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* commit '591a5f86bd4b3f1732b935daad909e4e63149f01':
Doc change: fix adt archive filename.
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Change-Id: I1cf33678e52b4e85dca472763bfe53b7ecfcfb8b
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Change-Id: I615bf5177abbc680292e0116491da58835c7396f
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Change-Id: Ief2adca0f7f5c37d9381bd69d39a27aa11ec85dc
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all sdk docs" into gingerbread
* commit 'a7ce2f04ef0ae548370eebd1314558f24f40e8f0':
catchup the gingerbread branch to gingerbread-docs for all sdk docs
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into gingerbread
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This is just a manual copy of all files that changed, rather than a dozen cherry picks.
Change-Id: I672f74a2ec9242a1d96713d1d84bfc4d2d47dd85
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MPEG4-LATM audio with extra trailing bytes.
* commit '2dce338e01678620db0734fc3d84bcb3f2512d62':
DO NOT MERGE: This particular RTSP server streams MPEG4-LATM audio with extra trailing bytes.
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extra trailing bytes.
And now we're just ignoring them. Yay standards.
Change-Id: Ia8c0b9161e606152fb681f0dda3ba901954dc749
related-to-bug: 3353752
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perm." into gingerbread
* commit '1d07b5bc9b0a9e3de4495834242548a600cae1d8':
Make CHANGE_COMPONENT_ENABLED_STATE a signatureOrSystem perm.
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gingerbread
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Change-Id: I53a6671d5478e7dbca44b17d90f1edfa8e3420bc
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* commit '2ba92c71b5684dce700cf848bf157153c156df1d':
do not merge bug 3370834 Cherrypick from master
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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
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* commit '138757db4f19cb626ef5c1d371d4fca004304a08':
Make OBB results a one-way call
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OBB result back to an application needs to be a one-way binder call.
Bug: 3353699
Change-Id: I0e625914d18a001c2fa9d764ea6463f34cf96743
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Change-Id: I52b5045cbef6e4927f86830498050977b2829859
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Change-Id: I9b02206042c74fd5c329fe2b70ed2b6b91d41382
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unresponsive, OTA fails (DO NOT MERGE)" into gingerbread
* commit '239fd805ef7c0e4116b0a89e20caaf287e91f017':
fix [3361121] hang in glClear() - device unresponsive, OTA fails (DO NOT MERGE)
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NOT MERGE)" into gingerbread
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Generally we never want to lock a buffer for write access if it is at
the "head" on the surfaceflinger side. The only exception (1) is when
the buffer is not currently in use AND there is at least one queued
buffer -- in which case, SurfaceFlinger will never use said buffer
anymore, because on the next composition around, it will be able to
retire the first queued buffer.
The logic above relies on SurfaceFlinger always retiring
and locking a buffer before composition -- unfortunately this
didn't happen during a screenshot.
This could leave us in a situation where a buffer is locked by the
application for write, and used by SurfaceFlinger for texturing,
causing a hang.
Here, we fix this issue by never assuming the exception (1), it was
intended as an optimization allowing ANativeWindow::lockBuffer() to
return sooner and was justified when most of SF composition was
done in software. The actual buffer locking is now ensured by
gralloc. We could have handled screenshots in a similar way to
a regular composition, but it could have caused glitches on screen,
essentially, taking a screenshot could cause to skip a frame.
now that we removed the notion of a "inUse" buffer in surfaceflinger
a lot of code can be simplified / removed.
noteworthy, the whole concept of "unlockClient" wrt. "compositionComplete"
is also gone.
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acceleration blackout (DO NOT MERGE)" into gingerbread
* commit '1955a5c9da421dc89bb1a1dd3d3193159192cde9':
partially fix [3306150] HTML5 video with H/W acceleration blackout (DO NOT MERGE)
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(DO NOT MERGE)" into gingerbread
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MERGE)
We used to guarantee that a layer in SurfaceFlinger would never be
destroyed before all references (to its ISurface) on the client
side would be released. At some point, this guarantee got
relaxed to allow to free gralloc resources sooner. This last
change was incorrect, because:
- in implementations with reference-counting the gralloc resources
wouldn't be released anyways, until all the mapping were gone
- in implementations without ref counting, the client side
would most likely crash or do something bad
- it also caused the SharedBufferStack slot to be reallocated
to another surface, which could be problematic if the client
continued to use the surface after the window manager destroyed it.
So, we essentially reinstate the guarantee that layers won't be
destroyed until after all references to their ISurface are
released.
NOTE: This doesn't entirely fix 3306150 because there is another
problem there where the Browser continues to use a surface after it
has been destroyed.
also improve SurfaceFlinger 'dumpsys' log
list the purgatory, which shows windows that have been closed,
but for which the client still has references.
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garbage in some cases (DO NOT MERGE)" into gingerbread
* commit 'd0441f921b2b1879fb44041f521d9bfaa5a547d0':
fix a small bug that caused screenshot to show garbage in some cases (DO NOT MERGE)
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(DO NOT MERGE)" into gingerbread
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MERGE)
we were not clearing the screen entirely, which caused garbage when
the screen wasn't entirely covered by windows.
Change-Id: Ia7aa13c36a8a314e0e8427d419b16b9aa2165ddf
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of gralloc errors (DO NOT MERGE)" into gingerbread
* commit '043a3cddf67e2baf38711bcb125084599f8f9802':
workaround [3201922] display not on: log full of gralloc errors (DO NOT MERGE)
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NOT MERGE)" into gingerbread
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we make sure to call compositionComplete after everytime we do
composition with the GPU (even for the screenshot case), which is
where the buffer locks are released.
Change-Id: I450430d1e4d1ee9ce1023970642378c42cdcfa4c
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libpixelflinger.so. (DO NOT MERGE)" into gingerbread
* commit 'd3fcd0a1c923498921107bb44fbc8b24582a170a':
minor clean-up. SurfaceFlinger doesn't need libpixelflinger.so. (DO NOT MERGE)
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NOT MERGE)" into gingerbread
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Change-Id: I1ddbbbec4fa5b2521ef3787bc28efe1bc90d0060
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Change-Id: I13a79033239d306fcf9291ca7fad7445f2cd1f47
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This makes the system more flexible and allows
adding new technology types without having to
update the API.
Change-Id: Iaee6b633965e501a70e8afc3f1d54d9d94a4d05a
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into gingerbread
* commit 'c929677e017759047d94599faa1b80ebe278fefb':
Increase the number of useful bits in the spectrum.
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Change-Id: I3efc2ac8db689285566df35cb6f594aab41141db
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* commit 'b134223f91c8801d577cb72e92a37cb65fec717a':
Make Mifare Classic increment/decrement operands little endian
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Also make sure they are non-negative.
This is not documented in Mifare Classic spec, but based on findings from NXP:
- Operand should be stored in little-endian format in the transceive buffer
- Tag ignores the sign bit on the operand, its effectively 31-bit unsigned
- Overflow and underflow generates an error.
Change-Id: Id3389b3894ded732c4b00d564ca53f5df651359e
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* commit 'cc019c0caa0dd984404dea4d6623ae9d7b8474f1':
Get mute state from active call.
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Currently, PhoneUtils.getMute() returns the mute state from the foreground phone.
When a SIP call is muted and then put on hold, the call is moved to background
and the SipPhone becomes background phone. At this point, PhoneUtils.getMute()
incorrectly returns false from the idle foreground phone (i.e., GSMPhone).
CallManager provides getMute() but it's not used anywhere. This CL fixes the
method and I'll have another CL to have PhoneUtils.getMute() take advantage of
it.
Bug: 3323789
Change-Id: I6c37500ae93f4e95db3bcd55e24e1ecb58a57c0a
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* commit '14b6d0620b42d1bb3a55778ba452d838a0d89223':
SipService: registers broadcast receivers on demand.
SipService: release wake lock for cancelled tasks.
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* changes:
SipService: registers broadcast receivers on demand.
SipService: release wake lock for cancelled tasks.
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The previous implementation registers receivers when SipService starts up.
If the user doesn't use SIP at all, SipService will still process connecivity
and wifi state change events, which involves holding wake lock and thus
consumes power unnecessarily.
With this CL, SipService is completely idle if the user doesn't use SIP at all.
It registers receivers only when at least one account is opened.
Bug: 3326998
Change-Id: Ib70e0cf2c808e0ebab4c3c43dcab5532d24e5eeb
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Bug: 3327004
Change-Id: I0691cd70edf61f815ecb0613aca85babd89f6cc4
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