| 111df679afd212115bff88481874715d98b04cdf |
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27-Jan-2011 |
Eric Laurent <elaurent@google.com> |
Fix issue 2988031.
Limit SYSTEM stream volume when a headset is connected and music is playing.
Change-Id: Ieb44ae5bb53ffa9cd5fe8e317798eed279b78df8
/frameworks/base/services/audioflinger/AudioPolicyManagerBase.cpp
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| 2ba92c71b5684dce700cf848bf157153c156df1d |
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15-Nov-2010 |
Jean-Michel Trivi <jmtrivi@google.com> |
do not merge bug 3370834 Cherrypick from master
Cherripick from master CL 79833, 79417, 78864, 80332, 87500
Add new audio mode and recording source for audio communications
other than telelphony.
The audio mode MODE_IN_CALL signals the system the device a phone
call is currently underway. There was no way for audio video
chat or VoIP applications to signal a call is underway, but not
using the telephony resources. This change introduces a new mode
to address this. Changes in other parts of the system (java
and native) are required to take this new mode into account.
The generic AudioPolicyManager is updated to not use its phone
state variable directly, but to use two new convenience methods,
isInCall() and isStateInCall(int) instead.
Add a recording source used to designate a recording stream for
voice communications such as VoIP.
Update the platform-independent audio policy manager to pass the
nature of the audio recording source to the audio policy client
interface through the AudioPolicyClientInterface::setParameters()
method.
SIP calls should set the audio mode to MODE_IN_COMMUNICATION,
Audio mode MODE_IN_CALL is reserved for telephony.
SIP: Enable built-in echo canceler if available.
1. Always initialize AudioRecord with VOICE_COMMUNICATION.
2. If echo canceler is available, disable our echo suppressor.
Note that this CL is intentionally not correcting the
getAudioSourceMax() return value in MediaRecorder.java as the
new source is hidden here.
Change-Id: Ie68cd03c50553101aa2ad838fe9459b2cf151bc8
/frameworks/base/services/audioflinger/AudioPolicyManagerBase.cpp
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| e4eaa317f611b76467ea25ba03c528a03f2bc780 |
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01-Dec-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2641884: Bluetooth volume is dependent on in call volume.
The problem is that the code in AudioPolicyManagerBase::checkAndSetVolume() that forces
voice volume to max when setting bluetooth SCO volume is not called if the bluetooth stream
volume did not actually change. So even if we re apply volumes when switching to bluetooth
device, the volume voice volume is not changed and remains what it was when routed to earpiece
What makes things worse on Passion is that stream volumes are limited when connected to bluetooth
and their actual value does not change as soon as they exceed the limit threshold.
Change-Id: Id7c317db45b392a1c20dca2859678e3c64a371ed
/frameworks/base/services/audioflinger/AudioPolicyManagerBase.cpp
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| b87b53d7a863da9049b66621d98caf720b8dec60 |
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02-Nov-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 3142808.
There is a bug in the way audio policy manager handles A2DP interface suspend/restore
when SCO is used. This bug is not new but has been triggered by a change in the timing
of the events received by audio policy manager when a call is setup and torn down
introduced by commit 164a8f86c7e48992691368c4895709c3bdb835a4.
The fix consists in grouping the control of A2DP suspended state in a single function
that is called systematically when conditions affecting this state are changed:
- call state change
- device connection/disconnection
- change in forced usage.
Change-Id: I46ee2399ee5547b60511fc6cfd32e2720091b0f8
/frameworks/base/services/audioflinger/AudioPolicyManagerBase.cpp
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| b8453f4e0e32d11926f4c5badf656cf1062fbe08 |
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28-Aug-2010 |
Eric Laurent <elaurent@google.com> |
Fix issue 2952766.
The problem is that the audio policy manager does not handle the input devices
when forced use for telephony is changed.
The problem does not appear in a call over PSTN becasue only teh output devices drives the
routing of in call audio to/from the base band.
The fix consists in modifying AudioPolicyManagerBase::setForceUse() to check for active inputs
and update the input device if needed.
Change-Id: I0d36d1f5eef1cce527929180c29b025439902f10
/frameworks/base/services/audioflinger/AudioPolicyManagerBase.cpp
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| 8ed6ed0b6216a9dfcbcd6a5ba6a62d28a901baec |
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13-Jul-2010 |
Eric Laurent <elaurent@google.com> |
Audio policy manager changes for audio effects
Added methods for audio effects management by audio policy manager.
- control of total CPU load and memory used by effect engines
- selection of output stream for global effects
- added audio session id in parameter list for startOutput() and stopOutput().
this is not used in default audio policy manager implementation.
Modifications of audio effect framework in AudioFlinger to allow moving and reconfiguring
effect engines from one output mixer thread to another when audio tracks in the same session
are moved or when requested by audio policy manager.
Also fixed mutex deadlock problem with effect chains locks.
Change-Id: Ida43484b06e9b890d6b9e53c13958d042720ebdb
/frameworks/base/services/audioflinger/AudioPolicyManagerBase.cpp
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| 08e83bb3b7cc41f603867acbeb1168019cf535fe |
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15-Jul-2010 |
Mathias Agopian <mathias@google.com> |
move native services under services/
moved surfaceflinger, audioflinger, cameraservice
all native services should now reside in this location.
Change-Id: Iee42b83dd2a94c3bf5107ab0895fe2dfcd5337a8
/frameworks/base/services/audioflinger/AudioPolicyManagerBase.cpp
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