| c447ded04f11169e9b96b31cd196b2c4ffa9f31c |
|
06-Jan-2015 |
Eric Laurent <elaurent@google.com> |
fix record from default input source
Commit caf7f48a0e broke capture from default
input source on some devices: the input source is now specified as part of
the audio attributes and the default source was not remapped to mic source.
Bug: 18736417.
Change-Id: I3f5da33cdb56f959aaef68ed1952342babc63aa0
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| caf7f48a0ef558689d39aafd187c1571ff4128b4 |
|
26-Nov-2014 |
Eric Laurent <elaurent@google.com> |
AudioRecord: use audio attributes instead of audio source.
Added AudioRecord constructor with audio attributes.
Replaced AudioPolicymanager::getInput() by getInputForAttr().
No new functionality for now.
Also:
- Fixed warnings in AudioPolicyManager
- Allocate audio session ID before calling getOutputForAttr() in
AudioTrack.
Bug: 16006090.
Change-Id: I15df21e4411db688e3096dd801cf579d76d81711
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| e83b55dc29ca16092ba02f36f55fa6e0e37fd78c |
|
14-Nov-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: new getOutputForAttr() prototype.
Update getOutputForAttr() prototype and group all logic
dealing with audio attributes to stream type conversion in
audio policy manager.
getOutputForAttr():
- specifies the audio session (for future use)
- returns a status code
- receives either stream type (for legacy) or audio attributes
- returns an updated streamtype
Remove logic dealing with legacy stream types to attributes conversion
from AudioTrack.
Use correct type for audio sessions in other APIs (startOutput() ...).
releaseOutput() specifies the audio session (for future use).
Bug: 18067208.
Change-Id: I1bfbe9626c04c7955d77f8a70aecfad2cb204817
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 4dc680607181e6a76f4e91a39366c4f5dfb7b03e |
|
29-Jul-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: pass session ID to capture activity methods
Indicate the audio session ID when calling getInput(),
startInput(), stopInput(), releaseInput().
Bug: 12378680.
Change-Id: I763793752f93e2f4e1445a5ab217c895af011038
/frameworks/av/media/libmedia/AudioRecord.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/AudioRecord.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/AudioRecord.cpp
|
| b3b1660ecb67f61f9da54efced8677fa3a6f4863 |
|
16-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Add audio_input_flags_t to IAudioPolicyService::getInput
Change-Id: I9f37be05f8dc7b85a8827a94e76ca0f45453e170
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 7df8c0b799d8f52d6386e03313286dbd7d5cdc7c |
|
03-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
IAudioFlinger::openRecord now suggests notificationFrames
Change-Id: I08885cc381d03c522a23289e74f0e1ed46563863
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 34fb29696b0f3abf61b10f8d053b1f33d501de0a |
|
19-Jun-2014 |
Mark Salyzyn <salyzyn@google.com> |
libmedia: 64-bit compile warnings
Change-Id: I600f062fa7148c01851023c1240c39939e648002
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 3bcffa136909c1fb6e88ee4efd12ccac18360a85 |
|
13-Jun-2014 |
Eric Laurent <elaurent@google.com> |
force command flush after releasing a reference to shared memory
After releasing a reference to a shared memory interface,
IPCThreadState::self()->flushCommands() must be called to ensure
that the corresponding reference is also cleared immediately in the
remote process. Otherwise the binder implementaiton will not
push the clear systematically causing the memory region not to be freed.
See issues 10711502 and 2801375.
Bug: 14057166.
Change-Id: If55f36f00d452d6cf00cf83bd2fba1c8f3abcb57
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| c263ca0ad8b6bdf5b0693996bc5f2f5916e0cd49 |
|
05-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
Squashed commit of the following:
commit 9128d6ffec43731d723f9b394f243d940f4c7e41
Author: Glenn Kasten <gkasten@google.com>
Date: Tue May 13 10:38:42 2014 -0700
Use of fast capture by normal capture
Will only configure fast capture path if the input buffer size is less than
10 ms and the input sample rate is same as the primary output sample rate.
Change-Id: I4a7cdc6069d750845412c626d27e83f72a1ab397
commit 2e5e0806a5abe7499848358ef5fde5c26405000d
Author: Glenn Kasten <gkasten@google.com>
Date: Mon Jun 2 08:29:22 2014 -0700
Add mPrimaryOutputSampleRate
Change-Id: I46b527fc3f2b5a5720a74b4f0b9a8f2e0d570b09
commit baf1d73467923996d1b1f2a9237260cc5697e050
Author: Andy Hung <hunga@google.com>
Date: Fri May 30 10:42:03 2014 -0700
Change parameter type for volume to float in AudioMixer
Change-Id: I4da1505ce852505f86f8e5b87f60e8edceeb30e0
commit 40fe20fa9760cd03c69778c2021cf7a490d75ece
Author: Andy Hung <hunga@google.com>
Date: Fri May 30 10:35:47 2014 -0700
Rename UNITY_GAIN to UNITY_GAIN_INT in AudioMixer
Change-Id: Ic040311305026f0b4c4280a5b3bef7a447ac1da3
commit 37c9a2b49f876abc5ff537a9ec036d7f0a423775
Author: Andy Hung <hunga@google.com>
Date: Thu May 29 21:33:13 2014 -0700
Refactor setVolumeRampVariables in AudioMixer
Change-Id: I8fcf3101bcea292de7c65433fa578f1c9cdd0974
commit 397070eca31f121d5d3993de1bfea99aaea5d4f3
Author: Andy Hung <hunga@google.com>
Date: Thu May 29 18:52:38 2014 -0700
Fix floating point output from mixer
A buffer pointer was being erroneously reset to buffer start,
potentially causing an audio glitch. The floating point
output mode is not enabled at this time, but will be in
the future.
Change-Id: If8b6414d232f064f3a2e2c5a6da889a91b27fb24
commit 2e61aa5b33b2247bbc5d4eaa0b519df9accd4bbc
Author: Andy Hung <hunga@google.com>
Date: Fri May 23 21:22:17 2014 -0700
Add multiple format capability to FastMixer
Floating point data from MixerThread into FastMixer.
Multiple output format capability from FastMixer to Sink.
Change-Id: I0da17810ee71381a39a006c46faec71108d22c26
commit b9ea653c702a785bbd23a66c5e588d40b4192c4e
Author: Andy Hung <hunga@google.com>
Date: Thu May 29 15:53:09 2014 -0700
Avoid resetting BufferProviders in mixer unnecessarily
Change-Id: Iad85c4dfd21be1dbf89dc11906106b34219376f8
commit 7f1a6d6da21c616f80cf9ba21bea11b419ec561b
Author: Andy Hung <hunga@google.com>
Date: Tue May 27 12:32:17 2014 -0700
Update dynamic resampler buffer fetching
Make the criteria tight for fetching to avoid storing
excessive frame data internal to the resampler.
This should reduce jitter in frame delivery computation.
Bug: 14962343
Change-Id: I7adaf714d11c272696ccdbf218bda994c7217477
commit b5e4aac07b9a02f0c803c090058602b03ac09ebb
Author: Glenn Kasten <gkasten@google.com>
Date: Tue May 27 12:30:54 2014 -0700
Allow kFastTrackMultiplier to be specified per device
Change-Id: I4eaaaf038df720cec4f5d9221d1b632970f9e3dd
commit b93cd97a52af31122df2da2cc0415cda888c8c73
Author: Andy Hung <hunga@google.com>
Date: Fri May 23 21:13:31 2014 -0700
Rename mixBuffer to mMixerBuffer in FastMixer
Likewise mixBufferState becomes mMixerBufferState.
This harmonizes with the naming in AF::MixerThread.
Change-Id: I1255d7c07cc2c6ee925d7430925236d2bd163122
commit 8340758622b9711365a8801806cbdf934803c63f
Author: Andy Hung <hunga@google.com>
Date: Mon May 12 16:51:41 2014 -0700
Add multiple format capability to AudioMixer
Change-Id: I04ac1cafd90b6ed652f8d51888ad07576678f0bc
Signed-off-by: Andy Hung <hunga@google.com>
commit 6b695b9d094820c232a897a3fabbe83d2b7193fe
Author: Glenn Kasten <gkasten@google.com>
Date: Thu Mar 13 14:59:31 2014 -0700
Start adding FastCapture based on FastThread WIP
This version supports at most one fast capture client.
Change-Id: Idf609bfc80ae22433433d66a5232c043c65506df
commit e951ad05a2c388471d7e2806d91e7d51325a150a
Author: Glenn Kasten <gkasten@google.com>
Date: Mon May 12 11:06:26 2014 -0700
Move validation of frameCount from set to openRecord_l
This move is needed because frameCount is validated on server side for
fast tracks (as should be done for normal tracks too).
Change-Id: I6d99e80869fd90fab373cf60ef348c01f075fbca
commit 73e76992dbba794894837c38e5472312ea829cf3
Author: Glenn Kasten <gkasten@google.com>
Date: Tue May 13 10:41:52 2014 -0700
Allow track buffer "allocation" to be from pipe
Change-Id: Ib9ac170f8e8b7746b3588157a56cbee3b753a1cb
commit 60de1d7ded05c6304037d4858b401094b1d2b4d3
Author: Andy Hung <hunga@google.com>
Date: Fri May 9 15:02:21 2014 -0700
Add format parameter to getTrackName() and track_t
Change-Id: Ia152a839014e235fbfb656104c15d7c1b456d02e
Signed-off-by: Andy Hung <hunga@google.com>
Change-Id: Ied0ade8b25d23e89bb03319a7e3135c238f735b9
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| e541269be94f3a1072932d51537905b120ef4733 |
|
16-May-2014 |
Andy Hung <hunga@google.com> |
Use new channel count functions for audio masks
Change-Id: Ia658ab4b6320d19fdb50f123c930918724ff0ef3
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/media/libmedia/AudioRecord.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/AudioRecord.cpp
|
| 828f883a43f66f77d776a75d0ea2b87c7c826071 |
|
07-May-2014 |
Glenn Kasten <gkasten@google.com> |
Fix a couple of fast capture typo bugs
AudioRecord constructor was not passing flags through to set().
Server-side check was using wrong kind of channel mask.
Change-Id: Ifaa880ec323771e9fd168262be05f3e539f53390
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 142f519aa1acd5804d111e60d100f170fed28405 |
|
26-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Use symbol AUDIO_IO_HANDLE_NONE from <system/audio.h>
Change-Id: Id6b1aa17558eb73e17f22b8eab6cd02e00a96dff
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 879707057cde0a83b5f6143c918c66b8f4a97581 |
|
28-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Make openRecord_l more like createTrack_l for fast tracks: part 3
Create the callback thread earlier, before creating the IAudioRecord,
so that the thread's tid is available as a parameter to openRecord().
Also move initialization of mCbf to same point as in AudioTrack.cpp.
Change-Id: I61ea4c5e2724ccfc691aaf51bc02a7c10d5a7495
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 2c48f27be6b3ae58e451d9b56c1dfd00e606f345 |
|
27-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Make openRecord_l more like createTrack_l for fast tracks: part 2
Assume double-buffering because we don't know the true HAL sample rate,
and for fast tracks we must accomodate kernel scheduling and app
computation jitter.
Change-Id: I983d6048a8b4814cfa5bf789397cdd9f1572256c
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 33ca5cf8826f5fcd9c011e8d336138b62d982f2a |
|
28-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Make openRecord_l more like createTrack_l for fast tracks: part 1"
|
| bce50bfc3846ab008bafa75c5d3f29fd7b5395f7 |
|
28-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Fix type of AudioTrack/AudioRecord parameter frameCount
It's size_t consistently
Change-Id: I29638ef59ac773218025f2403a3508a307b487e0
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 838b3d8bafa4a781e277870dee4e0390165cff52 |
|
28-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Fix type of AudioTrack/AudioRecord parameter notificationFrames
It's uint32_t consistently
Change-Id: If8298c7e9aeea2b951fe47b675adbdf48d104846
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| c6ba823a397abf865b02f4f48fe18231f94d8e87 |
|
27-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Make openRecord_l more like createTrack_l for fast tracks: part 1
Use the transfer mode to distinguish use cases, as the presence of
a callback handler is not sufficient. For example, the track could be
configured for synchronous transfer with write() or read(), and also
have a callback handler for position updates. But that does not mean
the track can operate in fast track mode.
Change-Id: I2a7f1f0ca98e68efe180b524496985109d8ce291
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| c08d20b6a37122ebf116262c9372509ed060d4c1 |
|
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Simplify error handling after track creation
IAudioFlinger::createTrack and IAudioFlinger::openRecord
both guarantee that (status == OK) == (sp<> != 0).
Change-Id: I91cb4f7e843019efb65cace7ba146f7da7aa5b59
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 045e739161f5ae00321a6cfba20935abb791005b |
|
26-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Move initialize of mInput to match AudioTrack
Change-Id: I4dc977f22f51cd618dc83d800b4b8756929a4612
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| f7cc3631c5c264e13ce8c6bc15bd829b2ff9b036 |
|
26-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Update comments to match AudioTrack
Change-Id: I7dd14eabd78c9130a157da2063a1d65ec4f9c65a
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| a5ed48d3476df7dd1e10b380a68e3333f2b646fd |
|
26-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Move initialization of mRefreshRemaining to match AudioTrack
This also fixes a bug where, for a re-created IAudioRecord,
mRefreshRemaining was not being reset correctly.
Change-Id: I9f721a4edf92aab859cf3f247ab7f65562d14fb0
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 591d9a3652f868652ccc48dd9e9714f3a9813963 |
|
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "mAudioRecord is always non-0 if set() is successful"
|
| 7c1b8941bb54e38f2a717e41af0d9565f5977c27 |
|
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Unify comments and whitespace between AudioTrack and AudioRecord"
|
| 882469cfe767188a4c67d2d83f3d72ab553a4818 |
|
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Update mReqFrameCount at same point in AudioTrack and AudioRecord"
|
| 99ef241a8d310b3319c02eac18de9bb7422c5ab0 |
|
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Permit AudioRecord to support non-linear formats in future"
|
| 9f474b7824abb0d5408d02e365cf37a986663979 |
|
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify and cleanup error handling in AudioRecord::getMinFrameCount"
|
| b42f318d9733f88c7eb9bedfd33b086b8ea5dff5 |
|
24-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Simplify and cleanup error handling in AudioRecord::getMinFrameCount
Change-Id: I8721ecedfb429c4e233453d1e768ddf69ecabbe4
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 089e87201522c8979ac8f00fa729e907f54c790b |
|
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
mAudioRecord is always non-0 if set() is successful
Change-Id: I0c2483210903c922f06f097ada373a37b9a90a02
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 5f631515d098c29603cda88f7a7e7580a2d55b57 |
|
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Unify comments and whitespace between AudioTrack and AudioRecord
Change-Id: I5320a6b2d7f7077cb12d7da4f2ca30a940100bf2
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 11cb175005db0090345f44bfa049b52fd8c809b8 |
|
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Update mReqFrameCount at same point in AudioTrack and AudioRecord
Change-Id: I12369dfbb9e75389f2cab015a706decdaf310a0d
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| c85df82b1e8b05714268926a8bb3deb9c65f9a22 |
|
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Permit AudioRecord to support non-linear formats in future
But still restricted to 16-bit PCM currently
Change-Id: I5df0e5033da9144ca73e44addf14a63d31406034
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 86f04663032ddaa25110149d709bbf896ad83b02 |
|
25-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Add log at entry to set() in AudioTrack and AudioRecord
Change-Id: Ife23b88474c1d62c0cf682c1a310d951f2c0f54a
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 6dd62fb91d82dedcfa3ab38c02eb0940b4ba932a |
|
06-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Multi-client recording
Supports multiple clients both at native sample rate and with resampling.
Change-Id: Icea55b4fd30751761b7debaa3ce016c79e712d8d
/frameworks/av/media/libmedia/AudioRecord.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/AudioRecord.cpp
|
| cac3daa6332bf6d1f7d26adc4a9915f3d7992dd9 |
|
07-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Use printf format %#x for audio_format_t in logs
Change-Id: I1c611d1037685d52ccc84efe0fccd6413ec938e9
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| b3b2e23fcf7e050710d23b82a6682c0f3d869b69 |
|
05-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Add AudioRecord::mReqFrameCount similar to AudioTrack
Change-Id: I62d6534a9581e84ae20c2422f7ad9aeda9b7c4df
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 0f11b51a57bc9062c4fe8af73747319cedabc5d6 |
|
01-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Fix unused parameter warnings in audio
Change-Id: I665ba3358dd9502f0adec70d486e7bf8a2e1b0fe
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| df576995a5aad3428aeeef765387d650945c161d |
|
28-Jan-2014 |
Eric Laurent <elaurent@google.com> |
AudioTrack: fix obtainBuffer timeout calculation
AudioTrack::obtainBuffer() passes a pointer to a timeout
variable that has gone out of scope when calling
ClientProxy::obtainBuffer().
Same fix for AudioRecord.
Bug: 11968591.
Change-Id: I22af8f94fa2f8cc54ab5c25c89167c805e754317
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 38e905b3cbba4da443d799b16999989781afc6d8 |
|
13-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Refactor code related to I/O handles to reduce chance for leaks
The AudioRecord input handle code was refactored earlier
to fix a potential handle leak, and to simplify the code:
> Change-Id: I124dce344b1d11c2dd66ca5e2c9aec0c52c230e2
This changelist refactors AudioTrack similarly,
and adds further cleanup of both AudioTrack and AudioRecord.
We attempt to implement the rules for referencing counting I/O handles,
but there is still the possibility of a handle leak if the client process
dies after allocating the handle reference but before releasing it.
That issue is being tracked separately.
Details:
- AudioSystem::getOutput() is now called within createTrack_l
- restoreTrack_l was missing offload info
now it has the info available,
but is not yet being called for offloaded tracks
- AudioTrack::getOutput() is now const
- Remove getOutput_l()
Change-Id: I44a0a623d24fc5847bcac0939c276400568adbca
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| f0002d142e6d24c5438600b2c259679de710f8ac |
|
24-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Replace control block frameCount_ by explicit in/out parameter"
|
| 3a0b6bd22aa32daa729b05c33896400807027eee |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use uint32_t sequence numbers for IAudioRecord"
|
| 269a7e210949c4b28ba84dc71b9f99f6ef14f94e |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Cleanup loop termination logic in AudioRecord::obtainBuffer()"
|
| 6d88aaf9cd810d96a4888dff8bd33d44cd01ccaa |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioRecord: fix unwanted overrun when restarting"
|
| 6a507298357c9ff869b97db57a890e58bd6bdec0 |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioRecord::getInputFramesLost() cleanup"
|
| 32a7e0a5f7b000951b9adb80b17796f35b024816 |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Unify comments between AudioTrack and AudioRecord"
|
| 5e1f79baa5c7355effbd2c9183787d6604487908 |
|
15-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
AudioRecord: fix unwanted overrun when restarting
Apply the same fix to AudioRecord as this CL did for AudioTrack:
> Change-Id: Id703f8dc092a6f07c905eee194054b4a978f979d
Details:
When restarting an AudioRecord from stopped state, it is necessary
to force refresh of mRemainingFrames by processAudioBuffer() as
the last read before stop() could be partial. No doing so will lead
into unnecessary sleep before filling the non contiguous part
of the buffer returned by obtainBuffer() when processAudioBuffer()
is executed for the first time after start().
Change-Id: I0ff04648e990e97bad5ea3db5be0c4151250d5e1
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| d8a9d0220e84ae0b5049385aa9b1a0d8ea02b5bb |
|
15-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Cleanup loop termination logic in AudioRecord::obtainBuffer()
Apply the same change to AudioRecord, that was already done for AudioTrack in this CL:
> Change-Id: I0fc48117946364cb255afd653195498891f622bd
If the proxy->obtainBuffer at the end of the "do" loop fails with status
DEAD_OBJECT, and the restoreRecord_l during the next trip of the "do"
loop also fails, then the value of the buffer fields will be based on
the previous proxy->obtainBuffer that returned DEAD_OBJECT. This will
have cleared the buffer, but it's tricky. So instead explicitly clear
the buffer after restoreRecord_l, before breaking out of the loop.
Change-Id: Ifc97f2ab7b8195d6b149502bb5d765d2983b07be
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 8ff50e7526d24aca11713006933b8dcb64ef6be9 |
|
15-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Use uint32_t sequence numbers for IAudioRecord
This makes it the same as for IAudioTrack.
Previously we had a mix of int, int32_t and uint32_t,
which could cause comparisons to fail.
Change-Id: Ic72ef650ee81f65db11e42ed16d818fdf3e310f4
/frameworks/av/media/libmedia/AudioRecord.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/AudioRecord.cpp
|
| 5b27ccd67c845aa20a12a1fb58339e7e81e3d536 |
|
15-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use AUDIO_SESSION_ALLOCATE instead of 0"
|
| 2b2165c75790050810460c8de3f414876bce4c0e |
|
13-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Unify comments between AudioTrack and AudioRecord
Change-Id: I00a1025e2891a1c96218b3c2187eaddda6614ebc
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 41721bb796da589f8a669d1db7687a0da8f88761 |
|
13-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Fix race condition in AudioRecord::pause followed by start
Bug: 11148722
Change-Id: Ia1e14133d73ac301fe06a047e70a573911822630
/frameworks/av/media/libmedia/AudioRecord.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/AudioRecord.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/AudioRecord.cpp
|
| 7c7be1e05634d96d08210efb4bdeb012ffba440d |
|
20-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Fix some (but not all) unused parameter warnings
Change-Id: Ia99e23a0b46db3f3e6aa46f9018e63c14f4af369
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| e93cf2ca27ae6f4a81d4ef548bbf10a34db6d98f |
|
24-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Cleanup openRecord error handling
Bug: 10888816
Change-Id: I84897dd7d30b370640b54e928f230604b873cb68
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 9d2c78c4798ffd8c276c1bf0eaa0b34bc255a2da |
|
23-Sep-2013 |
Eric Laurent <elaurent@google.com> |
AudioTrack: fix music resume
Fix regression introduced by commit 5a6cd22 in AudioTrack resume:
the callback thread was not signaled if paused internaly.
Bug: 10895013.
Change-Id: Ic356b115132d6fccbcee2d9bb855e92671dc20c5
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 5a6cd224d07c05b496b6aca050ce5ecf96f125af |
|
20-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Fix slow AudioTrack and AudioRecord destruction
There were two causes for the slowness:
When thread was paused, it used nanosleep and sleep. These usually
run to completion (except for POSIX signal, which we avoid because it
is low-level). Instead, replace the nanosleep and sleep by condition
timed wait, as that can be made to return early by a condition signal.
Another advantage of condition timed wait is that a condition wait was
already being used at top of thread loop, so it is a simpler change.
The AudioRecord destructor was missing a proxy interrupt that was correct
in AudioTrack. This proxy interrupt is needed in case another thread
is blocked in proxy obtainBuffer.
Does not address the 1 second polling for NS_WHENEVER.
Bug: 10822765
Change-Id: Id665994551e87e4d7da9c7b015f424fd7a0b5560
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| ac0bbe16f3eba46b3d8057b66c2aff9101fc6f7d |
|
10-Aug-2013 |
Jeff Brown <jeffbrown@google.com> |
Fix potential leak of audio input handle.
The audio input handle is ultimately owned by the audio recorder
object but it could be dropped on the floor if an error occurred
before that object was fully initialized.
Rearranged some of the argument validation and merged getInput_l
with openRecord_l to simplify the code and prevent such a leak
from occurring.
Bug: 10265163
Change-Id: I124dce344b1d11c2dd66ca5e2c9aec0c52c230e2
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 4a0efb77198c69df711ab369ac482a42dbdfab07 |
|
09-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Fix regression for AudioRecord streaming callback mode
The notification period (frequency of client wakeups) was broken in
streaming callback mode. Fast tracks were OK, but non-fast tracks with
normal latency weren't getting woken up at all.
Bug: 10222943
Change-Id: Ife9a2f57fa73c6eb921f1c5ba62de0bfcc20557b
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 230c5bde2779c618b26edc73443280b2d9371c70 |
|
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioRecord notification frames"
|
| 33f3177c08d238285b296d137e527ec99e34228f |
|
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioRecord::openRecord_l use flags"
|
| 743649fa70392b668377fb507d251b346c7b2769 |
|
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioRecord::openRecord_l now take flags"
|
| 3aa03e40668dd90390d9f1702f8c576e15b366c3 |
|
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "IAudioFlinger::openRecord track_flags_t flags is in/out"
|
| 77baebfe89c99415a0ee4c2ea87bfd8a0f9e51f1 |
|
05-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioRecord::set and constructor now take flags"
|
| f86b18b1ce3cf20a39aea1fcaf90ec66f9d6d63e |
|
05-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioRecord callback thread waits for priority boost"
|
| 6c104a4a245ea72b7bdf8c7c016121189045d4bc |
|
05-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioRecord has no default sample rate"
|
| 7cd9cf70e36ad4b8eb12e24f9adbbe6fd69edebd |
|
01-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
AudioRecord notification frames
Change-Id: I76ec536d1504eb9a558178b62bf225aace4b40d1
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 3151427b6b0adf99929433715bab6f1e505100c1 |
|
01-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
AudioRecord::openRecord_l use flags
Use the flags to determine input parameters for IAudioFlinger::openRecord.
Change-Id: I98d2726503af75c8830ce80ceaf3b94a755b342f
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 73493688f4190f790ee15d9ca54831cd64f4e195 |
|
01-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
AudioRecord::openRecord_l now take flags
The new parameter 'flags' of type audio_input_flags_t will be used for
requesting a fast track, but is currently ignored.
Change-Id: If68dfda8b2d4eaaca42927d721b4630c47f71f3b
/frameworks/av/media/libmedia/AudioRecord.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/AudioRecord.cpp
|
| 27f7b2a8fe899565487d8a326676a5f7d0a05a37 |
|
01-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
AudioRecord::set and constructor now take flags
The new optional parameter 'flags' of type audio_input_flags_t will be
used for requesting a fast track.
Change-Id: Ia7e070cb57c833e608352da354fb30dc26df6918
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 28f1351369682801e1bb40a835bdae3c97b73c1c |
|
31-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
AudioRecord callback thread waits for priority boost
Change-Id: Iae38fa4ac20a45751566169213a08a15deb0a2f6
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 954315a10089fa3684ac94db5be77c6655c08fc0 |
|
02-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Recording of non-linear formats is not supported
Such formats are already rejected in AudioRecord::set()
Change-Id: I5ba1fd9e4cd659e5226c75aa4f63e52f655e0521
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| f0f33c4acd231fa95deb9eeef2c46b0129e64463 |
|
31-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
AudioRecord has no default sample rate
Change-Id: I72feefdd6f3a623fd3669b80d4b264518fdc0929
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 96f60d8f04432a1ed503b3e24d5736d28c63c9a2 |
|
12-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Rename control block flags to mFlags
Change-Id: I7b6d31e24531954ab1ecdf3ed56c19433700bd89
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 291bb6d8947c5b0c062f0895d623c529259bfa39 |
|
17-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
AudioRecord and HAL input stream must be 16-bit PCM only
Currently there are 16-bit PCM assumptions in several places for capture:
- resampler API
- mRsmpInBuffer and mRsmpOutBuffer
- RecordThread::threadLoop upmix, downmix, and resampling
- possibly other places
Until those assumptions are removed, this CL enforces 16-bit PCM in both
client and server at all places where a format is checked.
Change-Id: I08b0570bff626ad0d341804825a72c14e61b4233
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 1ab85ec401801ef9a9184650d0f5a1639b45eeb9 |
|
31-May-2013 |
Glenn Kasten <gkasten@google.com> |
Include what is needed
Remove old includes.
Header files only include other header files that they directly need themselves.
Change-Id: Ic471386808d9f42ea19ccbd59cb50a5f83a89dd0
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 9f80dd223d83d9bb9077fb6baee056cee4eaf7e5 |
|
19-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
New control block for AudioTrack and AudioRecord
Main differences between old and new control block:
- removes the mutex, which was a potential source of priority inversion
- circular indices into shared buffer, which is now always a power-of-2 size
Change-Id: I4e9b7fa99858b488ac98a441fa70e31dbba1b865
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 7c027248e1a4ccd5b22bc4deafb03e2d87ac8f38 |
|
26-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Consistent whitespace
Change-Id: I118cce68d3b777f9ec9b6bfb70367496422a40f2
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| c9b2e20f7c9a71e07ef398152709c76079decbcd |
|
26-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Miscellaneous cleanup
Abbreviation framesReady to fRdy for new systrace.
Put inline const on one line.
Use local copy of mState in state.
Improve logging.
Line length 100.
Change-Id: I8201c3ce0e53fd464fd33d02544e52c342d40b68
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 2e4664677d72ce54201d3fd0beb0e10280add93c |
|
10-Jan-2013 |
Glenn Kasten <gkasten@google.com> |
Fix AudioRecord
Bug: 7965744
Change-Id: Ic024e7fb32f7459b8093c2cf6cd5752aade21ddb
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| a589764c3c0617c7a8996e929ce2d6db1cc01d77 |
|
05-Jan-2013 |
Glenn Kasten <gkasten@google.com> |
Fix bug in AudioRecord() constructor
It was calling set() with wrong parameter list.
This goes back to commit be916aa1267e2e6b1c148f51d11bcbbc79cb864c from 2010.
Change-Id: I2f6917765baf58260bf35e89a2cc59c199734ff6
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| d5681bc9a38fe4cd1d591e6ae62b9c68fb851041 |
|
22-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Start isolating control block accesses in a proxy"
|
| 49d24d3fcd01aef0ec1fa4abd9c1b85473f64c0f |
|
21-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove unnecessary parameter"
|
| 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/AudioRecord.cpp
|
| 89b629b398e87095cf262692f4e476d605fe87ed |
|
17-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Fix bug with discarded AudioRecord::read count
Formerly, if an AudioRecord::read() got a timeout on obtainBuffer()
after already successfully transferring some data, then it returned zero.
This had the effect of discarding a partial transfer, which resulted in
a gap in the audio data delivered to the app. Now if a timeout occurs
after a partial transfer, it returns that partial transfer count so that
no data is lost.
Change-Id: I0d9c2f4e495a400b56ef916a06613ba26537ca97
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| e3aa659e9cee7df5c12a80d285cc29ab3b2cbb39 |
|
04-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Start isolating control block accesses in a proxy
The proxy object will eventually be the only code that understands the
details of the control block. This should make it easier to change the
control block in the future.
Initial set of control block fields that are isolated:
- sample rate
- send level
- volume
Prepare for streaming/static separation by adding a union to the control
block for the new fields.
Fix bug in handling of max sample rate on a track. It was only checking
at re-configuration, not at each mix.
Simplify OutputTrack::obtainBuffer.
Change-Id: I2249f9d04f73a911a922ad1d7f6197292c74cd92
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| a42ff007a17d63df22c60dd5e5fd811ee45ca1b3 |
|
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Clean up channel count and channel mask
Channel count is uint32_t.
Remove redundant mask parameter to AudioTrack::createTrack_l()
and AudioRecord::openRecord_l().
Change-Id: I5dc2b18eb609b2c0dc3091994cbaa4628062c17f
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| aa25c0321bd4b9eb46a4d1cb4ac5fa47af30eeb4 |
|
17-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Merge ""if" statements use curly braces per media style"
|
| b603744e96b07b1d5bf745bde593fb2c025cefcf |
|
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Don't use control block frame count after create
This is part of a series to clean up the control block.
Change-Id: I7f4cb05aef63053f8e2ab05b286d302260ef4758
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| d65d73c4ae74d084751b417615a78cbe7a51372a |
|
23-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
"if" statements use curly braces per media style
Change-Id: I130e7849fd1da7a0b7fe56c3c53919d26e3843b8
/frameworks/av/media/libmedia/AudioRecord.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/AudioRecord.cpp
|
| 3b16c766d1ae2cfd8487e8ffb2b23936fc0a8e17 |
|
14-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Use uint32_t for sample rate
Change-Id: Ie240b48fb54b08359f69ecd4e5f8bda3d15cbe80
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| b36a7a68af073b1e7fd5cad6aa2c52223fd30efd |
|
13-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify AudioRecord::restoreTrack_l()
Finish removing CBLK_RESTORING and CBLK_RESTORED from control block flags,
and remove constant RESTORE_TIMEOUT_MS.
Also minor cleanup:
- Cache mCblk in local variable cblk and make cblk allocatable in a register.
- Use "iMem" for sp<IMemory>.
- Add missing error log to AudioRecord; it was already in AudioTrack.
This is part of a series to clean up the control block.
Change-Id: Ia5f5ab4763c392bc06a45851b167ddaee29e3455
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 83a0382dc17364567667a4e6135db43f5bd92efc |
|
12-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Move frame size out of the control block
This is part of a series to clean up the control block.
Change-Id: Ifab1c42ac0f8be704e571b292713cd2250d12a3f
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| b929e417853694e37aba1ef4399f188987b709d9 |
|
08-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Move buffers pointer out of the control block
This is part of a series to clean up the control block.
Change-Id: Ie474557db7cb360f2d9a0f11600a68f5a3d46f07
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 864585df53eb97c31e77b3ad7c0d89e4f9b42588 |
|
07-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Remove CBLK_DIRECTION from control block flags
This is part of a series to clean up the control block.
Change-Id: I0265fece3247356b585d4d48fbda6f37aea8a851
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 26ba972eafde73a26271ecf027a1d5988ce50eb8 |
|
22-Jun-2012 |
Glenn Kasten <gkasten@google.com> |
Removed unused fields in AudioRecord::Buffer
Change-Id: I89fc6d8f695b48516d956b0a9a4a43d408f369f9
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 9c5fdd83f9b9f49be35107971feb33528d60b945 |
|
05-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Simplify control block flag names
Use only one symbol per flag
Change-Id: Ia3582e2134abd60c896d11337face65383e79c7c
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 85ab62c4b433df3f1a9826bed1c9bec07a86c750 |
|
01-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Line length 100
Change-Id: Ib28fd7b9ce951a6933f006e7f8812ba617625530
/frameworks/av/media/libmedia/AudioRecord.cpp
|
| 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
|