Age | Commit message (Collapse) | Author |
|
Bug: 153277327
Test: atest TelecomUnitTests
Change-Id: Icad12c4144bf185c24ee80bcbdd1aec29550bf4e
|
|
Most of this was previously hidden; these last stragglers were missed.
Test: make update-api ; verify hidden
Fixes: 152394802
Change-Id: I41bda5b8ad368e1c88e4dd9e45d978a111a22e53
|
|
Test: Manual
Bug: 62170207
Change-Id: I06a256adb0e1910d40809c91bcdd42c56a142842
|
|
|
|
- New @SystemApis on Conference and PhoneAccount were missing the required
permissions annotations.
- Rename PROPERTY_ASSISTED_DIALING_USED to PROPERTY_ASSISTED_DIALING
- Standardize get/setConnectionStartElapsedRealtimeMillis method naming
across Connection and Conference classes.
- Clarify Conference#sendConferenceEvent API documentation to match the
docs present for similar method in Connection; include some examples of
valid event/extras combinations.
- Update TelecomManager#getDefaultdialerPackage to use UserHandle instead
of userId.
- Move Conference#getConnectionStartElapsedRealtimeMillis to public API
since the setter is already part of the public API.
Test: Run Telecom and Telephony CTS tests.
Test: Run Telephony unit tests.
Test: Perform manual single-party-conference regression test to confirm
that conference behavior does not regress.
Bug: 147301297
Bug: 148286830
Bug: 148284863
Bug: 148284843
Bug: 148287068
Bug: 148285484
Bug: 148285560
Change-Id: I1f446d81859fa109d74af3661a42a0bd224de5aa
Merged-In: I1f446d81859fa109d74af3661a42a0bd224de5aa
|
|
Supports initiation of a conference call
by directly adding participants to existing call
Test: Manual
Bug: 62151032
Change-Id: I4e60efafab4761ae65a460fdc6c4cacc3e233220
|
|
Adding a new Call API which supports passing a user-specified call
rejection reason down to the lower layers for reporting to the network.
Part of the VERSTAT spec involves support for this type of signaling, so
it makes sense to also support it here as well.
There are two potential types of reject reason:
declined - user declined the call because they want it to go to voicemail
or don't want to talk to the caller right now.
unwanted - this is a nuisance call and the user never wanted to receive it.
Bug: 135929421
Test: Added new CTS test to validate API pathways.
Test: Ran existing telecom and telephony unit tests.
Test: Modified test dialer app to use the new reject API and verified that
the reject reason signals down to the modem and translates to the correct
reject cause.
Change-Id: I6f25fafa2b2620e2839e5d3a9fb986f1130fa165
|
|
Add support for Adhoc Conference calls
Test: Manual
Bug: 62151032
Change-Id: Id50d235595d2133f867848ffdebdfe11e2f1c896
|
|
Test: Added new CTS tests to verify operation.
Bug: 135929421
Change-Id: I42360dad677060e03ecec865f31145b1760cf46a
|
|
Add new TelecomManager#getDefaultDialerPackage which is multiuse aware;
this is used when showing the voicemail notification.
Remove some @hide methods from Conference; push these inline.
Move ConferenceParticipantConnection into frameworks/opt/net/ims since
it is just an IMS implementation detail.
Bug: 141576016
Test: Manual smoke test.
Test: Run unit tests.
Test: Run CTS tests.
Change-Id: I39b6955cb14cc1ca68b05c620c3d09a2cdfe30c9
Merged-In: I39b6955cb14cc1ca68b05c620c3d09a2cdfe30c9
|
|
Test: atest TelecomUnitTests
Change-Id: Ibb80b5739083ad9f85ee06f4d9f0017f4cb605bd
|
|
This is important for ensuring the original call direction for existing
connections added from Conference Event Packages matches the call direction
of the original calls merged into the conference.
Also moved a utility function into ConferenceParticipant from
ConferenceParticipantConnection to make it generically usable inside
Telephony.
Test: Run all unit tests.
Test: Manual testing using VoLTE conference calls with mix of MO and MT
calls; verify call logging is appropriate.
Bug: 134471046
Change-Id: Iab09397b811782ab0f876aac02070e3447d81f09
|
|
When a device A creates a conference call containing device B and C, both
B and C can receive IMS signaling to indicate that they are in a conference
call. This occurs on most domestic carriers; the Telephony framework
uses the "multiparty" indicator on the IMS call to switch the call to a
conference call.
We made some changes to how conference calls are logged in Q which improves
the accuracy of the call durations. We used to log calls as they're merged
into a conference. In the case of a participant in a remotely hosted
conference call, we'd log the call as soon as they are remotely added
to the conference. This is unfortunate as the call durations are grossly
under-reported.
The conference call logging changes now assume we'll log the conference
event package children in the conference instead of the participants which
merge into the conference itself. On domestic carriers, since the
call on B (or C) becomes a conference, we would then no longer log the
call.
This is FURTHER complicated because on some carriers, B and C will ALSO
receive a conference event package from the network showing all the
participants in the conference. So if B hangs up on the conference, they
will have entries in their call log for A and C, which is really strange
because a call to C was never originated on their device.
In Telecom we need to ensure we do not log remotely hosted conference
participants, and we need to ensure that we DO log a remotely hosted
conference as if its just a single party call.
To accomplish this we need:
1. the address and name display information associated with the call from
A-B / A-C prior to the call turning into a remotely hosted conference.
We need this to log to the call log
2. the remotely hosted conference and participants need to be marked in a
manner that Telecom can identify them.
Test: Manual test.
Test: Add unit test to cover this logging scenario.
Bug: 132325382
Change-Id: I65e713f68d1695a48d96dacbf7faa4476cd8d815
|
|
The method queryRemoteConnectionServices had the issue that it assumes
there is a single connection manager for the device. This assumption does
not work on a multisim device. Since the ConnectionManager is associated
with a particular carrier, this means that the connection mgr for one
carrier could try to impact calls destined for another carrier.
This change ensures that the calling package is passed into Telecom so
that we can determine which RemoteConnectionServices which are available to
the calling connection manager.
Test: Manual test on DSDS with a connection mgr carrier and another carrier.
Bug: 131856987
Change-Id: I46d80dc68adaab7fd4374f023d7ba4242804c253
|
|
Pipe through the onConnectionEvent call between the conference host
connection and the android.telecom.Conference object.
Fixes: 130404376
Test: manual
Change-Id: Ifad3e59dc9764aa2efb2b9766271e18853bf1c76
|
|
call." am: 89e53d81ce am: 955c9adb36
am: 4fd8e1b376
Change-Id: I7ab00c1b82858014fb3135474a64a9fa44d8283b
|
|
call." am: 89e53d81ce
am: 955c9adb36
Change-Id: I5af4c4591cd825adfeab8be56d7c60d29905f1c4
|
|
Adding @hide APIs which Telephony can use to make a conference call with
a single participant look like its a standalone call.
Test: Manual testing
Bug: 75975913
Change-Id: Id8532234ab295785fc749b120898f43911e12637
|
|
am: 57adb99451
Change-Id: I477a610e0602e464847c1a1ccb13b21644ee1ae8
|
|
am: 14aa42cfda
Change-Id: Id461e2430301c62be5ee76f5046370069cb1a34c
|
|
Test: make ds-docs
Bug: 117449040
Change-Id: I282a2e960bbf722bf3a72dd932e3bf685abb74e5
Exempt-From-Owner-Approval: Docs-only change
|
|
2cbf44dc2f am: e784d5e5ca am: a4c0632746
am: 47dceea7d8
Change-Id: I28e69d1b32e71b467dded2d90118a46f9ef3fffc
|
|
am: 2cbf44dc2f
Change-Id: I823efd5765b77fcf55f4c38a6091d980530abae8
|
|
In android original design, the duration of CDMA MO call is started from
the dial command sent, so it is not the real duration of the active time.
In this patch, a new message is registered to listen the event of the call
accepted, and then reset the duration when the event happens.
Change-Id: Icc447012030ae243f200ec2c83b7d5210af9b31c
|
|
am: fd3e10b1bc" into pi-dev-plus-aosp
am: 6823c8a822
Change-Id: I1352a44cbb55fdeeb79771b27055a70b83a3cbb8
|
|
am: fd3e10b1bc
Change-Id: I8f7842fa84fcce91a292b32189cd827b4a125fb0
|
|
Although Telecom ensures that new calls use voip audio mode, the default
value from a Connection would override the Telecom default.
Bug: 76362663
Test: Modify test app to ensure it does not set voip audio mode, ensure
Telecom uses voip audio mode for new calls.
Change-Id: Ie6477659cf6dabd08f371d4958ece1d258cd3106
|
|
1. Update handover API docs for clarity.
2. Added an unknown value per API review comments.
3. Renamed HANDOVER_FAILURE_DEST_USER_REJECTED to
HANDOVER_FAILURE_USER_REJECTED
3. Removed the HANDOVER_FAILURE_DEST_INVALID_PERM constant since it isn't
used (methods which deal with permissions throw security exceptions).
Test: Make doc and verify documentation.
Fixes: 73751004
Fixes: 73750515
Fixes: 73750817
Merged-In: I7860fcd813f25adaaccf632f2c61dd4138a0a889
Change-Id: I7860fcd813f25adaaccf632f2c61dd4138a0a889
(cherry picked from commit c7a86b14a8e50d979b6b1c9e3dffe94748e2bc93)
|
|
am: 168a77237f
Change-Id: I7860fcd813f25adaaccf632f2c61dd4138a0a889
|
|
am: 8917fc21cc
Change-Id: Ic4a159b838c952594b0860ded69fc07c74180961
|
|
|
|
Bug: 73750116
Test: current telecom test
Change-Id: I74e9636c305b164bf01c3136c53e9a432101945b
|
|
1. Update handover API docs for clarity.
2. Added an unknown value per API review comments.
3. Renamed HANDOVER_FAILURE_DEST_USER_REJECTED to
HANDOVER_FAILURE_USER_REJECTED
3. Removed the HANDOVER_FAILURE_DEST_INVALID_PERM constant since it isn't
used (methods which deal with permissions throw security exceptions).
Test: Make doc and verify documentation.
Change-Id: Id21d6b4c83d5c773ab38d78eb6b1886a1ac4dadf
Fixes: 73751004
Fixes: 73750515
Fixes: 73750817
|
|
* Add a new API to allow Telecom to inform ConnectionServices when the
RTT text stream changes
* No longer set the RTT property from ConnectionService. Client apps
should be setting properties themselves.
* Add Dialer-side RTT property in order to remove the dependence on
checking the RTT streams, which have a complex lifecycle
Bug: 72951201
Bug: 72648661
Test: manual, with real RTT calls and Dialer's SimulatorConnection, also
cts
Change-Id: Ic4c7d883d2dc6baf8e8c0eaa4df58d7de8762b9e
Merged-In: Ic4c7d883d2dc6baf8e8c0eaa4df58d7de8762b9e
|
|
|
|
* Add a new API to allow Telecom to inform ConnectionServices when the
RTT text stream changes
* No longer set the RTT property from ConnectionService. Client apps
should be setting properties themselves.
* Add Dialer-side RTT property in order to remove the dependence on
checking the RTT streams, which have a complex lifecycle
Bug: 72951201
Bug: 72648661
Test: manual, with real RTT calls and Dialer's SimulatorConnection, also
cts
Change-Id: Ic4c7d883d2dc6baf8e8c0eaa4df58d7de8762b9e
|
|
d1134525fc am: a0b41a82ce
am: 4c09901931
Change-Id: I493ade2029398983a76fd41fcf9f080ee2f06b95
|
|
|
|
Change name of conference connection elapsed time method to make it more
clear what it is for. Updated documentation of this method and its
companion to make it more clear what they are for.
Test: Compile - this is a docs and naming change only.
Bug: 70639525
Change-Id: I02662cb0331cba0d1fe2d4353438a68f334f9192
|
|
ba0f9d2973 am: f39bef4801
am: ac48eed9c0
Change-Id: I4695aca7d886f37341672ea5815770e3ccdf7a51
|
|
Call deflection feature is useful to deflect MT call to another
number.
Test: Manual
Bug: 62170348
Change-Id: Idfbcc175a856aa0bb9476f8c73d7a614a3af0700
|
|
Adding the ACCEPT_HANDOVER runtime permission which an app must have in
order to accept handovers (this is per design).
Adding missing onHandoverComplete method in the android.telecom.Connection
API (per design).
Finishing plumbing for android.telecom.Call#onHandoverComplete API.
Fix issue where the new handover API methods would never get called; the
legacy handover extra was being used in this case when it should not have
been.
Bug: 65415068
Test: Verified using new CTS tests
Change-Id: If1558f6a23911862c02ac5b18fb62d86911ed7e2
Merged-In: If1558f6a23911862c02ac5b18fb62d86911ed7e2
|
|
Adding the ACCEPT_HANDOVER runtime permission which an app must have in
order to accept handovers (this is per design).
Adding missing onHandoverComplete method in the android.telecom.Connection
API (per design).
Finishing plumbing for android.telecom.Call#onHandoverComplete API.
Fix issue where the new handover API methods would never get called; the
legacy handover extra was being used in this case when it should not have
been.
Bug: 65415068
Test: Verified using new CTS tests
Change-Id: If1558f6a23911862c02ac5b18fb62d86911ed7e2
|
|
This add new api interface to ConnectionService to support the
connection service focus api.
Bug: 69651192
Test: manually
Change-Id: Iea49d95b086d32a0ebaf8e9f34fe4556953a0fd5
Merged-In: Iea49d95b086d32a0ebaf8e9f34fe4556953a0fd5
|
|
7661e81f85 am: c840efcf7e
am: 5d7c823e16
Change-Id: I57395d052d284ca3b8071b71dc762c1147d1107a
|
|
|
|
Bug: 65415068
Test: Manual
Design doc:
https://docs.google.com/document/d/1qY3oAzjff_4A1ttYb_CGrE_OwTRmXMG_KGsIuPT1ey8/edit#Bug:
Change-Id: Ic0c4af19098252389648007628affc19a44f89dd
Merged-in: Ic0c4af19098252389648007628affc19a44f89dd
|
|
Bug: 65415068
Test: Manual
Design doc:
https://docs.google.com/document/d/1qY3oAzjff_4A1ttYb_CGrE_OwTRmXMG_KGsIuPT1ey8/edit#
Change-Id: I692bb14fba66733154378c2dda525aa85c471a38
Merged-in: I692bb14fba66733154378c2dda525aa85c471a38
|
|
When a ConnectionService implementation returns a null connection, log this
and also set a unique disconnect reason to indicate in the telecom
dumpsys what happened.
Test: Manual
Bug: 70385625
Change-Id: Iff9846d434d400c4cf036e9ac46167cfb6f6b58c
|
|
Bug: 65415068
Test: Manual
Design doc:
https://docs.google.com/document/d/1qY3oAzjff_4A1ttYb_CGrE_OwTRmXMG_KGsIuPT1ey8/edit#Bug:
Change-Id: Ic0c4af19098252389648007628affc19a44f89dd
|