| 0f7b5f2b231caf87da9b20b74d086e5a9d6f4a9d |
|
19-Dec-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: pause HW A/V sync output when AudioTrack underruns
Do not standby, starve or feed 0s to the audio HAL on direct
output using HW A/V sync mode.
Bug: 17883772.
Change-Id: I11e6c97ec24360d75f9b602814d40a54b60cb7a7
/frameworks/av/services/audioflinger/Threads.cpp
|
| d1f69b0b17acbd96987ecb2f3378abd394d05903 |
|
15-Dec-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: implement pause/resume for direct outputs
Extend pause/resume support to direct output threads
(was only for offload threads).
If the HAL implements pause/resume, track pause/resume is forwarded to
the HAL.
Pause, flush, resume sequence is respected by executing the HAL
calls in the playback thread (same as offload).
Make sure the track flags on client side are consistent with the
flags on server side.
Bug: 17883772.
Change-Id: I89b360d69818f7a9204bd36e3ec63a79e106ecf1
/frameworks/av/services/audioflinger/Threads.cpp
|
| 94a92c69af528edf6ec17d7978a0c3bb6ab51e63 |
|
11-Dec-2014 |
Phil Burk <philburk@google.com> |
Merge "[audio][audioflinger] Consume all frames when a DirectThread is done." into lmp-mr1-dev
|
| 99adee3c3d9cde6819741a38163954808fea270a |
|
11-Dec-2014 |
Phil Burk <philburk@google.com> |
[audio][audioflinger] Consume all frames when a DirectThread is done.
This bug was causing movies to think there was still data left to play so they never finished.
Bug: 18360054
Change-Id: If7aee81d62b7af8532abd14579677848a06fe2b1
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| 275e8e9de2e11b4b344f5a201f1f0e51fda02d9c |
|
01-Dec-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: add support for custom mixes
Add support for custom mixes in AudioPolicyManager.
Two methods are added to register or unregister a list of custom mixes
with their attributes and format.
getOutputForAttr() and getInputForAttr() first look for a match in
registered mixes before defaulting to normal output/input selection
Remote submix device connection disconnection now takes address into
account to identify the correspnoding custom mix.
Bug: 16009464.
Change-Id: I3f1c2a485a0fb71b1f984ed0adc9b68aa971e408
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9a1c8892f6835325db6931529dc74bf98cf0aee8 |
|
03-Dec-2014 |
Andy Hung <hunga@google.com> |
Fix getTimestamp if offload thread is not initialized.
Returns INVALID_OPERATION.
Bug: 18605649
Change-Id: I440fb687fbf9249098e049982fa83bcd1515fd80
/frameworks/av/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| 223fd5c9738e9665e495904d37d4632414b68c1e |
|
11-Nov-2014 |
Eric Laurent <elaurent@google.com> |
audio: new routing strategies and stream types
Added new routing strategies and stream type for internal use
by audio policy manager and audio flinger:
- One for accessibility to allow different routing than media
- One for re-routing (remote submix) in preparation of dynamic
policies
- Added stream type for "internal" audio flinger tracks used
for audio patches and duplication.
Bug: 18067208.
Change-Id: I88f884b552e51e4a49c29125e5a1204cf58ff434
/frameworks/av/services/audioflinger/Threads.cpp
|
| ec0eeafa8a7fd882e8206e9cf8c4c7e1db81857b |
|
20-Nov-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
Fix format conversion in DuplicatingThread
DuplicatingThread converts PCM data from the format of the source thread
to PCM_16BIT before writing to all connected output tracks.
This needs to be done just once before writing to all tracks.
Bug: 18461844
Change-Id: If05254f87f9198e09a0fa3c4e857afbb53ae7073
/frameworks/av/services/audioflinger/Threads.cpp
|
| 02b5708776ba2a9b4ff8c09008483aef7dbe38c7 |
|
08-Nov-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix duplicating thread output buffers clear
Also clear mix buffer in DuplicatingThread::threadLoop_mix()
instead of just sink buffer when output threads are not
ready.
Bug: 18262127.
Change-Id: I4c22bbeaccdcc8d7c10c5882dde3439eb73d6ede
/frameworks/av/services/audioflinger/Threads.cpp
|
| ac302143551a8b964f026385a524dda9ff8ea5ba |
|
20-Oct-2014 |
Marco Nelissen <marcone@google.com> |
Fix the fix
We need to clear the buffer a little more often.
Bug: 18023017
Change-Id: If8a522b9133a4269e2435af8958f6ced322e71e9
/frameworks/av/services/audioflinger/Threads.cpp
|
| 57088b5c8e76855b99b3e6b3e410de5b6382670e |
|
18-Oct-2014 |
Marco Nelissen <marcone@google.com> |
Clear effect buffer before sending to chain
otherwise the effect might get a buffer with previous data instead
of silence.
Bug: 18023017
Change-Id: Ic6be5ce0e74552aac2a0cce9c660893f9ad27f5c
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2b217bb3aee87ce8486014f261c0f498f6209e80 |
|
03-Oct-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: fix pre processing transfer between record threads." into lmp-dev
|
| 1b92868010b5c1409692a86f6b27e4a265b64c1a |
|
03-Oct-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix pre processing transfer between record threads.
Fix two problems remaining with pre processing effects transfer from
one record thread to the next in case of tear down due to device connection:
1 - the enabled state of the effects was not communicated to the new HAL
input stream.
2 - the effects saved in orphan chains list were not transfered to the
new thread when a AudioRecord was created.
Bug: 17757378.
Change-Id: I0923c98470db3b51154dc89846157780a4c21e86
/frameworks/av/services/audioflinger/Threads.cpp
|
| 4c053ea158b29fa2cdd4c6f39d3c8da4ee5a7a02 |
|
28-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp
The per-thread timestamp latch was not synchronized with the per-track
released frames. Now the value of each track's released frames is
latched along with the timestamp.
Bug: 17531839
Bug: 17669342
Change-Id: I9d50c8c6a5de55a3f4561ac40e20d497376c1257
/frameworks/av/services/audioflinger/Threads.cpp
|
| a49c94ea49c8329a1bdefe980e1f7d335c0d1ec5 |
|
02-Oct-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Revert "Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp"" into lmp-dev
|
| 6954127b7ace022677ac407ff943c2793f8a11be |
|
02-Oct-2014 |
Glenn Kasten <gkasten@google.com> |
Revert "Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp"
This reverts commit 54464ba861aaafd11ee5645f5d1ecd1171c6e9fe.
Bug: 17733142.
Change-Id: I9886d40032b2f7d96bee03e600e14f78943d8052
/frameworks/av/services/audioflinger/Threads.cpp
|
| 955e24d3a8e218d4711cabc6558781e095011132 |
|
30-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp" into lmp-dev
|
| f71224ce266abb39f83ecba87783c2347631654a |
|
30-Sep-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: forward flush to direct output streams" into lmp-dev
|
| 54464ba861aaafd11ee5645f5d1ecd1171c6e9fe |
|
28-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp
The per-thread timestamp latch was not synchronized with the per-track
released frames. Now the value of each track's released frames is
latched along with the timestamp.
Bug: 17531839
Bug: 17669342
Change-Id: I96252782911bd42a2534cc4030521b7eeb99a7b9
/frameworks/av/services/audioflinger/Threads.cpp
|
| e659ef420dae0caae84ab78f9df8952acb9ad3a0 |
|
29-Sep-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: forward flush to direct output streams
Track flush should be forwarded to the audio HAL when playing
on a direct output stream.
Bug: 17704953.
Change-Id: Iafc94327fe9aebf1d4eaa97b76c6767b7ac5babe
/frameworks/av/services/audioflinger/Threads.cpp
|
| ba0b34c18da93681c0813ecdab19b0e215b6d261 |
|
28-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
Fix tee sink
Note: this does not fix the bug below, but it helps to investigate the bug.
Bug: 17578238
Change-Id: I0fd27a477b73ba0e1c5a658bdfad17c500f5dc67
/frameworks/av/services/audioflinger/Threads.cpp
|
| 80b617cf82e0030deef4f968a6341e2867166135 |
|
17-Sep-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: remove debug log.
Remove debug log left by commit ab5cdbaf.
bug: 17537746.
Change-Id: I10f672f607f8e87caaa8228fc05eff8aee7dfa15
/frameworks/av/services/audioflinger/Threads.cpp
|
| aaa44478a373232d8416657035a9020f9c7aa7c3 |
|
13-Sep-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix pre processing effect leak
When a capture thread was closed, the effects attached to this thread
were left dangling and the associated effect chain destroyed.
When their last client was disconnected, the effects were not released
properly from the effect library because the destruction process could
not be completed without the effect being attached to a thread.
A similar problem prevented a RecordTrack to be properly released if
its client was destroyed after the capture thread.
The fix consists in allowing the effect or record track to be properly
released even if its parent thread cannot be promoted.
Also save any effect chain still present on a closed capture thread
in case a new client wants to reuse the effects on the same session later.
Bug: 17110064.
Change-Id: I5cd644daa357afd1f3548f9bcb28e6152d95fdb8
/frameworks/av/services/audioflinger/Threads.cpp
|
| 3abc2ded40066f3b1df23aceb553f22d569c5cd3 |
|
06-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
Make IPowerManager native conform to .aidl for oneway
But provide a temporary escape hatch for AudioFlinger.
This oneway option will be removed as soon as possible.
Bug: 16408906
Change-Id: I058a50906af810787e444a96819cb781b7f639ad
/frameworks/av/services/audioflinger/Threads.cpp
|
| 97d547da43c9c41711d1ed1e3f4fa87c2ee3cb9a |
|
02-Sep-2014 |
Eric Laurent <elaurent@google.com> |
audio flinger: fix use of global effect buffer
When float mixing format is enabled, audio effects use a specific
buffer in 16 bit format. This buffer was not enabled when only
global effects were present resulting in the mixer accumulating directly
to the output sink buffer.
Bug: 16202259.
Change-Id: I3537d36c63196d97772f0ab22dd31d04dfa7fdd6
/frameworks/av/services/audioflinger/Threads.cpp
|
| 735f45fbc37d7905ffb722f40727edbed82319b7 |
|
19-Aug-2014 |
Glenn Kasten <gkasten@google.com> |
Fix memory access error due to race during fast track removal
Bug: 16205169
Change-Id: Ie15ec2a737b6ddd16b2d26030d01564ff70177f5
/frameworks/av/services/audioflinger/Threads.cpp
|
| 49d00ad9164ea5ce48c85765a2b6460d9b457d38 |
|
21-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Fix overruns when capturing with resampling
Bug: 16658148
Change-Id: I2dd04b23a143e0948b299d408790e7a46b5a53b0
/frameworks/av/services/audioflinger/Threads.cpp
|
| cd04484f4837b8ca0041d118286ab6a98e84fc75 |
|
07-Aug-2014 |
Andy Hung <hunga@google.com> |
Extend downsampling ratios greater than 2:1
Also improve robustness to choice of sampling rate or buffer size
such that increasing either by 10x does not cause overflow.
Bug: 12979141
Bug: 15933066
Change-Id: If7989bd745d1bee3bdf811b8b7c978543ccafb65
/frameworks/av/services/audioflinger/Threads.cpp
|
| 389cfdbb9a92a438a0d7710321c2964c7ad55eca |
|
08-Aug-2014 |
Andy Hung <hunga@google.com> |
Fix effect and mono sink handling.
Do not permit mixing to mono sink in AudioFlinger.
Allow effects on mono sink if not Mixer thread (e.g. Record).
Bug: 16863095
Change-Id: I8b232fc1fb3f07bf017020c5d4f9ace644dec6d8
/frameworks/av/services/audioflinger/Threads.cpp
|
| cf817a2330936947df94c11859f48771f5596a59 |
|
05-Aug-2014 |
Eric Laurent <elaurent@google.com> |
audio: fix crashes upon USB device connection
- Audio policy:
Replace unknown device assertion by debug log in setDeviceConnectionState().
It is possible that some kernels indicate connection of a legacy device type
when a USB audio device is connected.
We should just ignore this device.
- Audio flinger:
Make sure FastMixer thread is idle before exiting the normal mixer thread and
closing the output stream.
Change-Id: Ia10a20176a60a8aa56765538093a555fc998508a
/frameworks/av/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| 8661aaff9e9a4d4c1b57f6a68cdbcab006354ab2 |
|
28-Jul-2014 |
Andy Hung <hunga@google.com> |
Fix rounding in RecordThread resampler buffer computation
Bug: 16525305
Change-Id: Ie010822b0fa9dec74adde1d1b091ee44c183fc94
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9f81de3452dfb2385bd57dc05456a045174a1ab1 |
|
28-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Enable lower latency audio capture on more devices
Bug: 16601366
Change-Id: I573b98631f81b7768f7cc00aa09e560008dea587
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9a59276fb465e492138e0576523b54079671e8f4 |
|
22-Jul-2014 |
Andy Hung <hunga@google.com> |
Add multichannel capability to AudioFlinger
But not enabled (kEnableExtendedChannels == false).
Change-Id: I62f7e31fbd29ad703a9a02f5d1a280b6972dd423
/frameworks/av/services/audioflinger/Threads.cpp
|
| 83b8808faad1e91690c64d7007348be8d9ebde73 |
|
21-Jun-2014 |
Eric Laurent <elaurent@google.com> |
audio flinger: add patch connection between hw modules
Add support for audio device connections between different audio
hw modules.
The patch is performed by creating a bridge between the playback
thread connected to the sink device and the record thread connected
to the source device using a pair of specialized PlaybackTrack and
RecordTrack.
- Added PatchTrack and PatchRecord classes.
- Added TrackBase type to indicate more clearly the track behavior.
- A TrackBase can allocate the buffer or reuse an existing one.
- Factored some code in openOutput() and openInput() for internal use
by PatchPanel.
Bug: 14815883.
Change-Id: Ib9515fcda864610458a4bc81fa8f59096ff4d7db
/frameworks/av/services/audioflinger/Threads.cpp
|
| a7e04803ffa0d8d4cc51a122f68bf5038422aabb |
|
21-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Remove execute bit
Change-Id: I1cf0c0d9375779b3074af12f4cbf9c1ec07bf4ff
/frameworks/av/services/audioflinger/Threads.cpp
|
| 91be1fbc424035e70688e60306b755893e0fb0ae |
|
17-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Move AudioRecord frame count calculations to server" into lmp-dev
|
| 2e8186a258c934798129847f66171df36e95d23e |
|
17-Jul-2014 |
Andy Hung <hunga@google.com> |
Merge "Fix initial audio glitch on startup" into lmp-dev
|
| 5567aaf4818007cd8e77329683a91c0f5d7a8837 |
|
17-Jul-2014 |
Andy Hung <hunga@google.com> |
Fix initial audio glitch on startup
Also happens when restarting AudioFlinger.
Bug: 16322824
Change-Id: I382b20174cb0639e04316130141f2094d7dd0cb3
/frameworks/av/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| 6e6704c06d61bc356e30c164081e5bcffb37920c |
|
03-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Improve dumpsys for capture threads
Show per-track sample rate, and whether there is a fast capture thread
Change-Id: If6a08a40d1eeba0690aa8da3c541f845ec101b7f
/frameworks/av/services/audioflinger/Threads.cpp
|
| cd704219d22ec51660103684a680caf2c3a12dad |
|
15-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
const correctness on downmix and upmix
Change-Id: I0f51470a6d58a22310ce7cda80afc55e9d6df1dc
/frameworks/av/services/audioflinger/Threads.cpp
|
| 463be250de73907965faa6a216c00312bf81e049 |
|
11-Jul-2014 |
Andy Hung <hunga@google.com> |
Return negotiated format with HAL in AudioFlinger
Fixes an issue caused by extended precision and the FastMixer
when the format to the FastMixer was returned instead of the
HAL format.
Change-Id: I2be9d55d5c87d82e46b5200b07fcdd986ebde08a
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5c68f959eaa2e02fed5643c78e281fff42bcc0a2 |
|
07-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "IAudioFlinger::openRecord now suggests notificationFrames"
|
| 7df8c0b799d8f52d6386e03313286dbd7d5cdc7c |
|
03-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
IAudioFlinger::openRecord now suggests notificationFrames
Change-Id: I08885cc381d03c522a23289e74f0e1ed46563863
/frameworks/av/services/audioflinger/Threads.cpp
|
| 665470b36f202bcc8ee2f7417f68fd2608dd07c1 |
|
04-Jul-2014 |
Eric Laurent <elaurent@google.com> |
audio: deprecate audio_stream_frame_size()
Bug: 15000850.
Change-Id: I955a6efa620b8f70fa5193c7853cbe88e879c9c3
/frameworks/av/services/audioflinger/Threads.cpp
|
| f92f22becdf7fce1f55d5ebd80ac2caa2ad55602 |
|
30-Jun-2014 |
Andy Hung <hunga@google.com> |
Merge "Add enabling variable for extended precision audio"
|
| 6146c08f0c3dd8b9e5788063aa433f304a810602 |
|
18-Mar-2014 |
Andy Hung <hunga@google.com> |
Add enabling variable for extended precision audio
Set AudioFlinger::kEnableExtendedPrecision = true to enable
extended precision. Enabling will be required for devices (such as
USB) which report 24 bit or 32 bit sink formats.
Change-Id: I0dc1d7a4f7607086d7b536ea0e43aef0e696f2ee
/frameworks/av/services/audioflinger/Threads.cpp
|
| 4d23ca370dd0ce584f49a80ef9dfcdbb75ba2c8e |
|
13-May-2014 |
Glenn Kasten <gkasten@google.com> |
Use sp<> instead of raw pointer for FastMixer
This avoids worry about the thread lifetime during tear-down.
Bug: 15858655
Change-Id: I896d24317bd6f53a574645cdf4cd514eaa205dd5
/frameworks/av/services/audioflinger/Threads.cpp
|
| ab5cdbaf65ca509681d2726aacdf3ac8bfb6b3fa |
|
10-Jun-2014 |
Eric Laurent <elaurent@google.com> |
AudioTrack: add support for compressed audio
Add support for compressed audio playback by use
of an AudioTrack attached to a direct output thread.
Bug: 9428304.
Change-Id: I4a61be9cf0e31003ca85935d6e0ee38ca8192e03
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5094675aa55458414e8b00a86a104a32ea1d55c0 |
|
16-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
am a3603693: Merge "Fix deadlock" into lmp-preview-dev
* commit 'a3603693fb784d7e1a820ee4629a69730a0bb424':
Fix deadlock
|
| 3d61bc1ffc8afc8d7be3b0d4205c9b5ba6daf2e8 |
|
16-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
Fix deadlock
Bug: 15591052
Change-Id: I76154e1640646d2ae8fae65e163e1f3ed1be17dc
/frameworks/av/services/audioflinger/Threads.cpp
|
| b6be7f22a82ee3bad8bcc709d21e72fc4727da09 |
|
12-Jun-2014 |
Bryant Liu <bryant_liu@htc.com> |
frameworks: av: fix bug in the volume management
which can cause volume levels to be applied twice. After applying volume
in the effect chain, the remaining volume needs to be converted back to
floating point before providing to mAudioMixer.
Bug: 15583440
Change-Id: I3c884f95b0fa9ad1b554285b069601c850c54776
Signed-off-by: Glenn Kasten <gkasten@google.com>
/frameworks/av/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| 8f415570afb174f0781d8bf6b1c0fa6bdecca7cd |
|
03-Jun-2014 |
Andy Hung <hunga@google.com> |
Merge "Change parameter type for volume to float in AudioMixer"
|
| 6dbb5e3336cfff1ad51d429fcb847307c06efd61 |
|
13-May-2014 |
Glenn Kasten <gkasten@google.com> |
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
/frameworks/av/services/audioflinger/Threads.cpp
|
| 6be494077f8d7970f3a88129c5d139c5a0c88f6d |
|
30-May-2014 |
Andy Hung <hunga@google.com> |
Change parameter type for volume to float in AudioMixer
Change-Id: I4da1505ce852505f86f8e5b87f60e8edceeb30e0
/frameworks/av/services/audioflinger/Threads.cpp
|
| 1258c1ab592a899fabb1e31eb5db2ef413b6f38a |
|
24-May-2014 |
Andy Hung <hunga@google.com> |
Add multiple format capability to FastMixer
Floating point data from MixerThread into FastMixer.
Multiple output format capability from FastMixer to Sink.
Change-Id: I0da17810ee71381a39a006c46faec71108d22c26
/frameworks/av/services/audioflinger/Threads.cpp
|
| 951f455566775e5f01e67c5ee26863d7d19209d7 |
|
20-May-2014 |
Eric Laurent <elaurent@google.com> |
DO NOT MERGE - audioflinger: first patch panel implementation.
Added a new PatchPanel subclass to AudioFlinger
to handle audio ports and audio patches configuration
and connection.
The first implementation does not add new functionnality.
AudioPolicyManager uses patch panel interface to control
device routing.
AudioFlinger:
- Added PatchPanel class. The first implementation does not
add new functionnality. PatchPanel handles routing commands
for audio HAL after 3.0 or converts to setParameters for audio
HALs before 3.0.
- Added config events to ThreadBase to control synchronized
audio patch connection.
AudioPolicyManager:
- Use PatchPanel API to control device selection isntead of setParameters.
- New base class AudioPort common to audio device descriptors
and input output stream profiles. This class is RefBase and groups
attributes common to audio ports.
- Use same device selection flow for input as for outputs:
getNewInputDevice -> getDeviceForInptusiource -> setInputDevice
Change-Id: Idaa5a883b19a45816651c58cac697640dc717cd9
/frameworks/av/services/audioflinger/Threads.cpp
|
| ecc03733bfd3262ffadef3166e6be23b539c505c |
|
28-May-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: first patch panel implementation."
|
| 0349009fd19f89f8414c428f6b71b369f7546085 |
|
27-May-2014 |
Glenn Kasten <gkasten@google.com> |
Allow kFastTrackMultiplier to be specified per device
Change-Id: I4eaaaf038df720cec4f5d9221d1b632970f9e3dd
/frameworks/av/services/audioflinger/Threads.cpp
|
| 1c333e252cbca3337c1bedbc57a005f3b7d23fdb |
|
20-May-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: first patch panel implementation.
Added a new PatchPanel subclass to AudioFlinger
to handle audio ports and audio patches configuration
and connection.
The first implementation does not add new functionnality.
AudioPolicyManager uses patch panel interface to control
device routing.
AudioFlinger:
- Added PatchPanel class. The first implementation does not
add new functionnality. PatchPanel handles routing commands
for audio HAL after 3.0 or converts to setParameters for audio
HALs before 3.0.
- Added config events to ThreadBase to control synchronized
audio patch connection.
AudioPolicyManager:
- Use PatchPanel API to control device selection isntead of setParameters.
- New base class AudioPort common to audio device descriptors
and input output stream profiles. This class is RefBase and groups
attributes common to audio ports.
- Use same device selection flow for input as for outputs:
getNewInputDevice -> getDeviceForInptusiource -> setInputDevice
Change-Id: Idaa5a883b19a45816651c58cac697640dc717cd9
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5912117110ae6efadd862fcb0e395fdcd18aa00c |
|
27-May-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Move validation of frameCount from set to openRecord_l"
|
| c62476f0c0c1cf9283a38852bde0a4c9434df712 |
|
24-May-2014 |
Andy Hung <hunga@google.com> |
Merge "Add format parameter to getTrackName() and track_t"
|
| 29b703eec27b305e7b5b2343bf257643e38f6b68 |
|
12-May-2014 |
Glenn Kasten <gkasten@google.com> |
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
/frameworks/av/services/audioflinger/Threads.cpp
|
| 87cebadd48710e42474756fc3513df678de045ce |
|
22-May-2014 |
Elliott Hughes <enh@google.com> |
resolved conflicts for merge of f285c91e to master
Change-Id: I4c995e297d09c0854a42142cc57dce8b771acf65
|
| c10b874c4d0d6208fb93a7aa0dc8af4418faad45 |
|
22-May-2014 |
Calin Juravle <calin@google.com> |
am d5516339: Merge "Move frameworks/av from fdprintf to POSIX dprintf."
* commit 'd551633916d3677da4dda10aba6837bdc6803ee0':
Move frameworks/av from fdprintf to POSIX dprintf.
|
| 8b5f642eb2364ea7fe46a5b3af51b48b58f12183 |
|
22-May-2014 |
Elliott Hughes <enh@google.com> |
Move frameworks/av from fdprintf to POSIX dprintf.
Bug: 11156955
Change-Id: Ia98cd16b4c1f7be87cf060b7456de4f40896bacb
/frameworks/av/services/audioflinger/Threads.cpp
|
| e348c5b72ad889389c7c1c900c121f0fbee221b5 |
|
22-May-2014 |
Elliott Hughes <enh@google.com> |
resolved conflicts for merge of c0bf836f to master
Change-Id: I5cf238fa56d6fff443b212870cf9dd7f1110e7da
|
| ad71caf25e31ecfc4ecf2c925e38fd87905fbc8e |
|
22-May-2014 |
Elliott Hughes <enh@google.com> |
am 3b6da6bf: Merge "Move frameworks/av off private API."
* commit '3b6da6bfb50a1c0324fb4c87326d3a477944d5fe':
Move frameworks/av off private API.
|
| ee499291404a192b059f2e04c5afc65aa6cdd74c |
|
22-May-2014 |
Elliott Hughes <enh@google.com> |
Move frameworks/av off private API.
Bug: 11156955
Change-Id: Ib3bb9d66a2bf92977c2445b62aa074a2d7a45aea
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9382378b6b271949bdce527ea3da748c13c4c692 |
|
22-May-2014 |
Elliott Hughes <enh@google.com> |
resolved conflicts for merge of b1080917 to master
Change-Id: Iabbe3e95e29f28461ff722b550e8a9f35bd2528f
|
| acbd9492996348bf4565fbaa5a14772f0f4ace38 |
|
22-May-2014 |
Elliott Hughes <enh@google.com> |
am d084206d: Merge "Move frameworks/av off __futex_syscall3."
* commit 'd084206d5a4a5364211af46e4201311933e83bfc':
Move frameworks/av off __futex_syscall3.
|
| 77e1512aee8092069514973379a59a318e721be3 |
|
22-May-2014 |
Elliott Hughes <enh@google.com> |
Move frameworks/av off __futex_syscall3.
Bug: 11156955
Change-Id: I943080bc51b6efa781059b48b5b5ecc4fa287512
/frameworks/av/services/audioflinger/Threads.cpp
|
| b4db73d022f3de3530bc2b3c9c831ccfdd1a2ead |
|
20-May-2014 |
Andy Hung <hunga@google.com> |
Merge "Use new channel count functions for audio masks"
|
| 1e54e6dfc1bb8d33089ed8857f0f194e696f85e2 |
|
19-May-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Pass stereo gains as packed minifloat"
|
| e8a1ced4da17dc6c07803dc2af8060f62a8389c1 |
|
10-May-2014 |
Andy Hung <hunga@google.com> |
Add format parameter to getTrackName() and track_t
Change-Id: Ia152a839014e235fbfb656104c15d7c1b456d02e
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| c56f3426099a3cf2d07ccff8886050c7fbce140f |
|
22-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Pass stereo gains as packed minifloat
This will allow (eventually) a greater dynamic range for gains.
However there are still a few remaining places in effects and mixer
that will also need to be changed in order to get the full benefit.
Also fixes a minor bug: was not checking for NaN in AudioTrack C++.
Change-Id: I63bce9e82e0a61546d8ff475fb94bcb700d99c96
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5859b98476de3bbb4ae4dc00d4b2f010ca7897d3 |
|
19-May-2014 |
Eric Laurent <elaurent@google.com> |
am be2664da: am 4c9bde7a: am ce473c32: Merge "Codec Offload: Add NULL check as mCallbackThread is NULL when offloaded HAL is running BLOCKING mode"
* commit 'be2664dac34a6a9d9f754d4097b32158d4947cd2':
Codec Offload: Add NULL check as mCallbackThread is NULL when offloaded HAL is running BLOCKING mode
|
| 4c9bde7a3e636d3ecccd7905d142dbbb80e977a0 |
|
19-May-2014 |
Eric Laurent <elaurent@google.com> |
am ce473c32: Merge "Codec Offload: Add NULL check as mCallbackThread is NULL when offloaded HAL is running BLOCKING mode"
* commit 'ce473c32705e2d1c5fcc817198101e1220b3c8f7':
Codec Offload: Add NULL check as mCallbackThread is NULL when offloaded HAL is running BLOCKING mode
|
| 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/services/audioflinger/Threads.cpp
|
| deb9081a61b446dbe50461082974a4228899f178 |
|
14-May-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Pull up mFlags and isFastTrack from Track to TrackBase"
|
| 56604aa3a56dc8e15532597a0a74b3c7b165e006 |
|
13-May-2014 |
Uday Gupta <guptau@broadcom.com> |
Codec Offload: Add NULL check as mCallbackThread is NULL
when offloaded HAL is running BLOCKING mode
b/14897952
Change-Id: I159a33e1181c1a62d80ab4bd3337a35dafaad884
Signed-off-by: Uday Gupta <guptau@broadcom.com>
Signed-off-by: Pierre Couillaud <pierre@broadcom.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| 755b0a611f539dfa49e88aac592a938427c7e1b8 |
|
13-May-2014 |
Glenn Kasten <gkasten@google.com> |
Pull up mFlags and isFastTrack from Track to TrackBase
Change-Id: I6bd48292310be4e05758a11f617f50585904422d
/frameworks/av/services/audioflinger/Threads.cpp
|
| 021cf9634ab09c0753a40b7c9ef4ba603be5c3da |
|
13-May-2014 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: add specific mutex for client lists
Add a specific mutex to protect access to mClients and
mNotificationClients lists. This avoids locking the main AudioFlinger
mutex from inside thread loops and allows not to worry about
cross deadlocks when sending a config event with status reply while
keeping the ThreadBase or AudioFlinger mutex locked.
As a way of consequence, remove notification client list passed to
processConfigEvents_l() and audioConfigChanged() as the list
can now be accessed by locking client mutex only.
Change-Id: I228022204b6709a8bb60cc96d9514a6ffe59b62e
/frameworks/av/services/audioflinger/Threads.cpp
|
| 110135b9610fee8bfa5bac2d9ef4fc4c24991c36 |
|
09-May-2014 |
Glenn Kasten <gkasten@google.com> |
Explicitly ignore return value of __futex_syscall3
Change-Id: Ic65f257bfc4a0eff7bdd3c44fab68a3f17e2bd08
/frameworks/av/services/audioflinger/Threads.cpp
|
| 26d5ff926fa3323b39ae4408bcd29826a9523c9b |
|
10-May-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: refactor thread config events"
|
| 1035194cee4fbd57e35ea15c56e66cd09b63d56e |
|
09-May-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: refactor thread config events
Merge config events and set parameters messaging mechanism.
- setting parameters now uses the config event mechanism
- config event now allow to wait for a condition and synchronize caller
binder thread with execution thread and return an execution status.
- simplify locking mechanism to avoid unlocking the thread
mutex while processing events.
Change-Id: Ia49cb3e617abec4bacb6c1f9a8cb304c4ed3902e
/frameworks/av/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| b880f5e5fc07397ddd09a94ba18bdf4fa62aae00 |
|
07-May-2014 |
Glenn Kasten <gkasten@google.com> |
Add a per-thread heap that is read-only to clients
Change-Id: I401263566ca20fbfb565689c8fa99458d3b283b2
/frameworks/av/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| 993fa0603707e94ce259e95e56838a85b5ccbdc5 |
|
02-May-2014 |
Glenn Kasten <gkasten@google.com> |
Use audio_is_linear_pcm()
Change-Id: I17b30fdb6dbb6454ad0a20dd703fd603a37a8397
/frameworks/av/services/audioflinger/Threads.cpp
|
| c125f38cd0ae35409a01b98a99e483550daa1313 |
|
12-Apr-2014 |
Glenn Kasten <gkasten@google.com> |
audioflinger: forward device change to effects in direct thread
Device change during offload playback is not informed to effect chain
which causes effects still work on unexpected output devices. Add device
change notification in direct output thread.
Original author wjiang <wjiang@codeaurora.org>
CRs-Fixed: 630408
Bug: 14053172
Signed-off-by: Glenn Kasten <gkasten@google.com>
Change-Id: I094a99bdf540479cee2fca6614ec35c2fa7d6046
/frameworks/av/services/audioflinger/Threads.cpp
|
| 84a0c6e87c48f58a0d3be71961432c086a4d24cc |
|
02-Apr-2014 |
Andy Hung <hunga@google.com> |
Change references of Q19.12 to Q4.27 for clarity
Change-Id: I5beb7daf6ff9bc123ff3582f7c294edcaf8652f6
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| 3a6c90aa0617666d9abc94c02b752d9eb3d64772 |
|
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Add FIXMEs to createRecord for fast capture
Change-Id: I759be200fae32969212c52a409f46f2e704081e3
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5c4fa907328c8ee586367272c7525cb87742f14d |
|
26-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use LOG_ALWAYS_FATAL instead of LOG_FATAL"
|
| adad3d7d935da176ff24941b4ae9edf7340e9b96 |
|
21-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Use LOG_ALWAYS_FATAL instead of LOG_FATAL
LOG_FATAL is compiled out in most builds, so the
assertion checks were not being performed.
Change-Id: I774f0985ab9c5ccecd8989a0f1c940386b73fc35
/frameworks/av/services/audioflinger/Threads.cpp
|
| 66e4635cb09fadcaccf912f37c387396c428378a |
|
17-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Use symbolic constants from <system/audio.h>
AUDIO_INTERLEAVE_*
AUDIO_STREAM_MIN
AUDIO_SESSION_ALLOCATE
Change-Id: I31dd6f327204685e50716079ce21c4ba206dff11
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2d3ca68363f723fbe269d3ce52dab4985dfc7154 |
|
07-Mar-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: OffloadThread fix for resume underrun
AudioFlinger can miss resuming h/w on a pause->resume transition
if sufficient data isn't available
Bug: 11358524.
Change-Id: Ic3c75256290d3515fd4a96dfcc900909fbe5bc15
/frameworks/av/services/audioflinger/Threads.cpp
|
| 8d6c292a0bed3d63b5b7297d09a604af6327c663 |
|
05-Feb-2014 |
Krishnankutty Kolathappilly <kkolat@codeaurora.org> |
audioflinger: Handle pause /resume in stopping state
-Pre-requisite:
Perform seek on the clip. After seek the data remaining till EOS
is little more than the driver and common block buffering.
-Framework state:
Offload thread is waiting for signal from the HAL for a free
buffer. Audio Player calls sink stop on reaching EOS. Audio
track is waiting on obtain buffer for a free space in common
block to send the last buffer. The track is moved to stopping
state as input EOS is reached.
-Issue:
Perform pause/ resume in this state(STOPPING), Audio playback
does not resume.
-Fix
Ensure resume is called in stopping state if frames ready is
greater than zero.
Bug: 12870871
Change-Id: Ib1378c4ee5ce4bea655691e93de0775f7b1d2804
Signed-off-by: Glenn Kasten <gkasten@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| 15e5798908ccac14e10c84834eaf08c42931bd06 |
|
24-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Make error handling for audio driver initCheck more similar
Change-Id: I8222f00f7ac89d52235af0f64f3fee7a8c471e1e
/frameworks/av/services/audioflinger/Threads.cpp
|
| c3df838434b37d8400eea2438083cc01a4c1cc71 |
|
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Replace if/else by switch
Change-Id: Ibe96c8fb1f0ba7f850c0561f60769ce63aceba5d
/frameworks/av/services/audioflinger/Threads.cpp
|
| 05997e21af6c4517f375def6563af4b9ebe95f39 |
|
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Whitespace and comments
Change-Id: I99752cca74a459c4746ea4718acb0115439e50bd
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5b10a2037a835e790994b9ebec3c2e55052f1f3b |
|
13-Mar-2014 |
Andy Hung <hunga@google.com> |
Fix Offload playback in AudioFlinger
This fixes Offload playback regression introduced by adding
flexible format to mSinkBuffer. Test case is AAC file playback.
Bug: 13450717
Change-Id: I0fa11978295ed4793be90c565e5b8abedf156914
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| 010a1a1a552cdaad362cea8a0333b8906402dbcb |
|
13-Mar-2014 |
Andy Hung <hunga@google.com> |
Revert "Revert "Convert AudioFlinger mSinkBuffer to flexible format""
This reverts commit e2a9c29f35e0c09782558542fc4cf9823779590e.
/frameworks/av/services/audioflinger/Threads.cpp
|
| e2a9c29f35e0c09782558542fc4cf9823779590e |
|
13-Mar-2014 |
Eric Laurent <elaurent@google.com> |
Revert "Convert AudioFlinger mSinkBuffer to flexible format"
This reverts commit e7e676fd2866fa4898712c4effa9e624e969c182.
Bug: 13450717.
Change-Id: Ib80b0d14428fecce33c62003a1fcf83f71cee03b
/frameworks/av/services/audioflinger/Threads.cpp
|
| e7e676fd2866fa4898712c4effa9e624e969c182 |
|
05-Mar-2014 |
Andy Hung <hunga@google.com> |
Convert AudioFlinger mSinkBuffer to flexible format
Change-Id: I618d9c99a5f6f8c8d6a9f4b2d19e82c9ddc3b06e
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| 98ef978df4e928f486d244c4d7f7ad9f13111e98 |
|
04-Mar-2014 |
Andy Hung <hunga@google.com> |
Add mEffectBuffer to AudioFlinger
Change-Id: Icf97c50040bc127723d56eb4d2fb6e790a7253d9
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| f69f9869514730aebe5724c461768507084dfff7 |
|
07-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
NBAIO::Format_from_SR_C requires audio_format_t parameter
Change-Id: I4b65f6ed2f6ca3608b3a5f88f52a93af0b9b1f4a
/frameworks/av/services/audioflinger/Threads.cpp
|
| e857b65c1d3aa055281cb48f59c9b5eb4a062dd0 |
|
07-Mar-2014 |
Narayan Kamath <narayan@google.com> |
am 3c9130a8: Merge "Make frameworks/av 64-bit compatible"
* commit '3c9130a8a12ce962dc8d22e04243b135b9e98b01':
Make frameworks/av 64-bit compatible
|
| 10ac79930120efea20b125f5faedfe89e8674676 |
|
07-Mar-2014 |
Glenn Kasten <gkasten@android.com> |
am 5930e8eb: Merge "AudioFlinger: Remove code for supporting resampling in fast tracks"
* commit '5930e8ebdacf3a59c13eef11fd03658ba58b443e':
AudioFlinger: Remove code for supporting resampling in fast tracks
|
| 32fbb48de0248518fc5a14099b32fe4b8e9fc102 |
|
07-Mar-2014 |
Glenn Kasten <gkasten@android.com> |
am 4361749d: Merge "AudioTrack: Never try to use the fast path if resampling is required"
* commit '4361749d74113069aafb0620a1189404205c58d0':
AudioTrack: Never try to use the fast path if resampling is required
|
| 55fbfbbbf00722dac3f789be69138445b771a1e1 |
|
06-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use larger capture pipe since we support resampling by 6:1"
|
| 6cbccee701e74fa43a5ea49c15af7dd3267b6699 |
|
14-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: Fix for a deadlock in track creation
AudioFlinger enters a deadlock (with itself) on trying to free a
RecordTrack or Track object that failed initialization. Clear this
bad object from the caller instead.
Bug: 12423233
Change-Id: I926f2beb922a70f6924e593e2bbf1a5b5df85b16
/frameworks/av/services/audioflinger/Threads.cpp
|
| 4f4c9a22f4676c65bcae77f0805dcd699b3ba475 |
|
06-Mar-2014 |
Eric Laurent <elaurent@google.com> |
am 281dd4e1: audioflinger: fix static track end detection
* commit '281dd4e13309973dbb85bce531f884237e0d8fb0':
audioflinger: fix static track end detection
|
| 7028a2c8106ee5a22db88da5dad581820ee07254 |
|
05-Mar-2014 |
Eric Laurent <elaurent@google.com> |
am 7e92abea: audioflinger: fix offload write buffer offset
* commit '7e92abeafb184e8a34213d7149592e95a72601b0':
audioflinger: fix offload write buffer offset
|
| 281dd4e13309973dbb85bce531f884237e0d8fb0 |
|
21-Dec-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix static track end detection
If a static track is not a fast track,
prepareTracks_l() must rely on framesReady() to
detect end of buffer and remove the track from the active
track list.
Failing to do so results in the track staying active but
not processed by the mixer because in underrun. This leaves the
mix buffer content uninitialized and causes the effect process
function to accumulate its output onto undefined data.
Bug: 12013676.
Change-Id: I4b0819a9d93141ac3307b8786fc6a451dd585220
/frameworks/av/services/audioflinger/Threads.cpp
|
| 7e92abeafb184e8a34213d7149592e95a72601b0 |
|
22-Nov-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix offload write buffer offset
Fix current audio HAL write buffer offset calculation
which assumes that the frame size is a multiple of 2.
'
Bug: 12823725.
Change-Id: I0195ed5cfef225a6f114e7dd405a02680bb7254e
/frameworks/av/services/audioflinger/Threads.cpp
|
| 21e81bca4d1e6f7d1ccac61b65da7c7a4a121064 |
|
05-Mar-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
am e010f65e: audioflinger: Fix for a deadlock in track creation
* commit 'e010f65e6337267cb15f8894c950a3f64370dd36':
audioflinger: Fix for a deadlock in track creation
|
| 50c3157c5a3e0617be77716beff1ae8801d8a72f |
|
04-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: check for condition before waiting
AsyncCallbackThread must check for any condition that
was already been satisfied before waiting.
Bug: 11824817
Change-Id: I04683a1f355de4f440106cab47fd916aa39d5e35
/frameworks/av/services/audioflinger/Threads.cpp
|
| e010f65e6337267cb15f8894c950a3f64370dd36 |
|
14-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: Fix for a deadlock in track creation
AudioFlinger enters a deadlock (with itself) on trying to free a
RecordTrack or Track object that failed initialization. Clear this
bad object from the caller instead.
Bug: 12423233
Change-Id: I926f2beb922a70f6924e593e2bbf1a5b5df85b16
/frameworks/av/services/audioflinger/Threads.cpp
|
| 788207057ed4b8df4719ed8089f376ef52de9ca1 |
|
01-Mar-2014 |
Andy Hung <hunga@google.com> |
Rename mSinkFormat to mMixerFormat for AudioMixer::track_t
AudioMixer::SINK_FORMAT also changes to AudioMixer::MIXER_FORMAT
Change-Id: Ic3f8be77d2c75c082c4fd140bc907e30c304d285
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| e842614837e5401adf77e90485300c288b9a7876 |
|
01-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Use larger capture pipe since we support resampling by 6:1
This avoids overruns on the client AudioRecord side,
without requiring client to use a large buffer.
It should not increase input latency, since a newly started AudioRecord
always joins the stream starting at the latest data.
Change-Id: Ib2b8de75cc40a6a3d493a1f8b46b41220f69264f
/frameworks/av/services/audioflinger/Threads.cpp
|
| 69aed5f0f4a3be3996d1e78a0473e1a72c1547da |
|
26-Feb-2014 |
Andy Hung <hunga@google.com> |
Add MixerBuffer for accumulation of float audio
Path into and out of MixerBuffer is set up, but not
currently enabled.
Change-Id: I9d50752607d22dd2a3d9cc7e053babf8dfb22958
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| 25c2dac12114699e90deb1c579cadebce7b91a97 |
|
27-Feb-2014 |
Andy Hung <hunga@google.com> |
Rename mixBufferSize to mSinkBufferSize in AudioFlinger
Change-Id: I4d1d18d716fa3ad8c0e7cc87f5a2823422f504eb
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| 09a5007b17acb49d25cfa386a2e2534d942e8854 |
|
27-Feb-2014 |
Andy Hung <hunga@google.com> |
Rename constants for MixBufferSize to SinkBufferSize
Updated variable names are
kMinNormalSinkBufferSizeMs
kMaxNormalSinkBufferSizeMs
Change-Id: I91969eae3f8c10275db8778634cbf05fdec0712b
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2098f2744cedf2dc3fa36f608aa965a34602e7c0 |
|
27-Feb-2014 |
Andy Hung <hunga@google.com> |
Rename mMixBuffer to mSinkBuffer in AudioFlinger
Change-Id: I1f55630425b8ce36e9df2aaf64e06a8960b69a47
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| 323da1015a758243c2c43017b026a01c6cf0c6f2 |
|
22-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Account for unreleased frames when predicting resampler needs"
|
| 607fa3e928de696eba49f198af72d68e4591ca1b |
|
21-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Account for unreleased frames when predicting resampler needs
Other changes related to resampling:
- improve logs
- use correct form of assert
- don't round up max usable circular buffer sizes to power of 2;
this gives more predictable results
- add some FIXMEs
Bug: 13133176
Change-Id: I8d3c90eed4eb0c0c1fbac3c864e179a2806c2301
/frameworks/av/services/audioflinger/Threads.cpp
|
| 8ea16e4b0a7d398d26887c18675b3899de5d779d |
|
21-Feb-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix race condition in SyncEvent callback
Now that the SyncEvent callback is implemented by the
RecordTrack instead of the RecordThread, there is a possibility
that the callback is called after the track deletion.
SyncEvent callback now uses a weak pointer instead of
a raw pointer as cookie. This allows the callback implementer to
acquire a strong reference on the object pointed to by the cookie.
Bug: 13114128.
Change-Id: Id61b8f06044ed1e52c6f7e7c666cdede68340de2
/frameworks/av/services/audioflinger/Threads.cpp
|
| 4cc0a6a835c806d200ef83ef31fe5bef327c355c |
|
17-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Multi-client recording continued
Remove mReqChannelCount and mReqSampleRate.
Increase number of input frames needed for sample rate converter.
Add TODO comments to investigate later.
Fix bugs in sample rate conversion.
Change-Id: I3aa54b862843d518b6e4ece9a7f52fd27d741629
/frameworks/av/services/audioflinger/Threads.cpp
|
| 25f4aa83efaa9179e65a20583a6d441de2c3ff3f |
|
07-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Move handleSyncStartEvent and clearSyncStartEvent to RecordTrack
Change-Id: I611a63e63acf3eb17e2072c1a77b91fe9e62e36a
/frameworks/av/services/audioflinger/Threads.cpp
|
| deca2ae0a7cf8bc54ff3f30b7dc39bbc78b94c0d |
|
07-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Add _l suffix to readInputParameters/OutputParameters
They are always called while thread mutex is locked,
or in the thread constructor.
Change-Id: I56df71563a52a5cd6cea6fc0914df993519e1b74
/frameworks/av/services/audioflinger/Threads.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/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| 4b76d27d6c4751b31a1cb8ac5e6da1d4b7724a7b |
|
10-Feb-2014 |
Glenn Kasten <gkasten@android.com> |
am 5a458ede: am 5930e8eb: Merge "AudioFlinger: Remove code for supporting resampling in fast tracks"
* commit '5a458ede3b7d303723fde4b856532bdc3b94971b':
AudioFlinger: Remove code for supporting resampling in fast tracks
|
| 3523e8c40bc60af0c95d1aa71a51a13d69ec80a2 |
|
10-Feb-2014 |
Glenn Kasten <gkasten@android.com> |
am 4e12dade: am 4361749d: Merge "AudioTrack: Never try to use the fast path if resampling is required"
* commit '4e12dadeec10993d2f77cce96600e73ca0dd86e2':
AudioTrack: Never try to use the fast path if resampling is required
|
| 9601c6efcb2552960d6f125d073525b581c1b7ec |
|
10-Feb-2014 |
Marco Nelissen <marcone@google.com> |
Merge "Pretty up audioflinger dumpsys"
|
| ec95f0e82a680931e9c7c9011dbbd68d5ab484c8 |
|
10-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use printf format %#x for audio_format_t in logs"
|
| b220884bf3129253cc5bc8d030bc475411ea4911 |
|
07-Feb-2014 |
Marco Nelissen <marcone@google.com> |
Pretty up audioflinger dumpsys
Change-Id: I57e44b4c36b99f7149542bbcf9645521c6152dfa
/frameworks/av/services/audioflinger/Threads.cpp
|
| 4944acb7355b3aa25748fd25945a363a69d65444 |
|
19-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Fix whitespace
Change-Id: I9add0872030a26933f6b6dad1be299154611cc56
/frameworks/av/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| 4c6a433d74d5ae8b9bc0557207e3ced43bf34a25 |
|
15-Jan-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
AudioFlinger: Offload track transition fix
Trigger state changes in OffloadThread on adding a new offload track.
Bug: 12530661
Change-Id: I95b62764c7ac041d470702e86b951f812eb0e70f
/frameworks/av/services/audioflinger/Threads.cpp
|
| 7844f679be8d94c5cdf017f53754cb68ee2f00da |
|
15-Jan-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
AudioFlinger: Modify flush handling for offload path
Do not allow an offload track to directly control the offload thread
behavior. OffloadThread can check for any pending flush reporting
by its active tracks and decide to flush the HW or not.
Bug: 12530661
Change-Id: Ib33f023c942f6c091b618004136b153c38a6eef6
/frameworks/av/services/audioflinger/Threads.cpp
|
| 0f02f265123b7ef2fd6ac09ff70cde26eb5559ad |
|
11-Jan-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: OffloadThread state fix
When switching from one offload session to another that
reuses the same offload output, audio HAL is paused & flushed in
preparation for the new track
OffloadThread must reset mHwPaused to allow proper state transitions
for the second track. In addition, Offload thread need not
explicitly call resume on HAL while handling a flush.
Bug: 12530661
Change-Id: I8173845e1782b5a8b372a82c7d337299a61e0efe
/frameworks/av/services/audioflinger/Threads.cpp
|
| 3ce28aa3cb5262775180a8b423cfb4a5670ebc59 |
|
05-Feb-2014 |
Martin Storsjo <martin@martin.st> |
AudioFlinger: Remove code for supporting resampling in fast tracks
This isn't used at the moment.
Change-Id: I4e0fb2af5f7d959dbafd5ddb7defa1c6b8e8636a
/frameworks/av/services/audioflinger/Threads.cpp
|
| 7332c37c459f5f382a9e4105d522d1c5a7f6f0b5 |
|
05-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "audioflinger: Fix for a deadlock in track creation"
|
| 0f11b51a57bc9062c4fe8af73747319cedabc5d6 |
|
01-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Fix unused parameter warnings in audio
Change-Id: I665ba3358dd9502f0adec70d486e7bf8a2e1b0fe
/frameworks/av/services/audioflinger/Threads.cpp
|
| 398f21348e5100289f6e5be30c8b5257fa04aaf9 |
|
31-Jan-2014 |
Martin Storsjo <martin@martin.st> |
AudioTrack: Never try to use the fast path if resampling is required
Unless AudioFlinger was built with FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
enabled, AudioFlinger would deny using the fast path (and internally
fall back to the normal codepath) when it realized that resampling
was required. Since the buffer size calculations within AudioFlinger
don't take resampling into account properly (see the calculation
below "AUDIO_OUTPUT_FLAG_FAST denied" in audioflinger/Threads.cpp,
just below the hunk that this patch changes), make sure AudioTrack
doesn't try to use the fast path if resampling is required.
This removes the possibility to enable
FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE in AudioFlinger since it
AudioTrack now won't even try to use the fast path for content
that requires resampling, regardless of the AudioFlinger configuration.
Change-Id: Icf0f8ad50bf0fdb84657f518c0120aa0535f23f9
/frameworks/av/services/audioflinger/Threads.cpp
|
| 745e9a8283b56c7772ee7d72383a3f2e012e1ef9 |
|
21-Dec-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix static track end detection
If a static track is not a fast track,
prepareTracks_l() must rely on framesReady() to
detect end of buffer and remove the track from the active
track list.
Failing to do so results in the track staying active but
not processed by the mixer because in underrun. This leaves the
mix buffer content uninitialized and causes the effect process
function to accumulate its output onto undefined data.
Bug: 12013676.
Change-Id: Iad72c921fa18d34811abf7d1073890c093a27725
/frameworks/av/services/audioflinger/Threads.cpp
|
| 03e9e83c47ab4a518da0a1f36b8f702f59221c95 |
|
14-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: Fix for a deadlock in track creation
AudioFlinger enters a deadlock (with itself) on trying to free a
RecordTrack or Track object that failed initialization. Clear this
bad object from the caller instead.
Bug: 12423233
Change-Id: I926f2beb922a70f6924e593e2bbf1a5b5df85b16
/frameworks/av/services/audioflinger/Threads.cpp
|
| 000a4193dd82549192277fd4b9bb571d8a4c262f |
|
30-Jan-2014 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix record thread exit pending check
RecordThread loop must not release the mutex after checking for
exitPending and before waiting for a new wake up condition.
This can happen under the hood when methods like processConfigEvents_l()
or checkForNewParameters_l() are called. So exitPending() must
be checked after calling these functions.
Bug: 12787961.
Change-Id: Ia18c518bd5344fbb2401067303fcfe76a86879c4
/frameworks/av/services/audioflinger/Threads.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/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| 13e4c960ea3db03a43e084fbd85d52aa77f7b871 |
|
21-Dec-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix static track end detection
If a static track is not a fast track,
prepareTracks_l() must rely on framesReady() to
detect end of buffer and remove the track from the active
track list.
Failing to do so results in the track staying active but
not processed by the mixer because in underrun. This leaves the
mix buffer content uninitialized and causes the effect process
function to accumulate its output onto undefined data.
Bug: 12013676.
Change-Id: Iad72c921fa18d34811abf7d1073890c093a27725
/frameworks/av/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| c95616357a911e5c1c55873492b05ebfa3cf1894 |
|
04-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: check for condition before waiting
AsyncCallbackThread must check for any condition that
has already been satisfied before waiting.
Bug: 11824817
Change-Id: Ic8c2090d521ecd6a30b76ee75635258d35eb1eff
/frameworks/av/services/audioflinger/Threads.cpp
|
| 688ca6330a5e7d072f67a1791e464fa7dec08b72 |
|
17-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
am 5c474238: am 5893736d: am a67c7653: Merge "Increase kFastTrackMultiplier from 1 to 2"
* commit '5c474238bc579224ec8f7267d05651b0aa934100':
Increase kFastTrackMultiplier from 1 to 2
|
| a67c76538fff102cd44b2f38dd073df5f2a33605 |
|
17-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Increase kFastTrackMultiplier from 1 to 2"
|
| 24a325d6f8c4bbf9330e6ce0c769d46e04266ffc |
|
04-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: check for condition before waiting
AsyncCallbackThread must check for any condition that
was already been satisfied before waiting.
Bug: 11824817
Change-Id: Ic8c2090d521ecd6a30b76ee75635258d35eb1eff
/frameworks/av/services/audioflinger/Threads.cpp
|
| 70aca464f7e39a3b208341f442b6fc2316e17147 |
|
05-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Add RecordThread media.log and deferred deallocation"
|
| 0312b69e65e4c8bf6a0c0757d9100dee3651c16e |
|
05-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Change mActiveTrack to mActiveTracks vector"
|
| 481fb67a595f23c5b7f5be84b06db9b84a41a42f |
|
30-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Add RecordThread media.log and deferred deallocation
This change allows a media.log buffer for RecordThread.
Unlike playback threads which stick around forever, the RecordThread comes
and goes for every capture session. This means that the media.log buffer
for a RecordThread would disappear too, and so was useless. Now when a
thread exits, it's associated media.log buffer is just marked for deferred
deallocation. It is only actually freed when the memory is needed.
Other changes:
- Fix bug in unregistering comparison, it was comparing the wrong pointers
- Increased size of log area so we can log for RecordThread also
Change-Id: If45d4c03a793b86390a0112ec3acc5d41b2e3635
/frameworks/av/services/audioflinger/Threads.cpp
|
| b5fed68bcdd6f44424c9e4d12bfe9a3ff51bd62e |
|
03-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Increase kFastTrackMultiplier from 1 to 2
Change-Id: I158f147295eebcea96e4047d7618069bc48bdd7d
/frameworks/av/services/audioflinger/Threads.cpp
|
| 866d90595a177f6027e97161781948baebf2aa68 |
|
02-Dec-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Code simplification and movement to prepare for mActiveTracks"
|
| 480acec40aa2d96119530c7d451282af8cf07327 |
|
23-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix bug in creation of EffectHandle when out of memory"
|
| 04733dbdede313fc5e291070014c1a0641fe6166 |
|
22-Nov-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix offload write buffer offset
Fix current audio HAL write buffer offset calculation
which assumes that the frame size is a multiple of 2.
Change-Id: I0195ed5cfef225a6f114e7dd405a02680bb7254e
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2b806406ac1ec680b6fe3aaa84c54bdc4e43ad8d |
|
21-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Change mActiveTrack to mActiveTracks vector
The size of mActiveTracks is still limited to 1 for now.
Change-Id: Ia6b06b25d8c4709f087110ae29c589999407a2a2
/frameworks/av/services/audioflinger/Threads.cpp
|
| f10ffec18f930d92e1abe9200d60e746831841a7 |
|
21-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Code simplification and movement to prepare for mActiveTracks
Share code to acquire wake lock at entry to loop
Change-Id: I01fb645de9417055c0adfc6a4c4011bf22ea4f94
/frameworks/av/services/audioflinger/Threads.cpp
|
| e75da4004b2c814987aa2adf8a76190f92d99c65 |
|
20-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Fix bug in creation of EffectHandle when out of memory
If there is insufficient client memory to create the EffectHandle,
it was returning a handle which would be useless. Now it correctly
reports the out-of-memory error back to client.
Change-Id: I894e65d5d17df39383032c1527be6ccd35f578bb
/frameworks/av/services/audioflinger/Threads.cpp
|
| 731076cc7857fe322eed018569a2a67f6eac0e54 |
|
15-Nov-2013 |
Eric Laurent <elaurent@google.com> |
am 06d62548: am d7e59228: audioflinger: do not use raw pointer for tracks
* commit '06d6254823b8f7a46690369e57b5f149c94c9f65':
audioflinger: do not use raw pointer for tracks
|
| d7e59228caad3867794d847f6bf163c6495e9506 |
|
15-Nov-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: do not use raw pointer for tracks
Commit 9da3d95 surfaced a problem caused by the use of a raw
pointer to a track in offload thread implementation.
Pointers to tracks should always be weak or strong pointers.
Bug: 11708529.
Change-Id: Ic48632532d186c9be8261f73cefdf824b9fbbd2b
/frameworks/av/services/audioflinger/Threads.cpp
|
| 561879df42b65f44b83c391d78d351e9ad4cb162 |
|
14-Nov-2013 |
Eric Laurent <elaurent@google.com> |
am 307aa870: am 4215e616: Merge "audioflinger: fix offload track transition" into klp-dev
* commit '307aa87038017dff8ae1dd0262c0ff87723894d7':
audioflinger: fix offload track transition
|
| b04b25321dd323142eec7c5136c81dea3629cf14 |
|
14-Nov-2013 |
Eric Laurent <elaurent@google.com> |
am b50fa77b: am 9f357f31: Merge "audioflinger: fix offload resume after drain" into klp-dev
* commit 'b50fa77b2cc7dadae1b8ebb591d42976879ac43c':
audioflinger: fix offload resume after drain
|
| 9da3d9573a18ffe08365557c706cf52f09118d1c |
|
13-Nov-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix offload track transition
Make sure that when transitioning from one active
offloaded track to the next we flush DSP content
if both tracks are not on the same audio session.
This happens when switching between two applications
playing the same type of content (e.g MP3, stereo, 44.1).
In this case, we reuse the same output thread because the track
formats are compatible.
Bug: 11247103.
Change-Id: I2b9031591149adeb70766da5e0d21ff2933a37e8
/frameworks/av/services/audioflinger/Threads.cpp
|
| 1b9f9b134e732a48198e51f16424f330cbf03143 |
|
13-Nov-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix offload resume after drain
When pausing and resuming during the drain,
the drain sequence number shuold not be modified
otherwise the drain callback will be ignored.
This causes failure to notify end of stream to audio
player and transition to next song.
Bug: 11247103.
Change-Id: I2a35c5cc3fd6aa667cdd257f9e9cc8715cef5159
/frameworks/av/services/audioflinger/Threads.cpp
|
| 579dd27d96497022e534e859c6ebbec675ee07aa |
|
08-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Fix build
Change-Id: I1ecccada22047d9c828152a53ad68b5599ff8827
/frameworks/av/services/audioflinger/Threads.cpp
|
| 8594843c15b4722ced39436fe9e64f3e57e7ace4 |
|
19-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Pull HAL reads out of RecordThread resample
Previously, the resample was indirectly calling HAL read
from within the BufferProvider's getNextBuffer. Now the HAL
read is done ahead of time into a circular buffer,
and the getNextBuffer just reads out of that memory.
This change will permit some future planned features.
Also removed the mono to mono resampling optimization,
as that optimization will no longer be valid in the future.
Change-Id: I59f5016a2df91078d697c37a7dec2d9d9f44542e
/frameworks/av/services/audioflinger/Threads.cpp
|
| d8a1de6204829b0379bbd13042f1450bdb203f2b |
|
08-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Channel counts are uint32_t"
|
| f551e9961a6f66d63bb57f288c4be01b00725289 |
|
20-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Channel counts are uint32_t
Change-Id: I5676b6a291b72d7db6326f7eaf6d17588de51b05
/frameworks/av/services/audioflinger/Threads.cpp
|
| 09faf14b434bec0d184fecec947e3d750bcc64ff |
|
07-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Rename standby methods"
|
| 0699ed950b307663e0f7e01aecdfb505e62df7d6 |
|
06-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Move local declarations to point of first use"
|
| 44f412a81dcca8f7149d483eeeee9057ad0101cf |
|
06-Nov-2013 |
Eric Laurent <elaurent@google.com> |
am 44dc9155: am 48b6cd98: Merge "audioflinger: fix direct output underrun" into klp-dev
* commit '44dc91554521eaab478674a7f812cb8a4fa4f809':
audioflinger: fix direct output underrun
|
| 7bb88779cdc1a1fe1445d6d1691fb7daac932e74 |
|
06-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Rename TrackPaused to trackPaused"
|
| 1b3ec0f33cbdcac5e148c23abc5bc4f68d780ed4 |
|
06-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Use const more places"
|
| a23f17ac334ff20a11ee63dd177cb1080e44c483 |
|
06-Nov-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix direct output underrun
Underruns on tracks handled by a direct or offloaded
threads were not properly reported to the applications
causing a failure to automatically recover.
Bug: 11535001.
Change-Id: I7a8696b4d646f78b16710addfe7c108d0dd0038d
/frameworks/av/services/audioflinger/Threads.cpp
|
| 93e471f620454f7de73d190521568b1e25879767 |
|
19-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Rename standby methods
ThreadBase::standby() returns the value of mStandby.
RecordThread::standby() "overrode" ThreadBase::standby(), but with a
different meaning and no return value. This meant you couldn't
access the parent's class definition within RecordThread.
This could be confusing, so this CL renames RecordThread::standby()
to standbyIfNotAlreadyInStandby().
Change-Id: I9196ee12701dd2dccdfaed922af2d7b8e8774676
/frameworks/av/services/audioflinger/Threads.cpp
|
| b91aa63779ad1ce1846fbc604f9ec935e0c67852 |
|
19-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Move local declarations to point of first use
Change-Id: I8a078386e371caab5174fc07b47f48fb9f21609a
/frameworks/av/services/audioflinger/Threads.cpp
|
| e6f35b13567d920af19f0b8dc35f9120b7cf1dc9 |
|
19-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Rename TrackPaused to trackPaused
Change-Id: I3b38e0c9a6bbb6e18cb5b1f72670139a3071cb8f
/frameworks/av/services/audioflinger/Threads.cpp
|
| b2737d0b33c17e408d96d6f9eeaa3381479c94c7 |
|
19-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Use const more places
Change-Id: Ibc068d319d6fff26f2d11248e17481d8f7f027e0
/frameworks/av/services/audioflinger/Threads.cpp
|
| 1c7258b3c34b5656de92d845f6366a0f3ac1e0c6 |
|
31-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am 4cf799fb: am 5c7ffaed: Merge "AudioFlinger: fix duplicating thread standby." into klp-dev
* commit '4cf799fbbb74de102d34ccb014ba7fbd9bfa5e14':
AudioFlinger: fix duplicating thread standby.
|
| 2c3740f01acca69c3e0bcc5e01bb0edc51b6777f |
|
31-Oct-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix duplicating thread standby.
Commit fd47797 fixing issue 11247103 introduced a regression
where the duplicating thread does not exit standby state as
expected. This in turn prevents the tracks sending audio to
other threads to complete their activity cycle normally.
The fix conists in clearing mStandby state also in the specialized
threadLoop_write() method of DuplicatingThread subclass.
Bug: 11451415.
Change-Id: Ie8a92c819a56a2834ea25fa70a8b7b2125721775
/frameworks/av/services/audioflinger/Threads.cpp
|
| d21952fe5bc94f7bb6ee729f806c0e641fdb1a0d |
|
29-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am 7f8f7585: am fd477974: AudioFlinger: more fixes for offload audio lost
* commit '7f8f758506a95d6fe5c0e52aa7b5ef82fb18861a':
AudioFlinger: more fixes for offload audio lost
|
| fd4779740ec3e9e865d5514464df26d015354388 |
|
26-Oct-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: more fixes for offload audio lost
mStandby being false was not a reliable indication that
something had been written to audio HAL.
Considering the last track in mActiveTracks vector as the
latest activated track was wrong as this is a SortedVector.
Bug: 11247103.
Change-Id: I397d011c5afcdd779def3fe25aaae7669a472cfc
/frameworks/av/services/audioflinger/Threads.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/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| 6a7327f110b86613e6e8c7de105c49803acacc1d |
|
18-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am 6edd8ca5: am 6a51d7ed: audioflinger: fix track terminated before playing
* commit '6edd8ca5b7e3d169db4314d2b994bc9d48e9e082':
audioflinger: fix track terminated before playing
|
| 6a51d7ed7062536ccc892c8850a34ed55cbc8d5c |
|
18-Oct-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix track terminated before playing
When a track is terminated before any audio was written,
we should not send a drain request to the HAL. The drain
will be ignored and we will wait the async callback for ever.
Also fix a potential deadlock by not deleting a track with the
thread mutex held: mPreviousTrack changed from a sp to a raw pointer.
Bug: 11247103.
Change-Id: I0a4b47b099204c7c326d45d700234ab8f72d8eae
/frameworks/av/services/audioflinger/Threads.cpp
|
| 3278eed31fcf82045ac00dcd549dda5270c88971 |
|
15-Oct-2013 |
Narayan Kamath <narayan@google.com> |
am 8580679e: am 014e7fa2: Make Audio wakelock names less unique.
* commit '8580679e1780617d978d28a43e932314b86608ce':
Make Audio wakelock names less unique.
|
| 014e7fa2e90827d911c37bb0ce4d2e10e14d0bb3 |
|
14-Oct-2013 |
Narayan Kamath <narayan@google.com> |
Make Audio wakelock names less unique.
These wakelocks now show up in the client process, where they
might cause wakelock overflows if AudioFlinger threads die often.
The client process should be agnostic of audio flinger threading,
so don't make the wakelock names rely on audioflinger thread names.
Wakelock names now depend only on the audio flinger thread type, and
not the ID itself.
bug: 11200551
Change-Id: Ia1f71b569cb7502d617b77596abc245f461a6ec0
/frameworks/av/services/audioflinger/Threads.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
|
| 61f58c0c8d02970ea6d94ff816c54bf606f755b7 |
|
05-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am c22a1a49: am ea0fadeb: audioflinger: offload: fix pause/flush/resume
* commit 'c22a1a495c26b8a36fa15233cc136cb6ce6d6bea':
audioflinger: offload: fix pause/flush/resume
|
| ea0fadeb5d81ef3cb7f9db458c9033d628bdb86a |
|
05-Oct-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: offload: fix pause/flush/resume
If a pause/flush/resume sequence is fast enough, resume is received while
we are still in PAUSING state in which case it is a NOP. If this happens,
flush is still forwarded to the audio HAL but is not preceeded by a pause
which can cause problems to the audio DSP.
It is necessary to preserve the flush as this sequence is typical to a seek.
The fix consists in forcing a pause/resume when a flush request must be
executed and the audio HAL has not been paused previously.
Bug: 11081559.
Change-Id: Ib84ed26d503a61c05933b923ec556b10cedfe140
/frameworks/av/services/audioflinger/Threads.cpp
|
| 1c7feca9cc3e408dfeb632e7060cf31e5015227e |
|
04-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am 6c48a011: am 026c5b07: Merge "audioflinger: fix wake lock acquisition check" into klp-dev
* commit '6c48a01188746202164f8e10d63ff3775dfcf727':
audioflinger: fix wake lock acquisition check
|
| 547789d25dc6bd6561553bcf6b384fb0d4fee834 |
|
04-Oct-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix wake lock acquisition check
Commit e14a5d6 introduced a bug in ThreadBase::acquireWakeLock() where
the wake lock acquisition return code is not initialized before
being checked causing the wake lock token to be sometines not set
and potential failure to release the wake lock later.
Bug: 10985160.
Change-Id: Iffd40e773ae3bcfec3c148a844d5dbebbf474eaf
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5918e0cb64200f64632595f7aea6062c65d43353 |
|
04-Oct-2013 |
Marco Nelissen <marcone@google.com> |
am b5677882: am 8097068d: Merge "Better recording wake lock accounting" into klp-dev
* commit 'b5677882cf9a34bd3f2daf76d10951282ea9b083':
Better recording wake lock accounting
|
| 8097068de97c0cdab8150931f290edbd5b3721e5 |
|
03-Oct-2013 |
Marco Nelissen <marcone@google.com> |
Merge "Better recording wake lock accounting" into klp-dev
|
| e14a5d6d2cc91dd2fc09ffdf7aa670b37da0795d |
|
03-Oct-2013 |
Marco Nelissen <marcone@google.com> |
Better recording wake lock accounting
Count wake lock use against the app that initiated the recording,
not against the media server.
b/10985160
Change-Id: Iae6e6c030b7f1c6081d9c79725d6990f60dadaaa
/frameworks/av/services/audioflinger/Threads.cpp
|
| 34717c83733def81287e2b4ba2f62b416325c7ae |
|
02-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am 3424d6e1: am 1adf20ce: Merge "fix volume and effect enable delay on offloaded tracks" into klp-dev
* commit '3424d6e17637e0743ddf3bf4688af8ee36e69264':
fix volume and effect enable delay on offloaded tracks
|
| a6c7d98033c161592a062c61eaa6b7fd635e3006 |
|
30-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am 0adc67df: Merge "audioflinger: fix crash when starting offload thread" into klp-dev
* commit '0adc67dfc5cedb211c36f06849681a60a32f5805':
audioflinger: fix crash when starting offload thread
|
| 59fe010bcc072597852454a2ec53d7b0a2002a3b |
|
28-Sep-2013 |
Eric Laurent <elaurent@google.com> |
fix volume and effect enable delay on offloaded tracks
Volume: add a method to wake up the mediaserver playback
thread when a volume command is received on an offloaded track.
Effects: call effect chain process on offloaded playback threads
asynchronously from writes to allow effect state updates while
waiting for async write callback.
Bug: 10796540.
Change-Id: Id2747ae88783575d1d7ffd6fc86fbd054ab2c739
/frameworks/av/services/audioflinger/Threads.cpp
|
| 4de95592980dba88a35b3dc8f3fd045588387a4f |
|
27-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix crash when starting offload thread
Passing a sp to parent thread to AsyncCallbackThread() constructor
causes a strong reference to be acquired on the OffloadThread inside
its constructor which causes an early launch of the thread loop
with unpredictable consequences.
Pass a wp to parent thread instead.
Also move the creation of the AsyncCallbackThread to
readOutputParameters() where mUseAsyncWrite is initialized which
makes more sense.
Also change the type of AsyncCallbackThread parent thread to PlaybackThread
instead of OffloadThread to allow a broder use of non blocking write which
in theory is not limited to audio offload use case.
Bug: 8174034.
Change-Id: I4b093b022030cd4f5eb8b8e477333e91098a6549
/frameworks/av/services/audioflinger/Threads.cpp
|
| e93cf2ca27ae6f4a81d4ef548bbf10a34db6d98f |
|
24-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Cleanup openRecord error handling
Bug: 10888816
Change-Id: I84897dd7d30b370640b54e928f230604b873cb68
/frameworks/av/services/audioflinger/Threads.cpp
|
| c0dab528e8c63abce2022a391dface95c3ef67e5 |
|
24-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am a57e2ab5: am 664539d2: audioflinger: fix ro.audio.silent in offload again
* commit 'a57e2ab58a17d3da35593a6c85f65e19249e9ebc':
audioflinger: fix ro.audio.silent in offload again
|
| 664539d25180ab8f77e0521533ea2821cf28985f |
|
24-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix ro.audio.silent in offload again
Commit 1abbdb4 was not working 100% of the times because
it is possible that the offload thread loop never sleeps
after being created in which case the property is never read.
The loop now reads the property once when starting.
Bug: 10899309.
Change-Id: I2e2ca332f2d7086e59d65f6010378c4d9618ba9e
/frameworks/av/services/audioflinger/Threads.cpp
|
| 0230a2a8a413076a138db4c4e1dea018104fd5e2 |
|
20-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am c6bed216: am a3b97ad2: Merge "audioflinger: implement getTimestamp() for offloaded tracks" into klp-dev
* commit 'c6bed216e3a759a855ac4b9b743fbaed2d5929c1':
audioflinger: implement getTimestamp() for offloaded tracks
|
| d2b80a1fb90cb4dc3f569e716af0279c1e1ea72d |
|
20-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am ce3e7cb0: am fbb2609f: Merge "audioflinger: fix lost offload thread resume event" into klp-dev
* commit 'ce3e7cb0bd09e7861d417b7d45c1e5921f10a93f':
audioflinger: fix lost offload thread resume event
|
| a3b97ad288e6808c54c6275eaac1bd53108d8f11 |
|
20-Sep-2013 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: implement getTimestamp() for offloaded tracks" into klp-dev
|
| accc147666bfd37fc8b4ef745f18a8c751555ec2 |
|
20-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: implement getTimestamp() for offloaded tracks
Bug: 9587132.
Change-Id: Ie9d5f4cca96306d08bc9a2dbd6edd8953096702d
/frameworks/av/services/audioflinger/Threads.cpp
|
| ede6c3b8b1147bc425f7b923882f559a513fe23b |
|
19-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix lost offload thread resume event
It was possible that a resume request signaled by addTrack_l()
while waiting for an async write callback is lost. This is because
mSignalPending was not set and waitingAsyncCallback_l() would pause the
thread loop before executing prepareTracks_l().
The fix consists in using signal_l() method to wake the thread
loop o that mSignalPending is set.
Also make sure that sleepTime is reset to 0 when resuming to make sure
that we write any remaining bytes to the HAL.
Bug: 10810347.
Change-Id: If9a3b22cc3b9e6eb384a56c48c40e6258d0896ad
/frameworks/av/services/audioflinger/Threads.cpp
|
| dc8cae8c118e4aef4ef1f7b2c6f79becc1df4a05 |
|
18-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am bf5e2397: am 5baf2af5: more support for audio effect offload
* commit 'bf5e23979a03da96ce1d63126c480103232f174b':
more support for audio effect offload
|
| 5baf2af52cd186633b7173196c1e4a4cd3435f22 |
|
13-Sep-2013 |
Eric Laurent <elaurent@google.com> |
more support for audio effect offload
Offloading of audio effects is now enabled for offloaded
output threads. If an effect not supporting offload is enabled,
the AudioTrack is invalidated so that it can be recreated in PCM
mode.
Fix some issues in effect proxy related to handling of effect
commands to offloaded and non offloaded effects.
Also fixed a bug on capture index in software Visualizer effect.
Bug: 8174034.
Change-Id: Ib23d3c2d5a652361b0aaec7faee09102f2b18fce
/frameworks/av/services/audioflinger/Threads.cpp
|
| 6d07d9bd2e0549ca2682ad9e94118feee1f733f6 |
|
14-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am 029838a9: am 7f91e5e0: Merge "audioflinger: fix ro.audio.silent in offload mode" into klp-dev
* commit '029838a98c9377299ef3341e21e9378fd5562ab6':
audioflinger: fix ro.audio.silent in offload mode
|
| d455cdfad40ca0558b8f4f800ec192027e272c14 |
|
14-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am c91eb605: am 972a173d: audioflinger: longer offload thread standby delay
* commit 'c91eb6059a6af8b84c48e0ce727b01829f9cd789':
audioflinger: longer offload thread standby delay
|
| 7f91e5e09cf3bd8068894b2e54cbdaac70bf53a0 |
|
14-Sep-2013 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: fix ro.audio.silent in offload mode" into klp-dev
|
| 1abbdb4429479975718421c4fef3f79ce7c820e3 |
|
14-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix ro.audio.silent in offload mode
Change-Id: I5a1a79000d53146689b0a198cc5419c36509703f
/frameworks/av/services/audioflinger/Threads.cpp
|
| 972a173d7d1de1a3b5a617aae3e2abb6e05ae02d |
|
04-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: longer offload thread standby delay
- Increase offloaded output thread standby delay to 1 second
to allow transition between tracks with going to stanby
if reusing the same audio track (gapless)
- Make sure pause/flush/resume sequence is sent to the HAL
in the right order
- Fix format display in track dump
Bug: 8174034.
Change-Id: I43ef6f8fdbf7427e4eff6cc2d0665d7d1463ea8a
/frameworks/av/services/audioflinger/Threads.cpp
|
| d5577f26de1ae3a0dc6fbea9c60a07d585f894bf |
|
12-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am 4823964d: am efc4d489: Merge "audioflinger: add HOTWORD audio source." into klp-dev
* commit '4823964d198228289ff21530bf8b01bb2ef196bb':
audioflinger: add HOTWORD audio source.
|
| 9a54bc27876acd5d8be5b1fc3dc46701fe76fbb3 |
|
09-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: add HOTWORD audio source.
HOTWORD is a special capture audio source that uses
the same audio tuning as VOICE_RECOGNITION but allows an
active capture to be preempted.
Bug: 10640877.
Change-Id: Iea09a11cbcdbacc8d434e5230e7559dcd1b44ac0
/frameworks/av/services/audioflinger/Threads.cpp
|
| bfc4214079875f0dc8c349e8c475d1813c234e67 |
|
06-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am 99b84e8b: am 3b4529e0: audioflinger: remove async write race conditions
* commit '99b84e8b013a6e0b926693067b00e82cfbe2ca4f':
audioflinger: remove async write race conditions
|
| 3b4529e03c5fc7a44c22f9091ad15a269bfca3a8 |
|
06-Sep-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: remove async write race conditions
Remove possible race conditions between async callback
thread and offload thread when clearing and setting the
draining and write blocked flags.
Bug: 8174034.
Change-Id: I7af10491f39dc0e7d7414862a9d8e763daa2e2b7
/frameworks/av/services/audioflinger/Threads.cpp
|
| a7fef85e7d419a4f5d6a3144f9ba70ceff2f122a |
|
04-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
am f81d5fc1: am 943ba3cc: Merge "Add NBAIO_Sink::getTimestamp()" into klp-dev
* commit 'f81d5fc1d67a5252d0da1a45274460c4dd33b87b':
Add NBAIO_Sink::getTimestamp()
|
| 767094dd98b01baf21de2ad09c27b3c98776cf73 |
|
23-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Add NBAIO_Sink::getTimestamp()
with a real implementation in AudioStreamOutSink
for dummy implementation initially in MonoPipe.
Use in AudioFlinger::PlaybackThread::threadLoop_write()
to keep the input to the timestamp latch up-to-date.
Change-Id: I10ef277991b63bb43d55d6f3df75116ef32246cd
/frameworks/av/services/audioflinger/Threads.cpp
|
| 0fd582e3ce5243c3e5a429fee3330aafc69b69fa |
|
31-Aug-2013 |
Eric Laurent <elaurent@google.com> |
am 5d8cbf41: am 6bf9ae20: audioflinger: offloaded tracks flush/pause order
* commit '5d8cbf41ee7e511d20a649e240d6aecbbac3fae7':
audioflinger: offloaded tracks flush/pause order
|
| 6bf9ae20b3bd2dbb8f2e89ee167a6785222301cf |
|
31-Aug-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: offloaded tracks flush/pause order
Make sure that a flush received after a pause is forwarded
to the HAL after the pause.
Change-Id: Ib3221f70c59f8b3dbbac20b23104f6b779b75be0
/frameworks/av/services/audioflinger/Threads.cpp
|
| f557872b594d15599bdb65efd91eb5adeb1d0840 |
|
30-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
am 908d3c09: am 51ec03c2: Merge "Implement Track::getTimestamp()" into klp-dev
* commit '908d3c09ca7f2ccb280aa5dc8d876099ff9a9d0f':
Implement Track::getTimestamp()
|
| bd096fd9d8e5fc0e62f98807f4818a06f70d0812 |
|
23-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Implement Track::getTimestamp()
using a new timestamp latch in PlaybackThread, and
AudioTrackServerProxy::framesReleased() which returns mServer.
Change-Id: I1ebfba968c773faaab95648c272fd3ebd74718d6
/frameworks/av/services/audioflinger/Threads.cpp
|
| 7165268ffa6c7b6b405b6afad82e2a346500e8ee |
|
15-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Don't lock and then immediately unlock effect chains
Change-Id: Id9a4b1e9c49cecfd3a008a7814b5c14dad9a5831
/frameworks/av/services/audioflinger/Threads.cpp
|
| b86432b1d62e7c9034223f0120b9bceb44215c41 |
|
15-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Fix races in reading activeTrack->mState without lock
Change-Id: I8cfbd9884fa5959a846ddb52cdc797266b896769
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9e98235a9e9bb870e1c76911e3b4d00386a52c39 |
|
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Simplify flow by having all mActiveTrack.clear() continue
Change-Id: Icd160b0aa79a747bd19fabaa7feee813afb155e0
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5edadd46c76c5ff1c3edabf2ea943c2278e82e1c |
|
15-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Don't sleep with effect chains locked
Change-Id: I6eae037304d3dc8a8f3ef55c5b210fd4f8856e99
/frameworks/av/services/audioflinger/Threads.cpp
|
| ad5bcc29b216412dbc4d936c270d30620c68f361 |
|
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Copy mActiveTrack for stability and easier access
Change-Id: If0d88610ec35128054a22f42b2d79f07c4b724eb
/frameworks/av/services/audioflinger/Threads.cpp
|
| 1ba19cd7fcdf18ab6efab2a1b831affab9a46157 |
|
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Simplify code by pulling up mActiveTrack == 0 test
Change-Id: I225bb032b2fde97d891191f3599d9850c2d27bb2
/frameworks/av/services/audioflinger/Threads.cpp
|
| d9fc34fb0fcfcc739f868b116edf50c62af19d5e |
|
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Remove dead 'if'
Change-Id: I23c082f0115ae598cad293be0ac66a130afb692a
/frameworks/av/services/audioflinger/Threads.cpp
|
| 4ef0b463a56c19bad9197aa9f90d792090461429 |
|
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Fix race in checking exitPending() in RecordThread::threadLoop()
Change-Id: I97a1c5ab17a6c334d275be794392dcf6c9a88d0e
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2cfbf88b89854f30b295e8ae26a031edb8d712f8 |
|
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Simplify code that checks for record standby
Take advantage of the new processConfigEvents_l()
and it's post condition that mConfigEvents is empty.
Change-Id: Icd40fb175f3e374203c9c8d2e9df2c3929bed89b
/frameworks/av/services/audioflinger/Threads.cpp
|
| 26a4029c95620a2b98187cf003cd3c58eea03747 |
|
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
checkForNewParameters_l returns a value which is currently ignored
Change-Id: I19b9cc9df094d67560f6e832d94fa284dca18687
/frameworks/av/services/audioflinger/Threads.cpp
|
| d5418eb594435c958d6c37fa9938161a0112adbd |
|
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Use Autolock where possible
Change-Id: Id0e1836831c45beda4e5cb11e983005ea699fc2a
/frameworks/av/services/audioflinger/Threads.cpp
|
| f777331418a86cd9fd709af898ef24a69967aeb4 |
|
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Add ThreadBase::processConfigEvents_l()
Change-Id: I6bcdc212a984c52ab98b0b0744943b4ecc1e1542
/frameworks/av/services/audioflinger/Threads.cpp
|
| c527a7c2b1bfd26e8f3086e1b653d56e521379d9 |
|
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Pull in scope of effectChains
Change-Id: Id5336f8663099088a0958fb8bd59fc5df9f92829
/frameworks/av/services/audioflinger/Threads.cpp
|
| 47c2070b2ce8aedb7300c0aad91caccf3c383841 |
|
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Add record thread locking comments and FIXMEs
Change-Id: Ia5bdc9b8b013c2e40af17c82833051290bf4df70
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2d94426cd3302cb1215c92c5f1c4b90c24ceb72b |
|
13-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Convert if/else-if in threadLoop to switch
Change-Id: I7428abb332ead6c2a1df5903f6c4e166be1f7abb
/frameworks/av/services/audioflinger/Threads.cpp
|
| 3468e8a4d79cc6a7bb0f03f8382426195bed44df |
|
14-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Fix switch indentation to follow our audio style
Change-Id: I137763917962b276838715319b163c763cb54c71
/frameworks/av/services/audioflinger/Threads.cpp
|
| 34fca34606b448e6b71c2942f63cb13a0aebd620 |
|
13-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Explicit comparison to 0 for integral types
Change-Id: I2bfe864d1bda719581e297a38d96abfc0e3748b9
/frameworks/av/services/audioflinger/Threads.cpp
|
| 6e2ebe97f2ad0a21907f20f9ee644c4eacbb7a40 |
|
13-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Use curly braces in 'if' to make it easier to add logs
Change-Id: I58b33fefdd8bf703647414157a99a3223be3531c
/frameworks/av/services/audioflinger/Threads.cpp
|
| e198c360d5e75a9b2097844c495c10902e7e8500 |
|
13-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Whitespace
Change-Id: Id3ccc183a03421330d0498faaa62a45915cdc3d6
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2fc14730e4697a6f456b4631549c9981f6b0b115 |
|
05-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Comments
Change-Id: I56f82904cb3855bad6f33ad1ff4ddf66a61c5720
/frameworks/av/services/audioflinger/Threads.cpp
|
| c0029a8551d65929cbf51a55bd3bae3342a1f2fe |
|
13-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Check input HAL channel mask instead of channel count"
|
| 030033342a6ea17003e6af38a56c7edc6d2ead01 |
|
07-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Clean up error handling in createTrack and openRecord
Outside callers now use initCheck() to determine whether a TrackBase
has been created successfully, instead of relying on internal knowledge.
Previously, callers needed to know that a TrackBase was only valid if it's
getCbk() != 0. For a Track (playback), they needed to know to also check
the track's name (track index). Now, outsiders can just call initCheck().
Other changes:
- Return a 0 reference if track creation fails
- Remove a dead line of code in AudioFlinger::openRecord
Change-Id: If374924a3f6fd27906f625aa83dd0a1e3f506e00
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9156ef3e11b68cc4b6d3cea77f1f63673855a6d1 |
|
07-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Status pointer passed by caller is always non-NULL
in createTrack, openRecord, createEffect, createTrack_l,
createRecordTrack_l, and createEffect_l.
Change-Id: I2e459e4de9c78145f4d496e6abf289479a2f0941
/frameworks/av/services/audioflinger/Threads.cpp
|
| e92db4bdb575d897d6bbd5086761fa9387393c83 |
|
12-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Pull up readyToRun into ThreadBase"
|
| b0e6b967c290c905c4c696df4a8ac9e9d591928b |
|
12-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify the code that allocates mMixBuffer by removing mAllocMixBuffer"
|
| 309f7abb3c170ba764e67e9b6fed31fb442e9953 |
|
09-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Use 0 for sp<>, NULL for raw pointers"
|
| f874f1375d758c8a7a84572a94371fbd4085f0df |
|
08-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Pull up mBufferSize from RecordThread to ThreadBase"
|
| ec3fb50c8ee25ac263a9c2cbe9242f877bef960c |
|
17-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Check input HAL channel mask instead of channel count
The only supported channel masks are mono and stereo;
arbitrary channel masks are not yet allowed.
Change-Id: I9cebbe56480b53b98e59aa55245a1f03a30d2839
/frameworks/av/services/audioflinger/Threads.cpp
|
| 937098b9b564a779ff1c8c2d9e60769ee5c69810 |
|
26-Jun-2013 |
Glenn Kasten <gkasten@google.com> |
Use 0 for sp<>, NULL for raw pointers
Change-Id: Ia0595afdc41e1dec428d7a3db73558b54c417bf3
/frameworks/av/services/audioflinger/Threads.cpp
|
| cf04c2cb8e031acc03c1c91cb1ccab15098c89b6 |
|
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Pull up readyToRun into ThreadBase
Change-Id: Ib150afd72c3b35d83599a61664e89302a8feb4b5
/frameworks/av/services/audioflinger/Threads.cpp
|
| c1fac191069774c7bfcb062edbb821ea56e7dbc0 |
|
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Simplify the code that allocates mMixBuffer by removing mAllocMixBuffer
Change-Id: I5ac41210251789a034d760309c50fe018f2442ce
/frameworks/av/services/audioflinger/Threads.cpp
|
| 70949c47fbae3f836d15f040551d7631be3ed7c2 |
|
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Pull up mBufferSize from RecordThread to ThreadBase
Now both playback and record threads have a HAL buffer size for fast
tracks, and a potentially larger-sized buffer for normal tracks.
Change-Id: I6f57b85ecece3cbd4e9e64b8c41c3352f7656912
/frameworks/av/services/audioflinger/Threads.cpp
|
| 90e58b1fefc8caf70b34301a92bc86179580b6fc |
|
01-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
RecordThread::createRecordTrack_l use flags for fast tracks
Look at client's request for a fast track in the flags parameter,
and check whether the request can be satisfied.
Change-Id: I65ddaeb1b85670fe9066076f638bfed0cda01c0d
/frameworks/av/services/audioflinger/Threads.cpp
|
| ddb0ccf3fb6fe8da8c71a6deb30561b821f3c0a2 |
|
01-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
RecordTrack::createRecordTrack_l flags are in/out
The flags are currently unused, but will be used for requesting a fast
track. Making flags in/out will allow reporting back up to client that
the request is denied.
Change-Id: Ifbee57da3632ce130551065a426577fb97b1a68d
/frameworks/av/services/audioflinger/Threads.cpp
|
| f36dbfe3d517e9311be0a4d0fd6658380690e84c |
|
02-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Report underruns for fast tracks also"
|
| 1a3739169becc850999c3c06db402541119a428c |
|
01-Aug-2013 |
Glenn Kasten <gkasten@android.com> |
am 6d4d8790: am b687ff9f: Merge "Fix mismatched delete/new couple"
* commit '6d4d879037a495f9a4276a6996d1845594ce9090':
Fix mismatched delete/new couple
|
| 82aaf94a5b18939e4d790bbc752031f3070704a3 |
|
18-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Report underruns for fast tracks also
This fixes a regression that was introduced earlier
by commit 9f80dd223d83d9bb9077fb6baee056cee4eaf7e5
called "New control block for AudioTrack and AudioRecord".
That commit broke underrun reporting for fast tracks.
Also remove Track::mUnderrunCount, which counted the number of underrun
events, and was only used by dumpsys media.audio_flinger.
Now dumpsys media.audio_flinger reports the number of underrun frames,
Isolated underrun-related control block accesses via the proxy, so that
the server is not directly poking around in the control block.
The new proxy APIs are AudioTrackServerProxy::getUnderrunFrames() and
AudioTrackServerProxy::tallyUnderrunFrames(). getUnderrunFrames() returns
a rolling counter for streaming tracks, or zero for static buffer tracks
which never underrun, but do a kind of 'pause' at end of buffer.
tallyUnderrunFrames() increments the counter by a specified number of frames.
Change-Id: Ib31fd73eb17cbb23888ce3af8ff29f471f5bd5a2
/frameworks/av/services/audioflinger/Threads.cpp
|
| b08ab81bab9e8cd2e3a7fff02725a387070fd078 |
|
31-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Add resampler comments and fix a typo"
|
| 34af02647b387a252fb02bab8e2cb9f7bd9c8abb |
|
30-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Add resampler comments and fix a typo
Change-Id: Ie071673875f663de4212eed4a4dff89d51a5a915
/frameworks/av/services/audioflinger/Threads.cpp
|
| 96f60d8f04432a1ed503b3e24d5736d28c63c9a2 |
|
12-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Rename control block flags to mFlags
Change-Id: I7b6d31e24531954ab1ecdf3ed56c19433700bd89
/frameworks/av/services/audioflinger/Threads.cpp
|
| 97252bb0e40a5a9b41f020163393cf6c5da10b7e |
|
30-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "AudioRecord and HAL input stream must be 16-bit PCM only"
|
| f20e1d8df84c5fbeeace0052d100982ae39bb7a4 |
|
12-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Rename control block server to mServer and add comments
Change-Id: Ieabd91acee92d0e84e66fbd358df5282b856306e
/frameworks/av/services/audioflinger/Threads.cpp
|
| a30e75897934da2ce7b1b03bcb4b58e139d3e81e |
|
30-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix theoretical race using TrackBase::sampleRate()"
|
| eb144bbbd0b18427562b0d7db5b38e11f9484739 |
|
09-Oct-2012 |
Andrei V. FOMITCHEV <andreix.fomitchev@intel.com> |
Fix mismatched delete/new couple
In RecordThread::readInputParameters(), mRsmpInBuffer is allocated
with new[] and should be freed with delete[] instead of delete.
This error has been found by valgrind.
Change-Id: I2f947abb658a21a7d1e61eebb99270cb37d2d147
Signed-off-by: Andrei V. FOMITCHEV <andreix.fomitchev@intel.com>
Signed-off-by: David Wagner <david.wagner@intel.com>
Signed-off-by: Jack Ren <jack.ren@intel.com>
Signed-off-by: Bruce Beare <bruce.j.beare@intel.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| fa319e6d918b84f93fb5457af5d1cca6421ac517 |
|
30-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Remove CC_LIKELY and CC_UNLIKELY where not needed
Only keep them in performance-sensitive code
Change-Id: Ib257ddd7bc39ce9896997ffae008c524ac743d01
/frameworks/av/services/audioflinger/Threads.cpp
|
| 548efc94813c1dec6e8cf6c085ae41ccb04827f1 |
|
29-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
Rename RecordThread::mInputBytes to the more generic mBufferSize
This prepares for using it in PlaybackThreads later
Change-Id: Id90a92aa6372e4b69914b0008cef07296ca5d6a3
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9b58f63e45ef2fdfb839b9b9bb3411d81eb96128 |
|
16-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Move members from ThreadBase to PlaybackThread
Move mNormalFrameCount and frameCountHAL(), since they're not used by
record threads. Also comment which fields are updated by readParameters().
Change-Id: I5fc0a8a89cc637976f22d49271a5a3e136dab4e1
/frameworks/av/services/audioflinger/Threads.cpp
|
| 7f249fa9bcb64da324d19f551943fac7686d221c |
|
29-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Use standard name and type for channel mask"
|
| 175f8338e67aae6d0ca420b8f45a5f7cab4394d5 |
|
29-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify getParameters()"
|
| f6ed423af92a56ef54bba23eba883b1f21448b54 |
|
16-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Treat mChannelCount as uint32_t consistently
mChannelCount was 8-, 16-, or 32-bits
Change-Id: I2cc2fedf3e33144e5c8bbd9894763282d9217f63
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2eeca1f8e18106d118f7268cb9d8597539685c5c |
|
29-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Move local variable declarations to point of first use"
|
| b283374cbf79dc5a36f1eef0015d7baf8881792e |
|
29-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Move delete AudioMixer closer to point of re-allocation"
|
| afbbc6c5f6bc470d504dcf24ceb3ac21d57fb78d |
|
29-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify RecordTrack::stop()"
|
| 422801c5e9b795105f28e639c4bed5ee3f4e56f3 |
|
26-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "HAL stream format for mixer output threads must be stereo 16-bit PCM"
|
| a8356f663014e7d4c27869629af83d8bb3441e19 |
|
25-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Simplify RecordTrack::stop()
Change-Id: Ib959c1e9dc9544d12277ce11bea445118b2e0521
/frameworks/av/services/audioflinger/Threads.cpp
|
| bfb1b832079bbb9426f72f3863199a54aefd02da |
|
07-Jan-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: offload playback, non-blocking write
- Added specialized playback thread class for offload playback,
derived from directoutput thread.
This thread type handles specific state transitions for offloaded
tracks and offloading commands (pause/resume/drain/flush..) to audio HAL.
As opposed to other threads, does not go to standby if the track is paused.
- Added support for asynchronous write and drain operations at audio HAL.
Use a thread to handle async callback events from HAL: this avoids locking
playback thread mutex when executing the callback and cause deadlocks when
calling audio HAL functions with the playback thread mutex locked.
- Better accouting for track activity: call start/stop and release Output
methods in audio policy manager when tracks are actually added and removed
from the active tracks list.
Added a command thread in audio policy service to handle stop/release commands
asynchronously and avoid deadlocks with playback thread.
- Track terminated status is not a state anymore. This condition is othogonal
to state to permitted state transitions while terminated.
Change-Id: Id157f4b3277620568d8eace7535d9186602564de
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9e8fcbcd8efa51d70d1207ff57bfbfe31324287a |
|
25-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Move delete AudioMixer closer to point of re-allocation
No need to delete AudioMixer before readOutputParameters
Change-Id: Icafa785c9021bbe8d985e4f9527f8fc8c5e62622
/frameworks/av/services/audioflinger/Threads.cpp
|
| 7fc97ba08e2850f3f16db704b78ce78e3dbe1ff0 |
|
17-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
HAL stream format for mixer output threads must be stereo 16-bit PCM
Direct and tunnel output threads can support various HAL stream formats,
included encoded. But currently there are stereo 16-bit PCM assumptions
in several places for mixer and duplicating output threads:
- mMixBuffer and mixBuffer()
- AudioMixer including resampler
- FastMixer's mixBuffer
- effects
- NBAIO_Format
- anywhere FCC_2 is used
- possibly other places
Until those assumptions are removed, this CL enforces stereo 16-bit
PCM in mixer and duplicating threads at the place where the HAL format
is read. It was already being checked in checkForNewParameters_l(),
but not in readOutputParameters().
Change-Id: Ibe344cc922743da234299097aa1bb1f54795cc9b
/frameworks/av/services/audioflinger/Threads.cpp
|
| d8ea699dc8e7dac58bb32e9cdb31b0758da25817 |
|
16-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Simplify getParameters()
Change-Id: Iedfeca3cd477d023c350d6d4e6eed874ee467a32
/frameworks/av/services/audioflinger/Threads.cpp
|
| 34542acfa25c6413c87a94b6f7cc315a0c496277 |
|
26-Jun-2013 |
Glenn Kasten <gkasten@google.com> |
Move local variable declarations to point of first use
Change-Id: Ideb83dea2c3002651c34fa646753cba598e29e93
/frameworks/av/services/audioflinger/Threads.cpp
|
| fad226abd12435dbcd232f7de396f1a097b2bd5f |
|
17-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Use standard name and type for channel mask
Former name 'channels' was ambiguous with respect to channel count.
Change-Id: I716f792d95a7e0c787d27514ad6e93dbcef8a415
/frameworks/av/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| 4182c4e2a07e2441fcd5c22eaff0ddfe7f826f61 |
|
15-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Use AudioSystem::setLowRamDevice() to configure memory
Bug: 9798886
Change-Id: I9321e3f369f1ed9429ae222e3926ebdeb012b8b0
/frameworks/av/services/audioflinger/Threads.cpp
|
| 153b9fe667e6e78e0218ff0159353097428c7657 |
|
15-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Make AudioFlinger::instantiate() more resilient when called from separate module
Bug: 8834855
Change-Id: I4cd842cdfb09d2aaaaab9df9ac3bec6179709bd3
/frameworks/av/services/audioflinger/Threads.cpp
|
| c6ae3c8a261794fd4445e4e152d1ada074a3f92f |
|
17-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Fix theoretical memory leak in mConfigEvents
Change-Id: I137f70676c8919661e716c33e0dd9c25c2b6285c
/frameworks/av/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| 9fdcb0a9497ca290bcf364b10868587b6bde3a34 |
|
27-Jun-2013 |
Glenn Kasten <gkasten@google.com> |
Fix theoretical race using TrackBase::sampleRate()
In two places we assumed that TrackBase::sampleRate() would return the
same value when it is called twice in the same function. This is not
guaranteed; sampleRate() reads from the control block so the return
value could change. To fix this, only call sampleRate() once and cache
the return value to get a consistent value.
This was only a theoretical race. In MixerThread::prepareTracks_l()
it would have no bad effect. In TimedTrack::getNextBuffer() it could
cause a real problem, but we don't currently support dynamic sample rate
ratios for timed tracks.
Change-Id: I8e5c33f0121fc058d1e70c2ab5e9135397d3e0b7
/frameworks/av/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| fe9a901a445633e70073b1b91fca3af34d8e8132 |
|
12-Jun-2013 |
Eric Laurent <elaurent@google.com> |
am bb35be24: am 7be48189: Merge "audioflinger: fix effects on direct output threads" into jb-mr2-dev
* commit 'bb35be24239f05f59715960a93cd491a4cd3b4d6':
audioflinger: fix effects on direct output threads
|
| d0107bcd44fe608b0c00a8843d19fb6356c4cb69 |
|
11-Jun-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix effects on direct output threads
PlaybackThread::addTrack_l() uses the assumption that
effects are attached to a track only if the track accumulation
buffer is different from the mixer thread output buffer.
This is not true for direct output threads where only one track is active
an only one buffer is needed.
This assumption is an optimization to avoid checking for effect chains with
the same session ID each time a track is processed. The optimization
is not key if only one track is attached to the thread which is the case for
direct outputs.
Current code fails to increment the active track count in the effect chain
on direct output threads when a track is started thus making the effect
framework clear the mix buffer and produce silence each time the mixer runs.
The fix consists in removing the optimization described above.
Bug: 9324989.
Change-Id: Id7a6337450ed90d326299c2ce9fc02f4b9e2fa6f
/frameworks/av/services/audioflinger/Threads.cpp
|
| 04b035e3ccbf2919e4447c66e6483c11f2889f01 |
|
07-Jun-2013 |
Eric Laurent <elaurent@google.com> |
am d3a5b48a: am 18b46679: Merge "AudioFlinger: do not cache audio device when 0." into jb-mr2-dev
* commit 'd3a5b48a1bc4f5d6ff4472f54f909abaadd5fa39':
AudioFlinger: do not cache audio device when 0.
|
| 7e1139c0377b6806942fb2a043737b3b9cf0ae91 |
|
07-Jun-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: do not cache audio device when 0.
AudioFlinger should not cache the audio device
passed to the audio HAL when it is AUDIO_DEVICE_NONE
but keep previous valid selection instead.
Bug: 9323399.
Change-Id: I6f9480e55a21be4115453e6a5eebc5cf2536c476
/frameworks/av/services/audioflinger/Threads.cpp
|
| 7c027248e1a4ccd5b22bc4deafb03e2d87ac8f38 |
|
26-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Consistent whitespace
Change-Id: I118cce68d3b777f9ec9b6bfb70367496422a40f2
/frameworks/av/services/audioflinger/Threads.cpp
|
| 61d404efeaaf3ea1b615c160b843ac9f274d0018 |
|
20-May-2013 |
Dianne Hackborn <hackbod@google.com> |
Update to supply correct name for new power manager app ops.
Change-Id: Ia686319509d98a4f467479ea494e8a54f2c1e238
/frameworks/av/services/audioflinger/Threads.cpp
|
| f8197a6a9d9363cb52bb8a2c15c0e5a52064355e |
|
23-Apr-2013 |
Glenn Kasten <gkasten@google.com> |
Remove timing jitter during startup of audio
This fixes a regression introduced recently, that increased timing jitter
during the startup of the FastMixer and AudioTrack callback threads.
The regression was to make requestPriority() asynchronous as a way to
avoid an apparent priority inversion in system_server. This means that
the target thread could run briefly with the initial priority, before
the new priority takes effect.
This change removes the startup jitter for FastMixer, by making the
requestPriority() synchronous again for that case. It doesn't matter
that this restores the priority inversion involving normal mixer thread,
because it happens during startup of both threads.
The change also removes the startup jitter for the AudioTrack callback
thread, by having the target thread check whether the requestPriority()
has completed yet. If not, the target thread blocks with a timeout
until the priority boost finishes.
Finally, we now log an error message if the expected priority boost
doesn't happen.
Bug: 8698989
Change-Id: Id590e9a274b70ec1ba85b44a585ee37a22e41cbc
/frameworks/av/services/audioflinger/Threads.cpp
|
| a07f17ca46db04c9d5d9e7d6b2878db59ca2b9ea |
|
23-Apr-2013 |
Glenn Kasten <gkasten@google.com> |
Remove timing jitter during startup of audio
This fixes a regression introduced recently, that increased timing jitter
during the startup of the FastMixer and AudioTrack callback threads.
The regression was to make requestPriority() asynchronous as a way to
avoid an apparent priority inversion in system_server. This means that
the target thread could run briefly with the initial priority, before
the new priority takes effect.
This change removes the startup jitter for FastMixer, by making the
requestPriority() synchronous again for that case. It doesn't matter
that this restores the priority inversion involving normal mixer thread,
because it happens during startup of both threads.
The change also removes the startup jitter for the AudioTrack callback
thread, by having the target thread check whether the requestPriority()
has completed yet. If not, the target thread blocks with a timeout
until the priority boost finishes.
Finally, we now log an error message if the expected priority boost
doesn't happen.
Bug: 8698989
Change-Id: Id590e9a274b70ec1ba85b44a585ee37a22e41cbc
/frameworks/av/services/audioflinger/Threads.cpp
|
| 7fd54ff906c69400727fc8e90118a8a0429d7ab2 |
|
04-Apr-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: multiple tracks on direct output
Fix problems when multiple tracks exist on a direct output
thread: when multiple tracks are active, continue to update all
track states but only take into account the last track started
for audio HAL volume and mixer state control.
Bug 8388941
Change-Id: I57e6757286f41651dda99b11a5449e431812431b
/frameworks/av/services/audioflinger/Threads.cpp
|
| d595b7c858c481a07745674ce2d8a6690e980e74 |
|
04-Apr-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: multiple tracks on direct output
Fix problems when multiple tracks exist on a direct output
thread: when multiple tracks are active, continue to update all
track states but only take into account the last track started
for audio HAL volume and mixer state control.
Bug 8388941
Change-Id: I57e6757286f41651dda99b11a5449e431812431b
/frameworks/av/services/audioflinger/Threads.cpp
|
| a265845ecafe06c9668ce7ac67636657f4bedf86 |
|
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/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| 10774e144b3c99afad9a5070ccf50fdb12b5fb4e |
|
02-Mar-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "media.log cleanup" into jb-mr2-dev
|
| bf04b5860182d8f4130dcb5d6d88ee68a58c99cd |
|
02-Mar-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "media.log cleanup" into jb-mr2-dev
|
| d3c4b134a87b96227b90b9ec052d8a6e9880bbdf |
|
27-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
media.log cleanup
Remove almost all of the specific logs, but leave the media.log
logging infrastructure in place for the next time we need it.
Re-apply a few good changes that were reverted earlier:
- check logf format vs. argument list compatibility
- distinguish potentially modified and actually modified tracks in FastMixer
- fix benign bug where sq->end() was called more than once
- fix a build warning
Bug: 6490974
Change-Id: I02d3e83646c738acaebb415bd0d6b548638b4ef5
/frameworks/av/services/audioflinger/Threads.cpp
|
| ab7d72f0804fbb7e91ad9d2a16f826d97e20e5d0 |
|
27-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
media.log cleanup
Remove almost all of the specific logs, but leave the media.log
logging infrastructure in place for the next time we need it.
Re-apply a few good changes that were reverted earlier:
- check logf format vs. argument list compatibility
- distinguish potentially modified and actually modified tracks in FastMixer
- fix benign bug where sq->end() was called more than once
- fix a build warning
Bug: 6490974
Change-Id: I02d3e83646c738acaebb415bd0d6b548638b4ef5
/frameworks/av/services/audioflinger/Threads.cpp
|
| dd0bda0c47871d4964b5035a096c7c8c27fa7469 |
|
26-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Remove tee sink debugging at compile time
Bug: 8223560
Change-Id: Iddbfb06c45d43d9f20bb428215dd4094931e19a7
/frameworks/av/services/audioflinger/Threads.cpp
|
| 46909e7eb074ce1b95b8a411eb71154f53f84f77 |
|
26-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Remove tee sink debugging at compile time
Bug: 8223560
Change-Id: Iddbfb06c45d43d9f20bb428215dd4094931e19a7
/frameworks/av/services/audioflinger/Threads.cpp
|
| dd4abb5177b8d94cd30a787d4ce336d5110c185c |
|
10-Jan-2013 |
Glenn Kasten <gkasten@google.com> |
Update tee sink
Implement rotation to reduce long-term storage use.
Implement optional per-track tee.
Dynamically enable at runtime based on property, instead of at compile-time.
Dynamic frame count not yet implemented.
Bug: 8223560
Change-Id: I3706443c6ec0cb0c6656dc288715a02ad5fea63a
/frameworks/av/services/audioflinger/Threads.cpp
|
| da6ef1320d0161b1640dc84d7a9c5a25860c3619 |
|
10-Jan-2013 |
Glenn Kasten <gkasten@google.com> |
Update tee sink
Implement rotation to reduce long-term storage use.
Implement optional per-track tee.
Dynamically enable at runtime based on property, instead of at compile-time.
Dynamic frame count not yet implemented.
Bug: 8223560
Change-Id: I3706443c6ec0cb0c6656dc288715a02ad5fea63a
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5f6f3764989601d3b3ee65beaf1447d2a6b5f284 |
|
16-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 32584a7d672864b20ab8b83a3cb23c1858e908b7
Change-Id: I9dc680578b955b1af462eeb7a49d61a0d45eb81b
/frameworks/av/services/audioflinger/Threads.cpp
|
| 7f5d335f7b4caecd0dfb8f1085f352f1d2da5d2e |
|
16-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 32584a7d672864b20ab8b83a3cb23c1858e908b7
Change-Id: I9dc680578b955b1af462eeb7a49d61a0d45eb81b
/frameworks/av/services/audioflinger/Threads.cpp
|
| dd9764290b3c1d801fea9505189cae29db919902 |
|
13-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
- log mFastIndex when AudioMixer sees an invalid bufferProvider.
- log both potentially modified and actually modified tracks in FastMixer.
- fix benign bug where sq->end() was called more than once.
- log StateQueue push() call and return.
- increase StateQueue size from 4 to 8 entries
- log mixer->enable(), bufferProvider, and currentTrackMask
- log buffer provider addresses
- increase fast mixer log buffer again
- check logf format vs. argument list compatibility
- add logging to AudioMixer
- add checking of magic field in AudioMixer to detect overwrites
- add bool AudioMixer::enabled()
- increase log buffer sizes yet again
- enable assertion checking without ALOGV
- improve a few log messages
- check for corruption in more places
- log in all the process hooks
- add new mixer APIs so we can check for corruption of mixer state
- fix a build warning
Bug: 6490974
Change-Id: Ib0c4a73dcf606ef9bd898313b3b40ef61ab42f51
/frameworks/av/services/audioflinger/Threads.cpp
|
| 32584a7d672864b20ab8b83a3cb23c1858e908b7 |
|
13-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
- log mFastIndex when AudioMixer sees an invalid bufferProvider.
- log both potentially modified and actually modified tracks in FastMixer.
- fix benign bug where sq->end() was called more than once.
- log StateQueue push() call and return.
- increase StateQueue size from 4 to 8 entries
- log mixer->enable(), bufferProvider, and currentTrackMask
- log buffer provider addresses
- increase fast mixer log buffer again
- check logf format vs. argument list compatibility
- add logging to AudioMixer
- add checking of magic field in AudioMixer to detect overwrites
- add bool AudioMixer::enabled()
- increase log buffer sizes yet again
- enable assertion checking without ALOGV
- improve a few log messages
- check for corruption in more places
- log in all the process hooks
- add new mixer APIs so we can check for corruption of mixer state
- fix a build warning
Bug: 6490974
Change-Id: Ib0c4a73dcf606ef9bd898313b3b40ef61ab42f51
/frameworks/av/services/audioflinger/Threads.cpp
|
| ce70d400ff9fb614e334ab2b661df2933533ef57 |
|
15-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 639482c24c911b125398b31883ba6d55faebe28b
Change-Id: I11f2829072ab11e18b0663024f27bf31192f1d39
/frameworks/av/services/audioflinger/Threads.cpp
|
| e186b51e0a9834b287d7a509e960eaf1b688db75 |
|
15-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 639482c24c911b125398b31883ba6d55faebe28b
Change-Id: I11f2829072ab11e18b0663024f27bf31192f1d39
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5881f18029deb80eb83ea88046d0593441be79c7 |
|
13-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
- log mFastIndex when AudioMixer sees an invalid bufferProvider.
- log both potentially modified and actually modified tracks in FastMixer.
- fix benign bug where sq->end() was called more than once.
- log StateQueue push() call and return.
- increase StateQueue size from 4 to 8 entries
- log mixer->enable(), bufferProvider, and currentTrackMask
- log buffer provider addresses
- increase fast mixer log buffer again
- check logf format vs. argument list compatibility
- add logging to AudioMixer
- add checking of magic field in AudioMixer to detect overwrites
- add bool AudioMixer::enabled()
Bug: 6490974
Change-Id: I1f3f18aa62d9fbd35bc32285b669f5ba40efe28e
/frameworks/av/services/audioflinger/Threads.cpp
|
| 639482c24c911b125398b31883ba6d55faebe28b |
|
13-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
- log mFastIndex when AudioMixer sees an invalid bufferProvider.
- log both potentially modified and actually modified tracks in FastMixer.
- fix benign bug where sq->end() was called more than once.
- log StateQueue push() call and return.
- increase StateQueue size from 4 to 8 entries
- log mixer->enable(), bufferProvider, and currentTrackMask
- log buffer provider addresses
- increase fast mixer log buffer again
- check logf format vs. argument list compatibility
- add logging to AudioMixer
- add checking of magic field in AudioMixer to detect overwrites
- add bool AudioMixer::enabled()
Bug: 6490974
Change-Id: I1f3f18aa62d9fbd35bc32285b669f5ba40efe28e
/frameworks/av/services/audioflinger/Threads.cpp
|
| 7f2f8042cf335ab1323dec3edbe9143a06109f4e |
|
14-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 0ddd56316262ac74a95e9edb595697c163136d6d
Change-Id: I180a928af6f5a38d15a5efe44cd1fe927b5d961c
/frameworks/av/services/audioflinger/Threads.cpp
|
| 51eb3965caa8ba135bcdd8ffb7a2024a042ecdc0 |
|
14-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 0ddd56316262ac74a95e9edb595697c163136d6d
Change-Id: I180a928af6f5a38d15a5efe44cd1fe927b5d961c
/frameworks/av/services/audioflinger/Threads.cpp
|
| b900234f9e772bb7591a551f06ba17ee1969368f |
|
13-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
- log mFastIndex when AudioMixer sees an invalid bufferProvider.
- log both potentially modified and actually modified tracks in FastMixer.
- fix benign bug where sq->end() was called more than once.
- log StateQueue push() call and return.
Bug: 6490974
Change-Id: Iee7c8f40e20b6000cd8286c0ec6a14fff4a37af1
/frameworks/av/services/audioflinger/Threads.cpp
|
| 0ddd56316262ac74a95e9edb595697c163136d6d |
|
13-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
- log mFastIndex when AudioMixer sees an invalid bufferProvider.
- log both potentially modified and actually modified tracks in FastMixer.
- fix benign bug where sq->end() was called more than once.
- log StateQueue push() call and return.
Bug: 6490974
Change-Id: Iee7c8f40e20b6000cd8286c0ec6a14fff4a37af1
/frameworks/av/services/audioflinger/Threads.cpp
|
| c93aff0a698265501f46393f3146b884aa357aee |
|
12-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 3051df27261e9952c0e642dec548515250e85f6a
Change-Id: I8bf5c3e91b65bd20de26f480c367c2854b62373c
/frameworks/av/services/audioflinger/Threads.cpp
|
| ecd9389c8712aedeb2a79823ea0e4fb842684269 |
|
12-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Revert "Temporary additional logging to investigate bug"
This reverts commit 3051df27261e9952c0e642dec548515250e85f6a
Change-Id: I8bf5c3e91b65bd20de26f480c367c2854b62373c
/frameworks/av/services/audioflinger/Threads.cpp
|
| a1bf28b533907b0b234e52c72bc372b3fa49c5a3 |
|
12-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
Bug: 6490974
Change-Id: I4c030f171343fe4b483eae0ddea4427118d8d4b1
/frameworks/av/services/audioflinger/Threads.cpp
|
| 3051df27261e9952c0e642dec548515250e85f6a |
|
12-Feb-2013 |
Glenn Kasten <gkasten@google.com> |
Temporary additional logging to investigate bug
The bug appears related to continuing to use an invalid buffer provider
in fast mixer after track destruction, so focus the added logs in that area.
Also includes a bug fix: was calling log in an unsafe place
near Threads.cpp AudioFlinger::PlaybackThread::createTrack_l line 1250.
Details:
- include caller pid or client pid where appropriate
- increase log buffer size
Bug: 6490974
Change-Id: I4c030f171343fe4b483eae0ddea4427118d8d4b1
/frameworks/av/services/audioflinger/Threads.cpp
|
| 83fa606bd629ac32a2dab5af255188acfa30db4a |
|
04-Feb-2013 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger: fix RecordThread initial device"
|
| 59686720182dcaa1a1a4757964df38c16f1a5246 |
|
04-Feb-2013 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger: fix RecordThread initial device"
|
| 201fc9c9feb4765a12bc39e95d25f3aa1deb8ffe |
|
02-Feb-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix RecordThread initial device
A regression was introduced when the audio device enums where
modified for a 32 bit representation: the device passed when
constructing a RecordThread was still the concatenation of input
device and output device bit fields on one 32 bit value which
is not possible anymore.
The fix consists in modifying the RecordThread constructor to accept
separate values for input and output devices.
Change-Id: I81fb5f4718428b54251e65d74b86e198ce15193e
/frameworks/av/services/audioflinger/Threads.cpp
|
| d3922f72601d82c6fc067a98916fda0bd1291c5f |
|
02-Feb-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix RecordThread initial device
A regression was introduced when the audio device enums where
modified for a 32 bit representation: the device passed when
constructing a RecordThread was still the concatenation of input
device and output device bit fields on one 32 bit value which
is not possible anymore.
The fix consists in modifying the RecordThread constructor to accept
separate values for input and output devices.
Change-Id: I81fb5f4718428b54251e65d74b86e198ce15193e
/frameworks/av/services/audioflinger/Threads.cpp
|
| 011aa652a9349750dd6bca1dcb1b82f19e07504e |
|
19-Jan-2013 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger uses media.log service for logging
Change-Id: Ia0f8204334f6b233f644d897762a18c95d936b4b
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9e58b552f51b00b3b674102876bd6c77ef3da806 |
|
19-Jan-2013 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger uses media.log service for logging
Change-Id: Ia0f8204334f6b233f644d897762a18c95d936b4b
/frameworks/av/services/audioflinger/Threads.cpp
|
| 3a948fceb6b59121cc2e5eeb458064150870ffde |
|
18-Jan-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix build with debug log enabled
Change-Id: Id397155ec884c41cb84b35462ea09a97a04d3ed2
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2592f6e68edbed386e004d5b045233f15e7b9ba1 |
|
18-Jan-2013 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix build with debug log enabled
Change-Id: Id397155ec884c41cb84b35462ea09a97a04d3ed2
/frameworks/av/services/audioflinger/Threads.cpp
|
| 856ff4e4c3c43550f013e80277358fdf514342bf |
|
22-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Start isolating control block accesses in a proxy"
|
| d5681bc9a38fe4cd1d591e6ae62b9c68fb851041 |
|
22-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Merge "Start isolating control block accesses in a proxy"
|
| 92b1343f5fbc2fc43198a0a1252fe023cc9a5061 |
|
14-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Fix build warnings
Change-Id: I9e3cfa0c6b3467fe763ce55f759d179f02c5deea
/frameworks/av/services/audioflinger/Threads.cpp
|
| c4974312e5a1e2ab94eca56045f991baf1508d73 |
|
14-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Fix build warnings
Change-Id: I9e3cfa0c6b3467fe763ce55f759d179f02c5deea
/frameworks/av/services/audioflinger/Threads.cpp
|
| 552f274d82fb4988fb972df69134e031d375abfe |
|
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/services/audioflinger/Threads.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/services/audioflinger/Threads.cpp
|
| 30c01816ddee2268fcd3d965240cba9c00301e7d |
|
04-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Prepare for track invalidation to be done by proxy
Don't rely on control block to determine whether track has been marked
invalid. Instead, use a local flag that can't be corrupted by client.
Change-Id: I783dafe828f93c1c3d2d0e5a08105ea536436efb
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5736c35b841de56ce394b4879389f669b61425e6 |
|
04-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Prepare for track invalidation to be done by proxy
Don't rely on control block to determine whether track has been marked
invalid. Instead, use a local flag that can't be corrupted by client.
Change-Id: I783dafe828f93c1c3d2d0e5a08105ea536436efb
/frameworks/av/services/audioflinger/Threads.cpp
|
| 4b3a49e3a9ced6e63adab08acc5b720f8feddefa |
|
29-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
AudioTrack::mute() is unused so remove it
If ever needed again, it could be implemented on client side by forcing
a track volume of 0.
Change-Id: I88a9b4f675b6dca2948549414f9ec2c192d29269
/frameworks/av/services/audioflinger/Threads.cpp
|
| e4756fe3a387615acb63c6a05788c8db9b5786cb |
|
29-Nov-2012 |
Glenn Kasten <gkasten@google.com> |
AudioTrack::mute() is unused so remove it
If ever needed again, it could be implemented on client side by forcing
a track volume of 0.
Change-Id: I88a9b4f675b6dca2948549414f9ec2c192d29269
/frameworks/av/services/audioflinger/Threads.cpp
|
| af348741539ae4a84aa929a54ecf359dae96f3ee |
|
30-Nov-2012 |
Alex Ray <aray@google.com> |
Remove conditional compilation of ATRACE functions
Tracing functions are meant to be dynamically controlled via sysprops.
Conditional compilation removes this functionality.
Change-Id: I26bc473d104d0b3c50a228dddfda3fa2428d157a
/frameworks/av/services/audioflinger/Threads.cpp
|
| 371eb9756c32109ea572b91216b19bb623f6d3fd |
|
30-Nov-2012 |
Alex Ray <aray@google.com> |
Remove conditional compilation of ATRACE functions
Tracing functions are meant to be dynamically controlled via sysprops.
Conditional compilation removes this functionality.
Change-Id: I26bc473d104d0b3c50a228dddfda3fa2428d157a
/frameworks/av/services/audioflinger/Threads.cpp
|
| 7a90bc9265782788675af577c7b1c56e5d5be709 |
|
30-Nov-2012 |
Simon Wilson <simonwilson@google.com> |
Use ATRACE macros instead of Tracer statics
ATRACE_BEGIN and ATRACE_END have replaced the static
Tracer::traceBegin and Tracer::traceEnd functions, so
use them instead.
Fixes compilation errors when tracing is enabled.
Change-Id: I4d1147d2f76afcdf113e9986f0544cb848802b15
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2d590964aa58e137d17a43e095e6443dd0fe2e98 |
|
30-Nov-2012 |
Simon Wilson <simonwilson@google.com> |
Use ATRACE macros instead of Tracer statics
ATRACE_BEGIN and ATRACE_END have replaced the static
Tracer::traceBegin and Tracer::traceEnd functions, so
use them instead.
Fixes compilation errors when tracing is enabled.
Change-Id: I4d1147d2f76afcdf113e9986f0544cb848802b15
/frameworks/av/services/audioflinger/Threads.cpp
|
| ca7cc8273ffd88b9b89655808ee7e3df74162b83 |
|
19-Nov-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger files reorganization
Audioflinger.cpp and Audioflinger.h files must be split to
improve readability and maintainability.
This CL splits the files as follows:
AudioFlinger.cpp split into:
- AudioFlinger.cpp: implementation of IAudioflinger interface and global methods
- AFThreads.cpp: implementation of ThreadBase, PlaybackThread, MixerThread,
DuplicatingThread, DirectOutputThread and RecordThread.
- AFTracks.cpp: implementation of TrackBase, Track, TimedTrack, OutputTrack,
RecordTrack, TrackHandle and RecordHandle.
- AFEffects.cpp: implementation of EffectModule, EffectChain and EffectHandle.
AudioFlinger.h is modified by inline inclusion of header files containing
the declaration of complex inner classes:
- AFThreads.h: ThreadBase, PlaybackThread, MixerThread, DuplicatingThread,
DirectOutputThread and RecordThread
- AFEffects.h: EffectModule, EffectChain and EffectHandle
AFThreads.h includes the follownig headers inline:
- AFTrackBase.h: TrackBase
- AFPlaybackTracks: Track, TimedTrack, OutputTrack
- AFRecordTracks: RecordTrack
Change-Id: I512ebc3a51813ab7a4afccc9a538b18125165c4c
/frameworks/av/services/audioflinger/Threads.cpp
|
| 81784c37c61b09289654b979567a42bf73cd2b12 |
|
19-Nov-2012 |
Eric Laurent <elaurent@google.com> |
AudioFlinger files reorganization
Audioflinger.cpp and Audioflinger.h files must be split to
improve readability and maintainability.
This CL splits the files as follows:
AudioFlinger.cpp split into:
- AudioFlinger.cpp: implementation of IAudioflinger interface and global methods
- AFThreads.cpp: implementation of ThreadBase, PlaybackThread, MixerThread,
DuplicatingThread, DirectOutputThread and RecordThread.
- AFTracks.cpp: implementation of TrackBase, Track, TimedTrack, OutputTrack,
RecordTrack, TrackHandle and RecordHandle.
- AFEffects.cpp: implementation of EffectModule, EffectChain and EffectHandle.
AudioFlinger.h is modified by inline inclusion of header files containing
the declaration of complex inner classes:
- AFThreads.h: ThreadBase, PlaybackThread, MixerThread, DuplicatingThread,
DirectOutputThread and RecordThread
- AFEffects.h: EffectModule, EffectChain and EffectHandle
AFThreads.h includes the follownig headers inline:
- AFTrackBase.h: TrackBase
- AFPlaybackTracks: Track, TimedTrack, OutputTrack
- AFRecordTracks: RecordTrack
Change-Id: I512ebc3a51813ab7a4afccc9a538b18125165c4c
/frameworks/av/services/audioflinger/Threads.cpp
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