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270172bcbd91f96d4a38a3d73e23d744f57a25b8) and joal (commit 03f4bb63ce8a358b1c2ef303480e1887d72ecb2e)
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Removed Deprecated Class:
- com/jogamp/opengl/util/TGAWriter.java
- Use TextureIO w/ .tga suffix
- com/jogamp/opengl/util/awt/Screenshot.java
- Use:
- com.jogamp.opengl.util.GLReadBufferUtil, or
- com.jogamp.opengl.util.awt.AWTGLReadBufferUtil
The latter for reading into AWT BufferedImage
See: TestBug461FBOSupersamplingSwingAWT, TestBug605FlippedImageAWT
- javax/media/opengl/GLPbuffer.java
- Use:
caps.setPBuffer(true);
final GLAutoDrawable pbuffer = GLDrawableFactory.getFactory( caps.getGLProfile() ).createOffscreenAutoDrawable(null, caps, null, 512, 512);
- See: TestPBufferDeadlockAWT, ..
Removed Deprecated Methods:
- Constructor of AWT-GLCanvas, SWT-GLCanvas, AWT-GLJPanel
with argument 'final GLContext shareWith'
See GLSharedContextSetter, i.e. glCanvas.setSharedContext(..) !
- GLDrawableFactory.createOffscreenAutoDrawable(..)
with argument 'final GLContext shareWith'
See GLSharedContextSetter, i.e. offscreenAutoDrawable.setSharedContext(..) !
- GLDrawableFactory.createGLPbuffer(..),
see above!
- com.jogamp.opengl.util.av.AudioSink 'enqueueData(AudioDataFrame audioDataFrame)',
use 'enqueueData(int, ByteBuffer, int)'
- GLSharedContextSetter.areAllGLEventListenerInitialized(),
migrated to GLAutoDrawable !
- GLBase's
- glGetBoundBuffer(int), use getBoundBuffer(int)
- glGetBufferSize(int), use getBufferStorage(int).getSize()
- glIsVBOArrayBound(), use isVBOArrayBound()
- glIsVBOElementArrayBound(), use isVBOElementArrayBound()
- NEWT MouseEvent.BUTTON_NUMBER, use BUTTON_COUNT
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c47bc86ae2ee268a1f38c5580d11f93d7f8d6e74)
- Change non static accesses to static members using declaring type
- Change indirect accesses to static members to direct accesses (accesses through subtypes)
- Add final modifier to private fields
- Add final modifier to method parameters
- Add final modifier to local variables
- Remove unnecessary casts
- Remove unnecessary '$NON-NLS$' tags
- Remove trailing white spaces on all lines
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NullAudioSink shall return the last enqueued PTS in getPTS()
not causing a-v delta measure based on lagging audio in player.
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Signed-off-by: Harvey Harrison <[email protected]>
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Signed-off-by: Harvey Harrison <[email protected]>
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adds AL_SOFT_buffer_samples support w/ full AL caps
- Fixe type names:
- Remove AudioDataType, we only support PCM here anyways
- AudioDataFormat -> AudioFormat / Add 'planar' attribute to distingush packed/planar data type
- Validate float types
- Enhance AudioFormat negotiation
- Add 'isSupported(AudioFormat format)' which _shall_ be used before 'init(..)'
to test/negotiate format
- Add getMaxSupportedChannels(), which may be used w/ getPreferredFormat() if orig requested format fails
via 'isSupported(..)'
- 'init(..)' returns boolean only.
- ALAudioSink adds AL_SOFT_buffer_samples support w/ full AL caps
- Determine whether AL_SOFT_buffer_samples is supported
- Use new JOAL ALHelper to convert AudioFormat -> AL-types,
which also answers the 'isSupported(..)' query.
- Now allows multiple: channles, sample-types, etc.
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allowing to change the audio volume.
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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.
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Reuses ALAudioFrames to ease GC, Ringbuffer changes
- Adapt to Ringbuffer changes of GlueGen commit f9f881e59c78e3036cb3f956bc97cfc3197f620d
- Favor AudioSink 'AudioFrame enqueueData(int pts, ByteBuffer bytes, int byteCount)',
- Impl. shall reuse AudioFrame's instead of creating them on the fly
- User shall simply pass the net data required, while receiving an internal AudioFrame
- Add byte/time calc to AudioDataFormat:
- Add getDuration(byteCount) and getByteCount(ms).
- *AudioFrame OO changes:
- abstract AudioFrame extends TimeFrameI
- allow setting of all components to reuse instanced (GC clean)
- ALAudioSink reuses ALAudioFrames to ease GC:
- Remove creating temporary objects to ease GC
- ALAudioFrame holds ALBuffer name, remove ActiveBuffer type.
- Use ALAudioFrame similar to TextureFrame in GLMediaPlayerImpl,
i.e. fill them in 'full' Ringbuffer and move them in-between 'full'/'playing' Ringbuffer.
-
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frameLimit allowing an optional used Ringbuffer to grow in implementation.
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- Use Platform.currentTimeMillis() for accurate timing!
- GLMediaPlayer / GLMediaPlayerImpl
- Add DEBUG_NATIVE property jogl.debug.GLMediaPlayer.Native
for verbose impl. messages, i.e. ffmpeg/libav
- Add 'synchronization' section in GLMediaPlayer API doc (WIP)
- Use passive non-blocking video synchronization,
i.e. repeat frames instead of 'sleep'.
Thx to Xerxes's suggestion.
- Add flushing of cached decoded frames,
allowing to remove complicated 'videoSCR_reset_latch'
- FramePusher (threaded decoding):
- Always create a shared context!
- Release context while pausing
- Pre/post 'getNextTextureImpl()' actions only
at makeCurrent/release.
- newFrameAvailable(..) signal after decoded frame is enqueued
- FFMPEGDynamicLibraryBundleInfo
- Bind add. functions of libavcodec:
+ "av_init_packet",
+ "av_new_packet",
+ "av_destruct_packet",
- Bind add. functions of libavformat:
+ "avformat_seek_file",
+ "av_read_play",
+ "av_read_pause",
- DEBUG property := FFMPEGMediaPlayer.DEBUG || DynamicLibraryBundleInfo.DEBUG;
- FFMPEGMediaPlayer
- Use libavformat's 'av_read_play()' and 'av_read_pause()',
which may get utilized for network streams, e.g. RTSP
- getNextTextureImpl(..):
- Fix retry loop
- Use postNextTextureImpl/preNextTextureImpl if desired (PSM)
- Native:
- Use fixed my_av_q2i32(..) macro (again)
- Use INVALID_PTS marker (synced w/ Java code)
- DEBUG: Dump more detailed frame information
- TODO: Consider passing frame_delay, especially for repeated frames!
- Tests (MovieSimple, MovieCube):
- Refine KeyEvents control for seek and speed.
- TODO:
- Proper audio clock calculation - difficult w/ OpenAL !
- Video / Audio sync:
- seek !
- streams w/ very async A/V frames
- Test Streams:
- Five-minute-sync-test.mp4
- Audio-Video-Sync-Test-Calibration-23.98fps-24fps.mp4
- sound_in_sync_test.mp4
- big_buck_bunny_1080p_surround.avi
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- 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
-
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- AudioSink w/ AudioFrame and formats public
- ALAudioSink uses a circular buffer now, hence relaxes the one-threaded player mode
- FFMPEGMediaPlayer uses multiple audio frames (equal to the ALAudioSink number)
and wraps data to NIO buffer w/o copy.
- FFMPEGMediaPlayer audio threading currently disabled: distorted sound
Seems that the ALAudioSink's circular buffer usage is good enough for now.
- Verbosity only w/ DEBUG flag
- New SyncedRingbuffer for effcient synced buffering
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