| Commit message (Collapse) | Author | Age | Files | Lines |
|
|
|
|
|
|
| |
Warn allocation failures explicitly rather than crash
trying to access unallocated memory
Change-Id: Ie86c3ac130917e1f4030eb8207ac8350cba7711d
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
Add support for QTI audio resampler
audio: Audio resampler support for 192Khz playback
Change-Id: Ia8f24a0874ebf6e16ef7bd1f2759a14f47149875
audio: Add audio mixing support for qti resampler
Change-Id: Ib657aa12b2a72323564148c302ff8891e1bb7433
AudioMixer: Extend use of QTI resampler for 44.1Khz sampling rate
Change-Id: I2a819dbc9f1e3e280cb4fa79328e331883a3e981
AudioMixer: fill 0s at right place when no more buffers available
Change-Id: I50504c5a02eb0c69abfc9b047792b0f6f85b9ce8
audioflinger: add channel count check to use QTI resampler
Change-Id: I8f76dd82b72a0dd8b77343e77e0d0545e1be2114
Change-Id: Ia8f24a0874ebf6e16ef7bd1f2759a14f47149875
|
|
|
|
|
|
| |
- add support for effects on direct pcm output
Change-Id: I2fbac63c623bf51a03e5e91828369739d33329f3
|
|
|
|
|
|
|
|
|
|
|
| |
When an input stream is active on USB headset and if a voice call
is received, the proxy_open() for voice call fails and the screen freezes.
All active inputs must be closed before opening input stream for
voice call and also all new requests to open input stream must be
blocked.
Change-Id: I3fb0d482a77495ff6fe9fcdc8a1f8915ade52c9a
CRs-fixed: 876993
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
This is a squashed commit of the following fixes in
AudioPolicyManager. Only some parts of the fixes have been
ported as the custom audio policy has the other required
changes.
audiopolicy: Fix MT call delay
Change-Id: I790479eaad9d2d6fd4204cc4cb7698761c30f7cf
audiopolicy: support extended feature in audiopolicymanager
Change-Id: I1925339b591cd29f11a71c287a2e53c0627e9e62
audiopolicy: additional change for extended feature
Change-Id: I9bad6a294ddd7aee72f6f6a314666b892b730c8e
Change-Id: I7738d4b0ac11ee6d93bfd67e2553eae8518ff719
|
|
|
|
|
|
| |
allow effects in case outout is direct pcm
Change-Id: I2ad7eacf11642a4ca9f892b61124293d0dc503a9
|
|
|
|
|
|
|
|
|
|
|
| |
It's not needed to update the channel mask based on source because
the source is sent to audio HAL through set_parameters() and if source
equal to VOICE_CALL does not mean that two channels need to capture.
If recorder app selects AMR as encoding format but source as RX+TX means
both RX and TX are captured in ONE channel. So use channels set by the app
and use source for the type of capture (RX only, TX only, or RX+TX).
Change-Id: Iedf23318356480ba939fc06df4ae4f12906322b3
|
|
|
|
|
|
|
|
|
|
|
|
| |
- Playback of 24 bit 192kHz clips fails during device switch
between wired headset and BT when repeat track is enabled.
- Memory allocation for AudioTrack fails because the heap gets
fragmented and free chunks of the size requested are not available.
- Increase the current heap size to 2MB to ensure that there is
always a free chunk to accommodate the requested size.
CRs-Fixed: 855910
Change-Id: I2eb18b15557fa264fb66ff282746cad4e6c718f7
|
|
|
|
|
|
|
| |
-dumpsys logs show unknown format for PCM offload playback.
-Add PCM offloading formats for logging.
Change-Id: I4dbb8721c7e1d1f9d51bb1f964648046e7c09875
|
|
|
|
|
|
| |
- Add support for 5.1 channel recording
Change-Id: If060fffb2e198f516f40e85390489de2108be5d1
|
|
|
|
|
|
|
|
|
|
|
| |
For 6.1 channel ALAC clips, the compress offload profile is
not found even though the channel mask is defined in the
audio_policy.conf file. This is because the channel mask enum
is not defined and hence the profile's channel masks
are not getting enumerated properly.
This change is needed for 6.1 channel ALAC clips playback.
Change-Id: I6b820776c0dc6e68a402886f0931439edab24a8b
|
|
|
|
|
|
| |
add support for APE decoding
Change-Id: I55e8f4b3b87f4bdf1c99774d702506eb7c2f05b5
|
|
|
|
|
|
|
| |
add support for decoding/offloading ALAC
audio formats
Change-Id: Id66f0cb6c140113741962e119148bf434de3d064
|
|
|
|
|
|
|
|
| |
Effect command doesn't necessarily fill cmd return code into &cmdStatus,
so we should initialize cmdStatus to avoid uninitialized value propagates
to upper stack caller.
Change-Id: I5694616c9d3d66071241958f54a03b8d3b9d881e
|
|
|
|
|
|
|
| |
-Offload track invalidation is needed during SSR
to switch from Offload to deep buffer playback.
Change-Id: I728cfcadc8cd734914b94000a711d1e86bcfad9d
|
|
|
|
|
|
| |
Added WMA as a valid audio format in audio policy
Change-Id: Iace14a011ebb89b9deeebd7fe04d0f9b1ab27c9e
|
|
|
|
|
|
| |
-add 16 and 24 bit PCM offload formats
Change-Id: I23de9b6663be15971c62cba75e2476a503ef4e52
|
|
|
|
|
|
|
|
|
|
|
| |
make function virtual or protected so that
they can be extended in custom audio policy
also add flag in Android.mk so that proxy
device get added in device list when
proxy is enabled
Change-Id: Ida7992f6b327491fab1f4ea376e85e8eb34b89ca
|
|
|
|
|
|
|
|
|
|
|
| |
This increase is needed to accommodate higher sampling rate clip
playback over devices like BT and to support gapless playback with
larger buffer sizes. In both cases, the cblk memory allocated for a track
can be high enough that a new allocation (either due to restoreTrack_l or
opening a new track) can fail.
Change-Id: I96f674706184f029259802d5552f5ceeebc689c1
CRs-Fixed: 768106
|
|
|
|
|
|
|
|
|
|
|
| |
-Clear effect buffers in threadloop_mix() in case audio
effects enabled when output threads are not ready
-Also clear mix buffers in threadLoop_sleepTime()when tracks
are not ready
CRs-Fixed: 765749
Change-Id: I475d42ac0cc68e4856002a9bd4c6c256a6fca70c
|
|
|
|
|
|
|
| |
Add support for FLAC playback in
compressed offload mode
Change-Id: I617b41b867277272212d6cf1a6f82f646c5b1032
|
|
|
|
|
|
| |
Add audio policy for WFD and visualiser usecase.
Change-Id: Idf3856a373eb7a05362f19d6cb117e9d4fb757ef
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
This change includes the following gerrits:
# This is a combination of 4 commits.
# The first commit's message is:
Camera: Enable Histogram feature.
Link the histogram enable/disable commands from
application to the HAL layer.
Change-Id: I510c4e1798285ed1315bfb0d234fa76090659ba2
# This is the 2nd commit message:
Camera: Add support for ZSL burst mode.
Added ability to set number of snapshots in burst mode.
Change-Id: Ie0e7c8c0117b7adc985cfc92df79747ee6a5ea51
# This is the 3rd commit message:
CameraService: Adds support for longshot mode
- This change introduces additional functionality inside
CameraClient for supporting continuous compressed data
callbacks. This is needed for 'Burst/Long shot' mode
where we could have indefinite number of callbacks after
capture is triggered.
(cherrypicked from commit e4f502aa7cbe8875e8a1589024cdcf227c228a2b)
Change-Id: Ia18ca9bdda7736c679db557e510870115089537a
# This is the 4th commit message:
CameraClient: Enables meta data notifications.
Adds the needed functionality for enabling/disabling
metadata messages depending on the camera client
commands.
Change-Id: I39d632b4742e83df5db5f86b12742aefc2480dfc
Cherrypicked from 25bd97f5ec30e7942c3b1fdc96115da6028736f0
Change-Id: Ie930d20c962593e40a0767f9cf7d4385df8e2561
|
|
|
|
|
|
|
|
| |
Use a dedicated mutex for torch UID map so it won't cause a deadlock
after flashlight app gets killed while the torch is on.
Bug: 23722318
Change-Id: I228377aa0412052d56b6b948361d9abaecbbc686
|
|\ |
|
| |
| |
| |
| |
| |
| |
| | |
If no data, should return 0 frame count and NULL ptr.
Bug: 23293002
Change-Id: Ib5364e5bceb15c2dddc4a16e85299b409cf4e137
|
| |
| |
| |
| |
| |
| |
| |
| |
| | |
Also determine the number of 'normal' cameras present on
camera service startup, and ensure that all normal cameras have
IDs lower than the 'strange' cameras.
Bug: 23194168
Change-Id: I1f7b14825cb52707de698a955f85da1eaa932663
|
|\ \ |
|
| |/
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
Potential deadlock conditions this addresses, include:
- Not waking up waiting threads for several situations where
the status had been updated.
- Not waking up all waiting thread when status had been updated
(only one thread was awoken due to use of signal).
- Threads clear status transitions before other waiting threads
have a chance to examine them.
Bug: 22448586
Change-Id: I53ba669d333a83d2bfa1ca3170d34acc6d8fe6e3
|
|\ \
| |/
|/| |
|
| |
| |
| |
| |
| |
| |
| |
| | |
Relax InFlightMap size check for high speed configurations to
allow more pending capture requests.
Bug: 23162274
Change-Id: I955fe9a0754f0daed001f4a2b34ccb50f2465a11
|
|\ \
| |/
|/|
| | |
mnc-dev
|
| |
| |
| |
| |
| | |
Bug: 20537722
Change-Id: I9fa2fcdcfd41cd3370732c70414914993d3dc94e
|
|/
|
|
|
|
|
|
| |
Add permission check if the capture device selected is telephony
RX path.
Bug: 23017158.
Change-Id: Iaa34d836e6cf46b7cbbf2483fcd4306dcd27ce90
|
|
|
|
|
|
|
|
|
|
| |
Populate supported sampling rates, channel masks and formats when enumerating
attached build in capture devices.
Having this information for build-in mic is important for some applications.
Bug: 22729461.
Change-Id: I93f03296447a87c10f2615fa1b1c45e9879b4aa7
|
|
|
|
|
|
|
|
|
|
|
| |
Add a bit to tell the HAL that the PCM data is really encoded audio
wrapped in a data burst.
Otherwise the device may try to play the encoded data directly
which will sound like modulated white noise.
Bug: 22576112
Change-Id: Ib140da96876e849023858fe2510612310501d1ee
Signed-off-by: Phil Burk <philburk@google.com>
|
|\ |
|
| |
| |
| |
| |
| |
| |
| |
| |
| | |
and small buffer size. Also:
Pull out the magic number "12 ms" to a named constant.
Remove obsolete AudioFlinger::mPrimaryOutputSampleRate.
Bug: 22662814
Change-Id: I261f75a222c4505a84aad2493d251bd2dea59f68
|
|\ \
| | |
| | |
| | | |
mnc-dev
|
| |/
| |
| |
| |
| | |
Bug: 22173057
Change-Id: I8f5056ff5a1252c71a3d3b354440551bcd9fd466
|
|/
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
When evaluating on which device a sound is to be played, the policy
must consider which current routing strategy, if any, has priority
for overriding the choice. Here the playback of notifications,
when duplicated over speaker and headphones, was causing the
accessibility prompt to be muted because the it was of a higher
priority, and incompatible with the accessibility routing.
The fix consists in assigning a higher priority to the accessbility
routing strategy over the notification routing strategy.
Bug 18834451
Change-Id: I8228b30a7d80bd61d1c223afb030d9421d4f33cf
|
|\ |
|
| |
| |
| |
| |
| | |
Bug: 22496209
Change-Id: I73311573e8d1ac15fec668a9ef6e6af7a07a1d30
|
|\ \ |
|
| | |
| | |
| | |
| | |
| | | |
Bug: 22542551
Change-Id: I2fe5791a6554a8e2f7fd94593d552d8af18257db
|
|\ \ \ |
|
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | | |
The AudioFlinger kept pausing the audio when playing compressed AC3 or DTS.
This caused pause/resume loops that were hard to break out of.
The AudioFlinger was thinking that the compressed audio was PCM
because the HAL was in PCM mode playing SPDIF data bursts.
It also thought that EAC3 was at 192000 Hz instead of 48000
Hz because the data bursts are played at a higher rate.
This CL adds more calls to the shim that separates the AudioFlinger.
Now the AudioFlinger gets information about the HAL sample rate,
channel masks and format from the shim instead of calling the HAL directly.
The AudioFlinger now uses a different threshold for detecting
underruns when the audio is compressed.
Bug: 19938315
Bug: 20891646
Change-Id: Ib16f539346d1c7a273ea4feb3d3afcc3dc60237d
Signed-off-by: Phil Burk <philburk@google.com>
|
|\ \ \ \ |
|
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | |
| | | | | |
If largest jpeg stream cannot sustain 30 FPS, don't
create jpeg stream until takePicture is called and remove
it after still capture is done.
Also, disable video snapshot for such sensors so video snapshot
won't slow down video recording.
Bug: 22231605
Change-Id: I2b34d2537c224694ae10f2006b5a46be45a1b1a6
|
|\ \ \ \ \
| |_|/ / /
|/| | | | |
|