| Commit message (Collapse) | Author | Age | Files | Lines |
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These wakelocks now show up in the client process, where they
might cause wakelock overflows if AudioFlinger threads die often.
The client process should be agnostic of audio flinger threading,
so don't make the wakelock names rely on audioflinger thread names.
Wakelock names now depend only on the audio flinger thread type, and
not the ID itself.
bug: 11200551
Change-Id: Ia1f71b569cb7502d617b77596abc245f461a6ec0
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Bug: 10888816
Change-Id: I84897dd7d30b370640b54e928f230604b873cb68
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If a pause/flush/resume sequence is fast enough, resume is received while
we are still in PAUSING state in which case it is a NOP. If this happens,
flush is still forwarded to the audio HAL but is not preceeded by a pause
which can cause problems to the audio DSP.
It is necessary to preserve the flush as this sequence is typical to a seek.
The fix consists in forcing a pause/resume when a flush request must be
executed and the audio HAL has not been paused previously.
Bug: 11081559.
Change-Id: Ib84ed26d503a61c05933b923ec556b10cedfe140
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Commit e14a5d6 introduced a bug in ThreadBase::acquireWakeLock() where
the wake lock acquisition return code is not initialized before
being checked causing the wake lock token to be sometines not set
and potential failure to release the wake lock later.
Bug: 10985160.
Change-Id: Iffd40e773ae3bcfec3c148a844d5dbebbf474eaf
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Count wake lock use against the app that initiated the recording,
not against the media server.
b/10985160
Change-Id: Iae6e6c030b7f1c6081d9c79725d6990f60dadaaa
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Volume: add a method to wake up the mediaserver playback
thread when a volume command is received on an offloaded track.
Effects: call effect chain process on offloaded playback threads
asynchronously from writes to allow effect state updates while
waiting for async write callback.
Bug: 10796540.
Change-Id: Id2747ae88783575d1d7ffd6fc86fbd054ab2c739
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Passing a sp to parent thread to AsyncCallbackThread() constructor
causes a strong reference to be acquired on the OffloadThread inside
its constructor which causes an early launch of the thread loop
with unpredictable consequences.
Pass a wp to parent thread instead.
Also move the creation of the AsyncCallbackThread to
readOutputParameters() where mUseAsyncWrite is initialized which
makes more sense.
Also change the type of AsyncCallbackThread parent thread to PlaybackThread
instead of OffloadThread to allow a broder use of non blocking write which
in theory is not limited to audio offload use case.
Bug: 8174034.
Change-Id: I4b093b022030cd4f5eb8b8e477333e91098a6549
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Commit 1abbdb4 was not working 100% of the times because
it is possible that the offload thread loop never sleeps
after being created in which case the property is never read.
The loop now reads the property once when starting.
Bug: 10899309.
Change-Id: I2e2ca332f2d7086e59d65f6010378c4d9618ba9e
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Bug: 9587132.
Change-Id: Ie9d5f4cca96306d08bc9a2dbd6edd8953096702d
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It was possible that a resume request signaled by addTrack_l()
while waiting for an async write callback is lost. This is because
mSignalPending was not set and waitingAsyncCallback_l() would pause the
thread loop before executing prepareTracks_l().
The fix consists in using signal_l() method to wake the thread
loop o that mSignalPending is set.
Also make sure that sleepTime is reset to 0 when resuming to make sure
that we write any remaining bytes to the HAL.
Bug: 10810347.
Change-Id: If9a3b22cc3b9e6eb384a56c48c40e6258d0896ad
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Offloading of audio effects is now enabled for offloaded
output threads. If an effect not supporting offload is enabled,
the AudioTrack is invalidated so that it can be recreated in PCM
mode.
Fix some issues in effect proxy related to handling of effect
commands to offloaded and non offloaded effects.
Also fixed a bug on capture index in software Visualizer effect.
Bug: 8174034.
Change-Id: Ib23d3c2d5a652361b0aaec7faee09102f2b18fce
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Change-Id: I5a1a79000d53146689b0a198cc5419c36509703f
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- Increase offloaded output thread standby delay to 1 second
to allow transition between tracks with going to stanby
if reusing the same audio track (gapless)
- Make sure pause/flush/resume sequence is sent to the HAL
in the right order
- Fix format display in track dump
Bug: 8174034.
Change-Id: I43ef6f8fdbf7427e4eff6cc2d0665d7d1463ea8a
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HOTWORD is a special capture audio source that uses
the same audio tuning as VOICE_RECOGNITION but allows an
active capture to be preempted.
Bug: 10640877.
Change-Id: Iea09a11cbcdbacc8d434e5230e7559dcd1b44ac0
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Remove possible race conditions between async callback
thread and offload thread when clearing and setting the
draining and write blocked flags.
Bug: 8174034.
Change-Id: I7af10491f39dc0e7d7414862a9d8e763daa2e2b7
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with a real implementation in AudioStreamOutSink
for dummy implementation initially in MonoPipe.
Use in AudioFlinger::PlaybackThread::threadLoop_write()
to keep the input to the timestamp latch up-to-date.
Change-Id: I10ef277991b63bb43d55d6f3df75116ef32246cd
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Make sure that a flush received after a pause is forwarded
to the HAL after the pause.
Change-Id: Ib3221f70c59f8b3dbbac20b23104f6b779b75be0
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using a new timestamp latch in PlaybackThread, and
AudioTrackServerProxy::framesReleased() which returns mServer.
Change-Id: I1ebfba968c773faaab95648c272fd3ebd74718d6
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Look at client's request for a fast track in the flags parameter,
and check whether the request can be satisfied.
Change-Id: I65ddaeb1b85670fe9066076f638bfed0cda01c0d
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The flags are currently unused, but will be used for requesting a fast
track. Making flags in/out will allow reporting back up to client that
the request is denied.
Change-Id: Ifbee57da3632ce130551065a426577fb97b1a68d
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This fixes a regression that was introduced earlier
by commit 9f80dd223d83d9bb9077fb6baee056cee4eaf7e5
called "New control block for AudioTrack and AudioRecord".
That commit broke underrun reporting for fast tracks.
Also remove Track::mUnderrunCount, which counted the number of underrun
events, and was only used by dumpsys media.audio_flinger.
Now dumpsys media.audio_flinger reports the number of underrun frames,
Isolated underrun-related control block accesses via the proxy, so that
the server is not directly poking around in the control block.
The new proxy APIs are AudioTrackServerProxy::getUnderrunFrames() and
AudioTrackServerProxy::tallyUnderrunFrames(). getUnderrunFrames() returns
a rolling counter for streaming tracks, or zero for static buffer tracks
which never underrun, but do a kind of 'pause' at end of buffer.
tallyUnderrunFrames() increments the counter by a specified number of frames.
Change-Id: Ib31fd73eb17cbb23888ce3af8ff29f471f5bd5a2
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* commit '6d4d879037a495f9a4276a6996d1845594ce9090':
Fix mismatched delete/new couple
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In RecordThread::readInputParameters(), mRsmpInBuffer is allocated
with new[] and should be freed with delete[] instead of delete.
This error has been found by valgrind.
Change-Id: I2f947abb658a21a7d1e61eebb99270cb37d2d147
Signed-off-by: Andrei V. FOMITCHEV <andreix.fomitchev@intel.com>
Signed-off-by: David Wagner <david.wagner@intel.com>
Signed-off-by: Jack Ren <jack.ren@intel.com>
Signed-off-by: Bruce Beare <bruce.j.beare@intel.com>
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Change-Id: Ie071673875f663de4212eed4a4dff89d51a5a915
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Change-Id: I7b6d31e24531954ab1ecdf3ed56c19433700bd89
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Currently there are 16-bit PCM assumptions in several places for capture:
- resampler API
- mRsmpInBuffer and mRsmpOutBuffer
- RecordThread::threadLoop upmix, downmix, and resampling
- possibly other places
Until those assumptions are removed, this CL enforces 16-bit PCM in both
client and server at all places where a format is checked.
Change-Id: I08b0570bff626ad0d341804825a72c14e61b4233
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Change-Id: Ieabd91acee92d0e84e66fbd358df5282b856306e
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In two places we assumed that TrackBase::sampleRate() would return the
same value when it is called twice in the same function. This is not
guaranteed; sampleRate() reads from the control block so the return
value could change. To fix this, only call sampleRate() once and cache
the return value to get a consistent value.
This was only a theoretical race. In MixerThread::prepareTracks_l()
it would have no bad effect. In TimedTrack::getNextBuffer() it could
cause a real problem, but we don't currently support dynamic sample rate
ratios for timed tracks.
Change-Id: I8e5c33f0121fc058d1e70c2ab5e9135397d3e0b7
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Only keep them in performance-sensitive code
Change-Id: Ib257ddd7bc39ce9896997ffae008c524ac743d01
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This prepares for using it in PlaybackThreads later
Change-Id: Id90a92aa6372e4b69914b0008cef07296ca5d6a3
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Move mNormalFrameCount and frameCountHAL(), since they're not used by
record threads. Also comment which fields are updated by readParameters().
Change-Id: I5fc0a8a89cc637976f22d49271a5a3e136dab4e1
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Former name 'channels' was ambiguous with respect to channel count.
Change-Id: I716f792d95a7e0c787d27514ad6e93dbcef8a415
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Change-Id: Iedfeca3cd477d023c350d6d4e6eed874ee467a32
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mChannelCount was 8-, 16-, or 32-bits
Change-Id: I2cc2fedf3e33144e5c8bbd9894763282d9217f63
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Change-Id: Ideb83dea2c3002651c34fa646753cba598e29e93
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No need to delete AudioMixer before readOutputParameters
Change-Id: Icafa785c9021bbe8d985e4f9527f8fc8c5e62622
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Change-Id: Ib959c1e9dc9544d12277ce11bea445118b2e0521
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