| Commit message (Collapse) | Author | Age | Files | Lines |
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other ALC_EXT names
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last PTS value against System Clock Reference (SCR)
See GlueGen commit 52725b4c6525487f93407f529dc0a758b387a4fc
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updateQueue() dequeues w/o wait 1st, then returns adjusted PTS; Simplify/split waitFroReleaded*(); Use TSPrinter for DEBUG
Returning the time-adjusted PTS from the last dequeued frame seems to be the most accurate
value we can deliver.
Hence we store the Clock.currentMillis() in playing_pts_t0 when updating playing_pts
and add the difference to current Clock.currentMillis() when retrieving.
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JogAmp Version [2.4.0 - 2.5.0]
- Adopt to simplified AudioSink
- Add lastBufferedPTS and expose it
- Cleanup short* and perf*String() trace/debug presentations to simplify review
- Hence drop growBuffers()
- Set initial avgFrameDuration to latency, at least a good start
+++
dequeueBuffer(..):
- Pass releaseBufferCountReq directly, tangible only if wait == true,
have enqueueData(..) determine the wait and releaseBufferCountReq value.
- Drop dequeueBuffer(..) overload caller, simplifying code
- Don't change playingPTS(..) in overload caller, enqueueData(..) takes care of it
- Align DEBUG trace with enqueueData(..) to simplify review
- Otherwise no semnatic change in dequeueBuffer(..)
enqueueData(..):
- Dropped growBuffers()
- Show DEBUG trace before actual dequeueBuffer(..) to have meanigful output
- SOFT (no-wait) dequeueBuffer(..) triggers on 2/3rd full queue
- HARD (wait) dequeueBuffer(..) if queue is full
- Set playingPTS, either use
- old queue-tip (too old) and add (forward) 60% of queue-buffer time
- new queue-tail (too young), subtract (delay) 40% of queue-buffer time
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Actually new GlueGen WorkerThread was created from GLMediaPlayer, which was also the template for this one
and hence lead to generalization to WorkerThread.
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not running
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'sometimes wrong'. Workaround: Query released buffers after receiving event and use minimum.
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hasAL_SOFT_events, 1st disable all events); growBuffers(): No pre-condition exception for hasAL_SOFT_events
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released buffer count instead of polling
With wait == true, we simply wait until enough buffers have arrived,
otherwise take what we got - both w/o polling and querying the alSource.
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GlueGen JavaCallback
https://openal-soft.org/openal-extensions/SOFT_events.txt
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current; Expose hasALC_thread_local_context to avoid double checks
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remove dead code
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a simple sound source to test
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JOAL/OpenAL easier and reuse Context context locking
Context locking logic has been fixed and moved to Sound3D Context class (beside many other transparency changes),
see commit afb386e13fd00fde1401d4551ee4790b1f6d5e09.
This also aligns w/ AudioSink API change of Gluegen commits
- c04726720a57f8db42f2621ad58ff3bd42006c63
- 6a74d16a805a4204093972bb91361b2aa633065c
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ALCcontext recreation, ..
Context
- Recursive context locking (only 1st shall do native makeCurrent, only last shall do native release)
- Access to the current Context instance (thread local storage)
- Obey "One context can only be current on one thread,
and one thread can only have one context current!"
- ALCcontext recreation within lock, allowing to change native OpenAL specifics via attr list
- ALCcontext creation (initial) w/ attr list
Device
- Retrieve name if default name null has been given
- Expose device name
- Allow to open() again
Source
- Allow lazy creation w/ invalid ID
- Allow create() post instantiation (for a single source)
- Throw ALException in all queued buffer methods as they are crucial
in multithreading streaming.
- Add queue buffer with OpenAL buffer-id int[] arrays variant
to be used w/o Buffer
Listener
- Fix (get|set)Orientation() API doc: It's 'at' vector, then 'up' vector.
General:
- Have toString()
- Added versatile AudioSystem3D.check*Error(..)
Earlier Sound3D changes
- 7f73d50c90d05cf7388f23977ca956a4933019ad
- 64b40bd4359cad46ebf62751ea342d80205bd98b
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'avgFrameDuration') to avoid losing precision when dealing with stats, averages etc
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unused field 'avgFrameDuration'.
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accumulative while waiting (not yet dequeueing), ...
Further:
- brackets were missed in 'sleep =', i.e.
'releaseBufferLimes-releasedBuffers * avgBufferDura' -> '(releaseBufferLimes-releasedBuffers) * avgBufferDura)'
- The minimum sleep of avgFrameDuration 'sleep = Math.max(avgFrameDuration, ..'
lead to cut-off smaller sleep cycles and the else branch would only sleep for less (1ms) multiple times.
Hence use the minimum of 2ms, where we subtract 1ms for busy polling.
Notable, this is an extreme situation of small buffer sizes (duration),
but may happen on like synthesizer applications (jsyn).
We actually could use latency (refresh cycle) as used in OpenAL-Soft,
but this is an undocumented feature .. sort of.
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ALExtConstants directly: Fixes failing al.alGetEnumValue("AL_FORMAT_STEREO_DOUBLE")
al.alGetEnumValue("AL_FORMAT_STEREO_DOUBLE") failed w/ OpenAL-Soft,
despite having AL_EXT_MCFORMATS, AL_EXT_FLOAT32 and AL_EXT_DOUBLE supported.
Notable, al.alGetEnumValue("AL_FORMAT_MONO_FLOAT32") did work.
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using actual OpenAL frame processing refresh value
Also pre-allocate sampleBuffer to 30ms actual format, using AudioFormat.getDurationsByteSize(..)
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representation)
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locking and exit.
initImpl() shall just return false, not throwing an exception.
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(refresh-rate) if frameDuration < defaultLatency
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(mono, stereo) using undocumented queries
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makeContextCurrent()/alcSetThreadContext() fails (returns false)
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`hasALC_thread_local_context`, benefitting from thread-local-context
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mix-and-match; Align Context makeCurrent/release w/ ALAudioSink ...
Align Context makeCurrent/release w/ ALAudioSink
- use ALC_EXT_thread_local_context if available
- use a recursive lock
- same release method as makeCurrent
- ALC_EXT_thread_local_context or traditional
- destroy also releases the context and fully unwinds lock
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context-locking and release same way; Destroy shall also release context.
Result is 'ALSOFT(WW)' free.
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and OpenAL paremeter. Can be 'plugged' into existing OpenAL logic.
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audio-backens per platform and contained openal-soft library
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