimit) {
+ alBufferNames = new int[initialFrameCount];
+ al.alGenBuffers(initialFrameCount, alBufferNames, 0);
+ final int err = al.alGetError();
+ if( ALConstants.AL_NO_ERROR != err ) {
+ alBufferNames = null;
+ throw new RuntimeException(getThreadName()+": ALAudioSink: Error generating Buffers: 0x"+Integer.toHexString(err));
+ }
+ final ALAudioFrame[] alFrames = new ALAudioFrame[initialFrameCount];
+ for(int i=0; i<initialFrameCount; i++) {
+ alFrames[i] = new ALAudioFrame(alBufferNames[i]);
+ }
+
+ alFramesAvail = new LFRingbuffer<ALAudioFrame>(alFrames);
+ alFramesPlaying = new LFRingbuffer<ALAudioFrame>(ALAudioFrame[].class, initialFrameCount);
+ this.frameGrowAmount = requestedFormat.getFrameCount(
+ queueGrowAmount > 0 ? queueGrowAmount : AudioSink.DefaultQueueGrowAmount, useFrameDuration);
+ this.frameLimit = requestedFormat.getFrameCount(
+ queueLimit > 0 ? queueLimit : AudioSink.DefaultQueueLimitWithVideo, useFrameDuration);
+ if( DEBUG_TRACE ) {
+ alFramesAvail.dump(System.err, "Avail-init");
+ alFramesPlaying.dump(System.err, "Playi-init");
+ }
+ }
+ } finally {
+ unlockContext();
+ }
+
+ chosenFormat = requestedFormat;
+ if( DEBUG ) {
+ System.err.println(getThreadName()+": ALAudioSink.init: OK "+requestedFormat+", "+toString());
+ }
+ return true;
+ }
+
+ @Override
+ public final AudioFormat getChosenFormat() {
+ return chosenFormat;
+ }
+
+ private static int[] concat(final int[] first, final int[] second) {
+ final int[] result = Arrays.copyOf(first, first.length + second.length);
+ System.arraycopy(second, 0, result, first.length, second.length);
+ return result;
+ }
+ /**
+ private static <T> T[] concat(T[] first, T[] second) {
+ final T[] result = Arrays.copyOf(first, first.length + second.length);
+ System.arraycopy(second, 0, result, first.length, second.length);
+ return result;
+ } */
+
+ private boolean growBuffers() {
+ if( !alFramesAvail.isEmpty() || !alFramesPlaying.isFull() ) {
+ throw new InternalError("Buffers: Avail is !empty "+alFramesAvail+" or Playing is !full "+alFramesPlaying);
+ }
+ if( alFramesAvail.capacity() >= frameLimit || alFramesPlaying.capacity() >= frameLimit ) {
+ if( DEBUG ) {
+ System.err.println(getThreadName()+": ALAudioSink.growBuffers: Frame limit "+frameLimit+" reached: Avail "+alFramesAvail+", Playing "+alFramesPlaying);
+ }
+ return false;
+ }
+
+ final int[] newALBufferNames = new int[frameGrowAmount];
+ al.alGenBuffers(frameGrowAmount, newALBufferNames, 0);
+ final int err = al.alGetError();
+ if( ALConstants.AL_NO_ERROR != err ) {
+ if( DEBUG ) {
+ System.err.println(getThreadName()+": ALAudioSink.growBuffers: Error generating "+frameGrowAmount+" new Buffers: 0x"+Integer.toHexString(err));
+ }
+ return false;
+ }
+ alBufferNames = concat(alBufferNames, newALBufferNames);
+
+ final ALAudioFrame[] newALBuffers = new ALAudioFrame[frameGrowAmount];
+ for(int i=0; i<frameGrowAmount; i++) {
+ newALBuffers[i] = new ALAudioFrame(newALBufferNames[i]);
+ }
+ // alFrames = concat(alFrames , newALBuffers);
+
+ alFramesAvail.growEmptyBuffer(newALBuffers);
+ alFramesPlaying.growFullBuffer(frameGrowAmount);
+ if( alFramesAvail.isEmpty() || alFramesPlaying.isFull() ) {
+ throw new InternalError("Buffers: Avail is empty "+alFramesAvail+" or Playing is full "+alFramesPlaying);
+ }
+ if( DEBUG ) {
+ System.err.println(getThreadName()+": ALAudioSink: Buffer grown "+frameGrowAmount+": Avail "+alFramesAvail+", playing "+alFramesPlaying);
+ }
+ if( DEBUG_TRACE ) {
+ alFramesAvail.dump(System.err, "Avail-grow");
+ alFramesPlaying.dump(System.err, "Playi-grow");
+ }
+ return true;
+ }
+
+ private void destroyBuffers() {
+ if( !staticAvailable ) {
+ return;
+ }
+ if( null != alBufferNames ) {
+ try {
+ al.alDeleteBuffers(alBufferNames.length, alBufferNames, 0);
+ } catch (final Throwable t) {
+ if( DEBUG ) {
+ System.err.println("Caught "+t.getClass().getName()+": "+t.getMessage());
+ t.printStackTrace();
+ }
+ }
+ alFramesAvail.clear();
+ alFramesAvail = null;
+ alFramesPlaying.clear();
+ alFramesPlaying = null;
+ alBufferBytesQueued = 0;
+ // alFrames = null;
+ alBufferNames = null;
+ }
+ }
+
+ @Override
+ public final void destroy() {
+ available = false;
+ if( !staticAvailable ) {
+ return;
+ }
+ if( null != context ) {
+ lockContext();
+ }
+ try {
+ stopImpl(true);
+ if( null != alSource ) {
+ try {
+ al.alDeleteSources(1, alSource, 0);
+ } catch (final Throwable t) {
+ if( DEBUG ) {
+ System.err.println("Caught "+t.getClass().getName()+": "+t.getMessage());
+ t.printStackTrace();
+ }
+ }
+ alSource[0] = -1;
+ }
+
+ destroyBuffers();
+ } finally {
+ destroyContext();
+ }
+ if( null != device ) {
+ try {
+ alc.alcCloseDevice(device);
+ } catch (final Throwable t) {
+ if( DEBUG ) {
+ System.err.println("Caught "+t.getClass().getName()+": "+t.getMessage());
+ t.printStackTrace();
+ }
+ }
+ device = null;
+ }
+ chosenFormat = null;
+ }
+
+ @Override
+ public final boolean isAvailable() {
+ return available;
+ }
+
+ /**
+ * Dequeuing playing audio frames.
+ * @param wait if true, waiting for completion of audio buffers
+ * @param ignoreBufferInconsistency
+ * @return dequeued buffer count
+ */
+ private final int dequeueBuffer(final boolean wait, final boolean ignoreBufferInconsistency) {
+ int alErr = ALConstants.AL_NO_ERROR;
+ final int releaseBufferCount;
+ if( alBufferBytesQueued > 0 ) {
+ final int releaseBufferLimes = Math.max(1, alFramesPlaying.size() / 4 );
+ final int[] val=new int[1];
+ final int avgBufferDura = chosenFormat.getBytesDuration( alBufferBytesQueued / alFramesPlaying.size() );
+ final int sleepLimes = releaseBufferLimes * avgBufferDura;
+ int i=0;
+ int slept = 0;
+ int releasedBuffers = 0;
+ final long t0 = DEBUG ? Clock.currentNanos() : 0;
+ do {
+ val[0] = 0;
+ al.alGetSourcei(alSource[0], ALConstants.AL_BUFFERS_PROCESSED, val, 0);
+ alErr = al.alGetError();
+ if( ALConstants.AL_NO_ERROR != alErr ) {
+ throw new RuntimeException(getThreadName()+": ALError "+toHexString(alErr)+" while quering processed buffers at source. "+this);
+ }
+ releasedBuffers += val[0];
+ if( wait && releasedBuffers < releaseBufferLimes ) {
+ i++;
+ // clip wait at [avgFrameDuration .. 100] ms
+ final int sleep = Math.max(avgFrameDuration, Math.min(100, releaseBufferLimes-releasedBuffers * avgBufferDura));
+ if( slept + sleep <= sleepLimes ) {
+ if( DEBUG ) {
+ System.err.println(getThreadName()+": ALAudioSink: Dequeue.wait-sleep["+i+"]: avgBufferDura "+avgBufferDura+", releaseBuffers "+releasedBuffers+"/"+releaseBufferLimes+", sleep "+sleep+"/"+slept+"/"+sleepLimes+" ms, playImpl "+(ALConstants.AL_PLAYING == getSourceState(false))+", processed "+val[0]+", "+shortString());
+ }
+ unlockContext();
+ try {
+ Thread.sleep( sleep );
+ slept += sleep;
+ } catch (final InterruptedException e) {
+ } finally {
+ lockContext();
+ }
+ } else {
+ // Empirical best behavior w/ openal-soft (sort of needs min ~21ms to complete processing a buffer even if period < 20ms?)
+ unlockContext();
+ try {
+ Thread.sleep( 1 );
+ slept += 1;
+ } catch (final InterruptedException e) {
+ } finally {
+ lockContext();
+ }
+ }
+ }
+ } while ( wait && releasedBuffers < releaseBufferLimes && alBufferBytesQueued > 0 );
+ releaseBufferCount = releasedBuffers;
+ if( DEBUG ) {
+ final long t1 = Clock.currentNanos();
+ System.err.println(getThreadName()+": ALAudioSink: Dequeue.wait-done["+i+"]: "+TimeUnit.NANOSECONDS.toMillis(t1-t0)+" ms, avgBufferDura "+avgBufferDura+", releaseBuffers "+releaseBufferCount+"/"+releaseBufferLimes+", slept "+slept+" ms, playImpl "+(ALConstants.AL_PLAYING == getSourceState(false))+", processed "+val[0]+", "+shortString());
+ }
+ } else {
+ releaseBufferCount = 0;
+ }
+
+ if( releaseBufferCount > 0 ) {
+ final int[] buffers = new int[releaseBufferCount];
+ al.alSourceUnqueueBuffers(alSource[0], releaseBufferCount, buffers, 0);
+ alErr = al.alGetError();
+ if( ALConstants.AL_NO_ERROR != alErr ) {
+ throw new RuntimeException(getThreadName()+": ALError "+toHexString(alErr)+" while dequeueing "+releaseBufferCount+" buffers. "+this);
+ }
+ for ( int i=0; i<releaseBufferCount; i++ ) {
+ final ALAudioFrame releasedBuffer = alFramesPlaying.get();
+ if( null == releasedBuffer ) {
+ if( !ignoreBufferInconsistency ) {
+ throw new InternalError("Internal Error: "+this);
+ }
+ } else {
+ if(DEBUG_TRACE) {
+ System.err.println("< [al "+buffers[i]+", q "+releasedBuffer.alBuffer+"] <- "+shortString()+" @ "+getThreadName());
+ }
+ if( releasedBuffer.alBuffer != buffers[i] ) {
+ if( !ignoreBufferInconsistency ) {
+ alFramesAvail.dump(System.err, "Avail-deq02-post");
+ alFramesPlaying.dump(System.err, "Playi-deq02-post");
+ throw new InternalError("Buffer name mismatch: dequeued: "+buffers[i]+", released "+releasedBuffer+", "+this);
+ }
+ }
+ alBufferBytesQueued -= releasedBuffer.getByteSize();
+ if( !alFramesAvail.put(releasedBuffer) ) {
+ throw new InternalError("Internal Error: "+this);
+ }
+ if(DEBUG_TRACE) {
+ System.err.println("<< [al "+buffers[i]+", q "+releasedBuffer.alBuffer+"] <- "+shortString()+" @ "+getThreadName());
+ }
+ }
+ }
+ }
+ return releaseBufferCount;
+ }
+ private final void dequeueForceAll() {
+ if(DEBUG_TRACE) {
+ System.err.println("< _FLUSH_ <- "+shortString()+" @ "+getThreadName());
+ }
+ final int[] val=new int[1];
+ al.alSourcei(alSource[0], ALConstants.AL_BUFFER, 0); // explicit force zero buffer!
+ if(DEBUG_TRACE) {
+ al.alGetSourcei(alSource[0], ALConstants.AL_BUFFERS_PROCESSED, val, 0);
+ }
+ final int alErr = al.alGetError();
+ while ( !alFramesPlaying.isEmpty() ) {
+ final ALAudioFrame releasedBuffer = alFramesPlaying.get();
+ if( null == releasedBuffer ) {
+ throw new InternalError("Internal Error: "+this);
+ }
+ alBufferBytesQueued -= releasedBuffer.getByteSize();
+ if( !alFramesAvail.put(releasedBuffer) ) {
+ throw new InternalError("Internal Error: "+this);
+ }
+ }
+ alBufferBytesQueued = 0;
+ if(DEBUG_TRACE) {
+ System.err.println("<< _FLUSH_ [al "+val[0]+", err "+toHexString(alErr)+"] <- "+shortString()+" @ "+getThreadName());
+ ExceptionUtils.dumpStack(System.err);
+ }
+ }
+
+ /**
+ * Dequeuing playing audio frames.
+ * @param wait if true, waiting for completion of audio buffers
+ * @param inPTS
+ * @param inDuration
+ * @return dequeued buffer count
+ */
+ private final int dequeueBuffer(final boolean wait, final int inPTS, final int inDuration) {
+ final int dequeuedBufferCount = dequeueBuffer( wait, false /* ignoreBufferInconsistency */ );
+ final ALAudioFrame currentBuffer = alFramesPlaying.peek();
+ if( null != currentBuffer ) {
+ playingPTS = currentBuffer.getPTS();
+ } else {
+ playingPTS = inPTS;
+ }
+ if( DEBUG ) {
+ if( dequeuedBufferCount > 0 ) {
+ System.err.println(getThreadName()+": ALAudioSink: Write "+inPTS+", "+inDuration+" ms, dequeued "+dequeuedBufferCount+", wait "+wait+", "+getPerfString());
+ }
+ }
+ return dequeuedBufferCount;
+ }
+
+ @Override
+ public final AudioFrame enqueueData(final int pts, final ByteBuffer bytes, final int byteCount) {
+ if( !available || null == chosenFormat ) {
+ return null;
+ }
+ final ALAudioFrame alFrame;
+
+ // OpenAL consumes buffers in the background
+ // we first need to initialize the OpenAL buffers then
+ // start continuous playback.
+ lockContext();
+ try {
+ final int duration = chosenFormat.getBytesDuration(byteCount);
+ if( alFramesAvail.isEmpty() ) {
+ // try to dequeue w/o waiting first
+ dequeueBuffer(false, pts, duration);
+ if( alFramesAvail.isEmpty() ) {
+ // try to grow
+ growBuffers();
+ }
+ if( alFramesAvail.isEmpty() && alFramesPlaying.size() > 0 && isPlayingImpl0() ) {
+ // possible if grow failed or already exceeds it's limit - only possible if playing ..
+ dequeueBuffer(true /* wait */, pts, duration);
+ }
+ }
+
+ alFrame = alFramesAvail.get();
+ if( null == alFrame ) {
+ alFramesAvail.dump(System.err, "Avail");
+ throw new InternalError("Internal Error: avail.get null "+alFramesAvail+", "+this);
+ }
+ alFrame.setPTS(pts);
+ alFrame.setDuration(duration);
+ alFrame.setByteSize(byteCount);
+ if( !alFramesPlaying.put( alFrame ) ) {
+ throw new InternalError("Internal Error: "+this);
+ }
+ final int[] alBufferNames = new int[] { alFrame.alBuffer };
+ if( hasSOFTBufferSamples ) {
+ final int samplesPerChannel = chosenFormat.getBytesSampleCount(byteCount) / chosenFormat.channelCount;
+ // final int samplesPerChannel = ALHelpers.bytesToSampleCount(byteCount, alChannelLayout, alSampleType);
+ alExt.alBufferSamplesSOFT(alFrame.alBuffer, chosenFormat.sampleRate, alFormat,
+ samplesPerChannel, alChannelLayout, alSampleType, bytes);
+ } else {
+ al.alBufferData(alFrame.alBuffer, alFormat, bytes, byteCount, chosenFormat.sampleRate);
+ }
+
+ if(DEBUG_TRACE) {
+ System.err.println("> "+alFrame.alBuffer+" -> "+shortString()+" @ "+getThreadName());
+ }
+
+ al.alSourceQueueBuffers(alSource[0], 1, alBufferNames, 0);
+ final int alErr = al.alGetError();
+ if( ALConstants.AL_NO_ERROR != alErr ) {
+ throw new RuntimeException(getThreadName()+": ALError "+toHexString(alErr)+" while queueing buffer "+toHexString(alBufferNames[0])+". "+this);
+ }
+ alBufferBytesQueued += byteCount;
+ enqueuedFrameCount++; // safe: only written-to while locked!
+
+ if(DEBUG_TRACE) {
+ System.err.println(">> "+alFrame.alBuffer+" -> "+shortString()+" @ "+getThreadName());
+ }
+
+ playImpl(); // continue playing, fixes issue where we ran out of enqueued data!
+ } finally {
+ unlockContext();
+ }
+ return alFrame;
+ }
+
+ @Override
+ public final boolean isPlaying() {
+ if( !available || null == chosenFormat ) {
+ return false;
+ }
+ if( playRequested ) {
+ lockContext();
+ try {
+ return isPlayingImpl0();
+ } finally {
+ unlockContext();
+ }
+ } else {
+ return false;
+ }
+ }
+ private final boolean isPlayingImpl0() {
+ if( playRequested ) {
+ return ALConstants.AL_PLAYING == getSourceState(false);
+ } else {
+ return false;
+ }
+ }
+ private final int getSourceState(final boolean ignoreError) {
+ final int[] val = new int[1];
+ al.alGetSourcei(alSource[0], ALConstants.AL_SOURCE_STATE, val, 0);
+ final int alErr = al.alGetError();
+ if( ALConstants.AL_NO_ERROR != alErr ) {
+ final String msg = getThreadName()+": ALError "+toHexString(alErr)+" while querying SOURCE_STATE. "+this;
+ if( ignoreError ) {
+ if( DEBUG ) {
+ System.err.println(msg);
+ }
+ } else {
+ throw new RuntimeException(msg);
+ }
+ }
+ return val[0];
+ }
+
+ @Override
+ public final void play() {
+ if( !available || null == chosenFormat ) {
+ return;
+ }
+ playRequested = true;
+ lockContext();
+ try {
+ playImpl();
+ if( DEBUG ) {
+ System.err.println(getThreadName()+": ALAudioSink: PLAY playImpl "+(ALConstants.AL_PLAYING == getSourceState(false))+", "+this);
+ }
+ } finally {
+ unlockContext();
+ }
+ }
+ private final void playImpl() {
+ if( playRequested && ALConstants.AL_PLAYING != getSourceState(false) ) {
+ al.alSourcePlay(alSource[0]);
+ final int alErr = al.alGetError();
+ if( ALConstants.AL_NO_ERROR != alErr ) {
+ throw new RuntimeException(getThreadName()+": ALError "+toHexString(alErr)+" while start playing. "+this);
+ }
+ }
+ }
+
+ @Override
+ public final void pause() {
+ if( !available || null == chosenFormat ) {
+ return;
+ }
+ if( playRequested ) {
+ lockContext();
+ try {
+ pauseImpl();
+ if( DEBUG ) {
+ System.err.println(getThreadName()+": ALAudioSink: PAUSE playImpl "+(ALConstants.AL_PLAYING == getSourceState(false))+", "+this);
+ }
+ } finally {
+ unlockContext();
+ }
+ }
+ }
+ private final void pauseImpl() {
+ if( isPlayingImpl0() ) {
+ playRequested = false;
+ al.alSourcePause(alSource[0]);
+ final int alErr = al.alGetError();
+ if( ALConstants.AL_NO_ERROR != alErr ) {
+ throw new RuntimeException(getThreadName()+": ALError "+toHexString(alErr)+" while pausing. "+this);
+ }
+ }
+ }
+ private final void stopImpl(final boolean ignoreError) {
+ if( ALConstants.AL_STOPPED != getSourceState(ignoreError) ) {
+ playRequested = false;
+ al.alSourceStop(alSource[0]);
+ final int alErr = al.alGetError();
+ if( ALConstants.AL_NO_ERROR != alErr ) {
+ final String msg = "ALError "+toHexString(alErr)+" while stopping. "+this;
+ if( ignoreError ) {
+ if( DEBUG ) {
+ System.err.println(getThreadName()+": "+msg);
+ }
+ } else {
+ throw new RuntimeException(getThreadName()+": ALError "+toHexString(alErr)+" while stopping. "+this);
+ }
+ }
+ }
+ }
+
+ @Override
+ public final float getPlaySpeed() { return playSpeed; }
+
+ @Override
+ public final boolean setPlaySpeed(float rate) {
+ if( !available || null == chosenFormat ) {
+ return false;
+ }
+ lockContext();
+ try {
+ if( Math.abs(1.0f - rate) < 0.01f ) {
+ rate = 1.0f;
+ }
+ if( 0.5f <= rate && rate <= 2.0f ) { // OpenAL limits
+ playSpeed = rate;
+ al.alSourcef(alSource[0], ALConstants.AL_PITCH, playSpeed);
+ return true;
+ }
+ } finally {
+ unlockContext();
+ }
+ return false;
+ }
+
+ @Override
+ public final float getVolume() {
+ return volume;
+ }
+
+ @Override
+ public final boolean setVolume(float v) {
+ if( !available || null == chosenFormat ) {
+ return false;
+ }
+ lockContext();
+ try {
+ if( Math.abs(v) < 0.01f ) {
+ v = 0.0f;
+ } else if( Math.abs(1.0f - v) < 0.01f ) {
+ v = 1.0f;
+ }
+ if( 0.0f <= v && v <= 1.0f ) { // OpenAL limits
+ volume = v;
+ al.alSourcef(alSource[0], ALConstants.AL_GAIN, v);
+ return true;
+ }
+ } finally {
+ unlockContext();
+ }
+ return false;
+ }
+
+ @Override
+ public final void flush() {
+ if( !available || null == chosenFormat ) {
+ return;
+ }
+ lockContext();
+ try {
+ // pauseImpl();
+ stopImpl(false);
+ // Redundant: dequeueBuffer( false /* wait */, true /* ignoreBufferInconsistency */);
+ dequeueForceAll();
+ if( alBufferNames.length != alFramesAvail.size() || alFramesPlaying.size() != 0 ) {
+ throw new InternalError("XXX: "+this);
+ }
+ if( DEBUG ) {
+ System.err.println(getThreadName()+": ALAudioSink: FLUSH playImpl "+(ALConstants.AL_PLAYING == getSourceState(false))+", "+this);
+ }
+ } finally {
+ unlockContext();
+ }
+ }
+
+ @Override
+ public final int getEnqueuedFrameCount() {
+ return enqueuedFrameCount;
+ }
+
+ @Override
+ public final int getFrameCount() {
+ return null != alBufferNames ? alBufferNames.length : 0;
+ }
+
+ @Override
+ public final int getQueuedFrameCount() {
+ if( !available || null == chosenFormat ) {
+ return 0;
+ }
+ return alFramesPlaying.size();
+ }
+
+ @Override
+ public final int getFreeFrameCount() {
+ if( !available || null == chosenFormat ) {
+ return 0;
+ }
+ return alFramesAvail.size();
+ }
+
+ @Override
+ public final int getQueuedByteCount() {
+ if( !available || null == chosenFormat ) {
+ return 0;
+ }
+ return alBufferBytesQueued;
+ }
+
+ @Override
+ public final int getQueuedTime() {
+ if( !available || null == chosenFormat ) {
+ return 0;
+ }
+ return chosenFormat.getBytesDuration(alBufferBytesQueued);
+ }
+
+ @Override
+ public final int getPTS() { return playingPTS; }
+
+ private static final String toHexString(final int v) { return "0x"+Integer.toHexString(v); }
+ private static final String getThreadName() { return Thread.currentThread().getName(); }
+}
diff --git a/src/java/com/jogamp/openal/util/ALHelpers.java b/src/java/com/jogamp/openal/util/ALHelpers.java
index a666049..65869cc 100644
--- a/src/java/com/jogamp/openal/util/ALHelpers.java
+++ b/src/java/com/jogamp/openal/util/ALHelpers.java
@@ -1,7 +1,8 @@
-/*
+/**
* OpenAL Helpers
*
* Copyright (c) 2011 by Chris Robinson <[email protected]>
+ * Copyright (c) 2013-2023 JogAmp Community
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
@@ -21,53 +22,201 @@
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
-
-/* This file contains routines to help with some menial OpenAL-related tasks,
- * such as opening a device and setting up a context, closing the device and
- * destroying its context, converting between frame counts and byte lengths,
- * finding an appropriate buffer format, and getting readable strings for
- * channel configs and sample types. */
package com.jogamp.openal.util;
import com.jogamp.openal.AL;
import com.jogamp.openal.ALConstants;
import static com.jogamp.openal.ALConstants.*;
import com.jogamp.openal.ALExt;
-// import com.jogamp.openal.ALExtConstants;
import static com.jogamp.openal.ALExtConstants.*;
+import com.jogamp.common.av.AudioFormat;
-/* This file contains routines to help with some menial OpenAL-related tasks,
- * such as converting between frame counts and byte lengths,
- * finding an appropriate buffer format, and getting readable strings for
+
+/* This class contains routines to help with some menial OpenAL-related tasks,
+ * such as finding an audio format and getting readable strings for
* channel configs and sample types. */
public class ALHelpers {
+ /**
+ * [openal-soft >= 1.18.0](https://github.com/kcat/openal-soft/blob/master/ChangeLog)
+ * - Removed support for the AL_SOFT_buffer_samples and AL_SOFT_buffer_sub_data
+ * extensions. Due to conflicts with AL_EXT_SOURCE_RADIUS.
+ */
+ public static final String AL_SOFT_buffer_samples = "AL_SOFT_buffer_samples";
+ public static final String AL_EXT_MCFORMATS = "AL_EXT_MCFORMATS";
+ public static final String AL_EXT_FLOAT32 = "AL_EXT_FLOAT32";
+ public static final String AL_EXT_DOUBLE = "AL_EXT_DOUBLE";
+
+ public static final String ALC_EXT_thread_local_context = "ALC_EXT_thread_local_context";
+
+ /**
+ * Returns a compatible {@link AudioFormat} based on given OpenAL channel-layout, sample-type and format,
+ * as well as the generic sample-rate and sample-size.
+ * <p>
+ * The resulting {@link AudioFormat} uses {@link AudioFormat#planar} = false and {@link AudioFormat#littleEndian} = true.
+ * </p>
+ * @param alChannelLayout OpenAL channel layout
+ * @param alSampleType OpenAL sample type
+ * @param alFormat OpenAL format
+ * @param sampleRate sample rate, e.g. 44100
+ * @param sampleSize sample size in bits, e.g. 16
+ * @return a new {@link AudioFormat} instance or null if parameter are not conclusive or invalid.
+ */
+ public static AudioFormat getAudioFormat(final int alChannelLayout, final int alSampleType, final int alFormat,
+ final int sampleRate, final int sampleSize) {
+ if( ALConstants.AL_NONE == alChannelLayout || ALConstants.AL_NONE == alSampleType ||
+ ALConstants.AL_NONE == alFormat || 0 == sampleRate || 0 == sampleSize ) {
+ return null;
+ }
+ final int channelCount = getALChannelLayoutChannelCount(alChannelLayout);
+ if( 0 == channelCount ) {
+ return null;
+ }
+ final boolean signed = isALSampleTypeSigned(alSampleType);
+ final boolean fixedP = isALSampleTypeFixed(alSampleType);
+ return new AudioFormat(sampleRate, sampleSize, channelCount, signed, fixedP,
+ false /* planar */, true /* littleEndian */);
+ }
/**
- * Returns a compatible AL buffer format given the AL channel layout and
- * AL sample type. If <code>hasSOFTBufferSamples</code> is true,
+ * Returns a compatible AL buffer format given the {@link AudioFormat},
+ * which determines the AL channel layout and AL sample type.
+ * </p>
+ * <p>
+ * If <code>hasEXTMcFormats</code> or <code>hasSOFTBufferSamples</code> is true,
* it will be called to find the closest-matching format from
- * <code>AL_SOFT_buffer_samples</code>.
+ * <code>AL_EXT_MCFORMATS</code> or <code>AL_SOFT_buffer_samples</code>.
+ * </p>
* <p>
* Returns {@link ALConstants#AL_NONE} if no supported format can be found.
* </p>
+ * <p>
+ * Function uses {@link AL#alIsExtensionPresent(String)}, which might be context dependent,
+ * otherwise function is context independent.
+ * </p>
*
- * @param alChannelLayout AL channel layout, see {@link #getDefaultALChannelLayout(int)}
- * @param alSampleType AL sample type, see {@link #getALSampleType(int, boolean, boolean)}.
+ * @param audioFormat used to derive AL channel layout {@link #getDefaultALChannelLayout(int)}
+ * and AL sample type {@link #getALSampleType(int, boolean, boolean)}
+ * @param al AL instance
+ * @param alExt ALExt instance
+ * @return AL buffer format
+ */
+ public static int getALFormat(final AudioFormat audioFormat,
+ final AL al, final ALExt alExt) {
+ final int alChannelLayout = ALHelpers.getDefaultALChannelLayout(audioFormat.channelCount);
+ final int alSampleType = ALHelpers.getALSampleType(audioFormat.sampleSize, audioFormat.signed, audioFormat.fixedP);
+ if( ALConstants.AL_NONE != alChannelLayout && ALConstants.AL_NONE != alSampleType ) {
+ return ALHelpers.getALFormat(alChannelLayout, alSampleType, al, alExt);
+ } else {
+ return ALConstants.AL_NONE;
+ }
+ }
+
+ /**
+ * Returns a compatible AL buffer format given the {@link AudioFormat},
+ * which determines the AL channel layout and AL sample type.
+ * </p>
+ * <p>
+ * If <code>hasEXTMcFormats</code> or <code>hasSOFTBufferSamples</code> is true,
+ * it will be called to find the closest-matching format from
+ * <code>AL_EXT_MCFORMATS</code> or <code>AL_SOFT_buffer_samples</code>.
+ * </p>
+ * <p>
+ * Returns {@link ALConstants#AL_NONE} if no supported format can be found.
+ * </p>
+ * <p>
+ * Function is context independent.
+ * </p>
+ *
+ * @param audioFormat used to derive AL channel layout {@link #getDefaultALChannelLayout(int)}
+ * and AL sample type {@link #getALSampleType(int, boolean, boolean)}
+ * @param al AL instance
+ * @param alExt ALExt instance
* @param hasSOFTBufferSamples true if having extension <code>AL_SOFT_buffer_samples</code>, otherwise false
* @param hasEXTMcFormats true if having extension <code>AL_EXT_MCFORMATS</code>, otherwise false
* @param hasEXTFloat32 true if having extension <code>AL_EXT_FLOAT32</code>, otherwise false
* @param hasEXTDouble true if having extension <code>AL_EXT_DOUBLE</code>, otherwise false
+ * @return AL buffer format
+ */
+ public static int getALFormat(final AudioFormat audioFormat,
+ final AL al, final ALExt alExt,
+ final boolean hasSOFTBufferSamples,
+ final boolean hasEXTMcFormats,
+ final boolean hasEXTFloat32, final boolean hasEXTDouble) {
+ final int alChannelLayout = ALHelpers.getDefaultALChannelLayout(audioFormat.channelCount);
+ final int alSampleType = ALHelpers.getALSampleType(audioFormat.sampleSize, audioFormat.signed, audioFormat.fixedP);
+ final int alFormat;
+ if( ALConstants.AL_NONE != alChannelLayout && ALConstants.AL_NONE != alSampleType ) {
+ alFormat = ALHelpers.getALFormat(alChannelLayout, alSampleType, al, alExt,
+ hasSOFTBufferSamples, hasEXTMcFormats,
+ hasEXTFloat32, hasEXTDouble);
+ } else {
+ alFormat = ALConstants.AL_NONE;
+ }
+ return alFormat;
+ }
+
+ /**
+ * Returns a compatible AL buffer format given the AL channel layout and AL sample type.
+ * <p>
+ * If <code>hasEXTMcFormats</code> or <code>hasSOFTBufferSamples</code> is true,
+ * it will be called to find the closest-matching format from
+ * <code>AL_EXT_MCFORMATS</code> or <code>AL_SOFT_buffer_samples</code>.
+ * </p>
+ * <p>
+ * Returns {@link ALConstants#AL_NONE} if no supported format can be found.
+ * </p>
+ * <p>
+ * Function uses {@link AL#alIsExtensionPresent(String)}, which might be context dependent,
+ * otherwise function is context independent.
+ * </p>
+ *
+ * @param alChannelLayout AL channel layout, see {@link #getDefaultALChannelLayout(int)}
+ * @param alSampleType AL sample type, see {@link #getALSampleType(int, boolean, boolean)}.
* @param al AL instance
* @param alExt ALExt instance
* @return AL buffer format
*/
public static final int getALFormat(final int alChannelLayout, final int alSampleType,
+ final AL al, final ALExt alExt) {
+ final boolean hasSOFTBufferSamples = al.alIsExtensionPresent(AL_SOFT_buffer_samples);
+ final boolean hasEXTMcFormats = al.alIsExtensionPresent(AL_EXT_MCFORMATS);
+ final boolean hasEXTFloat32 = al.alIsExtensionPresent(AL_EXT_FLOAT32);
+ final boolean hasEXTDouble = al.alIsExtensionPresent(AL_EXT_DOUBLE);
+ return ALHelpers.getALFormat(alChannelLayout, alSampleType, al, alExt,
+ hasSOFTBufferSamples, hasEXTMcFormats,
+ hasEXTFloat32, hasEXTDouble);
+ }
+
+ /**
+ * Returns a compatible AL buffer format given the AL channel layout and AL sample type.
+ * <p>
+ * If <code>hasEXTMcFormats</code> or <code>hasSOFTBufferSamples</code> is true,
+ * it will be called to find the closest-matching format from
+ * <code>AL_EXT_MCFORMATS</code> or <code>AL_SOFT_buffer_samples</code>.
+ * </p>
+ * <p>
+ * Returns {@link ALConstants#AL_NONE} if no supported format can be found.
+ * </p>
+ * <p>
+ * Function is context independent.
+ * </p>
+ *
+ * @param alChannelLayout AL channel layout, see {@link #getDefaultALChannelLayout(int)}
+ * @param alSampleType AL sample type, see {@link #getALSampleType(int, boolean, boolean)}.
+ * @param al AL instance
+ * @param alExt ALExt instance
+ * @param hasSOFTBufferSamples true if having extension <code>AL_SOFT_buffer_samples</code>, otherwise false
+ * @param hasEXTMcFormats true if having extension <code>AL_EXT_MCFORMATS</code>, otherwise false
+ * @param hasEXTFloat32 true if having extension <code>AL_EXT_FLOAT32</code>, otherwise false
+ * @param hasEXTDouble true if having extension <code>AL_EXT_DOUBLE</code>, otherwise false
+ * @return AL buffer format
+ */
+ public static final int getALFormat(final int alChannelLayout, final int alSampleType,
+ final AL al, final ALExt alExt,
final boolean hasSOFTBufferSamples,
final boolean hasEXTMcFormats,
- final boolean hasEXTFloat32,
- final boolean hasEXTDouble,
- final AL al, final ALExt alExt) {
+ final boolean hasEXTFloat32, final boolean hasEXTDouble) {
int format = AL_NONE;
/* If using AL_SOFT_buffer_samples, try looking through its formats */
@@ -262,6 +411,22 @@ public class ALHelpers {
}
/**
+ * Returns the channel count of the given AL channel layout
+ */
+ public static final int getALChannelLayoutChannelCount(final int alChannelLayout) {
+ switch(alChannelLayout) {
+ case AL_MONO_SOFT: return 1;
+ case AL_STEREO_SOFT: return 2;
+ case AL_REAR_SOFT: return 2;
+ case AL_QUAD_SOFT: return 4;
+ case AL_5POINT1_SOFT: return 6;
+ case AL_6POINT1_SOFT: return 7;
+ case AL_7POINT1_SOFT: return 8;
+ }
+ return 0;
+ }
+
+ /**
* Returns the AL sample type matching the given audio type attributes, or {@link ALConstants#AL_NONE}.
* @param sampleSize sample size in bits
* @param signed true if signed number, false for unsigned
@@ -311,6 +476,45 @@ public class ALHelpers {
}
/**
+ * Returns whether the given AL sample type is signed
+ */
+ public static final boolean isALSampleTypeSigned(final int alSampleType) {
+ switch(alSampleType) {
+ case AL_BYTE_SOFT:
+ case AL_SHORT_SOFT:
+ case AL_INT_SOFT:
+ case AL_FLOAT_SOFT:
+ case AL_DOUBLE_SOFT:
+ return true;
+ case AL_UNSIGNED_BYTE_SOFT:
+ case AL_UNSIGNED_SHORT_SOFT:
+ case AL_UNSIGNED_INT_SOFT:
+ default:
+ return false;
+ }
+ }
+
+ /**
+ * Returns true if the given AL sample type is a fixed point (byte, short, int, ..)
+ * or false if a floating point type (float, double).
+ */
+ public static final boolean isALSampleTypeFixed(final int alSampleType) {
+ switch(alSampleType) {
+ case AL_BYTE_SOFT:
+ case AL_SHORT_SOFT:
+ case AL_INT_SOFT:
+ case AL_UNSIGNED_BYTE_SOFT:
+ case AL_UNSIGNED_SHORT_SOFT:
+ case AL_UNSIGNED_INT_SOFT:
+ return true;
+ case AL_FLOAT_SOFT:
+ case AL_DOUBLE_SOFT:
+ default:
+ return false;
+ }
+ }
+
+ /**
* Returns the byte size of the given AL sample type
* @throws IllegalArgumentException for unknown <code>alChannelLayout</code> or <code>alSampleType</code> values.
*/