History log of /frameworks/av/media/libmedia/AudioTrack.cpp
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
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
73431968e3c5cf420b5d63c54cf72d10ff7ee7bc 13-Sep-2012 Jean-Michel Trivi <jmtrivi@google.com> Workaround for track recreation bug

Ignore thread can call Java parameter to work around bug where
AudioTrack instance gets recreated when key clicks are enabled
and WFD gets turned on, with the wrong parameters.

Change-Id: Ia42c8704b46fe3ffea560b05b60939fa2e4b29e1
c3ae93f21280859086ae371428ffd32f39e76d50 30-Jul-2012 Glenn Kasten <gkasten@google.com> Update audio comments

Change-Id: Ie7504d0ddb252f7e4d4f99ed0b44cfc7b1049816
a997e7a7c5aa2fc7e95316218316f8b7b98786ba 07-Aug-2012 Glenn Kasten <gkasten@google.com> Revert "Swap the order of creating IAudioTrack and thread"

This reverts commit 5d464eb0b8cffb994a754ff108795e858a882414.
It caused the wrong thread ID -1 to be passed to IAudioFlinger::createTrack().

Change-Id: Ic221d2bb4af572d3d2d752af19238c52f6728e3a
5d464eb0b8cffb994a754ff108795e858a882414 23-Jun-2012 Glenn Kasten <gkasten@google.com> Swap the order of creating IAudioTrack and thread

Simplifies the error recovery in case IAudioTrack fails.

Change-Id: I6aee41a2ac747a5689fb4836b04174e6107bf32f
28b76b334f92a15a2be3cc9e2f7d229a3275d1ac 04-Jul-2012 Glenn Kasten <gkasten@google.com> Use audio_channel_mask_t for channel mask

Change-Id: I1c1896da48983aa9f1462a4b471f910498816f60
04cd0186305e2b59d23c9147787046c6662029cc 25-Jun-2012 Glenn Kasten <gkasten@google.com> getMinFrameCount error handling

Convention is for "get" APIs that directly return status_t and indirectly
return a value via a pointer, to return BAD_VALUE if the pointer is NULL.
Also indirectly return 0 for other errors.

Change-Id: I1599f20ecb26e9723f9fb384ffbf911ff3a2ce1c
d4070955e28ae62aa4be1657f9d32acde104bb86 12-Jul-2012 Glenn Kasten <gkasten@google.com> Remove dead code in libmedia

Change-Id: I7d8201590cda29c9fa99662a4fdba222091febfe
d3a9ff4b725de612bf0354c035ba8f8564dbc6e8 21-Jun-2012 Glenn Kasten <gkasten@google.com> Move declarations of local variables to first use

Change-Id: I48b193a742b32b6746aa938b84dc405124a6a5c3
192cbbad773979a6fe3b5a0c223356de4fc3309c 13-Jun-2012 Eric Laurent <elaurent@google.com> Fix audio track pause.

AudioTrack::pause() should signal the control block condition
to release threads waiting for available buffers in obtainBuffer().
Otherwise the behavior relies on the timout on the condition
or the fact that audioflinger will mix a new audio buffer while executing
the pause.

Bug 6653769.

Change-Id: I5f8f73c471fe306070f30b814f32fd4b4dc1d575
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
df839841d2db4cb8e2acb10205b3942622b3e7a2 31-May-2012 Eric Laurent <elaurent@google.com> Do not keep audio wake lock when apps underrun.

Do not automatically restart an AudioTrack after an underrun
if the callback is executed but no data is written by the app.

Bug 6541286.

Change-Id: I11e7ab8dc968d7ff087058fec68f44490d3a7731
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
093000f7d11839b920e8dfaa42ed1d09f48e24b8 03-May-2012 Glenn Kasten <gkasten@google.com> Don't allow AudioTrack frameCount to decrease

This is a workaround for bug that client can cache return value of
frameCount(), and is not notified when this value changes due to automatic
re-recreation of the underlying IAudioTrack.

A better long-term fix would be to notify clients when these kinds of
parameters change, and to fix assumptions in client code that they are
constant (e.g. in SoundPool and maybe obtainBuffer).

Also, once a fast track request is denied, don't request it again.

Bug: 6431187
Change-Id: I55b4ff30bbd9ed3a402e39452a38de52cdea53a9
f4022f90db5acb680870db8c1150b673cdd211d9 02-May-2012 Glenn Kasten <gkasten@google.com> Fix race in AudioTrack destruction

Bug: 6427369
Change-Id: Id3b4487406235b881f6f0b4b95c5a02a9b797e75
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
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
a636433cbd09c0708b85f337ef45c0cdef3bcb4d 19-Apr-2012 Glenn Kasten <gkasten@google.com> Use C APIs instead of C++ APIs for policy

The C++ APIs are going away.

Note: we use tid == 0 which is not supported yet by the C APIs,
do not submit this until that is added.

Change-Id: I0e90789e6c81c69f2544e899c52421ea5d1342be
ca8b28013c0558a4a3323a1a0f58520277200086 23-Apr-2012 Glenn Kasten <gkasten@google.com> Fix regression in AudioTrack::pause()

Bug: 6379646
Change-Id: I12b53bc4118499ddc73a53a981f3f56328140868
3acbd053c842e76e1a40fc8a0bf62de87eebf00f 28-Feb-2012 Glenn Kasten <gkasten@google.com> Configure policy of mediaserver threads

Change-Id: Ifd825590ba36996064a458f64453a94b84722cb0
1948eb3ea6eee336e8cdab9b0c693f93f5f19993 14-Apr-2012 Eric Laurent <elaurent@google.com> Add support for deep audio buffers

Allow AudioSink to use deep audio buffering when the
source is audio only and its duration is more than
a certain threshold.
This helps improve battery life but implies higher
audio latency.

Change-Id: Ie79915b61c370292f05aabda9779356570e03cbb
0ca3cf94c0dfc173ad7886ae162c4b67067539f6 18-Apr-2012 Eric Laurent <elaurent@google.com> rename audio policy output flags

Change-Id: I27c46bd1d1b2b5f96b87af7d05b951fef18a1312
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
4a4a0959bca78e03e3c3f486ba17829c28314d8c 19-Mar-2012 Glenn Kasten <gkasten@google.com> AudioTrack client fast policy

Change-Id: I42ce691df3f586ac061b62237f35a263a0116f1f
a1472d9883e35edd280201c8be3191695007dfd4 30-Mar-2012 Marco Nelissen <marcone@google.com> Make AudioTrack/AudioRecord handle more than 2^32 frames

Change-Id: I471815012c6a113ec2c4dd7676e8fa288a70bc76
b83d38feeeb88a8a2a6219e1fca2480b5a14fb0d 26-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "IAudioFlinger::createTrack and openRecord flags"
291f4d505aff81969e6666049d3cc3446f836af2 19-Mar-2012 Glenn Kasten <gkasten@google.com> Remove enforceFrameCount

It was only used to decide whether to issue a warning.
The warning was issued the first time track was created but
not at re-creation. Now it is a verbose message every time,
not a warning since it happens all the time with key clicks on A2DP.

Change-Id: I9d39f53c0a7eb84b666e55b1b76ff830cf8f37ba
63c1faa8dea7feb90255d31ef2a133d8f2818844 21-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "Update comments"
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
a075db4ff9b086ac2885df77bb6da0869293df92 06-Mar-2012 Glenn Kasten <gkasten@google.com> IAudioFlinger::createTrack and openRecord flags

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

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

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

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

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

Use git diff -b or -w to verify.

Change-Id: I966b7ba852faa5474be6907fb212f5e267c2874e
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
d8d6185c978c9b27ca69e7791785e0983ed9e8b8 06-Mar-2012 Eric Laurent <elaurent@google.com> AudioTrack: relax check on minimum buffer size

Current AudioTrack implementation enforces that the requested audio
buffer size is at least corresponding the audio latency.
This requirement is too strong and leads to problems with current
stagefright and AudioSink implementations when playing over output
streams with long latency.

Ultimately, the AudioSink design should be changed to specify a minimum
buffer size in time or frames units but not in buffer count units.

Change-Id: I8ba603956f92ac49143a8249572665aa548f2f0f
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
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>
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
ed15977476a3d53103866e6d527fa3fb65d4166c 14-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Remove dead code AudioTrack::getLoop"
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
38f5d71e72f3b76c5b519614d27f051d53cd2712 08-Feb-2012 Glenn Kasten <gkasten@google.com> Remove dead code AudioTrack::getLoop

Change-Id: I868329c52f31bc20125f068500d8f892b4ec9796
4f9b0c54011eb8fd2ccfb393c2dcd51cd07800e0 08-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Remove dead mutex in AudioTrack/AudioRecord thread"
a5224f319e2ba4b51ddb4287705ccf8d4b8ecc51 04-Jan-2012 Glenn Kasten <gkasten@google.com> AudioTrack declare more methods const

Change-Id: I4999e984460893961d0d8092cff17f3cf07d7214
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
0e265cf36d201a7ccc0238b5c60b50f43d1dc450 02-Feb-2012 Glenn Kasten <gkasten@google.com> Remove dead mutex in AudioTrack/AudioRecord thread

The client callback threads had mutexes called AudioTrackThread::mLock
and ClientRecordThread::mLock. These mutexes were only used by start()
and stop(), and were unused by the thread itself. But start() and
stop() already have their own protection provided by AudioTrack::mLock
and AudioRecord::mLock. So the thread mutexes can be removed.

Change-Id: I098406d381645d77fba06a15511e179a327848ef
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
05bfe50e13793404a78c20c850d467d17734d496 20-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Move memcpy_to_i16_from_u8 to audio_utils"
c813985abd8ba61e999b3505f6a332574f87a1be 18-Jan-2012 Andreas Huber <andih@google.com> Temporarily restore AudioSystem/AudioTrack APIs with their former signatures

until we get updated prebuilts from vendor.

Change-Id: I8aae81d2513edca0ab268053a11c8c4206879e61
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
511754b5839fd9b09fc56b89ae007fbc39084a33 11-Jan-2012 Glenn Kasten <gkasten@google.com> Move memcpy_to_i16_from_u8 to audio_utils

This will make it easier for this kind of code to be optimized
for each target architecture.

Change-Id: I9efd27d6c0175b00b9a784353244805cec63c0b8
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"
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
4cc55d53d542c4f4ed645738cebb65b9e7eb6c44 12-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Use audio_format_t consistently"
83bc7f3cf78b28a818417f40a4f0c00593993366 04-Jan-2012 Glenn Kasten <gkasten@google.com> libmedia new can't fail on Android

Change-Id: Ie79dd5abb8078b35474bf0f1b3a6ff994a3a3360
5c7b3bcc88e2b472f1f7b416d89222714b96a567 09-Jan-2012 Glenn Kasten <gkasten@google.com> Merge "Clean up AudioTrack::mActive and stopped()"
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
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
e1c3962e268ffc12bfd1bd9ea84da1f135f36960 04-Jan-2012 Glenn Kasten <gkasten@google.com> Use audio_format_t consistently

Was int, uint32_t, uint16_t, and uint8_t with 2-bit bitfield.
Also replace 0 by AUDIO_FORMAT_DEFAULT and replace 1 by

Change-Id: Ia8804f53f1725669e368857d5bb2044917e17975
9a2aaf927e56a4b4acab23ef16b3f133a9f48a63 03-Jan-2012 Glenn Kasten <gkasten@google.com> Clean up AudioTrack::mActive and stopped()

mActive is protected by mLock; volatile is meaningless on SMP.

Fixed a couple of places where mActive was accessed without a lock:
- stopped()
- processAudioBuffer()

Added stopped_l() for cases where we already hold the lock.

Made mActive a bool not int.

Moved down a lock in setPosition that was being acquired too early.

Change-Id: I73ff368e991c0db9f9472df0b3f96fd33fcc7311
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
42968939dfce0954d6540011199045ec4ed7de80 15-Dec-2011 Glenn Kasten <gkasten@google.com> Merge "Fix indentation and whitespace"
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
1d334101f1289cf4c8967af6e78ac22619175982 15-Dec-2011 Glenn Kasten <gkasten@google.com> Merge "Remove unnecessary this->"
b299dc4ded29a226daac07f195d1558e660d2f9f 15-Dec-2011 Glenn Kasten <gkasten@google.com> Merge "Remove redundant clear()"
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
9054897ab9ffb307fadae81b774a1fc61cb542e7 13-Dec-2011 Glenn Kasten <gkasten@google.com> Remove unnecessary this->

Change-Id: I72038f5d4568f0633d3e4ab90f4b67e2dd22c332
f0c495012bad92230604a9a12a907812ec49ee8f 30-Nov-2011 Glenn Kasten <gkasten@google.com> AudioTrack::setVolume check range

Change-Id: Ie182bf0f741f1f49f68c02a1e7437a2a34d34fc5
879135196fd1c97deefc538c888037c56c2879a7 23-Jun-2011 Glenn Kasten <gkasten@google.com> Bug 4903178 Restore priority and cgroup on stop

On AudioTrack and AudioRecord stop or failed start, restore the priority
and cgroup of the caller to their previous values, rather than forcing
to NORMAL. Dependent on new thread APIs.

Also fixes bug where priority was set to AUDIO but cgroup not set.

Change-Id: Ib83893918fb4fdf57c6b87884b51038997a631d8
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
cfe2ba6b01a258e39f9c215ffc7b750e0b68f708 14-Sep-2011 Eric Laurent <elaurent@google.com> Issue 5298399: Lost speech after a crash in gTalk.

Fixed problem in AudioTrack::restoreTrack_l() causing a permanent
failure if the IAudioTrack interface to AudioFlinger could not be
restored at the first attempt.

Change-Id: I039d4fe2dca8d3baf71f1a6c51119f27a67b6611
2267ba18d0d2b2d4bd7f5411821ad89ac2659a88 07-Sep-2011 Eric Laurent <elaurent@google.com> AudioTrack: extend callback thread sleep time

Do not force wake up the AudioTrack thread every 10ms if no timed
events (loop, markers..) have to be processed.
This will help reduce power consumption.

Change-Id: Icb425b13800690008dd07c27ffac84739e3dbba3
408b8dc3c0a364c6f6b4991d15da9e6bcc2b8008 06-Sep-2011 Eric Laurent <elaurent@google.com> Issue 5247986: Battery drain due to audio wakelock

The problem occurs when activating or deactivating A2DP connection
while SoudPool has a channel active. This can happen quite frequently now
that the UI sound effects are enabled by default.
If PCM data is remaining in the AudioTrack buffer when it is restroyed and
re-created on the new AudioFlinger output thread, this data is flushed.
As a consequence, no underrun or request for new data callback is sent to
SoundPool and the sound channel remains active for ever as the end of the
sample is never detected.

Change-Id: I13e0c11e4ce3f83bff7f58d347ca814b6a86712b
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
cd07594333cbe8b2c86c6609cce01a74d6cc33f8 26-Aug-2011 Jean-Michel Trivi <jmtrivi@google.com> Bug 4364249 Play position is 0 after flushing AudioTrack

AudioTrack::stop() is not synchronous, so a stop() followed
by flush(), which is synchronous, will not always report
a playhead position of 0 after being called.
This CL adds a flag to mark a track as flushed, and report the
correct playhead position in this state.
Bug 5217011 has been created to address the real issue in the
future, where flush could be made synchronous, to properly
address bug 4364249.

Change-Id: Icf989d41a6bcd5985bb87764c287f3edb7e26d12
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
68cbeba4e21aa53f52fb99b74dfa1910af31a3eb 23-Jun-2011 Glenn Kasten <gkasten@google.com> Merge "Fix warnings for uninitialized local variables"
d0965dde97f2815ae0a15fe6b40946f8a741a81e 23-Jun-2011 Glenn Kasten <gkasten@google.com> Fix warnings for uninitialized local variables

Change-Id: Ic9b03b0fd215444e76c7b7bebb385f7831c557e0
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>
c6854100cea4fcd0f20cb2ac8235c02d1849b3a1 02-Jun-2011 Glenn Kasten <gkasten@google.com> Remove unnecessary level of priority indirection

Change-Id: I942d43973c20a7ace8b0d3f78b4da97e45e996c6
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
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>
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>
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
9b7d950f1f3b0c526712b713dbceb0e22762c015 21-Mar-2011 Eric Laurent <elaurent@google.com> Fix issue 3483718: audio streaming and A2DP.

The problem is that when switching from A2DP to device speakers or headset,
The AudioTrack binder interface to AudioFlinger must be destroyed and restored
to accomodate new buffer size requirements. Current AudioTrack implementation
did not restore properly the PCM buffer write index which caused a mismatch between
the written frame count in the mediaplayer renderer and the AudioTrack. The renderer
could then believe the AudioTrack buffer was full and stop writing data preventing the
AudioTrack to reach a bufffer full condition and resume playback.

The rendered was also modified to refresh the AudioTrack frame count (buffer size)
inside the write loop in NuPlayer::Renderer::onDrainAudioQueue() as this count can change
from one write to the next.

Also modified AudioTrack::obtainBuffer() to check for track invalidated status before
querying for available space in the buffer. This avoids writing to the old track's
buffer until full before detecting the invalidated condition and create a new track.

Change-Id: I16a857e464e466880847f52f640820aa271539ad
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
1703cdfee717b1b312bf8979816a9e2f16a82e5d 07-Mar-2011 Eric Laurent <elaurent@google.com> Fix issue 3439872: video chat and bluetooth SCO

This change fixes the stability problems experienced when using
a bluetooth headset supporting both A2DP and SCO. Problems occur
when starting the video chat at which time the A2DP output is being
stopped to start SCO. At that time, active AudioTracks are invalidated
by AudioFlinger so that a new AudioTrack binder interface can be
recreated by the client process on the new mixer thread with correct parameters.
The problem was that the process to restore the binder interface was not
protected against concurrent requests which caused 2 binder interfaces
to be created sometimes. This could lead to permanent client deadlock
if one of the client threads was waiting for a condition of the first
created binder interface while the second one was created (as the AudioFlinger
would only signal conditions on the last one created).
This concurrent request situation is more likely to happen when a client
uses the JAVA AudioTrack as the JNI implementation uses simultaneously the
native AudioTrack callback and write push mechanisms. By doing so, the code
that checks if the binder interface should be restored (in obtainBuffer()) is
much more likely to be called concurrently from two different threads.

The fix consists in protecting the critical binder interface restore phase
with a flag in the AudioTrack control block. The first thread acting upon the binder
interface restore request will raise the flag and the second thread will just wait for
a condition to be signaled when the restore process is complete.

Also protected all accesses to the AudioTrack control block by a mutex to prevent
access while the track is being destroyed and restored. If a mutex cannot be held
(e.g because we call a callback function), acquire a strong reference on the IAudioTrack
to prevent its destruction while the cblk is being accessed.

Modified AudioTrack JNI to use GetByteArrayElements() instead of
GetPrimitiveArrayCritical() when writing audio buffers. Entering a critical section would
cause the JNI to abort if a mediaserver crash occurs during a write due to the AudioSystem
callback being called during the critical section when media server process restarts.
Anyway with current JNI implementation, either versions do not copy data most of the times
and the criticial version does not guaranty no data copy.

The same modifications have been made to AudioRecord.

Change-Id: Idc5aa711a04c3eee180cdd03f44fe17f3c4dcb52
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
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
2beeb50b1bba9e92f6cacfeca37fe9fa9d36ead1 16-Jul-2010 Eric Laurent <elaurent@google.com> Added support for auxiliary audio effects to AudioTrack and MediaPlayer.

Added methods to AudioTrack and MediaPlayer java classes to enable use of
auxiliary audio effects. The effect can be attached and detached by specifying its
ID and the send level controlled.

Change-Id: Ie74ff54a453096a742688476f612ce355543b6f3
a514bdb58b5de4986679f72b7204b4764f7a2778 21-Jun-2010 Eric Laurent <elaurent@google.com> Added support for audio sessions in MediaPlayer and AudioTrack.

Audio sessions are used to associate audio effects to particular instances (or groups) of MediaPlayers or AudioTracks.

Change-Id: Ib94eec43241cfcb416590f435ddce7ab39a07640
33005a932c60a0780fe9b7307d5988df3d9f6c26 16-Jun-2010 Chia-chi Yeh <chiachi@android.com> media: add AudioTrack::getMinFrameCount().

Change-Id: If15162583d1d16d89f59be0793106afe77417f35
be916aa1267e2e6b1c148f51d11bcbbc79cb864c 02-Jun-2010 Eric Laurent <elaurent@google.com> Issue 2667801: [Audio Effect Framework] AudioFlinger, AudioMixer AudioTrack modifications.

First drop of audio framework modifications for audio effects support.

- AudioTrack/AudioRecord:
Added support for auxiliary effects in AudioTrack
Added support for audio sessions
Fixed left right channel inversion in setVolume()

- IAudioFlinger:
Added interface methods for effect enumeraiton and instantiation
Added support for audio sessions.

- IAudioTrack:
Added method to attach auxiliary effect.

- AudioFlinger
Created new classes to control effect engines in effect library and manage effect connections to tracks or
output mix:
EffectModule: wrapper object controlling the effect engine implementation in the effect library. There
is one EffectModule per instance of an effect in a given audio session
EffectChain: group of effects associated to one audio session. There is one EffectChain per audio session.
EffectChain for session 0 is for output mix effects, other chains are attached to audio tracks
with same session ID. Each chain contains a variable number of EffectModules
EffectHandle: implements the IEffect interface. There is one EffectHandle object for each application
controlling (or using) an effect module. THe EffectModule maintians a list of EffectHandles.

Added support for effect modules and effect chains creation in PlaybackThread.
modified mixer thread loop to allow track volume control by effect modules and call effect processing.

Each track now specifies its output buffer used by mixer for accumulation
Modified mixer process functions to process tracks by groups of tracks with same buffer
Modified track process functions to support accumulation to auxiliary channel

Change-Id: I26d5f7c9e070a89bdd383e1a659f8b7ca150379c
d1b449aad6c087a69f5ec66b7facb2845b73f1cb 14-May-2010 Eric Laurent <elaurent@google.com> Fix issue 2553359: Pandora does not work well with Passion deskdock / Cardock.

The problem is due to a too big difference between the buffer size used at the hardware interface and at the A2DP interface.
When no resampling occurs we don't notice problems but the timing is very tight. As soon as resampling is activated, the AudioTrack underruns.
This is because the AudioTrack buffers are not resized when moving the AudioTrack from hardware to A2DP output.
The AudioTrack buffers are calculated based on a hardware output buffer size of 3072 bytes. Which is much less than the A2DP output buffer size (10240).

The solution consists in creating new tracks with new buffers in AudioFlinger when the A2DP output is opened
instead of just transfering active tracks from hardware output mixer thread to the new A2DP output mixer thread.
To avoid synchronization issues between mixer threads and client processes, this is done by invalidating tracks
by setting a flag in their control block and having AudioTrack release the handle on this track (IAudioTrack)
and create a new IAudioTrack when this flag is detected next time obtainBuffer() or start() is executed.

AudioFlinger modifications:
- invalidate the tracks when setStreamOutput() is called
- make sure that notifications of output opening/closing and change of stream type to output mapping are sent synchronously to client process.
This is necessary so that AudioSystem has the new stream to output mapping when the AudioTrack detects the invalidate flag in the client process.
Previously their were sent when the corresponding thread loop was executed.

AudioTrack modifications:
- move frame count calculation and verification from set() to createTrack() so that is is updated every time a new IAudioTrack is created.
- detect track invalidate flag in obtainBuffer() and start() and create a new IAudioTrack.

AudioTrackShared modifications
- group all flags (out, flowControlFlag, forceReady...) into a single bit filed to save space.

Change-Id: I9ac26b6192230627d35084e1449640caaf7d56ee
54b1a0550411c2fd2084d82d28934d505c37349a 20-Mar-2010 Mathias Agopian <mathias@google.com> libutils Condition are now PRIVATE by default

Condition must be initialized with SHARED for the old behavior, where
they can be used accross processes.

Updated the two places android that require SHARED conditions.

PRIVATE conditions (and mutexes) use more efficient syscalls.

Change-Id: I9a281a4b88206e92ac559c66554e886b9c62db3a
867d2f6ce668968e463eb86b856d21525f12fd67 26-Jan-2010 Mathias Agopian <mathias@google.com> Simplify the MemoryDealer implementation

At some point the implementation became complicated because of
SurfaceFlinger's special needs, since we are now relying on gralloc
we can go back to much simpler MemoryDealer.

Removed HeapInterface and AllocatorInterface, since those don't need
to be paramterized anymore. Merged SimpleMemory and Allocation.
Made SimplisticAllocator non virtual.

Removed MemoryDealer flags (READ_ONLY, PAGE_ALIGNED)

Removed a lot of unneeded code.
6100d2d60517ff33ed8eb35d0b7ea63cde0831c9 19-Nov-2009 Eric Laurent <elaurent@google.com> Issue 2265163: Audio still reported routed through earpiece on sholes

This is a second attempt to fix the audio routed to earpiece syndrom.
The root cause identified this time is the crash of an application having an active AudioTrack playing on the VOICE_CALL stream type.
When this happens, the AudioTrack destructor is not called and the audio policy manager is not notified of the track stop.
Results a situation where the VOICE_CALL stream is considered as always in use by audio policy manager which makes that audio is routed to earpiece.

The fix consists in moving the track start/stop/close notification to audio policiy manager from AudioTrack to AudioFlinger Track objet.
The net result is that in the case of a client application crash, the AudioFlinger TrackHandle object (which implements the remote side of the IAudioTrack binder interface) destructor is called which in turn destroys the Track object and we can notify the audio policy manager of the track stop and removal.

The same modification is made for AudioRecord although no bug related to record has been reported yet.
Also fixed a potential problem if record stop is called while the record thread is exiting.
2b584244930c9de0e3bc46898a801e9ccb731900 10-Nov-2009 Eric Laurent <elaurent@google.com> Improvements for issue 2197683: English IME key-press latency is noticeably higher on passion than sholes

This change goes with a kernel driver change that reduces the audio buffer size from 4800 bytes (~27ms) to 3072 bytes (~17ms).
- The AudioFlinger modifcations in change 0bca68cfff161abbc992fec82dc7c88079dd1a36 have been removed: the short sleep period was counter productive when the AudioTrack is using the call back thread as it causes to many preemptions.
- AudioFlinger mixer thread now detects long standby exit time and in this case anticipates start by writing 0s as soon as a track is enabled even if not ready for mixing.
- AudioTrack::start() is modified to start call back thread before starting the IAudioTrack so that thread startup time is masked by IAudioTrack start and mixer thread wakeup time.
34f1d8ecd23169a5f299937e3aaf1bd7937578a0 04-Nov-2009 Eric Laurent <elaurent@google.com> Fix issue 2203561: Sholes: audio playing out of earpiece.

Create a new IAudioTrack interface to AudioFlinger when start() fails due to a broken pipe error.
Do the same if start fails due to the same error after time out in obtainBuffer().
Do not indicate that the AudioTrack is started to AudioPolicyManager if IAudioTrack start fails.
This avoids that an AudioTrack keeps a dead IAudioTrack after a media server crash.

Same modifications for AudioRecord.

Add a flag to ToneGenerator indicating that the callback thread can call Java. Without it, when the media server crashes and restarts, the AudioSystem error callback will crash in JNI if the IAudiotrack is created from AudioTrack callback thread.
3302526f6276911b2dc40c731ea7fa0e7972d908 04-Aug-2009 Eric Laurent <elaurent@google.com> Fix problem in AudioTrack with 8 bit PCM and direct output.

Do not perform 8 to 16 bit conversion in AudioTrack write() and processAudioBuffer() if direct output flag is set.
c2f1f07084818942352c6bbfb36af9b6b330eb4e 17-Jul-2009 Eric Laurent <elaurent@google.com> Fix issue 1795088 Improve audio routing code

Initial commit for review.
Integrated comments after patch set 1 review.
Fixed lockup in AudioFlinger::ThreadBase::exit()
Fixed lockup when playing tone with AudioPlocyService startTone()
b07c28b90b2d2793be2b8878d813b607f3eebbb7 14-Jul-2009 Mathias Agopian <mathias@google.com> add a ctor to Mutex to specify the type, which can be shared. This is used by sf and af an soon will allow some optimization in the kernel for non shared mutexes
349dba337e07e129f6ba49a132999f0b73fedbe3 07-Jul-2009 Eric Laurent <elaurent@google.com> am 88e209dc: Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR

Merge commit '88e209dcf8c2ebddda5c272f46d1bd5478bc639c'

* commit '88e209dcf8c2ebddda5c272f46d1bd5478bc639c':
Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR
573266210fb2b2e7d86fbd46d0dfe16763611d91 07-Jul-2009 Eric Laurent <elaurent@google.com> Fix issue 1743700: AudioTrack: setPlaybackRate can not set the playback rate to twice of the ouputSR

Store sample rate on 32 bits instead of 16 bits in audio_track_cblk_t.
Removed sampleRate() methods from AudioTrack and AudioRecord: replaced by getSampleRate().
AudioTrack::setSampleRate() no returns a status.
7562408b2261d38415453378b6188f74fda99d88 20-May-2009 Mathias Agopian <mathias@google.com> move libbinder's header files under includes/binder
1dd70b9f04961a06fcb73a97fca10a53b3245d3c 21-Apr-2009 Eric Laurent <elaurent@google.com> Fix issue 1745312: Various cleanups in media framework

AudioTrack, AudioRecord:
- remove useless mAudioFlinger member of AudioTrack and AudioRecord.
- signal cblk.cv condition in stop() method to speed up stop completion.
- extend wait condition timeout in obtainBuffer() when waitCount is -1 to avoid waking up callback thread unnecessarily

- remove some warnings in AudioFlinger.cpp.
- remove function AudioFlinger::MixerThread::removetrack_l() as its content is never executed.
- remove useless call to setMasterVolume in AudioFlinger::handleForcedSpeakerRoute().
- Offset VOICE_CALL stream volume to reflect actual volume that is never 0 in hardware (this fix has been made in the open source): 0.01 + v * 0.99.

- correct typo in comment

IAudioflinger, IAudioFlingerClient:
- make AudioFlinger binder interfaces used for callbacks ONEWAY.

- correct routeStrings[] table in AudioHardwareInteface.cpp
2c22aeb65e801f663a754d043062f85e49f77739 25-Mar-2009 Jean-Michel Trivi <> Automated import from //branches/donutburger/...@140663,140663
1179bc9b0e3d17c984e8f4ad38561c049dd102fa 19-Mar-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //branches/cupcake_rel/...@140373
8555d0867c3e8fe6cc5c7ad40af557fe6b92fa72 05-Mar-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //depot/cupcake/@136594
89fa4ad53f2f4d57adbc97ae1149fc00c9b6f3c5 04-Mar-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //depot/cupcake/@135843
15f767b960b38059a74a42a33e16d8df2aec8bc1 04-Mar-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //depot/cupcake/@135843
ad04d9201452001dbaac4349f084cc9316190b89 03-Mar-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //depot/cupcake/@132589
99ffda877980468a9ae31e013cd10fb3645df1b0 03-Mar-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //depot/cupcake/@137055
25658fd43d150a45fb37734a9f9f27f48bb5c133 19-Feb-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //branches/cupcake/...@132276
7a2146d5807030b2629f347736be5301b61e8811 13-Feb-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //branches/cupcake/...@131421
5e07b5774c8b376776caa4f5b0a193767697e97e 11-Feb-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //branches/cupcake/...@130745
e5198b620a9a208ec59ea8457282404725f8ff6e 20-Jan-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //branches/cupcake/...@127101
cce8bd12da6d8419a8770e4552a51ec297c250c2 10-Jan-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //branches/cupcake/...@125939
7b5eb023f8d87cca6d830ae6c11c6aadbe02aca8 18-Dec-2008 The Android Open Source Project <initial-contribution@android.com> Code drop from //branches/cupcake/...@124589
2729ea9262ca60d93047e984739887cfc89e82eb 21-Oct-2008 The Android Open Source Project <initial-contribution@android.com> Initial Contribution