| Commit message (Collapse) | Author | Age | Files | Lines |
... | |
|/ /
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
Add a method to query from the audio HAL the HW sync
source used for a given audio session.
Modify audio policy to select a direct output with HW sync
when requested.
Bug: 16132368.
Change-Id: I03038f9188f2d389f8a5fd76a671854013a4513e
|
|\ \
| |/ |
|
| |
| |
| |
| | |
Change-Id: Ibc07bff7710398929c135f38324dd29857fa0ea6
|
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
Audio policy:
- Added active capture indication to sound trigger service:
recognition stops if concurrent capture is not supported.
- Added generation of reserved I/O handle and session ID for
utterance capture.
Sound trigger service
- Added sound model update callback handling.
- Added service state callback
- Simplified callback shared memory allocation.
Bug: 12378680.
Change-Id: Ib0292c2733e6df90fdae480633dd9953d0016ef1
|
|\ \
| | |
| | |
| | | |
into lmp-dev
|
| |/
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
Currently CameraSource/AudioSource's stop() and read() are both called
from the puller's looper. This works if source operates normally (i.e.
read() returns regularly before source is stopped), as the stop() will
eventually be handled by the looper. However, if for some reason the
source hang, it will get stuck in read(), and the stop() will never
be processed, which could lead to ANR (in addition to the source hang).
We need to move the source's stop out of the puller's looper. It also
can't be on MediaCodecSource's looper, because the source's stop
synchrounously waits for all outstanding buffers to return, these
are only returned when MediaCodecSource's looper processes the buffer.
This change moves the stop to MediaCodecSource::stop, after encoder
is shutdown.
Bug: 16522726
Change-Id: Ie91f563c5d8a98ab091bf1945af4e51f662b9403
|
|/
|
|
|
| |
bug: 12247651
Change-Id: I564ac8de3da2430342a028f4058e2c5ac2d85d5e
|
|\ |
|
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
In async mode:
- codec must be restarted after flush
- dequeueIn/OutputBuffers fail
- getIn/OutputBuffers fail
Bug: 11990118
Change-Id: If2d6a76ab499ee9ed4a11486fb537acbc52e66f6
|
|/
|
|
|
| |
Bug: 10706245
Change-Id: I9a77631bfae0358be229b079228c1fcae0e77faf
|
|
|
|
|
| |
Bug: 16653284
Change-Id: I54165041da5a13498d627eee1b3ec59ef3c923b0
|
|\ |
|
| |
| |
| |
| | |
Bug: 15328708
Change-Id: I9dfca30745c3e4dda91c3894363462f8631c41a1
|
| |
| |
| |
| |
| |
| |
| |
| | |
Indicate the audio session ID when calling getInput(),
startInput(), stopInput(), releaseInput().
Bug: 12378680.
Change-Id: I763793752f93e2f4e1445a5ab217c895af011038
|
| |
| |
| |
| |
| |
| |
| |
| |
| | |
Rename AudioSystem::newAudioSessionId() to
AudioSystem::newAudioUniqueId() as it can be used
also for I/O handles.
Bug: 12378680.
Change-Id: I611ea3b5eb57a4b0774437f477ee87dc4ccc2cc2
|
|/
|
|
|
|
|
|
|
|
| |
Add parameters to openInput() and openOutput(): device address,
input source.
Allow caller to specify a given I/O handle
Group parameters in a struct audio_config.
Bug: 12378680.
Change-Id: I7e9af74c0d996561cc13cbee7d9012d2daf33025
|
|
|
|
|
| |
Bug: 10706245
Change-Id: Icd246f22edfc67ed5240d59f5a5bde3e5f749465
|
|
|
|
|
| |
Bug: 10706245
Change-Id: I8c4e96a2581a039e9e8237c3e09e2c22226da055
|
|\ |
|
| |
| |
| |
| | |
Change-Id: I8f783466f8c2560820db14488acc1a309d27ab0f
|
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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
|
|\ \ |
|
| | |
| | |
| | |
| | |
| | | |
Bug: 16329805
Change-Id: Idcd603545352e36a88589d3e23ccf1ee37704695
|
|\ \ \
| |/ / |
|
| | |
| | |
| | |
| | |
| | | |
Bug: 16329805
Change-Id: I8a0ecd100fca397add97a1416125bcc6aeb86364
|
|\ \ \
| |/ / |
|
| | |
| | |
| | |
| | |
| | | |
Bug: 16329805
Change-Id: Ib971dd95b54829438c8af97528f9e00b87ab3f1e
|
| |/
|/|
| |
| |
| |
| | |
Bug: 12979595
Change-Id: Iafd93046a4fd9f22bcd66084deace746a7ca5d3c
|
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
Add support for the audio_attributes_t structure in the
SoundPoool constructor.
Remove SRC quality which was never implemented.
Remove stream types.
Add file to contain audio helper functions related to policy.
Change-Id: I1720ff15e7b23ea7b713a4395fdfac26dc3fd4da
|
| |
| |
| |
| |
| | |
Bug: 12065651
Change-Id: Icfb73c0009621cd747e113d8a0cd84c966bf055d
|
| |
| |
| |
| |
| | |
Bug: 15699665
Change-Id: I31c1ab4413c62ff3dd4e0d5b06a398064b4aaddd
|
| |
| |
| |
| |
| | |
Bug: 15699665
Change-Id: I2aaddc4c937cf5c1e36386bafd7d396d5781bf6d
|
|\ \ |
|
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | | |
Buffer frame count and notification frame count
are now calculated by server instead of by client.
The server has more information and can do a better job.
Also fix a few bugs:
- If a fast track was re-created, even with same pipe depth, it would fail.
Now it can correctly re-create a fast track provided the pipe depth is same.
- Notification frame count for fast tracks was calculated by client
as 1/2 of the total frame count, which is a large value due to the pipe.
Now the notification frame count is set by server to the HAL frame count.
This should reduce latency for fast tracks.
- EVENT_OVERRUN were happening frequently when there was sample rate conversion,
because the client didn't know about the sample rate conversion,
and under-estimated the necessary buffer size. Now since server
calculates the buffer sizes, EVENT_OVERRUN is unlikely.
- RecordThread::createRecordTrack_l was checking for mono and stereo
for fast tracks. This is not necessary, and now we can handle a
multi-channel fast track.
Bug: 7498763
Change-Id: I0c581618e8db33084d5ff9ed50a592990c9749e8
|
|\ \ \
| |/ / |
|
| | |
| | |
| | |
| | | |
Change-Id: I9f37be05f8dc7b85a8827a94e76ca0f45453e170
|
|\ \ \
| |/ /
| | /
| |/
|/| |
|
| |
| |
| |
| |
| |
| |
| | |
For backward compatibility, until flags are correctly calculated,
we will assume that the request is for a low latency input stream.
Change-Id: I76746834e870df00833dc77cbdaa2edd2ffeec95
|
|\ \
| |/
|/| |
|
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
- Enable the normal partial result path for HAL3.2, the quirk is only used
for the HAL version lower than HAL3.2. The partial quirks is no longer supported
for HAL3.2 or higher versions.
- Add CameraDeviceBase getDeviceVersion API.
- Fix some build warnings
Change-Id: I7a1b03d4d5fd5258d2addfba4368bee2ba691337
|
|/
|
|
|
| |
Bug: 15153976
Change-Id: I0204c4188d485cda026497469c7cde24f7bd5c95
|
|
|
|
|
| |
Bug: 12034929
Change-Id: I326f1356df89474aa088c1c87f8505b33654139d
|
|
|
|
|
| |
Bug: 15116722
Change-Id: I3fcc9aea38afcbd665f86c511a9929fe9a6a3a8f
|
|\ |
|
| |
| |
| |
| |
| | |
Bug: 11990118
Change-Id: I3278aecb20df88c42fa2709a66e6166eb3cbe56f
|
|\ \
| |/ |
|
| |
| |
| |
| |
| |
| | |
Bug: 11990118
Change-Id: I6fe4b407d9c85cddec8d958620d5d356735273cf
|
|/
|
|
|
|
|
|
| |
Pass audio aac sub formats in offloadinfo according to
aac profile. Audio HAL can take decision about offload
using DSP capabilities
Change-Id: If269a3654b5d2b09c183212b0646ef03e06f2d8f
|
|
|
|
|
|
|
|
|
|
| |
These are designed to be called from the same thread as the one
calling dequeue?Buffer, and use a mutex to avoid switching
context. All other calls of MediaCodec are designed to be blocking
and synchronous.
Bug: 14297827
Change-Id: If341c6e4407ca6f10f5e0d47008dddc0e20b0a50
|
|\ |
|