Make the file name consistent as the class name.
Test: make
Bug: 135621476
Change-Id: Icbde26fe1798e344c3746f4678db098688456654
Merged-In: Icbde26fe1798e344c3746f4678db098688456654
This patch aims to make the structure for libaudiofoundation and policy
stuff simpler, including:
1. Make AudioPortBase as AudioPort in libaudiofoundation. Create
PolicyAudioPort that contains policy related stuff, which is used by
audiopolicy.
2. Make AudioPortConfigBase as AudioPortConfig in libaudiofoundation.
Create PolicyAudioPortConfig that contains policy related stuff.
3. DeviceDescriptor derives from AudioPort, AudioPortConfig, PolicyAudioPort
and PolicyAudioPortConfig. IOProfile derives from AudioPort,
PolicyAudioPort. AudioInputDescriptor/AudioOutputDescriptor derives from
AudioPortConfig, PolicyAudioPortConfig.
Test: atest AudioTrackTest, AudioRecordTest, AudioManagerTest
Test: atest audiopolicy_tests, AudioHostTest
Test: audio smoke test
Bug: 135621476
Change-Id: I40299d95dda3b3fc0ea88b079f2fe38d8f7e5b31
Merged-In: I40299d95dda3b3fc0ea88b079f2fe38d8f7e5b31
The file only contains helper functions for AudioProfileVector.
Test: make
Bug: 135621476
Change-Id: Ifd8eb281cbb7d6395fd5d4b4d6f308d58e6ded3d
Merged-In: Ifd8eb281cbb7d6395fd5d4b4d6f308d58e6ded3d
Make AudioProfileVector only in libaudiofoundation. In policy part,
instead of having policy specific AudioProfileVector deriving from
AudioProfileVector in libaudiofoundation, make all the functions as
global functions. In that case, AudioPortBase does not need to be a
template class. That helps make the structure clearer.
Test: audio smoke test
Test: CTS for AudioTrack, AudioRecord, AudioManager
Test: audiopolicy_tests, AudioHostTest
Bug: 135621476
Change-Id: I36b12123cf52c3f82cef09a965403791dff74093
Merged-In: I36b12123cf52c3f82cef09a965403791dff74093
The test uses HDMI and bt hfp to test setting of device connection
state. After the device is connected, the test tries to use explicit
routing to the new connected device to make sure the connection state is
set correctly.
Bug: 135621476
Test: atest audiopolicy_tests
Change-Id: I14759daac6d3522fa133a6e3b27b3863bb4ce3d1
Merged-In: I14759daac6d3522fa133a6e3b27b3863bb4ce3d1
* changes:
Add AudioPortFoundation, AudioPortBase and AudioPortConfigBase.
Copy AudioPort.h to libaudiofoundation.
Make AudioProfile as a common class and create AudioProfileVectorBase.
Preparation for moving AudioProfile to libaudiofoundation.
Return DeviceDescriptor or DeviceVector in Engine.
Add unit test for dyanmic policy. Including:
1. Use bp file instead of mk file
2. Test (un)registering policy mixes
3. Test dynamic policy playback rerouting
4. Test dynamic policy mix record injection
The idea of 3. and 4. is from pa/1201282.
Test: atest audiopolicy_tests
Bug: 135621476
Change-Id: Ic95c6f9193c2448168897c5441554cff735f401d
Merged-In: Ic95c6f9193c2448168897c5441554cff735f401d
The purpose of this change is to make a generic AudioPort class that
could be used by vendor.
1. Add AudioPortFoundation and AudioPortBase. AudioPortFoundation
contains generic members of an audio port. AudioPortBase is a template
class that derives from AudioPortFoundation. It takes audio profile
vector as a template parameters. That makes it more feasible to use
customized audio profile stuff as needed. In audio policy, AudioPort
derives from AudioPortBase and contain policy related stuff.
2. Add AudioPortConfigBase. AudioPortConfigBase contains generic members
of an audio port config. AudioPortConfig derives from AudioPortConfigBase
and contains policy related stuff.
3. Use std::string instead of String8 in AudioPort and AudioPortConfig.
Bug: 135621476
Test: CTS for AudioRecord, AudioTrack, AudioManager, RoutingTest
Test: audiopolicy_tests, AudioServiceHostTest
audio smoke test
Test: dumpsys media.audio_policy
Change-Id: I05611b7067c18006660de2c678acd56bdd34010a
Merged-In: I05611b7067c18006660de2c678acd56bdd34010a
AudioProfile is a class that contains information for an audio profile.
AudioProfileVectorBase is a base class that contains operations for a
vector of audio profiles. AudioProfileVector derives from
AudioProfileVectorBase and contains policy related stuff.
Bug: 135621476
Test: CTS for AudioRecord, AudioTrack, AudioManager
Test: audio smoke test, audiopolicy_tests
Test: dumpsys media.audio_policy
Change-Id: Ic2e08efcc5efa99e499a931811b7042fbd5ddf04
Merged-In: Ic2e08efcc5efa99e499a931811b7042fbd5ddf04
Copy AudioProfile from audio policy to libaudiofoundation. Note that
AudioProfile in libaudiofoundation has not been added into the make
file as it needs some more change to make it work. Doing this just to
make a cleaner diff for later change.
Rename AudioProfile in managerdefinitions folder as AudioProfileVector
since AudioProfile will be moved to libaudiofoundation.
Bug: 135621476
Test: make
Change-Id: I13ac067317648bbfd8d3960d16a1ea56ded83011
Merged-In: I13ac067317648bbfd8d3960d16a1ea56ded83011
To remove the limit on the number of audio device types, there is a
need to remove the return value as a combination of audio_devices_t.
Using DeviceDescriptor or DeviceVector in Engine is a starting point.
Test: audiopolicy_tests, smoke test
Test: CTS for AudioTrack, AudioRecord, AudioManager
Bug: 135621476
Change-Id: I0d87fd76385629e6e6b74cdd37dbd334ae02c04f
Merged-In: I0d87fd76385629e6e6b74cdd37dbd334ae02c04f
1. Add SampleRateSet, FormatSet, ChannelMaskSet, FormatVector
using std::vector and std::set in libaudiofoundation.
2. Use SampleRateSet, FormatSet, ChannelMaskSet, FormatVector in
audio framework.
Bug: 135621476
Test: make, CTS for AudioRecord, AudioTrack, AudioManagerTest
Test: play and record audio
Change-Id: Ic04e637bcc6ba9df84c5e7561ec3c03f18a7d242
Merged-In: Ic04e637bcc6ba9df84c5e7561ec3c03f18a7d242
1. Create folder for libaudiofoudation.
2. Move AudioGain to libaudiofoudation. The goal is to make AudioGain
be able to use in binder call and be vendor-available later.
Bug: 135621476
Test: make, CTS for AudioTrack, AudioRecord, AudioManagerTest
Change-Id: Ibf16ad9cb7f2ac657a02632ccbe4f934b5a12731
Merged-In: Ibf16ad9cb7f2ac657a02632ccbe4f934b5a12731
Sometimes after switching songs in Files, audio driver
enters "PAUSED" state, and no sound anymore.
In normal case, "paused" track will be removed from active tracks,
then audio hal will make this stream enter standby mode.
Actually there are two paths to move paused tracks into standby mode:
1> track underruns twice:
if (--(track->mRetryCount) <= 0) { // mRetryCount is 2
tracksToRemove->add(track);
} else {
doHwPause = true; // pause
}
Unfortunately, this issue happened exactly in the middle state:
track underrun for the first time, it enters to paused state;
then before it underrun for second time, it is stopped due to MediaPlayer.
2> track is paused and code goes here:
if (mStandby || !last ||
track->presentationComplete(framesWritten, audioHALFrames) ||
track->isPaused() ||
+ track->isStopped()) {
tracksToRemove->add(track);
}
But when the issue happens, track is in stopped state,
there is no logic to standby it anymore, so add one.
Change-Id: I70cac1f3d6b065690400f6a2363712d4642a9763
Otherwise we could return the uninitialized value when result != OK.
Bug: 143955791
Test: manual - Run the test in the bug and it doesn't crash.
Change-Id: I23b2b0a20202152648a3e40650965bba45127ff3
The internal buffer queue and respective 'mConsumer' may
never get initialized in case the connection and/or
configuration of client shared output surfaces fails.
To avoid possible instabilities check whether the
consumer interface is valid before trying to disconnect.
Bug: 143506890
Test: atest
cts/tests/camera/src/android/hardware/camera2/cts/MultiViewTest.java
Change-Id: Ia533233444fd548ddb52f4fde06212a21bc843bc
This fixes > a dozen potential leaks flagged by clang's static
analyzer, and fixes the use of other nearby manual memory management.
Bug: None
Test: TreeHugger
Change-Id: I8af2523804986b278150291fb2dc7ca5af8391ee
C++20 will require members in a designated initializer to be in order
unlike C99.
Bug: 139945549
Test: mm
Change-Id: I610f4e8a8a9d69eb650c354df4bd1438a1408897
Merged-In: Ie129344a36893adc82e67181012e3af817bda9f4
Signed-off-by: Nick Desaulniers <ndesaulniers@google.com>
* changes:
cameraserver: Avoiding deadlocks while calling isPublicallyHiddenSecureCamera().
Do not include hidden secure cameras in camera1: getNumberOfCameras
The following scenario can occur:
T1 serving Client A's disconnect() call:
T2 : serving Client B's connect() call
T2 : CameraProviderManager::openSession() locks mInterfaceMutex and waits on open() HAL
interface call
T1: updateStatus() locks mStatusListenerMutex
T1: CameraProviderManager::isPublicallyHiddenSecureCamera() waits
on mInterfaceMutex
T2: while waiting on open(), gets a torchModeStatus() callback from the camera HAL
T2: onStatusChanged()
T2: broadcastTorchModeStatus() which waits on mStatusListenerMutex
As a result there's a possible circular hold and wait between T1 and T2.
We cache isPublicallyHiddenSecureCamera in CameraState. That doesn't completely
avoid having to hold mInterfaceLock while calling isPublicallyHiddenSecureCamera() in CameraService
(cache updates), instead it reduces it. We instead need to hold mCameraStates lock which has a
smaller scope.
Bug: 141756275
Test: CTS
Test: GCA (sanity)
Merged-In: I4a697c1eaccc3603007be4a595febea981fbeb64
Change-Id: Ie5508afb126a874f76fbbfc2dd19ef79ae6255e0
Signed-off-by: Jayant Chowdhary <jchowdhary@google.com>
Apps cannot connect to hidden secure cameras. Do not include them in the
number of cameras reported by camera1 api.
Bug: 141247926
Test: Without CL -> mark all cameras as hidden secure cameras;
atest FastBasicsTest.java#testCamera1 fails
With CL -> mark all cameras as hidden secure cameras;
atest FastBasicsTest.java#testCamera1 passes
Test: camera CTS
Merged-In: I9d1721fd5e94fa7f692c3da52aa667ae9247d368
Change-Id: I229a336bed6b2695e16c1457cb8f74c26b07f7e8
Signed-off-by: Jayant Chowdhary <jchowdhary@google.com>
(cherry picked from commit cf93540965)
This is preparation for having a subclass of DataSourceFactory which
is only used in mediaserver process with OMA (forward-lock) use case.
Test: build
Bug: 142567168
Change-Id: I2a1ab3d1ae89f657a84376d9a95d4e814b545b4f
Fix issue where RecordTrack silencing didn't silence the
full buffer: the memset to 0 was using the RecordThread frame
size, not the RecordTrack frame size.
OP_RECORD_AUDIO was only enforced at the start of a recording
which would fail if not granted. This patch silences the recording
(i.e. silence is recorded instead) when it is lost, and undoes that
when granted again. This requires:
- propagating the package name of the client to the RecordTrack class
- registering an appOp callback in RecordTrack (through a new
OpRecordAudioMonitor class) to (un)silence the recording
- update the isSilenced() method to take into account the appOp.
Bug: 138968594
Bug: 138636979
Test: run app that records audio, then "adb shell appops __pack_name__ 27 2"
and verify recording is silent after that.
Change-Id: Ib33f5b592185a67204997213bab1ac2594d90d37
(cherry picked from commit ddf87effa4)
Merged-In: Ib33f5b592185a67204997213bab1ac2594d90d37
(cherry picked from commit 7307293e8b)
There were three issues with the implementation
of Audio Playback Capture patches on secondary outputs.
1) Patch tracks were always forced to be ready, even if there were not
enough audio to be played. That led to playing partial buffers if the
source output had smaller (time wise) buffer than the secondary output.
This is fixed by avoiding setting CBLK_FORCEREADY on every buffer push
from the primary output thread.
2) After 1 is fixed, the patch tracks now behave like regular (non fast)
tracks. This means that the track only starts when it is full.
That leads to overrun on startup as the primary and secondary outputs
usually do not have the same write period.
What makes the issue worst is that Remote submix has a fixed buffer in bytes,
so its write period changes depending on sample rate contrary to the
primary HAL which pulls with fixed time periods (usually 20/40ms).
This patch solution is to introduce a ready threshold. The track will be
considered ready to be active if there are more than frames in its buffer.
To avoid changing previous latency behaviour except for APC,
legacy tracks have this threshold set to their buffer size and
non APC patch tracks have it set to 1, similar to setting
CBLK_FORCEREADY.
3) The patch track buffer size calculation of the patch track did not take
into account primary and secondary output could be of different sampling
rate.
This mean that the patch track buffer was usually too small as the
secondary output usually had smaller sampling rate (Live caption
requests 16kHz) than the track (music usually plays in 48kHz).
This made the two problems even worst.
This was easily fixed by scaling the buffer size with each side sample rate.
Bug: 136691300
Test: play audio
adb shell audiorecorder --target /data/file1.raw
# Check that the recorded file has no underrun
Change-Id: Ib2846b2827afd1b953d4a25acb18c7cadf57cd3e
Signed-off-by: Kevin Rocard <krocard@google.com>
As python 2 is beeing deprecated, this CL updates all the tool
scripts to python3.
It also fix some issues revealed by pylint.
Test: lunch audio_configurable target & build
Change-Id: I69a201190c688be3825cbdaa238046367a5d09c7
Signed-off-by: Francois Gaffie <francois.gaffie@renault.com>
If no default device is available from configuration.xml, and if
the engine fallbacks on it anyway, it tries to get its type without checking
the pointer validity.
This CL fix the potential segfault and assert with explicit error message.
Test: build
Change-Id: Icf8d7f5ef998dad6f4033d934b48d408030c7e17
Signed-off-by: François Gaffie <francois.gaffie@renault.com>
To avoid bypassing PATH restrictions, this CL fixes the python
binary dependancy by retrieving the path from the host binaries.
Test: build
Change-Id: Ib3356ee1a69d1a5e5a9f9d22b46e3e70c28fa991
Signed-off-by: François Gaffie <francois.gaffie@renault.com>
into libdatasource, which contains:
DataSourceFactory
(Clear)FileSource
(Clear)MediaHTTP
DataURISource
HTTPBase
NuCachedSource2
This is needed to break a circular dependency in an upcoming CL.
Test: build, boot
Change-Id: I34d9937235c78f18f51b18945342a0743e209577
Merged-In: I34d9937235c78f18f51b18945342a0743e209577
Do not consider Accessibility services when determining
lastest active capture client. As an Accessibility service
capturing does not silence other captures, it should not
prevent another client from being selected as the latest active.
Bug: 135806801
Test: Start capture with Recorder app, activate Voice Access
Change-Id: I864be785752f27201d45ce24f95013de6391dbd6
Merged-In: I864be785752f27201d45ce24f95013de6391dbd6
Call initialize() after constructing AudioPolicyManager
from factory to avoid calling virtual methods from constructor.
Bug: 136491567
Test: make
Change-Id: Ie0b591d90e643b1fd1c326f9731dc656b38230dc
Merged-In: Ie0b591d90e643b1fd1c326f9731dc656b38230dc
Add missing calls to ATRACE for direct writes and frames ready
on direct thread.
Test: <start direct output>
external/chromium-trace/systrace.py audio
Merged-In: Iee6ea72442f9902819628735e19cce0ba1546165
Change-Id: Iee6ea72442f9902819628735e19cce0ba1546165
Do not consider Assistant capturing with HOTWORD source
when determining lastest active capture client. As assistant
capturing for HOTWORD does not silence other captures, it should not
prevent another client from being selected as the latest active.
Bug: 135806801
Test: Start capture with Recorder app, place in background and trigger
false OK G detection
Change-Id: Ic17bfd70675ef749d6a678d067112b1dbd205746
Merged-In: Ic17bfd70675ef749d6a678d067112b1dbd205746
* changes:
Use PassthruPatchRecord for DIRECT to DIRECT connections
Add PassthruPatchRecord for low latency software patches
Abstract access to HAL stream via Source in RecordThread
Move PlaybackThread::Track::writeFrames to PatchRecord
audioflinger: Add tracing of buffer frames to PatchTrack/Record
libaudioprocessing: Extract vendor-available part of AudioMixer
AudioMixer: Cleanups
libaudioprocessing: Trivial dependency cleanups
When both input and output connected by a software patch
are in 'DIRECT' mode (no framework processing), use
PassthruPatchRecord as it helps to reduce the latency
significantly by avoiding intermediate buffering.
Remove 'std::nothrow' from tracks creation in PatchPanel
for consistency with other code.
Bug: 117564323
Test: with MSD module
Merged-In: I52ec5b02a207548ebc4073c1033e396f444c041c
Change-Id: I52ec5b02a207548ebc4073c1033e396f444c041c
Implement a subclass of PatchRecord that uses PatchTrack's
thread for reading from HAL. This eliminates the need for
buffering that adds latency.
The only modification needed for PatchTrack is to indicate
unlimited amount of available frames to the playback thread.
This is to prevent PatchTrack from being deactivated by
DirectOutputThread due to lack of frames available.
RecordThread believes it reads audio data on its thread,
and manages timestamps as usual. The data that it "reads"
and passes to PassthruPatchRecord is discarded by the latter.
Bug: 117564323
Test: with MSD module
Merged-In: I376656e3c791e91e2196331ecdf2b425697c4e18
Change-Id: I376656e3c791e91e2196331ecdf2b425697c4e18
This allows to replace direct reading from HAL with obtaining
audio data from another source.
It should be possible to encapsulate reading from FastCapture
in the same manner, but it's not required for direct inputs.
Test: make
Merged-In: I3f005583410cc9c5d4b07c127d95e236abb4a85f
Change-Id: I3f005583410cc9c5d4b07c127d95e236abb4a85f
This code logically belongs to PatchRecord because it emulates
obtaining recorded data from a client.
Test: make
Merged-In: Icba4e33d9d0ca57e6ad964aaf3209e7db766f284
Change-Id: Icba4e33d9d0ca57e6ad964aaf3209e7db766f284
Seeing requested / obtained buffer frame counts in trace
helps to identify discontinuities in software patch pipeline.
There is no need to trace requested frame count for PatchRecord
as it always requests as much as possible ("-1" frameCount).
Bug: 117564323
Test: with software patch active,
external/chromium-trace/systrace.py audio
Merged-In: I78400f6cc1d1940b3290fb23bb1aacfbffe042e5
Change-Id: I78400f6cc1d1940b3290fb23bb1aacfbffe042e5
Split AudioMixer into the base part which doesn't rely on
the framework and can be used in vendor code and the derived part
which is intended to be used by Audioflinger.
Test: A/B compare output from test scripts, manual testing on the phone
Merged-In: I24c390f67f20baa8109902099359ca6e34eebcfd
Change-Id: I24c390f67f20baa8109902099359ca6e34eebcfd
- Remove unused dependencies on libnbaio and libnblog;
- Move dependencies on libaudiohal and libsonic to libaudioprocessing
as AudioMixer implementation does not use them.
Test: make
Merged-In: I258a450725bdacb4fcf437b6f86582d51d40e622
Change-Id: I258a450725bdacb4fcf437b6f86582d51d40e622
During the transition to Treble, "device connect" / "disconnect"
parameter was erroneously attributed to audio streams. This
issue got fixed in Audio HAL V4.0, but AudioParameter file
was still using incorrect name for the constant.
Test: make
Change-Id: I4056ad9e98af6dadd017d22bc4bde031cac75631
Merged-In: I4056ad9e98af6dadd017d22bc4bde031cac75631
jiabin
Treehugger Robot
Xin Li
Emilian Peev
Jindong
Nick Desaulniers
Tao Wu
Jayant Chowdhary
Marco Nelissen
George Burgess IV
Dongwon Kang
Jean-Michel Trivi
Kevin Rocard
Eric Laurent
Francois Gaffie
Mikhail Naganov
Yi Kong
Aniket Kumar Lata
Dean Wheatley