History log of /frameworks/av/include/media/AudioRecord.h
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
955e78180ac6111c54f50930b0c4c12395e86cf7 21-Feb-2012 Glenn Kasten <gkasten@google.com> AudioRecord locking

Fix race conditions for EVENT_MARKER and EVENT_NEW_POS callbacks.
Marker and new position update fields are protected by lock.

getSampleRate() doesn't need a lock because it reads from shared memory
control block.

Enforce that the parameter passed with EVENT_MARKER and EVENT_NEW_POS
cannot not be changed by the callback handler, and will not change during
the call by another thread.

Session ID should never change; log if it does.

Change-Id: Ia2c63cf1a71b10bb06c37981bd76437f83fffa91
d64cd233eef39430561c1e1df423336a199cc5d7 21-Feb-2012 Glenn Kasten <gkasten@google.com> AudioRecord::stop() return void

like AudioTrack::stop()

Change-Id: Iab62f4665151345f1ad5874c97a21d1a331f0154
68337edf595a0c345ba4b8adcd4f1e541a1d7eb7 12-Jul-2012 Glenn Kasten <gkasten@google.com> AudioRecord client threading cleanup

Rename ClientRecordThread to AudioRecordThread to be more similar to
AudioTrack naming.

Only create the thread once, and use resume() and pause() for start()
and stop(). This will allow us to have a known client callback thread
tid that we can pass to AudioFlinger before start().

Made mActive a bool not int.
mActive is protected by mLock; volatile is meaningless.
Fixed a few places where mActive was accessed without a lock:
- stopped()
- processAudioBuffer()
These aren't used internally, so no need for _l() versions.

Change-Id: I4b8a5c90f3a22d3894b344564cb1c5aef4f1fda2
d4070955e28ae62aa4be1657f9d32acde104bb86 12-Jul-2012 Glenn Kasten <gkasten@google.com> Remove dead code in libmedia

Change-Id: I7d8201590cda29c9fa99662a4fdba222091febfe
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
02e84eaff54414e9f10c0f605152728a682c6874 25-Jun-2012 Glenn Kasten <gkasten@google.com> AudioRecord comments

Group the private fields according to how they are used

Change-Id: I7ce3d0939510c10f34bd91a55f6e03afc8e7d43c
624a7fcb377f2a40109c16de5109ae8ea1f67a69 22-Jun-2012 Glenn Kasten <gkasten@google.com> Use audio_channel_mask_t in AudioRecord

Change-Id: I9e1b918b2635d961604a4a9d88eb1c7179a167a7
70be725da4d8aafb94d47c1962e897ecd5fdf823 22-Jun-2012 Glenn Kasten <gkasten@google.com> Remove unused AudioRecord::channels()

It was declared but never implemented or called.

Change-Id: If5846147fcbd7f6d4187971e2044dd6fd3252b42
868a6a357018e5872e064b7a13a9b891e2078962 22-Jun-2012 Glenn Kasten <gkasten@google.com> Include what you use

Change-Id: I12ef9367d05dbe069c037b1b4acd6347a8cf3ece
f92eec53f886f43e4374a36195be55f2a7bbcf36 07-Mar-2012 Glenn Kasten <gkasten@google.com> Remove AudioRecord record_flags

Change-Id: I021ddcc1bcb63132a4597d13e3d09db2a5f2c628
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
a0a98ca6ec9b599af79a597cb7c5350b61a77624 21-Apr-2012 Eric Laurent <elaurent@google.com> Made AudioRecord a subclasss of RefBase

Made AudioRecord a subclass of RefBase to allow using strong
references and solve concurrency issues.

Issue 6254582.

Change-Id: Ic1f3845958f477e8b2d23d3d25bf0f666addcb3b
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
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
679ab0b0792846a89162ce41c953819d70030112 07-Mar-2012 Glenn Kasten <gkasten@google.com> Use AudioRecord::record_flags consistently

Change-Id: I6f369a2b99eb515603bc7d5629a07db2b96783fe
985ed9a1a22ec7e6e245d3fb8e93d3a23bdc539b 02-Mar-2012 Glenn Kasten <gkasten@google.com> Merge "Fix typos and line length in AudioRecord comments"
606ee61616efdba4696ae591ad10a4be33d8c946 25-Feb-2012 Glenn Kasten <gkasten@google.com> AudioRecord const methods

Change-Id: Ifae4fd7820b650aaca2b13c8658c292db1c46c0f
417c27304c67057779693007a7bc08e4dd80c262 24-Feb-2012 Glenn Kasten <gkasten@google.com> Fix typos and line length in AudioRecord comments

Change-Id: I85cfb9a2b9b3ade098161aa7687b4d4f7eb226ea
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
4f9b0c54011eb8fd2ccfb393c2dcd51cd07800e0 08-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Remove dead mutex in AudioTrack/AudioRecord thread"
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
eba51fb3a361f67a6a64d5a16eba6084fe27d60e 23-Jan-2012 Glenn Kasten <gkasten@google.com> Use audio_source_t consistently

Was a mix of audio_source_t, uint8_t, and int.

Related fixes:
- fix comments in MediaRecorder.java
- AudioPolicyService server side was not checking source parameter at
all, so if the client wrapper was bypassed, invalid values could be
passed into audio HAL
- JNI android_media_AudioRecord_setup was checking source for positive
values, but not negative values. This test is redundant, since already
checked at Java and now checked by AudioPolicyService also, but might
as well make it correct.

Change-Id: Ie5e25d646dcd59a86d7985aa46cfcb4a1ba64a4a
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
a3f1fa308728976fc9ca1b4f37d26e633b32b9ac 18-Jan-2012 Glenn Kasten <gkasten@google.com> Fix incorrect includes of AudioTrack.h

Remove unnecessary includes of AudioTrack.h.
Use forward declaration of class names in preference to #include when possible.

Change-Id: I12982811fa75c2c7695d8bbfa595a7aaec047dc0
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
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
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>
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
15304d601cbf83be6519ca53e1a26b97d50d0192 22-Jun-2010 Chia-chi Yeh <chiachi@android.com> media: add AudioRecord::getMinFrameCount().

Change-Id: I952071ab10aa49aa96b727d157b68470d69fff3d
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
05bca2fde53bfe3063d2a0a877f2b6bfdd6052cf 26-Feb-2010 Eric Laurent <elaurent@google.com> Issue 2071329: audio track is shorter than video track for video capture on sholes

Add API to retrieve number of frames dropped by audio input kernel driver.

Submitted on behalf of Masaki Sato <masaki.sato@motorola.com>
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.
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.
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()
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.
95634c8b6ad5419e310a5196bcc37f5988ed82da 26-May-2009 Android (Google) Code Review <android-gerrit@google.com> am de8268d6: Merge change 2331 into donut

Merge commit 'de8268d6d1cd168510c490b17e93154d2eab767c'

* commit 'de8268d6d1cd168510c490b17e93154d2eab767c':
Fix issue 1846343 - part 1
f5879c1448cc6aebc51b26d3ec2399d66144f8f4 22-May-2009 Eric Laurent <elaurent@google.com> Fix issue 1846343 - part 1

This change is the first part of a fix for issue 1846343, :
- Added new enum values for input sources in AudioRecord and MediaRecorder for voice uplink, downlink and uplink+downlink sources.
- renamed streamType to inputSource in all native functions handling audio record.

A second change is required in opencore author driver and android audio input to completely fix the issue.
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
7d563247cdac0509009d579bbf849157d47c38a9 25-Mar-2009 Jean-Michel Trivi <> Automated import from //branches/donutburger/...@141200,141200
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
e5198b620a9a208ec59ea8457282404725f8ff6e 20-Jan-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //branches/cupcake/...@127101
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