Fix bug # 3136179.
Keep audio mode as IN_CALL during hangup DISCONNECTING state
to prevent the NORMAL and IN_CALL glitch in auiod setMode.
Change-Id: I5513a3d5c65bd13ac054c9718c4dbd7d6db9eaf3
Initial state should be unknown or we miss the first connected change.
Don't send a disconnected msg when changing network types.
Filter out redundent disconnects.
Add some logging.
bug:3060742
Change-Id: Idc797c1276b7417337a91ed60b12b1bf392d57c0
instead of silently returning null and causing NPE in applications as returning
null is not documented in the javadoc.
Add connection to the connection list in SipCall after dial() succeeds so that
we don't need to clean up if it fails. The original code will cause the failed
connection to continue to live in the SipCall and in next dial() attempt, a new
connection is created and the in-call screen sees two connections in the call
and thus shows conference call UI.
Bug: 3157234, 3157387
Change-Id: Iabc3235f781c4f1e09384a67ad56b09ad2c12e5e
Rewrote interceptKeyBeforeQueueing to make the handling more systematic.
Behavior should be identical except:
- We never pass keys to applications when the screen is off and the keyguard
is not showing (the proximity sensor turned off the screen).
Previously we passed all non-wake keys through in this case which
caused a bug on Crespo where the screen would come back on if a soft key
was held at the time of power off because the resulting key up event
would sneak in just before the keyguard was shown. It would then be
passed through to the dispatcher which would poke user activity and
wake up the screen.
- We propagate the key flags when broadcasting media keys which
ensures that recipients can tell when the key is canceled.
- We ignore endcall or power if canceled (shouldn't happen anyways).
Changed the input dispatcher to not poke user activity for canceled
events since they are synthetic and should not wake the device.
Changed the lock screen so that it does not poke the wake lock when the
grab handle is released. This fixes a bug where the screen would come
back on immediately if the power went off while the user was holding
one of the grab handles because the sliding tab would receive an up
event after screen turned off and release the grab handles.
Fixed a couple of issues where media keys were being handled inconsistently
or not at all, particularly in the case of the new PAUSE, PLAY
and RECORD keys.
Bug: 3144874
Change-Id: Ie630f5fb6f128cfdf94845f9428067045f42892c
In order to reduce object creation the RILRequest objects are
stored in an array when it is unused (max 4). This avoids GC
of the object. The object in turn has references to other
objects which sometimes hold large memory chunks. This fix
releases these references since they are not used anyway.
This will make it possible to GC the Message (mResult) which
in some cases holds references to a Bitmap which sometimes
leads to OutOfMemoryException. The reference is cleared
anyway in RILRequest.obtain(...)
Change-Id: I3b895bc39b5e2f3ab7cc8297c3583ea78e0ebc77
The problem was that when we did a contact lookup based on a SIP address,
the resulting CallerInfo object did not have the person_id field set
correctly. That meant we had no way to look up the photo for that person.
This was because of a missing case in the logic to determine which column
(in the resulting cursor) to use for the person_id lookup. We were
handling lookups fine in the PhoneLookup and Phone tables, but were
missing a case for direct lookups in the Data table (which is how we look
up SIP addresses.)
The fix is to add a case for URIs like
"content://com.android.contacts/data" when looking up the person_id.
Also, since the person_id lookup is pretty hairy (and includes ~20 lines
of comments to explain what it's doing!) refactor it out into a helper
method.
TESTED: Both SIP and PSTN calls; verified that contact name *and* photo
are displayed correctly in all cases.
Bug: 3121292
Change-Id: I2b0083cc5394c1a49bbdc9a4e5651854aedb82f7
When a SIP call is put on hold and no other call is active, the audio mode should not be
switched to incall.
Change-Id: I1307330f10cbfb9c4223bcb9dc4faa79778750af
Merge commit 'd691b36cdea314585b6aa17556ed4c35bb65a69e'
* commit 'd691b36cdea314585b6aa17556ed4c35bb65a69e':
Do another contact lookup if the first one fails and...
Merge commit 'fd0ba81c02c0881f7534333afc8c53e241158a3d'
* commit 'fd0ba81c02c0881f7534333afc8c53e241158a3d':
Silently reject a ringing call when another call is dialing/ringing.
Merge commit '6fe795ecd35c4d49822d349424fc71b660577dfc' into gingerbread-plus-aosp
* commit '6fe795ecd35c4d49822d349424fc71b660577dfc':
Do another contact lookup if the first one fails and...
Merge commit 'baced375ba5f374445c44a2115700d69693794a0' into gingerbread-plus-aosp
* commit 'baced375ba5f374445c44a2115700d69693794a0':
Silently reject a ringing call when another call is dialing/ringing.
Merge commit '1180f2a099a134c40f923c7e4162a5e7d7ca0184' into gingerbread-plus-aosp
* commit '1180f2a099a134c40f923c7e4162a5e7d7ca0184':
Remove ringtone API from SipAudioCall.
+ Avoid concurrent modification when forming >3-way conf call.
+ Revise SipConnection.separate() to put the newly separated call to foreground.
Bug: 3114987
Change-Id: If6204e7e3cc05f4a516c33657a368b53a0ad014d
it's a SIP call and the peer's username is all numeric. The all-numeric username
could be a PSTN number.
Bug: 3105116 (case #2)
Change-Id: I1de9cfac3aab1c4c89935176264d07693adb5e7d
(watch out auto-merge conflict for SipAudioCall).
Bug: 3113033, related CL: https://android-git/g/#change,75185
Change-Id: Ib48d3b990e229e0b341e47e10e76934e1a50d10f
Merge commit 'b595e094e3901ff8a224eaf7d4869e7d2e5649dd' into gingerbread-plus-aosp
* commit 'b595e094e3901ff8a224eaf7d4869e7d2e5649dd':
Return display name in SipConnection.getCnapName().
Remember, the system and main logs are
- Shared resources
- Primarily for recording problems
- To be used only for large grained events during normal operation
Bug: 3104855
Change-Id: I136fbd101917dcbc8ebc3f96f276426b48bde7b7
Merge commit 'f5b7c855fbb69e8614dab5ca15639228a2428784'
* commit 'f5b7c855fbb69e8614dab5ca15639228a2428784':
Keep original phone number in SipConnection.
