History log of /frameworks/av/services/audioflinger/AudioFlinger.h
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
2bfc6b42b3733c12485dd51ed95191956abc3e4e 28-Sep-2012 Jean-Michel Trivi <jmtrivi@google.com> bug 7253033 clean up before closing an output

An output can only be closed if there is no lock contention that
prevents ThreadBase::exit() from being blocked. If an output
device is waiting for an operation to complete (here a write
in the remote_submix module, because the pipe is full), signal
the module that it's entering the "exiting" state.

Change-Id: I8248add60da543e90c25a4c809866cdb26255651
cc0f1cfb69ce8b8985fc2c0984847a06a13ad22d 24-Sep-2012 Glenn Kasten <gkasten@google.com> Implement android.media.AudioManager.getProperty()

Bug: 6635041
Change-Id: I3386a4a6c226bc4eceaf65556119e4fb15f73224
896adcd3ae6a1c7010e526327eff54e16179987b 13-Sep-2012 Eric Laurent <elaurent@google.com> audioflinger: send priority request from a thread

When creating a fast AudioTrack, a request is sent to SchedulingPolicyService
to elevate the requesting thread priority. This generates a binder
call into system_server process and to a JAVA service via JNI.
If the thread from which the track was created is in the system_server
process and does not have the "can call java" attribute, a crash occurs because
the binder optimization reuses the same thread to process the returning binder
call and no JNI env is present.

The fix consists in sending the priority change request from the AudioFlinger
mixer thread, not from the binder thread.

This also reverts the workaround in commit 73431968

Bug 7126707.

Change-Id: I3347adf71ffbb56ed8436506d4357eab693078a3
fe3156ec6fd9fa57dde913fd8567530d095a6550 11-Sep-2012 Jean-Michel Trivi <jmtrivi@google.com> Communicate audio session ID to downmixer

The audio downmixer effect might need the audio session Id, pass it
from the track creation in AudioFlinger to the downmix effect
creation in AudioMixer.

Change-Id: I5e29540542ae89cf4a0cdb537b3e67f04442a20a
f1c04f952916cf70407051c9f824ab84fb2b6e09 28-Aug-2012 Eric Laurent <elaurent@google.com> audioflinger: changes for new audio devices enums

The ThreadBase class now has a separate member for input
and output devices (mInDevice, mOutDevice).

Only query get_supported_devices() from audio HAL if the function
is exposed and if the audio policy manager did not specify the
audio module to open.

Also fixed bug in AEC preprocessing that would reset
to default output device when an input device was given.

Change-Id: I19d4d06aeb920b068e3ef31e6e6be6345ce5d67a
57b2dd1e78af53115985f18d31ec5421c9da947e 01-Sep-2012 Eric Laurent <elaurent@google.com> AudioFlinger: send audio source to audio effects

Added support for EFFECT_CMD_SET_AUDIO_SOURCE audio effect
command to inform preprocessings of current audio source
selection for capture.

Change-Id: Ib2418a9aa8114e8457fe828ecd43b230ed86cdd6
c3ae93f21280859086ae371428ffd32f39e76d50 30-Jul-2012 Glenn Kasten <gkasten@google.com> Update audio comments

Change-Id: Ie7504d0ddb252f7e4d4f99ed0b44cfc7b1049816
2dd4bdd715f586d4d30cf90cc6fc2bbfbce60fe0 29-Aug-2012 Glenn Kasten <gkasten@google.com> Move libnbaio out of AudioFlinger

libnbaio is now a separate shared library from AudioFlinger, rather
than a static library used only by AudioFlinger.

AudioBufferProvider interface is now also independent of AudioFlinger,
moved to include/media/

Change-Id: I9bb62ffbc38d42a38b0af76e66da5e9ab1e0e21b
106e8a42038f9e90d5ff97f8ab6f1a42258bde9e 02-Aug-2012 Glenn Kasten <gkasten@google.com> const methods

Change-Id: I92e32ee16274c032c9d0ce910676be2a7fa52471
ee578c0330319f04a48bccbdb26b53fea0388d04 24-Jul-2012 John Grossman <johngro@google.com> AudioFlinger: Better handling for master volume/mute

(cherry picked from commit 93d906837e0e89aa1d9c913ab2b531b809f9bb9e)

> AudioFlinger: Better handling for master volume/mute
> Changes to address bug 6842827.
> When a HAL is loaded, cache whether or not the HAL supports
> set_master_volume/mute in the AudioHwDevice structure. Store an
> AudioHwDevice in AudioStream(In|Out) structures instead of just an
> audio_he_device_t. This give threads (PlaybackThreads in
> particular) access to the cached capabilities.
> When setting master volume/mute, change the system to always set the
> setting on all HAL which support it and also to set the setting on all
> PlaybackThreads. Change PlaybackThreads to apply the setting at the
> in SW mix stage of the pipeline if its assigned HAL does not support
> the setting, or to ignore the setting of the assigned HAL does support
> it.
> Change-Id: Ia14137a30b4c3ee6f2d7ddcc8cba87bf5eec87f4
> Signed-off-by: John Grossman <johngro@google.com>

Change-Id: Icb6bc13764e100a2003eb1dee2231132ab287d98
Signed-off-by: John Grossman <johngro@google.com>
d8f178d613821c3f61a5c5e391eb275339e526a9 20-Jul-2012 John Grossman <johngro@google.com> Change audio flinger to user HAL master mute if available

(cherry picked from commit 91de9b56282d126ffb36344266af5fee3cefcfdd)

> Change audio flinger to user HAL master mute if available
> Hand merge from ics-aah
> > Change audio flinger to user HAL master mute if available: DO NOT MERGE
> >
> > Replicate the pattern used for HAL master volume support to make use
> > of master mute support if the HAL supports it. This is part of the
> > change needed to address bug 6828363. Because of the divergences
> > between ICS and master, this change will need to be merged by hand.
> >
> > Signed-off-by: John Grossman <johngro@google.com>
> > Change-Id: I6d83be524021d273d093bcb117b8f2fe57c23685
> Change-Id: I32280582905c969aaec2bb166ec5c61df82d737a
> Signed-off-by: John Grossman <johngro@google.com>

Change-Id: I5cd709187221d307fe25c5117ccaadca5f6b197b
Signed-off-by: John Grossman <johngro@google.com>
510a3d6b8018a77683dac466127ffd0af34bef6e 16-Jul-2012 Glenn Kasten <gkasten@google.com> Start adding support for multiple record tracks

Replace single mTrack by vector mTracks.
Destroy record tracks similarly to playback tracks.
Dump all record tracks, in addition to the active record track.

Change-Id: I503f10b51928b6b92698fe1c51a9ddd3215df1f4
0ec23ce0d1ff79566c402bc30df3074f6e25a22b 10-Jul-2012 Glenn Kasten <gkasten@google.com> Clean up start() parameters

Document where int is used instead of AudioSystem::sync_event_t
(probably because of a header file dependency).
TrackBase::start() and RecordTrack::start() don't need default parameters.

Change-Id: I82f4a4d078be900f3aa4bd926697e32f5ed68ec8
e4e2a37dbe2a4d923232305549101f779a2e3638 23-Jul-2012 Glenn Kasten <gkasten@google.com> Extract methods to enter standby and standby mode

Also move initial standby from to threadLoop to avoid a race condition.

Change-Id: I65afca83c36fb41b983b3b1d3dab35d4029560e3
0a7af18d0308295405491f86603e3d119450aba0 10-Jul-2012 Glenn Kasten <gkasten@google.com> Use valueAt instead of editValueAt when possible

Change-Id: I885b169f4b176a6b5c2ca9a534214b4ffff1700e
58e5aa34f01d663654d8bafad65db1dda42161ff 20-Jun-2012 Glenn Kasten <gkasten@google.com> effect_descriptor_t const correctness

Change-Id: Iad008f20d35a18acf500f773900164552fd0c19e
1d491ff06f4b9c90ff24fe953b90d0843eaf1c04 16-Jul-2012 Glenn Kasten <gkasten@google.com> Fix races in AudioRecord stop()

Change-Id: Id0ac1915f57fef4a938c7f90989c1162a8b6c51c
be5f05e0fdfc4e3799653702187861a2afa072ee 19-Jul-2012 Glenn Kasten <gkasten@google.com> Internal dump methods return void not status_t

Only the IAudioFlinger::dump() needs to return a status_t.

Change-Id: Iffeb2a7db4846df850b6b2ed960276f1fd75dba0
bb4350d3b9e9485ae59e084de270f86aecef8066 04-Jul-2012 Glenn Kasten <gkasten@google.com> Use audio_devices_t more places

Change-Id: Id0ace02ca6f480a6c46e11880acf6cdc66d83184
d96c5724818fb47917bb5e7abe37799735e1ec0e 25-Apr-2012 Glenn Kasten <gkasten@google.com> Don't call virtual methods in destructor

The result of calling virtual methods from a destructor is undefined.

Change-Id: I0fd4a19626e5ae564a60b753315b5f6c4b8d1f2c
1ea6d23396118a9cfe912b7b8a4e6f231e318ea2 09-Jul-2012 Glenn Kasten <gkasten@google.com> Use atomic ops for thread suspend count

There was a theoretical but unlikely race if two binder threads
executed suspend() or restore() concurrently. Also added comments.

Change-Id: I0908acc810b83bdd66455b27ca3429de1662a2cd
1879fff068422852c1483dcf8365c2ff0e2fadfc 12-Jul-2012 Glenn Kasten <gkasten@google.com> Add tid parameter to IAudioFlinger::openRecord

Not yet implemented

Change-Id: I35523fb15ad71727ecc9f4bb870f07e4b7397dc4
39c54f68804c1ce5c85ec588f3c2c63447a807b4 09-Mar-2012 Glenn Kasten <gkasten@google.com> Remove dead code

Change-Id: If22a6c4e572b0734eba0c5a7ce29a2c61c581e5d
04270daf50f0c602d7c57a257a693e68246cbeb7 10-Jul-2012 Glenn Kasten <gkasten@google.com> Record overflow cleanup

Add comments and rename one method for clarity

Change-Id: I04a9147e46e88a072256c0211b112d52202419e2
254af180475346b6186b49c297f340c9c4817511 03-Jul-2012 Glenn Kasten <gkasten@google.com> Use audio_channel_mask_t more places

Use it in AudioSystem::getOutput(), AudioSystem::getInput(),
IAudioPolicyService::getOutput(), IAudioPolicyService::getInput(),
and various other places in AudioFlinger.

Not done: AudioTrack and OutputDescriptor.

Change-Id: I70e83455820bd8f05dafd30c63d636c6a47cd172
2df8f55055fe431a508148cf525df1ba40f03113 10-Jul-2012 Glenn Kasten <gkasten@google.com> Add comments on use of volatile for track count

and add acquire load at the read

Change-Id: Ib41a58f5b1f6af87a8bd63d3f77d2ec0e48cb479
a34f8ec169986c5a28600c0decaa4e2db70df8e4 21-Jun-2012 Glenn Kasten <gkasten@google.com> Remove 'volatile' from mMute and add comments

Change-Id: I386ba27b2305a397aba70331c6bf0d35ea727cf6
9f34a36d9cdb9595c288e50ffe00da038bc8abb9 21-Mar-2012 Glenn Kasten <gkasten@google.com> Whitespace and indentation

Change-Id: I424052b4ff9218147a5cfc8e6dcd67fe8105d229
6648821933dc06c0b09ab2c8b32135edddcd4291 21-Jun-2012 Glenn Kasten <gkasten@google.com> AudioFlinger::getBuffer() always returns non-NULL

Change-Id: I543d3db507597cacbfdad5d9ea71732137fe54fb
01542f2704f39956da09ae2840e192dab760091f 02-Jul-2012 Glenn Kasten <gkasten@google.com> Only write to mDevice once

This fixes a bug where readers might see intermediate values.
Also add comments about how mStandby and mDevice are used.

Change-Id: Idc84e56c21381a45137a2ca5ff9c57d437201869
c1dae24a08b67b98e18e4239d4f3a74d600d353c 03-Jul-2012 Glenn Kasten <gkasten@google.com> Remove debug code HAVE_REQUEST_PRIORITY and SOAKER

Change-Id: I73a2afe72d8acb53e57e6b4e6fb5133e22b7875a
a5f44ebaf58911805b4fb7fb479b19fd89d2e39b 25-Jun-2012 Eric Laurent <elaurent@google.com> audioflinger: fix effect disconnect deadlock

Fix possible deadlock when several EffectHandles on the same
EffectModule are destroyed simultaneously:
A wp on an EffectHandle should not be promoted to a local sp
with ThreadBase mutex held as the EffectHandle destructor can be
called when the sp gets out of scope which will call
ThreadBase::disconnectEffect() and try to acquire the mutex.

Use raw pointers instead of weak pointers for the list of handles
on an EffectModule.

Bug 6679606.

Change-Id: Ice8b602fb03a7d363c44ce3dced8a53540d96270
dd8104cc5367262f0e5f13df4e79f131e8d560bb 02-Jul-2012 Glenn Kasten <gkasten@google.com> Use audio_channel_mask_t more consistently

In IAudioFlinger::createTrack() and IAudioFlinger::openRecord(),
declare input parameter to use correct type audio_channel_mask_t.

In IAudioFlinger::getInputBufferSize(), input parameter is now channel mask
instead of channel count.

Remove unused IAudioFlinger::channelCount(audio_io_handle_t).

In AudioRecord::getMinFrameCount() and AudioSystem::getInputBufferSize(),
input parameter is channel mask instead of channel count.

Change-Id: Ib2f1c29bea70f016b3cfce83942ba292190ac965
f1da96d8cf60842538e00a9c950cc451f7da2c10 03-Jul-2012 Glenn Kasten <gkasten@google.com> Remove longStandbyExit

It was never set (the assignment was within an "if" that was never true).

Change-Id: I01cc68e9df6b190eece621b2aa9858b4361880ce
415fa7599f48494f99206b8d6e1974abb52c5923 03-Jul-2012 Glenn Kasten <gkasten@google.com> Fix uninitialized field EffectModule::mPinned

Also mark EffectModule::mId and EffectModule::mSessionId const, and
document the initialization of other fields in EffectModule.

Change-Id: Ic1ca008e75e9b5924743ffc35bef80057f3a0669
dbabf8a7dfe3aa8bf0ed169220d2009d5891fef2 01-Jul-2012 Eric Laurent <elaurent@google.com> am 651f9e7c: am 717e1286: audioflinger: fix auxiliary effect attachment

* commit '651f9e7c972b58a49066081187161268bcf9237a':
audioflinger: fix auxiliary effect attachment
717e128691f083a9469a1d0e363ac6ecd5c65d58 30-Jun-2012 Eric Laurent <elaurent@google.com> audioflinger: fix auxiliary effect attachment

Auxiliary effects (Reverb) are global effects and as such follow
the default rule which is to attach them to the output thread that
handles music streams by default. This causes a problem when several
threads are eligible to handle music streams as tracks can be attached
to either thread based on criteria unknown when teh effect is created.

The fix consists in moving the auxiliary effect if necessary when an
AudioTrack is attached to it and this track is not on the same
output thread.

Bug 6608561.

Change-Id: Ib32c3cabc731b2046aba728be1771982999c6069
22167855ff9af7b13fda669ca27c67a037a7d585 20-Jun-2012 Eric Laurent <elaurent@google.com> audioflinger: fix setStreamOutput()

AudioFlinger::setStreamOutput() should also work for direct outputs.
Also ignore the destination output specified to match the expected
behavior which is to invalidate all tracks using the specified stream
type so that they can be re created on the correct ouput thread.

Do not send STREAM_CONFIG_CHANGED event wich is ignored by AudioSystem
anyway since the stream to output cache has been removed.

Change-Id: I13d9d47922923b630dd755717875424c16be4637
c15d6657a17d7cef91f800f40d11760e2e7340af 30-May-2012 Glenn Kasten <gkasten@google.com> Add audio watchdog thread

Change-Id: I4ed62087bd6554179abb8258d2da606050e762c0
28ed2f93324988767b5658eba7c1fa781a275183 07-Jun-2012 Glenn Kasten <gkasten@google.com> Reduce underruns in screen off, esp. with EQ

Add MonoPipe APIs to specify setpoint.
Use screen state to configure pipe setpoint.
Fix a long-standing bug where pipe sleep time was excessive,
which interacted poorly with governor and low clock frequencies.
Now it deducts the elapsed time since last write(),
which was significant when there was EQ and low clock frequency.

Bug: 6618373
Change-Id: I6f3b0072c2244aeb033ef0795ad164491a164ff5
f06c2ed50e1db871ae9eb2bd15a196064f8c278c 06-Jun-2012 Marco Nelissen <marcone@google.com> Take latency and current time into account for visualization

Buffer more data, and return the data that is currently being
output from the audio output, to ensure that visualizations are
smooth and responsive even when the audio output has a large
latency and/or large buffers.

Change-Id: I401637f01be7600b3c594a55c869036c13b206c0
67c0a58e05f4c19d4a6f01fe6f06267d57b49305 02-May-2012 Eric Laurent <elaurent@google.com> audioflinger: various fixes on direct output

Various fixes in direct output playback thread implementation:
- threadLoop_write() was broken for playback threads that do not
use a pipe sink.
- output buffer size calculation was hard coded for stereo.
- removed software volume that was implemented for PCM stereo
format only: the audio HAL has to implement volume if needed
for direct outputs.

Change-Id: If211b4489be9af395435707b8cf0388cce1347b2
399930859a75d806ce0ef124ac22025ae4ef0549 31-May-2012 Glenn Kasten <gkasten@google.com> State queue dump

Bug: 6591648
Change-Id: Iac75e5ea64e86640b3d890c46a636641b9733c6d
893a05479c96f911d02beb0443da3ed6508143a7 30-May-2012 Glenn Kasten <gkasten@google.com> Fix fast track leak if out of normal track names

Bug: 6580402
Change-Id: I3ac7f012062c35833147f47ba822eb4bf532a824
91b14c4c144d0cc957a427cffc02ba10d0615677 30-May-2012 Eric Laurent <elaurent@google.com> audioflinger: fix effect problem during underrun

When an audio track underruns, the input buffer of the
corresponding effect chain (if any) must be cleared, otherwise
audio from previous mixer run will be fed again to the effect process

Bug 6551652.

Change-Id: I5cd02196745f756c85af82d6937e9dc54369b37f
e737cda649acbfa43fc1b74612a83f2fac9aa449 23-May-2012 Eric Laurent <elaurent@google.com> audioflinger: refine latency latency calculation.

There is an audio pipe between the normal mixer output and the fast
mixer to cope for scheduling delays and buffer size difference.
This pipe depth was not taken into account in latency calculation.

Adding the pipe contribution to the latency significantly improves A/V sync.

Bug 6520569.

Change-Id: I5584908e8aa8a02170eb38b22b4370eea800a235
fbae5dae5187aca9d974cbe15ec818e9c6f56705 21-May-2012 Glenn Kasten <gkasten@google.com> Keep a copy of most recent audio played

Change-Id: I6b2f97881c39998a2fae9ab79d669af6c0a37e94
88cbea8a918bbaf5e06e48aadd5af5e81d58d232 15-May-2012 Glenn Kasten <gkasten@google.com> Display pipe underrun counters in dumpsys

The normal mixer writes it's submix to a pipe, which is read by the fast
mixer. Now dumpsys media.audio_flinger display the raw underrun counters
when fast mixer tries to pull from the pipe but doesn't get enough frames.

Change-Id: I72505f149f9e12802784da654a651d43734e1c79
44a957f06400a338e7af20b3d16c4c4ae22a673c 16-May-2012 Eric Laurent <elaurent@google.com> Fix static track activity ref counting

When a static AudioTrack underruns, it means that playback is over.
As apps do not necessarily stop playback explicitly, AudioFlinger
should call stopOutput() to decrease activity ref count in
audio policy manager.

Bug 6486311.

Change-Id: I1ea722c443780329ded6310c958b24726e918d16
2986460984580833161bdaabc7f17da1005a8961 09-May-2012 Eric Laurent <elaurent@google.com> Fix issues with synchronous record start.

- Added a timeout in case the trigger event is never fired.
- Extend AudioRecord obtainBuffer() timeout in case of
synchronous start to avoid spurious warning.
- Make sure that the event is triggered if the track is
- Reject event if the triggering track is in an incompatible state.

Also fix a problem when restoring a static AudioTrack after
a mediaserver crash.

Bug 6449468.

Change-Id: Ib36e11111fb88f73caa31dcb0622792737d57a4b
09474df67278c0cd621b57c4aef1deaec4d8447f 10-May-2012 Glenn Kasten <gkasten@google.com> Improve underrun handling for fast tracks

Maintain more accurate accounting of type of underrun.
Automatically remove track from active list after a series of "empty" underruns.

Change-Id: If042bf80e1790dcaaf195c99dc9c0ed9b55382c1
d08f48c2ad2941d62b313007955c7145075d562c 02-May-2012 Glenn Kasten <gkasten@google.com> Fix stopping process for fast tracks

Previously, the state of a fast track "wiggled" back and forth at the end.

Now it goes through these transitions:
active -> stopping_1 -> stopping_2 -> stopped

This CL is only for fast tracks, and does not change how
normal tracks work.

Change-Id: Icc414f2b48c46dda63cfa6373ca22d033dd21cd4
810280460da5000785662f6c5b0c7ff3ee0a4cb3 01-May-2012 Glenn Kasten <gkasten@google.com> Temporary fix for both normal tracks & fast tracks

If there is at least one active fast track, it forces a mixer
status of ready, which messes up the logic for normal track underruns.

Change-Id: I9de2fcaef090e2c2f99682333af3d3dd618b0d6b
288ed2103d96f3aabd7e6bea3c080ab6db164049 26-Apr-2012 Glenn Kasten <gkasten@google.com> Fix race condition for non-started fast tracks

This required re-implementing how fast tracks are considered active.
Now, they use the same logic as normal tracks, except underrun is ignored.

Other changes:
- add framesReady() to AudioBufferProvider interface
- rebased
- add track underrun counter state to fast mixer dump state
- move dumpsys header to Track::appendDumpHeader()
so it closer to where tracks are dumped
- display track state in dumpsys as a character code
- measure and display warmup time and cycles in dumpsys
- copy in the presentation complete code
- add ExtendedAudioBufferProvider for framesReady() which returns size_t
- simplify underrun tracking
- deferred reset track after stop()
- add comments

Change-Id: I7db8821bc565230ec76da1f9380fe3fb09735e5b
83faee053cfd4251dbb591b62039f563ffdac399 28-Apr-2012 Eric Laurent <elaurent@google.com> AudioFlinger: fix stop detection for static tracks

The end of playback and end of presentation detection was broken for
static AudioTracks (tracks using shared memory buffers passed by client).

The mixer should not wait for a minimal amount of frames to be available to mix
a static track otherwise the last frames might never be consumed.

A static track should be removed from active list in case of underrun even if not

Issue 6411521.

Change-Id: I66a2c1a77e98149e5049a223a6f04c3b8c5ad11a
da747447c1d4b5205469b4e94485b8769df57a97 26-Apr-2012 Eric Laurent <elaurent@google.com> AudioFlinger: fix tracks ready for mixing logic.

Commit fec279f5 broke the logic allowing to wait for an application
to provide frames for mixing in the case of several active tracks.

This was causing audio gaps when playing music and superposing a
sound Fx (keyboard clicks...).

Issue 6185007.

Change-Id: Id0fad150d0b615646d6b1387c0de8ca944d228f6
58912562617941964939a4182cda71eaeb153d4b 03-Apr-2012 Glenn Kasten <gkasten@google.com> AudioFlinger normal mixer uses FastMixer

Change-Id: I3131bb22d2d057e9197a2ebfa6aa1cfaab9e5321
3acbd053c842e76e1a40fc8a0bf62de87eebf00f 28-Feb-2012 Glenn Kasten <gkasten@google.com> Configure policy of mediaserver threads

Change-Id: Ifd825590ba36996064a458f64453a94b84722cb0
0ca3cf94c0dfc173ad7886ae162c4b67067539f6 18-Apr-2012 Eric Laurent <elaurent@google.com> rename audio policy output flags

Change-Id: I27c46bd1d1b2b5f96b87af7d05b951fef18a1312
1c345196edc61694f29307a1826a64a0d26028dc 27-Mar-2012 John Grossman <johngro@google.com> TimedAudio: Track of the number of pending frames.

This is a manual merge from ics-aah

> TimedAudio: Track of the number of pending frames.
> Keep track of the number of frames pending in the timed audio queue so
> we can implement framesReady in O(1) time instead of O(N). This
> change partially addresses bug 6020970; the bug will be completely
> addressed once this change has been up-integrated into master.
> Change-Id: I599eb15ea1f6d715b97b30e65214fb6fadd169df
> Signed-off-by: John Grossman <johngro@google.com>

Change-Id: I6cbbbc3afc8efd066fe94865326ede0c6b3db2bd
Signed-off-by: John Grossman <johngro@google.com>
9fbdee13d09447550dd22ae72c2dbabdce7f0a80 27-Mar-2012 John Grossman <johngro@google.com> TimedAudio: Fix a cause of audio popping.

This is a manual merge from ics-aah

> TimedAudio: Fix a cause of audio popping.
> Fix an issue with buffer lifecycle management which could cause audio
> pops on timed outputs. There were two issues at work here.
> 1) During trim operations for the queued timed audio data, buffers
> were being trimmed based on their starting PTS instead of when the
> chunk of audio data actually ended. This means that if you have a
> very large chunk of audio data (larger than the mixer lead time),
> then a buffer at the head of the queue could be eligible to be
> trimmed before its data had been completely mixed into the output
> stream, even though the output stream was fully buffered and in no
> danger of underflow.
> 2) The implementation of getNextBuffer and releaseBuffer for timed
> audio tracks was not keeping anything like a reference to the data
> that it handed out to the mixer. The original architecture here
> seemed to be expecting a ring buffer design, but timed audio tracks
> use a packet based design. Pieces of packets are handed out to the
> mixer which then frequently will hold onto that chunk of data
> across two mix operations, using the first part of the chunk to
> finish a mix buffer and then using the end of the chunk for the
> start of the next mix buffer. If the buffer that the mixer is
> holding a piece of got trimmed before the start of the next mix
> operation, it would return to its heap and could be filled with who
> knows what by the time it actually got mixed. On debug builds,
> they seem to get zero'ed out as they go back to the heap causing
> obvious pops in presentation.
> This change addresses both issues. Trim operations are now based on
> ending presentation time for a chunk of audio, not the start. Also,
> when the head of the queue is in flight to the mixer, it can no longer
> be trimmed immediately, merely flagged for trim by the mixer when the
> mixer finally does call releaseBuffer.
> Signed-off-by: John Grossman <johngro@google.com>
> Change-Id: Ia1ba08cb9dea35a698723ab2d9bcbf804f1682fe

Change-Id: I2c5e2f0375c410f0de075886aac56ff6317b144c
Signed-off-by: John Grossman <johngro@google.com>
7d5b26230a179cd7bcc01f6578cd80d8c15a92a5 05-Apr-2012 Jean-Michel Trivi <jmtrivi@google.com> AudioMixer uses downmix effect for multichannel content

In the AudioMixer structure associated with each track, add an object
that acts as the buffer provider when the track has more than two
channels of input in the mixer. This object, DownmixerBufferProvider,
gets audio from the actual buffer provider of the track, and applies
a downmix effect on it.
The downmix effect is created and configured when the track gets
created in AudioFlinger, which causes AudioMixer::getTrackName()
to be called with the new track's channel mask. It is released
when the track is disabled in the mixer.

Change-Id: I05281ed5f61bef663a8af7ca7d5ceac3517c82db
a4c5a550e2a3bc237179b8684e51718e05894492 29-Mar-2012 Eric Laurent <elaurent@google.com> audio policy: load audio hw modules.

Audio HW modules are now loaded upon request from audio policy manager
according to the configuration in audio_policy.conf.
Removed hard coded HW module loading by AudioFlinger at init time.
Added methods to IAudioFlinger and AudioPolicyInterface
to control the loading of audio HW modules.
Added methods to open an output or input stream on a specific hw module.

Change-Id: I361b294ece1a9b56b2fb39cc64259dbb73b804f4
1a9ed11a472493cac7f6dfcbfac2064526a493ed 21-Mar-2012 Eric Laurent <elaurent@google.com> audio policy: add configuration file

removed outputs to stream mapping cache in audio system: the output for a
given stream type must always be queried from audio policy manager as the cache
is not always updated fast enough by audioflinger callback.

removed AudioFlinger::PlaybackThread::setStreamValid() not used anymore if
stream to output mapping is not cached.

Change-Id: Ieca720c0b292181f81247259c8a44359bc74c66b
73d227557ba5192735356bacab9f77b44980793b 19-Mar-2012 Glenn Kasten <gkasten@google.com> AudioFlinger track flags and server's fast policy

Change-Id: I72358c8e6829d173b3e60ced8a8babc089869fac
0bf65bdde04b8e66c998ff37e2b2afafddddfa33 29-Feb-2012 Glenn Kasten <gkasten@google.com> const methods and comments

Change-Id: Ifd16750174fdb15b72507787502b587562ffc99e
a011e35b22f95f558d81dc9c94b68b1465c4661d 30-Mar-2012 Eric Laurent <elaurent@google.com> implemented synchronous audio capture

Added the infrastructure to support the synchronization of playback and
capture actions on specific events.
The first requirement for this feature is to synchronize the audio capture
start with the full rendering of a given audio content.
The applications can further be extended to other use cases
(synchronized playback start...) by adding new synchronization events and
new synchronous control methods on player or recorders.

Also added a method to query the audio session from a ToneGenerator.

Change-Id: I51f1167290d9cafdf2fbcdf9e4785156973af44c
b83d38feeeb88a8a2a6219e1fca2480b5a14fb0d 26-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "IAudioFlinger::createTrack and openRecord flags"
c5c49398584f2399af905a931e556ed6e0a29cd4 21-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "Clean up Track constructor"
f99590187e2e3f1cf6f093063170edec269cac5d 19-Mar-2012 Glenn Kasten <gkasten@google.com> Clean up Track constructor

The 'thread' parameter can never be NULL.
Use constructor initialization list when possible.
Make more members const.
Only put the relevant code under "if (mCblk != NULL)".
Add comment about track name leak.

Change-Id: Ib963390a69bed1999638cc982a759edd1d5f4712
17a736c3e1d062d7fc916329eb32aef8935614af 14-Feb-2012 Glenn Kasten <gkasten@google.com> Update comments

Change-Id: I327663a020670d0a72ff57bd0b682e2ce0528650
a075db4ff9b086ac2885df77bb6da0869293df92 06-Mar-2012 Glenn Kasten <gkasten@google.com> IAudioFlinger::createTrack and openRecord flags

createTrack and openRecord don't need the "old" flags parameter,
which was either audio_policy_output_t or audio_in_acoustics_t
shifted left by 16 bits. But they do need "new" flags, which
are defined by the application use case. Initially, the only
application use case flag is timed output, but others are planned.

For output, the audio_policy_output_t flags are passed to
AudioSystem::getOutput, which returns an audio_io_handle_t, and that
handle is then passed to createTrack. So createTrack doesn't need the
old flags parameter.

For input, the audio_in_acoustics_t flags are passed to
AudioSystem::getInput, which returns an audio_io_handle_t, and that
handle is then passed to openRecord. So openRecord doesn't need the
old flags parameter.

Change-Id: I18a9870911846cca69d420c19fe6a9face2fe8c4
e53b9ead781c36e96d6b6f012ddffc93a3d80f0d 13-Mar-2012 Glenn Kasten <gkasten@google.com> Whitespace and indentation

Fix indentation to be multiple of 4.
Make it easier to search:
sp< not sp < to
"switch (...)" instead of "switch(...)" (also "if" and "while")
Remove redundant blank line at start or EOF.
Remove whitespace at end of line.
Remove extra blank lines where they don't add value.

Use git diff -b or -w to verify.

Change-Id: I966b7ba852faa5474be6907fb212f5e267c2874e
73f4bc33e0d458933460250a47c64aa868c05f97 09-Mar-2012 Glenn Kasten <gkasten@google.com> Inline applyVolume() into threadLoop_mix()

Also the declaration of applyVolume in PlaybackThread was dead.

Change-Id: I4b1a9848d07d3d7f340baea05b17f667c78df868
470aa50c36089fbe0427557f7cf4464dd26a1c52 12-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "Remove unnecessary friend declarations"
2d6ef93773465cd3e66146fac35050a472c589f7 12-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "Remove virtual from methods that don't need it"
66fcab972e9218d47c58a915f391b2f48a09903a 24-Feb-2012 Glenn Kasten <gkasten@google.com> Merge dup code at thread entry and param change

This CL is mostly just cleanup, but there are a couple of fixes marked
"FIX" below.

Merge the duplicate code that was at the beginning of threadLoop() and
after a parameter change. cacheParameters_l() is now called at entry to
threadLoop() and after any parameter change. It re-calculates all values
that are derived from parameters, and caches them in instance variables.

- FIX activeSleepTime depends on mWaitTimeMs, which was initially set
to infinity. updateWaitTime_l() was not called at entry to
threadLoop(), so activeSleepTime was not set correctly before the
first parameter change.

- FIX reversed the order of calls after parameter change
for the same reason so that updateWaitTime_l() is called before
calculating values that are derived from wait time.

- marked it private since now it's only called from DuplicatingThread

Change-Id: If2607d2ed66c6893d910433e48208a93c41fb7e9
18868c5db2f90309c6d11e5837822135e4a0c0fa 07-Mar-2012 Glenn Kasten <gkasten@google.com> Use audio_policy_output_flags_t consistently

This affects:
- IAudioFlinger::openOutput
- AudioTrack::AudioTrack
- AudioTrack::set
- apps that call these

Change-Id: I26fb281bac6cb87593d17697bc9cb37a835af205
688aac7675f18bdd7bff13334759e20bc4e6c390 09-Mar-2012 Glenn Kasten <gkasten@google.com> Remove virtual from methods that don't need it

Change-Id: I30e17e61aae25b036436c0e270313c80c43e5f06
1998661fdb6b0b5ae103e047e3d653c5da1b99e3 09-Mar-2012 Glenn Kasten <gkasten@google.com> Remove unnecessary friend declarations

Add comments to the remaining friends, so we know what is left if we
decide to remove them later.

Change-Id: I1de929257dc4700960f77902cda3d303177c72cf
53d76dbe7c55821e89d9da02e7a563f7fb45de87 08-Mar-2012 Glenn Kasten <gkasten@google.com> Replace hard-coded 3 by FCC_2 to simplify searches

Change-Id: I92881d04e8378307f849fb343071a58d181a68b4
fec279f5a0bfaa2a42e91ab6dfa0282baeee308b 08-Mar-2012 Glenn Kasten <gkasten@google.com> Mixer status cleanup

Use mPrevMixerStatus for DirectOutputThread also.
Remove the MIXER_CONTINUE logic and use MIXER_IDLE instead.
Rename the field mixerStatus to mMixerStatus.
Rename local variable back to mixerStatus.

Change-Id: I0a8145fc856c6c5ff8b784b6176ef3c4d8eb7408
b071e9bc248865ef87a339044c0c5cbabfac175c 08-Mar-2012 Glenn Kasten <gkasten@google.com> Cleanup DirectOutputThread::mActiveTrack

Rename activeTrack to mActiveTrack.
Release the reference earlier, at the end of threadLoop_mix().
This allows the field to be made private and to
move the declaration from PlaybackThread to DirectOutputThread.

Change-Id: I02be7a254638f7d85e92aaf0002d20ca0092a5c3
b279312a9038b9c5b9b05b31b1b1db86f748efd8 08-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "IAudioFlingerClient::ioConfigChanged param2 const"
fa26a859bacb597587a8f49acfc3127351fa688c 06-Mar-2012 Glenn Kasten <gkasten@google.com> Isolate references to outputTracks/mOutputTracks

Move all references to DuplicatingThread::outputTracks and
DuplicatingThread::mOutputTracks from the common threadLoop() into
virtual methods. This allows them to be moved from PlaybackThread to
DuplicatingThread, and to be marked private.

Also use vector assignment to copy mOutputTracks to outputTracks.

Change-Id: Ieb1cf1ad36b8a65143e61e6c92a65fb43427e5e2
a13ad6e84db128eadf23b154d3346f0bb473a5f7 07-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "Make applyVolume private to DirectOutputThread"
d4513b09123deebf8023b73a82d2d46d35806cea 07-Mar-2012 Glenn Kasten <gkasten@google.com> Make applyVolume private to DirectOutputThread

Change-Id: I7ca4a59505857cbd106b6f274c66e9580dead271
a3707a280177e934a1e0a15660d9176663b7fc17 07-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "Add comments about sequence for setting parameters"
1465f0c3df04c3166155a852a6a5c10069c1fd0a 06-Mar-2012 Glenn Kasten <gkasten@google.com> Merge the calls to prepareTracks_l

Change-Id: I1dd759581333e2908d980180d44db7bf5ed6591d
b81cc8c6f3eec9edb255ea99b6a6f243585b1e38 01-Mar-2012 Glenn Kasten <gkasten@google.com> IAudioFlingerClient::ioConfigChanged param2 const

The 3rd parameter (param2) to AudioFlingerClient::ioConfigChanged
is used as an input. So changed it from void * to const void *.
It is then cast to const OutputDescriptor *
or const audio_stream_type_t * depending on the event.

Change-Id: Ieec0d284f139b74b3389b5ef69c7935a8e5650ee
f8edf68a1e39da273eafe8c85bdbedc26636c4ec 07-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "Rename fields of AudioSessionRef"
cbc52bfb3b51c81c945b1e35990324bde892829a 01-Mar-2012 Glenn Kasten <gkasten@google.com> Add comments about sequence for setting parameters

Change-Id: Iffa59a34c3c47bdc1d3234cdcb4d8ff99c102825
438b036deaf4413503567a75a2861c000d21da5b 06-Mar-2012 Glenn Kasten <gkasten@google.com> Rename updateWaitTime since a lock is held

Change-Id: I9bb978cbd0debf5b21676467060f72eebafea3e6
012ca6b4f69fb24385025c0e84b8f816525a3032 06-Mar-2012 Glenn Kasten <gkasten@google.com> Rename fields of AudioSessionRef

Change-Id: I9f2a66094135c4ac6bec2d3e9db3ac5fbf988ede
000f0e39b4d0c88441297a05ab5f8da6832c1640 02-Mar-2012 Glenn Kasten <gkasten@google.com> threadLoop merge

Change-Id: Id8e6330ac6be76f9c2debba94f856de87e2d98f7
e8286332f3817a8b7cc4cfd8f6450a3913533660 29-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Shorten thread names"
3e07470f3b122097cacfe5b85cdb1359279a2f33 29-Feb-2012 Glenn Kasten <gkasten@google.com> Prepare for threadLoop merge - active tracks

Continued work on making the copies of threadLoop more similar:
- Remove alias for mActiveTracks in MixerThread and DuplicatingThread.
- Pull in declaration of activeTrack in DirectOutputThread.
- Remove redundant parameter of prepareTracks_l().
- Comment prepareTracks_l().

Change-Id: If1087c1902b454acec01ddfdd9f055f0ca7abf04
bc4b08ba67a0245e092aee8f08ba30ef22d421bf 29-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Update AudioFlinger comments"
c455fe9727d361076b7cead3efdac2d32a1a1d6d 29-Feb-2012 Glenn Kasten <gkasten@google.com> mSuspend comments and usage

Emphasize that playbackthread::mSuspend is a counter, not a bool

Change-Id: I7188e56814e1c54dbc65e560f3627f138257d644
d805b718b1fd2d5407ef665c8d4bb42e63dc71a9 25-Feb-2012 Glenn Kasten <gkasten@google.com> Update AudioFlinger comments

Add comments to enum mixer_state
Note side-effect of lockEffectChains_l
Fix a typo

Change-Id: Ibd51678bac2193201cbcbe081ff5664046fbc494
480b46802bef1371d5caa16ad5454fce04769c57 28-Feb-2012 Glenn Kasten <gkasten@google.com> Shorten thread names

prctl(PR_SET_NAME) limits to 15 characters. Before we had names like
"Binder Thread #" and the counter was cut off :-( Also remove redundant
"thread" at end of name; it's always a thread.

Change-Id: I1f99c2730ba0787ed9b59c15914356cddf698e2f
a17c820c556fddf7ddd96b82b3e9874e340ffafd 28-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "AudioFlinger const methods and parameters"
fadb2c73fce479205432652530663e1e90fd546c 28-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "AudioBufferProvider comments and cleanup"
843a12d146bd64642bf85a4e56c274246e3893a6 27-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Fix tracking of hardware state for dump"
02fe1bf923bbe5789202dbd5810e2c04794562e6 25-Feb-2012 Glenn Kasten <gkasten@google.com> AudioFlinger const methods and parameters

Change-Id: I93ec28024005ed23aa141518092a012a4a7c44c5
37d825e72a6c606553a745da1212590a425996d3 24-Feb-2012 Glenn Kasten <gkasten@google.com> Pull out duplicated copies of silent mode check

Also fix the error handling for the property_get.

This is part of preparation for the threadLoop() merge.

Change-Id: I6405190ea18146d1271575e1dfe9f279e8f36b17
01c4ebf6b794493898114a502ed36de13137f7e5 22-Feb-2012 Glenn Kasten <gkasten@google.com> AudioBufferProvider comments and cleanup

Add comments about which methods implement the AudioBufferProvider interface.

Simplified the definition of kInvalidPts. <stdint.h> is very hard to work
with, there seems to be no way to use it reliably to get INT64_MAX without
having a separate source file, which is ugly because it means kInvalidPts
is not a compile-time constant. So I just deleted AudioBufferProvider.cpp
and used a hard-coded constant instead.

Added a default constructor for Buffer so that the fields aren't random
(especially .raw which is used to determine if the buffer is valid).

Make the pts for getNextBuffer default to kInvalidPTS so code that
doesn't need a pts doesn't have to specify a value.

Rename the parameter to AudioMixer::setBufferProvider to make it clearer.

Change-Id: I87e7290884d4ed975b019f62d1ab6ae2bc5065a5
8abf44d2f2bcd20a2835570efe89d89c19db426a 02-Feb-2012 Glenn Kasten <gkasten@google.com> Fix tracking of hardware state for dump

At end of AudioFlinger::onFirstRef(), the hardware status was being left
in wrong state. It should be AUDIO_HW_IDLE but was AUDIO_HW_INIT.

mHardwareStatus was being set to AUDIO_HW_OUTPUT_OPEN too early, and so
a return would leave it in the wrong state until next hardware operation.

Take the hardware lock for dev->get_parameters, and update mHardwareStatus
before and after.

Keep hardware lock only for the duration of the dev->set_parameters.

Rename two constants in enum hardware_call_state to have the prefix
AUDIO_HW so they follow the naming conventions.

Add comments.

Change-Id: I6c7450b11f9b13adaeef9cec874333e478a58fc0
5cf034d92d901169ca6e36c90475f40715827fcd 21-Feb-2012 Glenn Kasten <gkasten@google.com> Remove TrackBase::mFlags

The bit-field TrackBase::mFlags was supposed to have track-specific
flags in the upper 16 bits, and system flags in the lower 16 bits.

The upper 16 bits of mFlags were initialized in the TrackBase
constructor from the flags parameter of IAudioFlinger::createTrack()
and IAudioFlinger::openRecord(), and the lower 16 bits were cleared.

However, the upper 16 bits of mFlags were never acccessed again.
So really there are no track-specific flags. I left the flags
in the parameter list of createTrack() and openRecord() but made a
note that these should be removed eventually as they are dead.

This leaves only the one system flag "step server failed". I replaced
the bit-field mFlags by bool mStepServerFailed, which is simpler and
slightly faster.

Change-Id: I6650f5487be72791b4a67d73adcd10ffa04e2aa5
9eaa55756c5b245970447019250ce852f5189525 20-Jan-2012 Glenn Kasten <gkasten@google.com> Avoid wp<>::unsafe_get() with a few exceptions

Avoid using wp<>::unsafe_get() except in a log, and other specific cases
when it's known to be safe.

Use more specific subclass types for parameters to avoid down-casts.

When a constructor or method parameter is "this" of an object that is
currently being constructed, it's better to use a raw pointer rather
than either sp<> or wp<>.

Using the raw pointer is safe, provided either:
- it is "this" of an object being constructed (which has sp<> refcount of 0),
- or the caller already holds an sp<>

The raw pointer is simpler and faster, and it avoids the problem of the
sp<> reference count being incremented and then decremented to zero on
scope exit, which would cause the object's destructor to run while the
object is still being constructed.

Also removed some dead code per a review comment.

Change-Id: I7375f64da3aec11b928c33cb01faff186252ef5e
a111792f1314479c649d1d44c30c2caf70c00c2a 26-Jan-2012 Glenn Kasten <gkasten@google.com> Simplify code

Use DefaultKeyedVector::valueFor to avoid extra test
Make local variables as local as possible
No double parentheses
No typedef for single use
No parentheses around indirect function call
No AudioFlinger:: prefix when not needed
Remove unnecessary casts
Remove block with only one line

Saves 128 bytes

Change-Id: I3a87430eeb01b81e7b81a1c38f6fdd3274ec48f3
4ff14bae91075eb274eb1c2975982358946e7e63 09-Feb-2012 John Grossman <johngro@google.com> Upintegrate Audio Flinger changes from ICS_AAH

Bring in changes to audio flinger made to support timed audio tracks
and HW master volume control.

Change-Id: Ide52d48809bdbed13acf35fd59b24637e35064ae
Signed-off-by: John Grossman <johngro@google.com>
6dad4378f2a78d967defc8912ecf47f6ed117584 14-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Fix races related to volume and mute"
99e53b86eebb605b70dd7591b89bf61a9414ed0e 19-Jan-2012 Glenn Kasten <gkasten@google.com> Update comments

We no longer put the filename at start of file.

Change-Id: Ic435b159a23105681e3d4a6cb1ac097bc853302e
6dbc1359f778575d09d6da722b060a6d72c2e7c5 02-Feb-2012 Glenn Kasten <gkasten@google.com> AudioRecord and AudioTrack client tid

Inform AudioFlinger of the tid of the callback thread.

Change-Id: I670df92dd06749b057238b48ed1094b13aab720b
2b213bc220768d2b984239511cd4554a96bc0079 02-Feb-2012 Glenn Kasten <gkasten@google.com> mAudioHwDevs and related cleanup

Inline AudioFlinger::initCheck and remove unnecessary lock.

Remove redundant check of mAudioHwDevs.size().

No need to lock mHardwareLock for each device separately
during initialization.

Use size_t not int to loop through Vector, since size() returns size_t.

Add missing hardware lock for get_mic_mute() and get_input_buffer_size().

Add comments.

Change-Id: Iafae78ef78bbf65f703d99fcc27c2f4ff221aedc
b6333aa8317ce5162ab006c4baed6b0890936dc7 11-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Simplify ThreadBase::exit() aka requestExitAndWait"
0d9302d7830b46542821b3e5f3e4f96942bd3cb3 11-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Move header declarations around for clarity"
858df80948ee64f478782a6a6c06533ba1651ef1 11-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Camel case readability & private disconnect(bool)"
b28686f95daee16edeb5f39af2cd5274ac3dc99f 06-Jan-2012 Glenn Kasten <gkasten@google.com> Simplify ThreadBase::exit() aka requestExitAndWait

We can remove mExiting and use Thread::exitPending() instead.

The local sp<> on "this" in exit() is not needed, since the caller must
also hold an sp<> in order to be calling us. (Unless it was using a raw
pointer, but that would be dangerous for other reasons.)

Add comment explaining the mLock in exit().

Change-Id: I319e5107533a1a7cdbd13c292685f3e2be60f6c4
12018d80add66f5558675614d73fa6549150806e 10-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Move declaration of stream_type_t up earlier"
2f732eb768004c6362fae8a02c60b69c9400b032 26-Jan-2012 Glenn Kasten <gkasten@google.com> Move header declarations around for clarity

Put IAudioFlinger methods in binder opcode order.
Move hardware call state closer to where it is used.
getMode() and btNrecIsOff() are private.

Change-Id: Ie50340b396c39c763f2b155cbc08da8a0d0f2424
58123c3a8b5f34f9d1f70264a3c568ed90288501 03-Feb-2012 Glenn Kasten <gkasten@google.com> Camel case readability & private disconnect(bool)

Change-Id: If66516ed2703e048c5e6ccc6cd431446a024f4a1
6637baae4244aec731c4014da72418d330636ae1 09-Jan-2012 Glenn Kasten <gkasten@google.com> Fix races related to volume and mute

Fix race conditions when setting master volume, master mute, stream
volume, stream mute for a playback thread, and when reading stream
volume of a playback thread. Lock order is AudioFlinger, then thread.

Rename streamVolumeInternal to streamVolume_l, comment, and use it to
implement streamVolume().

Code size reduction:
- Remove dead code: AudioFlinger::PlaybackThread::masterVolume, masterMute, streamMute.
- Change return type of non-binder methods that always succeed from status_t to void.
- Remove virtual from volume and mute methods that don't need it.

This change saves 228 bytes but decreases performance of binder operations
due to the added locks.

Change-Id: Iac75abc1f54784873a667d1981b2e08f8f31e5c9
b7bf796b758e144f94f6ed4d16c21bf22a118fb3 08-Feb-2012 Glenn Kasten <gkasten@google.com> Move declaration of stream_type_t up earlier

stream_type_t is used by AudioFlinger class, so it should be declared there.
This way we don't have to peek into PlaybackThread to get the declaration.

Change-Id: Ie08bab1604699214d1e8df2d48d3fbfbbc436e96
02bbd20cece1785c223ac4ca2ddc635931a80673 08-Feb-2012 Glenn Kasten <gkasten@google.com> Rename type() to streamType()

This avoids possible confusion with thread's type().
Also remove redundant cast "(audio_stream_type_t)".

Change-Id: I320b9177b6c267a102d215f002228bcf988c437a
98ec94c5854daccc3474758524e7f4adfe535ce0 25-Jan-2012 Glenn Kasten <gkasten@google.com> Combine duplicate code & document wp<> in mClients

Change-Id: Iea8cfe8e57563337fb2484a1246ef79d6ad3db18
72ef00de10fa95bfcb948ed88ab9b7a177ed0b48 17-Jan-2012 Glenn Kasten <gkasten@google.com> Use audio_io_handle_t consistently instead of int

- add a comment to nextUniqueId
- made ThreadBase::mId const, since it is only assigned in constructor.

Change-Id: I4e8b7bec4e45badcde6274d574b8a9aabd046837
5e92a7861196ddae14638d4b7a63fc4892b7ef59 30-Jan-2012 Glenn Kasten <gkasten@google.com> Effect UUID inputs passed by pointer are const

Change-Id: I1f5c338bcb7368e3dd8cd5f804b2e6d9fbe087f8
63d2daed17ab749baa80bc808fb5083b688b771b 08-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "AudioFlinger methods const and inline"
86feafe15b3f9609e1e9f64184688c2b6f2e4834 08-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Use virtual destructors"
e98bbd36d67243fe987b09904956550a68af1cc7 08-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Declare more IAudioFlinger methods const"
d5e54f7a36daedc3b2a642d1499c262da04e6280 26-Jan-2012 Glenn Kasten <gkasten@google.com> Remove dead code

mFormat is unused in resampler
mClientTid is unused
local variable pid is unused in dump

Change-Id: Ib156e38029366620bfeff2a13e73471867155a5b
f587ba5b991c7cd91e4df093d0d796bd419e5d67 27-Jan-2012 Glenn Kasten <gkasten@google.com> Declare more IAudioFlinger methods const

This is just documentation, as C++ method const-ness doesn't mean anything
for a binder API. Instead, here const means "no side effects".

Change-Id: Iaa9cd2fe477db10ae9a40cac4f79f0faa9b4e5e6
c59c004a3a6042c0990d71179f88eee2ce781e3c 02-Feb-2012 Glenn Kasten <gkasten@google.com> AudioFlinger methods const and inline

This saves 1063 bytes and probably improves performance.

Change-Id: I11cf0dfd925fbaec75e3d1b806852a538eae5518
c19e22450e6e3d07594c935c7a9522e85e909e82 30-Jan-2012 Glenn Kasten <gkasten@google.com> Use virtual destructors

It turns out to be just a comment, as all except AudioMixer are RefBase.

There are only a few performance-sensitive cases where it's worth thinking
about whether you need a virtual destructor, and the headache usually
outweighs the benefit.

Change-Id: I716292f9556ec17c29ce8c76ac8ae602cb496533
787bae0578fbaab6219ebf23494866b224d01438 03-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Use audio_in_acoustics_t consistently"
87f155d6655b2d3b27e69281a29e85c6407e4d26 03-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "For performance, return large objects by reference"
84afa3b51ac48f84ed62489529ce78cba7fca00e 26-Jan-2012 Glenn Kasten <gkasten@google.com> Constructor initialization and const fields

In constructors, initialize member fields in the initialization list
rather than constructor body where possible. This allows more fields
to be const, provided they are never modified.

Also initialize POD fields in constructor, unless it's obvious they
don't need to be initialized. In that case, put a comment instead.

Remove explicit clear() in destructors on fields that are now const.

Give AudioSessionRef a default constructor, so it's immutable fields can
be marked const.

Add comment about ~TrackBase() trick.

Initialize fields in declaration order to make it easier to confirm that
all fields are set.

Move initialization of mHardwareStatus from onFirstRef() to constructor.

Use NULL not 0 to initialize raw pointers in initialization list.

Rename field mClient to mAudioFlingerClient, and getter from client()
to audioFlingerClient().

Change-Id: Ib36cf6ed32f3cd19003f40a5d84046eb4c122052
5c0ad10b14ec2287f90f95912d98e66eef006e2a 03-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Cleanup thread types"
d05397144be774f2f3623c754e865f51753e4e31 30-Jan-2012 Glenn Kasten <gkasten@google.com> For performance, return large objects by reference

Change-Id: Ibf737018ef1d3c7d717584615dcb2d4ecdb50c99
435dbe6c3ecd04bcb4bd80584064e287ebccd720 30-Jan-2012 Glenn Kasten <gkasten@google.com> Fix const sp<>& in parameter list and return value

EffectModule::addHandle and Client::heap() were declared incorrectly.

As a parameter, an sp<> should be & for efficiency, and for input
parameters it should also be const to protect the caller's value.

But as a return value, an sp<> should have neither const or &. The "e"
in "return e;" might be located on the stack, and if there is "&" then
the caller would see the address of a variable which no longer exists.
Also, an & would make it hard to do "return 0;".
A "const" without & is meaningless in the return type.
(In this particular case, the "e" is a member field, so it was safe.)

Change-Id: I3df5f294214eb15a9d4d596c6d5ef29de97b5c27
de9719b3ec71472e6bf75117152176af51d1a515 27-Jan-2012 Glenn Kasten <gkasten@google.com> Use audio_in_acoustics_t consistently

Change-Id: I0a9dd668fb2e57b1c3ece3190588194974b99062
a3a2cd4072aaa2d93c91251a786eb7323f8d2c27 27-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "AudioStreamIn and AudioStreamOut"
6f5980b75df837231365d238c1b0d6f386363fbb 27-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Declare methods in binder opcode order"
28f52c84c22e129063a576e1269a39ae0cc0bfb3 27-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Use enum effect_state consistently"
114c458f2b80a252ec627add1d5fda2093c79068 27-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Use enum track_state consistently"
aed850d0d3b3c8cf3feaf1438076f33db2a60946 26-Jan-2012 Glenn Kasten <gkasten@google.com> AudioStreamIn and AudioStreamOut

These are immutable, so make the fields const.
getOutput() and getInput() methods are now const.

Change-Id: I128246ebd56ea50b3e542be43f2aa1bcb55f1373
23bb8becff20449a9b1647d5a1a99b14c83f0cce 26-Jan-2012 Glenn Kasten <gkasten@google.com> Cleanup thread types

Use type_t instead of int for thread types.
Initialize ThreadBase::mType in constructor and make it const.

Change-Id: I43d141388b9639e4783c30b97dbda5688bf7555f
90716c5728b37637b2d0a730a721bfc9fad299e0 26-Jan-2012 Glenn Kasten <gkasten@google.com> Declare methods in binder opcode order

This makes it easier to compare interface and implementation.

Change-Id: Ie060e43dec348902abcf40f5a610cec639d6d0d3
29c23c3aee5ae799b3480dc6876a46c46b019710 26-Jan-2012 Glenn Kasten <gkasten@google.com> Use enum mixer_state consistently

Change-Id: I5b71ed20f939dfc4b98143334b7aa064d282f584
28243dd563fee1c82f0fff6cc27b5cbf21fa2585 26-Jan-2012 Glenn Kasten <gkasten@google.com> Use enum effect_state consistently

Also fix indentation

Change-Id: I393ef9e37ffceed5ad4a78df439726ae1fe139df
b853e986caf43408ad95b9014f194aadff385e3c 26-Jan-2012 Glenn Kasten <gkasten@google.com> Use enum track_state consistently

Change-Id: Ie5ebb7befa092e1de1e4df9c6e2d51e6bcfd176a
335787fe43596f38ea2fa50b24c54d0823a3fb1d 21-Jan-2012 Glenn Kasten <gkasten@google.com> Remove AudioFlinger dependencies on client

Change-Id: Ibb591e41a3ca5d7015e2b66b98b8fef5f415fb37
58f30210ea540b6ce5aa6a46330cd3499483cb97 12-Jan-2012 Glenn Kasten <gkasten@google.com> Use audio_format_t consistently, continued

Was int or uint32_t.

When AudioFlinger::format can't determine the correct format,
return INVALID rather than DEFAULT.

Init mFormat to INVALID rather than DEFAULT in the constructor.
Subclass constructors will set mFormat to the correct value.

Change-Id: I9b62640aa107d24d2d27925f5563d0d7407d1b73
aeeb7e219e34d2d657d829913659a4e10e976375 19-Jan-2012 Eric Laurent <elaurent@google.com> resolved conflicts for merge of 05683c85 to master

Change-Id: I7846b7da8c5813b7a9b1f3f71aede0229689ff0d
2774144fa8283f1a7b43e17a53c97dec0c366dd3 18-Jan-2012 Eric Laurent <elaurent@google.com> AudioFlinger: mix track only when really ready (2)

This problem due to the way audio buffers are mixed when
low power mode is active was addressed by commits 19ddf0eb
and 8a04fe03 but only partially. As a matter of fact, when more
than one audio track is playing, the problem is still present.
This is most noticeable when playing music with screen off
and a notification or navigation instruction is played: in this case,
the music or notification is likely to skip.

The fix consists in declaring the mixer ready if all active tracks
are ready. Previous behavior was to declare ready if at least one track was
ready. To avoid that one application failing to fill the track buffer blocks other
tracks indefinitely, this condition is respected only if the mixer was ready
in the previous round.

Issue 5799167.

Change-Id: Iabd4ca08d3d45f563d9824c8a03c2c68a43ae179
ad0f6cc5e115ca167ff122c83451b46d85c590ac 17-Jan-2012 Glenn Kasten <gkasten@google.com> Remove dead setVolume() and mVolume[2]

Change-Id: I94b835434093e920432614eb5007101e87758f32
0696400a6bb9abbed62b3b9c6aa105495dc600a2 17-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Use audio_mode_t consistently"
263709e7be37c7040aaef385bc5c9389a9b5f514 06-Jan-2012 Glenn Kasten <gkasten@google.com> Check stream type in AudioFlinger::createTrack

A bad parameter to AudioFlinger::createTrack could cause mediaserver to crash.

Other AudioFlinger stream type cleanup:
- Simplify range check for audio_stream_type_t
- Add comment about mStreamTypes array initialization.

Change-Id: Ia33aa1cce0fdd694b08d9288816ffc097a9543d0
3944e0326a286bcb931551e61e79c033b10d09d4 17-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Fix locking for mMasterVolume and mMute"
613882293184e575a44bff681a3decaefe889e69 17-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Use size_t for frame size"
9806710f5d6722cfc5783c7eca3512451a0f2035 13-Dec-2011 Glenn Kasten <gkasten@google.com> Fix locking for mMasterVolume and mMute

mMasterVolume and mMute are both protected by mutex in AudioFlinger class, but
there were two places where they were accessed without a mutex.

Also make AudioFlinger::mMasterMute private not protected.

Change-Id: Ia3897daeb5c50313df5bcc071824357526237f3e
b9980659501d0428d65d8292f3c32da69d37fbd2 11-Jan-2012 Glenn Kasten <gkasten@google.com> Use size_t for frame size

except in the control block, where we don't have room.

In AudioFlinger::ThreadBase::TrackBase::getBuffer,
read the frame size from control block only once.

Change-Id: Id6c4bccd4ed3e07d91df6bbea43bae45524f9f4e
fff6d715a8db0daf08a50634f242c40268de3d49 13-Jan-2012 Glenn Kasten <gkasten@google.com> Use audio_stream_type_t consistently

At native level it was a mixture of audio_stream_type_t, int, uint32_t,
and uint8_t. Java is still int. Also fixed a couple of hard-coded -1
instead of AUDIO_STREAM_DEFAULT, and in startToneCommand a hard-coded 0

Change-Id: Ia33bfd70edca8c2daec9052984b369cd8eee2a83
f78aee70d15daf4690de7e7b4983ee68b0d1381d 04-Jan-2012 Glenn Kasten <gkasten@google.com> Use audio_mode_t consistently

It was int or uint32_t.
Also make getMode() const.

Change-Id: Ibe45aadbf413b9158e4dd17f2b3bcc6355288d37
c40256146bee58bff09e1c16ef99ea06d31f89f9 11-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Use correct type for hardware call state"
5f29ca38b71506ad7c7cb9925efbddf588e9655b 09-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "By convention const goes before the type specifier"
a4454b4765c5905f14186893b0688be375642283 04-Jan-2012 Glenn Kasten <gkasten@google.com> Use correct type for hardware call state

Change-Id: Ic6d98b129e3ec653df1d8f7e829adf8dccb4f378
54c3b66444ebfb9f2265ee70ac3b76ccefa0506a 06-Jan-2012 Glenn Kasten <gkasten@google.com> By convention const goes before the type specifier

Change-Id: I70203abd6a6f54e5bd9f1412800cc01212157e58
a3a854868a80fd9b9b8720e06a172754943f9417 04-Jan-2012 Glenn Kasten <gkasten@google.com> suspended() and isSuspended() are const

Change-Id: I04b95970b5a645b64e7e64fffd46d868354dda66
f6b1678f8f508b447155a81b44e214475ab634a8 15-Dec-2011 Glenn Kasten <gkasten@google.com> Use the standard CC_LIKELY and CC_UNLIKELY macros

Several source files privately defined macros LIKELY and UNLIKELY in terms
of __builtin_expect. But <cutils/compiler.h> already has CC_LIKELY and
CC_UNLIKELY which are intended for this purpose. So rename the private
uses to use the standard names.

In addition, AudioFlinger was relying on the macro expanding to extra ( ).

Change-Id: I2494e087a0c0cac0ac998335f5e9c8ad02955873
079123ee3d2e20bbc17a7ddbd96ca46bed27898f 16-Dec-2011 Glenn Kasten <gkasten@google.com> Merge "Improve resistance to leaks for ConfigEvent"
f3990f2cc8fd824ae52a880a7b22248e1bdfb192 13-Dec-2011 Glenn Kasten <gkasten@google.com> Improve resistance to leaks for ConfigEvent

A Vector of pointers is risky, as there is no ownership (and the
ThreadBase destructor was not deleting them, so if there were any left
over at end it would leak). Replaced by a Vector of values.

Change-Id: Iddde72dc30134adfcf724dec26cbe0a742509b8c
e0feee3da22beeffbd9357540e265f13b2119cbb 13-Dec-2011 Glenn Kasten <gkasten@google.com> Use NULL not 0 for pointers

Change-Id: Iab3f9abbdab617dc5a599e657ec46a0b0a002eef
162b40bbaf3c3a24f61a6636bef6f80a9c0a31dd 05-Dec-2011 Eric Laurent <elaurent@google.com> audioflinger: fix audio skipping over A2DP

The maximum sleep time allowed in the mixer thread when audio tracks
are enabled but not ready for mixing is derived from the latency
reported by the output stream.
This does not work for A2DP where the latency also reflects encoding, decoding
and transfer time.

Modified activeSleepTimeUs() to take A2DP case into account.

Issue 5682206.

Change-Id: I3784ac01fb6f836b5a6ce6f764fb15347586de35
544fe9b6e9325701df4ab8c1d29774fc13c4cf6c 12-Nov-2011 Eric Laurent <elaurent@google.com> audioflinger: fix noise when skipping to next song

When audio effects are enabled, a noise can be heard at the
beginning of the new song when skipping to next song in music app.

This is because some effects (especially virtualizer) have a tail.
This tail was not played when previous song was stopped because effects were
not processed when no tracks were present on a given session. This is to
reduce CPU load when effects are enabled but no audio is playing.
The tail was then rendered when the new song was started.

Added a delay before stopping effect process after all tracks have been removed from a session.

Issue 5584880.

Change-Id: I815e0f7441f9302e8dfe413dc269a94e4cc6fd95
a85a74a8219c03f2b1d1ef98f3f02e55f89f89a3 19-Oct-2011 Eric Laurent <elaurent@google.com> Fix issue 381905: BassBoostTest CTS tests fail...

When AudioEffectTest is executed, an Equalizer is created
and enabled on a MediaPlayer session. Effects on the output
mix are therefore suspended.
Then the MediaPlayer is released with the effect still enabled.
In this case, Audioflinger::purgeStaleEffects_l() fails to restore
the suspended effects when the effect attached to the released audio session
is removed.
When subsequent tests are executed on output mix effects, these effects cannot be
enabled as they are still suspended.

Fixed purgeStaleEffects_l() to restore suspended effects if the effect removed is enabled.

Also fixed EffectHandle::disconnect() to only restore suspended effects if the disconnected
handle actually has control over the effect.

Change-Id: I67232e7c34680b0cc01abfd57d5d510a524e5d4f
ec35a1416472865dbebc22b10199ad718ed2cc95 06-Oct-2011 Eric Laurent <elaurent@google.com> Fix issue 5381089: problem with A2DP music volume

This problem only occurs when audio effects are present and
the music volume is applied by one effect engine.
When connecting or disconnecting A2DP, audio effects are moved from
one mixer thread to another. When removed from the source thread,
the effect is stopped but it is not restarted when added to the
destination thread.
This regression was introduced by commit 21b5c47e.

Change-Id: I4cc578d8d760ec65b185032b6fda98c739d331bc
b76e90de3c64626fe07a68469d0a59a31c8efb6b 30-Aug-2011 Eric Laurent <elaurent@google.com> Merge "226483: A2DP connected, but music out to speaker"
9f6530f53ae9eda43f4e7c1cb30d2379db00aa00 30-Aug-2011 Eric Laurent <elaurent@google.com> 226483: A2DP connected, but music out to speaker

When the A2DP headset is connected, there is a possible
race condition when the audio tracks are moved from
the mixer thread attached to the speaker output to the thread
attached to A2DP output.
As the request to clear the stream type to output mapping cache in
the client process is asynchronous, it is possible that the flag
indicating to the client audio track to re-create the IAudioTrack
on the new thread is processed before the cache is invalidated.
In this case, the track will be attached to the old thread and
music will continue playing over the device speaker instead of being
redirected to A2DP headset.

Change-Id: Ib2ce1eb5320eaff83287b93779061bf4e7a330df
bee5337da7659b3b7128622ba1f42618b11df5be 29-Aug-2011 Eric Laurent <elaurent@google.com> Audioflinger: reverse logic of BT NREC indication

The interpretation of BT NREC by AudioFlinger to enable
or disable AEC and NS was wrong: NREC to ON (default) means
the phone (Audio Gateway) must enable local AEC and NS.

Change-Id: I88a264e7fc9831c43bbace4f6b585baec73f2006
db7c079f284f6e91266f6653ae0ec198b1c5006e 10-Aug-2011 Eric Laurent <elaurent@google.com> Audio effects: track CPU and memory use separately

Before this change, CPU and memory usage for an audio effect were
registered and checked against the limit by audio policy manager
upon effect instantiation. Even if an effect was not enabled
it would prevent another effect to be created if the CPU load budget
was exceeded, which was too restrictive.

This change adds a method to register/unregister CPU load only when
an effect is enabled or disabled.
It also adds a mechanism to place all effects on the global output mix
in suspend state (disabled) when an effect is enabled on a specific session.
This will allow applications using session effects to have the priority
over others using global effects.

Also fixes some issues with suspend/restore mechanism:
- avoid taking actions when an effect is disconnected and was not enabled.
- do not remove a session from the suspended sessions list when corresponding
effect chain is destroyed.

Change-Id: I5225278aba1ae13d0d0997bfe26a0c9fb46b17d3
3a34befc6fb04a4945a849e8bda8b84e4bf973fe 02-Aug-2011 Marco Nelissen <marcone@google.com> Keep effects sessions active when the caller dies.

Don't remove effects until the session they are in goes away or all
AudioEffects have been explicitly released. This allows the control
panel process to die without stopping the effects.

Change-Id: I4496e5df080230ca1af149dec95c1309ab8ea888
c3e6572e0ff535932b1f6ffb7bcf5acd891675fb 08-Aug-2011 Eric Laurent <elaurent@google.com> Merge "AudioFlinger: protect input/output stream access"
b8ba0a979067a4efb0b3819bf17770793e41c15e 08-Aug-2011 Eric Laurent <elaurent@google.com> AudioFlinger: protect input/output stream access

Some methods would not check that the output orinput stream of a thread
was still valid before calling functions on its interface.
This could cause a crash if those methods where called while the output or
input was being closed by another thread.

Make sure that the output or input stream pointer is cleared before closing the
Always check that the output or input pointer is not null before calling
functions at the stream interface.
Generalize the use of initCheck() method to verify that the output or input
stream is not null.

Change-Id: I9d9ca6b744d011bcf3a7bbacb4a581ac1477bfa5
59bd0da8373af0e5159b799495fda51e03120ea4 01-Aug-2011 Eric Laurent <elaurent@google.com> AudioFlinger: disable AEC and NS with BT headsets

Disable AEC and NS when the Bluetooth SCO headset in use indicates it
implements those pre processings.

Change-Id: I93f3d10b0a27243d5dbff7182639576fc0c6d862
59255e4fc7d8ff52874b85b1988dc0785140cf81 28-Jul-2011 Eric Laurent <elaurent@google.com> Audio Effect Framework: add effect suspend/restore

Add the possibility for the effect framework to suspend
(temporarily disable process) and restore audio effects.
This feature will be usefull to disable pre processing under certain
conditions and better control coexistence of audio effects
on output mix and specific sources.

Change-Id: I79b195982cc48748d5708308fb1647b9c3c34cc6
a7280a59259018d997896c043fd2db95f631f12e 27-Jul-2011 Eric Laurent <elaurent@google.com> Merge "AudioFlinger: fix crash when deleting pre process."
ec437d8d3db79459d7b19e1734e6fe309bd621e8 27-Jul-2011 Eric Laurent <elaurent@google.com> AudioFlinger: fix crash when deleting pre process.

If a pre processing effect is detroyed while enabled and capture is active,
there was a possibility that the effect engine is released by the framework
while still processed by the audio HAL.

The fix consists in not releasing the engine in EffectModule::removeHandle()
but just flag the effect as being detroyed to avoid further calls to functions
on the engine effect interface.
The effect interface is then removed from the audio HAL safely in
EffectChain::removeEffect_l() while holding the EffectChain mutex.

Change-Id: I71fab30d9145062af8644f545a1f1d4d3e7e7f02
feb0db689c17dced50afaee54c659f1676e2d505 22-Jul-2011 Eric Laurent <elaurent@google.com> Fix issue 4604090: notification sound interrupted.

The problem is that the audio HAL fails to acquire the wake lock when playing the notification.
This is because of a change that removed the mediaserver process form the system group for honeycomb.

The fix consists in requesting the wake lock from PowerManagerService when AudioFlinger mixer
wakes up.

A consequence of this change is that audio HALs or pcm drivers do not have to hold wake locks
anymore as in the past.

Change-Id: I4fb3cc84816c9c408ab7fec75886baf801e1ecb5
1d2bff0e588afe183a1baaae731519b4e957bbdb 25-Jul-2011 Eric Laurent <elaurent@google.com> AudioFlinger: add dump of audio pre processing.

Dump of media.audio_flinger service was only listing effects on output threads.
Moved the dump of effect chains from PlaybackThread to ThreadBase class so that
pre processings on RecordThread are also listed.

Change-Id: If8bc74023c12b9c2371f1b300743b156ceca7b87
7c7f10bd4fda9a084e5e7f0eb3a040dfcbf01745 18-Jun-2011 Eric Laurent <elaurent@google.com> Audio framework: support for audio pre processing

Audio effect framework is extended to suport effects on
output and input audio path.

AudioFlinger: Support for audio effects and effect chains is
moved from PlaybackThread class to ThreadBase class so that
RecordThread can manage effects.
Effects of type pre processing are allowed on record thread
only. When a pre processing is enabled, the effect interface handle is
passed down to the input stream so that the audio HAL can call the
process function. The record thread loop calls the effect chain process
function that will only manage the effect state and commands and skip the
process function.

AudioRecord: The audio session is allocated before calling getInput() into
audio policy serice so that the session is known before the input theead is
created and pre processings can be created on the correct session.

AudioPolicyService: default pre processing for a given input source are
loaded from audio_effects.conf file.
When an input is created, corresponding effects are created and enabled.

Change-Id: Id17119e0979b4dcf189b5c7957fec30dc3478790
7394a4f358fa9908a9f0a7c954b65c399f4268e6 14-Jun-2011 Dima Zavin <dima@android.com> audio: update for audio/audio_policy header names/locations

Change-Id: I36c49352eee57559403cd1597f56a8485a360289
Signed-off-by: Dima Zavin <dima@android.com>
0d255b2d9061ba31f13ada3fc0f7e51916407176 25-May-2011 Jean-Michel Trivi <jmtrivi@google.com> Use channel mask instead of channel count for track creation

Record and playback objects (resp AudioRecord and AudioTrack)
are created using a channel mask, but this information is lost
in the mixer because only the channel count is known to
AudioFlinger. A channel count can always be derived from a
channel mask.

The change consists in:
- disambiguiting variable names for channel masks and counts
- passing the mask information from the client to AudioFlinger
and the mixer.
- when using the DIRECT ouput, only verifying the format of
the track is compatible with the output's for PCM.

Change-Id: I50d87bfb7d7afcabdf5f12d4ab75ef3a54132c0e
65580f9adf6c4d98449ad0716488f9fe3869aa5a 28-May-2011 Eric Laurent <elaurent@google.com> Removed interface to load audio effects libraries

Removed unused functions allowing dynamic loading of audio effects libraries
from effects factory API.

Change-Id: I06cc5a51dc10aca87c7a8687bbb874babd711eca
e1315cf0b63b4c14a77046519e6b01f6f60d74b0 18-May-2011 Eric Laurent <elaurent@google.com> New effect library API

Moved and renamed media/EffectApi.h to hardware/audio_effect.h
Modified the effect library API to expose a library info structure
containing an interface functions table.
Also removed enums for audio channels, audio format and devices
from effect API and use values from system/audio.h instead.

Modified effects factory to support new library interface format and
load libraries and efffects listed in audio_effects.conf file.
The file audio_effects.conf is first loaded from /vendor/etc and
then from /system/etc/audio_effects.conf if not found.

Modified existing effect libraries to implement the new library interface.

Change-Id: Ie52351e071b6d352fa2fbc06c3846686f8c45df9
6708b9a3fb654f5623ba5a696288fdba310a5e1a 13-May-2011 Eric Laurent <elaurent@google.com> Merge "Fix audio effect framework issues"
b469b9490b3cd9e0f0466d9b9ab228f6c793b82e 09-May-2011 Eric Laurent <elaurent@google.com> Fix audio effect framework issues

Fix two issues in audio effect framework reported by partners.

1 - Fixed duplicated audio buffer sent to effect process function when
pausing a track.
Modified Effectchain::process_l() function to clear the effect chain
input buffer before calling the effect process functions when no track
is active on the session. Previous code was clearing the buffer after
calling the process functions and when transitioning from active
to inactive, the last processed buffer was passed again once to effect
process function before being cleared.

2 - Fixed potential mutex cross deadlock when disconnecting an effect
while playback is active. This is because EffectChain::process_l()
was calling PlaybackThread::hasAudioSession() thus creating an inversion
in the mutex lock order (EffectChain mutex locked before ThreadBase mutex).
The fix consists in removing the call to hasAudioSession() from process_l()
and requires each effect chain to keep count of the number of audio tracks
attached to it (previously only the active tracks were accounted for).

Change-Id: Iee4246694ea8c7a66c012120c629d72dd38f9c35
64760240f931714858a59c1579f07264d7182ba2 11-May-2011 Dima Zavin <dima@android.com> update for new audio.h header location

Change-Id: Ic4c62c4037800802427eb7d3c7f5eb8b25d18876
Signed-off-by: Dima Zavin <dima@android.com>
5a61d2f277af3098fc10b2881babca16391362da 20-Apr-2011 Dima Zavin <dima@android.com> audioflinger: don't do work in constructor, instead do it in onFirstRef

Change-Id: I22d9e01821816c3beb52b014330386c7fd2f0411
Signed-off-by: Dima Zavin <dima@android.com>
799a70e7028a4d714436c3a744a775acfbd31aae 19-Apr-2011 Dima Zavin <dima@android.com> audioflinger: enumerate all the possible audio interfaces

Keep track of the primary interface that handles the master volume,

Change-Id: Ib0701fccff8d8783a99035a241ab7c8ec75c00ac
Signed-off-by: Dima Zavin <dima@android.com>
fce7a473248381cc83a01855f92581077d3c9ee2 20-Apr-2011 Dima Zavin <dima@android.com> audio/media: convert to using the audio HAL and new audio defs

Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
117cd9286424888c1c5bf202ebf1e08ae1e6affe 04-Feb-2011 Glenn Kasten <gkasten@google.com> Merge "Bug 3366885 Remove LVMX switch"
db130fbd3ccd37e247e49494a84f8a9841ecd593 04-Feb-2011 Glenn Kasten <gkasten@google.com> Bug 3366885 Remove LVMX switch

Change-Id: I0bf98c6f85f00b3296874571e1c049dcc4e2fcca
eda6c364c253ba97ee45a3adeb8c2b45db1f81db 02-Feb-2011 Eric Laurent <elaurent@google.com> Fix issue 3371080

Modified default volume control logic in AudioService:
1 IN_CALL volume if in video/audio chat
2 NOTIFICATION if notification is playing or was playing less than 5s ago.

Modified silent mode:
- now also affect MUSIC stream type
- entering silent mode when VOL- hard key is pressed once while selected
stream volume is already at 0 (except for VOICE_CALL stream).
- exiting silent mode when pressing VOL+ hard key while in silent mode

Play sound FX (audible selections, keyboard clicks) at a fixed volume.

Modified audio framework:
- isStreamActive() method now implemented in AudioPolicyManagerBase (previously AudioFlinger)
- iStreamActive() now specifies a time window during which the stream is considered
active after it actually stopped.

Change-Id: I7e5a0724099450b9fc90825224180ac97322785f
f5aafb209d01ba2ab6cb55d1a12cfc653e2b4be0 18-Nov-2010 Eric Laurent <elaurent@google.com> Fix issue 3157123.

Use a Mutex wherever atomic operations were used in AudioTrack,
AudioRecord, AudioFlinger and AudioEffect classes.

Change-Id: I6f55b2cabdcd93d64ef19446735b8f33720f8dbc
8f45bd725549436eeacd12ee69349e2332ed8da5 31-Aug-2010 Eric Laurent <elaurent@google.com> Audio Effects: fix problems in volume control.

- Fixed click when re-enabling effect during the turn off phase:
make sure the effect states where effect is processed are the same
where volume control is delegated to effect.
- Fixed click when effect is deleted while still active: do not apply
volume ramp if an effect having volume control was just removed from the
effect chain.

Also fixed a crash when PCM dump is enabled in effect bundle wrapper.

Change-Id: Ib562f5cf75c69af75df0e862536262e2514493e4
25cbe0ecd6df8be7e40537c5d85c82f105038479 19-Aug-2010 Eric Laurent <elaurent@google.com> Fix issue 2929440

Fixed regression introduced by change a54d7d3d7dd691334189aab20d23c65710092869 in audioflinger mixer thread:
When the output stream is suspended, the sleep time between two writes must match the actual duration
of one output stream buffer otherwise the playback rate is not respected.

Change-Id: Ic5bebe890290d1f44aeff9dd3c142d18e26fff2a
39e94f8f723d445447fdee0822291e664b631f60 28-Jul-2010 Eric Laurent <elaurent@google.com> Allow creation of an audio effect on a session with no audio tracks.

This is necessary to allow creating and enabling an effect attached to a particular player
session before the playback is started. As a matter of fact, the implementation of the mediaplayer
does not create the AudioTrack before playback starts.

Change-Id: I1266e8885f9d756acc949303321aaac0fbf83e34
25f4395b932fa9859a6e91ba77c5d20d009da64a 28-Jul-2010 Eric Laurent <elaurent@google.com> Audio effects: modified command() parameter types.

The type of the cmd, cmdSize and *pReplySize parameters of the effect control interface command()
function have been modified from int to uint32_t. This is more consistent with their role.

Change-Id: I84d289fc262d6753747910f06f485597dfee6591
de070137f11d346fba77605bd76a44c040a618fc 13-Jul-2010 Eric Laurent <elaurent@google.com> Audio policy manager changes for audio effects

Added methods for audio effects management by audio policy manager.
- control of total CPU load and memory used by effect engines
- selection of output stream for global effects
- added audio session id in parameter list for startOutput() and stopOutput().
this is not used in default audio policy manager implementation.

Modifications of audio effect framework in AudioFlinger to allow moving and reconfiguring
effect engines from one output mixer thread to another when audio tracks in the same session
are moved or when requested by audio policy manager.
Also fixed mutex deadlock problem with effect chains locks.

Change-Id: Ida43484b06e9b890d6b9e53c13958d042720ebdb
f997cabca292d70d078ae828e21c28e6df62995f 19-Jul-2010 Eric Laurent <elaurent@google.com> Fixed problems in audio effect volume control.

Fixed the following problems in audio effect volume control in AudioFlinger:
- Make sure that the volumes returned by EffectChain::setVolume_l() are correct even is
no change is detected since last call
- Do not use isEnabled() to validate volume control but mState >= ACTIVE instead as the volume control
must be also active in STOPPING and STOPPED states.

Change-Id: Id62da3164fad500ee8a5efd6cd78c77e8fdcb541
cab112421da6e8eac19ffddbbe3d76067cffee78 15-Jul-2010 Eric Laurent <elaurent@google.com> Several improvements in audio effects volume control.

- Fixed crash when deleting an effect chained before an effect having volume control
(not need to set both in effect descriptor).
- Volume control changes from one effect to another if needed according to effect enable state
- EFFECT_CMD_SET_VOLUME is only sent when their is an actual change in volume

Change-Id: Ieebaf09157e2627366023569d95516646e03e26c
5462fc9a38fa8c9dff434cd53fa5fb1782ae3042 15-Jul-2010 Mathias Agopian <mathias@google.com> added BinderService<> template to help creating native binder services

Change-Id: Id980899d2647b56479f8a27c89eaa949f9209dfe
65ab47156e1c7dfcd8cc4266253a5ff30219e7f0 15-Jul-2010 Mathias Agopian <mathias@google.com> move native services under services/

moved surfaceflinger, audioflinger, cameraservice

all native services should now reside in this location.

Change-Id: Iee42b83dd2a94c3bf5107ab0895fe2dfcd5337a8