History log of /frameworks/av/media/libmedia/AudioRecord.cpp
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
6f744d75d3439f7984245e3c489cc7cf91cea41c 06-Sep-2012 Eric Laurent <elaurent@google.com> AudioRecord: Fix minimum frame count calculation.

AudioRecord::set() was calling getMinFrameCount() with
a channel count instead of a channel mask.

Change-Id: Iabace7686426430fd53deac0c71b0c36aa64171c
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
d64cd233eef39430561c1e1df423336a199cc5d7 21-Feb-2012 Glenn Kasten <gkasten@google.com> AudioRecord::stop() return void

like AudioTrack::stop()

Change-Id: Iab62f4665151345f1ad5874c97a21d1a331f0154
/frameworks/av/media/libmedia/AudioRecord.cpp
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().

mActive:
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
1879fff068422852c1483dcf8365c2ff0e2fadfc 12-Jul-2012 Glenn Kasten <gkasten@google.com> Add tid parameter to IAudioFlinger::openRecord

Not yet implemented

Change-Id: I35523fb15ad71727ecc9f4bb870f07e4b7397dc4
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
20010053daabfa43fcfe781bbf004473b4c08538 22-Jun-2012 Glenn Kasten <gkasten@google.com> Remove acoustics from AudioSystem::getInput()

Change-Id: I29fb3ee5664c1f0ee0409c1bb2be087ecca637db
/frameworks/av/media/libmedia/AudioRecord.cpp
624a7fcb377f2a40109c16de5109ae8ea1f67a69 22-Jun-2012 Glenn Kasten <gkasten@google.com> Use audio_channel_mask_t in AudioRecord

Change-Id: I9e1b918b2635d961604a4a9d88eb1c7179a167a7
/frameworks/av/media/libmedia/AudioRecord.cpp
868a6a357018e5872e064b7a13a9b891e2078962 22-Jun-2012 Glenn Kasten <gkasten@google.com> Include what you use

Change-Id: I12ef9367d05dbe069c037b1b4acd6347a8cf3ece
/frameworks/av/media/libmedia/AudioRecord.cpp
f92eec53f886f43e4374a36195be55f2a7bbcf36 07-Mar-2012 Glenn Kasten <gkasten@google.com> Remove AudioRecord record_flags

Change-Id: I021ddcc1bcb63132a4597d13e3d09db2a5f2c628
/frameworks/av/media/libmedia/AudioRecord.cpp
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
destroyed.
- 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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
3acbd053c842e76e1a40fc8a0bf62de87eebf00f 28-Feb-2012 Glenn Kasten <gkasten@google.com> Configure policy of mediaserver threads

Change-Id: Ifd825590ba36996064a458f64453a94b84722cb0
/frameworks/av/media/libmedia/AudioRecord.cpp
a1472d9883e35edd280201c8be3191695007dfd4 30-Mar-2012 Marco Nelissen <marcone@google.com> Make AudioTrack/AudioRecord handle more than 2^32 frames

b/6160363
Change-Id: I471815012c6a113ec2c4dd7676e8fa288a70bc76
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
679ab0b0792846a89162ce41c953819d70030112 07-Mar-2012 Glenn Kasten <gkasten@google.com> Use AudioRecord::record_flags consistently

Change-Id: I6f369a2b99eb515603bc7d5629a07db2b96783fe
/frameworks/av/media/libmedia/AudioRecord.cpp
e8286332f3817a8b7cc4cfd8f6450a3913533660 29-Feb-2012 Glenn Kasten <gkasten@google.com> Merge "Shorten thread names"
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
/frameworks/av/media/libmedia/AudioRecord.cpp
606ee61616efdba4696ae591ad10a4be33d8c946 25-Feb-2012 Glenn Kasten <gkasten@google.com> AudioRecord const methods

Change-Id: Ifae4fd7820b650aaca2b13c8658c292db1c46c0f
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
83bc7f3cf78b28a818417f40a4f0c00593993366 04-Jan-2012 Glenn Kasten <gkasten@google.com> libmedia new can't fail on Android

Change-Id: Ie79dd5abb8078b35474bf0f1b3a6ff994a3a3360
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
d1a243e41caffa8fd346907eed4625c9c47c1a86 27-Jul-2011 Eric Laurent <elaurent@google.com> AudioRecord: Fix getInput()

AudioRecord::getInput() was issuing a query to get a new input stream from
audio policy service instead of returning the cached input stream in AudioRecord.

Change-Id: Ice324b7c60bc0898149023797bcb56a72091b9d3
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
671a636931295d9c33ffca74551a804479d01241 17-Jun-2011 Eric Laurent <elaurent@google.com> Added audio_bytes_per_sample() helper function

Change-Id: Ibfcd75c4c241a53d5f052c25ada091904991048a
/frameworks/av/media/libmedia/AudioRecord.cpp
c6854100cea4fcd0f20cb2ac8235c02d1849b3a1 02-Jun-2011 Glenn Kasten <gkasten@google.com> Remove unnecessary level of priority indirection

Change-Id: I942d43973c20a7ace8b0d3f78b4da97e45e996c6
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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>
/frameworks/av/media/libmedia/AudioRecord.cpp
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>
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
15304d601cbf83be6519ca53e1a26b97d50d0192 22-Jun-2010 Chia-chi Yeh <chiachi@android.com> media: add AudioRecord::getMinFrameCount().

Change-Id: I952071ab10aa49aa96b727d157b68470d69fff3d
/frameworks/av/media/libmedia/AudioRecord.cpp
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.

-AudioMixer
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
/frameworks/av/media/libmedia/AudioRecord.cpp
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
/frameworks/av/media/libmedia/AudioRecord.cpp
88335b1a749fe0157547907a2ce6c9632f4d2592 03-Mar-2010 Eric Laurent <elaurent@google.com> Fix issue 2428563: Camera rendered inoperable by voice call interruption.

The problem is that AudioRecord never exits read() when a timeout occurs while trying
to get new PCM data from audio hardware input buffer: it just keeps waiting and retrying until stop() is called.
In the same time, opencore AndroidAudioInput::audin_thread_func() loop cannot be exited when stuck
in AudioRecord::read() because the iExitAudioThread flag can only be sampled when AudioRecord::read()
returns. We remain stuck with the audio input thread running.

The fix consists in modifying AudioRecord behavior in case of timeout when getting new PCM samples.
We now wait only one timeout period and try to restart audio record, in case the problem is due to a media_server
process crash. If this fails, we exit read() with a number of bytes read equals to 0 so that
AndroidAudioInput::audin_thread_func() loop can exit.

Also modified Audioflinger::RecordThread() loop so that we attempt to recover from HAL read errors.
In case of read error, the input stream is forced to standby so that next read attempt does a
reconfiguration and restart of the audio input device.
/frameworks/av/media/libmedia/AudioRecord.cpp
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>
/frameworks/av/media/libmedia/AudioRecord.cpp
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.
/frameworks/av/media/libmedia/AudioRecord.cpp
148b266afe2ac92b5616c24e8d5160e6f9242f69 05-Dec-2009 Eric Laurent <elaurent@google.com> Fix issue 2304669: VoiceIME: starting and canceling voice IME yields persistent "error 8" state on future attempts and breaks voice search.

Fixed AudioFlinger::openInput() broken in change ddb78e7753be03937ad57ce7c3c842c52bdad65e
so that an invalid IO handle (0) is returned in case of failure.
Applied the same correction to openOutput().
Modified RecordThread start procedure so that a failure occuring during the first read from audio input stream is detected and causes
the record start to fail.
Modified RecordThread stop procedure to make sure that audio input stream fd is closed before we exit the stop function.

Fixed AudioRecord JAVA and JNI implementation to take status of native AudioRecord::start() into account
and not change mRecordingState to RECORDSTATE_RECORDING if start fails.
/frameworks/av/media/libmedia/AudioRecord.cpp
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.
/frameworks/av/media/libmedia/AudioRecord.cpp
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.
/frameworks/av/media/libmedia/AudioRecord.cpp
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()
/frameworks/av/media/libmedia/AudioRecord.cpp
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.
/frameworks/av/media/libmedia/AudioRecord.cpp
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.
/frameworks/av/media/libmedia/AudioRecord.cpp
7562408b2261d38415453378b6188f74fda99d88 20-May-2009 Mathias Agopian <mathias@google.com> move libbinder's header files under includes/binder
/frameworks/av/media/libmedia/AudioRecord.cpp
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

AudioFlinger:
- 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.

AudioSystem.java:
- correct typo in comment

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

AudioHardwareInterface:
- correct routeStrings[] table in AudioHardwareInteface.cpp
/frameworks/av/media/libmedia/AudioRecord.cpp
cd6725a333395ffeac3215ea4bf834a95aaa8def 25-Mar-2009 Eric Laurent <> Automated import from //branches/donutburger/...@142065,142065
/frameworks/av/media/libmedia/AudioRecord.cpp
7d563247cdac0509009d579bbf849157d47c38a9 25-Mar-2009 Jean-Michel Trivi <> Automated import from //branches/donutburger/...@141200,141200
/frameworks/av/media/libmedia/AudioRecord.cpp
89fa4ad53f2f4d57adbc97ae1149fc00c9b6f3c5 04-Mar-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //depot/cupcake/@135843
/frameworks/av/media/libmedia/AudioRecord.cpp
15f767b960b38059a74a42a33e16d8df2aec8bc1 04-Mar-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //depot/cupcake/@135843
/frameworks/av/media/libmedia/AudioRecord.cpp
ad04d9201452001dbaac4349f084cc9316190b89 03-Mar-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //depot/cupcake/@132589
/frameworks/av/media/libmedia/AudioRecord.cpp
99ffda877980468a9ae31e013cd10fb3645df1b0 03-Mar-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //depot/cupcake/@137055
/frameworks/av/media/libmedia/AudioRecord.cpp
925a349b45d1d16eaaca6a1f4827191831271ca0 20-Feb-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //branches/cupcake/...@132569
/frameworks/av/media/libmedia/AudioRecord.cpp
7a2146d5807030b2629f347736be5301b61e8811 13-Feb-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //branches/cupcake/...@131421
/frameworks/av/media/libmedia/AudioRecord.cpp
5e07b5774c8b376776caa4f5b0a193767697e97e 11-Feb-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //branches/cupcake/...@130745
/frameworks/av/media/libmedia/AudioRecord.cpp
e5198b620a9a208ec59ea8457282404725f8ff6e 20-Jan-2009 The Android Open Source Project <initial-contribution@android.com> auto import from //branches/cupcake/...@127101
/frameworks/av/media/libmedia/AudioRecord.cpp
7b5eb023f8d87cca6d830ae6c11c6aadbe02aca8 18-Dec-2008 The Android Open Source Project <initial-contribution@android.com> Code drop from //branches/cupcake/...@124589
/frameworks/av/media/libmedia/AudioRecord.cpp
2729ea9262ca60d93047e984739887cfc89e82eb 21-Oct-2008 The Android Open Source Project <initial-contribution@android.com> Initial Contribution
/frameworks/av/media/libmedia/AudioRecord.cpp