| 1bb9082596c9cac2995dc9b4465a7c6d5666d099 |
|
03-May-2017 |
rago <rago@google.com> |
Fix wait time for audio threads in suspend mode
Better estimation of wait time in suspended audio threads.
Bug: 34772697
Test: phone calls with BT headset.
Change-Id: Icf99c0671def810b57332a321e4de4680697715b
/frameworks/av/services/audioflinger/Threads.cpp
|
| dff774afa22569774eb2c8c25b1b3f934badea4b |
|
22-Apr-2017 |
Eric Laurent <elaurent@google.com> |
audioflinger: improve mmap stream volume
First try to use HW volume control and then fall back
to volume control in AAudio service.
Bug: 37518243
Test: play a sound and check AAudio volume callback is called
when pressing volume keys.
Change-Id: Ie18be9f553ae23c247974d8923e7dcddeb1047c8
/frameworks/av/services/audioflinger/Threads.cpp
|
| d9534fb35245b8bcd2489e717bdd1ad9269b3000 |
|
12-Apr-2017 |
Phil Burk <philburk@google.com> |
Merge "AAudioService: integrated with audioserver" into oc-dev
|
| c0c70e3c7dd10bc2c0caffcab1f3f5fb406b35fb |
|
09-Feb-2017 |
Phil Burk <philburk@google.com> |
AAudioService: integrated with audioserver
Call the MmapStreamInterface from AudioFlinger instead of the FakeHAL.
Fix sending timestamps from the thread.
Add shared mode in service.
Bug: 35260844
Bug: 33398120
Test: CTS test_aaudio.cpp
Change-Id: I44c7e4ecae4ce205611b6b73a72e0ae8a5b243e5
Signed-off-by: Phil Burk <philburk@google.com>
(cherry picked from commit 7f6b40d78b1976c78d1300e8a51fda36eeb50c5d)
/frameworks/av/services/audioflinger/Threads.cpp
|
| ddb332138cc8f3b4efad9b14b038f0a462b1b661 |
|
12-Apr-2017 |
Glenn Kasten <gkasten@google.com> |
Merge "Improve audio logs and dumpsys media.audio_flinger" into oc-dev
|
| 38f86713cbd22caae9fdd4ac2f0bb76ff91df34d |
|
11-Apr-2017 |
Jean-Michel Trivi <jmtrivi@google.com> |
Audio policy / flinger: updates to recording callback and record stop
Do not trigger the callback for recording events when the patch
handle is NONE, the patch handle will be updated for the
recording session later.
When stopping recording, only stop the record thread if the
track is not active, not if it's not the first one.
Test: cts-tradefed run cts-dev -m CtsMediaTestCases -t android.media.cts.AudioRecordingConfigurationTest
Bug: 36569938
Change-Id: I1b7445ef4115d6337ba7b926f7e6b068361926a7
/frameworks/av/services/audioflinger/Threads.cpp
|
| 6d8018f0b7be9deec6b0acab10a0dca6e91d0fb8 |
|
21-Feb-2017 |
Glenn Kasten <gkasten@google.com> |
Improve audio logs and dumpsys media.audio_flinger
Include mFrameCount in shared memory corrupt error log.
Fix typo in log for primary audio interface.
Include the new server frame count in logs for fast tracks.
Always log when fast is successful for output tracks, as it helpful
(was already there for input).
Cleanup dumpsys for threads.
Bug: 37153050
Test: builds OK, and logs contain more information
Change-Id: I2b5b4d700e8eb7c261a3125fb03ddc00bf08537e
(cherry picked from commit 1bfe09a0b1755a79abd32b41c0dd433b88fc260c)
/frameworks/av/services/audioflinger/Threads.cpp
|
| 293558ad1977e24e65f7ba78f47382d33fc77d64 |
|
21-Mar-2017 |
Andy Hung <hunga@google.com> |
AudioFlinger: Improve Thread logging.
Test: dumpsys media.audio_flinger
Bug: 30572472
Change-Id: I43f72354a6ece045f5f9f664946a406166974258
/frameworks/av/services/audioflinger/Threads.cpp
|
| 886deb506ea2938cfec40fc0dd2bff072850386b |
|
13-Mar-2017 |
Glenn Kasten <gkasten@google.com> |
Merge "Re-format to decrease the maximum line length of files to 100 characters"
|
| d3bb645f0b7f567b033b8664499d685f8ec10628 |
|
06-Dec-2016 |
Glenn Kasten <gkasten@google.com> |
Re-format to decrease the maximum line length of files to 100 characters
Test: compiles OK
Change-Id: Ibe663032cd390ed2bcca6dc921d47732e6e15e21
/frameworks/av/services/audioflinger/Threads.cpp
|
| 1e039394f351ef77592689f32294a1c73840ad0e |
|
13-Mar-2017 |
Glenn Kasten <gkasten@google.com> |
Merge "Rename audio_utils/conversion.h to mono_blend.h"
|
| 6bf707f4c0752643757a7389384817da41d34fab |
|
24-Feb-2017 |
Glenn Kasten <gkasten@google.com> |
Rename audio_utils/conversion.h to mono_blend.h
There is only one function declared in that header file, and it is
mono_blend, so a more specific filename may be easier to understand.
Test: compiles OK
Change-Id: Ie2aec6c706b1822581b34d3702818b8200a1db8a
/frameworks/av/services/audioflinger/Threads.cpp
|
| 89e5a2b102b484fe19dbda2a0b118edfefc70241 |
|
09-Mar-2017 |
Nicolas Roulet <nicoroulet@google.com> |
Merge "NBLog periodic merging thread"
|
| dcdfaecc1fa630a799e1fdb508a9b92da55abc36 |
|
14-Feb-2017 |
Nicolas Roulet <nicoroulet@google.com> |
NBLog periodic merging thread
Bug: 35468674
Test: no change in functionality, everything works as before
Change-Id: Id2cea243bc15767ca6803c9505bf23a18411500e
/frameworks/av/services/audioflinger/Threads.cpp
|
| 18b570146c971fe729c391bfbb869391084e623d |
|
14-Feb-2017 |
Eric Laurent <elaurent@google.com> |
audioflinger: add standby() method to MmapStreamInterface
Bug: 33398120
Test: open/start/stop/close MMAP no IRQ stream for capture and playback
Change-Id: I48ec202a71565f759c441c2a835d8c8190e76334
/frameworks/av/services/audioflinger/Threads.cpp
|
| 10cbff139360f3f642e0e3b3ccf2d463dbed22cf |
|
22-Feb-2017 |
Andy Hung <hunga@google.com> |
VolumeShaper: Enable for offload and direct tracks
Test: Play Music in offload mode
Bug: 31015569
Change-Id: I00bb59e3e6809d4682f42057b1cc083f4fa9b9d1
/frameworks/av/services/audioflinger/Threads.cpp
|
| d5973043bed640cf634a26d65dedcde5d5ab3037 |
|
17-Feb-2017 |
Nicolas Roulet <nicoroulet@google.com> |
Merge "Minor code style changes and namespace fix"
|
| 8dc323ef737babc1b1c0b0347b4b895969aee77b |
|
17-Feb-2017 |
Nicolas Roulet <nicoroulet@google.com> |
Merge "Added process name to PID log"
|
| 4da7820be451847bad698ac0f687b964d9b5d34f |
|
03-Feb-2017 |
Nicolas Roulet <nicoroulet@google.com> |
Minor code style changes and namespace fix
Bug: 35329090
Test: recompile, no change of functionality
Change-Id: Ied3ec7d95ce29896e8bf1851b08345da74bf8299
/frameworks/av/services/audioflinger/Threads.cpp
|
| c20cb50c1c14cd765874511872f087c875f43ed9 |
|
01-Feb-2017 |
Nicolas Roulet <nicoroulet@google.com> |
Added process name to PID log
Bug: 35357686
Test: included in CL
Change-Id: Ie4d3cb66a2c766ce87ebabcc9c0d3aa988702e09
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9ee35d72c4b89db99c2df0b0125373f360b337f8 |
|
16-Feb-2017 |
TreeHugger Robot <treehugger-gerrit@google.com> |
Merge "AudioFlinger: Prevent FastMixer for A2DP sink"
|
| fda69406966f8578a7827e889bff615f0144b441 |
|
15-Feb-2017 |
Andy Hung <hunga@google.com> |
AudioFlinger: Prevent FastMixer for A2DP sink
Also avoids issues with FastMixer and BT suspend.
Test: BT with Play Movies
Bug: 34335167
Bug: 35166960
Bug: 35384619
Change-Id: Ie1bfaf2d09b7cfe6b86a1d26fc1d5fdabf82c020
/frameworks/av/services/audioflinger/Threads.cpp
|
| f228531c5d01f497f5802c44545e0000ec44ab83 |
|
15-Feb-2017 |
Andy Hung <hunga@google.com> |
AudioFlinger: Do not wait for FastMixer if in cold idle.
Test: BT outgoing call
Bug: 34335167
Bug: 35166960
Change-Id: I1da0de844560bf77b4149dc30979eebc131d009e
/frameworks/av/services/audioflinger/Threads.cpp
|
| 30d48d9542fb3f85889108c1ee2daff98a4860e7 |
|
14-Feb-2017 |
Andy Hung <hunga@google.com> |
Merge "VolumeShaper: Initial implementation"
|
| 9fc8b5cd4a64ef07e84c69112461324d5c13a0b0 |
|
24-Jan-2017 |
Andy Hung <hunga@google.com> |
VolumeShaper: Initial implementation
The VolumeShaper is used to apply a volume
envelope to an AudioTrack or a MediaPlayer.
Test: CTS
Bug: 30920125
Bug: 31015569
Change-Id: I42e2f13bd6879299dc780e60d143c2d465483a44
/frameworks/av/services/audioflinger/Threads.cpp
|
| f5f4056014118f6ddd16affe853fd0605001ab40 |
|
13-Feb-2017 |
Eric Laurent <elaurent@google.com> |
Merge "audio: several concurrent capture fixes"
|
| 7863c791dba8c4e5e9591b6837d410e580be0a25 |
|
10-Feb-2017 |
Eric Laurent <elaurent@google.com> |
Merge changes from topic 'mmap_no_irq'
* changes:
Add support for mmap stream
audioflinger: define MMAP HAL Stream control interface
|
| 555530ab903504ad3586bc24cd8a4f200f5c39ae |
|
08-Feb-2017 |
Eric Laurent <elaurent@google.com> |
audio: several concurrent capture fixes
Make sure AudioRecord clients are invalidated synchronously
when closing an input stream instead of when the capture thread
exits.
Fix AudioPolicyManager::getInputForDevice() logic to make sure that
not only the first opened input encountered is considered when looking
for a compatible input to reuse for concurrent capture.
Bug: 22702906
Test: run CTS test
Change-Id: I7f69609d4ee70a37ab06753ff970b12df17d885b
/frameworks/av/services/audioflinger/Threads.cpp
|
| 83f042776b131b149f803fff6ab184ae2c4d98cd |
|
07-Feb-2017 |
Mikhail Naganov <mnaganov@google.com> |
audioflinger: Adjust priority of hal threads
HALs are prohibited from using framework binder, and there is
no equivalent scheduling policy service in hwbinder. Thus, in order
to match priorities of FastCapture / Mixer threads with their
counterparts in the HAL, it is needed to request the priority boost
from audioflinger on behalf of the HAL.
Test done to verify the priority was correctly set.
Bug: 34131400
Change-Id: If8b6b031c0fcba771fae901a5b8e7da89b3a1570
Test: check priority match between audioflinger's and hal's threads
/frameworks/av/services/audioflinger/Threads.cpp
|
| 8eacd8090e16c022dcbb11850cff83f984fe2357 |
|
03-Feb-2017 |
Nicolas Roulet <nicoroulet@google.com> |
Merge "Implemented typed, thread-specific logging system. Currently supported types are: * Strings * Integers * Floats * Timestamps * Process IDs"
|
| fe1e1449cadff4f946c33403aecc73b4b4a11e56 |
|
30-Jan-2017 |
Nicolas Roulet <nicoroulet@google.com> |
Implemented typed, thread-specific logging system. Currently supported types are:
* Strings
* Integers
* Floats
* Timestamps
* Process IDs
Added printf-like formatted logging using mentioned types.
Test: included in this CL
Bug: 29421410
Change-Id: Ie07b78d8d39c32fcc8a122ffa9b1b7082800b990
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9bb03b4e10bfd26247eeb1865e40bb0645489816 |
|
14-Jan-2017 |
TreeHugger Robot <treehugger-gerrit@google.com> |
Merge "audioflinger: Fix effect creation handling"
|
| 2247f7b84bf0ce3cc9c909ef987eedb086e3b4e8 |
|
13-Jan-2017 |
Mikhail Naganov <mnaganov@google.com> |
audioflinger: Fix effect creation handling
The code flow was incorrect for sure. The 'if' branch is chosen
if 'effect == 0', and 'effectRegistered' is set to 'true' before
the effect is even created.
Test: audioflinger doesn't crash if effect failed to get created
Change-Id: I00d5f28e8b96acd765867e212c5b8193e21f9b4a
/frameworks/av/services/audioflinger/Threads.cpp
|
| 8981605d43e24c46d395acb5f145b99589d45917 |
|
12-Jan-2017 |
Andy Hung <hunga@google.com> |
AudioFlinger: Move RecordBufferConverter to libaudioprocessing
Test: Recording loopback
Bug: 31015569
Change-Id: I7897d959f36ac7424544e35f47576c99a442dd54
/frameworks/av/services/audioflinger/Threads.cpp
|
| 6acd1d432f526ae9a055ddaece28bf93b474a776 |
|
04-Jan-2017 |
Eric Laurent <elaurent@google.com> |
Add support for mmap stream
Add support for MMAP streams created by Oboe service.
- A new audio thread class MmapThread deriving from ThreadBase
is added to manage permissions, volume, routing and wakelocks for activity
on MMAP streams.
- Requests received over MmapStreamInterface to open, start and stop
a MMAP stream are forwarded to audio policy manager so that activity
on MMAP capture and playback streams is visible to audio policy and
taken into account in volume and routing management.
`
Bug: Bug: 33398120
Test: manual test
Change-Id: I957906495fe9277d2972d810b633f50cd1e14ec3
/frameworks/av/services/audioflinger/Threads.cpp
|
| 068561c8e84569d51df2adbbb53b56fdfd09c06b |
|
04-Jan-2017 |
Andy Hung <hunga@google.com> |
AudioFlinger: Split off audio processing library
Test: native AudioResampler test, general playback test
Bug: 31015569
Change-Id: Ifb248f4402a583438d756c014dcd7a4577aef713
/frameworks/av/services/audioflinger/Threads.cpp
|
| 022b9953153bdb1984f0abb17d21ef8c1826ad49 |
|
05-Jan-2017 |
Mikhail Naganov <mnaganov@google.com> |
Re-implement HIDL effect processing using FMQ and IMemory
Result: no hwbinder calls due music processing.
Test: make, use Play Music with effects, check traces
Bug: 30222631
Change-Id: I06d0e94e603688874b31824427f3b0878b5f7c8e
/frameworks/av/services/audioflinger/Threads.cpp
|
| e1c4b5d7a94c21b8ce0c5707b4af84de596fbb79 |
|
22-Dec-2016 |
Mikhail Naganov <mnaganov@google.com> |
Re-implement HIDL stream read and write using FMQ
Result: no hwbinder calls due read / write session.
Test: make, perform Loopback RTT, check traces
Bug: 30222631
Change-Id: I4a8792525ec374111302cfd5c0a2e41f9f4cc418
/frameworks/av/services/audioflinger/Threads.cpp
|
| d01b0f18491c355d808a57cb272404480e69618f |
|
08-Nov-2016 |
Andy Hung <hunga@google.com> |
AudioFlinger: update to use uid_t
Test: Power manager shows correct uids for audio
Bug: 32713790
Change-Id: I60134f60631ea00886efc912ba298443aadd3c00
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5b4051afc26367d7accd06580c5135c8295759f9 |
|
20-Dec-2016 |
Andy Hung <hunga@google.com> |
Merge "AudioFlinger: Associate audio time with client uid"
|
| dae27707fc7d8370eb200d25d1a7c6dd7ad5e201 |
|
31-Oct-2016 |
Andy Hung <hunga@google.com> |
AudioFlinger: Associate audio time with client uid
Group active track operations for add and remove together.
PlaybackThread now uses strong pointer for active tracks.
Test: Play Music, Youtube, Batterystats, CTS AudioTrack
Bug: 32361950
Change-Id: I101df081dd8d090560a83c44c2fa9ffcbf39c84d
/frameworks/av/services/audioflinger/Threads.cpp
|
| 20b9ef0b55c9150ae11057ab997ae61be2d496ef |
|
05-Dec-2016 |
Eric Laurent <elaurent@google.com> |
Add unique audio port IDs to AudioTrack and AudioRecord
This will allow to track activity at the track level instead of
at audio session level as only possible with current implementation.
AudioTracks and AudioRecords will receive a unique audio port ID the
first time they register to audio policy with
getOutputForAttr()/getInputForAttr() and keep this ID for their
lifetime.
This CL is the first partial change and just updates the
audio policy and audio flinger APIs used at track creation time.
Test: basic regression test of audio playback and capture use cases
Change-Id: I8d612e67738e120494f61e3f7c60bfd0b2c6a329
/frameworks/av/services/audioflinger/Threads.cpp
|
| abf6ff26df459d991cdbc2dca3b78046c97469db |
|
03-Dec-2016 |
Andy Hung <hunga@google.com> |
Merge "Log audio information to ensure complete delivery"
|
| 0d5a2ed0a05a2bf337c68edb54f24c60e18032c1 |
|
02-Dec-2016 |
Eric Laurent <elaurent@google.com> |
improve audio effect framwework thread safety
- Reorganize handle effect creation code to make sure the effect engine
is created with both thread and effect chain mutex held.
- Reorganize handle disconnect code to make sure the effect engine
is released with both thread and effect chain mutex held.
- Protect IEffect interface methods in EffectHande with a Mutex.
- Only pin effect if the session was acquired first.
- Do not use strong pointer to EffectModule in EffectHandles:
only the EffectChain has a single strong reference to the EffectModule.
Test: Tested with EffectTest app. Regression tests in main audio use cases.
Bug: 32707507
Change-Id: Ia1098cba2cd32cc2d1c9dfdff4adc2388dfed80e
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2148bf0e79c436b8764b9edc4c8f2730cce98a32 |
|
29-Nov-2016 |
Andy Hung <hunga@google.com> |
Log audio information to ensure complete delivery
Test: Audio playback
Bug: 30572472
Change-Id: Ibad6fc202692cd3480ae726627252afdead083f3
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9cab746b116b8aed38ca6b97bfea35103535e522 |
|
10-Nov-2016 |
Eric Laurent <elaurent@google.com> |
Revert "AudioFlinger: Associate audio time with client uid"
This reverts commit 2f366df67c31119bb6dd726becd32d14b18e6573.
Bug: 32728805
Change-Id: If0a158154814492eda427de0af8a49c0b0a91180
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5850c4c8eecbe0db3cee8511a4f82cb443e27d08 |
|
10-Nov-2016 |
Eric Laurent <elaurent@google.com> |
Revert "AudioFlinger: Remove naked track pointers"
This reverts commit 062bfceaefab29f0f71db7d5a248b3f5f0572b6a.
Bug: 32728805
/frameworks/av/services/audioflinger/Threads.cpp
|
| 77109420fe285b4e9618bd5f0cd12e0fcef6eced |
|
10-Nov-2016 |
Andy Hung <hunga@google.com> |
Merge "Audio: Use uid_t for uids"
|
| 1f12a8ad958344c50733b948628ffa06db9c5bc6 |
|
08-Nov-2016 |
Andy Hung <hunga@google.com> |
Audio: Use uid_t for uids
Test: Power manager shows correct uids for audio
Bug: 32713790
Change-Id: If5337e17283268f74dc0f00cc66ece9153d680ef
/frameworks/av/services/audioflinger/Threads.cpp
|
| 0871d93cbadad8d736a09454b47846e09a793504 |
|
08-Nov-2016 |
Andy Hung <hunga@google.com> |
Merge "AudioFlinger: Remove naked track pointers"
|
| 19b396db94fa9dc3dfe713e8687dad4804e06327 |
|
08-Nov-2016 |
Andy Hung <hunga@google.com> |
Merge "AudioFlinger: Associate audio time with client uid"
|
| 062bfceaefab29f0f71db7d5a248b3f5f0572b6a |
|
01-Nov-2016 |
Andy Hung <hunga@google.com> |
AudioFlinger: Remove naked track pointers
Test: Play Music offloaded and kill audioserver
Change-Id: I3114a67f248e067e0b5334d76e4b5e6c39858040
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2f366df67c31119bb6dd726becd32d14b18e6573 |
|
31-Oct-2016 |
Andy Hung <hunga@google.com> |
AudioFlinger: Associate audio time with client uid
Group active track operations for add and remove together.
PlaybackThread now uses strong pointer for active tracks.
Test: Play Music, Youtube, Batterystats, CTS AudioTrack
Bug: 32361950
Change-Id: Ica96e2b84b9dfa5766571e24de49401a81a30e9d
/frameworks/av/services/audioflinger/Threads.cpp
|
| 913d06c099bd689375483a839e11057ccf284d1c |
|
01-Nov-2016 |
Mikhail Naganov <mnaganov@google.com> |
Move TypeConverter into a shared library
This will be needed for the default implementation of the audio HAL
in TREBLE for parsing supported formats etc. provided by HAL in
a form of string literals.
As a bonus, remove some hand-written type conversions in AudioFlinger
used in dumps.
Example changes in the dump output:
HAL format: 0x1 (pcm16) ==> HAL format: 0x1 (AUDIO_FORMAT_PCM_16_BIT)
Processing format: 0x5 (pcmfloat) ==> Processing format: 0x5 (AUDIO_FORMAT_PCM_FLOAT)
Output device: 0x2 (SPEAKER) ==> Output device: 0x2 (AUDIO_DEVICE_OUT_SPEAKER)
Input device: 0 (NONE) ==> Input device: 0 (AUDIO_DEVICE_NONE)
AudioStreamOut: 0x... flags 0x6 (PRIMARY|FAST) ==>
AudioStreamOut: 0x... flags 0x6 (AUDIO_OUTPUT_FLAG_PRIMARY|AUDIO_OUTPUT_FLAG_FAST)
Test: make & run
Change-Id: I9cde640e6827b7aa6d62e9caade9e738227e299f
/frameworks/av/services/audioflinger/Threads.cpp
|
| f2ae3e19080938db8cbf29a963fd744a3964fcc2 |
|
28-Oct-2016 |
Wei Jia <wjia@google.com> |
BatteryNotifier: attribute battery usage of video to correct uid.
Test: manual check batterystats
Bug: 32361950
Change-Id: Id052220e87f6667ca908d5bf6be6164dc8c14c53
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9ee0540d3a61bff03d561ca431a371c3d9335d2b |
|
14-Oct-2016 |
Mikhail Naganov <mnaganov@google.com> |
Remove last references to hardware/audio.h
DeviceHalInterface transitioned to "capabilities" model
(similar to the one already used by streams, e.g. 'supportsDrain').
No direct checking of the HAL version is needed.
AudioPolicy uses its own version read from the configuration,
and these values never checked against the actual HAL version,
thus it does not need versions and macroses from hardware/*.
Test: make & run on N6P
Change-Id: Ic4a56bfa19a9a61edac2b9f9a163fd8f63a0ff87
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9fe94012187a29eeeca2c74e75f121192560fba0 |
|
14-Oct-2016 |
Mikhail Naganov <mnaganov@google.com> |
Split audio effects headers
There can be different use cases with the code that imports
headers from audio_effects (each subsequent case is a more
specialized version of the previous one):
1. The code only needs access to general structures
like effect_descriptor_t:
==> include system/audio_effect.h
Example: AudioSystem in libmedia
2. The code that needs access to effect UUID and parameter
enums:
==> include system/audio_effects/effect_foo.h
Examples: media framework effects interfaces, libwilhelm
3. The code that needs to call into HAL:
==> include hardware/audio_effect.h
Examples: libaudiohal (implementation part) and libeffects
4. The code that implements a particular effect:
==> include audio_effects/effect_foo.h
Examples: cts/.../CTSTestEffect.cpp
stuff in hardware/, vendor/
Note that AudioPolicy still deals with the effects HAL directly,
this needs to be fixed.
Change-Id: I963335ede6a196f9225caf21e1a01d03635e363c
Test: make
/frameworks/av/services/audioflinger/Threads.cpp
|
| 00260b5e6996b0a4b12f71c5b84e44adea040534 |
|
13-Oct-2016 |
Mikhail Naganov <mnaganov@google.com> |
Remove dependencies on hardware/audio_effects.h and EffectsFactoryApi.h
Also remove some outstanding dependencies on hardware/audio.h.
Most of the time, only constants and structs from audio_effects.h
are used, and no actual calls into HAL are made. Move these
definitions into system/audio_effect.h, and generate constants
from TREBLE HAL definition.
For parameters, always use values from AudioParameter class.
There are still some outstanding includes of hardware/audio.h
in code that directly deals with audio modules. Need to extend
libaudiohal interfaces to support required functionality.
Test: make
Change-Id: I0bcdf8ae199485d4d65f3eeb7e7a3b9b5cdb6ab6
/frameworks/av/services/audioflinger/Threads.cpp
|
| 473de569eed6438ee3eea955efe309900122f051 |
|
12-Oct-2016 |
Andy Hung <hunga@google.com> |
AudioFlinger: Improve effect compatibility with RAW and FAST am: d3bb0adfe4 am: 4a7ef5c1d7
am: 5bc5b90bbe
Change-Id: I681d68a632a527983972df160ba5dab1c50967cc
|
| 5bc5b90bbe3eae27c6de5a556eda531d4430dfab |
|
12-Oct-2016 |
Andy Hung <hunga@google.com> |
AudioFlinger: Improve effect compatibility with RAW and FAST am: d3bb0adfe4
am: 4a7ef5c1d7
Change-Id: Ia7d516d957b0582ec6776d4306019e6f2dfc10c0
|
| d3bb0adfe40e60bb94e9a87e8d5bffb4ea6863b0 |
|
12-Oct-2016 |
Andy Hung <hunga@google.com> |
AudioFlinger: Improve effect compatibility with RAW and FAST
Effects without process function now are allowed with RAW
and FAST playback and record tracks.
Bug: 32053844
Change-Id: I58064eb2c357043c2da1a781a20988f42570d97e
/frameworks/av/services/audioflinger/Threads.cpp
|
| cbc8f617c1aebef5d041fa40dcd38a5466690b99 |
|
12-Oct-2016 |
Mikhail Naganov <mnaganov@google.com> |
Eliminate dependencies on hardware/audio.h (trivial cases)
frameworks/av must not depend on hardware/ (except for the code
from libaudiohal that actually calls into HAL).
This CL deals with simple cases where depending on system/audio.h
is enough.
Change-Id: Ia2cb66cc8c92316ce5ab884a008d5e531263c2e4
Test: make
/frameworks/av/services/audioflinger/Threads.cpp
|
| a0c91339814f37ea78365afb436c9f3d1f0a0090 |
|
19-Sep-2016 |
Mikhail Naganov <mnaganov@google.com> |
Update NBAIO to use the new audio HAL abstraction layer
Moved the HAL access abstraction layer to a separate library so it
can be used both by audioflinger and libnbaio.
Bug: 30222631
Test: manual with Loopback app, Hangouts, YouTube
Change-Id: Id622c2f1aa8f55a775d34f369a596c2c4d29d5be
/frameworks/av/services/audioflinger/Threads.cpp
|
| 711ee66377eb053cd7a3467ca519239ebb7bf300 |
|
25-Sep-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: limit max number of tracks per client am: ad7dd9610b am: 6cfa8fdc9f
am: 5c24d28bfc
Change-Id: I3653bed5e0a0ef1476aa4409b256a8a81c6bc3fa
|
| 5c24d28bfc72423ff5bb6de3d46fe4747d9b64b1 |
|
23-Sep-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: limit max number of tracks per client am: ad7dd9610b
am: 6cfa8fdc9f
Change-Id: I1952387cd010a11ce9512a8b56a363cc4bb7a078
|
| 6cfa8fdc9f28dabe98c44e7317330f06f7a88813 |
|
23-Sep-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: limit max number of tracks per client
am: ad7dd9610b
Change-Id: I0dd67dfc1324cd4969561370faf105bcf7286949
|
| 7bafda7a153cda393ce018241fa7304845225ae5 |
|
23-Sep-2016 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: limit max number of tracks per client" into nyc-mr1-dev
|
| 39ed2aa31b0c6c8e485660c2aedff74f10d14411 |
|
23-Sep-2016 |
Andy Hung <hunga@google.com> |
Add logging of written frame counts am: 2c45393d84
am: 6353f121d7
Change-Id: I3f569f4587c09f9f3f85d2466a3c742e2720b529
|
| ad7dd9610b6fafa81baf69607f4ac669da88182a |
|
22-Sep-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: limit max number of tracks per client
Limit max number of audio tracks for a given application to
limit the chance of denial of service by misbehaving apps
not releasing tracks.
Bug: 7007023
Change-Id: Id7ac37450aabdbeaa5f83cb6e4b2a2b0c749f99c
/frameworks/av/services/audioflinger/Threads.cpp
|
| b54c854d1ef1bb66e093c94099f915178eac570e |
|
21-Sep-2016 |
Andy Hung <hunga@google.com> |
Add logging of written frame counts
Bug: 31591209
Change-Id: Ife5917aa65b02c3b0791dfe1975c70ccf58f9a75
(cherry picked from commit 2c45393d842a5d2968ef78fa7a08667efdf15f2a)
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2c45393d842a5d2968ef78fa7a08667efdf15f2a |
|
21-Sep-2016 |
Andy Hung <hunga@google.com> |
Add logging of written frame counts
Bug: 31591209
Change-Id: Ife5917aa65b02c3b0791dfe1975c70ccf58f9a75
/frameworks/av/services/audioflinger/Threads.cpp
|
| 1dc98674f701dada94143b4d31b7221c58346c6c |
|
19-Aug-2016 |
Mikhail Naganov <mnaganov@google.com> |
Abstract away access to audio streams HAL in AudioFlinger
In this CL all direct access to audio_stream_t, audio_stream_out_t, and
audio_stream_in_t their functions is encapsulated within the new
hierarchy of Stream[In|Out]HalLocal classes. AudioFlinger uses
interface classes Stream[In|Out]HalInterface to access these functions.
Note that NBAIO still receives raw HAL stream handles and needs to be
converted separately.
Bug: 30222631
Test: manual with Loopback app
Change-Id: I6388cfa2006791c9c0aa7bb186719209726a2d48
/frameworks/av/services/audioflinger/Threads.cpp
|
| db0fd69d5dd2cb70009884458d5c78c9e7256c15 |
|
16-Sep-2016 |
Eric Laurent <elaurent@google.com> |
Merge commit '77b44035440f4dfc1c8d29da7d800f111ebe6376' into fix_merge
Change-Id: I0f1d2b605a8d90f86f8fa27cbf971018981c6c38
|
| 77b44035440f4dfc1c8d29da7d800f111ebe6376 |
|
16-Sep-2016 |
Eric Laurent <elaurent@google.com> |
audio flinger: add support for effects without process function am: 6dd0fd92d6
am: 79c3335e35
Change-Id: I719e0379c19d6e0e4e50597726f83b3a1d0b3cd8
|
| 6dd0fd92d6cdeb2cf5b7127c0e880e5eacfd4574 |
|
15-Sep-2016 |
Eric Laurent <elaurent@google.com> |
audio flinger: add support for effects without process function
Add support for effects indicating they do not implement a process
function and do not consume CPU or add latency.
Enable those effects on RAW and FAST outputs and inputs.
Do not call the process function.
Bug: 31491112.
Change-Id: If020c8bb3b180568dec5138b087edec8c2524182
/frameworks/av/services/audioflinger/Threads.cpp
|
| 1f12a380bef8aa162dec458cc7849e886815aeaa |
|
14-Sep-2016 |
Mikhail Naganov <mnaganov@google.com> |
Merge "Fix RecordThread running too often if fast capture is active"
|
| e4f1f63a2c54ee8687ad8cca18df0f6639ad7c81 |
|
31-Aug-2016 |
Mikhail Naganov <mnaganov@google.com> |
Abstract away access to audio devices HAL
In this CL all direct access to audio_hw_device_t and its functions is
encapsulated within the new class DeviceHalLocal. Loading of hardware
modules is encapsulated withing DevicesFactoryHalLocal. AudioFlinger
uses interface classes DevicesFactoryHalInterface and DeviceHalInterface
to access these functions.
Bug: 30222631
Change-Id: Ic88b20c55813a24b898f4a832e082c17d81935b7
/frameworks/av/services/audioflinger/Threads.cpp
|
| 1b291841a58ce6b2b291232dfdd56134b2185c48 |
|
18-Jul-2016 |
Glenn Kasten <gkasten@google.com> |
Fix RecordThread running too often if fast capture is active
Bug: 30159297
Change-Id: Ieca804450168391c162fc4862398586fcc5d7e4e
/frameworks/av/services/audioflinger/Threads.cpp
|
| 18e6e46cfa616c34095e269505121f5033312924 |
|
30-Aug-2016 |
Andy Hung <hunga@google.com> |
Do not park the FastMixer on AudioPatch changes am: f60abceac7 am: 927fefa85b
am: 85cda4dc02
Change-Id: Iabc962fec667ad2f77dfb2721cccb05e00adb2b2
|
| 85cda4dc026ca3ce8725202738b2deb24ce94829 |
|
27-Aug-2016 |
Andy Hung <hunga@google.com> |
Do not park the FastMixer on AudioPatch changes am: f60abceac7
am: 927fefa85b
Change-Id: I175e8ebb94c97638707480b1725cea9491ea9a98
|
| f60abceac70c7a30dc2f888c4b80256b4a01a6dd |
|
26-Aug-2016 |
Andy Hung <hunga@google.com> |
Do not park the FastMixer on AudioPatch changes
Can be turned on again by property af.patch_park.
Bug: 30952714
Change-Id: Id7f7750e28ecf950d110379309d18819e169e999
/frameworks/av/services/audioflinger/Threads.cpp
|
| 4e6ff4151f03c6b7211136b83e71fca34bbc402c |
|
23-Aug-2016 |
Mikhail Naganov <mnaganov@google.com> |
audioflinger: Fix battery accounting for offload tracks am: e94c27a17f am: 7cba7beec4
am: e50af19356
Change-Id: I80e26985a69800d31b0aa9331bccdc0a91caa08b
|
| e50af19356b852781a28233d50f06b7adde7d470 |
|
23-Aug-2016 |
Mikhail Naganov <mnaganov@google.com> |
audioflinger: Fix battery accounting for offload tracks am: e94c27a17f
am: 7cba7beec4
Change-Id: Id5d5b2e5ac25a167583243d84fd060698c18930f
|
| e94c27a17fd56fd5ac63fc2de9df08cfcc80b1e3 |
|
19-Aug-2016 |
Mikhail Naganov <mnaganov@google.com> |
audioflinger: Fix battery accounting for offload tracks
Ensure that the list of wakelock UIDs is only cleared if
the offload playback wakelock is released during async completion
wait.
Bug: 30830106
Change-Id: Icf18b9e88e55c27bc03d8a904cc91f593cc95815
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5ca8f32280377dd923e72c3c6bd3994217461b8b |
|
11-Aug-2016 |
Chih-Hung Hsieh <chh@google.com> |
resolve merge conflicts of fd923e7 to stage-aosp-master am: df5d9246f9
am: 63ef5a38f8
Change-Id: I516a43c5e56323e99ad7a8ca2e4811b77c9704f4
|
| df5d9246f9607b1c2f8b134c46a05af06e206da3 |
|
11-Aug-2016 |
Chih-Hung Hsieh <chh@google.com> |
resolve merge conflicts of fd923e7 to stage-aosp-master
Change-Id: I75b44b89bae41197a1fd68362d20b8ba2b4dd192
|
| e964d4e421e2d1ca937227a580c0c837091a11e3 |
|
09-Aug-2016 |
Chih-Hung Hsieh <chh@google.com> |
Fix clang-tidy warnings in audio and playerservice.
* Add explicit keyword to conversion constructors.
Bug: 28341362
* Use const reference type for read-only parameters.
Bug: 30407689
Test: build with WITH_TIDY=1
Change-Id: I265f3b094e08d5705b506b3fbba51439c134af84
/frameworks/av/services/audioflinger/Threads.cpp
|
| 36d0ca16024820df9a12903d2ac443fabcc180bc |
|
09-Aug-2016 |
Chih-Hung Hsieh <chh@google.com> |
Fix clang-tidy warnings in audio and playerservice.
* Add explicit keyword to conversion constructors.
Bug: 28341362
* Use const reference type for read-only parameters.
Bug: 30407689
Test: build with WITH_TIDY=1
Change-Id: I265f3b094e08d5705b506b3fbba51439c134af84
Merged-In: I265f3b094e08d5705b506b3fbba51439c134af84
/frameworks/av/services/audioflinger/Threads.cpp
|
| b7809a709886c5dc58e737cd3c356579a088dc65 |
|
02-Aug-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix OutputTrack initial check am: af3ec7c9a0 am: 5271857554
am: 54efaa90a1
Change-Id: I7c4f4db022099abbcd20cdfb0971efa2d46998f6
|
| 54efaa90a110e50b1b9a5e4badaf30d494be56a0 |
|
02-Aug-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix OutputTrack initial check am: af3ec7c9a0
am: 5271857554
Change-Id: Icf61ce31fb83cc5796be5122c65b483437b17365
|
| af3ec7c9a0cfc483017d4457e0e3d77de4b1a33a |
|
01-Aug-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix OutputTrack initial check
Call initCheck() on newly created OutputTrack in
DuplicatingThread::addOutputTrack() to make sure that
the track is successfully attached to the AudioMixer.
Also make invalid track name always fatal in AudioMixer::deleteTrackName().
Bug: 29953737
Change-Id: I47f2517bd53de940e125391de99c40611ad0fc9d
/frameworks/av/services/audioflinger/Threads.cpp
|
| 57ef37350cd88ded7b83a2f4362fcbaab36f9372 |
|
29-Jul-2016 |
Andy Hung <hunga@google.com> |
Merge "Flush FastCapture PipeReader when starting after stop"
|
| f2c7f0043f13a0317e343082d3dc2913d3699f70 |
|
29-Jul-2016 |
Andy Hung <hunga@google.com> |
Merge \\\"AudioFlinger: Account for frames dropped in suspend mode\\\" into nyc-mr1-dev am: f769ab76d1 am: 4166015417
am: 648bc951b8
Change-Id: I23516b04456b3d8f49c44ca42f4e62bd8dc134e4
|
| 6d988186ffa998a1e28251b3e655cfcbccca9db5 |
|
29-Jul-2016 |
Andy Hung <hunga@google.com> |
AudioFlinger: Account for frames dropped in suspend mode am: 238fa3deff am: ef0eed3d26
am: 5c42082da7
Change-Id: I9c51ed42c901c054fd25e7629725655d763ff390
|
| 648bc951b86860dd11463340029eaef4b4d90003 |
|
29-Jul-2016 |
Andy Hung <hunga@google.com> |
Merge \\"AudioFlinger: Account for frames dropped in suspend mode\\" into nyc-mr1-dev am: f769ab76d1
am: 4166015417
Change-Id: Ied8c294777ddf380876197fe620c8931e681deb5
|
| 5c42082da799fa6469321e621216922444678900 |
|
29-Jul-2016 |
Andy Hung <hunga@google.com> |
AudioFlinger: Account for frames dropped in suspend mode am: 238fa3deff
am: ef0eed3d26
Change-Id: I0790c3ac10654142856c9e252277fc3c7880b7c4
|
| 416601541797b132ee8abe1e1d15d773ad975ce5 |
|
29-Jul-2016 |
Andy Hung <hunga@google.com> |
AudioFlinger: Prevent offload underrun during active playback
am: f804475807
Change-Id: Iefc9c73590b5aede7ef3ee1a2eaf728bdd01113e
|
| f929f9b203ef4787d45f8b24aab053aaab993be9 |
|
29-Jul-2016 |
Andy Hung <hunga@google.com> |
Merge "AudioFlinger: Prevent offload underrun during active playback" into nyc-mr1-dev
|
| 238fa3deff26d7e4d9e81bd0a88c936f16226c4e |
|
28-Jul-2016 |
Andy Hung <hunga@google.com> |
AudioFlinger: Account for frames dropped in suspend mode
These frames must be added to the kernel position in timestamps.
Bug: 30025777
Change-Id: I30c14503c9467059c3782e5c96c01ae1cf3709f6
/frameworks/av/services/audioflinger/Threads.cpp
|
| f804475807407442d5596ab7378ed07d50664063 |
|
27-Jul-2016 |
Andy Hung <hunga@google.com> |
AudioFlinger: Prevent offload underrun during active playback
Bug: 29897830
Change-Id: Id99d416cdaa8bf62daca8f92e3564f2895f15490
/frameworks/av/services/audioflinger/Threads.cpp
|
| ad6d52d38917069e3d8e68cec1625855b9d96200 |
|
18-Jul-2016 |
Andy Hung <hunga@google.com> |
Flush FastCapture PipeReader when starting after stop
Discards stale frames.
Bug: 30199985
Change-Id: Ie93a3784bf052aba6989d2ff1be92b1980b0c207
/frameworks/av/services/audioflinger/Threads.cpp
|
| b566e2ed097985da01d4218a3216b45049b25767 |
|
20-Jul-2016 |
Andy Hung <hunga@google.com> |
Merge \\"Only read timestamps from HAL if not FastCapture\\" into nyc-mr1-dev am: b5aac433cd
am: cbbb9a50d3
Change-Id: Ic983502e4b1802208deb3bd85cd537caa45314c9
|
| cbbb9a50d30360794724a0eecc10fff726706cec |
|
20-Jul-2016 |
Andy Hung <hunga@google.com> |
Merge \"Only read timestamps from HAL if not FastCapture\" into nyc-mr1-dev
am: b5aac433cd
Change-Id: I35220717ab32936421a8dc906c4af2cac979a5f5
|
| b5aac433cda041c1fde2fcfadf1835f3f8d19eeb |
|
20-Jul-2016 |
Andy Hung <hunga@google.com> |
Merge "Only read timestamps from HAL if not FastCapture" into nyc-mr1-dev
|
| 69ce44d164c7b9963c06cb96ea9d25cf99e069c9 |
|
18-Jul-2016 |
Andy Hung <hunga@google.com> |
Only read timestamps from HAL if not FastCapture
Prevents blocking FastCapture from the RecordThread
Bug: 29392543
Change-Id: Icd00534677be121bc4f4e9a7a6bd5b44f6f14f50
/frameworks/av/services/audioflinger/Threads.cpp
|
| d7e04b3fe9f61fdba21116dd7c50b86979fe2497 |
|
18-Jul-2016 |
Eric Laurent <elaurent@google.com> |
Merge \\"AudioFlinger: fix offload volume on resume\\" into nyc-mr1-dev am: cecb30a22f
am: 97a480927d
Change-Id: I00f8643d420ab63ce63658aebb98794c7517958c
|
| 97a480927d83e4d79ec915a86262ac3489377bff |
|
18-Jul-2016 |
Eric Laurent <elaurent@google.com> |
Merge \"AudioFlinger: fix offload volume on resume\" into nyc-mr1-dev
am: cecb30a22f
Change-Id: I7c4165affd590a4f43d631006e02f92bc4fd0a9e
|
| cecb30a22f3657483f07f259231b81b02b2a7305 |
|
18-Jul-2016 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger: fix offload volume on resume" into nyc-mr1-dev
|
| 3df841a10b4d553949c518ad52f1af8dcb365058 |
|
16-Jul-2016 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix offload volume on resume
Make sure that new volume is sent to audio effects and HAL
immediately upon resuming playback on direct and offload threads.
Also make sure that only the latest active track on thread controls
volume.
Bug: 30057465
Change-Id: I846958b4aae5abd3740b8a2545b57bd8f3614031
/frameworks/av/services/audioflinger/Threads.cpp
|
| 8221d9e855909c8e097f5f0e986755baca4dd1b7 |
|
15-Jul-2016 |
Glenn Kasten <gkasten@google.com> |
Distinguish systrace timelines for playback and capture am: f9715e43ea
am: 78f9c3ad66
Change-Id: If66bafab6f0cd2aa1f5478b963faec0632f297ca
|
| 78f9c3ad666d832d423550ded32f3626abbe6e28 |
|
15-Jul-2016 |
Glenn Kasten <gkasten@google.com> |
Distinguish systrace timelines for playback and capture
am: f9715e43ea
Change-Id: I67d7f23998c10a554d53cea37ad443ca1fad3f6f
|
| f9715e43ea73361321663514c44129c939c5db2f |
|
13-Jul-2016 |
Glenn Kasten <gkasten@google.com> |
Distinguish systrace timelines for playback and capture
Bug: 30120068
Change-Id: If416540f0785e83e7b1e8ba986ee55b5b9298a2a
/frameworks/av/services/audioflinger/Threads.cpp
|
| 8da2d981fb78fd75f9d0d2cab9c23fb651be334d |
|
14-Jul-2016 |
Eric Laurent <elaurent@google.com> |
Merge \\"AudioFlinger: fix capture stop sequence\\" into nyc-mr1-dev am: a5bcf56619
am: d7fdb3b5c3
Change-Id: Icd9355083c77ae2e0b7fb2ed12d675105483d3dd
|
| d7fdb3b5c3f8eca33b7462cc46093752d47203c1 |
|
14-Jul-2016 |
Eric Laurent <elaurent@google.com> |
Merge \"AudioFlinger: fix capture stop sequence\" into nyc-mr1-dev
am: a5bcf56619
Change-Id: I17cde8f3113d5371e24ce1d4adafa2413752b0e3
|
| a5bcf56619a70b0b46c737a29d873bbff8fb77a4 |
|
14-Jul-2016 |
Eric Laurent <elaurent@google.com> |
Merge "AudioFlinger: fix capture stop sequence" into nyc-mr1-dev
|
| 5c25d5637ef3406883e0fc97a713f9d5cdca426f |
|
14-Jul-2016 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: fix capture stop sequence
Make sure that the HAL input stream is actually in standby when audio policy
is notified by stopInput().
There was a race condition where it was possible for a new capture stream to be started
in the interval between stopInput() notification and input stream going into
standby because of the state update and sleep sequence in the record thread loop.
Also make sure we interrupt the sleep period if a command is pending.
Bug: 29865791
Change-Id: Ife720fbd2e5d0df8792669dc724b1fd70f63ab8a
/frameworks/av/services/audioflinger/Threads.cpp
|
| 4099f1a290c4b7980681afa7ba562e6c2450380e |
|
12-Jul-2016 |
Eric Laurent <elaurent@google.com> |
Merge \\"audioflinger: async callback error handling\\" into nyc-mr1-dev am: 93540a6b59
am: 6a7f864490
Change-Id: I289e192efe71c599229a10771a08d8685a75dd57
|
| 6a7f864490766020ef791d6c803364fdb7e79cbb |
|
12-Jul-2016 |
Haynes Mathew George <hgeorge@codeaurora.org> |
Merge \"audioflinger: async callback error handling\" into nyc-mr1-dev
am: 93540a6b59
Change-Id: Ifa8e386c0da9e30b90884e0a787e581ace151c64
|
| 93540a6b596b1660a453d344573571462a7e6587 |
|
12-Jul-2016 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: async callback error handling" into nyc-mr1-dev
|
| bbe0cfb249f872fe68215fd1baa7d6426aef2d16 |
|
12-Jul-2016 |
Jean-Michel Trivi <jmtrivi@google.com> |
Merge \\"Fix non-offloadable effect creation on offload threads\\" into nyc-mr1-dev am: b28d6117a5
am: 8611d5135b
Change-Id: I038d377d8489b95f6eb119f27fd968e97bd83b14
|
| 8611d5135b6a62cc4bbf89404090c9f6ac3245f5 |
|
12-Jul-2016 |
Jean-Michel Trivi <jmtrivi@google.com> |
Merge \"Fix non-offloadable effect creation on offload threads\" into nyc-mr1-dev
am: b28d6117a5
Change-Id: I273585d967945cadddcd70d10514dabb64b1f2e3
|
| 4527b9ebc6a37b861f5a3bba68bcc63dc8d69ada |
|
08-Jul-2016 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: async callback error handling
Invalidate all tracks on receiving an async error
callback from the output stream.
Change-Id: I92f441ed7fd904b85f464a041847880faac5b383
/frameworks/av/services/audioflinger/Threads.cpp
|
| 773ee957aadbe86ee53ac96a108ce47f8e02b961 |
|
12-Jul-2016 |
Jean-Michel Trivi <jmtrivi@google.com> |
Fix non-offloadable effect creation on offload threads
Do not prevent the creation of non-offloadable effects on
offload threads.
Bug 29229872
Change-Id: I1deebb5c0973f67a9e925ef1278d22ef7f62c779
/frameworks/av/services/audioflinger/Threads.cpp
|
| d82bd98d0dd8bfa792d7e5223eb3a266d132f2b0 |
|
07-Jul-2016 |
Andy Hung <hunga@google.com> |
Merge \\"Attribute throttle time to previous loops write time\\" into nyc-mr1-dev am: bf6dc1af5b
am: 4b06aec0d4
Change-Id: I2c1362d5544c73934bc5b1b7c402d4371a09ba6f
|
| 4b06aec0d4ce89859c0376096fdffeec50c65c2f |
|
07-Jul-2016 |
Andy Hung <hunga@google.com> |
Merge \"Attribute throttle time to previous loops write time\" into nyc-mr1-dev
am: bf6dc1af5b
Change-Id: I4bc857144d49760097f7344c1d58ad7444b66548
|
| bf6dc1af5bd88135f47c2489c03cdb9f95d57927 |
|
07-Jul-2016 |
Andy Hung <hunga@google.com> |
Merge "Attribute throttle time to previous loops write time" into nyc-mr1-dev
|
| 0a31dddf67e739684f60aecc85311ff446fb68f9 |
|
07-Jul-2016 |
Andy Hung <hunga@google.com> |
Attribute throttle time to previous loops write time
Bug: 29991292
Change-Id: Ie3ecc9631e1dfa83fafc7b003495f59e981a0b6b
/frameworks/av/services/audioflinger/Threads.cpp
|
| 6217c5233ad77f8e8f22f2b54edf8b6271275f09 |
|
30-Jun-2016 |
Eric Laurent <elaurent@google.com> |
Merge \\"audioflinger: fix logcat\\" into nyc-mr1-dev am: bb3bee86d5
am: bde5acd3a3
Change-Id: Ie443e96b8231b3fff5773253e7886fe7f5c42150
|
| bde5acd3a31b19a73e6877309ce3de27b211855a |
|
30-Jun-2016 |
Eric Laurent <elaurent@google.com> |
Merge \"audioflinger: fix logcat\" into nyc-mr1-dev
am: bb3bee86d5
Change-Id: I1b5e4c626d686379c570316181c020c5e16aa378
|
| 122f7e793fe6fb8904634cc6d2e35ac4b014ea72 |
|
29-Jun-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix logcat
Fix build when verbose logging is enabled in Threads.cpp.
Remove log spam for timestamp errors.
Change-Id: I704527c60e48012c5dcc00cb0be08d14ec2ab2a6
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9ab9f6219a0274bed6a8d64169ab695ef77d29de |
|
06-May-2016 |
Andy Hung <hunga@google.com> |
Remove throttle logcat for bluetooth
Due to reduced buffer sizes for bluetooth it is possible to throttle
regularly, do not logcat throttle for bluetooth to avoid spamming.
Bug: 28286313
Change-Id: I9a572077a58c8e5cce3d5ac4de3cc6d97d94d7a0
/frameworks/av/services/audioflinger/Threads.cpp
|
| 516b743f3c19f38816ecdead90ae74bc1b3c84e4 |
|
21-Jun-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: add check for effect vs thread compatibility am: 4c415062ad
am: ad67b138fc
Change-Id: I38015a85b7df2d620daa2c9e5d7435de510a8fbe
|
| ad67b138fcec25ba99254e826c559778adf0e7f3 |
|
21-Jun-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: add check for effect vs thread compatibility
am: 4c415062ad
Change-Id: Idbdb7591b291c09ced94d4eabae0edc3581e1bcd
|
| 4c415062ad1bb53e9af8f644d8215837262b79bb |
|
18-Jun-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: add check for effect vs thread compatibility
Make sure that only HW accelerated effects are added on FAST
playback or record threads and no effect is added on threads with
RAW flag.
Make sure that tracks with RAW and FAST flags are compatible with exiting
effects on a playback or record threads and clear flags accordingly.
Bug: 4999001
Change-Id: I0c060f1ee4dfb9f1e206633536e67f7c3b7b3788
/frameworks/av/services/audioflinger/Threads.cpp
|
| 4cfa355eedae56e047b63b1b02fc6d789e21980d |
|
04-Jun-2016 |
Eric Laurent <elaurent@google.com> |
Merge "Add AudioTrack and AudioRecord flag checks" into nyc-mr1-dev am: 2f8e63b5be
am: b70868bb33
* commit 'b70868bb337f0b12f993494f4d110979a33ddd24':
Add AudioTrack and AudioRecord flag checks
Change-Id: Ia6ab9998dcaa065d18fc048ae5ea6bb3e087ba62
|
| b70868bb337f0b12f993494f4d110979a33ddd24 |
|
04-Jun-2016 |
Eric Laurent <elaurent@google.com> |
Merge "Add AudioTrack and AudioRecord flag checks" into nyc-mr1-dev
am: 2f8e63b5be
* commit '2f8e63b5be5b0160302cffe1f1f18e560d31d3e7':
Add AudioTrack and AudioRecord flag checks
Change-Id: If9043721f4763ce0f4db96c63595f92c5ba22556
|
| 050677873c10d4da308ac222f8533c96cca3207e |
|
02-Jun-2016 |
Eric Laurent <elaurent@google.com> |
Add AudioTrack and AudioRecord flag checks
Verify that the requested flags are compatible with the input
or output flags when creating and AudioRecord or AudioTrack
Get rid of IAudioFlinger::track_flags_t which was redundant
with audio_input_flags_t and audio_output_flags_t.
Change-Id: I0dd9232f857b2737e99a8c668806e45bce09cdbd
/frameworks/av/services/audioflinger/Threads.cpp
|
| 8a4348bb7a598eab9b692d782595689112df367a |
|
25-May-2016 |
Andy Hung <hunga@google.com> |
resolve merge conflicts of 5d8e380 to nyc-mr1-dev am: b09a6d0832
am: 6b0a8693e1
* commit '6b0a8693e169a97f5fd92260bcbd437b514d7691':
AudioTrack: Fix timestamp startup condition
Change-Id: Ie11abf463d4ca8b065fffc66bfb1c4db37a3ca4c
|
| 6b0a8693e169a97f5fd92260bcbd437b514d7691 |
|
25-May-2016 |
Andy Hung <hunga@google.com> |
resolve merge conflicts of 5d8e380 to nyc-mr1-dev
am: b09a6d0832
* commit 'b09a6d08321b9484d80a2a9dc11ed52623942c8d':
AudioTrack: Fix timestamp startup condition
Change-Id: Id2c883949a47daac936266310ebe4d69a5965eb3
|
| b09a6d08321b9484d80a2a9dc11ed52623942c8d |
|
25-May-2016 |
Andy Hung <hunga@google.com> |
resolve merge conflicts of 5d8e380 to nyc-mr1-dev
Change-Id: Ic3aacd191cae0749bb33fe9ccfdac743ff82a21f
|
| 8572ee135c9af5aacf74f02dd3ce8e9eb3956ec4 |
|
25-May-2016 |
Andy Hung <hunga@google.com> |
Merge "AudioTrack: Fix timestamp startup condition" into nyc-dev
am: 5d8e3808b4
* commit '5d8e3808b4012fa72bedf5898803682831f88e84':
AudioTrack: Fix timestamp startup condition
Change-Id: Ibc54738fb4beee0db7559e5b9f76910d639e966f
|
| 69488c4ef115c9de52c85f4fcae27c7774720298 |
|
17-May-2016 |
Andy Hung <hunga@google.com> |
AudioTrack: Fix timestamp startup condition
Suppress timestamps without valid kernel information or
a server update.
Update timing of server timestamp location to better indicate
when the server pull occurs.
Update write timing logic to suppress unneeded time calls
and to ensure proper throttling.
Bug: 28771691
Change-Id: I2cd25dad638ca1d0b1355d87d175d908f3bddb26
/frameworks/av/services/audioflinger/Threads.cpp
|
| 791fce20ee88597ea7cd64cc728010088a2e7a8f |
|
24-May-2016 |
Chih-Hung Hsieh <chh@google.com> |
resolve merge conflicts of 53529cf to nyc-dev-plus-aosp am: 4d1522bdc7
am: a15b56e484
* commit 'a15b56e48406d8242e91629e748a1b90d5977381':
Fix misc-macro-parentheses warnings.
Change-Id: I5fcae0f83f1f6820ec8da61c46e778e30353195f
|
| a15b56e48406d8242e91629e748a1b90d5977381 |
|
24-May-2016 |
Chih-Hung Hsieh <chh@google.com> |
resolve merge conflicts of 53529cf to nyc-dev-plus-aosp
am: 4d1522bdc7
* commit '4d1522bdc7efaa9677ac92b620cea9e9110ec2a4':
Fix misc-macro-parentheses warnings.
Change-Id: Id828c632b701e19de541fab868484ae3d4bfab45
|
| 4d1522bdc7efaa9677ac92b620cea9e9110ec2a4 |
|
24-May-2016 |
Chih-Hung Hsieh <chh@google.com> |
resolve merge conflicts of 53529cf to nyc-dev-plus-aosp
Change-Id: Iae19c31b9ce2d9c604040e54d6fbf125a3d3d362
|
| bf29173d6d63536ae9f8cfd1d02153237ee23612 |
|
18-May-2016 |
Chih-Hung Hsieh <chh@google.com> |
Fix misc-macro-parentheses warnings.
Bug: 28705665
Change-Id: I97d1dec9a4397fd7a3cab5180a833066b8abe0ab
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2ffd63e2b6762e9cbd0eff2c0fdd8297ba9e5393 |
|
17-May-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix invalidateTracks() on offload thread am: 1308462107 am: 514d8c512d
am: a043ba313b
* commit 'a043ba313b68a7e6694b17faaca7f10dd4e5f1d2':
audioflinger: fix invalidateTracks() on offload thread
Change-Id: Ica11f6f42cdfbadc63ffd665410f483da0b1b2da
|
| a043ba313b68a7e6694b17faaca7f10dd4e5f1d2 |
|
17-May-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix invalidateTracks() on offload thread am: 1308462107
am: 514d8c512d
* commit '514d8c512d08824b288fa1784b33a294b0f7e81d':
audioflinger: fix invalidateTracks() on offload thread
Change-Id: I659f616463f5df569b958e8c6178a1f1da006839
|
| 57fc788e8c924823c9026f1239282d39433da821 |
|
17-May-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix invalidateTracks() on offload thread
am: 1308462107
* commit '13084621072ea2e4c34e2a9d79793c621d9bf005':
audioflinger: fix invalidateTracks() on offload thread
Change-Id: Ib4f9306ababd292b71d4a5026bc157ef40e61120
|
| 13084621072ea2e4c34e2a9d79793c621d9bf005 |
|
17-May-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix invalidateTracks() on offload thread
Commit 05317d29 introduced a regression causing a systematic flush
on offload output thread when invalidateTracks() is called even if
no track is using the stream type specified.
Bug: 28786996
Change-Id: Iee393ee8b1093c40647235bd35f4a6fa9b62937a
/frameworks/av/services/audioflinger/Threads.cpp
|
| ea2bf087bc6d1f9e9e0e5242f0d29d1498845879 |
|
12-May-2016 |
Haynes Mathew George <hgeorge@codeaurora.org> |
Merge "audioflinger: update multiplier logic to calculate frameCount" into nyc-mr1-dev am: 64a253562a
am: b7bcb3c46f
* commit 'b7bcb3c46f0501a214329509d912908a85646d49':
audioflinger: update multiplier logic to calculate frameCount
Change-Id: I60f4d2b2493cf6f93e70624c35a29f07c82b660b
|
| c77ce274f1f8355ccddb6d8b1c5f589922573f7b |
|
12-May-2016 |
Haynes Mathew George <hgeorge@codeaurora.org> |
Merge "audioflinger: Update throttling logic in MixerThread" into nyc-mr1-dev am: 86c54273d0
am: 1e6323fa09
* commit '1e6323fa092f51fad8ee5f563780dba1d62a1f5a':
audioflinger: Update throttling logic in MixerThread
Change-Id: I85cab0203af31bd8de1f645293fc3b182d88023d
|
| ca8e6805991c0c058c3d6315556c7612d663edab |
|
12-May-2016 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: various offload playback fixes" into nyc-dev am: 8b43838c48 am: d9e1a02f39
am: 9d0956cf44
* commit '9d0956cf445b6839ebba3b81d33c8e8289dd086f':
audioflinger: various offload playback fixes
Change-Id: I3455daba5e1d154bab7cbf4a44e64388212ae07b
|
| 64a253562a0521f5f0b3205dae98367102ebe532 |
|
12-May-2016 |
Glenn Kasten <gkasten@google.com> |
Merge "audioflinger: update multiplier logic to calculate frameCount" into nyc-mr1-dev
|
| 86c54273d035ff1ee6efab1a56766f883e1c98f7 |
|
12-May-2016 |
Glenn Kasten <gkasten@google.com> |
Merge "audioflinger: Update throttling logic in MixerThread" into nyc-mr1-dev
|
| e93cc03da360a1a0d2ad937c745ce8c8e8be81c2 |
|
05-May-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: various offload playback fixes
Revert underrun sleep time optimization added for
offload by commit 51716185 which can cause music to pause for a few
seconds in some corner cases.
Allow underruns in STOPPING_1 state to avoid dropping last buffer
received by the AudioTrack callback after stop() is called by the
client.
Allow interruption of thread loop sleep if a command is pending to
speed up track start sequence.
Do not wait for a full AudioTrack buffer before writing to audio HAL
when resuming playback or transitioning to next track.
Also moved log level for underruns in AudioSink from I to D to reduce
spam on user builds.
Bug: 28545177
Bug: 27682362
Bug: 28347796
Change-Id: I05b651b7878a2d2eedcac43cd669e32add171d40
/frameworks/av/services/audioflinger/Threads.cpp
|
| 227a14be87e8c63eb7a464b857a1d29c86c90e7c |
|
09-May-2016 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: update multiplier logic to calculate frameCount
If the value of the multiplier used in calculating
mNormalFrameCount is odd, it is rounded off to a higher even value.
This results in an increase of mNormalFrameCount and thereby
the latency which is not expected.
Do not prefer an even multiplier and let the value remain as is
even if it is odd.
Bug: 25740205
authored-by: Satya Krishna Pindiproli<satyak@codeaurora.org>
Change-Id: Ia60d87d01caef6f45998bffeafc3d6a24f7c7fb4
/frameworks/av/services/audioflinger/Threads.cpp
|
| f92b2175cd4a8d0fa9aca6e660ed4e85f6b7078b |
|
09-May-2016 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: Update throttling logic in MixerThread
The throttling logic in MixerThread only keeps track of
time spent during sink write. However, on low tier
APs and with high sample rate tracks, it is possible
that time spent in mixing the tracks is significant.
This inturn leads to sink writes returning earlier than
expected kicking in (incorrectly) the throttling logic.
To fix this, update throttling logic to include
the time spent in mixing buffers along with the time
spent in sink write.
authored-by: Dhananjay Kumar <dhakumar@codeaurora.org>
Bug: 28672515
Change-Id: Iac4994c3084850bb93640c978b4a519db8d7951d
/frameworks/av/services/audioflinger/Threads.cpp
|
| c3afc78811e976a2aa8204a9b8147e9fee5868c7 |
|
10-May-2016 |
Andy Hung <hunga@google.com> |
Merge "Improve timestamp jitter fix" into nyc-dev am: 5adc73643f am: 2815769b9f
am: bd46093e09
* commit 'bd46093e09879ef207105d53d8221c24bfce5da0':
Improve timestamp jitter fix
Change-Id: I8672b654c0649429871aae697c168308a7fce0bc
|
| 6d7b119a416c9f10288051e562f294365e5d954c |
|
08-May-2016 |
Andy Hung <hunga@google.com> |
Improve timestamp jitter fix
Use last valid kernel timestamp on server side, not client side.
Bug: 28465713
Change-Id: I33590d6922980f288355f947bc56279245058429
/frameworks/av/services/audioflinger/Threads.cpp
|
| d36e5779ca922925e96a544bbb5ae790e8a5d2fe |
|
06-May-2016 |
Andy Hung <hunga@google.com> |
Merge "Remove throttle logcat for bluetooth" into nyc-dev am: 823b18ef45 am: f6ef1a972d
am: 43c8a40899
* commit '43c8a4089937865b6cdebf25eeb8551a10f23026':
Remove throttle logcat for bluetooth
Change-Id: I5be66bc7c9dc8ac609745487ad7ba9a74e556d04
|
| 823b18ef45f0593386d9d5d20fbf9a0379ad0ebb |
|
06-May-2016 |
Andy Hung <hunga@google.com> |
Merge "Remove throttle logcat for bluetooth" into nyc-dev
|
| 3ea004dd80c06c5b366e1a2547ea9b28fffc1865 |
|
06-May-2016 |
Andy Hung <hunga@google.com> |
Remove throttle logcat for bluetooth
Due to reduced buffer sizes for bluetooth it is possible to throttle
regularly, do not logcat throttle for bluetooth to avoid spamming.
Bug: 28286313
Change-Id: I9a572077a58c8e5cce3d5ac4de3cc6d97d94d7a0
/frameworks/av/services/audioflinger/Threads.cpp
|
| be3d5d5671fd2c38baa5d66bfc0775c54af06740 |
|
05-May-2016 |
Haynes Mathew George <hgeorge@codeaurora.org> |
Merge "audioflinger: set flush pending on invalidating offload track" into nyc-dev am: 0c43cb3a98 am: 08686471b0
am: 9ec791f42a
* commit '9ec791f42a68887114a67a880ba1c73e6497ae1a':
audioflinger: set flush pending on invalidating offload track
Change-Id: I671e9facbdf952a6b854a7f8c311802f8246f04b
|
| 05317d29b27e5fda654bea21b80d4423a03f49b3 |
|
04-May-2016 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: set flush pending on invalidating offload track
On invalidating an offload track, the IAudioTrack instance is
destroyed and the offload output is released. If it so happens
that APM::getOutputForAttr for the new IAudioTrack is called before
OffloadThread::prepareTracks_l checks and removes an invalid track,
the same output can get reused.
The side effect of this is data present in HAL and below from before the
invalidate will be rendered before data from the new seek position
is rendered. This is unexpected.
To fix this, set hint to issue flush when an offload track is invalidated.
Bug: 28566885
Change-Id: Ib6c38a3abb600598b87591bac90d03b7150d5216
/frameworks/av/services/audioflinger/Threads.cpp
|
| 9d9138534bb1736b662b31025b0e412273269684 |
|
29-Apr-2016 |
Glenn Kasten <gkasten@google.com> |
Merge "Can now indicate the desired number of notifications (sub-buffers)" into nyc-dev am: d0389b874d am: c03ab34de0
am: aba26a54a8
* commit 'aba26a54a8b007fafca3772082c0db5e43e16aec':
Can now indicate the desired number of notifications (sub-buffers)
Change-Id: I5650eb7f47e4daccdd068950146601c98e44ea39
|
| d0389b874d592b960bdc456c2514a36b082c2c94 |
|
29-Apr-2016 |
Glenn Kasten <gkasten@google.com> |
Merge "Can now indicate the desired number of notifications (sub-buffers)" into nyc-dev
|
| c7285abd573c046250c3bf03b5720d3a2bf55eb0 |
|
29-Apr-2016 |
Glenn Kasten <gkasten@google.com> |
Make max fast tracks configurable using a property am: dc2c50bad4 am: 3b2d92e2ec
am: 3b5f5f51b5
* commit '3b5f5f51b51439b60ac46ec07bb6bb03525a908e':
Make max fast tracks configurable using a property
Change-Id: Ie3a015ffd5638cf4ea353ad14863ff1a09858daf
|
| ea38ee7742e799b23bd8675f5801ef72f94de0f4 |
|
18-Apr-2016 |
Glenn Kasten <gkasten@google.com> |
Can now indicate the desired number of notifications (sub-buffers)
per fast track buffer when using the default buffer size.
There is no change for normal tracks, or fast tracks with a non-default buffer size.
Also fix related bugs:
Notification period was not set correctly when fast multiplier is not 1.
Incorrect warning when client adjusted notification frames.
Bug: 27819623
Bug: 28117362
Change-Id: Ifc2121e874f51718cd56ec04e4bd5f89f4963132
/frameworks/av/services/audioflinger/Threads.cpp
|
| dc2c50bad491d2c0c8a2efc0e24491076701c63c |
|
21-Apr-2016 |
Glenn Kasten <gkasten@google.com> |
Make max fast tracks configurable using a property
ro.audio.max_fast_tracks
Rename the currently configured maximum number of fast tracks
from FastMixerState::kMaxFastTracks to FastMixerState::sMaxFastTracks.
There is no guarantee that the CPU will be able to handle
the configured number of fast tracks.
Bug: 27564141
Change-Id: If9af226d839b226503488c3cb20a4bb8950b429d
/frameworks/av/services/audioflinger/Threads.cpp
|
| 6a3b25f04f04ec9bb4133042af431a810dc70868 |
|
20-Apr-2016 |
Glenn Kasten <gkasten@google.com> |
Merge "Add AudioSystem::getFrameCountHAL()" into nyc-dev am: eb7f394004 am: 5fb71f6247
am: b137e82406
* commit 'b137e8240624e363951bcd3a42cb5d68b7622513':
Add AudioSystem::getFrameCountHAL()
Change-Id: Iab15db51b03e80e39d6711d46967995fb796a2f3
|
| 4a8308b11b92e608cdaf29f73f7919e75706f9a2 |
|
18-Apr-2016 |
Glenn Kasten <gkasten@google.com> |
Add AudioSystem::getFrameCountHAL()
And add comments about declaring methods in binder opcode order.
Bug: 28117362
Change-Id: I3c4426fa4bb3ce9c4a207a44d3bb1103d7fef160
/frameworks/av/services/audioflinger/Threads.cpp
|
| 8c797481a10ada8b678fd831feebcf68c787dacc |
|
12-Apr-2016 |
Glenn Kasten <gkasten@google.com> |
Fix audio tee sink debugging tool am: fc302fd
am: aaf2475
* commit 'aaf2475680709d665a194832f2713ced15ac7cd3':
Fix audio tee sink debugging tool
Change-Id: I844ff35140bf17567192fddb03a11250fd69d545
|
| fc302fd9e2d2cbafef771de77c47799488e1c044 |
|
11-Apr-2016 |
Glenn Kasten <gkasten@google.com> |
Fix audio tee sink debugging tool
Bug: 28119692
Change-Id: Ieb7b41754fab0111602dd29b63b0fd8c9c4b4d2d
/frameworks/av/services/audioflinger/Threads.cpp
|
| 0df20cfb9fdbf756c9339c2fad610a179754e021 |
|
02-Apr-2016 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: fix offload underrun sleep logic" into nyc-dev am: eebc3ad
am: 707e11c
* commit '707e11cfd5aed5b035ba6ba0954de1c2b4a820c2':
audioflinger: fix offload underrun sleep logic
Change-Id: I713eb483d25a0ce0c46ca9bdf78217294473f589
|
| eebc3ad457d5b9a1acc0fe0393b770004016a014 |
|
02-Apr-2016 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: fix offload underrun sleep logic" into nyc-dev
|
| 3eaf66b860f9a0d8af0dd4d5ac6adb5b67d7b73a |
|
01-Apr-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix offload underrun sleep logic
Do not sleep in case of underrun if not enough data has been
sent to the audio HAL since last flush or standby.
Bug: 27898813
Change-Id: Ic44bed97c7397ec1156d4fe9026c4fa9f00d651c
/frameworks/av/services/audioflinger/Threads.cpp
|
| 356e0b99303c04b43e609755b683e1a03225078c |
|
01-Apr-2016 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: keep wakelock during offload playback" into nyc-dev am: c526e1e
am: 1719e8f
* commit '1719e8ffa0b44e72b3d44a98f0340613949bc04a':
audioflinger: keep wakelock during offload playback
Change-Id: I4bfe4dff0cf363d48ead2335bff1cfd3a677447b
|
| c526e1ef36f8f0520a059c6cf304e12e86170c26 |
|
01-Apr-2016 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: keep wakelock during offload playback" into nyc-dev
|
| 3a5ea10722790d961237a68f1704b33845664c10 |
|
01-Apr-2016 |
Jean-Michel Trivi <jmtrivi@google.com> |
Merge "Ignore ro.audio.silent property when playing on REMOTE_SUBMIX" into nyc-dev am: 601b5a5
am: 1a15a10
* commit '1a15a1048c8a99ae812bc2d0bd617ead8d924387':
Ignore ro.audio.silent property when playing on REMOTE_SUBMIX
Change-Id: Ic5ab1ea7923ebef344a72098db7329f9be24a0f0
|
| 32f37c22f60b7a1a6ccfa351700f80c03918d4fe |
|
01-Apr-2016 |
Jean-Michel Trivi <jmtrivi@google.com> |
Ignore ro.audio.silent property when playing on REMOTE_SUBMIX
Ignore the effect of the ro.audio.silent property for playback
threads playing on AUDIO_DEVICE_OUT_REMOTE_SUBMIX as playing
in those mixes doesn't make a sound.
Bug 27336527
Change-Id: I933cfed29cd72636fd33282e5f2ad05693688b83
/frameworks/av/services/audioflinger/Threads.cpp
|
| 646679779a8f952980a5d4219ad9c6f93efc4b92 |
|
31-Mar-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: keep wakelock during offload playback
Add a system property ro.audio.offload_wakelock to select
if a wakelock should be kept during offload playback while waiting for
write completion callback.
The default is yes, keep the wakelock.
On some platforms, the cost of entering or exiting suspend is so high
that is it better to not release a wakelock at all rather than releasing
it periodically.
Bug: 26208297
Change-Id: Ib41c7e4ed9c8c0c063363eb9fe99a2ecc81b521b
/frameworks/av/services/audioflinger/Threads.cpp
|
| 0e8d86c5a0eb6ded973649795d3b0e0c75870db5 |
|
29-Mar-2016 |
Andy Hung <hunga@google.com> |
Merge "Fix FastTrack creation buffer count irregularity" into nyc-dev am: 0be3be0
am: 994ea26
* commit '994ea268043e476c1f4e5dbd57d55d6114f59eb8':
Fix FastTrack creation buffer count irregularity
|
| 0be3be0cf5530d6faf655c325ec9d94a2bd53564 |
|
29-Mar-2016 |
Andy Hung <hunga@google.com> |
Merge "Fix FastTrack creation buffer count irregularity" into nyc-dev
|
| fda6504ae3ee3ebbe1b85c7e249dd771d5962a7c |
|
25-Mar-2016 |
Haynes Mathew George <hgeorge@codeaurora.org> |
Park FastCapture in HOT_IDLE while processing new parameters am: 9ce67b5
am: 9f9d824
* commit '9f9d8241f72ff01072561ead51b2b0e3d5ce6475':
Park FastCapture in HOT_IDLE while processing new parameters
|
| 9ce67b524236db52eadf558e7765775acad89f1f |
|
30-Sep-2015 |
Haynes Mathew George <hgeorge@codeaurora.org> |
Park FastCapture in HOT_IDLE while processing new parameters
Without this, FastCapture can starve RecordThread delaying
handling of new parameters.
Bug: 24667895
Change-Id: Idb7113b7b0f96f2c35c655b6fdb3279f119ad1a3
/frameworks/av/services/audioflinger/Threads.cpp
|
| ac4dacc0ba7384f86ccfa2feb8f0d78d96d42dc8 |
|
25-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Merge "Add AutoPark template class and use to control FastMixer" into nyc-dev am: c1c545a
am: d210059
* commit 'd2100593a01b4a36745b394b649e027a125513cc':
Add AutoPark template class and use to control FastMixer
|
| c05b8d7df46619d3474356241d47655478b8bc82 |
|
24-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Add AutoPark template class and use to control FastMixer
Bug: 24667895
Change-Id: Id739a109aca13664b9390d2f9d9616fa2f88d188
/frameworks/av/services/audioflinger/Threads.cpp
|
| e0a269a5f75956efdf78a9cacaefc428b352730c |
|
23-Mar-2016 |
Andy Hung <hunga@google.com> |
Fix FastTrack creation buffer count irregularity
For small values of buffer framecount, AudioFlinger may create a
normal track.
Bug: 27817933
Change-Id: I7c69491fc1f2b204e14286fb4c2210c5983afb8b
/frameworks/av/services/audioflinger/Threads.cpp
|
| 59aa43931103cebb833cb9316c5b7e9011191d33 |
|
21-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix build warnings" into nyc-dev am: 311fab5
am: 7b81b7e
* commit '7b81b7e026053615f93af1efc130205eef547e57':
Fix build warnings
|
| c42e9b462661673dff480ee71757a58b0f806370 |
|
21-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Fix build warnings
Change-Id: Idfdaff3a7a8f60cd8817bd306bfba28020a391c8
/frameworks/av/services/audioflinger/Threads.cpp
|
| 638d4c21fa86e6e525eee1e7cc14eca018658f8b |
|
19-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix build warnings" into nyc-dev am: 7be6866
am: da631f5
* commit 'da631f5894b2cfb77e47179a0f401c266b5eedb1':
Fix build warnings
|
| 7be6866af78a86a89113e9b680f1a6416ae1f711 |
|
19-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix build warnings" into nyc-dev
|
| 57c4e6f7464d458eb52d209c2a63524913d6406d |
|
18-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Fix build warnings
Change-Id: I33178dbe0bfc087d6599579ca0529ad853c669ed
/frameworks/av/services/audioflinger/Threads.cpp
|
| b3341577d32aafba7f66fa715e614c12762a63e1 |
|
19-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix copy and paste error" into nyc-dev am: 76f2713
am: 066ba12
* commit '066ba128e3641ddfb74adc3095e357d8b6f8452a':
Fix copy and paste error
|
| 584be6ac37276544809eb9369b0ee9b1614e76d4 |
|
19-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Merge "Add systrace for normal read()" into nyc-dev am: 1c65a1d
am: 07fc599
* commit '07fc5995732aa62bfb1a4e76a5e2fe45abb4ab53':
Add systrace for normal read()
|
| 76f27134fb546340154216212d84243d56a8928a |
|
19-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix copy and paste error" into nyc-dev
|
| 1c65a1d6e4a4a7c7a534aa6b178f4bb6e0967fb9 |
|
19-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Merge "Add systrace for normal read()" into nyc-dev
|
| 04dec0ea88c86761d765d6b9054ab20be18657ff |
|
19-Mar-2016 |
vivek mehta <mvivek@codeaurora.org> |
Merge "audio: allow standby for direct track" into nyc-dev am: 1606b3f
am: 929290f
* commit '929290fa589824833a7eef4e80f9a934c2246f4d':
audio: allow standby for direct track
|
| 9cd7ad101315e0bff4c5f171706c6f80c7e22644 |
|
17-Mar-2016 |
vivek mehta <mvivek@codeaurora.org> |
audio: allow standby for direct track
Delay standby for offload track use case is not suitable for direct
track in PCM format, because player torn down doesn't necessarily happen.
To avoid AP looping infinitely, allow standby right after direct track
in PCM format is paused.
Bug: 27746493
Change-Id: Id0026ac9891a12acc4affa446b2bc841559fcc28
/frameworks/av/services/audioflinger/Threads.cpp
|
| 8379b72b3c7febc5268bc8f9c152fd433a1e6c70 |
|
18-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Fix copy and paste error
Change-Id: I0976f5651e0004948108ef43eed3f89e21ecb38b
/frameworks/av/services/audioflinger/Threads.cpp
|
| ec6a70345fc99cd9f8461749a7656b8240874a62 |
|
14-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Add systrace for normal read()
Change-Id: Icba9c9fa648909604556305da7e2d9548f5b4b0c
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5bba2f6916dbe00aea7c0521faa0c6ed42114a75 |
|
18-Mar-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: use sp<> for track proxy
AudioTrack and AudioRecord shared memory proxy (Proxy class) is
RefBase so should not be refenced by raw pointer.
Change-Id: Ic63e31cef97503b8e262d5dafd8ce8f1d72d5eb8
/frameworks/av/services/audioflinger/Threads.cpp
|
| d848eb48c121c119e8ba7583efc75415fe102570 |
|
08-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Use audio_session_t consistently
Bug: 27562099
Change-Id: I328d8226191386b163f2ace41233607294c50dcd
/frameworks/av/services/audioflinger/Threads.cpp
|
| 4d231dc0ee34380956c71cbe18a750e487a69601 |
|
12-Mar-2016 |
Eric Laurent <elaurent@google.com> |
audio: Fix race condition in AudioTrack underrun.
When audio flinger mixer removes an AudioTrack from the
active list in case of underrun, it is possible that the
client has written a full buffer just after the underrun detection and
is blocked waiting for more space to write. In this case, the client
will never detect the DISABLED flag and the track never be restarted.
Also implement missing DISABLE flag detection in server side audio tracks
(OutputTrack and PatchTrack).
bug: 27567768
Change-Id: I8d0753429d4113498258b1f61bd8ac5939a612f0
/frameworks/av/services/audioflinger/Threads.cpp
|
| 517161856d74f5fe39cce131f29b977bc1745991 |
|
01-Mar-2016 |
Eric Laurent <elaurent@google.com> |
audio flinger: fix offload track underrun
Fix offload track underrun detection causing early disabling
of offloaded audio tracks.
Optimize sleep time in case of underrun to avoid spinning while
waiting for new data.
Re-enable offload when streaming
Bug: 26668110
Change-Id: I874728c69647ac05e1effb84c48700a6d0ac7435
/frameworks/av/services/audioflinger/Threads.cpp
|
| 7fd0422fbd17af3b24eb04421d37fce50f3826e2 |
|
02-Feb-2016 |
Glenn Kasten <gkasten@google.com> |
Sample rate 0 means a route-dependent rate
Remove check for primary _output_ [sic] sampling rate for fast capture.
Clean up AudioRecord handling of frame count and sample rate.
Clean up AudioTrack handling of notification period updates.
Make AudioRecord and AudioTrack more similar in order of operation, comments, and whitespace.
Bug: 25641253
Bug: 21019153
Change-Id: I24a6677945987fc39a9bf93f70357e4bc7410f98
/frameworks/av/services/audioflinger/Threads.cpp
|
| eeecb980ff4c202d0a3c4b0bfe040dce2f73336d |
|
26-Feb-2016 |
Glenn Kasten <gkasten@google.com> |
Add use for audio_unique_id_t
Bug: 25641253
Bug: 21019153
Change-Id: I65dc128e760c245f3d90559635a8981b186c87d7
/frameworks/av/services/audioflinger/Threads.cpp
|
| e736431c69e2a00468d140d2af97b63e0e5aa83a |
|
25-Feb-2016 |
Eric Laurent <elaurent@google.com> |
Merge "audio policy: add device for audio bus" into nyc-dev
|
| c54b1ffc92b8ad27608a8af21033d7cab33cb3a0 |
|
23-Feb-2016 |
Andy Hung <hunga@google.com> |
Fix AudioTrack presentationComplete regression
Bug: 27310929
Change-Id: Id52299660d90b07220a4fd9996da53fe755a41f1
/frameworks/av/services/audioflinger/Threads.cpp
|
| 58545be2ce4e701c8c37401edcc126a8b683890d |
|
23-Feb-2016 |
Eric Laurent <elaurent@google.com> |
audio policy: add device for audio bus
Add audio device category representing a bus in the
audio subsystem. E.g a set of channels in MOST automotive bus.
Bug: 25448664
Change-Id: Id00da57dc45037d4d8062e9a747c423940611539
/frameworks/av/services/audioflinger/Threads.cpp
|
| 818e7a32ce3633980138aff2c2bfcc5158b3cfcc |
|
17-Feb-2016 |
Andy Hung <hunga@google.com> |
Implement server side playback timestamps with 64 bit accuracy
Provide server timestamps if the HAL doesn't provide it.
Provide monotonic - boottime translation.
Integrate record timestamps and playback timestamps together.
Bug: 17472992
Bug: 22871200
Bug: 26400089
Bug: 26682703
Change-Id: If1974f94232fcce7ba0bbcdf63d9e54ed51918ff
/frameworks/av/services/audioflinger/Threads.cpp
|
| 6ae5843c281301a9ffd1059d185620a9337e15a2 |
|
17-Feb-2016 |
Andy Hung <hunga@google.com> |
Enable FastTrack timestamps
Bug: 26413951
Change-Id: Id54c32cb0f0be83d443f6cf1007181ebc461c586
/frameworks/av/services/audioflinger/Threads.cpp
|
| b0a144d705a021994628d136a170a6d2bd4282e7 |
|
12-Feb-2016 |
Phil Burk <philburk@google.com> |
Merge "AudioTrack: support ENCODING_IEC61937" into nyc-dev
|
| fdb3c07db5d44535eb8c3ec46dc78ad8446c01eb |
|
09-Feb-2016 |
Phil Burk <philburk@google.com> |
AudioTrack: support ENCODING_IEC61937
Set DIRECT flag.
Use audio_has_proportional_frames() instead of audio_is_linear_pcm()
where appropriate.
Bug: 24541671
Bug: 20891646
Bug: 26373761
Change-Id: Ia32036b18683b084d6c9887593df87397ea0afd9
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| e10393e72454bfd8298017dc193faf424f4e9a8f |
|
12-Jun-2015 |
Andy Hung <hunga@google.com> |
Fix audio timestamp computation for pause, stop, and dynamic speed changes
Timestamp on pause and underrun (stop) do not reflect actual position.
Timestamps do not account for dynamic changes to track speed / sample rate.
Bug: 11085154
Bug: 17552775
Change-Id: I0e5e40ab3eaee82f0c91b9f399089698a0b1947e
/frameworks/av/services/audioflinger/Threads.cpp
|
| 3f0c902beb53a245c9db35e871607dba05b8d391 |
|
16-Jan-2016 |
Andy Hung <hunga@google.com> |
Add AudioRecord timestamps
Bug: 13569372
Bug: 22886739
Change-Id: Ibc81afefb733d23676a632a0f2da31163fdbe05f
/frameworks/av/services/audioflinger/Threads.cpp
|
| 03c48d5afcb3dfc1e43df93e1f3b3b3e9292e148 |
|
28-Jan-2016 |
Glenn Kasten <gkasten@google.com> |
Add limiter for mono blend
Bug: 26904791
Change-Id: I76b16e70871cd9fc0a9b8badc66f27069fabe1cb
/frameworks/av/services/audioflinger/Threads.cpp
|
| d79072e9dff59f767cce2cda1caab80ce5a0815b |
|
06-Jan-2016 |
Glenn Kasten <gkasten@google.com> |
Remove TimedAudioTrack and associated code
Bug: 8278435
Change-Id: I095c1a4888e645e14d93b0b15fbef4524a831ca1
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2812d9ea3a3a33142dd8e23c9d949c498d6f7a12 |
|
04-Jan-2016 |
Phil Burk <philburk@google.com> |
AudioTrack: Add getUnderrunCount()
This allows an app to detect application-level output glitches.
Underrun counts survive track recreation.
Change-Id: I8eb14e92f6fc1007718a29b0666ab51ace30cdb8
Bug: 25641253
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| 113efbb3ac8e7306ea8645cb53a96d1e81c2d041 |
|
09-Jan-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix standby delay on A2DP output
Make sure that standby delay is never less than the audio flinger
default on A2DP output.
Due to variable latency and amount of buffering in A2DP sinks,
an agressive standby delay could lead to truncated audio.
Bug: 25830539.
Change-Id: I38be37ad346f5f4bf8303d3db4e3e911bf637968
(cherry picked from commit 42537be61479e59c4718e1304364551c1454f63c)
/frameworks/av/services/audioflinger/Threads.cpp
|
| 197f766a0e3c37efe4fe941553511c6022cf10b1 |
|
12-Jan-2016 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: fix standby delay on A2DP output"
|
| 42537be61479e59c4718e1304364551c1454f63c |
|
09-Jan-2016 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix standby delay on A2DP output
Make sure that standby delay is never less than the audio flinger
default on A2DP output.
Due to variable latency and amount of buffering in A2DP sinks,
an agressive standby delay could lead to truncated audio.
Bug: 25830539.
Change-Id: I38be37ad346f5f4bf8303d3db4e3e911bf637968
/frameworks/av/services/audioflinger/Threads.cpp
|
| 64f1901bdddf464c8f29bed34e3449762873b3e9 |
|
09-Jan-2016 |
Andy Hung <hunga@google.com> |
Merge "Add setMasterMono and getMasterMono"
|
| 2ddee19245641e86bca436dda23a0f5089bf2ab5 |
|
19-Dec-2015 |
Andy Hung <hunga@google.com> |
Add setMasterMono and getMasterMono
Bug: 15283594
Bug: 22700363
Change-Id: I32dc1fcecf285967a61bd508af3bb299595db57d
/frameworks/av/services/audioflinger/Threads.cpp
|
| 8a397d583a4f4cf24ad88facaf2fd33990cfb811 |
|
02-Dec-2015 |
rago <rago@google.com> |
Adding UNPROCESSED to the available audio_sources
bug: 25564291
Change-Id: I86d8feda4d687352f98b640819211207a06d6b50
/frameworks/av/services/audioflinger/Threads.cpp
|
| 438e7572c83674f4b9e6184f32f3dc94cd50524e |
|
15-Dec-2015 |
Andy Hung <hunga@google.com> |
AudioFlinger: change logcat error to warning
Change wakelock token error to warning if AudioServer.java
hasn't issued systemReady() yet (from boot or audioserver died).
Bug: 26069693
Change-Id: I8ab1ec06a30a476e058bf0dde50bfbbc60243570
/frameworks/av/services/audioflinger/Threads.cpp
|
| dcb346b7dc5b88c3e85db8a70bbd6a2fee8192b9 |
|
09-Sep-2015 |
Marco Nelissen <marcone@google.com> |
Run audio services in audioserver instead of mediaserver
audioflinger, audiopolicy, (fm) radio, soundtrigger moved to audioserver.
medialog is removed from mediaserver.
audioserver has same permissions and rights as mediaserver.
mediaserver is allowed to specify client uid to audioserver.
Subsequent CL will add medialog to audioserver.
TBD: Pare down permissions.
Bug: 24511453
Change-Id: Iccaa08b874a502710d142e72f8abbb3d97c44cfa
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2a4e18da8ee0521a48285643586159946298caef |
|
03-Dec-2015 |
Glenn Kasten <gkasten@google.com> |
Merge "Add logging to help diagnose a rare bug"
|
| 818da521b6a487518f54614b9eba68957a8d8aeb |
|
02-Dec-2015 |
Glenn Kasten <gkasten@google.com> |
Fix build warnings
Change-Id: Ic329a9852589247ed797b6a6cfed735ff3af184a
/frameworks/av/services/audioflinger/Threads.cpp
|
| f7d65ee34f64e8fad9c5af3f11da783193caf5f9 |
|
02-Dec-2015 |
Glenn Kasten <gkasten@google.com> |
Add logging to help diagnose a rare bug
Bug: 25911696
Change-Id: Idca44ed873f0a9fa911e9cb1c3a09b4e7e4f23eb
/frameworks/av/services/audioflinger/Threads.cpp
|
| 2f90c51d42efa881a9e54e4f4efadf99398304f9 |
|
02-Dec-2015 |
Glenn Kasten <gkasten@google.com> |
Don't place large objects on the stack
Bug: 25020816
Change-Id: Ife4da9fc3000e645f654f2eb28b37ad3a89d61f9
(cherry picked from commit fc39dd9f653754315df3ff7f76b8a193829a76e8)
/frameworks/av/services/audioflinger/Threads.cpp
|
| 3f273d10817ddb2f792ae043de692efcdf1988ae |
|
24-Nov-2015 |
Wei Jia <wjia@google.com> |
Uses AudioFlinger to signal battery stats for audio.
Bug: 18177221
Change-Id: I704010c037cb99bb74b98798ee8e494fa8bdc875
/frameworks/av/services/audioflinger/Threads.cpp
|
| d60560af7cb559762593161c8202459cc01fb0f5 |
|
10-Apr-2015 |
Eric Laurent <elaurent@google.com> |
audio policy: add support for external audio sources
Add support for activity on external audio sources.
An external source reflects activity on an input audio device
that must be controlled (both routing and volume) by the
audio policy manager.
First, the input device must be connected with setDeviceConnectionState().
Then, the source activity is indicated with startAudioSource() and
stopAudioSource() APIs.
startAudioSource() indicates the source device with an audio port configuration
and the use case by the audio attributes.
Once a source is active, its routing and volume are controlled by the policy manager
as it would for a software source (AudioTrack).
Change-Id: If5805d58a4356b2f681f1aabf54375f62b55b98a
/frameworks/av/services/audioflinger/Threads.cpp
|
| 403bff76dbde57fdcfce84292a5e42fa33c37842 |
|
13-Oct-2015 |
Andy Hung <hunga@google.com> |
am 4c6e77ff: AudioFlinger: Clear record buffers when starting RecordThread
* commit '4c6e77ff8e18a1551320a6b42f6a45e19dcce748':
AudioFlinger: Clear record buffers when starting RecordThread
|
| 4c6e77ff8e18a1551320a6b42f6a45e19dcce748 |
|
21-Sep-2015 |
Andy Hung <hunga@google.com> |
AudioFlinger: Clear record buffers when starting RecordThread
Bug: 24211743
Bug: 24267152
Change-Id: I58c55e56b85067b71e4e300f947b4dfc159637ba
/frameworks/av/services/audioflinger/Threads.cpp
|
| 10cfc14fadae08cd5806c4834e28aa9f743f550e |
|
28-Sep-2015 |
Glenn Kasten <gkasten@google.com> |
Merge "Add missing entries for input and output flags"
|
| 937c34741e476f26ddd0ff28a2af5e1bcd96895c |
|
28-Sep-2015 |
Glenn Kasten <gkasten@google.com> |
Add missing entries for input and output flags
Bug: 24141510
Change-Id: Ia151bc7d25d72864992888c33547a34adf460a9b
/frameworks/av/services/audioflinger/Threads.cpp
|
| 060b582b37f55b758a41b0cc26dc1ff6d112814f |
|
27-Sep-2015 |
Eino-Ville Talvala <etalvala@google.com> |
am b171ba8c: am ffce26c0: am f99498ee: CameraService: Use SCHED_FIFO for request queue thread in HFR
* commit 'b171ba8ce83ebb5abc2de9a171a15dc974756d3d':
CameraService: Use SCHED_FIFO for request queue thread in HFR
|
| f99498ee4de7123e2fd71778c6877be44fbd1506 |
|
26-Sep-2015 |
Eino-Ville Talvala <etalvala@google.com> |
CameraService: Use SCHED_FIFO for request queue thread in HFR
- Move SchedulingPolicyService from audioservice to mediautils
- When starting up a high speed stream config, set request queue thread
to SCHED_FIFO using SchedulingPolicyService
Bug: 24227252
Change-Id: I224b59142bd111caf563779f55cddd62385b9bac
/frameworks/av/services/audioflinger/Threads.cpp
|
| 0a01c2fb68e6f35905de8dfaf938d099cd48587d |
|
21-Sep-2015 |
Andy Hung <hunga@google.com> |
AudioFlinger: Clear record buffers when starting RecordThread
Bug: 24267152
Change-Id: I58c55e56b85067b71e4e300f947b4dfc159637ba
/frameworks/av/services/audioflinger/Threads.cpp
|
| c06f0ef67f02f88e01e2b2679e83c87628dbfc1a |
|
11-Sep-2015 |
Eric Laurent <elaurent@google.com> |
am b7b97b9d: am 48e879db: am a2ab4505: audio flinger: force audio path start from normal mixer
* commit 'b7b97b9d897581b30f4bba78ec360474646c230a':
audio flinger: force audio path start from normal mixer
|
| a2ab4505c807f42afe34809409c4e85d91618a8c |
|
09-Sep-2015 |
Eric Laurent <elaurent@google.com> |
audio flinger: force audio path start from normal mixer
The audio HAL wakes up and configures the audio path when receiving
the first write() in standby state. This causes a certain amount of
process to take place in the mixer threads which is problematic for
fast mixer running at FIFO priority.
We now force a fake write() of 0 bytes from normal mixer to trigger
the audio path configuration before starting the fast mixer.
Bug: 23791972.
Change-Id: I54311b337fda956444846f5d2f53a3263d54e04b
/frameworks/av/services/audioflinger/Threads.cpp
|
| f87c2f5eb4e37f0950962e31b9ca49e32f5b0864 |
|
21-Aug-2015 |
Glenn Kasten <gkasten@google.com> |
dumpsys was missing "processing" for format and frame size
Change-Id: I6931b18a6a85bf0638d659e31bca5aa6c8cf62bf
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5ce059661dcbab8924edab400681fc408ee76123 |
|
24-Jul-2015 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix capture overruns at non-primary sample rate" into mnc-dev
|
| a97630bda8815958587ca31adb2d37090022af25 |
|
23-Jul-2015 |
Andy Hung <hunga@google.com> |
Fix record memory buffer allocation for legacy upmix/downmix
Bug: 22173057
Change-Id: I8f5056ff5a1252c71a3d3b354440551bcd9fd466
/frameworks/av/services/audioflinger/Threads.cpp
|
| eb9487e10294a4e73977f460f30eeaff503acd21 |
|
22-Jul-2015 |
Glenn Kasten <gkasten@google.com> |
Fix capture overruns at non-primary sample rate
and small buffer size. Also:
Pull out the magic number "12 ms" to a named constant.
Remove obsolete AudioFlinger::mPrimaryOutputSampleRate.
Bug: 22662814
Change-Id: I261f75a222c4505a84aad2493d251bd2dea59f68
/frameworks/av/services/audioflinger/Threads.cpp
|
| ca5e6143740299c877d69e97f7968cd04476d32c |
|
14-Jul-2015 |
Phil Burk <philburk@google.com> |
AudioFlinger: fix repeated underruns for compressed audio
The AudioFlinger kept pausing the audio when playing compressed AC3 or DTS.
This caused pause/resume loops that were hard to break out of.
The AudioFlinger was thinking that the compressed audio was PCM
because the HAL was in PCM mode playing SPDIF data bursts.
It also thought that EAC3 was at 192000 Hz instead of 48000
Hz because the data bursts are played at a higher rate.
This CL adds more calls to the shim that separates the AudioFlinger.
Now the AudioFlinger gets information about the HAL sample rate,
channel masks and format from the shim instead of calling the HAL directly.
The AudioFlinger now uses a different threshold for detecting
underruns when the audio is compressed.
Bug: 19938315
Bug: 20891646
Change-Id: Ib16f539346d1c7a273ea4feb3d3afcc3dc60237d
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| 4b33e0838fdb1b5e545449add02005916b512c99 |
|
13-Jul-2015 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: flush HAL when transitioning to next direct track" into mnc-dev
|
| 7c1ec5f038e63a5eb8b04434577c25bc23f5f410 |
|
09-Jul-2015 |
Eric Laurent <elaurent@google.com> |
audio: several fixes in audio routing callbacks
- audio policy:
Force device change to ensure new audio patch creation
upon first track activity on a given output.
Fix function device_distinguishes_on_address() which could mistake
some output device with remote submix input device.
- audio flinger:
Reduce number of binder calls upon new client registration by only
sending ioConfigChanged() callbacks to newly registered client.
Fix first patch after output thread creation not triggering an
ioConfigChanged() callback.
-audio system:
Force client registration upon routing callback installation to force
new ioConfigChanged() callback from audio flinger.
Bug: 22381136.
Change-Id: Ieb0d9f92f563a40552eb31bc0499c8ac65f78ce4
/frameworks/av/services/audioflinger/Threads.cpp
|
| 0f0631eb55b1f0a7f4b62212b78a3faa0b49919b |
|
07-Jul-2015 |
Eric Laurent <elaurent@google.com> |
audioflinger: flush HAL when transitioning to next direct track
Send flush command to the audio HAL when transtioning to
next track on direct output thread, even if both tracks are in the
same audio session.
Commit 43b4dcc to fix issue 21145353 did only flush the HAL if the
audio session was different for the new track because the logic was
copied from the offload thread.
Bug: 22019044.
Change-Id: I89b217580023ed7449a58e9bf3dc068ce7a84487
/frameworks/av/services/audioflinger/Threads.cpp
|
| b9d73333cce3f9da3a7a0b33589f6bbe0f992a92 |
|
01-Jul-2015 |
Eric Laurent <elaurent@google.com> |
audio: add definitions for devices connected over IP
Bug: 22068684.
Change-Id: Idde0eaf7121d2e43f32eee3e6b10e99d8cff4912
/frameworks/av/services/audioflinger/Threads.cpp
|
| e8726fea8a53bf3474aa3c6deaf2f6c1f565e694 |
|
26-Jun-2015 |
Eric Laurent <elaurent@google.com> |
reduce number of binder calls from mediaserver
Reduce the number of audio port, audio patch and
IO config changed binder calls from mediaserver to
client processes:
- Do not call IO config changed callback if selected
device is the same as previously selected one on a given
audio flinger playback or capture thread.
- Do not call the audio port or audo patch list update
callback on a client if this client as no listener registered.
Bug: 22045560.
Change-Id: If780e105404de79b7cb5c80c27b793ceb6b1c423
/frameworks/av/services/audioflinger/Threads.cpp
|
| 40eb1a1f8871909c272e72afaf7d5af84fea2412 |
|
18-Jun-2015 |
Andy Hung <hunga@google.com> |
Reduce log messages on AudioFlinger throttle
Bug: 21858740
Change-Id: I8f291b64c1033867bb57ffceaa3b7d94aa998715
/frameworks/av/services/audioflinger/Threads.cpp
|
| ad9ef61e770c0751a9983aa5c9844dfeb9ed665b |
|
11-Jun-2015 |
Phil Burk <philburk@google.com> |
Merge "AudioFlinger: flush stream when switching tracks" into mnc-dev
|
| 43b4dcc660e6da96285e4672ae371070ab845401 |
|
10-Jun-2015 |
Phil Burk <philburk@google.com> |
AudioFlinger: flush stream when switching tracks
For direct threads, when recycling a stream, perform a flush so
that the frame position is reset.
Bug: 21145353
Change-Id: I08611cd64bb249a9659c44f9e4c47e7455f4838f
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| e1635ec096d1110c33a5aa46847af59c261fb7fa |
|
09-Jun-2015 |
Glenn Kasten <gkasten@google.com> |
Take advantage of audio_channel_in_mask_from_count
Bug: 21375533
Bug: 21721483
Change-Id: I1ccd5d1d68a25f415dc4a62bf7a44d9db12a256b
/frameworks/av/services/audioflinger/Threads.cpp
|
| 08fb1743f80437c38b4094070d851ea7f6d485e5 |
|
01-Jun-2015 |
Andy Hung <hunga@google.com> |
Throttle MixerThread data pull to no more than twice expected rate
This helps prevent underruns with NuPlayer and other applications
which set up buffers that are close to minimum size or use deep
buffers, and rely on a double-buffering sleep strategy to fill.
Enabled by default. Disabled by setting af.thread.throttle 0
Bug: 19062223
Bug: 21198655
Change-Id: Ia52b48e0c99588af5db53c43fede2afd775b8899
/frameworks/av/services/audioflinger/Threads.cpp
|
| e1f939bf0e1bac806b7da1b316e70c96426dc1b6 |
|
30-May-2015 |
Glenn Kasten <gkasten@google.com> |
Merge "Add more device codes for symbolic display" into mnc-dev
|
| ad9cb8b88e4f8face23f01d8dc89fa769dacb50f |
|
27-May-2015 |
Eric Laurent <elaurent@google.com> |
audio flinger: add suffix to time variables names
Add suffix to clarify units for the following variables:
standbyTime -> mStandbyTimeNs
standbyDelay -> mStandbyDelayNs
activeSleepTime -> mActiveSleepTimeUs
idleSleepTime -> mIdleSleepTimeUs
sleepTime -> mSleepTimeUs
Change-Id: I7f5d602c39e0ef3f6fe9ef99eaf1b351c7bd4fc3
/frameworks/av/services/audioflinger/Threads.cpp
|
| 3ab77c1e93f3da5da0d18ff998f1bd6db19534f3 |
|
27-May-2015 |
Eric Laurent <elaurent@google.com> |
Merge "audio flinger: pause direct output when underrunning" into mnc-dev
|
| 84d61caa6040e346df68dbe6f8e4fe339906de08 |
|
07-May-2015 |
Glenn Kasten <gkasten@google.com> |
Add more device codes for symbolic display
Change-Id: I08553f0e94d0a0931ccf98ee04f53686b96c8b03
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5cff403679fc44c8293de81aed31c459c6129243 |
|
26-May-2015 |
Eric Laurent <elaurent@google.com> |
audio flinger: pause direct output when underrunning
If a direct output supports pause, pause the stream
instead of placing it into standby when the audio track
underruns. This will avoid resetting the presented frame count
and preserve A/V sync.
Bug: 21437855.
Change-Id: I598346edb62a1864126acdb1d9a937c82eac2191
/frameworks/av/services/audioflinger/Threads.cpp
|
| 748a792be85838c429ebf46acf7d6eb02e79f00b |
|
21-May-2015 |
Eric Laurent <elaurent@google.com> |
Merge "audio flinger: do not call JAVA services until system is ready" into mnc-dev
|
| 72e3f39146fce4686bd96f11057c051bea376dfb |
|
20-May-2015 |
Eric Laurent <elaurent@google.com> |
audio flinger: do not call JAVA services until system is ready
Wait for system ready indication form AudioService before enabling
calls to scheduling service or power manager.
Bug: 11520969.
Change-Id: I221927394f4a08fd86c9d457e55dd0e07949f0cf
/frameworks/av/services/audioflinger/Threads.cpp
|
| 8ef66c6662002279a606df70f41651491a349126 |
|
20-May-2015 |
Andy Hung <hunga@google.com> |
Merge "Update FastTrack channel conversion check" into mnc-dev
|
| 1f439e1cf16a29347288ba9ddd06c0b6d086a145 |
|
19-May-2015 |
Andy Hung <hunga@google.com> |
Update FastTrack channel conversion check
Change-Id: Ia1a3124e6408859bf4d95ff9fd95dda6970a4a7f
/frameworks/av/services/audioflinger/Threads.cpp
|
| f98ec8d0d42e6952c0a7cc5027935851073f7426 |
|
19-May-2015 |
Andy Hung <hunga@google.com> |
Fix channel mask dump in AudioFlinger
Change-Id: Iba5ccd1885775b14c44342c7b169a0672b93549b
/frameworks/av/services/audioflinger/Threads.cpp
|
| d9f493ebcd1830c76d7b1782e64c7ba9ad8ab4bd |
|
12-May-2015 |
Andy Hung <hunga@google.com> |
Merge "Update sampling rate to 192kHz for devices" into mnc-dev
|
| 69dce3343ffe33d2ba60ab4c6755953a7ec96899 |
|
12-May-2015 |
Marco Nelissen <marcone@google.com> |
Merge "Work around bug in offload driver" into mnc-dev
|
| 078538cfc6c4682889ed52de460d29c0d15bb9eb |
|
12-May-2015 |
Marco Nelissen <marcone@google.com> |
Work around bug in offload driver
There is a bug in the offload driver that causes the last offload buffer(s)
to be dropped unless the device is on power or holding a wake lock.
To avoid truncated playback, we now hold a wake lock during the drain phase
of offloaded playback.
Bug: 19928717
Change-Id: I8df22e965ec791448aa5d9b74e743f48ef886fc4
/frameworks/av/services/audioflinger/Threads.cpp
|
| db4c031f518ae5806af73756273ff32cd8d0e4f8 |
|
06-May-2015 |
Andy Hung <hunga@google.com> |
Update sampling rate to 192kHz for devices
Change-Id: I0a83206be51d7ae18ccf85b94b2127356307be69
/frameworks/av/services/audioflinger/Threads.cpp
|
| af49160aa75d4e1750f3ed56b812ec9cc448fa46 |
|
12-May-2015 |
Andy Hung <hunga@google.com> |
Merge "Allow 8 channels in RecordThread" into mnc-dev
|
| f6870aefc5e31d4220f3778c4e79ff34a61f48ad |
|
08-May-2015 |
Eric Laurent <elaurent@google.com> |
audio flinger: fix fuzz test crash
Clear output stream pointer in duplicating thread
when the main output to which it is attached is closed.
Also do not forward master mute and volume commands to
duplicating threads as this is not applicable.
Also fix logic in AudioFlinger::primaryPlaybackThread_l()
that could accidentally return a duplicating thread.
This never happens because the primary thread is always
first in the list.
Bug: 20731946.
Change-Id: Ic8869699836920351b23d09544c50a258d3fb585
/frameworks/av/services/audioflinger/Threads.cpp
|
| 296fb13dd9b5e90d6a05cce897c3b1e7914a478a |
|
01-May-2015 |
Eric Laurent <elaurent@google.com> |
Implement audio device callback
Add class AudioSystem::AudioDeviceCallback notifying
AudioSystem clients upon device selection change on a given
input or output thread.
Maintain a list of installed callback per I/O handle in AudioSystem
and call registered callbacks when an OPEN of CONFIG_CHANGED event
is received on IAudioFlingerClient::ioConfigChanged().
Add methods to AudioTrack and AudioRecord to add and remove device
change callbacks.
Add methods to AudioTrack and AudioRecord to query currently selected
device.
ioConfigChanged() events now convey the audio patch describing
the input or output thread routing.
Fix AudioRecord failure to start when invalidation is
handled by start().
Change-Id: I9e938adf025fa712337c63b1e02a8c18f2a20d39
/frameworks/av/services/audioflinger/Threads.cpp
|
| d1abb8f94d7ed749ee959655db1e07d26dad074d |
|
06-May-2015 |
Andy Hung <hunga@google.com> |
Allow 8 channels in RecordThread
Change-Id: I738d4975188695e568015e1bc64d160550e958f5
/frameworks/av/services/audioflinger/Threads.cpp
|
| 8ce74c3c11458faa34395591a3424e90db856bfc |
|
02-May-2015 |
Phil Burk <philburk@google.com> |
Merge "AudioTrack: fix direct tracks not pausing" into mnc-dev
|
| 6fc2a7c81f62b1e21487ae37e11aae6241bc3ead |
|
01-May-2015 |
Phil Burk <philburk@google.com> |
AudioTrack: fix direct tracks not pausing
When a Direct Track is paused and the HAL does not support
pause() and resume() then the HW never gets paused.
The app can just keep writing data, which gets played.
Bug: 18899620
Change-Id: Ice0f360956ff7ca425f6f24a0a2a8640d8b43fa8
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| 73e26b661af50be2c0a4ff6c9ac85f7347a8b235 |
|
28-Apr-2015 |
Eric Laurent <elaurent@google.com> |
AudioSystem: refactor audio config cache and callbacks
Clean up implementation of audio configuration cache and
callback events from AudioFlinger:
- Define class AudioIoDescriptor for audio input and output
configurations outside of AudioSystem class.
- Do not use void * but an AudioIoDescriptor as argument to
audio config callbacks from AudioFlinger.
- Remove unused configuration events.
- Move AudioSystem audio input and output cache from static singletons to
members of AudioFlingerClient subclass.
Change-Id: I67c196c32c09ce2756af0755ee1fe631040c3270
/frameworks/av/services/audioflinger/Threads.cpp
|
| 054d9d3dea1390294650ac704acb4aa0a0731217 |
|
24-Apr-2015 |
Eric Laurent <elaurent@google.com> |
PatchPanel: do not use setParameters() internally.
Do not use setParameters() with AUDIO_PARAMETER_STREAM_ROUTING
when communicating the input or output device selected to playback or
record threads, even for HAL version less than 3.0.
Use createAudioPatch()/releaseAudioPatch() instead.
This allows to send more information on the output or input device being
selected.
Also fix a regression introduced in L where the output device selection
was not communicated to effects on record threads.
Change-Id: I4780ada53241d56694b005c992171e173c3bf8f5
/frameworks/av/services/audioflinger/Threads.cpp
|
| d330ee46022f34da76d14d0c4d2910526ecc2321 |
|
20-Apr-2015 |
Andy Hung <hunga@google.com> |
Add floating and multichannel record to AudioFlinger
Change-Id: Ia388fb012a0b6d81613ef87142a97d76836338f9
/frameworks/av/services/audioflinger/Threads.cpp
|
| 04f07fd61022dac46ddda9c815d65bc3d3278b84 |
|
22-Apr-2015 |
Andy Hung <hunga@google.com> |
Merge "Make record buffer in RecordThread variable format"
|
| 5a8a95de6dad1a3bcf3da5a37b35766e89086e13 |
|
18-Apr-2015 |
Ricardo Garcia <rago@google.com> |
Use AudioPlaybackRate to hold TimestretchBufferProvider parameters
Use this struct to handle the parameters for TimestretchBufferProvider all
across the system.
Add stretch mode and fallback mode to TimestretchBuffer Provider.
Change-Id: I19099924a7003c62e48bb6ead56c785cb129fba2
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5744661e85981f8a9456bf470e2761235fc026da |
|
20-Apr-2015 |
Andy Hung <hunga@google.com> |
Make record buffer in RecordThread variable format
Change-Id: Id4bb9b973eeea16946fba3bc084c7ac270d9fa33
/frameworks/av/services/audioflinger/Threads.cpp
|
| 8edb8dc44b8a2f81bdb5db645b6b708548771a31 |
|
27-Mar-2015 |
Andy Hung <hunga@google.com> |
Add playback rate to AudioTrack
Bug: 19196501
Change-Id: I6411e1d3ce652b711a71a6d9df020cb5f60d4714
/frameworks/av/services/audioflinger/Threads.cpp
|
| 6770c6faa3467c92eabc5ec9b23d60eb556a0d03 |
|
07-Apr-2015 |
Andy Hung <hunga@google.com> |
Enable 8 bit and float pcm record formats for AudioFlinger
Update sampling rate handling as well.
Bug: 19570772
Change-Id: I872248e64c0578b2e48869a68fee0d51bd0640c3
/frameworks/av/services/audioflinger/Threads.cpp
|
| 73c02e4277b399c2ec1555d32b6ad5df23bb83dc |
|
29-Mar-2015 |
Andy Hung <hunga@google.com> |
Improve ResamplerBufferProvider
Change-Id: I3cc3af221ad5797ff219d75227350733afa180db
/frameworks/av/services/audioflinger/Threads.cpp
|
| 97a893eb34f8687485c88eaf15917974a203f20b |
|
29-Mar-2015 |
Andy Hung <hunga@google.com> |
Add RecordBufferConverter for RecordThread data processing
Change-Id: Ia3aab8590cd41e8a7cba0a7345d70d2866d92045
/frameworks/av/services/audioflinger/Threads.cpp
|
| b369cafd67beb63dd0278dba543f519956208a7f |
|
31-Mar-2015 |
Eric Laurent <elaurent@google.com> |
audio flinger: fix standby on output with HW A/V sync
Fix a bug in audio HAL pause logic on output threads with
HW A/V sync preventing the HAL to enter standby when
the audio track is stopped and detroyed.
Bug: 19980184.
Change-Id: Ia497dad23159038b447fcbc18a67bb61b70b79cc
/frameworks/av/services/audioflinger/Threads.cpp
|
| ae9710473b88592c557488bfcae71616d9e9b75f |
|
24-Mar-2015 |
Phil Burk <philburk@google.com> |
Merge "AudioFlinger: call SPDIF wrapper from AudioFlinger"
|
| 062e67a26e0553dd142be622821f493df541f0c6 |
|
11-Feb-2015 |
Phil Burk <philburk@google.com> |
AudioFlinger: call SPDIF wrapper from AudioFlinger
Create an interface layer between the AudioFlinger and the HAL
that manages the wrapping and format conversion.
Removed unnecessary includes.
Handle rate conversion in getRenderPosition().
Try to open HAL with encoded format before wrapping with SPDIF.
Bug: 17566660
Change-Id: I00ad888ca15ff0f85b85efb8167c7f5ea761a244
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/services/audioflinger/Threads.cpp
|
| b7fbf7ecc6b034243ec64f79f3113675b5e3c941 |
|
18-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
AudioRecord TRANSFER_OBTAIN can be used with FAST flag
Change-Id: I001ba1a88150dddf79d99baf5927f31799745eef
/frameworks/av/services/audioflinger/Threads.cpp
|
| d198b85a163330b03e7507c9e8bfeb5f4d958a6c |
|
16-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Fix typos
Fix typos in comments
Add formal parameter name to declaration where it was missing
Fix out of order comments
Change-Id: I1de81ae82af5ca507864e4c7b959111bac898b98
/frameworks/av/services/audioflinger/Threads.cpp
|
| 5a23aa949188fafe1d8d35966ff5a73974294dfd |
|
19-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Merge "Initial implementation of fast capture dump"
|
| b4f18f964cba9f124e5a264428290ada6eb4aa00 |
|
09-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Merge "Allow TRANSFER_OBTAIN to be used for fast tracks"
|
| 1dfe2f9c2d03fc8d0ed0cdfe0b9fb894bc0bcc11 |
|
09-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Allow TRANSFER_OBTAIN to be used for fast tracks
Change-Id: I8fa20c26f076567b38210af4a680fe1cb2eacee4
/frameworks/av/services/audioflinger/Threads.cpp
|
| 17c9c998afed5ed9df7495eeed5822f3ed53ebec |
|
03-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Initial implementation of fast capture dump
Change-Id: I898d903e539f760ef7caa80f41ca21c223f67264
/frameworks/av/services/audioflinger/Threads.cpp
|
| 44182c206f7c5584ef2cf504da6be98fab665dbf |
|
06-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Dump generic information first for each thread
also mBufferSize was already being displayed as part of dumpBase
Change-Id: I17f3062fcc076c594b5fd6b8fca286b27067e07c
/frameworks/av/services/audioflinger/Threads.cpp
|
| 0b89bc0d285b8fd4798df1ff0ba9f93851a3bd48 |
|
06-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Display more fields in thread dumpBase
Change-Id: Ice15e999dda2f6cf9d23685ade4a87f74180322d
/frameworks/av/services/audioflinger/Threads.cpp
|
| d7dca050c630bddbd73a6623271b34b4290460ee |
|
06-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Rename mName to mThreadName and kNameLength to kThreadNameLength
Change-Id: I0adfcdcab7923a07a840ec0e04528cb8bfc41f10
/frameworks/av/services/audioflinger/Threads.cpp
|
| bcb1486d052e329ae4790d93055d1c51017286c3 |
|
06-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Whitespace
switch indent
Change-Id: I652c798dd37a80634d247c4d881fb1cce92c4bd6
/frameworks/av/services/audioflinger/Threads.cpp
|
| 0f5b562737d6b5457aa83a4fdce9c6fb32584d9d |
|
18-Feb-2015 |
Glenn Kasten <gkasten@google.com> |
Add devicesToString, inputFlagsToString, and sourceToString
Change-Id: I0c09d76c204ffc5579f62d2ed1faef07922a5962
/frameworks/av/services/audioflinger/Threads.cpp
|
| dfb0e115d827887e2f56a8877fe41b256d24360e |
|
18-Feb-2015 |
Glenn Kasten <gkasten@google.com> |
Fix typo in string table
Change-Id: Iaf3d3c77129c62cf3dcad21fc754f390eb43b28e
/frameworks/av/services/audioflinger/Threads.cpp
|
| fbdb2aceab7317aa44bc8f301a93eb49e17b2bce |
|
02-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Pull up increaseSamplingN and kSamplingNforLowRamDevice
from FastMixerDumpState to FastThreadDumpState, and remove unused parameter
from FastMixerDumpState constructor.
Change-Id: Ib8937b106622a8da28a6ef6043de4528ae82cb05
/frameworks/av/services/audioflinger/Threads.cpp
|
| 63238efb0d674758902918e3cdaac322126484b7 |
|
03-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Remove redundant semicolon from namespace closing
Change-Id: I163f9d3d216c283ae1160ce4802e5247cf44fba7
/frameworks/av/services/audioflinger/Threads.cpp
|
| d797a9d5daad3051f9ac1d348abc2f434ea3ddcf |
|
02-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Fix compile error if FAST_THREAD_STATISTICS not defined
Change-Id: I3a4ac558e61ad956a7a6e325534e722066e49b2f
/frameworks/av/services/audioflinger/Threads.cpp
|
| d4bd93b22ab9f7dfa32dfc6d48e008db3ea9c634 |
|
28-Feb-2015 |
Andy Hung <hunga@google.com> |
Merge "Change AudioTrack resampling buffers from 3 to 2"
|
| ad8510a339ffab330c2c46e5c247dd1cf9e15c22 |
|
18-Feb-2015 |
Glenn Kasten <gkasten@google.com> |
Include what you use
Change-Id: I1967861f383bb5ed6743fb69e3bd439907ed7033
/frameworks/av/services/audioflinger/Threads.cpp
|
| 0e48d25606c82def035ad10a5b3923767a765cdd |
|
26-Jan-2015 |
Andy Hung <hunga@google.com> |
Change AudioTrack resampling buffers from 3 to 2
Move computation of minimum AudioTrack buffer size to server
for normal streaming PCM tracks.
Use server-side computation to exactly determine requirements
for the resampler to avoid triple buffering.
This reduces latency for normal audio tracks that require resampling,
and makes things consistent with the minimum buffer size.
Change-Id: I2f2ca0e599ee20e16559bc5c5dab61ed100da16c
/frameworks/av/services/audioflinger/Threads.cpp
|
| dce27d0ebab31e82543bb777ed3eb04955cd18ff |
|
17-Jan-2015 |
Andy Hung <hunga@google.com> |
Merge "Update duplicating thread to use float audio"
|
| 2d2bd6af573383bced45e2f610e11193e28fdcdb |
|
16-Jan-2015 |
Andy Hung <hunga@google.com> |
Merge "Allow duplicating thread to use native mixing audio format"
|
| fbfc3959f4aac839445edc7075532067fef497c2 |
|
15-Jan-2015 |
Andy Hung <hunga@google.com> |
Update duplicating thread to use float audio
Previously device format was used, typically pcm 16 bit.
Change-Id: I70a8b594820e948a2caa299194807ec17348b0e9
/frameworks/av/services/audioflinger/Threads.cpp
|
| c25b84abdd7ff229d0af663fbf3a37bd9512939d |
|
15-Jan-2015 |
Andy Hung <hunga@google.com> |
Allow duplicating thread to use native mixing audio format
Update OutputTrack to take variable formats.
Previously conversion to AUDIO_FORMAT_PCM_16_BIT was required.
Change-Id: I4a96a60ddd8d8dfe651405a0bcd4f98c89bc1ade
/frameworks/av/services/audioflinger/Threads.cpp
|
| 81879df8dc51df4d8a3880349e9e1e6b5e3cebfb |
|
07-Jan-2015 |
Eric Laurent <elaurent@google.com> |
am 69158e3e: am 6ab33981: am 610255e8: Merge "audioflinger: pause HW A/V sync output when AudioTrack underruns" into lmp-mr1-dev
* commit '69158e3e7b565a5ca131a2efaa9b76615ca80cbb':
audioflinger: pause HW A/V sync output when AudioTrack underruns
|
| 69158e3e7b565a5ca131a2efaa9b76615ca80cbb |
|
07-Jan-2015 |
Eric Laurent <elaurent@google.com> |
am 6ab33981: am 610255e8: Merge "audioflinger: pause HW A/V sync output when AudioTrack underruns" into lmp-mr1-dev
* commit '6ab33981d45cd69e683a143fec6530e66bd3e371':
audioflinger: pause HW A/V sync output when AudioTrack underruns
|
| b187de1ada34a9023c05d020a4592686ba761278 |
|
30-Dec-2014 |
Glenn Kasten <gkasten@google.com> |
Line length 100
Change-Id: I6c8fe626a3825fa9e139319656d682a57b887c97
/frameworks/av/services/audioflinger/Threads.cpp
|
| 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
|
| 23c8dc13787186dc98731055858b2fba3aedc105 |
|
17-Dec-2014 |
Eric Laurent <elaurent@google.com> |
am 85aca658: am d33712d7: am 145cf5d8: Merge "audioflinger: implement pause/resume for direct outputs" into lmp-mr1-dev
* commit '85aca658ac7d20584b0647427256df50a5f243ef':
audioflinger: implement pause/resume for direct outputs
|
| 85aca658ac7d20584b0647427256df50a5f243ef |
|
17-Dec-2014 |
Eric Laurent <elaurent@google.com> |
am d33712d7: am 145cf5d8: Merge "audioflinger: implement pause/resume for direct outputs" into lmp-mr1-dev
* commit 'd33712d7ec5dcf427cc0be9b7d2ca1c99823c8e6':
audioflinger: implement pause/resume for direct outputs
|
| 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
|
| 65575de268f4c0d75e6c1172469001ace765fc14 |
|
11-Dec-2014 |
Phil Burk <philburk@google.com> |
am 2592184c: am 6baa8fe7: am 94a92c69: Merge "[audio][audioflinger] Consume all frames when a DirectThread is done." into lmp-mr1-dev
* commit '2592184c976ef83aa9fdad7d63bfdb408c95aa7d':
[audio][audioflinger] Consume all frames when a DirectThread is done.
|
| 2592184c976ef83aa9fdad7d63bfdb408c95aa7d |
|
11-Dec-2014 |
Phil Burk <philburk@google.com> |
am 6baa8fe7: am 94a92c69: Merge "[audio][audioflinger] Consume all frames when a DirectThread is done." into lmp-mr1-dev
* commit '6baa8fe747621b7121ff18dcdfab15d99c349c15':
[audio][audioflinger] Consume all frames when a DirectThread is done.
|
| 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
|
| b67eb5856203e9869aa1e5b5a13c1eff62790335 |
|
10-Dec-2014 |
Eric Laurent <elaurent@google.com> |
am b279f5ad: am c49d9cda: am 2dda40d6: Merge "audio policy: add support for custom mixes" into lmp-mr1-dev
* commit 'b279f5adfa2fd6f22c87b3a7106fbb62a0b9a5d2':
audio policy: add support for custom mixes
|
| b279f5adfa2fd6f22c87b3a7106fbb62a0b9a5d2 |
|
10-Dec-2014 |
Eric Laurent <elaurent@google.com> |
am c49d9cda: am 2dda40d6: Merge "audio policy: add support for custom mixes" into lmp-mr1-dev
* commit 'c49d9cda49dab8b7dd0376cf56476b3e1895d5c6':
audio policy: add support for custom mixes
|
| 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
|
| f849e1984ae639112e125a91e27098cf0ebc68c1 |
|
09-Dec-2014 |
Andy Hung <hunga@google.com> |
am 900f6535: am 561f50de: am db995fe3: Merge "Fix getTimestamp if offload thread is not initialized." into lmp-mr1-dev
* commit '900f6535af232298e4a77477907a40f68e8dc61b':
Fix getTimestamp if offload thread is not initialized.
|
| 900f6535af232298e4a77477907a40f68e8dc61b |
|
08-Dec-2014 |
Andy Hung <hunga@google.com> |
am 561f50de: am db995fe3: Merge "Fix getTimestamp if offload thread is not initialized." into lmp-mr1-dev
* commit '561f50de82ce47f11ba8d4f25db1ca1d8467b2a7':
Fix getTimestamp if offload thread is not initialized.
|
| 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
|
| 3458bb2356e711419487056fe1dd474e100466a3 |
|
26-Nov-2014 |
Eric Laurent <elaurent@google.com> |
am 121143d5: am bf14c62e: am 72bf901c: Merge "audio policy: new getOutputForAttr() prototype." into lmp-mr1-dev
* commit '121143d5242a790d0bd01fe1b9cec5d28a1ba1d7':
audio policy: new getOutputForAttr() prototype.
|
| 121143d5242a790d0bd01fe1b9cec5d28a1ba1d7 |
|
26-Nov-2014 |
Eric Laurent <elaurent@google.com> |
am bf14c62e: am 72bf901c: Merge "audio policy: new getOutputForAttr() prototype." into lmp-mr1-dev
* commit 'bf14c62e7973ed793a3b00c2c67beca4484c1c7a':
audio policy: new getOutputForAttr() prototype.
|
| 3a3e87005fdfbfe4564605220a044b8023d0f539 |
|
26-Nov-2014 |
Eric Laurent <elaurent@google.com> |
am 72215491: am 1a475921: am 223fd5c9: audio: new routing strategies and stream types
* commit '72215491c60fbcdb9a2f0be782e24e39cca249c5':
audio: new routing strategies and stream types
|
| 72215491c60fbcdb9a2f0be782e24e39cca249c5 |
|
26-Nov-2014 |
Eric Laurent <elaurent@google.com> |
am 1a475921: am 223fd5c9: audio: new routing strategies and stream types
* commit '1a475921c0577a4650d1bbe40a85b732d1766939':
audio: new routing strategies and stream types
|
| 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
|
| 994242df089e1abbd7d1d21d8183b4ad32710948 |
|
21-Nov-2014 |
Glenn Kasten <gkasten@google.com> |
am c63c0973: am f50db74f: Merge "Fix format conversion in DuplicatingThread" into lmp-mr1-dev automerge: d48f1b6
* commit 'c63c09738772133e9b20a7c8576ad116ff855ee2':
Fix format conversion in DuplicatingThread
|
| c63c09738772133e9b20a7c8576ad116ff855ee2 |
|
21-Nov-2014 |
Glenn Kasten <gkasten@google.com> |
am f50db74f: Merge "Fix format conversion in DuplicatingThread" into lmp-mr1-dev
automerge: d48f1b6
* commit 'd48f1b68b2b824ffcfdbf6aa67ad4696020a8a11':
Fix format conversion in DuplicatingThread
|
| 3dd2eb810af471df146cb16a680b1b0ca7364be5 |
|
21-Nov-2014 |
Glenn Kasten <gkasten@google.com> |
am d48f1b68: am f50db74f: Merge "Fix format conversion in DuplicatingThread" into lmp-mr1-dev
* commit 'd48f1b68b2b824ffcfdbf6aa67ad4696020a8a11':
Fix format conversion in DuplicatingThread
|
| 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
|
| 77e79a509d19ab50b1867fe3903b5a8fae450591 |
|
17-Nov-2014 |
Marco Nelissen <marcone@google.com> |
resolved conflicts for merge of e78a27ca to lmp-mr1-dev-plus-aosp
Change-Id: If10a9cc17245f95d5e10b1507445abbb4020670e
|
| f888020c6e2735624f2b2a30e72aca24e17b8b4d |
|
14-Nov-2014 |
Marco Nelissen <marcone@google.com> |
Update calls to IInterface::asBinder()
to use the new static version.
Change-Id: Ia7b10eb38ca55b72278bfd33d3bf647f338b4e6a
Conflicts:
media/libmedia/IAudioFlinger.cpp
media/libmedia/IMediaPlayer.cpp
media/libstagefright/CameraSource.cpp
/frameworks/av/services/audioflinger/Threads.cpp
|
| 06b46062d2f8bc82ca3061a23d197734ae51918b |
|
14-Nov-2014 |
Marco Nelissen <marcone@google.com> |
Update calls to IInterface::asBinder()
to use the new static version.
Change-Id: I89a5988a0ac694ffc04d88cf939e8455bf925d4c
/frameworks/av/services/audioflinger/Threads.cpp
|
| e77540228e1f60b1129a1615d2e43e0bf8015d3c |
|
31-Oct-2014 |
Glenn Kasten <gkasten@google.com> |
Include normal output track buffer depth in systrace
Also log "second chance" sleeps
Change-Id: I6e352948926c246002642a50c92af213e6b79040
/frameworks/av/services/audioflinger/Threads.cpp
|
| 4b4ceaabd739b39e0690911afd1ae8f6d5ae9fae |
|
08-Nov-2014 |
Eric Laurent <elaurent@google.com> |
am ccfeadcd: audioflinger: fix duplicating thread output buffers clear automerge: 02b5708
automerge: 8510932
* commit '8510932d6e5a1d4628c700fbff8f836abe470713':
audioflinger: fix duplicating thread output buffers clear
|
| 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
|
| 97b7b75b6a31330e213bdb96ccc60916218ad903 |
|
28-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
Improve dumpsys for output threads
Display output thread type and flags.
Also add units and use consistent spelling.
Change-Id: I19eb627f41c4887b6ad0543024a8477a99890798
/frameworks/av/services/audioflinger/Threads.cpp
|
| 97502a19165de566eaf7525b6967f84adc8aabce |
|
20-Oct-2014 |
Marco Nelissen <marcone@google.com> |
am 9f38361f: am 5cc72274: am d61ba0aa: am 36d9256f: am ac302143: Fix the fix
* commit '9f38361fa0ba8fe04b9ba7e21bf25abe20620800':
Fix the fix
|
| 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
|
| 791854d962d8fd53dc267f73bfbd474a8887a1d1 |
|
20-Oct-2014 |
Marco Nelissen <marcone@google.com> |
am 412b8e71: am 0c2c1619: am 0cfc8f25: am 145a4c67: am 22bc10a3: Merge "Clear effect buffer before sending to chain" into lmp-dev
* commit '412b8e719e77e6e3284b87fd8ae1499d6ae76e06':
Clear effect buffer before sending to chain
|
| 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
|
| e49574c22849dd1f53cf78e97d22e61b80a04a6a |
|
03-Oct-2014 |
Eric Laurent <elaurent@google.com> |
am b17182d8: am 10b464cf: am 2b217bb3: Merge "audioflinger: fix pre processing transfer between record threads." into lmp-dev
* commit 'b17182d85f147a8d68bba2b4d7ed5e757691b4f4':
audioflinger: fix pre processing transfer between record threads.
|
| 2b217bb3aee87ce8486014f261c0f498f6209e80 |
|
03-Oct-2014 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: fix pre processing transfer between record threads." into lmp-dev
|
| e979d5bb0060ad4fb4fb9aa5aefc6f9b722a1bab |
|
03-Oct-2014 |
Glenn Kasten <gkasten@google.com> |
am 12d0cd1b: am 3a8bad2e: am be9ebd46: Merge "Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp" into lmp-dev
* commit '12d0cd1bf756ec96755fae2e573659898bfba6d6':
Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp
|
| 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
|
| 4113fe9dcbcda53d09f9b9c91f59d9a54e6c2408 |
|
02-Oct-2014 |
Glenn Kasten <gkasten@google.com> |
am 0e42a880: am e5095be1: am a49c94ea: Merge "Revert "Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp"" into lmp-dev
* commit '0e42a8807edec5541ecac4a050459fbb72c477dc':
Revert "Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp"
|
| 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
|
| ac2a2239dfc86c70f7382bebdb85c8cc1a9f6119 |
|
01-Oct-2014 |
Glenn Kasten <gkasten@google.com> |
am 785da8f5: am 955e24d3: Merge "Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp" into lmp-dev
* commit '785da8f542120e49d29289947e69aba180c1724b':
Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp
|
| a2dda4fcb3ab34a58c5218bb3a8a156cf04651b2 |
|
01-Oct-2014 |
Eric Laurent <elaurent@google.com> |
am e0270e79: am f71224ce: Merge "audioflinger: forward flush to direct output streams" into lmp-dev
* commit 'e0270e79066825bb232ef5f271ac9ef84ce91b0f':
audioflinger: forward flush to direct output streams
|
| 4cb7073c020fb7d37061be43347fa0a3a7da29ea |
|
30-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
am 00cd942e: am 4379c034: Merge "Fix tee sink" into lmp-dev
* commit '00cd942ebe4bf675cad1fbba0d09ba36e419b017':
Fix tee sink
|
| 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
|
| 137910189b005192ad4019894a970144f7377977 |
|
18-Sep-2014 |
Eric Laurent <elaurent@google.com> |
am 589e1241: am 2720c8c4: Merge "audioflinger: remove debug log." into lmp-dev
* commit '589e12415e10e291d48c19f63cb47193eb613fb0':
audioflinger: remove debug log.
|
| 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
|
| e716ae7a76941932535cd47ae454423dee598c1e |
|
16-Sep-2014 |
Eric Laurent <elaurent@google.com> |
am f5e98702: am cf7863ea: Merge "audioflinger: fix pre processing effect leak" into lmp-dev
* commit 'f5e9870228db68d9cb927ff49f2e46dabfb7bbec':
audioflinger: fix pre processing effect leak
|
| 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
|
| f4207cd4024e421bf51e5cd1457dd454f52be68a |
|
08-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
am b12d6a17: am 62ce2072: Merge "Make IPowerManager native conform to .aidl for oneway" into lmp-dev
* commit 'b12d6a17d33141789c974e3bd3eacfcb0be1c1e3':
Make IPowerManager native conform to .aidl for oneway
|
| 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
|
| 9d1cad2ba6a35168fa27a322518150193f19e53b |
|
03-Sep-2014 |
Eric Laurent <elaurent@google.com> |
am 3d640e0a: am 97d547da: audio flinger: fix use of global effect buffer
* commit '3d640e0a93e31bb53e671ed467774b63bb69750e':
audio flinger: fix use of global effect buffer
|
| 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
|
| efaa7ab620b2ee936efa4fb74dc4c670ed757a46 |
|
20-Aug-2014 |
Glenn Kasten <gkasten@google.com> |
Clear the timestamp latch valid flag if no new timestamp available
Bug: 11815245
Change-Id: I94885a1b4c2c7c055b8eacc7f7d0fcd064094b6d
/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
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| 81784c37c61b09289654b979567a42bf73cd2b12 |
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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
|