| f62e1a2887bedb440bb98c4e7071896dd3f46163 |
|
09-May-2018 |
Andy Hung <hunga@google.com> |
Fix PatchTrack retry for ClientProxy::obtainBuffer
Restore buffer mFrameCount as error clears it.
Test: phone call with usb device, simulate underrun.
Bug: 79423841
Change-Id: Ia00d08b6753b5c65cf7f07bf1c889fd6d1eefd0a
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 9378eb9634d07010bebf8a2d4bda84320ec80e57 |
|
03-May-2018 |
jiabin <jiabin@google.com> |
Restrict setting silenced in RecordTrack.
When initializing, always set mSilenced as false. Never set silenced for
patch track.
Bug: 77193174
Test: make phone call with usb headset multiple times.
Change-Id: I93f4646cc6d88749c602bdd0003afc48f7dec36f
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 331679c8a5cd21a0e6a7a1d851e965e4721f7420 |
|
17-Apr-2018 |
Eric Laurent <elaurent@google.com> |
Revert "Revert "audioflinger: implement silenced capture for mmap""
This reverts commit 611ebb95e76c230214780b7b583b8a3c5277ac7a.
Bug: 78118584
bug: 72134552
Test: 'adb shell write_sine_callback -m3 -pl -s100 -x' and Plug headphones in and out.
Test: AAudio CTS tests
Change-Id: I7558ad7b9c6c2d352ebce105382fe914c26fd35e
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 554e08c6b00b2a57348f4545d450915a9e8c870a |
|
17-Apr-2018 |
TreeHugger Robot <treehugger-gerrit@google.com> |
Merge "Revert "audioflinger: implement silenced capture for mmap"" into pi-dev
|
| 611ebb95e76c230214780b7b583b8a3c5277ac7a |
|
16-Apr-2018 |
Eric Laurent <elaurent@google.com> |
Revert "audioflinger: implement silenced capture for mmap"
This reverts commit a2f478ca0b2bda8db9b321f0cfa9e9d3cf6130f9.
Bug: 78118584
Change-Id: Ie0da3dea445f2474c7a889c5f9919305c1c96d40
/frameworks/av/services/audioflinger/Tracks.cpp
|
| c16f23e399969e9d82f0b245afb0870fa448e78e |
|
13-Apr-2018 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: implement silenced capture for mmap" into pi-dev
|
| fe463237ab61459ddc30120962c51fb089d25ec4 |
|
12-Apr-2018 |
TreeHugger Robot <treehugger-gerrit@google.com> |
Merge "AudioFlinger: input thread dump shows if RecordTrack is silenced" into pi-dev
|
| a2f478ca0b2bda8db9b321f0cfa9e9d3cf6130f9 |
|
11-Apr-2018 |
Eric Laurent <elaurent@google.com> |
audioflinger: implement silenced capture for mmap
Implement silencing of audio capture for idle UIDs for
MMAP streams.
Use audio HAL setMicGain() API to silence the mmap buffer for
exclusive mode and when only one client is active in shared mode.
Add new MmapStreamCallback method onStreamSilenced() for AAudio
service to selectively silence streams from silenced UIDs in
shared mode.
Bug: 72134552
Test: manual
Change-Id: I7c92ba0002bc5ba003c1a3c887edddf9ae53b538
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 7d665abb442a84d82760d88d228914a116e34532 |
|
12-Apr-2018 |
Jean-Michel Trivi <jmtrivi@google.com> |
AudioFlinger: input thread dump shows if RecordTrack is silenced
For each record thread, show if tracks are 's'ilenced
or 'n'ormal
Bug: 70304055
Test: during recording: adb shell dumpsys media.audio_flinger
look at active tracks under "Input thread"
Change-Id: I8b745bd5f8a9b6b228129dd8981acea7fb230acd
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 12381092fdb5810cb27f738435ea8a0e3d2d0f42 |
|
11-Apr-2018 |
Kevin Rocard <krocard@google.com> |
Revert "Revert "Audio V4: Send volume and duplicated track attr in update metadata""
The original patch was breaking the fast track volume, so it was
reverted.
In order to fix the original patch, revert the revert and in the
following patch apply the fix.
This makes review easier as only the difference to the original patch
will be present.
Bug: 38184704
Bug: 69623109
Bug: 77892019
Test: Patch already tested for original patch and is tested for following
patch that fixes this patch.
This reverts commit 80ee2722d64a73777fa76027b1568e61dec8d910.
Change-Id: I06914911c2435df167fff05ce5ca40dd1ed90f39
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 80ee2722d64a73777fa76027b1568e61dec8d910 |
|
11-Apr-2018 |
Kevin Rocard <krocard@google.com> |
Revert "Audio V4: Send volume and duplicated track attr in update metadata"
This reverts commit c86a7f7de52f0ff0059e8c6c4fa4619aa7fae5c5.
Reason for revert: fast track volume broken
Test: none, it is a revert
Bug: 77892019
Change-Id: If00616afae5c069cf0b6bd7c2da2a6d4af72180e
/frameworks/av/services/audioflinger/Tracks.cpp
|
| c86a7f7de52f0ff0059e8c6c4fa4619aa7fae5c5 |
|
03-Apr-2018 |
Kevin Rocard <krocard@google.com> |
Audio V4: Send volume and duplicated track attr in update metadata
Only the volume of MMAP thread was previously sent and
the attributes of duplicated tracks were skipped.
Bug: 38184704
Bug: 69623109
Test: check that the HAL receives correct track metadata volume
Test: check that the HAL receives correct duplicated track metadata
Signed-off-by: Kevin Rocard <krocard@google.com>
Change-Id: I1e3b14edd8c9a5a65d2b9f4cf8660997c81c701c
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 0967c6c06d030636e2c584ce3e24924357a3a854 |
|
05-Apr-2018 |
Andy Hung <hunga@google.com> |
Merge "AudioTrack: Prevent server from reading client data after stop" into pi-dev
|
| 1d3556d6a068f141ac390a04f6a9d92a70f8d7ed |
|
30-Mar-2018 |
Andy Hung <hunga@google.com> |
AudioTrack: Prevent server from reading client data after stop
Prior to this CL, if an AudioTrack client wrote audio data after
AudioTrack stop(), but before the track was drained by the server,
the newly written client data would be consumed in the drain.
We now limit the server read to the client write position on stop.
This interlocking is essential for rapid asynchronous AudioTrack
command processing.
Test: AudioTrack CTS, SoundPool looping, bug test case
Bug: 75788313
Change-Id: Ib70e3dc46afe047a8c6cf1fb906a618b3c66cc7f
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 1f564acca544826daaf7dca44e39cec6016b82fd |
|
29-Mar-2018 |
Kevin Rocard <krocard@google.com> |
Audio V4: Propagate audio attributes to tracks
This patch has not functional changes.
Bug: 38184704
Bug: 69623109
Test: play music and a audio
Change-Id: Ide6ec7de4dbfab4af2653475901c3b3daee36e52
Signed-off-by: Kevin Rocard <krocard@google.com>
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 1bc088a918d7038603230637d640941953b314d4 |
|
10-Feb-2018 |
Andy Hung <hunga@google.com> |
Call AudioMixer only from MixerThread threadLoop.
As part of change:
Remove track name offset by TRACK0.
Move track name management to the Tracks class.
Sync mixer track name to FastMixer track index.
Fixes regression introduced by commit 8ed196a.
Test: SoundPool, AudioTrack CTS, Usability
Bug: 72937362
Bug: 73004420
Change-Id: I2f1a33f6f0da66bcd7aa91e2a4b663ff822df645
/frameworks/av/services/audioflinger/Tracks.cpp
|
| fee1976a2849c37a53d8a01ac10327d522a1ba93 |
|
30-Jan-2018 |
Eric Laurent <elaurent@google.com> |
Revert "Revert "audio policy: refactor audio record APIs""
This reverts commit e9ebcdbb0580bd4a75f4530b957b1859e535c028.
Bug: 72628781
Test: manual camcorder tests
Change-Id: I9fb550a9f62d2d15cafbd7534e5e7eaa155a7213
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 5dd03a5825d06033c1e66353dbbfcfc7f60990c5 |
|
01-Feb-2018 |
TreeHugger Robot <treehugger-gerrit@google.com> |
Merge "Support query active microphones information in AudioRecord."
|
| e9ebcdbb0580bd4a75f4530b957b1859e535c028 |
|
29-Jan-2018 |
Eric Laurent <elaurent@google.com> |
Revert "audio policy: refactor audio record APIs"
This reverts commit 0f4b3c5449f85c1cd78e1b9ac4850de962b8edbe.
Bug: 72628781
Test: Capture several videos from camera app
Change-Id: I6bcc87c618ac9bfe5c911915cbb37de616bc727a
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 653cc0ab6d5b714d1f26adcd08f9bc3269b62fa0 |
|
18-Jan-2018 |
jiabin <jiabin@google.com> |
Support query active microphones information in AudioRecord.
This is part of device enumeration. With the new add API, developer
could get the active microphones information for each channel.
Bug: 64038649
Test: Run cts and check the print log.
Change-Id: Ic63d86e533a30e40697da7522a5a81f7cfcea988
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 0f4b3c5449f85c1cd78e1b9ac4850de962b8edbe |
|
05-Jan-2018 |
Eric Laurent <elaurent@google.com> |
audio policy: refactor audio record APIs
Refactor audio policy service APIs controlling audio record (startInput, stopInput, releaseInput)
To allow finer grain control per AudioRecord client and prepare for new concurrent capture
policy
Test: AudioRecord CTS test. manual test of capture use cases:
camcorder, OK Google VoIP
Change-Id: I4992fd5a115853c47be5dac5465989eb53f1287c
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 5ec161bb6845402ceed033c0f9d91127127b27e6 |
|
06-Dec-2017 |
Ivan Lozano <ivanlozano@google.com> |
Fix audioflinger in overflow sanitized builds.
The loop as constructed in Track::triggerEvents potentially leads to
two unsigned integer overflows on the i = 0 loop.
This refactors the loop to prevent the overflow.
Bug: 30969751
Test: Compiles and device boots.
Change-Id: I7ac3223ab3197f5c475a4d09c99e6f05d0ddb208
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 94a1ee822686e920a33e312f4032f991731aea07 |
|
22-Jul-2017 |
rago <rago@google.com> |
Converting effect chains to float point by default.
Removed additional conversions from/to int16/float.
Fixed config file restrictions.
Detection of native format (int16/float) of effects.
On-the-fly conversion to/from 16 bit effects if necessary.
WIP: testing effects in all possible configurations.
Bug: 63935479
Test: Bufflog dumps, frequency analysis comparison, SoloTester
Change-Id: Id4cabc5c5698befc90a5d68b0b996a9b9adebfc4
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 8cf3a0788df2ee184b498086b1b8da322eeadcca |
|
09-Aug-2017 |
Ivan Lozano <ivanlozano@google.com> |
Generate the IPlayer interface with AIDL.
This removes the manually rolled IPlayer implementation and instead uses
an aidl generated implementation. The IPlayer aidl file is moved from
frameworks/base/.
Test: run cts --module CtsMediaTestCases
Bug: 64223827
Change-Id: Ia609f7ba1cc6f71ce919d6174dd8b5e0e1b05575
/frameworks/av/services/audioflinger/Tracks.cpp
|
| ff6900d8f991aac0b67b625f8a1d4d0461b4c50e |
|
02-Aug-2017 |
Ivan Lozano <ivanlozano@google.com> |
Generate IAudioRecord with AIDL.
Use AIDL to generate the IAudioRecord interface rather than using
a manually implemented interface.
Bug: 64223827
Test: Ran the AudioNativeTest and CtsMediaTestCases CTS tests.
Change-Id: Iade23c73253a97c358050bbd697ebe592b89a4c0
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 2b01f009428797ee9154649411a64b522946e966 |
|
05-Jul-2017 |
Andy Hung <hunga@google.com> |
Replace old audio time conversion macros and functions
Test: Play Movies, Youtube, CTS
Bug: 25569135
Change-Id: Ib6e304f149c381dc7af7f4aa959c3d79d7bdb2aa
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 2c6c3bb76b2fa18ed3536331a1700250bd04c7ce |
|
16-Jun-2017 |
Andy Hung <hunga@google.com> |
AudioFlinger: Log Record and MMap thread events
Also update track dump for better grouping and consistency.
Test: adb shell dumpsys media.audio_flinger
Bug: 62700410
Bug: 62701585
Change-Id: I3a6e76eac871cb08ff14d45f7d3992a6e7fba045
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 8fe68036c2a670abb9eb0cc67ab9830c2c23de81 |
|
06-Jun-2017 |
Andy Hung <hunga@google.com> |
Track: Check buffer size
Test: Native POC
Bug: 38340117
Change-Id: Ia7edd9a802905214a27961dbcec6352f6ef98f73
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 8adc80866b6cc1bb6be62c1998dd424f1f339dea |
|
07-Jun-2017 |
Yunlian Jiang <yunlian@google.com> |
Fix a clang-analyzer-cplusplus.NewDelete warning.
This fixes warning: Argument to 'delete' is the address of the local
variable 'inBuffer', which is not memory allocated by 'new'
[clang-analyzer-cplusplus.NewDelete]
Bug: none
Test: The warning is gone.
Change-Id: I6c954a679ba22e0d7859caaafce24f4e50b0391d
/frameworks/av/services/audioflinger/Tracks.cpp
|
| ab37fbfcdc197429d59f6926ba970b3e68bbbadf |
|
05-May-2017 |
Mikhail Naganov <mnaganov@google.com> |
Merge "audioflinger: Do not report underruns for stopped or paused tracks" into oc-dev am: bc4040efd0
am: 8c9796628b
Change-Id: I819e1361dd3245f35fe06691a4758fe43febc3d0
|
| a66d3899f7d13e69ee451f1071f0d999678803ce |
|
04-May-2017 |
Mikhail Naganov <mnaganov@google.com> |
audioflinger: Do not report underruns for stopped or paused tracks
PlaybackThread::Track is missing checks for the track state
when reporting underruns. These checks present on other instances
when tallyUnderrunFrames is being called.
This eliminates fake underrun reportings when playing
multichannel tracks via deep buffer.
Bug: 36181621
Test: play multichannel track and check underrun count at the end
Change-Id: Ieff338b33807e66df91234d85b782ee383fe95b9
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 93d7b51f52a9a5b44d4873f69feefc1709d872c8 |
|
25-Apr-2017 |
Andy Hung <hunga@google.com> |
Merge "VolumeShaper: improve logging and dumpsys" into oc-dev am: 9706527f86
am: db59802a59
Change-Id: I119a0c5cf44a9efaa0190c33a302730a4c756970
|
| da540db0fc21bc9319d9602aefe1a109d00a7e6c |
|
20-Apr-2017 |
Andy Hung <hunga@google.com> |
VolumeShaper: improve logging and dumpsys
Test: CTS and dumpsys
Bug: 37536598
Change-Id: I9a838a8605defde3aaf42a872e48f01425df0f0e
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 5b2191a4a36746b66ba3e8f73b711f49fdb83b4b |
|
19-Aug-2016 |
Glenn Kasten <gkasten@google.com> |
Add a tee sink filename suffix to distinguish type
Test: tee sink filenames have a _C (capture), _P (playback), or _T (track) suffix
Change-Id: Ib8ddadb0f3e3ba3ac620e33072bb212d5cde4978
/frameworks/av/services/audioflinger/Tracks.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/Tracks.cpp
|
| 09cf7cab168663ca98a440a9473b1515e0eb0cfa |
|
15-Feb-2017 |
Andy Hung <hunga@google.com> |
Merge "AudioFlinger: Fix memory allocation for client-less tracks"
|
| afb31487f3156a7284d2f0d06646c7bc00d99537 |
|
14-Feb-2017 |
Andy Hung <hunga@google.com> |
AudioFlinger: Fix memory allocation for client-less tracks
Test: Ringtone with BT
Bug: 35350587
Change-Id: If247d319d58f8f4d18b49f58ec950491871ebb2d
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 1883f69de5f2c4e71df58d5b71d7c39f9779b50c |
|
14-Feb-2017 |
Andy Hung <hunga@google.com> |
AudioFlinger: Check framecount overflow when creating track
Test: Native POC
Bug: 34749571
Change-Id: I7529658e52ac7e64d162eb5338f10fb25eaa8fe7
/frameworks/av/services/audioflinger/Tracks.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/Tracks.cpp
|
| 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
|
| 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/Tracks.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/Tracks.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/Tracks.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/Tracks.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/Tracks.cpp
|
| 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/Tracks.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
|
| 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/Tracks.cpp
|
| e37736428f5393ea5f32729ad16f59c2c297c04f |
|
14-Sep-2016 |
Phil Burk <philburk@google.com> |
AudioFlinger: flush direct tracks that are paused am: 4bb650bb3c am: b4fb66000f
am: 1935f31023
Change-Id: I7e6e7e73f2dd77c91995570c8719ab9c8e5561e9
|
| 4bb650bb3c4f15789232f3f3cf778c5441122360 |
|
09-Sep-2016 |
Phil Burk <philburk@google.com> |
AudioFlinger: flush direct tracks that are paused
Normally the flush is processed on ObtainBuffer.
But that is not called when the track is paused.
So it was not possible to flush and re-prime a
track that was paused.
Now we flush synchronously for inactive tracks,
and flush from prepareTracks_l for active tracks.
Bug: 19193985
Bug: 27791443
Change-Id: I39a7e4921e45041c3a51cf91fd3995b5edee6dd4
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/services/audioflinger/Tracks.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
|
| 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/Tracks.cpp
|
| 9a3fbd9f84656ca25acdfd202ef7d4461a640f74 |
|
04-Jun-2016 |
Chih-Hung Hsieh <chh@google.com> |
Fix misc-macro-parentheses warnings in Tracks.cpp.
Bug: 28705665
Change-Id: Ifa59d95b0242bef71762d862e3437fa1e914c552
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 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
|
| 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/Tracks.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/Tracks.cpp
|
| 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
|
| 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/Tracks.cpp
|
| 8d9286e4d75b10da715e955ba91ba6fa2a2ae030 |
|
13-Apr-2016 |
Andy Hung <hunga@google.com> |
Merge "AudioTrack: Reset position on flush for extended timestamps." into nyc-dev am: aff4e46
am: 9f0fdd1
* commit '9f0fdd1edd70ab31c3b7d9b687afb44e83b868b7':
AudioTrack: Reset position on flush for extended timestamps.
Change-Id: Id2b09b1cfd21fcf4ec7c9d040f0f199f94e35c7a
|
| e1e9846a786a0d77a13b44f8e798e03b4cba9d77 |
|
12-Apr-2016 |
Andy Hung <hunga@google.com> |
AudioTrack: Reset position on flush for extended timestamps.
Bug: 27940058
Change-Id: If91320669e2e84803e601137b1ce66b361b129f1
/frameworks/av/services/audioflinger/Tracks.cpp
|
| adc4827a835087a2a7edd934967bc059aa0ee743 |
|
04-Apr-2016 |
Andy Hung <hunga@google.com> |
Merge "Implement client playback timestamps with 64 bit accuracy" into nyc-dev am: f3bcfeb
am: ba27fff
* commit 'ba27fff473a122a9d2ad3ac852eceb61002de3c6':
Implement client playback timestamps with 64 bit accuracy
Change-Id: I55f278982c03360a7c2e06f1e3810548afdda647
|
| ea2b9c07b34079f0dbd8610a511e006e69a15adc |
|
13-Feb-2016 |
Andy Hung <hunga@google.com> |
Implement client playback timestamps with 64 bit accuracy
Provide server timestamps if the HAL doesn't provide it.
Provide monotonic - boottime translation.
Bug: 17472992
Bug: 26682703
Bug: 27749434
Change-Id: I6c9b213d9f9284092e34d57f52870e02c72df62a
/frameworks/av/services/audioflinger/Tracks.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/Tracks.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/Tracks.cpp
|
| fc1337d079ba7e6ffd7ce07706cfeeeb355c0200 |
|
18-Mar-2016 |
Phil Burk <philburk@google.com> |
Merge "AudioTrack: fix stall if setBufferSizeInFrames() called before play()" into nyc-dev
|
| e8972b0a27fac327c997fceb20d9abae1dc1d899 |
|
04-Mar-2016 |
Phil Burk <philburk@google.com> |
AudioTrack: fix stall if setBufferSizeInFrames() called before play()
The server was waiting for a full buffer.
But the buffer was only getting partly filled.
So the stream was not starting.
The fix involves having the server look at the adjustable threshold.
Bug: 27505889
Change-Id: I5dbf686413e670dacbbecc9e0f838744e465f44f
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/services/audioflinger/Tracks.cpp
|
| d848eb48c121c119e8ba7583efc75415fe102570 |
|
08-Mar-2016 |
Glenn Kasten <gkasten@google.com> |
Use audio_session_t consistently
Bug: 27562099
Change-Id: I328d8226191386b163f2ace41233607294c50dcd
/frameworks/av/services/audioflinger/Tracks.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/Tracks.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/Tracks.cpp
|
| c54b1ffc92b8ad27608a8af21033d7cab33cb3a0 |
|
23-Feb-2016 |
Andy Hung <hunga@google.com> |
Fix AudioTrack presentationComplete regression
Bug: 27310929
Change-Id: Id52299660d90b07220a4fd9996da53fe755a41f1
/frameworks/av/services/audioflinger/Tracks.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/Tracks.cpp
|
| 6ae5843c281301a9ffd1059d185620a9337e15a2 |
|
17-Feb-2016 |
Andy Hung <hunga@google.com> |
Enable FastTrack timestamps
Bug: 26413951
Change-Id: Id54c32cb0f0be83d443f6cf1007181ebc461c586
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 8ce8e264d0bf09ec131e4a7701fe1ebc3a678f8b |
|
16-Feb-2016 |
Eric Laurent <elaurent@google.com> |
Revert "Enable FastTrack timestamps"
This reverts commit b3a486fecddc9e9264fe63d356eca0812493db27.
Reverting while investigating the root cause.
Bug: 27186414
/frameworks/av/services/audioflinger/Tracks.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/Tracks.cpp
|
| e99556b85081778027653bd3e716a529ab518b35 |
|
12-Feb-2016 |
Andy Hung <hunga@google.com> |
Merge "Fix audio timestamp computation for pause, stop, and dynamic speed changes" into nyc-dev
|
| b3a486fecddc9e9264fe63d356eca0812493db27 |
|
05-Aug-2015 |
Andy Hung <hunga@google.com> |
Enable FastTrack timestamps
Bug: 26413951
Change-Id: I99a69241a0bcd69f6bfebf785cac6d9ee1e2cd5f
/frameworks/av/services/audioflinger/Tracks.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/Tracks.cpp
|
| 3f0c902beb53a245c9db35e871607dba05b8d391 |
|
16-Jan-2016 |
Andy Hung <hunga@google.com> |
Add AudioRecord timestamps
Bug: 13569372
Bug: 22886739
Change-Id: Ibc81afefb733d23676a632a0f2da31163fdbe05f
/frameworks/av/services/audioflinger/Tracks.cpp
|
| d79072e9dff59f767cce2cda1caab80ce5a0815b |
|
06-Jan-2016 |
Glenn Kasten <gkasten@google.com> |
Remove TimedAudioTrack and associated code
Bug: 8278435
Change-Id: I095c1a4888e645e14d93b0b15fbef4524a831ca1
/frameworks/av/services/audioflinger/Tracks.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/Tracks.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/Tracks.cpp
|
| e6fb82a207f5256933e2d83e77262331af50a27f |
|
09-Sep-2015 |
Andy Hung <hunga@google.com> |
For static obtainBuffer(), do not set mUnreleased if acknowledging flush.
static audio tracks use obtainBuffer() to update position in start().
Bug: 22938515
Change-Id: I8ae32f6cce4d122386d2cf8982e158049b04ba9a
/frameworks/av/services/audioflinger/Tracks.cpp
|
| a542782d0045588e55e075a763cedcd2d504ad22 |
|
12-Sep-2015 |
Andy Hung <hunga@google.com> |
Workaround static fast track start-after-stop issue
Bug: 23291988
Bug: 23614327
Bug: 23924081
Change-Id: Id1a519ed4bb2a6f4cb197da8450f7069b55c0d48
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 564d144fc28917d42e3a67718ac51d61bfc36315 |
|
09-Sep-2015 |
Eric Laurent <elaurent@google.com> |
Revert "Do not acknowledge flush() on start() for static tracks"
This reverts commit cafe86a9cb6625bb1ec6383e16e28e4c9e455f87.
Bug: 23924093.
Change-Id: I186d1013b06a286eca93c30bb9b3545dc36695ff
/frameworks/av/services/audioflinger/Tracks.cpp
|
| cafe86a9cb6625bb1ec6383e16e28e4c9e455f87 |
|
14-Aug-2015 |
Andy Hung <hunga@google.com> |
Do not acknowledge flush() on start() for static tracks
Bug: 22938515
Change-Id: I1de653de169a3fbbaa693da6057897ea57772447
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 4c6afaf1053ecbfda6d95098e0f49772ecbcf2e1 |
|
13-Jun-2015 |
Andy Hung <hunga@google.com> |
Fix AudioRecord cleanup if construction fails
Bug: 11671194
Change-Id: Id48350264f8b5ce53bf7ce2b308bf4ba21e59241
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 1b42097f38e72574ed853a35f4e8a66e4739c421 |
|
22-Apr-2015 |
Phil Burk <philburk@google.com> |
AudioTrack: fix spurious retrograde messages
The retrograde motion was confused by some positions coming from
the DSP on offloaded tracks. So the retrograde check was moved up
into AudioTrack.cpp. This allows us to take advantage of the checks
for invalid positions based on timing.
Bug: 2047891
Change-Id: Ifcad2349201443a7f1711347c203297100449536
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 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/Tracks.cpp
|
| 8edb8dc44b8a2f81bdb5db645b6b708548771a31 |
|
27-Mar-2015 |
Andy Hung <hunga@google.com> |
Add playback rate to AudioTrack
Bug: 19196501
Change-Id: I6411e1d3ce652b711a71a6d9df020cb5f60d4714
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 86f79e3f3bc1de7b4e4dd045de8962b8a62ab5d5 |
|
09-Apr-2015 |
Andy Hung <hunga@google.com> |
Merge "Improve ResamplerBufferProvider"
|
| 73c02e4277b399c2ec1555d32b6ad5df23bb83dc |
|
29-Mar-2015 |
Andy Hung <hunga@google.com> |
Improve ResamplerBufferProvider
Change-Id: I3cc3af221ad5797ff219d75227350733afa180db
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 77a44e8d1f902a100500ea69a450e22889dcff31 |
|
09-Apr-2015 |
Andy Hung <hunga@google.com> |
Merge "Add RecordBufferConverter for RecordThread data processing"
|
| 97a893eb34f8687485c88eaf15917974a203f20b |
|
29-Mar-2015 |
Andy Hung <hunga@google.com> |
Add RecordBufferConverter for RecordThread data processing
Change-Id: Ia3aab8590cd41e8a7cba0a7345d70d2866d92045
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 3bcf8597189c592713675ec58326ecbef0ac4ae9 |
|
03-Apr-2015 |
Eric Laurent <elaurent@google.com> |
audio flinger: improve device to device audio patches
Allow creation of audio patches between input and output
devices managed by the same audio HW module.
Change-Id: I4b83268a4d5c41f3d5905d7581202cf5193efd32
/frameworks/av/services/audioflinger/Tracks.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/Tracks.cpp
|
| 6140c79c31f5dc237fba69554de5be641cedf113 |
|
19-Mar-2015 |
Phil Burk <philburk@google.com> |
AudioFlinger: prevent retrograde timestamps for direct mode
Prevent retrograde motion in the HAL or drivers from making it
up through the AudioFlinger to the app.
Bug: 19604395
Change-Id: I11d6c0f7cab69e75fd4b4a2a90f64960b4bb6d5a
Signed-off-by: Phil Burk <philburk@google.com>
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 240934ba80d6c6165749db7681d243c6857a092f |
|
12-Mar-2015 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: refresh fast track underrun state upon start
False underrun is detected when starting recycled fast tracks, which
leads to continuous fatal assertion failures and even AP reboot.
Track's last mObservedUnderruns isn't updated one at previous stop()
call. Hence, when we start the same track again, we should synchronize
it to the latest state instead of relying on stale one.
authored-by: Weiyin Jiang<wjiang@codeaurora.org>
CRs-Fixed: 803389
Bug: 19635137
Signed-off-by: Andy Hung <hunga@google.com>
Change-Id: Ia003a49c6896dba965798c062c98b8c367ef8369
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 63238efb0d674758902918e3cdaac322126484b7 |
|
03-Mar-2015 |
Glenn Kasten <gkasten@google.com> |
Remove redundant semicolon from namespace closing
Change-Id: I163f9d3d216c283ae1160ce4802e5247cf44fba7
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 7841f704fee0e5acd45fa6e47a336120c3cad42c |
|
27-Feb-2015 |
Andy Hung <hunga@google.com> |
Merge "Do not pre-fill OutputTrack buffer for duplicating thread"
|
| ad8510a339ffab330c2c46e5c247dd1cf9e15c22 |
|
18-Feb-2015 |
Glenn Kasten <gkasten@google.com> |
Include what you use
Change-Id: I1967861f383bb5ed6743fb69e3bd439907ed7033
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 5bedff60b2facaa1ec5b9433647ebf1504f065ca |
|
16-Jan-2015 |
Andy Hung <hunga@google.com> |
Do not pre-fill OutputTrack buffer for duplicating thread
It is not necessary to start the downstream MixerThread early,
rather allow stop() to flush the remaining frames.
Change-Id: Ibcf8e7f5eda09d0f915d50697ff0c24c32fd11c4
/frameworks/av/services/audioflinger/Tracks.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/Tracks.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
|
| 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/Tracks.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.
|
| 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
|
| 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/Tracks.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/Tracks.cpp
|
| 847cbecbfda35e27a73a767c93fd0150ba001882 |
|
18-Nov-2014 |
Andy Hung <hunga@google.com> |
am a248703b: am a3e2c146: am 4d0a5985: Merge "Prevent stopped audio tracks from mixing data" into lmp-mr1-dev
* commit 'a248703b0799a9a27481f0ec13df90214a661eb5':
Prevent stopped audio tracks from mixing data
|
| 27876c02c21cd3ab7ef01bdd1fc5400c5143584a |
|
10-Sep-2014 |
Andy Hung <hunga@google.com> |
Prevent stopped audio tracks from mixing data
Static tracks should have zero framesReady when stopping or stopped,
to prevent mixer from running potentially indefinitely with loop mode.
Bug: 16070883
Change-Id: Iab9968044d1a6f8543bbcbfb246f49f27d68cace
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 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
|
| 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/Tracks.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/Tracks.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
|
| 955e24d3a8e218d4711cabc6558781e095011132 |
|
30-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix uncertainty of one normal mix buffer in AudioTrack::getTimestamp" 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/Tracks.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/Tracks.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/Tracks.cpp
|
| 648c1ecee3f79ff9d754dd982be59484de08d752 |
|
03-Sep-2014 |
Glenn Kasten <gkasten@google.com> |
am 955d8f90: am 23c8cf95: Merge "Restore the ability to build tee sink for debugging" into lmp-dev
* commit '955d8f906e920fe8da66e7943a7f2bfa98422e19':
Restore the ability to build tee sink for debugging
|
| 329f6511ee4e03a4605c70bbda8d3a96d2544884 |
|
29-Aug-2014 |
Glenn Kasten <gkasten@google.com> |
Restore the ability to build tee sink for debugging
This changelist does not enable tee sink, but makes it possible to do so.
Tee sink had suffered some bit rot since it is not built by default.
Also fixes a crash for > 2 byte per sample or > 2 channels.
Still does not write correct header for floating-point;
that will be best solved by moving to libsndfile.
Bug: 16990102
Change-Id: I8e92c588ccc513d7802d696fcfb324e815772df6
/frameworks/av/services/audioflinger/Tracks.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/Tracks.cpp
|
| c2730ba7c5e9559b7499ef5e0d7742deb18c5110 |
|
21-Jul-2014 |
Eric Laurent <elaurent@google.com> |
audio policy: add support for USB devices for voice call
Add possibility to use sink and source devices not on
the primary HW module for voice calls (e.g. USB headsets).
Bug: 15520724.
Change-Id: Ib27db4ba759b6d91ea1104dc2e35c87733517b30
/frameworks/av/services/audioflinger/Tracks.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/Tracks.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/Tracks.cpp
|
| 529c61b7e4468a3e21f302f2a92a660249daa722 |
|
19-Jul-2014 |
Glenn Kasten <gkasten@google.com> |
Fix copy/paste bug
introduced by Change-Id: I4e9b7fa99858b488ac98a441fa70e31dbba1b865
Change-Id: I9a0c8800f5cf3767cc259776f95c96cad972e392
/frameworks/av/services/audioflinger/Tracks.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/Tracks.cpp
|
| 5e58b0abe5b6c8f5bd96a8f78bbeeeb4d3892020 |
|
24-Jun-2014 |
Andy Hung <hunga@google.com> |
Add floating point volume handling to AudioMixer
Use floating point volume in AudioMixer mixing when floating
point input is used with the new mixer engine.
AudioResampler is updated to take floating point volume to match.
Both legacy integer and floating point mixer engines work.
For now, integer volume is used when the new mixer engine
runs in integer input mode, for backward compatibility with
the legacy mixer. The new mixer engine will generally run in
floating point input mode. When the legacy path is removed,
the integer volumes will be removed.
Change-Id: I79e80c292ae7c8b8bdd0aa371a1b2c3a1b618290
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 3348e36c51e91e78020bcc6578eda83d97c31bec |
|
07-Jul-2014 |
Andy Hung <hunga@google.com> |
Replace int bitDepth with audio_format_t in Resampler
Remove mBitDepth from class (not used).
Replace with audio_format_t in factory method to distinguish
between float and pcm 16-bit.
Change-Id: I166860796c68285077ef4458d8758d19b82523f9
/frameworks/av/services/audioflinger/Tracks.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/Tracks.cpp
|
| 3bcffa136909c1fb6e88ee4efd12ccac18360a85 |
|
13-Jun-2014 |
Eric Laurent <elaurent@google.com> |
force command flush after releasing a reference to shared memory
After releasing a reference to a shared memory interface,
IPCThreadState::self()->flushCommands() must be called to ensure
that the corresponding reference is also cleared immediately in the
remote process. Otherwise the binder implementaiton will not
push the clear systematically causing the memory region not to be freed.
See issues 10711502 and 2801375.
Bug: 14057166.
Change-Id: If55f36f00d452d6cf00cf83bd2fba1c8f3abcb57
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 0ac53364b79dc72ee0c09f0efc46a22ab4374dbf |
|
11-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix non-monotonic AudioTrack::getTimestamp after pause" into lmp-preview-dev
|
| ced6e74215937182fe2f9f6b0867f7c28ccd02c1 |
|
10-Jun-2014 |
Glenn Kasten <gkasten@google.com> |
Fix non-monotonic AudioTrack::getTimestamp after pause
Bug: 15523502
Change-Id: Ifd4aa7fca197bc041c1620fc3f7d953a8902551a
/frameworks/av/services/audioflinger/Tracks.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/Tracks.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/Tracks.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/Tracks.cpp
|
| b4db73d022f3de3530bc2b3c9c831ccfdd1a2ead |
|
20-May-2014 |
Andy Hung <hunga@google.com> |
Merge "Use new channel count functions for audio masks"
|
| 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/Tracks.cpp
|
| e541269be94f3a1072932d51537905b120ef4733 |
|
16-May-2014 |
Andy Hung <hunga@google.com> |
Use new channel count functions for audio masks
Change-Id: Ia658ab4b6320d19fdb50f123c930918724ff0ef3
Signed-off-by: Andy Hung <hunga@google.com>
/frameworks/av/services/audioflinger/Tracks.cpp
|
| deb9081a61b446dbe50461082974a4228899f178 |
|
14-May-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Pull up mFlags and isFastTrack from Track to TrackBase"
|
| 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/Tracks.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/Tracks.cpp
|
| fe9570c7b937b49d3603ccb394aed732b79bc6be |
|
07-May-2014 |
Glenn Kasten <gkasten@google.com> |
Remove obsolete IAudioRecord::getCblk()
Change-Id: Id20b5efd765b9796b0e391610e06dc928a829ebf
/frameworks/av/services/audioflinger/Tracks.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/Tracks.cpp
|
| 1a7be1ec9c769203e6c8e26378de0ab8e2ad493d |
|
29-Apr-2014 |
Bill Yi <byi@google.com> |
Merge commit '91820d46b5f3065c2fded3cdf65d305715b33bb1' into HEAD
|
| 1b24cf32e02a1372e57b3a2a432bc0adfe496e6d |
|
16-Apr-2014 |
Mark Salyzyn <salyzyn@google.com> |
am e80631aa: am 839d11d1: Merge changes I0a744dc7,Id993a70d
* commit 'e80631aa1992ca50af679cd6a018c0ffda7f9b17':
media: use size_t for integer iterator to Vector::size()
media: 64 bit compile issues
|
| e80631aa1992ca50af679cd6a018c0ffda7f9b17 |
|
16-Apr-2014 |
Mark Salyzyn <salyzyn@google.com> |
am 839d11d1: Merge changes I0a744dc7,Id993a70d
* commit '839d11d1f7be9dff2f06c7d30a9eb39cb6782078':
media: use size_t for integer iterator to Vector::size()
media: 64 bit compile issues
|
| 3ab368e0810d894dcbc0971350c095049478a055 |
|
15-Apr-2014 |
Mark Salyzyn <salyzyn@google.com> |
media: use size_t for integer iterator to Vector::size()
Change-Id: I0a744dc7815a86a993df9b0623440be620ec8903
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 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/Tracks.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/Tracks.cpp
|
| 164985121796cf214c7a83d32005d9b01125b558 |
|
18-Mar-2014 |
Eric Laurent <elaurent@google.com> |
AudioFlinger: Change track ready condition in STOPPING_1 state
Set FS_FILLED only if framesReady() > 0
Change-Id: Ibb1d2e988ff17fcf3a7ab61031a3f85df82f18d5
/frameworks/av/services/audioflinger/Tracks.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/Tracks.cpp
|
| 3ef14ef30359376006a233f6a21a165d4b65a7df |
|
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Make error handling more similar for output and capture
And simplify error case
Change-Id: I0bb1ec252945d672cc4cef137977b912f1b23d51
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 05997e21af6c4517f375def6563af4b9ebe95f39 |
|
13-Mar-2014 |
Glenn Kasten <gkasten@google.com> |
Whitespace and comments
Change-Id: I99752cca74a459c4746ea4718acb0115439e50bd
/frameworks/av/services/audioflinger/Tracks.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
|
| 0504a198d6595e29e58d1d66ae59fac1d662384a |
|
05-Mar-2014 |
Haynes Mathew George <hgeorge@codeaurora.org> |
am 632c3790: am 7c206e5b: am e0cd1051: audioflinger: update track ready condition
* commit '632c3790523896260dc43fd6cd40a158765cdb15':
audioflinger: update track ready condition
|
| e0cd1051ed9fea0629745c29020516ae62298461 |
|
28-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: update track ready condition
Signal track ready if the track isStopping().
Bug: 12423190
Change-Id: I95e14905df10ebf301e398263478c8ca25d7e2ce
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 25f4aa83efaa9179e65a20583a6d441de2c3ff3f |
|
07-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Move handleSyncStartEvent and clearSyncStartEvent to RecordTrack
Change-Id: I611a63e63acf3eb17e2072c1a77b91fe9e62e36a
/frameworks/av/services/audioflinger/Tracks.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/Tracks.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/Tracks.cpp
|
| 9601c6efcb2552960d6f125d073525b581c1b7ec |
|
10-Feb-2014 |
Marco Nelissen <marcone@google.com> |
Merge "Pretty up audioflinger dumpsys"
|
| b220884bf3129253cc5bc8d030bc475411ea4911 |
|
07-Feb-2014 |
Marco Nelissen <marcone@google.com> |
Pretty up audioflinger dumpsys
Change-Id: I57e44b4c36b99f7149542bbcf9645521c6152dfa
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 8f762d577f2dcf34357eb8731070ae465649ec9a |
|
08-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix whitespace"
|
| a793625031a3e326a261a0843598c35a5784bbde |
|
07-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Use Format_isValid() and Format_isEqual() instead of direct comparison"
|
| 4944acb7355b3aa25748fd25945a363a69d65444 |
|
19-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Fix whitespace
Change-Id: I9add0872030a26933f6b6dad1be299154611cc56
/frameworks/av/services/audioflinger/Tracks.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/Tracks.cpp
|
| 6e0d67d7b496ce17c0970a4ffd3a6f808860949c |
|
31-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Use Format_isValid() and Format_isEqual() instead of direct comparison
Change-Id: Ie87607aa514976947540a77775e6425c4e56e7d9
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 0f11b51a57bc9062c4fe8af73747319cedabc5d6 |
|
01-Feb-2014 |
Glenn Kasten <gkasten@google.com> |
Fix unused parameter warnings in audio
Change-Id: I665ba3358dd9502f0adec70d486e7bf8a2e1b0fe
/frameworks/av/services/audioflinger/Tracks.cpp
|
| f0002d142e6d24c5438600b2c259679de710f8ac |
|
24-Jan-2014 |
Glenn Kasten <gkasten@google.com> |
Merge "Replace control block frameCount_ by explicit in/out parameter"
|
| 0bcfa88149e2404b34d13c622e3921e1b846cdf8 |
|
28-Dec-2013 |
Haynes Mathew George <hgeorge@codeaurora.org> |
audioflinger: update track ready condition
Signal track ready if the track isStopping().
Bug: 12423190
Change-Id: Ie279995d5f90fa8211a20dfbeacc1cf9b921d0bf
/frameworks/av/services/audioflinger/Tracks.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/Tracks.cpp
|
| 663c2247b71086e30bfd3192979d1dd7f15c539e |
|
24-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Consistent error checking for sp<IMemory> and pointer()
There have been concerns that an sp<IMemory> could be non-0, but the
associated pointer() still be NULL. There are rumors this may happen
when a non-0 sp<IMemory> is passed in by client but the shared memory cannot
be re-mapped into mediaserver.
There's also evidence in the early (2009/03/03) pre-git code of checking
pointer() for NULL, after a local allocate() returned a non-0 sp<IMemory>.
It's not clear if this is "cargo cult" paranoia, or if there was a
genuine reason for the check.
In any case, we now consistently check pointer() for sp<IMemory>
input parameters in createTrack() and queueTimedBuffer().
We also check after successful allocate(). If allocate() returns a
non-0 sp<> but NULL pointer(), then treat it as if the allocate() had
returned 0.
Change-Id: I3013ac5766b493d443ecef71711ec861076a623e
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 30ff92cba19c5acd747631365db1e1084e45ab34 |
|
20-Nov-2013 |
Glenn Kasten <gkasten@google.com> |
Code simplification for timed allocate
Change-Id: I5acccd7418c5b4a4dc0112147c750fef9f74bf33
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 1fc6cb5be26734938b3999e6bf4ce0e625e7d284 |
|
30-Oct-2013 |
Eric Laurent <elaurent@google.com> |
am ce275b7d: am 1c594b63: Merge "audioflinger: enable effects after registration if needed" into klp-dev
* commit 'ce275b7dd20e3d4781a46126c753e6eb998e6c0f':
audioflinger: enable effects after registration if needed
|
| 1c594b637df26499ce1dae2db34f2b3290efd838 |
|
30-Oct-2013 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: enable effects after registration if needed" into klp-dev
|
| 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/Tracks.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/Tracks.cpp
|
| a153b68f2260a8ed7fbb236fa659b13264ac5ac0 |
|
19-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
am 77177fa2: am 95c03858: Merge "Allow releaseBuffer after flush" into klp-dev
* commit '77177fa20773d02b4f9c4147ecb98107f019fa7d':
Allow releaseBuffer after flush
|
| 2e422c472c91aa7912befd0fc038d1e11f354bc1 |
|
18-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
Allow releaseBuffer after flush
After AudioTrack start checks for pending flush,
allow releaseBuffer on any previously obtained buffer.
For example, this can happen if the resampler has obtained
a buffer but not released the whole buffer yet.
Note that the resampler will be reading obsolete data.
Bug: 11285590
Change-Id: I0614fbb62e43604aac3089cce4b7797c87a306b5
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 9138d32050411c04cc45c25094352cf8b6e01df9 |
|
18-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
am e45796de: am dedc7b0f: Merge "Fix flush() followed by start()" into klp-dev
* commit 'e45796dee9c072cdb13a00b407e68be2cd55e2f2':
Fix flush() followed by start()
|
| 12022ff8d223ccf4dfae019c81e4a93a3f44985c |
|
17-Oct-2013 |
Glenn Kasten <gkasten@google.com> |
Fix flush() followed by start()
Bug: 11259166
Change-Id: I72526cdfd4e3ff4b06e6a317e577f57cc285205d
/frameworks/av/services/audioflinger/Tracks.cpp
|
| d72b7c0180ee83fc3754629ed68fc5887a125c4c |
|
13-Oct-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: enable effects after registration if needed
This fixes a bug where effects were not properly reenabled after
being moved from one playback thread to another. The effect
is enabled but the audio policy manager sees it as disabled.
Bug: 11181933.
Change-Id: I19cac7acbaf61c546e667fd85ed7d4eda1c716d8
/frameworks/av/services/audioflinger/Tracks.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
|
| 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/Tracks.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/Tracks.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/Tracks.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/Tracks.cpp
|
| 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
|
| 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/Tracks.cpp
|
| f523897787b64cf2217d0ab4946b564420c675fe |
|
12-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
am 8a220740: am dc33c542: Merge "Fix AudioTrack shared memory leak" into klp-dev
* commit '8a220740cd77eb844123d3914190a94513797b40':
Fix AudioTrack shared memory leak
|
| dc33c542351449ceee95cc50a9009f495e7c72ba |
|
12-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix AudioTrack shared memory leak" into klp-dev
|
| 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.
|
| efc4d4899e03fb8464017dd48539106cec2163de |
|
12-Sep-2013 |
Eric Laurent <elaurent@google.com> |
Merge "audioflinger: add HOTWORD audio source." into klp-dev
|
| 0c72b24f91c68442eb374bd1b338c394105b8262 |
|
11-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
Fix AudioTrack shared memory leak
Bug: 2801375
Change-Id: I50e15164fe310f69ea019dca5b49171a02bc6992
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 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/Tracks.cpp
|
| 8136cfae9c22ae8ff42eec9ed751833dda605444 |
|
09-Sep-2013 |
Eric Laurent <elaurent@google.com> |
am 8a910716: am 6ca83fad: Merge "audioflinger: no effects on offloaded tracks" into klp-dev
* commit '8a910716892d17a2ac62c7e9884af0e9d75b26bc':
audioflinger: no effects on offloaded tracks
|
| 813e2a74853bde19e37d878c596a044b3f299efc |
|
31-Aug-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: no effects on offloaded tracks
Invalidate offloaded tracks when an effect is enabled
so that the track is recreated in PCM mode and the effect
can be applied.
This is temporary until effect offloading is implemented.
Bug: 8174034.
Change-Id: I77b8b54a10db6cb8334be76d863ea7e720eaad09
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 344202e13a91c9d3d109fd8f4a67f5b23efa7800 |
|
04-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
am 55817ab9: am fe346c70: Fix miscellanous AudioTrack::getTimestamp() bugs
* commit '55817ab9c1868f1e6a59155f55c768f01509cda5':
Fix miscellanous AudioTrack::getTimestamp() bugs
|
| 1462a2b84a578b5541683c824e0f77ebf2aa82ae |
|
04-Sep-2013 |
Glenn Kasten <gkasten@google.com> |
am a4a82c5f: am 6466c9e6: Add ExtendedAudioBufferProvider::framesReleased and onTimestamp
* commit 'a4a82c5f8f3eccfa10ce30a52cbb4b1c5c983282':
Add ExtendedAudioBufferProvider::framesReleased and onTimestamp
|
| fe346c707f59d763ded93bc3d27b51f0c0408258 |
|
30-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Fix miscellanous AudioTrack::getTimestamp() bugs
Check that get_presentation_position is non-NULL before calling.
AudioTrack::getTimestamp not implemented for fast tracks.
Fix typo in Track::getTimestamp().
Fix bugs in AudioTrack::getTimestamp after stop:
- getTimestamp while stopped is not allowed.
- stop, start, getTimestamp now returns the correct value.
Change-Id: Ie8d9dc1f28d8927634e04175a68b147ffc2ea8eb
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 6466c9e6e6278c740aed77f695f679be9f5db478 |
|
23-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Add ExtendedAudioBufferProvider::framesReleased and onTimestamp
and implement them in SourceAudioBufferProvider using the associated NBAIO_Source,
and in Track using the associated AudioTrackServerProxy.
Change-Id: I60dc4adba63fc1dc452ff16caf347e4a7c8242c2
/frameworks/av/services/audioflinger/Tracks.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()
|
| 30c296c153447f5c2bb5251928bf4d3c4e28e552 |
|
30-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
am 3ae0a58a: am 9841f636: Merge "Add AudioFlinger::PlaybackThread::Track::getTimestamp()" into klp-dev
* commit '3ae0a58a83d092d96672295c8a62908478f28a9b':
Add AudioFlinger::PlaybackThread::Track::getTimestamp()
|
| ec9ad1b0947f5d6b465281312dbe92f096a8f881 |
|
30-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
am f94b2946: am 56b59224: Merge "Add IAudioTrack::getTimestamp()" into klp-dev
* commit 'f94b2946a511c5cbb6b9001449ca8278cb332bda':
Add IAudioTrack::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/Tracks.cpp
|
| 573d80a8f463f648a515fc0975bf83951b272993 |
|
26-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Add AudioFlinger::PlaybackThread::Track::getTimestamp()
with a dummy implementation initially,
and use it in AudioFlinger::TrackHandle::getTimestamp()
Change-Id: I2da88fc52a135a7f0d9fd9538986e919dc8ccd3b
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 53cec22821072719ee02c856e9ac2dda2496c570 |
|
29-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Add IAudioTrack::getTimestamp()
with dummy implementation in AudioFlinger::TrackHandle, and
implement AudioTrack::getTimestamp() using IAudioTrack.
Also document invariant that mAudioTrack and control block are always
non-0 after successful initialization.
Change-Id: I9861d1454cff7decf795d5d5898ac7999a9f3b7e
/frameworks/av/services/audioflinger/Tracks.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/Tracks.cpp
|
| 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/Tracks.cpp
|
| 6ae6b811666865815ebb1f670aacb1a0f2edaa73 |
|
06-Aug-2013 |
Glenn Kasten <gkasten@google.com> |
Remove obsolete field RecordTrack::mAudioRecordServerProxy
Change-Id: I054ba929e209797542940a0e734fc15f19f321c9
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 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/Tracks.cpp
|
| d054c32443a493513ab63529b0c8b1aca290278c |
|
12-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Move control block mName to createTrack() output
This is part of a series of CLs to clean up the shared memory
control block, by removing any fields that don't have to be there.
Change-Id: I6e51003a1293b6800258c31b22cff2eba42162e7
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 96f60d8f04432a1ed503b3e24d5736d28c63c9a2 |
|
12-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Rename control block flags to mFlags
Change-Id: I7b6d31e24531954ab1ecdf3ed56c19433700bd89
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 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/Tracks.cpp
|
| a30e75897934da2ce7b1b03bcb4b58e139d3e81e |
|
30-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Fix theoretical race using TrackBase::sampleRate()"
|
| 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/Tracks.cpp
|
| b59c1ad726b8bdf441f373d0bdc612539769334a |
|
29-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Declare methods in binder opcode order"
|
| 3dcd00dddec86a1c5133083ad7ba2265d49c048c |
|
17-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Declare methods in binder opcode order
Change-Id: I5f624b7a51ffe1a17a67c056cf984f74e4c56eac
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 35cc4f3127322ad3e3dd1e15e8ae29ff4b4a3af6 |
|
25-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Remove obsolete mBufferEnd
Change-Id: I507c6109d66000bb30933ca23b912a1316f55e5e
/frameworks/av/services/audioflinger/Tracks.cpp
|
| bd4c4fbb3b073e48963185d11a15da1fa18d2e54 |
|
25-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Remove obsolete TrackBase::step(), mStepCount, mStepServerFailed
Change-Id: I6347096f066b8b19451c6472db7b0671f0cf7702
/frameworks/av/services/audioflinger/Tracks.cpp
|
| afbbc6c5f6bc470d504dcf24ceb3ac21d57fb78d |
|
29-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Simplify RecordTrack::stop()"
|
| 420ea49539d23ab305a7b33c0854a7a475493dd4 |
|
27-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "CBLK_OVERRUN, not CBLK_UNDERRUN, is for record threads"
|
| 65d30dd9ee498f7b3bfc033735458a6ff399c207 |
|
27-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "Remove obsolete TrackBase::reset()"
|
| a8356f663014e7d4c27869629af83d8bb3441e19 |
|
25-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Simplify RecordTrack::stop()
Change-Id: Ib959c1e9dc9544d12277ce11bea445118b2e0521
/frameworks/av/services/audioflinger/Tracks.cpp
|
| fc38a2e0268b5e531db2975c3a81462a3593c861 |
|
25-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
CBLK_OVERRUN, not CBLK_UNDERRUN, is for record threads
Change-Id: I8948f76ef4717a423c37cd6ea7db4381636af612
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 0fea74cdbc09c1259e08215e2ea90e7988d62df8 |
|
25-Jul-2013 |
Glenn Kasten <gkasten@google.com> |
Remove obsolete TrackBase::reset()
Change-Id: I38100b7e28a12d7af8cb40ae3f4d9cb4a0ebe701
/frameworks/av/services/audioflinger/Tracks.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/Tracks.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/Tracks.cpp
|
| ad3af3305f024bcbbd55c894a4995e449498e1ba |
|
25-Mar-2013 |
Richard Fitzgerald <rf@opensource.wolfsonmicro.com> |
Public API changes for audio offload support.
NOTE: this does _not_ include all private member variables added
to classes as part of offload support. Only public/protected functions
and stubs functions/variables needed to make the changes buildable.
- isOffloadSupported() added to audio policy service
A stub implementation is required to build, this always returns false
- setParameters() added to IAudioTrack
A stub implementation is required to build, this always returns
INVALID_OPERATION
- CBlk flag for stream end
- Change AudioSystem::getRenderPosition() to take an audio_output_t
so caller can specify which output to query
- Add AudioSystem::isOffloadSupported()
This is fully implemented down to the AudioFlinger function
AudioPolicyServer::isOffloadSupported() which is just a stub
that always returns false.
- Add EVENT_STREAM_END to AudioTrack interface.
STREAM_END is used to signal when the hardware has actually finished
playing all the data it was sent.
- Add event type enumeration to media player interface AudioSink callbacks
so that the same callback can be used to handle multiple types of
event. For offloaded tracks we also have to handle STREAM_END and
TEAR_DOWN events
- Pass audio_offload_info_t to various functions used for opening outputs,
tracks and audio players. This passes additional information about the
compressed stream down to the HAL when using offload.
For publicly-available APIs this is an optional parameter (for some of
the internal and low-level APIs around the HAL interface it is mandatory)
- Add getParameters() and setParameters() API to AudioTrack
Currently dummy implementations.
- Change AudioPlayer contructor so that it takes a set of bitflags defining what
options are required. This replaces the original bool which only specified
whether to use deep buffering.
- Changes to StageFright class definition related to handling tearing-down of
an offloaded track when we need to switch back to software decode
- Define new StageFright utility functions used for offloaded tracks
Currently dummy implementations.
- AudioFlinger changes to use extended audio_config_t.
Fills in audio_offload_info_t member if this info is passed in when
opening an output.
- libvideoeditor changes required to add the new event type parameter
to AudioSink callback functions
- libmediaplayerservice changes required to add the new event type parameter
to AudioSink callback functions
Change-Id: I3ab41138aa1083d81fe83b886a9b1021ec7320f1
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Eric Laurent <elaurent@google.com>
/frameworks/av/services/audioflinger/Tracks.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/Tracks.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/Tracks.cpp
|
| 7c027248e1a4ccd5b22bc4deafb03e2d87ac8f38 |
|
26-Dec-2012 |
Glenn Kasten <gkasten@google.com> |
Consistent whitespace
Change-Id: I118cce68d3b777f9ec9b6bfb70367496422a40f2
/frameworks/av/services/audioflinger/Tracks.cpp
|
| 8d2d4932b96632e9eb3af4a3d4000192ef603960 |
|
25-Apr-2013 |
Eric Laurent <elaurent@google.com> |
audioflinger: fix duplicating track sampling rate
Add missing initialization of client proxy sampling rate
and volumes in OutputTrack constructor.
Bug: 8687522
Change-Id: I813e700315bb97083a63dd32279f1998ac775483
/frameworks/av/services/audioflinger/Tracks.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/Tracks.cpp
|
| bf04b5860182d8f4130dcb5d6d88ee68a58c99cd |
|
02-Mar-2013 |
Glenn Kasten <gkasten@google.com> |
Merge "media.log cleanup" into jb-mr2-dev
|
| 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/Tracks.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/Tracks.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/Tracks.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/Tracks.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/Tracks.cpp
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| e186b51e0a9834b287d7a509e960eaf1b688db75 |
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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/Tracks.cpp
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| 639482c24c911b125398b31883ba6d55faebe28b |
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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/Tracks.cpp
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| 51eb3965caa8ba135bcdd8ffb7a2024a042ecdc0 |
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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/Tracks.cpp
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| 0ddd56316262ac74a95e9edb595697c163136d6d |
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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/Tracks.cpp
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| ecd9389c8712aedeb2a79823ea0e4fb842684269 |
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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/Tracks.cpp
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| 3051df27261e9952c0e642dec548515250e85f6a |
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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/Tracks.cpp
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| 9e58b552f51b00b3b674102876bd6c77ef3da806 |
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19-Jan-2013 |
Glenn Kasten <gkasten@google.com> |
AudioFlinger uses media.log service for logging
Change-Id: Ia0f8204334f6b233f644d897762a18c95d936b4b
/frameworks/av/services/audioflinger/Tracks.cpp
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| e3aa659e9cee7df5c12a80d285cc29ab3b2cbb39 |
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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/Tracks.cpp
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| 5736c35b841de56ce394b4879389f669b61425e6 |
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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/Tracks.cpp
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| e4756fe3a387615acb63c6a05788c8db9b5786cb |
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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/Tracks.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/Tracks.cpp
|