History log of /frameworks/av/services/audioflinger/AudioFlinger.cpp
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
291f824e02ff517a34cfe50220b4e2b402ee998d 19-Oct-2012 Glenn Kasten <gkasten@google.com> Remove active track when thread goes to standby

Bug: 7369232
Change-Id: I7ff9f525dad4be0aef562a53015b06ee7d3d50f1
a045dcafd2b77036210f5b72e79d745ad4c1b848 16-Oct-2012 Jean-Michel Trivi <jmtrivi@google.com> Fix track estimation for presentation complete

Audio tracks were not using the right latency estimation for
signalling the completion of their presetation. This caused
the synchronization mechanism between playback and record to be
off, and a synchronized recording would contain some of the audio
that was meant to be over once recording would start.
Use the playback thread's latency reporting which takes the audio
pipe into account.

Bug 7237669

Change-Id: I23a907a53ad0b0d68d246789ec595a77a79fced5
087dd8e7232e4c009e9121ab7e8c37985522c9ad 27-Sep-2012 Glenn Kasten <gkasten@google.com> Disable audio watchdog

It's not critical, and is wasting power

Bug: 7241714
Change-Id: I6ad4375f0000c92529688723dbe0ff0caa809c5d
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
842c5d9553f3f8e97d04ed1bd0d37e4851240654 26-Sep-2012 Glenn Kasten <gkasten@google.com> Revert "Don't wait for presentation complete if terminated"

This reverts commit 44cda3a4e7ca3de0db9cb49145def3803b03ebb4

Change-Id: I7fd29b77690dab057ac966a42fb198b2772f092c
cc0f1cfb69ce8b8985fc2c0984847a06a13ad22d 24-Sep-2012 Glenn Kasten <gkasten@google.com> Implement android.media.AudioManager.getProperty()

Bug: 6635041
Change-Id: I3386a4a6c226bc4eceaf65556119e4fb15f73224
b6ba2fd0d61a4844c153e17843fbe2c841c4bf57 25-Sep-2012 Eric Laurent <elaurent@google.com> audioflinger: improve record start and thread exit

Use broadcast() instead of signal() on the
thread wake up condition when starting record or requesting thread
exit to make sure that if another thread is waiting for the same
condition (e.g binder thread calling setParameters()) the mixer
thread will be woken up.

Bug 7184317.

Change-Id: I3154a4509ca7af6ffae5236e522b0fab8e75ed06
1afc26db11b71c43f63a0f72a45a803f1a7910dd 24-Sep-2012 Eric Laurent <elaurent@google.com> fix end of track presentation on suspended output

The code detecting the end of an audio track presentation before
removing it from the active track list is based on the
count of audio frames sent to audio HAL. When an output stream
is suspended (e.g. A2DP when SCO is active), this count does not
change and a track in stopped state will never be removed from
active track list causing the mixer thread to never release
the wake lock.

The fix consists in incrementing the audio HAL frame count even
if the output is suspended.

Also fix a problem in getRenderPosition() when the output is suspended.

Bug 7167534.

Change-Id: I3be836cbbea29b65dc087199cac6a1cd84c0a41d
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
4362f5300162ed55f56cf167eccf7f2e5b89d435 13-Sep-2012 Jean-Michel Trivi <jmtrivi@google.com> Handle audio HAL returning 0 bytes in record thread

Enter standby when HAL returns an error, but also consider 0 bytes
returned as NOT_ENOUGH_DATA.

Change-Id: Ica83142310e9c176f936e0440571a6034cbc575f
52762410dbc9189cd92a4094a1dbd4cfe8e71cb6 13-Sep-2012 Jean-Michel Trivi <jmtrivi@google.com> Don't deadlock on AudioRecord start when reads return 0

When calling start() on an AudioRecord with a HAL that
returns 0 on a read() operation, the start blocking
condition was never unblocked.
Add a boolean to track the first read operation so the returned
number of bytes (mBytesRead) is only evaluated after that
first read.

Change-Id: I8c735a00d48cd6a0da467ccdf75d3616b38f6afa
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
7aa25591769685ae0e8349b3ca3534c724484375 03-Aug-2012 Glenn Kasten <gkasten@google.com> Remove dead code

RecordThread::isValidSyncEvent() returns false, so most of
RecordThread::setSyncEvent() is never executed.

Change-Id: I0cf848beb46a367a45126d2df3073c5afa2ca59c
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
0dbb356050d0db9e0043dd43045c1864a933332b 03-Aug-2012 Glenn Kasten <gkasten@google.com> Simplify AudioFlinger::PlaybackThread::isValidSyncEvent()

Change-Id: I3e4af69b929d4ca04afaac26c7e41c89fce25b9c
d23eedca9b5a1812891c05d89850ab7ee707040d 02-Aug-2012 Glenn Kasten <gkasten@google.com> Discard setSyncEvent() return value

setSyncEvent() returns a status_t which is sometimes ignored.
Emphasize this is intentional by casting to void.

Change-Id: Ic614988347cba36bd2504d7ad321594a355b0d9d
3ed292031dc50c56110cdadb1e3778117e3be76a 08-Aug-2012 Glenn Kasten <gkasten@google.com> Replace hard-coded "2" by a constant and comment

Bug: 6679403
Change-Id: I6c2701f9afedc26540dfad0b4e23348bbc4cb01a
2c3b2da3049627264b7c6b449a1622f002210f03 03-Aug-2012 John Grossman <johngro@google.com> AudioFlinger: fix timed audio

(cherry picked from commit e20ac92c564a2f4e8123885807abdf0a78de0dd7)

> AudioFlinger: fix timed audio
> Addresses Bug 6900517.
> Finish up support for timed audio in the new FastMixer world. Pay special
> attention to remaining lock-less and voluntary yield free on the FastMixer
> thread. This fixes audio playback for Q on JB.
> Change-Id: Iaf815e58a1b1d0a0190051794bec8dc5c9231785
> Signed-off-by: John Grossman <johngro@google.com>

Change-Id: I9bd687acc345a05867af48e71116690fdb0ce1b5
Signed-off-by: John Grossman <johngro@google.com>
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>
44cda3a4e7ca3de0db9cb49145def3803b03ebb4 01-Aug-2012 Glenn Kasten <gkasten@google.com> Don't wait for presentation complete if terminated

Change-Id: Ia04cf6c620693457dca87b4ffea5dd0fe71efdce
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
1d491ff06f4b9c90ff24fe953b90d0843eaf1c04 16-Jul-2012 Glenn Kasten <gkasten@google.com> Fix races in AudioRecord stop()

Change-Id: Id0ac1915f57fef4a938c7f90989c1162a8b6c51c
69d799679c8c0308e42057e7b5ad63a7ae806480 19-Jul-2012 Glenn Kasten <gkasten@google.com> Use upmix/downmix utility routines

Change-Id: I9ae2ec938fb695ec576ea008a42205325af7bbf1
e65c89113232d070dd9153c54ca19301bca7a162 21-Jul-2012 Eric Laurent <elaurent@google.com> fix audio effect not destroyed when needed.

commit a5f44eba contained an error which made that audio effect
modules where not destroyed but left in the destroyed state
rendering them unuseable after being released.

Bug 6805168.

Change-Id: Ia4e683b3c970ffd01846c482fde73d799ff219de
33e6e35b03a726e35203e97550f32154c91d5f13 17-Jul-2012 Glenn Kasten <gkasten@google.com> Miscellaneous audio record fixes

Call AudioSystem::stopInput() if exitPending() after wait() returns.

Acquire lock before clearing mActiveThread.

Change-Id: Ia55e4c4b3accc65ad5479cbdc094fd919152af9f
cd2d61016527bf48bd2e9a920bb3fdbb875eb3e4 19-Jul-2012 Glenn Kasten <gkasten@google.com> Use constructor to initialize instead of memcpy

Also don't check for thread parameter as it's always non-NULL

Change-Id: Id23ded1370556ef3f76f81f5f0c6fa644bcba681
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
5ad92f620fbbb6a8281f10169a23d38e3601e07a 19-Jul-2012 Glenn Kasten <gkasten@google.com> Revert 94479fd5405642c67efd14cebe722feb9cbe6e77

Change-Id: I5ca78d5462badf541868785b2ba2e3f6d0cf492a
bb4350d3b9e9485ae59e084de270f86aecef8066 04-Jul-2012 Glenn Kasten <gkasten@google.com> Use audio_devices_t more places

Change-Id: Id0ace02ca6f480a6c46e11880acf6cdc66d83184
a189a6883ee55cf62da1d7bf5bf5a8ab501938a4 20-Feb-2012 Glenn Kasten <gkasten@google.com> Use struct assignment instead of explicit memcpy

for POD structs effect_descriptor_t and effect_config_t

Change-Id: Ib2fc47f85fb65ed91b0abb1f87217c49b5eb571d
fd4e20c226eca185fc789de761beae64855bfbbb 04-Jun-2012 Glenn Kasten <gkasten@google.com> Run audio at priorities 2 and 3 above kernel 1

Also run the watchdog at same priority as the fast mixer.
requestPriority() originally used only the caller's pid to decide which
cgroup to assign, but in the future it might look at the priority also.
So it's safer to use same priority as the fast mixer to be sure we
run in the same cgroup.

Bug: 6461925
Change-Id: Ia59c93e4b22dacbb6746bfa6ad491be7b72f2b8d
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
bf04a5d7f287fc712e0ed91849dc85c90c1e182d 12-Jul-2012 Glenn Kasten <gkasten@google.com> Simplify AudioRecord::getInputFramesLost()

This also fixes a benign race in reading mActive without a lock.

Change-Id: I19e953d4f275e5c266ca1ca3fece7b6c02ad1707
39c54f68804c1ce5c85ec588f3c2c63447a807b4 09-Mar-2012 Glenn Kasten <gkasten@google.com> Remove dead code

Change-Id: If22a6c4e572b0734eba0c5a7ce29a2c61c581e5d
4fe1ec4f40b58abff6cec147aa786cb65698161a 28-Feb-2012 Glenn Kasten <gkasten@google.com> Fix check for invalid channel count

Change-Id: Id9e3dce0e3d5971786212d3f70e17a17e32ce92b
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
9f34a36d9cdb9595c288e50ffe00da038bc8abb9 21-Mar-2012 Glenn Kasten <gkasten@google.com> Whitespace and indentation

Change-Id: I424052b4ff9218147a5cfc8e6dcd67fe8105d229
7d6c35bf132a46c0a8a9826491882495fc98bd8c 02-Jul-2012 Glenn Kasten <gkasten@google.com> Move constant initializations from onFirstRef() to constructor

Change-Id: I57f55b0bd1edee105c58c3a055f95f1e4a2c9646
6648821933dc06c0b09ab2c8b32135edddcd4291 21-Jun-2012 Glenn Kasten <gkasten@google.com> AudioFlinger::getBuffer() always returns non-NULL

Change-Id: I543d3db507597cacbfdad5d9ea71732137fe54fb
94479fd5405642c67efd14cebe722feb9cbe6e77 10-Jul-2012 Glenn Kasten <gkasten@google.com> Fix build

Revert after system/core audio_devices_t is submitted

Change-Id: I5a8ee1a7b711e834501e927f41c62efa6a6600b6
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
d5903ec1332630f2992a6f0d5ca69d13a185c665 18-Mar-2012 Glenn Kasten <gkasten@google.com> Compare sp<> to 0 and raw pointers to NULL

Change-Id: I50ff8a010d349d1d7e3dffa04a6331814c2128b0
a01992a0675a06df7d0bbe7b977207dd0c33fdc7 02-Jul-2012 Eric Laurent <elaurent@google.com> am dbbd5b86: am 109347d4: audioflinger: fix regression in attachAuxEffect().

* commit 'dbbd5b860a3a26bea3376410f75f27530d9cd10b':
audioflinger: fix regression in attachAuxEffect().
109347d421413303eb1678dd9e2aa9d40acf89d2 02-Jul-2012 Eric Laurent <elaurent@google.com> audioflinger: fix regression in attachAuxEffect().

Commit 717e1286 introduced a regression in PlaybackThread::Track::attachAuxEffect()
when called with an effect ID of 0 to detach the auxiliary effect.

It is normal in this case that AudioFlinger::getEffectThread_l() returns 0.

Bug 6768757.

Change-Id: I7430bd1aad2f68da38f7c3e4794e7ad657bfc6be
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
362ebcbf100f7fccd37551c77e67c4faa7241b63 24-May-2012 Glenn Kasten <gkasten@google.com> DO NOT MERGE Remove log spam for fast track denied

Bug: 6531054
Change-Id: Iedf58e810a157aae88b5900da27c81054c437058
1d6573032ecde54a466ca32951e101b41a05c797 14-Jun-2012 Glenn Kasten <gkasten@google.com> am 48a0bfa6: am 49dd5cf3: Merge "Log track name on obtain/releaseBuffer warnings" into jb-dev

* commit '48a0bfa6f731386f2794ff36d6677d56e98fc6ea':
Log track name on obtain/releaseBuffer warnings
0c9d26d187017f7fb028ab52a0fbc6395142faa4 31-May-2012 Glenn Kasten <gkasten@google.com> Log track name on obtain/releaseBuffer warnings

This should help diagnose problems by allowing us to correlate
the logs with the dumpsys media.audio_flinger output.

Change-Id: I8c7c592b4f87d13b0f29c66ce7a2f301a0f063c9
8cc3651c04e48b755dcc579bfa4f9a7e9391d6e7 12-Jun-2012 Glenn Kasten <gkasten@google.com> am 57d9b728: am c15d6657: Add audio watchdog thread

* commit '57d9b72812d25dff1c33e37b8475a469accd0919':
Add audio watchdog thread
c15d6657a17d7cef91f800f40d11760e2e7340af 30-May-2012 Glenn Kasten <gkasten@google.com> Add audio watchdog thread

Change-Id: I4ed62087bd6554179abb8258d2da606050e762c0
796078f96c8a15fefcea70e666b22ea67cc35301 08-Jun-2012 Glenn Kasten <gkasten@google.com> am ea5008d8: am 28ed2f93: Reduce underruns in screen off, esp. with EQ

* commit 'ea5008d8abfdf1479b4efa266cdb7c842d168aa9':
Reduce underruns in screen off, esp. with EQ
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
bdbf0c6f1a11d6fd9b71aa765f7de39e248557ba 08-Jun-2012 Eric Laurent <elaurent@google.com> am 98e2e030: am a4f7e0e9: audioflinger: fix duplicating thread standby

* commit '98e2e030231ce99a4796d34d39a7517078d6735e':
audioflinger: fix duplicating thread standby
a4f7e0e9a0e92a063f1b3a08988cf46e2cf1fa94 08-Jun-2012 Eric Laurent <elaurent@google.com> audioflinger: fix duplicating thread standby

The code that waits for a track presentation to be complete
before disabling it caused a regression for duplicating threads.
Because of the way output tracks activity is managed, the number
of frames output by the duplicating thread would never
reach the target set for a track to be considered presented.
The track would not be removed from active list and the thread would
not go to standby and keep its wakelock held.

Bug 6606922.

Change-Id: I4b46b420ac4cbf79a86b6791ae6589d407b01c92
349d79668ea4ca078400995c70d5d3201e8d9ea1 07-Jun-2012 Marco Nelissen <marcone@google.com> am d89dea16: am e35a55fb: Merge "Take latency and current time into account for visualization" into jb-dev

* commit 'd89dea161ff3c3da515c02928cad4a9c73b23312':
Take latency and current time into account for visualization
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
a88ed026402d92d699c336aa11267616007e4a9d 05-Jun-2012 Eric Laurent <elaurent@google.com> am 717f9b73: am 67c0a58e: audioflinger: various fixes on direct output

* commit '717f9b7392b0fc3ba15b018c923b85e64c0662b5':
audioflinger: various fixes on direct output
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
b86f92d552c391f5d2471bd4f979135be2578578 05-Jun-2012 Eric Laurent <elaurent@google.com> am e506084e: am ab9071b8: audioflinger: fix active sleep time calculation.

* commit 'e506084e1c22f8f1797b9dc3eb789a699efd45c3':
audioflinger: fix active sleep time calculation.
ab9071b8d1b375418eb797c9a790da71de644344 04-Jun-2012 Eric Laurent <elaurent@google.com> audioflinger: fix active sleep time calculation.

When an audioTrack underruns, the audioflinger mixer thread
sleeps for a certain time to give the app a chance to recover.
This time is based on the reported audio HAL latency.
Some audio HALs implementing deep buffering have a variable
latency and this creates a problem if the sleep time is cached
when the output stream is opened and not updated afterwards.

This change derives the active sleep time from the mix buffer
framecount instead of the latency. This is more conservative
but works for variable latency.

Bug 6588525.

Change-Id: Ia892fc290fe06f836565c3ae15f7a2ce026c88c6
24a2fd0113da60785ce5af5dd905f8aaf9e0f0a1 04-Jun-2012 Glenn Kasten <gkasten@google.com> am f335f182: am 39993085: State queue dump

* commit 'f335f182e4b50249ac34b41da3566ddb016cc816':
State queue dump
5385b7b0f5d922ee38f8a54f11ee4462ef4b5e29 04-Jun-2012 Glenn Kasten <gkasten@google.com> am 2c00676c: am 1295bb4d: Fast track dumpsys

* commit '2c00676cd34d78460ad610a4a4fd7a68544a7b7d':
Fast track dumpsys
399930859a75d806ce0ef124ac22025ae4ef0549 31-May-2012 Glenn Kasten <gkasten@google.com> State queue dump

Bug: 6591648
Change-Id: Iac75e5ea64e86640b3d890c46a636641b9733c6d
510ba8b812d88f62968a2c9b0b638fff6d99ee84 03-Jun-2012 Glenn Kasten <gkasten@google.com> am f45dc2f1: am bf0d21fb: Count underruns for normal tracks also

* commit 'f45dc2f1751d9ac19826b99fab0b226f5c84570a':
Count underruns for normal tracks also
1295bb4dcff7b29c75cd23746816df12a871d72c 31-May-2012 Glenn Kasten <gkasten@google.com> Fast track dumpsys

Bug: 6591648
Change-Id: I696f51c682e7233ba690d97da26012084989b412
bf0d21fb1310e8677caa53b90e8c3aecebc7fc13 31-May-2012 Glenn Kasten <gkasten@google.com> Count underruns for normal tracks also

Bug: 6591648
Change-Id: Iff9cabe392bb2ce97062603adb9c9dc7aa4170d5
35d7bfc359b3aa87ade92d1ab55c6992418cad48 01-Jun-2012 Glenn Kasten <gkasten@google.com> am 92e5ee95: am 7dc5b165: Merge "Fix fast track leak if out of normal track names" into jb-dev

* commit '92e5ee9548542513791a70c81e0cd3fd70397269':
Fix fast track leak if out of normal track names
893a05479c96f911d02beb0443da3ed6508143a7 30-May-2012 Glenn Kasten <gkasten@google.com> Fix fast track leak if out of normal track names

Bug: 6580402
Change-Id: I3ac7f012062c35833147f47ba822eb4bf532a824
529e888738a91ca70cbdeeabd982f8fb2947780c 30-May-2012 Eric Laurent <elaurent@google.com> am 8c07f759: am 3bdb4fbf: Merge "audioflinger: fix effect problem during underrun" into jb-dev

* commit '8c07f7599a757fe51dc54253c480067cf01f13d3':
audioflinger: fix effect problem during underrun
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
6d80297a55ab12759ee00b7f99fa97584b430da0 24-May-2012 Eric Laurent <elaurent@google.com> am 0cc62570: am f436fdcf: audioflinger: change session check in createTrack.

* commit '0cc6257030d3e6c649ea3ad807ecb9327ceb5b3e':
audioflinger: change session check in createTrack.
f436fdcf93bd417fd3c9d2a8b19fd221d894b5e3 24-May-2012 Eric Laurent <elaurent@google.com> audioflinger: change session check in createTrack.

Do not refuse to create a track on an output thread if the same session
is present on another thread. It is now possible that two tracks
with the same session ID are on different threads if one can use deep
buffering and the other can't.

In this case, move effects attached to this session to the output
thread ion which the new track is created.

Bug 6530324.

Change-Id: I9019b3ee382e374c89d2319033afcfa7f886e4c4
852fca99e25db8d2180c2622ca55fca676490a08 24-May-2012 Glenn Kasten <gkasten@google.com> Remove log spam for fast track denied

Bug: 6531054
Change-Id: Iedf58e810a157aae88b5900da27c81054c437058
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
99c99d00beb43b939dedc9ffb07adb89f6a85ba5 15-May-2012 Glenn Kasten <gkasten@google.com> systrace for audio

Trace fast track buffer fill status for underruns etc.

Move the definition of macro to Android.mk.

No overhead if disabled.

Change-Id: If0e83e21b61b059ca38f543f8a6ffb58e08c79ee
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
9017e5e0ebad9664bb7b6f2057e5bb29c852c64f 15-May-2012 Glenn Kasten <gkasten@google.com> Increase normal mixer's pipe to fast mixer

Change-Id: I330925c7d07b6adb30b773bda3657e4efef9ae9b
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
4adcede0dc54a85c31abaf139921aebd7a072d8e 14-May-2012 Glenn Kasten <gkasten@google.com> Reduce video frame drop rate

The video playback engine depends on having relatively precise audio
progress updates for its A/V sync and frame drop calculations. For small
audio HAL buffer sizes, this was not a problem, but when the HAL buffer
size was > 12 ms, the normal mix buffer size became > 24 ms and this
then caused video problems. The new formula tries to keep the normal
mix buffer size within a closer tolerance of 20 ms to 24 ms.

Also use consistent term: multiplier instead of multiple.

Bug: 6479613
Change-Id: I903bad74461908e8c8f0a61e99ab5e24d5c44433
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
808e7d16504cbe5b28bb88c31afb2542ab488965 12-May-2012 Eric Laurent <elaurent@google.com> AudioFlinger: fix global effects suspend logic

Audio effects on the output mix should not be suspended when effects
on the output stage (post processing) are enabled.

Change-Id: I2e1c08fa9358ea3cbaec68856738d504b1be54e4
d8e6fd35ec2b59ee7d873daf1f1d9d348221c7bc 07-May-2012 Glenn Kasten <gkasten@google.com> Use audio tag for system tracing

Disabled by default; uncomment ATRACE_TAG to enable

Change-Id: I99af894022a859ee5644bd853cfd8a48e4735ff9
e213c86d36414a8fc75e37c52999522fe09c7328 25-Apr-2012 Glenn Kasten <gkasten@google.com> dumpsys fCnt and flags

Previously displayed TrackBase::mFrameCount but not control block frameCount.
Now displays both.

Also display the track flags in control block.

Change-Id: Ie53781d4784633d78b6f928d69ebd494d8f110ec
31dfd1db7a4d2228d9642008af6f3dd744368eb6 01-May-2012 Glenn Kasten <gkasten@google.com> Disable fast track log spam

except for "denied by client" and "denied by server"

Change-Id: I133ab747933729cc1f386813ee06ece055bdb294
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
300a2ee9327c05fbf9d3a5fd595b558097c7c5e8 25-Apr-2012 Glenn Kasten <gkasten@google.com> Fast mixer configuration

Add compile-time option for when to use fast mixer.

Double HAL frame count for fast tracks due to SRC, and make the normal
frame count multiplier an even number for compatibility. Sample rate
conversion can result in underruns if the HAL frame count is used as is,
due to jitter.

Change-Id: Ia1f8da1b8ac247d9807acfce3c318161db000905
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
e0fa467e1150c65a7b1b1ed904c579b40f97c9df 24-Apr-2012 Glenn Kasten <gkasten@google.com> Move frame count calculations for fast tracks

For fast tracks: move the default and minimum frame count calculations
from client to server. If accepted, the default and minimum frame count
is the fast mixer (HAL) frame count. If denied, the default and minimum
frame count is the same as it currently is for normal tracks.

For normal tracks: there is no change yet, preserve legacy behavior for
now but add a FIXME to change this later.

Bug fix: the test for buffer alignment matches channelCount was wrong.

Bug fix: check for 8-bit data in shared memory, which isn't supported.

- in set(), only call AudioSystem::getOutputSamplingRate() when needed
- in createTrack_l(), only call AudioSystem::getSamplingRate() and
AudioSystem::getFrameCount() when needed

Change-Id: I79d2fe507db1a8f7bb094c71da8a129951dbb82f
1dc28b794587be22c90a97070d928f94586db638 24-Apr-2012 Glenn Kasten <gkasten@google.com> Use scheduling policy service

Change-Id: I3c09da1dc0de5039d0c15ce7fb2bc373fa398712
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
c95cfbb87d0ac5e773037019a96bfc29972d4b4e 12-Apr-2012 John Grossman <johngro@google.com> TimedAudioTrack: Optimize the queue trim operation.

Hand merge from ics-aah

> TimedAudioTrack: Optimize the queue trim operation.
> Don't perform the end PTS calculation for each buffer during trimming.
> Instead, only calculate the ending PTS of a buffer if there is no next
> buffer in the queue. This optimization assumes that the buffers being
> queued are in monotonic media time order (a fair assumption for now)
> and that the timestamps in the audio are contiguous (not a requirement
> for this API, but a reality of how it is being used right now).
> In the case where the audio is discontinuous on purpose, it is
> that this optimization will cause the system hold one extra buffer
> which it could have safely trimmed. It should not be much of an issue
> since in real life the audio is almost always contiguous, and as long
> as the media clock is running and the mixer is mixing, the buffer will
> be used up and discard as part of the normal flow anyway.
> Change-Id: I00061e85ee7d5651fcf80751646c7d7415894a14
> Signed-off-by: John Grossman <johngro@google.com>

Change-Id: I0054b58e1389fa005aa990cb5710caf4af7b706a
Signed-off-by: John Grossman <johngro@google.com>
8d314b709fdd81bb64bdaa8d72a0b19c355cefb9 19-Apr-2012 John Grossman <johngro@google.com> TimedAudioTrack: Fix continuity threshold handling.

Hand merge from ics-aah

> TimedAudioTrack: Fix continuity threshold handling.
> Fix issues with continuity threshold handling; notably
> + If the steady-state continuity threshold is exceeded, be sure to
> clear the on-time flag. Failure to do this will result in the
> system picking a new mix point which simply satisfies the
> steady-state continuity threshold instead of the startup threshold.
> Since we are putting a discontinuity in presentation anyway, we
> really want to pick a perfect point, not just an OK point.
> + Tighten the steady-state continuity threshold. It was currently set
> to 100mSec which is enormous. 4mSec (the new setting) is much more
> appropriate. On systems with a VCXO (like tungsten) this should
> never be wrong by more than a sample. If TimedAudioTracks are ever
> to be used on VCXO-less systems, this threshold should probably be a
> a parameter configurable by applications on a track by track basis
> so they can make the tradeoff between allowed error and frequency of
> disruptive corrections.
> + Reset the on-time flag if the mixer provides no PTS during a mix
> operation. This makes for a convenient way for the HAL to reset
> timed tracks when it makes changes for delay compensation across
> multiple outputs.
> Change-Id: I2cb23de5a3d1f75618abc1c8ab903db883837aa8
> Signed-off-by: John Grossman <johngro@google.com>

Change-Id: Ibd28c9d290494b0b19eb01caf2d9bfdef606a9b4
Signed-off-by: John Grossman <johngro@google.com>
9bd23229fdec1657398abc682ccccfce1c95f8aa 16-Apr-2012 Jean-Michel Trivi <jmtrivi@google.com> Fix multichannel downmix pause bug on video player

Pausing a video player will cause the track audio mixer to be
disabled, which causes the downmixer to be deleted. When reenabled,
the track channel mask hasn't changed but the downmixer is there
Fixed by:
- instanciating a downmixer when the AudioMixer track
gets initialized (in getTrackName(), now taking a channel mask
as input), and deleted when in deleteTrackName().
- when the channel changes on a track, check whether it
needs a downmixer or not. Preparing a track for downmix
automatically removes the old downmixer if there was one.

Also: initialize the track downmixerBufferProvider field
when AudioMixer is instanciated, so we can safely call
delete on it in AudioMixer's destructor, in case
deleteTrackName() wasn't called before the mixer was

Change-Id: I589b0781cda5b3c82f85b561c52b08546cac21f8
0ca3cf94c0dfc173ad7886ae162c4b67067539f6 18-Apr-2012 Eric Laurent <elaurent@google.com> rename audio policy output flags

Change-Id: I27c46bd1d1b2b5f96b87af7d05b951fef18a1312
acb86cccbd9d245439a04cef0bcefa589addaa4c 16-Apr-2012 Jean-Michel Trivi <jmtrivi@google.com> Configure the resampler with the correct channel count when downmixing

When a track needs to be downmixed and resampled, it gets downmixed
first before being resampled. Therefore the resampler needs to
be configured with the channel count of the output of the downmixer
instead of that of the track.
Removed frame size checks that don't apply anymore now that we support
frame sizes that are not powers of 2 (e.g. 12 for 5.1 16bits), and
changed test performed for every buffer during playback into an

Change-Id: Ia220f00ee382f4f7848b661c58555bdca664e194
f7ffb8bf0a58037f0bc9662c5275005a4e539948 14-Apr-2012 Eric Laurent <elaurent@google.com> audioflinger: update to new audio HAL.

Updated audio flinger to new function prototypes
for open_output_stream() and open_input_stream().

Check audio hw device version when loading a module and
reject devices with a version different from current one.

Change-Id: I9d4c81a1f59a15db78b3989417c2a411c638fe88
b388e531613730572067e193b6b66afb5d042233 14-Apr-2012 Eric Laurent <elaurent@google.com> Fix ALOG_ASSERT in AudioFlinger

Fix broken ALOG_ASSERT in updateFramesPendingAfterTrim_l() introduced by
commit 1c345196.

Change-Id: Ie1b2653069283f23ff0367f2628828e37fb0749c
d3030da2ac3c0ebb8b7bdf38418263caf405b863 12-Apr-2012 John Grossman <johngro@google.com> Fix the build

forgot to upload final fixup during merge. sry about that

Change-Id: I2ddd2c08d8efa83c0a8d1e378ae4c28686145154
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>
3f9c84c0a5af83fceb8669390e2d71b75ec7b550 04-Apr-2012 Eric Laurent <elaurent@google.com> audio pre processing: test code for dual mic

Added functional test code to validate effect API for
multi mic simplementations.

Also fixed warning in AudioFlinger.

Change-Id: I07be4d2e4d17791d3626c804ba3e9f87ff26d05a
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
a1472d9883e35edd280201c8be3191695007dfd4 30-Mar-2012 Marco Nelissen <marcone@google.com> Make AudioTrack/AudioRecord handle more than 2^32 frames

Change-Id: I471815012c6a113ec2c4dd7676e8fa288a70bc76
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"
63c1faa8dea7feb90255d31ef2a133d8f2818844 21-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "Update comments"
7153494670bdac8b650cb10b8b1838651e0ca418 21-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "Whitespace"
a26ff6f22f4e86d09514c2819237bd9748455018 21-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "new doesn't fail on Android"
9a5f413a21b320fea0607e653bc75b1a4f0e7a2e 21-Mar-2012 Eric Laurent <elaurent@google.com> am a48285c4: am 165ee4c5: am 14958e21: Merge "audioflinger: fix issue with camcorder and A2DP" into ics-mr1

* commit 'a48285c4f22ffc43f1771ebd1ff35dcec48db2c7':
audioflinger: fix issue with camcorder and A2DP
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
ea7939a079b3600cab955760839b021326f8cfc3 14-Mar-2012 Glenn Kasten <gkasten@google.com> Whitespace

Fix indentation, and add blank lines in key places for clarity

Change-Id: I57a0a8142394f83203161aa9b8aa9276abf3ed7c
17a736c3e1d062d7fc916329eb32aef8935614af 14-Feb-2012 Glenn Kasten <gkasten@google.com> Update comments

Change-Id: I327663a020670d0a72ff57bd0b682e2ce0528650
a03567676e8766828ff970b87e13bc4c97b23473 19-Mar-2012 Glenn Kasten <gkasten@google.com> new doesn't fail on Android

Change-Id: I5079a3bf31097dd0807b2d806d5f8d3cff2077ab
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
9d7b4c074205609271f61e1a4741ac0c524a1795 19-Mar-2012 Eric Laurent <elaurent@google.com> am 14958e21: Merge "audioflinger: fix issue with camcorder and A2DP" into ics-mr1

* commit '14958e21c12f922d7501d32c3bec05109eb342d5':
audioflinger: fix issue with camcorder and A2DP
89d94e79dad032fb18ddc655e6068e4231d3f0aa 17-Mar-2012 Eric Laurent <elaurent@google.com> audioflinger: fix issue with camcorder and A2DP

Some audio HALs do not support well a device selection of 0 (no device)
received on an input stream.

This can happen because of a problem in the audioflinger code that handles
the forwarding of the output device selection to the record thread for use by
the pre processing modules that need it. If the output device is 0 (meaning
no op, which happens when stopping playback over A2DP) audioflinger could not
detect it was an output device selection and would forward it to the input
stream (see AudioFlinger::setParameters() and RecordThread::checkForNewParameters_l().

Issue 6179641.

Change-Id: Idae534521866538e0d12ba259a2834f402a922e2
dfaf549e3e310bc22444f4404b19f4907b24c286 15-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "AudioFlinger playback thread CPU measurement in Hz"
190a46f7c84e160386610c0c4cecb9767fb5503b 06-Mar-2012 Glenn Kasten <gkasten@google.com> AudioFlinger playback thread CPU measurement in Hz

Log statistics on CPU usage in Hz in addition to wall clock time

Use CPU statistics for all playback threads, not just MIXER
(but they are disabled by default by a compile-time debug macro).

ThreadCpuUsage library:
- Move statistics out of the library and leave that up to the caller
- Add API to determine a CPU's frequency

Change-Id: Ia1011123146e641fcf210ef26e78ae2b4d3b64ad
3b229ed97c0dfc85a8cf881341e29e595e0edea7 14-Mar-2012 Eric Laurent <elaurent@google.com> Merge "audioflinger: more info in dumpsys"
612bbb57c59397a540e96f06bdd16e437a583af5 14-Mar-2012 Eric Laurent <elaurent@google.com> audioflinger: more info in dumpsys

Added TID and io handle to AudioFlinger threads dump.

Change-Id: Ib1a856f3bad55c73e4c395b5e59d57435f4b9a4c
d3cee2f0f649c01e1153d593cbe723887b8e0ba0 14-Mar-2012 Glenn Kasten <gkasten@google.com> Break circular dependency on media player service

Bug: 6165157
Change-Id: I3c85bbcaf31f3cb9a009e273f7b6284015eb3bd8
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
fd83fbf5ead098070bae674b20e6f87f45ab5d4c 13-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "Clean up assertion checks"
aa4397f07c43bd83bc3100b749401dc3d15e7622 13-Mar-2012 Glenn Kasten <gkasten@google.com> Fix bug where mMixerStatus was set to IDLE

Change-Id: I55df1738fb7ba17ba6caeea6a17557526eac17a7
5798d4ebf236357a4b13246f40e52b90a34d09a4 08-Mar-2012 Glenn Kasten <gkasten@google.com> Clean up assertion checks

Use ALOG_ASSERT instead of assert.
Use compile-time asserts where appropriate.
Fix typo in an ALOGV.

Change-Id: I58f1c1ffc14319a022c88b5a88b8d0368660da8b
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
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
d69549665d412f1f6ebad48ad8cd05133ada8728 09-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "Replace hard-coded 3 by FCC_2 to simplify searches"
083c154162c88a9f63aeaa10a4b52dd454bda9ff 09-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "Don't ask policy manager about invalid stream type"
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
639dbee79140956c43926344c23af765f6e0c9a5 07-Mar-2012 Glenn Kasten <gkasten@google.com> Don't ask policy manager about invalid stream type

Change-Id: If50fbff9d34045d1398984da48da7e6428a74491
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
d4513b09123deebf8023b73a82d2d46d35806cea 07-Mar-2012 Glenn Kasten <gkasten@google.com> Make applyVolume private to DirectOutputThread

Change-Id: I7ca4a59505857cbd106b6f274c66e9580dead271
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"
92b8360fe9c3174dc0edaaab4b746d8a3d3f831f 07-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "Fix indentation for re-organized code"
438b036deaf4413503567a75a2861c000d21da5b 06-Mar-2012 Glenn Kasten <gkasten@google.com> Rename updateWaitTime since a lock is held

Change-Id: I9bb978cbd0debf5b21676467060f72eebafea3e6
952eeb27682a9b2ddfa761f24b6eb5e18fe5d814 06-Mar-2012 Glenn Kasten <gkasten@google.com> Fix indentation for re-organized code

Change-Id: I63471cebdbd095b7ad4e481611b785f9b02c7941
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
91cda8bdf8358a8b977a44e016b60a70bc1a5ee9 29-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Pull in declaration of effectChains to inner block"
73ca0f5837d5448f7a5eb159a09cd0ebe82b4de9 29-Feb-2012 Glenn Kasten <gkasten@google.com> Pull in declaration of effectChains to inner block

Change-Id: I09eacf72124942abd604132b9f4e774b1236fcf3
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
688a64030834ea2f52cc9765676ddf6aa34df767 29-Feb-2012 Glenn Kasten <gkasten@google.com> Mark similar and different sections in threadLoop

Most of these comments will be removed after the threadLoop merge.

Note: the trivial change in assignments to mixBufferSize, and the
comments about "tracks to remove" is to make them all identical.

Change-Id: I3b1a33a7f2cd12ad557a1986bb71f6171161974a
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
a3873833d518e032138cf70188b6f33cd7acec3d 28-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Unlock effect chains in the middle of two if's"
3e9c3a1d34960cd258f294d31135ab6bf76179d5 28-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Simplify removeNotificationClient"
a17c820c556fddf7ddd96b82b3e9874e340ffafd 28-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "AudioFlinger const methods and parameters"
cfbd62616ab2b12f0fee603658f04e5827cc7f8f 28-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Fix theoretical race condition in addOutputTrack"
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"
e628d515888baadba75442128678e747e930ed58 27-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Make threadLoop() logs identical"
d3cee0b1f77baa4fb7a049eb757e9f5006890726 27-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Move declaration of mixerStatus to inner block"
a3b09254d44cd8d66ec947abe547538c4cfeaa89 20-Jan-2012 Glenn Kasten <gkasten@google.com> Simplify removeNotificationClient

No need to check for presence of item before removing
(but we do lose the log of the previous value).

Change-Id: I2838430824de5f257f2ee15db0c22b1920c67d08
02fe1bf923bbe5789202dbd5810e2c04794562e6 25-Feb-2012 Glenn Kasten <gkasten@google.com> AudioFlinger const methods and parameters

Change-Id: I93ec28024005ed23aa141518092a012a4a7c44c5
c0b52836d07f823732f0ff98ca5ca9d7f5730cb8 24-Feb-2012 Glenn Kasten <gkasten@google.com> Make threadLoop() logs identical

Change the wording of the logs in the various copies of threadLoop()
to be identical. This will make it easier to merge them soon.

Change-Id: Idfa181e437738712c784dc7f746cac79f83d2931
5d4eeeaf76ebe177b43e87b2a9df5e55e39021f0 24-Feb-2012 Glenn Kasten <gkasten@google.com> Move declaration of mixerStatus to inner block

mixerStatus was being declared (and initialized) too early,
which also resulted in a duplicate initialization. Moved
the declaration into the block where it is actually used.

Change-Id: Ifdcfefe362a5efe3493dd616cdb44645c6f9aed5
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
04743e99e71c0da012508c7119f414027654ee94 24-Feb-2012 Glenn Kasten <gkasten@google.com> Unlock effect chains in the middle of two if's

As part of the upcoming threadLoop() merge, this CL makes it clearer
what are the similar and different parts before and after unlocking
effect chains.

In each threadLoop(), the old code was:

if (sleepTime == 0) {
// A
// B
} else {
// C

The new code is:

if (sleepTime == 0) {
// A
if (sleepTime == 0) {
// B
} else {
// C

Also this is slightly slower by one "if", it has the advantage of making
it much more obvious about what is done before and after the unlock,
and also to see the similarities and differences among the various
copies of threadLoop().

Change-Id: I7bf4369d2dcb072573ec43b7e52c637f0097dc00
5ce96d97feafc6989f6141bb2633eae3d87ddf28 24-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Pull CPU statistics code out of threadLoop()"
b6b740629c9f11535086e744465bada03f26df11 24-Feb-2012 Glenn Kasten <gkasten@google.com> Fix theoretical race condition in addOutputTrack

This is not a real race, because addOutputTrack was only called in two
places, and in both places there could be no other threads referencing
the DuplicatingThread instance.

Those two places are:
- the DuplicatingThread constructor, which is of course safe
- openDuplicateOutput - this is safe because it's called immediately
after the new DuplicatingThread, and there are no sp<> either in the
constructor or here which could cause onFirstRef() to do Thread::run().

But for safety in case addOutputTrack is ever called somewhere else,
or there are sp<> created earlier, it is safer to take the thread lock.

Change-Id: I1502d014fa37ec5dbf4bf40d3e2884af311cd5e9
83efdd0fc08cd5aedf50b45741a8a87be8dc4b41 24-Feb-2012 Glenn Kasten <gkasten@google.com> Pull CPU statistics code out of threadLoop()

This is to prepare for the threadLoop() merge

Change-Id: I118c7d5c6b011b5d5b95ec7d63fb03feb166a9cf
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
d6fd85a157ce2054b2304e6d171fa87ae09c363d 22-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Avoid wp<>::unsafe_get() with a few exceptions"
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
f063b49e95c28d63a58215ebda892a5fee4204cc 18-Feb-2012 Glenn Kasten <gkasten@google.com> Fix build warning

warning: pointer of type 'void *' used in arithmetic
warning: enumeral and non-enumeral type in conditional expression

Change-Id: I7b8d626a636145ef648e3b5d0e77068216dd012e
3b81acab52b7140c1b8b20be2d67be3e221637e7 28-Jan-2012 Glenn Kasten <gkasten@google.com> Remove bit fields to improve performance

uint16_t enabled is (mostly) changed to bool in a separate CL

Change-Id: Ied9f8c034b2479cee9a8778cee7b8ff92ae75b7b
1b094ee8f7fe7eca65bf3d2f983ba95eef6db93d 17-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Simplify code"
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
fe5b3ba4b332d5fc9aa4f453434329b9f38768c2 13-Feb-2012 John Grossman <johngro@google.com> Put a bandaid on a segfault in timed audio track handling.

Add a bandaid to prevent a segfault which can occur while handling
timed audio buffers. There is a deeper problem which should
eventually be addressed, but for now this fix should prevent any

The deeper problem is as follows.

When the AudioFlinger mixer gets data to mix from an AudioTrack, it
ends up getting a structure filled out which points into an IMemory
region owned by the AudioTrack. Unfortunately, this structure is not
holding a refcount on the IMemory which it points into. If the
IMemory refcount hits 0 and the chunk of RAM is retuned to the binder
heap it came from, there can still be a Buffer object being held by
the AudioFlinger mixer which points into the region of memory which
was retuned to the binfer heap. If AF reads from this buffer, it
could read corrupt data (if the region of memory gets handed back out
to a writer), or it could segfault (if the heap has been freed and the
pages unmapped). Similar problems could happen if AF attempts to
write to the buffer, heap corruption in one case, segfaulting in the

In the past, this has not been an issue for AF, because tracks
allocate a single IMemory (which serves as a ring buffer) and the
IMemory lives for as long as the track lives. As an artifact of the
way the code came out, the mixer cannot be holding a Buffer structure
pointing into the IMemory which used to be owned by a track if the
track no longer exists. Tracks cannot come into or out of existence
during a mix operation, which is the only thing which makes this safe.

TimedTracks work differently, however. Timed tracks each allocate a
small binder heap, and then hand out IMemory instances broken out of
this heap. The heap lives as long as the track, so the worst which
could happen here is that a TimedTrack's IMemory gets returned to the
heap while there is still a buffer structure in flight pointing into
the memory region, then the region gets handed out again and
overwritten by new data causing the mixer to mix the wrong audio. The
timing to cause this to happen is very difficult to encounter, and you
to generate the timing conditions required, you need to be in a pretty
bad failure state where audio is already breaking up and skipping, so
its unlikely that anyone would notice (which is why I'm band-aiding
the segfault and letting the deeper issue slide for now).

In general, however, it might be a good idea to revisit this buffering
design. On principal, if someone is going to hold pointers into a
refcounted object, they should be holding a ref on the object at the
same time. Failure to do this will usually lead to a situation where
there are corruption or segfault issues, or to a system where the
refcounted object's lifetime must be implicitly managed very carefully
in ways which are usually non-obvious and are easy to break by new
engineers on a project.

Change-Id: Ib391075395ed0ef46a03c37aa38a82d09e88abeb
ef7740be67a4d7b6b033ebed59c3d4a9c74a2c18 09-Feb-2012 John Grossman <johngro@google.com> Fix a segfault in AudioFlinger.

Check the string returned by a HAL's implementation of get_parameters
for NULL before attempting to make use of it. That way, we won't
bring down the mediaserver because of a poorly written HAL.

Change-Id: Ic99d7b004520d7d6347842a681c0595e889b68ea
Signed-off-by: John Grossman <johngro@google.com>
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"
d9b9b8d09e7471b0ffa21cfa9f944ef4ad300a71 14-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Update comments"
99e53b86eebb605b70dd7591b89bf61a9414ed0e 19-Jan-2012 Glenn Kasten <gkasten@google.com> Update comments

We no longer put the filename at start of file.

Change-Id: Ic435b159a23105681e3d4a6cb1ac097bc853302e
8d6a2449a91f5116d7243ab039393195ebd663fe 08-Feb-2012 Glenn Kasten <gkasten@google.com> Use size_t and ssize_t with Vector

Use size_t with size() and ssize_t with indexOfKey(). Exception:
use ssize_t for backwards loops, and indices that are overloaded as a
marker or error code.

Change-Id: Ibf2a360af4539b72b09c818dda22ea2a0de92431
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
44deb053252a3bd2f57a007ab9560f4924f62394 06-Feb-2012 Glenn Kasten <gkasten@google.com> Factor out and speed up permission-checking code

Use the caching permission check for dump to save IPC.

Cache getpid() to save kernel call for other permission checks.

The C runtime library getpid() can't cache due to a fork
race condition, but we know that mediaserver doesn't fork.

Don't construct String16 on the stack.

Change-Id: I6be6161dae5155d39ba6ed6228e7683e67be34ed
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"
858df80948ee64f478782a6a6c06533ba1651ef1 11-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Camel case readability & private disconnect(bool)"
c8ad36bbb30e99e49026cba78e5e0f83db5cb0f6 11-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Use mul from audioutils"
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
0ba18ec1b343a8de70924f87630dd1f329b00fe6 10-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "No newline or space at end of ALOG format string"
8b5980798ca06e57b1284e6e23fa220e1207bf41 10-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Fix typos in ALOG for pid vs tid"
58123c3a8b5f34f9d1f70264a3c568ed90288501 03-Feb-2012 Glenn Kasten <gkasten@google.com> Camel case readability & private disconnect(bool)

Change-Id: If66516ed2703e048c5e6ccc6cd431446a024f4a1
5b9ff43995f6a6b819d9ad37dd8cdc5ad4a088d7 09-Feb-2012 Glenn Kasten <gkasten@google.com> Use mul from audioutils

I verified that the disassembled output is identical.

Change-Id: I34a76f0842ebc4aef2c923e079e38d0bc1f98b5c
23d82a9bc9a43b49ba684ba40875b91db310d3b9 03-Feb-2012 Glenn Kasten <gkasten@google.com> Fix typos in ALOG for pid vs tid

Change-Id: I6dc70f137d0ff8a86427ab8882a81886e1de0782
90bebef5669a9385c706b042d146a31dca2e5d9b 28-Jan-2012 Glenn Kasten <gkasten@google.com> No newline or space at end of ALOG format string

Change-Id: I0bef580cbc818cb7c87aea23919d26f1446cec32
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
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
dbfafaffe2e97eaf8d74ec6b6c468418a1ad2443 26-Jan-2012 Glenn Kasten <gkasten@google.com> Simplify destructors

Remove explicit clear() when the order doesn't matter.

Change-Id: I5931bc7ef5f681c7ce329aa9ec0a6e46d34a56c5
5e92a7861196ddae14638d4b7a63fc4892b7ef59 30-Jan-2012 Glenn Kasten <gkasten@google.com> Effect UUID inputs passed by pointer are const

Change-Id: I1f5c338bcb7368e3dd8cd5f804b2e6d9fbe087f8
0a20fa9c41c96e31fa20e071074a4b6e7f6c41c3 08-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Use pid_t not int"
b61ec89bb0c701b3bd06eb658f854230681f8b39 08-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Don't double destruct audio_track_cblk_t"
63d2daed17ab749baa80bc808fb5083b688b771b 08-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "AudioFlinger methods const and inline"
e616d4e6de6d53ddebbc3d7fb381af94589c2232 08-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Improve performance for sp<> on stack"
1579d7948117e3e6541b0cfda02cc5234a3280ea 08-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Use 0 not NULL for sp<> and wp<>"
e98bbd36d67243fe987b09904956550a68af1cc7 08-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Declare more IAudioFlinger methods const"
1a0ae5be3d1273cba12584b33830d859510fbf82 03-Feb-2012 Glenn Kasten <gkasten@google.com> Don't double destruct audio_track_cblk_t

Fortunately audio_track_cblk_t doesn't have a destructor, but for clarity
remove the double destruction.

Also add warning not to add any virtuals to audio_track_cblk_t.

Change-Id: I70ebe1a70460c7002145b2cdf10f9f137396e6f3
bb001926447d0b7dc71ca8bb3c9856f3136d8f4c 03-Feb-2012 Glenn Kasten <gkasten@google.com> Use pid_t not int

Change-Id: Iad1c2fd4152e94080ad8c65c13ddf4519fc2ed27
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
7378ca506e4e20c2b2d4e94a131cf1b95831adb5 20-Jan-2012 Glenn Kasten <gkasten@google.com> Use 0 not NULL for sp<> and wp<>

Change-Id: Id1f0c89acefaceed6cb9ca7c165fce895e46d85b
787bae0578fbaab6219ebf23494866b224d01438 03-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Use audio_in_acoustics_t consistently"
a0d68338a88c2ddb4502f95017b546d603ef1ec7 28-Jan-2012 Glenn Kasten <gkasten@google.com> Use NULL not 0 for raw pointers

Use if (p != NULL) instead of if (ptr)

Change-Id: Iaec3413a59ccbf233c98fcd918cc7d70ac5da9fa
87f155d6655b2d3b27e69281a29e85c6407e4d26 03-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "For performance, return large objects by reference"
f81e97e4ec8b01965a5b36987f886cf5001f71ff 03-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "No need to check a wp<> for 0 before promote()"
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"
62da7fbd60bee2dd57f503126266e9f04311d400 03-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Make AudioTrack control block volume field private"
d45ee9d9d61af0791c7c3c51f8d4fe6794ef02a5 02-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Fix const sp<>& in parameter list and return value"
d05397144be774f2f3623c754e865f51753e4e31 30-Jan-2012 Glenn Kasten <gkasten@google.com> For performance, return large objects by reference

Change-Id: Ibf737018ef1d3c7d717584615dcb2d4ecdb50c99
090f01963e215f895020a31e22368cd44e086ce3 30-Jan-2012 Glenn Kasten <gkasten@google.com> Improve performance for sp<> on stack

Combine default constructor for sp<> immediately followed by assignment,
as the reference-counting is relatively slow. Also return sp<> directly
rather than via local variable, for the same reason.

Change-Id: If55931f1e407994f6591ddde41b53db72fb4fc40
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
e9dd0176933d6233916c84e18f3e8c0d644ca05d 28-Jan-2012 Glenn Kasten <gkasten@google.com> Unconditional delete

Don't check that pointer is non-NULL before delete.

Don't leave deleted member fields non-NULL, except in a destructor,
since it could be misleading in a dump or debugger. (mRsmpOutBuffer)

Change-Id: Ic0492a6b752f74a67f4c96dfb89ca2de4e69eecf
77c1119ea0b5cb32287088ceeeb7e3b6bd14a85d 25-Jan-2012 Glenn Kasten <gkasten@google.com> No need to check a wp<> for 0 before promote()

Also remove unnecessary wp<> local variable.

Change-Id: I620e67b5d559d28616f8e00609a525cfe19c5ddc
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"
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
b853e986caf43408ad95b9014f194aadff385e3c 26-Jan-2012 Glenn Kasten <gkasten@google.com> Use enum track_state consistently

Change-Id: Ie5ebb7befa092e1de1e4df9c6e2d51e6bcfd176a
9365ea9bf2e439b3e71abbabe22ce7382ebc4b3a 25-Jan-2012 Eric Laurent <elaurent@google.com> am 535b0264: am 7eeaf3f0: Merge "AudioFlinger: refine mixer sleep time logic" into ics-mr1

* commit '535b0264a4cfa790e549bd9cd09980788f1375f4':
AudioFlinger: refine mixer sleep time logic
84e19873fde204d73628ba1b5ca9e3f5778574fa 24-Jan-2012 Eric Laurent <elaurent@google.com> am 7eeaf3f0: Merge "AudioFlinger: refine mixer sleep time logic" into ics-mr1

* commit '7eeaf3f07aa6fb10639d9f96c1367eb98c3e8839':
AudioFlinger: refine mixer sleep time logic
7c5aea0a8d9b422999483f96a2566f77ff11abf2 24-Jan-2012 Eric Laurent <elaurent@google.com> am 41773d46: Merge "DO NOT MERGE Revert "Revert "AudioFlinger: mix track only when really ready (2)""" into ics-mr1

* commit '41773d46556aa47d4322ff89fdaf7d1345c2d1f2':
DO NOT MERGE Revert "Revert "AudioFlinger: mix track only when really ready (2)""
21e4b6ed00e814bffc70895847a4944d7a190020 24-Jan-2012 Eric Laurent <elaurent@google.com> AudioFlinger: refine mixer sleep time logic

When an AudioTrack is in underrun state, the AudioFlinger mixer will
sleep for a short period of time to give the app a chance to fill the
AudioTrack buffer. If the AudioTrack is still not ready during next mixing round,
the mixer will proceed with other tracks.

If an application keeps a steady underrun condition, the AudioFlinger mixer will
alternate between ready and not ready states. In the longer term this will cause the
audio HAL to underrun.
There is a mechanism to reduce the sleep period if the mixer is not ready several times in a
row but this mechanism is defeated by the alternating ready/not ready conditions.

The fix consists in only increasing sleep time if the mixer is ready for at least two
consecutive times.

Issue 5904527.

Change-Id: Id0139bca9be8c4e425ec6d428515c4d8f718e8c9
eaa0b5cc2f7723e9b25298126d0dcb48c56d5dac 24-Jan-2012 Eric Laurent <elaurent@google.com> DO NOT MERGE Revert "Revert "AudioFlinger: mix track only when really ready (2)""

This reverts commit b918035d34422a2041b6ec8c09c566bb93345b40.

Change-Id: I093bcfa56ad54a080b930208b6b79169d33581fb
15dfda272eec983508b89fb8bc9ca6f2bb825496 24-Jan-2012 Justin Ho <justinho@google.com> am fee5a860: Merge "DO NOT MERGE Revert "AudioFlinger: mix track only when really ready (2)"" into ics-mr1

* commit 'fee5a860a8355cda071ff23644e943414ba7f65d':
DO NOT MERGE Revert "AudioFlinger: mix track only when really ready (2)"
7baf7894bc2f1a62440f381eeb50143f210a5d61 23-Jan-2012 Justin Ho <justinho@google.com> DO NOT MERGE Revert "AudioFlinger: mix track only when really ready (2)"

This reverts commit 71c4496a9757438afd30b4404824f296f6158a49.

Change-Id: Iff10c49ea728bb10023ddeb50a3b708db770fff2
335787fe43596f38ea2fa50b24c54d0823a3fb1d 21-Jan-2012 Glenn Kasten <gkasten@google.com> Remove AudioFlinger dependencies on client

Change-Id: Ibb591e41a3ca5d7015e2b66b98b8fef5f415fb37
83d86538c4c479a9225c75ab27938e8f05abb9c8 17-Jan-2012 Glenn Kasten <gkasten@google.com> Make AudioTrack control block volume field private

This is part of the process of abstracting the control block
to make it easier to maintain.

Change-Id: Idb8f461e68dab3bcf268159cc0781651c6fb7094
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
d967f0a099db2b71597a3127134afd4a46287a4a 20-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Remove redundant get()"
7fc9a6fdf146ded90b51c52f4a05d797294dcb85 10-Jan-2012 Glenn Kasten <gkasten@google.com> Remove redundant get()

get() is almost always unnecessary, except in a LOG.
Also no need to check for != 0 before calling get().

Change-Id: Ib06e7a503f86cf102f09acc1ffb2ad085025516d
d746737921074e2a6c39c52b06022c5166689df5 20-Jan-2012 Jean-Baptiste Queru <jbq@google.com> am 6df477be: Merge "Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF)"

* commit '6df477be186233e36fc370c4d2db6c1ed928a740':
Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF)
daef36f5d4934bd055c694a8d54b86e2b50a6159 20-Jan-2012 Jean-Baptiste Queru <jbq@google.com> am a826f9e2: Merge "Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF)"

* commit 'a826f9e2c4f6329d8d48c927f6e942e78ffaf92f':
Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF)
3734cbca25c6d902677cfb5e59dff7a1cb17a45d 20-Jan-2012 Jean-Baptiste Queru <jbq@google.com> am 4f367f33: Merge "Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF)"

* commit '4f367f3387887c538c81c34cc8becaea6fa5e430':
Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF)
e744a90eb52bf9547848c08380cfb7ba7e63ffbc 20-Jan-2012 Jean-Baptiste Queru <jbq@google.com> am ba7f0d2a: Merge "Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF)"

* commit 'ba7f0d2a03643ce429421b81febf18fd50473070':
Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF)
ca05a87f4c69670a06bdee4b4f98bcdcd838beda 20-Jan-2012 Jean-Baptiste Queru <jbq@google.com> Merge ee4618bc

Change-Id: Ie1dc6ad38e7c30636d80f6caef11cf6673144940
9a8ded7348c5b2302dd27b285b395416bc842c49 06-Jan-2012 Steve Block <steveblock@google.com> Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF)

Change-Id: I1de629b4632a4b3187ca1a28d6416daccd35f924
aa70226152d2084f85a96b52359dbc8476a86a45 06-Jan-2012 Steve Block <steveblock@google.com> Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF)

Change-Id: I8fbdfa7a7581f481968dbb65aa40f7042936d7cb
53feeb42c721e8fc9285e35e679906a951f3277c 04-Jan-2012 Steve Block <steveblock@google.com> Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF)

Change-Id: I26f76452ac49e2890b14d133c065493d8df0fb4a
52546c0ef96aa3e7e21482e0f9b6e982557c8da9 20-Dec-2011 Steve Block <steveblock@google.com> Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF)

Change-Id: I44f267700356967dc51e8f85ebf457dc85cfb229
3812256de32e73e38ba16e50ac0451c10223d4eb 20-Oct-2011 Steve Block <steveblock@google.com> Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF)

Change-Id: I5321ebd12e9c6248a108529e82c4e1af2a4405e3
aeeb7e219e34d2d657d829913659a4e10e976375 19-Jan-2012 Eric Laurent <elaurent@google.com> resolved conflicts for merge of 05683c85 to master

Change-Id: I7846b7da8c5813b7a9b1f3f71aede0229689ff0d
f1d4592d4c3c99ebab55559e164ff102e825283e 14-Jan-2012 Glenn Kasten <gkasten@google.com> For booleans, use ! instead of == false

Change-Id: Ibc115936d2d0b0b7744ebe9b52839ea5b42c4edd
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
b1cf75c4935001f61057989ee3cf27bbf09ecd9c 17-Jan-2012 Glenn Kasten <gkasten@google.com> Track volume cleanup

Always read and write track volumes atomically. In most places this was
already being done, but there were a couple places where the left and
right channels were read independently.

Changed constant MAX_GAIN_INT to be a uint32_t instead of a float.
It is always used as a uint32_t in comparisons and assignments.
Use MAX_GAIN_INT in more places.

Now that volume is always accessed atomically, removed the union
and alias for uint16_t volume[2], and kept only volumeLR.

Removed volatile as it's meaningless.

In AudioFlinger, clamp the track volumes read from shared memory
before applying master and stream volume.

Change-Id: If65e2b27e5bc3db5bf75540479843041b58433f0
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"
0107954f72153db747a3727dc1157e9236dfed90 17-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Use audio_stream_type_t consistently"
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
05632a5fa4b88ca474294887fc92a9fcdf0e2352 03-Jan-2012 Glenn Kasten <gkasten@google.com> AudioTrack and AudioFlinger send level cleanup

Add an API to control block for getting/setting send level.
This allow us to make the mSendLevel field private.

Document the lack of barriers.

Use 0.0f to initialize floating-point values (for doc only).

Change-Id: I59f83b00adeb89eeee227e7648625d9a835be7a4
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
e3a067f8bc98134941ee1a4da8c2a92a15aaa9cc 11-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Simplify range check for audio_mode_t"
c40256146bee58bff09e1c16ef99ea06d31f89f9 11-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Use correct type for hardware call state"
2ea3410d0d3d592ce30c3ba0ce3e0e63b1244057 11-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Use consistent style of & reference for AutoMutex"
9770988e61961d34033fd2c12f0de85a267df68f 11-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "AudioFlinger new can't fail"
930f4caa1e311ef7ff538c421a324396157eb24f 07-Jan-2012 Glenn Kasten <gkasten@google.com> Simplify range check for audio_mode_t

AudioSystem::setMode previously allowed negative modes, but these were
then rejected by AudioFlinger.

Now negative modes (including AUDIO_MODE_INVALID and AUDIO_MODE_CURRENT)
are explicitly disallowed.

Change-Id: I0bac8fea737c8eb1f5b6afbb893e48739f88d745
c1dc1cb1d1eaf84e88669f1a5f22579a0d9237c2 09-Jan-2012 Steve Block <steveblock@google.com> Rename LOG_ASSERT to ALOG_ASSERT DO NOT MERGE

See https://android-git.corp.google.com/g/157519

Bug: 5449033
Change-Id: I8ceb2dba1b031a0fd68d15d146960d9ced62bbf3
25b248eb52a0a16adaef6b79c9d92cb88b9a2bc2 04-Jan-2012 Glenn Kasten <gkasten@google.com> Use cached reference to media.player service

This save unnecessary binder calls

Change-Id: I93a60efc54d9c8fb8fab706cd4477bbfd00ffec8
febdbfec3b1ed0e20aa4f10bfdd82702d3e41f4b 09-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "suspended() and isSuspended() are const"
29357bc2c0dd7c43ad3bd0c8e3efa4e6fd9bfd47 06-Jan-2012 Steve Block <steveblock@google.com> Rename (IF_)LOGE(_IF) to (IF_)ALOGE(_IF) DO NOT MERGE

See https://android-git.corp.google.com/g/#/c/157220

Bug: 5449033
Change-Id: Ic9c19d30693bd56755f55906127cd6bd7126096c
4a6f02833d8421b7d9b20f89729d8bb03b8b8102 07-Jan-2012 Glenn Kasten <gkasten@google.com> AudioFlinger new can't fail

Change-Id: I7dae05a5ea1c962a9975386eab1fedbbe106ffba
a7d8d6fc5e102a08b262a1b78aa1abeeb097d9e4 06-Jan-2012 Glenn Kasten <gkasten@google.com> Use consistent style of & reference for AutoMutex

AutoMutex, which is a typedef for Mutex::Autolock, is overloaded for
either a reference (&) or pointer (*) parameter, but we prefer to use
the reference form when the mutex is known at compile time.

Change-Id: I3515e6d6ab7959b2356a27fa3b04fd49e42cb31e
a4454b4765c5905f14186893b0688be375642283 04-Jan-2012 Glenn Kasten <gkasten@google.com> Use correct type for hardware call state

Change-Id: Ic6d98b129e3ec653df1d8f7e829adf8dccb4f378
5ff1dd576bb93c45b44088a51544a18fc43ebf58 06-Jan-2012 Steve Block <steveblock@google.com> Rename (IF_)LOGW(_IF) to (IF_)ALOGW(_IF) DO NOT MERGE

See https://android-git.corp.google.com/g/157065

Bug: 5449033
Change-Id: I00a4b904f9449e6f93b7fd35eac28640d7929e69
8a08dcc0a5de19a904e77d5f31bed3dff9a59890 05-Jan-2012 Steve Block <steveblock@google.com> Merge "Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGE"
a3a854868a80fd9b9b8720e06a172754943f9417 04-Jan-2012 Glenn Kasten <gkasten@google.com> suspended() and isSuspended() are const

Change-Id: I04b95970b5a645b64e7e64fffd46d868354dda66
88592eccaf6afcddd5f985955be92fe25205c680 05-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Remove the notion of "active track" from mixer"
a2a0a5d7d56baa831870f4bf2a0d942a477d92ef 05-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Use the standard CC_LIKELY and CC_UNLIKELY macros"
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
7ab41c9f773ba599646f1b0d00955c1be80f92fd 05-Jan-2012 Eric Laurent <elaurent@google.com> resolved conflicts for merge of 1a4b9939 to master

Change-Id: I0c910d391a38a916d8431f7d1f5b82e39e1a66c2
df64d15042bbd5e0e4933ac49bf3c177dd94752c 04-Jan-2012 Steve Block <steveblock@google.com> Rename (IF_)LOGI(_IF) to (IF_)ALOGI(_IF) DO NOT MERGE

See https://android-git.corp.google.com/g/156801

Bug: 5449033
Change-Id: Ib08fe86d23db91ee153e9f91a99a35c42b9208ea
b8a805261bf0282e992d3608035e47d05a898710 20-Dec-2011 Steve Block <steveblock@google.com> Rename (IF_)LOGD(_IF) to (IF_)ALOGD(_IF) DO NOT MERGE

See https://android-git.corp.google.com/g/156016

Bug: 5449033
Change-Id: I4c4e33bb9df3e39e11cd985e193e6fbab4635298
071ccd5a9702500f3f7d62ef881300914926184d 23-Dec-2011 Eric Laurent <elaurent@google.com> audioflinger: fix clicks on 48kHz audio.

The calculation done in prepareTracks_l() for the minimum amount
off frames needed to mix one output buffer had 2 issues:
- the additional sample needed for interpolation was not included
- the fact that the resampler does not acknowledge the frames consumed
immediately after each mixing round but only once all frames requested have been used
was not taken into account.
Thus the number of frames available in track buffer could be considered sufficient although
it was not and the resampler would abort producing a short silence perceived as a click.

Issue 5727099.

Change-Id: I7419847a7474c7d9f9170bedd0a636132262142c
9c56d4ae6212c21ce5fd71ed534eb195983a07c1 20-Dec-2011 Glenn Kasten <gkasten@google.com> Remove the notion of "active track" from mixer

This is a first step towards making the mixer more object-oriented.

Change-Id: Ifd445d0e471023a7f5c82e934736ffc95ba1b05b
bbaf8673f1d1dd79d1b7f474ca7111da58e84aff 20-Dec-2011 Eric Laurent <elaurent@google.com> Merge "audio effects: rename configure command"
3d5188bd6abe55898f10a0edf3c05aff8aa2ef67 17-Dec-2011 Eric Laurent <elaurent@google.com> audio effects: rename configure command

Renamed audio effect library interface command for audio format
This makes the naming more consistent with other exixsting commands
and allow adding a new command to get the configuration (EFFECT_CMD_GET_CONFIG).
Same change for reverse channel configuration renamed from

Implemented EFFECT_CMD_GET_CONFIG in exisitng effect libraries.

Change-Id: Ia7b1c620f13797fe5aceb3b0b4acbacce09fb067
d1e672acd8fa1af899f85ee2321327237028adf8 17-Dec-2011 Glenn Kasten <gkasten@google.com> Merge "Extract out audio DSP code to utility library"
3b21c50ef95fe4e7ac3426ca14b365749e66ff08 15-Dec-2011 Glenn Kasten <gkasten@google.com> Extract out audio DSP code to utility library

Change-Id: Ib8ce72028a7ea30e82baa518e381370e820ebbd0
b87c068727a15a3d3f0bfdcb758c76a097f5e869 16-Dec-2011 Glenn Kasten <gkasten@google.com> Merge "Simplify enable/disable mixing"
079123ee3d2e20bbc17a7ddbd96ca46bed27898f 16-Dec-2011 Glenn Kasten <gkasten@google.com> Merge "Improve resistance to leaks for ConfigEvent"
26fa039c3752eaaf74d1be53d9795f48e9f43de3 16-Dec-2011 Glenn Kasten <gkasten@google.com> Merge "Use NULL not 0 for pointers"
1c48c3c61970527b97892ab6a2daae8eaac26964 15-Dec-2011 Glenn Kasten <gkasten@google.com> Simplify enable/disable mixing

The MIXING enum isn't needed, and now returns void instead of status_t.

Change-Id: Ibe4ec24081d75ad4ab78b9c7191fc9077959c4e9
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
42968939dfce0954d6540011199045ec4ed7de80 15-Dec-2011 Glenn Kasten <gkasten@google.com> Merge "Fix indentation and whitespace"
a06a9a50b37d60e9c43c9de9f8ea3a8649cd5691 15-Dec-2011 Glenn Kasten <gkasten@google.com> Merge "Improve AudioFlinger error logging"
d879601ace079e3c0aed79cf3fa5fb4db6ad4a9f 28-Oct-2011 Glenn Kasten <gkasten@google.com> Improve AudioFlinger error logging

Change-Id: I8ce9aff4038cd7fa0067600faa8080b137db1939
2eda60a8485cfe70a60e72156beffdc470ecb093 15-Dec-2011 Glenn Kasten <gkasten@google.com> Merge "Audio C++ comments"
c5ac4cb3a5124860ccfc7e4ff66251c55a5595ca 12-Dec-2011 Glenn Kasten <gkasten@google.com> Fix indentation and whitespace

Use git diff -w to verify.

Change-Id: Ib65be0a1ecf65d6cad516110604e3855bf68a638
c23bd9b5b9e4be9c395789810fdd8522296fc50c 15-Dec-2011 Glenn Kasten <gkasten@google.com> Merge "Use const char correctly in AudioFlinger"
e5dfcd8c6792c4b64120fd03708729b70a887f2a 15-Dec-2011 Glenn Kasten <gkasten@google.com> Merge "Use units after all times"
362c4e697d8e9c034e964ac7b40227e054491547 14-Dec-2011 Glenn Kasten <gkasten@google.com> Audio C++ comments

Change-Id: I84906ebb9dfcfa5b96b287d18364b407f02a30c1
91eb8bfbe253a6b6fe1aa23fb884a601c28991c4 13-Dec-2011 Glenn Kasten <gkasten@google.com> Remove redundant clear()

Change-Id: Ie5e4e63cbc8fa85ef50451dddf8f149fa864b132
ec1d6b5e17281a066d618f7fcd2b63b3ce11f421 12-Dec-2011 Glenn Kasten <gkasten@google.com> Use const char correctly in AudioFlinger

Use const char [] instead of const char * to eliminate unnecessary pointer.
Make the array audio_interfaces also const, in addition to the strings.

Change-Id: I31f33d1dcb9a657ee136f4280fd2d46492496831
7dede876998ff56351d495ec3a798c1b131193e8 13-Dec-2011 Glenn Kasten <gkasten@google.com> Use units after all times

Change-Id: I48d3f29c37228b5d03189e4c9600824c9360cac9
2013d4d159bfc29b4143d3b5fd4735f51a03684c 06-Dec-2011 Eric Laurent <elaurent@google.com> am 5433e25f: am 7b6aff23: Merge "audioflinger: fix audio skipping over A2DP" into ics-mr1

* commit '5433e25f6ce013860ff2a074ad8d1158cc39ab91':
audioflinger: fix audio skipping over A2DP
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
926798f8c21ab002d9797ef8973852a2612c1f75 23-Nov-2011 Eric Laurent <elaurent@google.com> am f6422f5f: am 20398fac: Merge "audioflinger: reduce sleep time to avoid underrun" into ics-mr1

* commit 'f6422f5f5d04aab47f8f36a0ea92e2140bed0105':
audioflinger: reduce sleep time to avoid underrun
7cafbb32999049873d4746ba83bd20c88abe6ce6 23-Nov-2011 Eric Laurent <elaurent@google.com> audioflinger: reduce sleep time to avoid underrun

Progressively reduce the sleep time applied in MixerThread::threadLoop()
in case of consecutive application underruns to avoid starving the audio HAL.
As the default sleep time is longer than the duration of an audio buffer
we ended up writing less data than needed by the audio HAL if
the condition persisted.

Issue 5553055.

Change-Id: I2b23ee79c032efa945025db228beaecd1e07a2e5
c4795ecad4e5a0b3ec54862a40c82ef1ba53cd59 14-Nov-2011 Eric Laurent <elaurent@google.com> am 25924f8f: am 030bb998: Merge "audioflinger: fix noise when skipping to next song" into ics-mr1

* commit '25924f8f6c0a4ca4a2eb257b72d9625f69d2525e':
audioflinger: fix noise when skipping to next song
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
6977ca7d5ffdbc1610a95c74653b1fbe6a665f32 10-Nov-2011 Eric Laurent <elaurent@google.com> am db7d79e6: am 2b7f91b9: Merge "Fix regression for SoundPool playback" into ics-mr1

* commit 'db7d79e6f1e1860a9bfe4756a03c753435fd0ddf':
Fix regression for SoundPool playback
a47b69c6f7c6fe0044ebcb2d0790ce3548de56fd 09-Nov-2011 Eric Laurent <elaurent@google.com> Fix regression for SoundPool playback

Commit 19ddf0eb introduced a problem with applications (like SoundPool)
relying on an underrun condition to detect end of playback instead of
stopping the track when all data is written.
AudioFlinger would keep waiting for new data in case of partial buffer
filling and never reach the underrun condition.

Added a mechanism to wait no more than once if not enough frames are present
in the track buffer.

Issue 5585490.

Change-Id: I131e605ff6070831a01ddf734e68459e3bf2354b
3b86c964df855a9740c446e984309b719c3ec37c 08-Nov-2011 Eric Laurent <elaurent@google.com> am f3a892ab: Merge "AudioFlinger: mix track only when really ready" into ics-mr1

* commit 'f3a892ab9347ce733b81ccb4913a91c586f8f367':
AudioFlinger: mix track only when really ready
3dbe3201479828e84abe02e1fdd0a5d414c0ddb8 03-Nov-2011 Eric Laurent <elaurent@google.com> AudioFlinger: mix track only when really ready

The addition of low power audio playback mode made that audio buffer consumption
by audio HAL can now happen in bursts. This makes that requesting audio data
from an AudioTrack for mixing can happen at much shorter intervals than before.
This revealed an existing problem where AudioFlinger would consider a track ready
for mixing although not enough frames were ready to completely fill one output buffer,
thus creating short periods of silence.

The fix consists in waiting for enough frames to be ready in AudioTrack buffer before
declaring a track ready for mixing. This minimum is not applied when the track is stopped
to allow the buffer to be emptied completely.

Change-Id: I6d04f9b65db5af85b0b53f0a5674be7ec02f9e9f
3856b090cd04ba5dd4a59a12430ed724d5995909 20-Oct-2011 Steve Block <steveblock@google.com> Rename (IF_)LOGV(_IF) to (IF_)ALOGV(_IF) DO NOT MERGE

See https://android-git.corp.google.com/g/#/c/143865

Bug: 5449033
Change-Id: I0122812ed6ff6f5b59fe4a43ab8bff0577adde0a
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
5c4e818c39ac2d2739675fe907904a874f7623c5 19-Oct-2011 Eric Laurent <elaurent@google.com> Limit AudioFlinger log.

AudioFlinger logs a warning when a write to the audio HAL
takes too long to return. The threshold for this warning is
a rule of thumb based on the assumption that the audio HAL will consume
buffers at a regular pace.
The introduction of low power audio mode with larger buffers and writes
occuring in bursts makes that this threshold is often exceeded resulting
in excessive and misleading warnings.

The threshold is raised to remove unwanted warnings but we should reconsider
the usefulness of this warning altogether.

Change-Id: I5ef6898ea28d879cede3e47da542a64092a3cca4
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
9d18ec574f5d847a86a21594ac39394ef5b108fc 27-Sep-2011 Eric Laurent <elaurent@google.com> Fix issue 5373658: memory leak in AudioFlinger.

Unlink PowerManagerService binder interface death recipient
in ThreadBase destructor.

Change-Id: Iab06ae9a8a6737bb002b6416a157b0fb50c11ad5
6bffdb8b598a2399e57f6ca48660fb7bdb2490b5 23-Sep-2011 Eric Laurent <elaurent@google.com> Fix issue 5355047: Automated effect tests fail.

Fixed several regressions in automated audio effect tests due
to changes in effect framework and visualizer FFT output range.

- Do not suspend Volume effect on session 0 when effects are
enabled on specific sessions.
- Adapt energy detection thresholds to new visualizer FFT range.
- Leave more time for BassBoost and Virtualizer effects to ramp up
before measuring the effect.
- Removed second insert reverb left by mistake on the player session
in preset reverb test.

Change-Id: I7a1ad1372d783fa7900eb9dd1d3b47f54d8d766f
60cd0a0d488e604d27fc7dbb02b00348693dfde2 13-Sep-2011 Eric Laurent <elaurent@google.com> Issue 4345021: Audio routed to multiple devices...

There is a possiblility that the condition on which RecordThread::checkForNewParameters_l()
waits after updating the command completion status is never signalled.
This happens if the thread executing ThreadBase::setParameters() has timed out waiting
for the status (for instance if the audio HAL takes too long to execute the setParameters()
command. Then the RecordThread is stuck forever.

The fix consists in waiting for the condition with a timeout in RecordThread::checkForNewParameters_l().

Change-Id: I7fc671bc2fc43ba4acb65a2beb33ee05742f091e
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
a1884f9e9ec3836683efd7eb333ee442e8bc9d56 23-Aug-2011 Eric Laurent <elaurent@google.com> AudioFlinger: add check for audio HAL init failure

Do not call audio HAL functions on the primary HW interface
if it could not be initialized properly.

Change-Id: If54059c8fd188d6c1686f9e0439994fe9411478a
0270b188aa3929cc512ec6869caba1d6b60cc08c 12-Aug-2011 Marco Nelissen <marcone@google.com> Don't remove effects on session 0

Change-Id: Id6f29fb1c687069f7480dd81d4745a558f202226
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
6dbe883644940badc684957cfc381bfd115f205e 28-Jul-2011 Eric Laurent <elaurent@google.com> Fix issue 5090721: audio record broken

Commit 6dbdc40 introduced a deadlock when exiting the
AudioFlinger RecordThread.

Change-Id: I1f63e54c5aeff05da4e4d028b53f734c62c78677
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
edc15ad8fcde12dc4f642d80d077239b1532eeca 22-Jul-2011 Eric Laurent <elaurent@google.com> Fix issue 4988574: 8 bit PCM audio playback broken.

Fixed regression in audio track control block frame size
calculation introduced by commit c310dcb.

Change-Id: Ia731b946ae4e43316b98d80229e3b08a696e47d6
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
e0b5bb23f0a26d248275d203885b820659da7320 16-Jul-2011 Glenn Kasten <gkasten@google.com> Merge "Log CPU usage"
4d8d0c30abfa4b8d75866d42094cc797e05068fa 09-Jul-2011 Glenn Kasten <gkasten@google.com> Log CPU usage

Change-Id: Ie447e59be139153e526b7ad467c46c659d26816f
6d8b694d999e9be7d5dcc336535832a80fb6f61f 24-Jun-2011 Eric Laurent <elaurent@google.com> Moved and renamed effect API header files

Moved specific effect header files to
and renamed to lower case (effect_xxx.h).

Change-Id: Icfc2264bfd013cab0395d7e310ada636b9fe3621
671a636931295d9c33ffca74551a804479d01241 17-Jun-2011 Eric Laurent <elaurent@google.com> Added audio_bytes_per_sample() helper function

Change-Id: Ibfcd75c4c241a53d5f052c25ada091904991048a
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>
0512ab559d4670c2204078470d7ef5d376811c57 05-May-2011 Glenn Kasten <gkasten@google.com> Remove dead code related to gettid

The gettid system call is always available now.

Change-Id: Ib78b41781eda182dc8605daf456bbea7ff7c2dc0
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
162f7d15ac5c8c23d1c3de171239f3a4e6e06b2a 23-May-2011 Glenn Kasten <gkasten@google.com> Merge "HAVE_ANDROID_OS AUDIOFLINGER_SECURITY_ENABLED dead"
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>
249c6a61f21bc90e25e4b77f18c98af1ac363e69 05-May-2011 Glenn Kasten <gkasten@google.com> HAVE_ANDROID_OS AUDIOFLINGER_SECURITY_ENABLED dead

Remove dead code from the days when Android ran in emulator.

Change-Id: Ibadbbde0538239ad9b2811a3a2e8f8a6d3b6389c
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>
c30268b9d118309a0514bcf280a03ee69f81403f 06-Apr-2011 Glenn Kasten <gkasten@google.com> Merge "Miscellaneous code cleanup in audio framework"
4bcae82f9b07d1a39956c45a6f5bec0b696c4dd1 04-Apr-2011 Glenn Kasten <gkasten@google.com> Miscellaneous code cleanup in audio framework

- Move declaration of kClassPathName to top of file so it can be used
in more than one place, instead of "android/media/AudioSystem".
- Make private methods static.
- Add comment to stream_type, audio_mode, force_use types that they must match
values in AudioSystem.java.
- Add comment about unused types mp3_sub_format and vorbis_sub_format.
- Fix typos.
- Use @ in javadoc comments.
- Delete dead APIs setMode, getMode, setRouting, getRouting in AudioSystem.java
(they are all hidden, deprecated, and unused by rest of framework)
- Delete unused private log method.
- Fix pathname for android_media_AudioSystem.cpp.
- Improve code formatting for space after == and !=.
- Add logging of delta for changing audio policy manager ref count.

Change-Id: I18037c7beb8ab76d1fda08c11e589f6e591d36e1
38ccae2c0324daa305f3fe77d25fdf5edec0b0e1 29-Mar-2011 Eric Laurent <elaurent@google.com> New fix for issue 4111672: control block flags

The first fix (commit 913af0b4) is problematic because it makes threads
in mediaserver process block on the cblk mutex. This is not permitted
as it can cause audio to skip or worse have a malicious application
prevent all audio playback by keeping the mutex locked.

The fix consists in using atomic operations when modifying the control
block flags.

Also fixed audio_track_cblk_t::framesReady() so that it doesn't block
when called from AudioFlinger (only applies when a loop is active).

Change-Id: Ibf0abb562ced3e9f64118afdd5036854bb959428
33797ea64d067dfeaacbfd7ebe7f3383b73961b5 17-Mar-2011 Eric Laurent <elaurent@google.com> Fix issue 4111672: AudioTrack control block flags

Make sure that all read/modify/write operations on the AudioTrack
and AudioRecord control block flags field are protected by the
control block's mutex.

Also fix potential infinite loop in AudioTrack::write() if the
written size is not a multiple of frame size.

Change-Id: Ib3d557eb45dcc3abeb32c9aa56058e2873afee27
9ee159b79022b2e1a050acb3890ce948e99e9ccb 24-Feb-2011 Gloria Wang <gwang@google.com> - To track the usage of all audio output devices
- To track the currently used audio device
- The devices are separated as speaker and other audio devices
- Provide the collected data to battery application through pullBatteryData()

Change-Id: I374c755266b5ac6b1c6c630400f4daf901ea8acc
243f5f91755c01614a8cafe90b0806396e22d553 01-Mar-2011 Eric Laurent <elaurent@google.com> Fix issue 3479042.

The problem is that when an AudioRecord using the resampler is restarted,
the resampler state is not reset (as there is no reset function in the resampler).
The consequence is that the first time the record thread loop runs, it calls the resampler
which consumes the remaining data in the input buffer and when this buffer is released
the input index is incremented over the limit.

The fix consists in implementing a reset function in the resampler.

A similar problem was also present for playback but unoticed because the track buffer is always
drained by the mixer when a track stops. The only problem for playback was that the initial
phase fraction was wrong when restarting a track after stop (it was correct after a pause).

Change-Id: Ifc2585d685f4402d29f4afc63f6efd1d69265de3
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
73337489229cc9b50371c7a9fcd86e9f00ea46d0 20-Jan-2011 Eric Laurent <elaurent@google.com> Fix issue 3371096.

EffectModule::process() was copying effect chain input buffer to
output buffer if no effect was active instead of accumulating it.

Change-Id: I2838af2e7b6654d0a76547625929a5453da68d02
935752053ef2691dbb6d5a6d149e0e362c6e3c74 19-Jan-2011 Eric Laurent <elaurent@google.com> Tentative fix for issue 3362362.

The problem is likely that one method is called on the AudioPolicyManagerBase
instance while it is still being constructed by AudioPolicyService.

To avoid this, the AudioPolicyService mutex is held by the constructor until the
platform specific AudioPolicyManager is constructed and the member
mpPolicyManager initialized.

Also added an initCheck() method to AudioPolicyInterface to verify successful
initialization of AudioPolicyManager.

A similar change is done in AudioFlinger constructor.
Also added some missing protections in AudioFlinger methods where the
playback thread list is parsed.

Change-Id: I006b244ec057e1bb0aa5ebe426ef006e3b171056
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
f1fb01a7f00b8da90a36268aba8584a872e99175 15-Nov-2010 Jean-Michel Trivi <jmtrivi@google.com> Add new audio mode for audio communications other than telelphony.

The audio mode MODE_IN_CALL signals the system the device a phone
call is currently underway. There was no way for audio video
chat or VoIP applications to signal a call is underway, but not
using the telephony resources. This change introduces a new mode
to address this. Changes in other parts of the system (java
and native) are required to take this new mode into account.
The generic AudioPolicyManager is updated to not use its phone
state variable directly, but to use two new convenience methods,
isInCall() and isStateInCall(int) instead.

Change-Id: Id744cd26520ea1d1a4795eabe6a1f0c58789af76
af59ce2407fa4e6e5d8f1664a4df2daf1badd407 05-Oct-2010 Eric Laurent <elaurent@google.com> Fixed AudioFlinger not always pausing tracks

If the pause request is received before the AudioTrack buffer was
completelly filled and the track ready for mixing, the pause is
not executed: the track just underruns and stays in pausing state.

The fix consists in considering the track ready for mixing immediately
if pausing.

Change-Id: Ia6cb4703fee2126e41011a6400ea8eeb3a3e5456
44d9848d6656777a18019223e0d35f2fcc67719a 01-Oct-2010 Eric Laurent <elaurent@google.com> Issue 3032913: improve AudioTrack recovery time

This issue showed that when an AudioTrack underruns during a too long period
of time and is therefore disabled by audioflinger mixer, it takes an additional
delay of up to 3 seconds to recover.
This fix adds a simple mechanism to recover immediately when the client application
is ready to write data again in the AudioTrack buffer

Also throttle warnings on record overflows

Change-Id: I8b2c71578dd134b9e60a15ee4d91b70f3799cb3d
dac69110ed1073bf0a9827a3f78698896dd05d97 28-Sep-2010 Eric Laurent <elaurent@google.com> Fix several audio effects problems.

Fixed the following issues in LVM effect bundle wrapper:
- memory leaks in EffectCreate() in case effect creation fails at various stages
- Added saturation when accumulating to output buffer
- Fixed problems with enabled effects count when an effect is released while enabled
- Do not allocate temporary buffer for accumulation each time process() is called

Fixed the following issues in effects framework (AudioFlinger)
- Release effect synchronously in the library when deleted from effect chain
- Do not call the effect process function if no tracks are present in the same
audio session

Change-Id: Ifbd80a163415cfb3c0a337c12082853ea45d9c91
84e9a10fde8a4ae3da4f88d7911c154933aa457f 24-Sep-2010 Eric Laurent <elaurent@google.com> Fix issue 3007862

Removed a cross deadlock condition between audioflinger and audio policy
service mutexes.
Audioflinger::createEffect() locks audioflinger mutex and then calls
AudioSystem::getOutputForEffect() which ends up in
AudioPolicyService::getOutputForEffect() which locks audio policy service
mutex. If at the same time, the command thread in audio policy service is
processing a command(set volume, set route...), the mutex is locked and the
command will call one audioflinger method which in turn will attempt to
lock audioflinger mutex.
The fix consists in releasing audioflinger mutex before calling

Change-Id: Id44e7feb36e0a295731f6aa97cf32d022edd34d0
fac4895de4ae63928ff0cf2ccece106eb6d33f72 22-Sep-2010 Eric Laurent <elaurent@google.com> Request permission for global audio effects.

Applications creating an audio effect on the output mix must
have the MODIFY_AUDIO_SETTINGS permission.

Change-Id: I57d88533f91ad0d33680107d79abcec28f7263b5
e0aed6ddcb4e3c301b80aa26706b6052dab42c41 11-Sep-2010 Eric Laurent <elaurent@google.com> Fix volume problems with insert revert

- Use a constant input level to the reverb engine and implement volume control in the
insert reverb. This avoids the volume spikes when an effect that was inserted after
the reverb is disabled or removed.
- Fix clicks (one silent buffer) at the end of the reverb disable period.
- Modified volume management in audioflinger so that the volume ramp is also done by
the insert effect if present when the track is paused (avoids clicks).
- Increased room level for all presets.

Also fixed problems with output stage session (-1):
- effect bundle wrapper was not designed to support session -1
- the permission check in audioflinger for using session -1 failed due to a wrong usage of

Change-Id: Id1ff51327263364bf71d3f2668fa5cde4311d84f
aeae3de947fa0b1e670c8472b32288962f97b4f5 02-Sep-2010 Eric Laurent <elaurent@google.com> Fix problem in AudioEffect::command() status.

The *pReplyData argument of the command() function was left unitialized by EffectHandle::command()

Change-Id: I91a19817ead2a8cfbdd8e2d77ca270c7ce9d5bd4
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
571d49c1c316f5e07b74ed7b5df6bdec7cbc1a14 11-Aug-2010 Eric Laurent <elaurent@google.com> Fix issue 2909189: System property ro.audio.silent no longer mutes system.

Fixed regression introduced by commit 2a6b80bc65c4782b5a7168b300e1dc5ec9f617ee:
master mute was not working if no effect chains were present on session 0.

Change-Id: I66d107e045d159cb94d29c7476fa1e12d92f2ae7
8569f0d3bf4c6787707e348a7cf73b9c4199cb32 30-Jul-2010 Eric Laurent <elaurent@google.com> Fixed several audio effects problems.

- Fixed constant inversions in AudioEffect.java
- Do not return error when enabling an already enabled effect
- Update cached effect state in native AudioEffect class when effect is enabled/disabled by command() method
- Remove click when restarting effect during disable sequence
- Fixed problem in master mute management when volume control is delegated to effect.

Change-Id: I6df4ce9fcc54fdc7345df858f639d20d802d6712
60e182437228312cc28469a5b0dfde77ac848e1a 29-Jul-2010 Eric Laurent <elaurent@google.com> Fixed underrun in audioflinger mixer.

When all audio tracks have been disabled and the mixer is running idle before the output stream is placed in standby,
the mixer sometimes fails to write to the output stream on time to avoid underrun.

This is because the sleep period used to wait before the next write to output stream is too close to the actual buffer duration.
In fact this sleep time is not critical as if we write too early to the output stream, the kernel driver will wait for free buffers
from the audio DSP DMA and we will sleep anyways.

The fix consists in dividing the calculated wait period by 2 to increase the margin.

Change-Id: I5730887dc2ccce2a511bc858494a6f7da6b392a0
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