| 93c3d41bdb15e39dac0faea9c5b60f1637cd477c |
|
01-Aug-2014 |
Eric Laurent <elaurent@google.com> |
AudioSystem: add API to query audio HW sync source
Add a method to query from the audio HAL the HW sync
source used for a given audio session.
Modify audio policy to select a direct output with HW sync
when requested.
Bug: 16132368.
Change-Id: I03038f9188f2d389f8a5fd76a671854013a4513e
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| de3f8392fbf380ba6f09d009b00d7172477389a2 |
|
28-Jul-2014 |
Eric Laurent <elaurent@google.com> |
rename AudioSystem::newAudioSessionId()
Rename AudioSystem::newAudioSessionId() to
AudioSystem::newAudioUniqueId() as it can be used
also for I/O handles.
Bug: 12378680.
Change-Id: I611ea3b5eb57a4b0774437f477ee87dc4ccc2cc2
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| cf2c0210c8afbe7d0661ccbbae3835b5ce73c0bf |
|
26-Jul-2014 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: update openInput() and openOutput()
Add parameters to openInput() and openOutput(): device address,
input source.
Allow caller to specify a given I/O handle
Group parameters in a struct audio_config.
Bug: 12378680.
Change-Id: I7e9af74c0d996561cc13cbee7d9012d2daf33025
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 7410591dad836434c72ddee66680802708b70c10 |
|
03-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Move AudioRecord frame count calculations to server
Buffer frame count and notification frame count
are now calculated by server instead of by client.
The server has more information and can do a better job.
Also fix a few bugs:
- If a fast track was re-created, even with same pipe depth, it would fail.
Now it can correctly re-create a fast track provided the pipe depth is same.
- Notification frame count for fast tracks was calculated by client
as 1/2 of the total frame count, which is a large value due to the pipe.
Now the notification frame count is set by server to the HAL frame count.
This should reduce latency for fast tracks.
- EVENT_OVERRUN were happening frequently when there was sample rate conversion,
because the client didn't know about the sample rate conversion,
and under-estimated the necessary buffer size. Now since server
calculates the buffer sizes, EVENT_OVERRUN is unlikely.
- RecordThread::createRecordTrack_l was checking for mono and stereo
for fast tracks. This is not necessary, and now we can handle a
multi-channel fast track.
Bug: 7498763
Change-Id: I0c581618e8db33084d5ff9ed50a592990c9749e8
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| ec40d284218466d8f0e832e7eb88e6ea6c479c88 |
|
16-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Add audio_input_flags_t to IAudioFlinger::openInput
For backward compatibility, until flags are correctly calculated,
we will assume that the request is for a low latency input stream.
Change-Id: I76746834e870df00833dc77cbdaa2edd2ffeec95
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 7df8c0b799d8f52d6386e03313286dbd7d5cdc7c |
|
03-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
IAudioFlinger::openRecord now suggests notificationFrames
Change-Id: I08885cc381d03c522a23289e74f0e1ed46563863
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 4b123406c10c17852734a1b691bb9ce2a4cb7caf |
|
11-Apr-2014 |
Eric Laurent <elaurent@google.com> |
IAudioFlinger interface extension for patch panel
Change-Id: Iaabe0a7e315d5725e00a74a6ed40339b98f20330
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| d776ac63ce9c013c9626226e43f7db606e035838 |
|
07-May-2014 |
Glenn Kasten <gkasten@google.com> |
IAudioFlinger::openRecord returns IMemory(s)
openRecord() now explicitly returns the control block and data buffer
as separate IMemory references. If the IMemory for data buffer
is 0, this means it immediately follows the control block.
Change-Id: Ic098f88f0e037f8fbe30006689e18cacacf09d06
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| b53fc4ef8da47df8d5f580e8ee6135866ffebc33 |
|
02-May-2014 |
Glenn Kasten <gkasten@google.com> |
resolved conflicts for merge of 47c888a9 to master
Change-Id: I4ba2fdc6374a93a892bb7651b0d174e495f09bf6
|
| 2301acc6a9c7a3af4ad01f3d1d0f76f13eca7350 |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Update comments
Change-Id: I5776313b9b49072cd666d28880f0d07cc73f827b
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 51beebba9d4989d05831a13bbe5cdf53ecac1bf5 |
|
26-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use symbol AUDIO_DEVICE_NONE from <system/audio.h>"
|
| 45faf7e02791993a487d6e038d16ff46395f1975 |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Use symbol AUDIO_DEVICE_NONE from <system/audio.h>
Change-Id: I61f882c5e7c949bf00d3bfc745ebf3b5e1c42a58
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 85d109a4b0eddd76a8c5cee170bc2bcc99d00118 |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Document AudioSystem::newAudioSessionId() failures
Change-Id: Iaa168722f362c36bdfa87fe20dc0a59b43cf1ca3
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| c5a17425986b4ce3384e6956762c86018b49c4a0 |
|
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Remove name output parameter from createTrack
It was only used for one log.
A better solution will be a per-track unique ID.
Change-Id: Ia440e02ae4a5a4019a9a2d08970e1ee93ac4c3a3
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| d2304db2fcb5112292105a0949a55986a4c9875f |
|
03-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Rename setStreamOutput to invalidateStream
And simplify by removing the unused I/O handle parameter 'output'.
Change-Id: Ie9c4df17a7378066312d4ed8790fda7a9125c95e
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 7074296c72b3be919ebc9dfd504f4572a188d2e7 |
|
18-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Fix warning due to incorrect log format
Change-Id: I58ff399ace74965ebf8e5690c20637385f9425ad
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| e03dd22716e688faa56467561827d9b1ea0c4214 |
|
28-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
When passing a size_t via binder, use 64-bits unconditionally
64-bits is almost always over-kill. But it's easier and cleaner to change
the binder code to be accurate, than to rename all the the API parameter
types to be 32-bit.
Bug: 12381724
Change-Id: Ib8f198d814a2027760ef24e9e3feacee21a973b1
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| d457c970c8d08519cd77280a90b61ae1e342cfe3 |
|
11-Feb-2014 |
Marco Nelissen <marcone@google.com> |
Track pid for each session
so they can be properly freed.
Change-Id: I6f389035bc29e74e7c367c1c6d0252b180f666b3
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 1d6fa7af1288b550faabe4ec2cf98684236723db |
|
11-Feb-2014 |
Narayan Kamath <narayan@google.com> |
resolved conflicts for merge of 566be7c3 to master
Change-Id: I7b1cc71057b2bd4f771e7bcf508a8c3abd6017ce
|
| 377b2ec9a2885f9b6405b07ba900a9e3f4349c38 |
|
03-Feb-2014 |
Kévin PETIT <kevin.petit@arm.com> |
Make frameworks/av 64-bit compatible
Contains the necessary changes to make frameworks/av build and work
on a 64-bit machine.
Signed-off-by: Craig Barber <craig.barber@arm.com>
Signed-off-by: Kévin PETIT <kevin.petit@arm.com>
Signed-off-by: Ashok Bhat <ashok.bhat@arm.com>
Signed-off-by: Marcus Oakland <marcus.oakland@arm.com>
Change-Id: I725feaae50ed8eee25ca2c947cf15aee1f395c43
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| f0002d142e6d24c5438600b2c259679de710f8ac |
|
24-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Replace control block frameCount_ by explicit in/out parameter"
|
| 0ce5ca5e113d58cbcb7cb48615d6dbbf3fbe7c66 |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Cleanup createTrack error handling"
|
| 0cde076ddb283c84c3801a2df4cc3df99bd1577f |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Cleanup createTrack error handling
Similar to cleanup done earlier for openRecord in this CL:
> Change-Id: I84897dd7d30b370640b54e928f230604b873cb68
Bug: 10888816
Change-Id: I804a47f898e0319a60a9bd58906bbb037e45cc25
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 5f972c031d4061f4f037c9fda1ea4bd9b6a756cd |
|
13-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
AudioRecord::getInputFramesLost() cleanup
Fixed bug that if the binder call failed (for example if the
IAudioFlinger binder is dead), then getInputFramesLost was returning
garbage. Now it correctly returns zero, which is the error value for
this method.
The type declarations for getInputFramesLost were inconsistent:
a mixture of unsigned int, size_t, and uint32_t. Now it returns uint32_t
everywhere, which is what the underlying HAL API returns.
Added a FIXME about the side effect behavior. This will need review
for multi-client.
Change-Id: Ifa2e117a87dbd0c1f2c892a31d1c3dd919bf1a0a
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| aea7ea06394bcb155972d82055d4ea59962e4051 |
|
26-Jun-2013 |
Glenn Kasten <gkasten@google.com> |
Use AUDIO_SESSION_ALLOCATE instead of 0
Also fix a couple of places where we were using AUDIO_SESSION_OUTPUT_MIX,
which happens to also be equal to 0, but has a different meaning.
Change-Id: I90e39be3b89f5021a96d9e3b8d10929013ca977f
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 74935e44734c1ec235c2b6677db3e0dbefa5ddb8 |
|
19-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Replace control block frameCount_ by explicit in/out parameter
in IAudioFlinger::createTrack and IAudioFlinger::openRecord
Change-Id: I09c644c80e92c8e744b1b99055988a2588b2a83d
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 9cae217050aa1347d4ac5053c305754879e3f97f |
|
14-Jan-2013 |
Marco Nelissen <marcone@google.com> |
Assign blame for playback wakelocks.
Set a work source for the playback wakelock, so that playback is
counted against the requesting app instead of the media server.
Cherrypicked from master.
b/9464621
Change-Id: I7329f88a288a95a582a78005a1c3d16a5a611e31
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 462fd2fa9eef642b0574aa7409de0bde3fec8d43 |
|
14-Jan-2013 |
Marco Nelissen <marcone@google.com> |
Assign blame for playback wakelocks.
Set a work source for the playback wakelock, so that playback is
counted against the requesting app instead of the media server.
Change-Id: I7329f88a288a95a582a78005a1c3d16a5a611e31
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 35295078ab59c8c5d143a54d5a55557c3ca62c51 |
|
07-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
resolved conflicts for merge of 9a7d11f0 to master
Change-Id: Ie4d101185a93d37ec280f608813ad62a077a584e
|
| fee4ce338d78eeb58af1f66831ead53322d3859e |
|
07-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Cleanup openRecord error handling" into klp-dev
|
| c666b8f16310fc826471c0fb6ee3f7e4d3a2d067 |
|
24-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am 8ee02943: am 1c7f35d1: Merge "soundpool: allocate shared memory heap by client" into klp-dev
* commit '8ee02943b61e0a7dd78e5596720261b185a223d5':
soundpool: allocate shared memory heap by client
|
| e93cf2ca27ae6f4a81d4ef548bbf10a34db6d98f |
|
24-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Cleanup openRecord error handling
Bug: 10888816
Change-Id: I84897dd7d30b370640b54e928f230604b873cb68
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 3d00aa6de95fb46e36f2bab4e3facdf0b96acf06 |
|
24-Sep-2013 |
Eric Laurent <elaurent@google.com> |
soundpool: allocate shared memory heap by client
Current SoundPool implementation allocates the shared memory heap
containing decoded PCM samples in mediaserver process.
When mediaserver process crashes, the shared memory heaps allocated by
AudioCache cannot be mapped anymore in the new instance of mediaserver.
This causes a silent failure to end playback of new sounds because
AudioFlinger believes the new AudioTracks are opened in streaming mode
and not static mode: it sees a NULL shared memory pointer when the track
is created.
The fix consists in allocating the memory heap in the client process. Thus
the heap is not lost when mediaserver restarts. The global memory usage is
the same as this is shared memory.
Also added a way to detect that a shared memory is passed when the track is
created but cannot be mapped on mediaserver side.
Also fix a crash in SoundPool when ALOGV is enabled.
Bug: 10894793.
Change-Id: Ice6c66ec3b2a409d75dc903a508b6c6fbfb2e8a7
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 507b2860cfe0bb4a712064f1c503caa9a7325886 |
|
01-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Consistent style for comparisons of raw pointers to NULL
Change-Id: Iec102a5ccb2fe69229887b5432cd1fb66f26f0cf
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| eeca32671896739e84050da5992d5f151a1629de |
|
01-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
IAudioFlinger::openRecord track_flags_t flags is in/out
This will allow AudioFlinger to tell client it is denying a request.
Change-Id: Iff2be3ad6636371bbda9c9899a283c94620c1f06
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| d054c32443a493513ab63529b0c8b1aca290278c |
|
12-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Move control block mName to createTrack() output
This is part of a series of CLs to clean up the shared memory
control block, by removing any fields that don't have to be there.
Change-Id: I6e51003a1293b6800258c31b22cff2eba42162e7
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| b1a270d1e926fb9a01b4265a7675ed0c2c8f4868 |
|
14-May-2013 |
Richard Fitzgerald <rf@opensource.wolfsonmicro.com> |
libmedia: offloaded playback support
- start() returns a status so that upper layers can
recreate a non offloaded track in case of error.
- Added states to handle offloaded tracks specific:
- waiting for stream end (drain) notification by
audio flinger
- allow pause while waiting for stream end notification
- getPosition() queries the render position directly from
audio HAL.
- disable APIs not applicable to offloaded tracks
- Modified track restoring behavior for invalidated
offloaded tracks: just send the callback and wait for
upper layers to create a new track.
- Added wait for stream end management in audio track client
proxy. Similar to obtainBuffer and should be factored in.
Change-Id: I0fc48117946364cb255afd653195498891f622bd
Signed-off-by: Eric Laurent <elaurent@google.com>
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 4182c4e2a07e2441fcd5c22eaff0ddfe7f826f61 |
|
15-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Use AudioSystem::setLowRamDevice() to configure memory
Bug: 9798886
Change-Id: I9321e3f369f1ed9429ae222e3926ebdeb012b8b0
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 7c5977f0322204240b3d1874a44c1f3911275ae5 |
|
02-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Explicitly compare raw pointers to NULL
Change-Id: Id2c7828a36a6912333465475b21fa87e294c83c7
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| ad3af3305f024bcbbd55c894a4995e449498e1ba |
|
25-Mar-2013 |
Richard Fitzgerald <rf@opensource.wolfsonmicro.com> |
Public API changes for audio offload support.
NOTE: this does _not_ include all private member variables added
to classes as part of offload support. Only public/protected functions
and stubs functions/variables needed to make the changes buildable.
- isOffloadSupported() added to audio policy service
A stub implementation is required to build, this always returns false
- setParameters() added to IAudioTrack
A stub implementation is required to build, this always returns
INVALID_OPERATION
- CBlk flag for stream end
- Change AudioSystem::getRenderPosition() to take an audio_output_t
so caller can specify which output to query
- Add AudioSystem::isOffloadSupported()
This is fully implemented down to the AudioFlinger function
AudioPolicyServer::isOffloadSupported() which is just a stub
that always returns false.
- Add EVENT_STREAM_END to AudioTrack interface.
STREAM_END is used to signal when the hardware has actually finished
playing all the data it was sent.
- Add event type enumeration to media player interface AudioSink callbacks
so that the same callback can be used to handle multiple types of
event. For offloaded tracks we also have to handle STREAM_END and
TEAR_DOWN events
- Pass audio_offload_info_t to various functions used for opening outputs,
tracks and audio players. This passes additional information about the
compressed stream down to the HAL when using offload.
For publicly-available APIs this is an optional parameter (for some of
the internal and low-level APIs around the HAL interface it is mandatory)
- Add getParameters() and setParameters() API to AudioTrack
Currently dummy implementations.
- Change AudioPlayer contructor so that it takes a set of bitflags defining what
options are required. This replaces the original bool which only specified
whether to use deep buffering.
- Changes to StageFright class definition related to handling tearing-down of
an offloaded track when we need to switch back to software decode
- Define new StageFright utility functions used for offloaded tracks
Currently dummy implementations.
- AudioFlinger changes to use extended audio_config_t.
Fills in audio_offload_info_t member if this info is passed in when
opening an output.
- libvideoeditor changes required to add the new event type parameter
to AudioSink callback functions
- libmediaplayerservice changes required to add the new event type parameter
to AudioSink callback functions
Change-Id: I3ab41138aa1083d81fe83b886a9b1021ec7320f1
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Eric Laurent <elaurent@google.com>
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 8d6cc842e8d525405c68e57fdf3bc5da0b4d7e87 |
|
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Remove unnecessary parameter
Just get the parameter on server side
Change-Id: I433a63104dbb257e0d862be2ab61847cb36d1c15
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 5876f2f28f31c1bd99864ba3bb1590e3d6765018 |
|
30-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Remove last bits of IAudioFlinger::channel_count
Change-Id: I9e13678e0aa32a86eb27367a4aff4b32b8aec8cc
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 26c77556efc30800466b60b3975bc35a70c8c28b |
|
16-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Fix time vs. bytes units bug in getRenderPosition
Rename correctLatency since it requires thread to be locked.
Use size_t for byte and frame counts.
Change-Id: I178fdd18bdb823813b9563927bdff8c0d28ca5a5
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| e33054eb968cbf8ccaee1b0ff0301403902deed6 |
|
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Use size_t for frame counts
Also fix typo: bufferCount should be frameCount.
Change-Id: Ibed539504db75ef99dc21c8ff1bf2987122063a5
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 3b16c766d1ae2cfd8487e8ffb2b23936fc0a8e17 |
|
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Use uint32_t for sample rate
Change-Id: Ie240b48fb54b08359f69ecd4e5f8bda3d15cbe80
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| b26e3e9f2ab0334bff21a4fa4851dbf6e57fba5d |
|
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Fix build warnings
Change-Id: Ic43bcca166a529a6431711b05a7fa21849b6a38b
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| e0b07179a48ee50fda931d2aa1b3c751d167e4d7 |
|
07-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Remove CBLK_FAST from control block flags
This is part of a series to clean up the control block.
Change-Id: Ic881a3560d9547cb63fcc0cefec87aa3da480e0d
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 85ab62c4b433df3f1a9826bed1c9bec07a86c750 |
|
01-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Line length 100
Change-Id: Ib28fd7b9ce951a6933f006e7f8812ba617625530
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| cc0f1cfb69ce8b8985fc2c0984847a06a13ad22d |
|
24-Sep-2012 |
Glenn Kasten <gkasten@google.com> |
Implement android.media.AudioManager.getProperty()
Bug: 6635041
Change-Id: I3386a4a6c226bc4eceaf65556119e4fb15f73224
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 28b76b334f92a15a2be3cc9e2f7d229a3275d1ac |
|
04-Jul-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_channel_mask_t for channel mask
Change-Id: I1c1896da48983aa9f1462a4b471f910498816f60
/frameworks/av/media/libmedia/IAudioFlinger.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/IAudioFlinger.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/IAudioFlinger.cpp
|
| 3acbd053c842e76e1a40fc8a0bf62de87eebf00f |
|
28-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Configure policy of mediaserver threads
Change-Id: Ifd825590ba36996064a458f64453a94b84722cb0
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 0ca3cf94c0dfc173ad7886ae162c4b67067539f6 |
|
18-Apr-2012 |
Eric Laurent <elaurent@google.com> |
rename audio policy output flags
Change-Id: I27c46bd1d1b2b5f96b87af7d05b951fef18a1312
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| a4c5a550e2a3bc237179b8684e51718e05894492 |
|
29-Mar-2012 |
Eric Laurent <elaurent@google.com> |
audio policy: load audio hw modules.
Audio HW modules are now loaded upon request from audio policy manager
according to the configuration in audio_policy.conf.
Removed hard coded HW module loading by AudioFlinger at init time.
Added methods to IAudioFlinger and AudioPolicyInterface
to control the loading of audio HW modules.
Added methods to open an output or input stream on a specific hw module.
Change-Id: I361b294ece1a9b56b2fb39cc64259dbb73b804f4
/frameworks/av/media/libmedia/IAudioFlinger.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/IAudioFlinger.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/IAudioFlinger.cpp
|
| 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
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 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>
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 9d1f02d74fd395ec4de6861147da289423f0ab6f |
|
09-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Follow raw pointer and sp<> conventions
Unconditional delete for raw pointers.
Use "if (sp != 0)" not "if (sp.get() != 0)" or "if (sp != NULL)".
Use "if (raw != NULL)" not "if (raw)".
Change-Id: I531a8da7c37149261ed2f34b862ec4896a4b785b
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 72ef00de10fa95bfcb948ed88ab9b7a177ed0b48 |
|
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_io_handle_t consistently instead of int
Other:
- add a comment to nextUniqueId
- made ThreadBase::mId const, since it is only assigned in constructor.
Change-Id: I4e8b7bec4e45badcde6274d574b8a9aabd046837
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 5e92a7861196ddae14638d4b7a63fc4892b7ef59 |
|
30-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Effect UUID inputs passed by pointer are const
Change-Id: I1f5c338bcb7368e3dd8cd5f804b2e6d9fbe087f8
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| f587ba5b991c7cd91e4df093d0d796bd419e5d67 |
|
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Declare more IAudioFlinger methods const
This is just documentation, as C++ method const-ness doesn't mean anything
for a binder API. Instead, here const means "no side effects".
Change-Id: Iaa9cd2fe477db10ae9a40cac4f79f0faa9b4e5e6
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 787bae0578fbaab6219ebf23494866b224d01438 |
|
03-Feb-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use audio_in_acoustics_t consistently"
|
| a0d68338a88c2ddb4502f95017b546d603ef1ec7 |
|
28-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use NULL not 0 for raw pointers
Use if (p != NULL) instead of if (ptr)
Change-Id: Iaec3413a59ccbf233c98fcd918cc7d70ac5da9fa
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| de9719b3ec71472e6bf75117152176af51d1a515 |
|
27-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_in_acoustics_t consistently
Change-Id: I0a9dd668fb2e57b1c3ece3190588194974b99062
/frameworks/av/media/libmedia/IAudioFlinger.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/IAudioFlinger.cpp
|
| 0696400a6bb9abbed62b3b9c6aa105495dc600a2 |
|
17-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Use audio_mode_t consistently"
|
| 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
instead of AUDIO_STREAM_VOICE_CALL.
Change-Id: Ia33bfd70edca8c2daec9052984b369cd8eee2a83
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| f78aee70d15daf4690de7e7b4983ee68b0d1381d |
|
04-Jan-2012 |
Glenn Kasten <gkasten@google.com> |
Use audio_mode_t consistently
It was int or uint32_t.
Also make getMode() const.
Change-Id: Ibe45aadbf413b9158e4dd17f2b3bcc6355288d37
/frameworks/av/media/libmedia/IAudioFlinger.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/IAudioFlinger.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/IAudioFlinger.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/IAudioFlinger.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/IAudioFlinger.cpp
|
| 65580f9adf6c4d98449ad0716488f9fe3869aa5a |
|
28-May-2011 |
Eric Laurent <elaurent@google.com> |
Removed interface to load audio effects libraries
Removed unused functions allowing dynamic loading of audio effects libraries
from effects factory API.
Change-Id: I06cc5a51dc10aca87c7a8687bbb874babd711eca
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| eda6c364c253ba97ee45a3adeb8c2b45db1f81db |
|
02-Feb-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 3371080
Modified default volume control logic in AudioService:
1 IN_CALL volume if in video/audio chat
2 NOTIFICATION if notification is playing or was playing less than 5s ago.
3 MUSIC
Modified silent mode:
- now also affect MUSIC stream type
- entering silent mode when VOL- hard key is pressed once while selected
stream volume is already at 0 (except for VOICE_CALL stream).
- exiting silent mode when pressing VOL+ hard key while in silent mode
Play sound FX (audible selections, keyboard clicks) at a fixed volume.
Modified audio framework:
- isStreamActive() method now implemented in AudioPolicyManagerBase (previously AudioFlinger)
- iStreamActive() now specifies a time window during which the stream is considered
active after it actually stopped.
Change-Id: I7e5a0724099450b9fc90825224180ac97322785f
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| de070137f11d346fba77605bd76a44c040a618fc |
|
13-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Audio policy manager changes for audio effects
Added methods for audio effects management by audio policy manager.
- control of total CPU load and memory used by effect engines
- selection of output stream for global effects
- added audio session id in parameter list for startOutput() and stopOutput().
this is not used in default audio policy manager implementation.
Modifications of audio effect framework in AudioFlinger to allow moving and reconfiguring
effect engines from one output mixer thread to another when audio tracks in the same session
are moved or when requested by audio policy manager.
Also fixed mutex deadlock problem with effect chains locks.
Change-Id: Ida43484b06e9b890d6b9e53c13958d042720ebdb
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| ffe9c25ce85e1af55d58ec025adc6367d70db7e8 |
|
24-Jun-2010 |
Eric Laurent <elaurent@google.com> |
Various fixes and improvements in audio effects implementation
Effect API:
- Use different definitions for audio device, channels, formats... in AudioSystem and EffectApi:
Removed media/AudioCommon.h file created for initial version of EffectApi
- Indicate audio session and output ID to effect library when calling EffectCreate(). Session ID can be useful to optimize
the implementation of effect chains in the same audio session. Output ID can be used for effects implemented in audio hardware.
- Renamed EffectQueryNext() function to EffectQueryEffect() and changed operating mode:
now an index is passed for the queried effect instead of implicitly querying the next one.
- Added CPU load and memory usage indication in effects descriptor
- Added flags and commands to indicate changes in audio mode (ring tone, in call...) to effect engine
- Added flag to indicate hardware accelerated effect implementation.
- Renamed EffectFactoryApi.h to EffectsFactoryApi.h for consistency with EffectsFactory.c/h
Effect libraries:
- Reflected changes in Effect API
- Several fixes in reverb implementation
- Added build option TEST_EFFECT_LIBRARIES in makefile to prepare integration of actual effect library.
- Replaced pointer by integer identifier for library handle returned by effects factory
Audio effect framework:
- Added support for audio session -1 in preparation of output stage effects configuration.
- Reflected changes in Effect API
- Removed volume ramp up/down when effect is inserted/removed: this has to be taken care of by effect engines.
- Added some overflow verification on indexes used for deferred parameter updates via shared memory
- Added hardcoded CPU and memory limit check when creating a new effect instance
Change-Id: I43fee5182ee201384ea3479af6d0acb95092901d
/frameworks/av/media/libmedia/IAudioFlinger.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/IAudioFlinger.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/IAudioFlinger.cpp
|
| 342e9cf388cceb807def720e40e8b0a217f4bcaa |
|
20-Jan-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2285561: New AudioFlinger and audio driver API needed for A/V sync
Added getRenderPosition() API to IAudioFlinger to retreive number of audio frames
written by AudioFlinger to audio HAL and by DSP to DAC.
Added getRenderPosition() API to AudioHardwareInterface to retreive number of audio frames
written by DSP to DAC.
Exposed AudioTrack::getPosition() to AudioSink() to make it available to media player.
Removed excessive log in AudioHardwareGeneric.
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| b72a396826da8bd934b9531bbd40f86d7509e71c |
|
25-Jan-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2378022: AudioService should direct volume control to STREAM_VOICE_CALL stream when STREAM_VOICE_CALL stream is active.
Modified AudioService.getActiveStreamType() so that STREAM_VOICE_CALL is selected when a track using this stream
type is playing.
Chanded isMusicActive() for a more generic isStreamActive(stream) method in AudioSystem, IAudioFlinger and AudioFlinger.
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| f0ee6f4055e26fb35d9c526a596668a4dc9da5ba |
|
21-Oct-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 2192181: AudioFlinger must provide separated methods to set VOICE_CALL stream volume and down link audio volume.
Added setVoiceVolume() method to AudioSystem, AudioFlinger, IAudioFlinger, AudioPolicyService.
Removed call to AudioHardwareInterface::setVoiceVolume() from AudioFlinger::setStreamVolume().
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 5841db78dff14898538200287d246577b1fc37e2 |
|
09-Sep-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 2107584: media server crash when AudioFlinger fails to allocate memory for track control block.
AudioFlinger: verify that mCblk is not null before using it in Track and RecordTrack contructors.
IAudioFlinger: check result of remote transaction before reading IAudioTrack and IAudioRecord.
IAudioTrack and IAudioRecord: check result of remote transaction before reading IMemory.
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| fa2877b9ea48baed934b866d2ab3658b69c4c869 |
|
28-Jul-2009 |
Eric Laurent <elaurent@google.com> |
Fix issue 2001214: AudioFlinger and AudioPolicyService interfaces should not use pointers as handles to inputs and outputs.
Use integers instead of void* as input/output handles at IAudioFlinger and IAudioPolicyService interfaces.
AudioFlinger maintains an always increasing count of opened inputs or outputs as unique ID.
/frameworks/av/media/libmedia/IAudioFlinger.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/IAudioFlinger.cpp
|
| b7056fc8525e0515c6f46676e3307088f8b64cdc |
|
23-May-2009 |
Mathias Agopian <mathias@google.com> |
some work to try to reduce the code size of some native libraries
- make sure that all binder Bn classes define a ctor and dtor in their respective library.
This avoids duplication of the ctor/dtor in libraries where these objects are instantiated.
This is also cleaner, should we want these ctor/dtor to do something one day.
- same change as above for some Bp classes and various other non-binder classes
- moved the definition of CHECK_INTERFACE() in IInterface.h instead of having it everywhere.
- improved the CHECK_INTERFACE() macro so it calls a single method in Parcel, instead of inlining its code everywhere
- IBinder::getInterfaceDescriptor() now returns a "const String16&" instead of String16, which saves calls to String16 and ~String16
- implemented a cache for BpBinder::getInterfaceDescriptor(), since this does an IPC. HOWEVER, this method never seems to be called.
The cache makes BpBinder bigger, so we need to figure out if we need this method at all.
/frameworks/av/media/libmedia/IAudioFlinger.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/IAudioFlinger.cpp
|
| 7562408b2261d38415453378b6188f74fda99d88 |
|
20-May-2009 |
Mathias Agopian <mathias@google.com> |
move libbinder's header files under includes/binder
/frameworks/av/media/libmedia/IAudioFlinger.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/IAudioFlinger.cpp
|
| 89fa4ad53f2f4d57adbc97ae1149fc00c9b6f3c5 |
|
04-Mar-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //depot/cupcake/@135843
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 15f767b960b38059a74a42a33e16d8df2aec8bc1 |
|
04-Mar-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //depot/cupcake/@135843
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 7a2146d5807030b2629f347736be5301b61e8811 |
|
13-Feb-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //branches/cupcake/...@131421
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 5e07b5774c8b376776caa4f5b0a193767697e97e |
|
11-Feb-2009 |
The Android Open Source Project <initial-contribution@android.com> |
auto import from //branches/cupcake/...@130745
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 7b5eb023f8d87cca6d830ae6c11c6aadbe02aca8 |
|
18-Dec-2008 |
The Android Open Source Project <initial-contribution@android.com> |
Code drop from //branches/cupcake/...@124589
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|
| 2729ea9262ca60d93047e984739887cfc89e82eb |
|
21-Oct-2008 |
The Android Open Source Project <initial-contribution@android.com> |
Initial Contribution
/frameworks/av/media/libmedia/IAudioFlinger.cpp
|