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authorSven Gothel <sgothel@jausoft.com>2013-08-14 06:43:42 +0200
committerSven Gothel <sgothel@jausoft.com>2013-08-14 06:43:42 +0200
commitbc3776633ccad81199a96ff8116195133d862395 (patch)
treec5edc7fc7eb8d5ffb3def1efffea44af96f8515a /src/jogl/classes/com/jogamp
parentf53b7713e5eb58a86faf0db06db8be35cfa413d9 (diff)
GLMediaPlayer Multithreaded Decoding: AudioSink (Part-2) - WIP
- AudioSink.AudioDataFormat - add fixedP (fixed-point or floating-point) - AudioSink - rename 'buffer count' to 'frame count' - add setPlaySpeed(..) - add isPlaying() - add play() - add pause() - add flush() - add: getFrameCount(), getQueuedFrameCount(), getFreeFrameCount(), getEnqueuedFrameCount(), - rename: writeData() -> enqueueData(..) - ALAudioSink - multithreaded usage - make ALCcontext current per thread, now required for multithreaded use Use RecursiveLock encapsulating the ALCcontext's makeCurrent/release/destroy, since the native operations seem to be buggy. NOTE: Think about adding these general methods to ALCcontext - implement new methods -
Diffstat (limited to 'src/jogl/classes/com/jogamp')
-rw-r--r--src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java117
-rw-r--r--src/jogl/classes/com/jogamp/opengl/util/av/AudioSinkFactory.java5
2 files changed, 95 insertions, 27 deletions
diff --git a/src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java b/src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java
index 5caeb969a..ba785ac31 100644
--- a/src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java
+++ b/src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java
@@ -41,12 +41,13 @@ public interface AudioSink {
* Specifies the audio data format.
*/
public static class AudioDataFormat {
- public AudioDataFormat(AudioDataType dataType, int sampleRate, int sampleSize, int channelCount, boolean signed, boolean littleEndian) {
+ public AudioDataFormat(AudioDataType dataType, int sampleRate, int sampleSize, int channelCount, boolean signed, boolean fixedP, boolean littleEndian) {
this.dataType = dataType;
this.sampleRate = sampleRate;
this.sampleSize = sampleSize;
this.channelCount = channelCount;
this.signed = signed;
+ this.fixedP = fixedP;
this.littleEndian = littleEndian;
}
/** Audio data type. */
@@ -58,30 +59,32 @@ public interface AudioSink {
/** Number of channels. */
public final int channelCount;
public final boolean signed;
+ /** Fixed or floating point values. Floating point 'float' has {@link #sampleSize} 32, 'double' has {@link #sampleSize} 64, */
+ public final boolean fixedP;
public final boolean littleEndian;
public String toString() {
return "AudioDataFormat[type "+dataType+", sampleRate "+sampleRate+", sampleSize "+sampleSize+", channelCount "+channelCount+
- ", signed "+signed+", "+(littleEndian?"little":"big")+"endian]"; }
+ ", signed "+signed+", fixedP "+fixedP+", "+(littleEndian?"little":"big")+"endian]"; }
}
- /** Default {@link AudioDataFormat}, [type PCM, sampleRate 44100, sampleSize 16, channelCount 2, signed, littleEndian]. */
- public static final AudioDataFormat DefaultFormat = new AudioDataFormat(AudioDataType.PCM, 44100, 16, 2, true /* signed */, true /* littleEndian */);
+ /** Default {@link AudioDataFormat}, [type PCM, sampleRate 44100, sampleSize 16, channelCount 2, signed, fixedP, littleEndian]. */
+ public static final AudioDataFormat DefaultFormat = new AudioDataFormat(AudioDataType.PCM, 44100, 16, 2, true /* signed */, true /* fixed point */, true /* littleEndian */);
public static class AudioFrame {
public final ByteBuffer data;
public final int dataSize;
- public final int audioPTS;
+ public final int pts;
- public AudioFrame(ByteBuffer data, int dataSize, int audioPTS) {
+ public AudioFrame(ByteBuffer data, int dataSize, int pts) {
if( dataSize > data.remaining() ) {
throw new IllegalArgumentException("Give size "+dataSize+" exceeds remaining bytes in ls "+data+". "+this);
}
this.data=data;
this.dataSize=dataSize;
- this.audioPTS=audioPTS;
+ this.pts=pts;
}
- public String toString() { return "AudioFrame[apts "+audioPTS+", data "+data+", payloadSize "+dataSize+"]"; }
+ public String toString() { return "AudioFrame[apts "+pts+", data "+data+", payloadSize "+dataSize+"]"; }
}
/**
@@ -94,6 +97,19 @@ public interface AudioSink {
*/
public boolean isInitialized();
+ /** Returns the playback speed. */
+ public float getPlaySpeed();
+
+ /**
+ * Sets the playback speed.
+ * <p>
+ * Play speed is set to <i>normal</i>, i.e. <code>1.0f</code>
+ * if <code> abs(1.0f - rate) < 0.01f</code> to simplify test.
+ * </p>
+ * @return true if successful, otherwise false, i.e. due to unsupported value range of implementation.
+ */
+ public boolean setPlaySpeed(float s);
+
/**
* Returns the preferred {@link AudioDataFormat} by this sink.
* <p>
@@ -117,52 +133,101 @@ public interface AudioSink {
* The {@link #DefaultFormat} <i>should be</i> supported by all implementations.
* </p>
* @param requestedFormat the requested {@link AudioDataFormat}.
- * @param bufferCount number of buffers for sink
+ * @param frameCount number of frames to queue in this sink
* @return if successful the chosen AudioDataFormat based on the <code>requestedFormat</code> and this sinks capabilities, otherwise <code>null</code>.
*/
- public AudioDataFormat initSink(AudioDataFormat requestedFormat, int bufferCount);
-
+ public AudioDataFormat initSink(AudioDataFormat requestedFormat, int frameCount);
+
+ /**
+ * Returns true, if {@link #play()} has been requested <i>and</i> the sink is still playing,
+ * otherwise false.
+ */
+ public boolean isPlaying();
+
+ /**
+ * Play buffers queued via {@link #enqueueData(AudioFrame)} from current internal position.
+ * If no buffers are yet queued or the queue runs empty, playback is being continued when buffers are enqueued later on.
+ * @see #enqueueData(AudioFrame)
+ * @see #pause()
+ */
+ public void play();
+
+ /**
+ * Pause playing buffers while keeping enqueued data incl. it's internal position.
+ * @see #play()
+ * @see #flush()
+ * @see #enqueueData(AudioFrame)
+ */
+ public void pause();
+
+ /**
+ * Flush all queued buffers, implies {@link #pause()}.
+ * <p>
+ * {@link #initSink(AudioDataFormat, int)} must be called first.
+ * </p>
+ * @see #play()
+ * @see #pause()
+ * @see #enqueueData(AudioFrame)
+ */
+ public void flush();
/** Destroys this instance, i.e. closes all streams and devices allocated. */
public void destroy();
/**
- * Returns the number of bytes queued for playing.
+ * Returns the number of allocated buffers as requested by
+ * {@link #initSink(AudioDataFormat, int)}.
+ */
+ public int getFrameCount();
+
+ /** @return the current enqueued frames count since {@link #initSink(AudioDataFormat, int)}. */
+ public int getEnqueuedFrameCount();
+
+ /**
+ * Returns the current number of frames queued for playing.
+ * <p>
+ * {@link #initSink(AudioDataFormat, int)} must be called first.
+ * </p>
+ */
+ public int getQueuedFrameCount();
+
+ /**
+ * Returns the current number of bytes queued for playing.
* <p>
- * {@link #initSink(AudioDataFormat)} must be called first.
+ * {@link #initSink(AudioDataFormat, int)} must be called first.
* </p>
*/
public int getQueuedByteCount();
/**
- * Returns the queued buffer time in milliseconds for playing.
+ * Returns the current queued frame time in milliseconds for playing.
* <p>
- * {@link #initSink(AudioDataFormat)} must be called first.
+ * {@link #initSink(AudioDataFormat, int)} must be called first.
* </p>
*/
public int getQueuedTime();
/**
- * Returns the number of buffers in the sink available for writing.
- * <p>
- * {@link #initSink(AudioDataFormat)} must be called first.
- * </p>
+ * Return the current audio presentation timestamp (PTS) in milliseconds.
*/
- public int getWritableBufferCount();
+ public int getPTS();
/**
- * Returns true if data is available to be written in the sink.
+ * Returns the current number of frames in the sink available for writing.
* <p>
- * {@link #initSink(AudioDataFormat)} must be called first.
+ * {@link #initSink(AudioDataFormat, int)} must be called first.
* </p>
*/
- public boolean isDataAvailable(int data_size);
-
+ public int getFreeFrameCount();
+
/**
- * Writes the remaining bytes of the given direct ByteBuffer to this sink.
+ * Enqueue the remaining bytes of the given {@link AudioFrame}'s direct ByteBuffer to this sink.
* <p>
* The data must comply with the chosen {@link AudioDataFormat} as returned by {@link #initSink(AudioDataFormat)}.
* </p>
+ * <p>
+ * {@link #initSink(AudioDataFormat, int)} must be called first.
+ * </p>
*/
- public void writeData(AudioFrame audioFrame);
+ public void enqueueData(AudioFrame audioFrame);
}
diff --git a/src/jogl/classes/com/jogamp/opengl/util/av/AudioSinkFactory.java b/src/jogl/classes/com/jogamp/opengl/util/av/AudioSinkFactory.java
index 40321fb6f..a6a14f7dd 100644
--- a/src/jogl/classes/com/jogamp/opengl/util/av/AudioSinkFactory.java
+++ b/src/jogl/classes/com/jogamp/opengl/util/av/AudioSinkFactory.java
@@ -42,10 +42,13 @@ public class AudioSinkFactory {
sink = create(cl, JavaAudioSinkClazzName);
}
if( null == sink ) {
- sink = new NullAudioSink();
+ sink = createNull();
}
return sink;
}
+ public static AudioSink createNull() {
+ return new NullAudioSink();
+ }
public static AudioSink create(final ClassLoader cl, String implName) {
final AudioSink audioSink;