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* Because we get a hellish firestorm during offload fallback, and
our effects library really dislikes this.
Change-Id: I22adb55d04502eedf03ce22f35f8b9f61de1c7b0
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This is moved to custom policy manager in HAL sources.
Change-Id: I716580f7d3399f1628c26bd3b79bd6c6a52fbbf6
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Squash of the following:
commit 40c41a743a4345ad26b6eab41755c5b474f75853
Author: chrmhoffmann <chrmhoffmann@gmail.com>
Date: Sat Nov 29 19:32:27 2014 +0100
pre kitkat audio legacy policy fix for hotword (ok google)
Change-Id: I5bb90fb90f373461a98609226a5cea50ac5266e9
commit d47b306954b96142323c79b976edeb3eaa3b46dd
Author: chrmhoffmann <chrmhoffmann@gmail.com>
Date: Sun Nov 23 18:22:35 2014 +0100
commit d47b306954b96142323c79b976edeb3eaa3b46dd
Author: chrmhoffmann <chrmhoffmann@gmail.com>
Date: Sun Nov 23 18:22:35 2014 +0100
pre-kitkat audio policy blobs re-activated.
Change-Id: Idfcb03d0a4aa0a099bf5a17b7bf592ab402e3255
Change-Id: Iae65c6ccb6dcb120a6fc20978b8a69e6af39a51a
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* Some effects modules are racy and don't tolerate being destroyed
and immediately resurrected on the same session. This is
the common case when switching tracks, and the use of default
output effects makes the problem even worse. Certain apps
which handle gapless in a sloppy way are also to blame.
* Instead of immediately nuking the entire descriptor with the
stream, just decrease the refcount and defer it for 10 seconds.
If it needs resurrected, the refcount will be increased and
the delayed release command will not shoot it in the face.
Change-Id: I068dd72c4180023a74eb9ccbe8a180f6f0683dbf
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* In M, we now have the ability to define a default set of audio effect
on a per-stream basis. This allows us to get around the problem of
apps not sending the control intents so we can implement smart
global effects for specific media types.
* We still need a session id in order to get a handle and configure them
from an app like AudioFX, so we'll need to add some plumbing in
order to send an event to interested applications.
* This patch implements the native side of this. The Java layer will
call down thru AudioSystem and register a callback which will be
invoked by the audio policy when default effects are updated on
a stream. This callback will receive both the stream type as well
as the session id.
* Attaching this listener requires that the caller hold the
MODIFY_AUDIO_ROUTING permission.
Change-Id: I142b15f2585ffca6a953c3e828e2a7c07b24f56c
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* In case a device has a /vendor partition which cannot be
modified, this allows us to short-circuit the audio_effects.conf
file which might be placed there with one of our own which
lives on /system.
Change-Id: Ief87bd4cfba2c3188b0dff122d91f773b7f3d92d
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* Fixes the build for legacy audio policy
* Broken by: de3b1c01146d2103f6bf56e2659a9c650734945a
Change-Id: I990ce609b3da81c462b35c7dd3f7b11d189e2d2e
Signed-off-by: Xuefer <xuefer@gmail.com>
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Handle startOutput on AudioPolicyService output command thread
to serialize with concurrent releaseOutput/stopOutput calls.
CRs-Fixed: 944129
Change-Id: Ie333c736750c7dfb31d3036d79dfff13cd0486fc
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Change-Id: If91536d2be0165c90effccdfa2b92722223eb905
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* Fix compilation with debug enabled
* Allow FLAC offload to be enabled independent of QCOM_HARDWARE,
tested with the Nexus 5X and is working!
* Disable the FLACExtractor if offload is enabled since it can
only output decoded PCM. This will force FFMPEG to be used,
which does everything we need for FLAC offload
Change-Id: I7d71c153a6a6ea7df8e32bc73f5cbe9f51cdcf64
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Direct output is not selected for voip usecases as
AUDIO_OUTPUT_FLAG_VOIP_RX is not added to
sOutputFlagNameToEnumTable.
Add this flag to output flags list.
Change-Id: Ifccb78a7b8579da0a65eb3ea7347756c664246a8
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https://android.googlesource.com/platform/frameworks/av into cm-13.0
Android 6.0.1 release 3
Change-Id: I2f2a1fe1b58c828e8341556996211562d6e195ab
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Make sure no active capture takes place while in call and the
capture device is the same as the call TX device.
Bug: 23977713.
Change-Id: Ia379d5e626c150ea49f0cdd581c881dbf0b8399d
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When transitioning from speaker phone to normal mode and a USB
headset is connected, we must force a teardown of active input
as the new input device cannot be reached via current input stream.
Bug: 24406915.
Change-Id: I9f7c1d3f3cba285c25a6da1d54a3d15a5c1f378a
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Audio can be heard when muting volume as the minimum volume
value takes 0.0f instead of VOLUME_MIN_DB (-758).
This patch fixes the minimum volume value which should be
VOLUME_MIN_DB instead of 0.0f.
Change-Id: I4cb8093b6d9f18e5330ba5f770b9133baa77542f
Signed-off-by: Safa Boumaiza <Safax.boumaiza@intel.com>
Signed-off-by: Zhiquan Liu <zhiquan.liu@intel.com>
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For Proxy thread to be started , a2dp state has to updated from
the audiopolicy.
Now , added changes to send the BT A2DP status to audio HAL.
- A- Family doesn't support separate output for a2dp , removed the
getA2dpOutput(). getDeviceForStrategy() is selecting the proper
device using mA2dpSuspended flag.
CRs-Fixed: 751090
Change-Id: I599de562a8674ee611e10575fcc85e4715707a63
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Change-Id: If037b4eb6bbd864ccab050565899ce508ee90f9e
Signed-off-by: AdrianDC <radian.dc@gmail.com>
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git://codeaurora.org/platform/frameworks/av into cm-13.0
Change-Id: Ia41df61c3ccfeb809572b63a4b1a8ca6bd85dfb2
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Synchronize public APIs of EffectDescriptorCollection
CRs-Fixed: 920103
Change-Id: I04ccac526c6f99e61e43288776653d6b7ff325c4
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Fixe race condition in AudioCommandThread::threadLoop() where a
command can be inserted in first position in the queue after the sleep
time has been calculated causing a longer delay than expected.
Also fix a failure to hold a wake lock while commands are still in the queue.
Bug: 22707905.
Change-Id: I813626986677bf00106acb37ee20d3dd75d5cf33
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Synchronize access to APM when getDevicesForStream is called
on APM.
CRs-Fixed: 913227
Change-Id: I2ba6922341f035375270b02000ef5a7e078f6b5a
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Change-Id: I774f75b493c47386ca1eaf004d663432f1041a66
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- Send USB connected/disconnected events to Audio HAL
only in case of USB AOA as these are used only in
case of USB AOA
Change-Id: I7db508f6eff5e8601d54f1e1bfa603358f9d6703
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setPhoneState calls getDeviceForInputSource to check if
voice_tx device is available as part of Primary Audio
Module. getDeviceForInputSource always returns
DEVICE_IN_COMMUNICATION for SOURCE_VOICE_COMMUNICATION
on A Family and this is not added to Available devices
This is leading to creation of txPatch when it is not
needed
Fix is to use AUDIO_SOURCE_VOICE_CALL instead of
AUDIO_SOURCE_VOICE_COMMUNICATION
Change-Id: I9aa7dd2139b8374ec7f06676d06ef5addea4b2a9
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- On A family, Only DEVICE_IN_COMMUNICATION is supported
by voip_tx profile. getDeviceForInputSource should
always return DEVICE_IN_COMMUNICATION to load
voip_tx profile
Change-Id: I3bb2fb70a89eadc35c2ced0c7f717ec15120bfe7
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USB AOA is supported by primary HAL on A family through
customized connect/disconnect parameters.Added support
for that in AudioPolicyManager
Change-Id: I5212027b2711c24a9750629774f69d669a9d164f
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Added entry for DEVICE_IN_COMMUNICATION to support
VOIP usecase.
Change-Id: Ie7e5131b776d7b93c53ca0796dffa11316091805
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Currently, audio doesn't play on Brillo because the policy manager's
default value for all streams is set to 0. In Android, this value is
changed by calls from AudioService.java. If we set the default value to
1 in Brillo, then even if there are no calls, we will have audio.
BUG=none
TEST=manual
Change-Id: I1418c9ecfb4f1c81c9c0946eee11deed3c65b358
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Synchronize access to APM when getDevicesForStream is called
on APM.
Change-Id: I2ba6922341f035375270b02000ef5a7e078f6b5a
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Android 6.0.0 release 26
Change-Id: I8a57007bf6efcd8b95c3cebf5e0444345bdd4cda
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Because a preemtible capture session can preempt another one,
we end up in an endless loop situation were each session is allowed to restart
after being preempted, thus preempting the other one which restarts and so on.
To avoid this situation, we store which audio session was preempted when
a particular input started and prevent preemption of this active input by this session.
We also inherit sessions from the preempted input to avoid a 3 way preemption loop etc...
Bug: 24007220.
Change-Id: I0eab5299440ef3ab9e987635dc9a300cf42f2c79
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Fixe race condition in AudioCommandThread::threadLoop() where a
command can be inserted in first position in the queue after the sleep
time has been calculated causing a longer delay than expected.
Also fix a failure to hold a wake lock while commands are still in the queue.
Bug: 22707905.
Change-Id: I813626986677bf00106acb37ee20d3dd75d5cf33
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No routing operation should be allowed between setMode() and setPhoneState()
when starting a call as the audio HAL relies on a precise sequence of mode change
and routing change to select the initial audio device for the call.
Bug: 24083591.
Change-Id: I2d5ef62c11cf7aedc2ec7ca5e5fadd7ac875afbc
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Only change the volume for AUDIO_STREAM_ACCESSIBILITY when
changing the media volume.
Bug 23366575
Change-Id: Ic42c726ec4d47615c20500c20e4d43cef159b3c6
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Always play Transmitted Through Speaker stream at max volume instead
of tracking media stream volume.
Bug: 23729461.
Change-Id: I61021f59617bb851a4cd7da80924e587029e96a3
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Do not mute the TTS stream if a dedicated output for
TTS is available.
Bug: 22100304.
Change-Id: Ic126100ce1144f765bc0ae42c8ea87fa47cfb822
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Output handle of AudioOutputDescriptor is required by derived
implementation of some AudioPolicyManager methods.
These methods need to access attributes of an output which is not
accessible through AudioOutputDescriptor base class unless output
handle is made available here.
Change-Id: I96ef9fd1c5a94874acb897245501ba2f9c9ab0c0
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- handle non default input buffer size
- fix issues associated with non-default buffer size
Change-Id: Iad31f5c85601983b01e18e130b1c3eecb3988e30
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Warn allocation failures explicitly rather than crash
trying to access unallocated memory
Change-Id: Ie86c3ac130917e1f4030eb8207ac8350cba7711d
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- add support for effects on direct pcm output
Change-Id: I2fbac63c623bf51a03e5e91828369739d33329f3
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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
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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
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allow effects in case outout is direct pcm
Change-Id: I2ad7eacf11642a4ca9f892b61124293d0dc503a9
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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
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- Add support for 5.1 channel recording
Change-Id: If060fffb2e198f516f40e85390489de2108be5d1
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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
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add support for APE decoding
Change-Id: I55e8f4b3b87f4bdf1c99774d702506eb7c2f05b5
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add support for decoding/offloading ALAC
audio formats
Change-Id: Id66f0cb6c140113741962e119148bf434de3d064
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