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authorSven Gothel <[email protected]>2013-08-24 23:38:42 +0200
committerSven Gothel <[email protected]>2013-08-24 23:38:42 +0200
commit517371b2c200783890e2f6a173748cf65d3c8c91 (patch)
treeca4c43e48762c7f67c13e5fb5175476f4c5de57c /src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java
parentd0e01cb5c0ec3e48b8a9b9b79a7795b214c6e3ea (diff)
AudioSink.init(..) abstract 'frame count' -> duration [ms] allowing non-frame based AudioSink's to deal w/ desired queue sizes.
- Rename AudioSink.initSink(..) -> AudioSink.init(..) - Move: "int initialFrameCount, int frameGrowAmount, int frameLimit" to "int initialQueueSize, int queueGrowAmount, int queueLimit" based on milliseconds instead of frame count. - Passing hint 'float frameDuration' to calculate frame count for fame based audio sink, i.e. ALAudioSink. - Adding sensible static final default values - AudioDataFormat: Add convenient conversion routines (samples/bytes/frame-count) - FFMPEGMediaPlayer: Retrieve audio frame size in samples per channel, pass it to AudioSink.init(..) to properly calculate frame count/limits based on duration.
Diffstat (limited to 'src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java')
-rw-r--r--src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java104
1 files changed, 82 insertions, 22 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 cacf8c5a4..2b8da8af9 100644
--- a/src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java
+++ b/src/jogl/classes/com/jogamp/opengl/util/av/AudioSink.java
@@ -36,6 +36,18 @@ import jogamp.opengl.Debug;
public interface AudioSink {
public static final boolean DEBUG = Debug.debug("AudioSink");
+ /** Default frame duration in millisecond, i.e. 1 frame per {@value} ms. */
+ public static final int DefaultFrameDuration = 32;
+
+ /** Initial audio queue size in milliseconds. {@value} ms, i.e. 16 frames per 32 ms. See {@link #init(AudioDataFormat, float, int, int, int)}.*/
+ public static final int DefaultInitialQueueSize = 16 * 32; // 512 ms
+ /** Audio queue grow size in milliseconds. {@value} ms, i.e. 16 frames per 32 ms. See {@link #init(AudioDataFormat, float, int, int, int)}.*/
+ public static final int DefaultQueueGrowAmount = 16 * 32; // 512 ms
+ /** Audio queue limit w/ video in milliseconds. {@value} ms, i.e. 96 frames per 32 ms. See {@link #init(AudioDataFormat, float, int, int, int)}.*/
+ public static final int DefaultQueueLimitWithVideo = 96 * 32; // 3072 ms
+ /** Audio queue limit w/o video in milliseconds. {@value} ms, i.e. 32 frames per 32 ms. See {@link #init(AudioDataFormat, float, int, int, int)}.*/
+ public static final int DefaultQueueLimitAudioOnly = 32 * 32; // 1024 ms
+
/** Specifies the audio data type. Currently only PCM is supported. */
public static enum AudioDataType { PCM };
@@ -66,21 +78,65 @@ public interface AudioSink {
public final boolean littleEndian;
/**
- * Returns the duration in milliseconds of the given byte count according
- * to {@link #sampleSize}, {@link #channelCount} and {@link #sampleRate}.
+ * Returns the byte size of the given milliseconds
+ * according to {@link #sampleSize}, {@link #channelCount} and {@link #sampleRate}.
*/
- public final int getDuration(int byteCount) {
+ public final int getByteSize(int millisecs) {
final int bytesPerSample = sampleSize >>> 3; // /8
- return byteCount / ( channelCount * bytesPerSample * ( sampleRate / 1000 ) );
+ return millisecs * ( channelCount * bytesPerSample * ( sampleRate / 1000 ) );
}
/**
- * Returns the byte count of the given milliseconds according
- * to {@link #sampleSize}, {@link #channelCount} and {@link #sampleRate}.
+ * Returns the duration in milliseconds of the given byte count
+ * according to {@link #sampleSize}, {@link #channelCount} and {@link #sampleRate}.
*/
- public final int getByteCount(int millisecs) {
+ public final int getBytesDuration(int byteCount) {
final int bytesPerSample = sampleSize >>> 3; // /8
- return millisecs * ( channelCount * bytesPerSample * ( sampleRate / 1000 ) );
+ return byteCount / ( channelCount * bytesPerSample * ( sampleRate / 1000 ) );
+ }
+
+ /**
+ * Returns the duration in milliseconds of the given and sample count per frame and channel
+ * according to the {@link #sampleRate}, i.e.
+ * <pre>
+ * ( 1000f * sampleCount ) / sampleRate
+ * </pre>
+ * @param sampleCount sample count per frame and channel
+ */
+ public final float getSamplesDuration(int sampleCount) {
+ return ( 1000f * (float) sampleCount ) / (float)sampleRate;
+ }
+
+ /**
+ * Returns the rounded frame count of the given milliseconds and frame duration.
+ * <pre>
+ * Math.max( 1, millisecs / frameDuration + 0.5f )
+ * </pre>
+ * <p>
+ * Note: <code>frameDuration</code> can be derived by <i>sample count per frame and channel</i>
+ * via {@link #getSamplesDuration(int)}.
+ * </p>
+ * @param millisecs time in milliseconds
+ * @param frameDuration duration per frame in milliseconds.
+ */
+ public final int getFrameCount(int millisecs, float frameDuration) {
+ return Math.max(1, (int) ( (float)millisecs / frameDuration + 0.5f ));
+ }
+
+ /**
+ * Returns the byte size of given sample count
+ * according to the {@link #sampleSize}, i.e.:
+ * <pre>
+ * sampleCount * ( sampleSize / 8 )
+ * </pre>
+ * <p>
+ * Note: To retrieve the byte size for all channels, you need to pre-multiply <code>sampleCount</code>
+ * with {@link #channelCount}.
+ * </p>
+ * @param sampleCount sample count
+ */
+ public final int getSamplesByteSize(int sampleCount) {
+ return sampleCount * ( sampleSize >>> 3 );
}
public String toString() {
@@ -175,12 +231,16 @@ public interface AudioSink {
* The {@link #DefaultFormat} <i>should be</i> supported by all implementations.
* </p>
* @param requestedFormat the requested {@link AudioDataFormat}.
- * @param initialFrameCount initial number of frames to queue in this sink
- * @param frameGrowAmount number of frames to grow queue if full
- * @param frameLimit maximum number of frames
- * @return if successful the chosen AudioDataFormat based on the <code>requestedFormat</code> and this sinks capabilities, otherwise <code>null</code>.
+ * @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 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 initialFrameCount, int frameGrowAmount, int frameLimit);
+ public AudioDataFormat init(AudioDataFormat requestedFormat, float frameDuration,
+ int initialQueueSize, int queueGrowAmount, int queueLimit);
/**
* Returns true, if {@link #play()} has been requested <i>and</i> the sink is still playing,
@@ -207,7 +267,7 @@ public interface AudioSink {
/**
* Flush all queued buffers, implies {@link #pause()}.
* <p>
- * {@link #initSink(AudioDataFormat, int, int, int)} must be called first.
+ * {@link #init(AudioDataFormat, float, int, int, int)} must be called first.
* </p>
* @see #play()
* @see #pause()
@@ -220,17 +280,17 @@ public interface AudioSink {
/**
* Returns the number of allocated buffers as requested by
- * {@link #initSink(AudioDataFormat, int, int, int)}.
+ * {@link #init(AudioDataFormat, float, int, int, int)}.
*/
public int getFrameCount();
- /** @return the current enqueued frames count since {@link #initSink(AudioDataFormat, int, int, int)}. */
+ /** @return the current enqueued frames count since {@link #init(AudioDataFormat, float, int, int, int)}. */
public int getEnqueuedFrameCount();
/**
* Returns the current number of frames queued for playing.
* <p>
- * {@link #initSink(AudioDataFormat, int, int, int)} must be called first.
+ * {@link #init(AudioDataFormat, float, int, int, int)} must be called first.
* </p>
*/
public int getQueuedFrameCount();
@@ -238,7 +298,7 @@ public interface AudioSink {
/**
* Returns the current number of bytes queued for playing.
* <p>
- * {@link #initSink(AudioDataFormat, int, int, int)} must be called first.
+ * {@link #init(AudioDataFormat, float, int, int, int)} must be called first.
* </p>
*/
public int getQueuedByteCount();
@@ -246,7 +306,7 @@ public interface AudioSink {
/**
* Returns the current queued frame time in milliseconds for playing.
* <p>
- * {@link #initSink(AudioDataFormat, int, int, int)} must be called first.
+ * {@link #init(AudioDataFormat, float, int, int, int)} must be called first.
* </p>
*/
public int getQueuedTime();
@@ -259,7 +319,7 @@ public interface AudioSink {
/**
* Returns the current number of frames in the sink available for writing.
* <p>
- * {@link #initSink(AudioDataFormat, int, int, int)} must be called first.
+ * {@link #init(AudioDataFormat, float, int, int, int)} must be called first.
* </p>
*/
public int getFreeFrameCount();
@@ -270,7 +330,7 @@ public interface AudioSink {
* The data must comply with the chosen {@link AudioDataFormat} as returned by {@link #initSink(AudioDataFormat)}.
* </p>
* <p>
- * {@link #initSink(AudioDataFormat, int, int, int)} must be called first.
+ * {@link #init(AudioDataFormat, float, int, int, int)} must be called first.
* </p>
* @returns the enqueued internal {@link AudioFrame}, which may differ from the input <code>audioDataFrame</code>.
* @deprecated User shall use {@link #enqueueData(int, ByteBuffer, int)}, which allows implementation
@@ -284,7 +344,7 @@ public interface AudioSink {
* The data must comply with the chosen {@link AudioDataFormat} as returned by {@link #initSink(AudioDataFormat)}.
* </p>
* <p>
- * {@link #initSink(AudioDataFormat, int, int, int)} must be called first.
+ * {@link #init(AudioDataFormat, float, int, int, int)} must be called first.
* </p>
* @returns the enqueued internal {@link AudioFrame}.
*/