1015 Commits

Author SHA1 Message Date
Robert Greenwalt
e12aec941d Add some network types that OEM's are asking for.
Adding them hidden so that if OEM's are rolling their own at least they can
use the same values.  Will mark them unhidden in a future sdk release.

bug:3395729
Change-Id: I90eabe036a96e1aa7c8cac49ca51efd9b1776a0c
2011-01-28 14:48:37 -08:00
Jean-Michel Trivi
1155efdc71 am 2ba92c71: do not merge bug 3370834 Cherrypick from master
* commit '2ba92c71b5684dce700cf848bf157153c156df1d':
  do not merge bug 3370834 Cherrypick from master
2011-01-26 14:05:18 -08:00
Jean-Michel Trivi
2ba92c71b5 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
2011-01-26 11:20:01 -08:00
Hung-ying Tyan
df1cc4ef92 am cc019c0c: Merge "Get mute state from active call." into gingerbread
* commit 'cc019c0caa0dd984404dea4d6623ae9d7b8474f1':
  Get mute state from active call.
2011-01-24 21:46:23 -08:00
Hung-ying Tyan
cc019c0caa Merge "Get mute state from active call." into gingerbread 2011-01-24 21:13:08 -08:00
John Wang
78eb92388c am 93300ce2: Merge "Enable recovery in RIL wakelock release check." into gingerbread
* commit '93300ce2d398195d5616a2e924eb4a785274538e':
  Enable recovery in RIL wakelock release check.
2011-01-24 09:26:56 -08:00
John Wang
696794fc13 Enable recovery in RIL wakelock release check.
Wakelock will get released while
1) no request pending to be sent out, in which mRequestMessagesPending increases
before calling EVENT_SEND and decreases while handling EVENT_SEND.

and

2) no waiting requests sent to RIL but no replied, in which mRequestMessagesWaiting
increases while sending request and decreases while handling response.

Both will be cleared while WAKE_LOCK_TIMEOUT occurs to recovery from out of sync situation.

bug: 3369427, 3370827
Change-Id: Ib2fc54db3b155bd3fb1296ad83720b7836708caf
2011-01-21 17:46:08 -08:00
Hung-ying Tyan
65a7f147de Get mute state from active call.
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
2011-01-11 15:32:30 +08:00
Hung-ying Tyan
05c53067b6 am 273d2ea3: Merge "Fix setting audio group mode in SipPhone." into gingerbread
* commit '273d2ea3f986f1611d2cf303cc5b93f820c14dd3':
  Fix setting audio group mode in SipPhone.
2011-01-04 17:47:47 -08:00
Hung-ying Tyan
273d2ea3f9 Merge "Fix setting audio group mode in SipPhone." into gingerbread 2011-01-04 17:45:36 -08:00
John Wang
e85229988e am 06fccc32: Merge "Clear request list while timeout." into gingerbread
* commit '06fccc325123bf4c9ebd04ac9300b504436724fe':
  Clear request list while timeout.
2011-01-03 08:46:47 -08:00
Hung-ying Tyan
1d12ef09a8 Fix setting audio group mode in SipPhone.
Bug: 3119690
Change-Id: I495d3c031ee4c272d360fe19553ef9726a3f8771
2010-12-29 16:07:17 +08:00
John Wang
00d520b66c Clear request list while timeout.
The wakelock will be kept held if there is outstanding requests
in request list. When WAKE_LOCK_TIMEOUT occurs, all requests
in mRequestList already waited at least DEFAULT_WAKE_LOCK_TIMEOUT
but no response. Those lost requests return GENERIC_FAILURE and
request list is cleared.

bug:3292426
Change-Id: I369c6ba4d6836d65ef616140e48c7304faf888f0
2010-12-28 17:14:18 -08:00
Jeff Brown
fa93584a4f am c6f2b3b3: Merge "Fix policy issues when screen is off. (DO NOT MERGE)" into gingerbread
* commit 'c6f2b3b302c06b8b7b81ec7e3a43a7df1813d0e0':
  Fix policy issues when screen is off. (DO NOT MERGE)
2010-12-23 12:43:48 -08:00
Jeff Brown
eb9f7a01b0 Fix policy issues when screen is off. (DO NOT MERGE)
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.

Bug: 3144874
Change-Id: Iebb91e10592b4ef2de4b1dd3a2e1e4254aacb697
2010-12-22 16:00:21 -08:00
Jean-Baptiste Queru
b3177c135b am 749c627f: Merge "Support for KSC5601 on SIM."
* commit '749c627fc0f30dd3db051f22f20b69a51dc19e59':
  Support for KSC5601 on SIM.
2010-12-21 12:05:04 -08:00
Erik Zivkovic
aad6f806c5 Support for KSC5601 on SIM.
Korean phones write to the ADN record of the SIM in a non-standard way.
When UCS2 is not used, the alphaTag will be written in the KSC5601
encoding. This contribution adds support for reading that format when
a Korean SIM card is used.

Also adds support for KSC5601 in SMS.

Change-Id: I81a4a6949359b4d23a937ac2d813bafed2b85ff6
2010-12-21 10:26:48 -08:00
Wink Saville
a740413022 am d0ffef4b: am 19b23afa: Merge "Fix for phone app crash in Icc Card."
* commit 'd0ffef4b43594e540cb867da18d4403b4f583622':
  Fix for phone app crash in Icc Card.
2010-12-09 14:02:09 -08:00
Wink Saville
d0ffef4b43 am 19b23afa: Merge "Fix for phone app crash in Icc Card."
* commit '19b23afadf1053a8e06cb3444d9cdae3405ad9a1':
  Fix for phone app crash in Icc Card.
2010-12-09 13:58:26 -08:00
Wink Saville
19b23afadf Merge "Fix for phone app crash in Icc Card." 2010-12-09 13:50:28 -08:00
Wink Saville
cc41b92ed7 am 0a5ae453: am 7f7474dd: Merge "frameworks/base/telephony: Release wakelock on RIL request send error"
* commit '0a5ae453e50144b31c7f544714feb5bbc7b77d18':
  frameworks/base/telephony: Release wakelock on RIL request send error
2010-12-08 21:57:03 -08:00
Wink Saville
0a5ae453e5 am 7f7474dd: Merge "frameworks/base/telephony: Release wakelock on RIL request send error"
* commit '7f7474ddd6170b68b8b58cc03b75df85c96f08f2':
  frameworks/base/telephony: Release wakelock on RIL request send error
2010-12-08 21:53:24 -08:00
Wink Saville
7f7474ddd6 Merge "frameworks/base/telephony: Release wakelock on RIL request send error" 2010-12-08 21:37:38 -08:00
Uma Maheswari Ramalingam
cc7605ce61 Fix for phone app crash in Icc Card.
- Check for active phone in ICC handler before processing messages.

- Boundary check for gsm/cdma subscription app index
while retrieving the ICC Card App.

Change-Id: I3d54447e8d48e3482763e78eeb2a737a34cec321
2010-12-08 10:33:24 -08:00
Anshul Jain
60bb9c9a81 frameworks/base/telephony: Release wakelock on RIL request send error
Android telephony does not release the partial wakelock right away if
there is an error in sending the RIL request. The wake lock is released
only after EVENT_WAKE_LOCK_TIMEOUT occurs that prevents the phone to go
in power collpase. The change is to release the wake lock as soon as the
error in send is detected.

Also, change RIL#send not not send a request if there is no connection to
vendor RIL, as the request will always fail.

Change-Id: Ia39a4b9ac12f4064e301a65abfd26409d49babe1
2010-12-08 10:25:41 -08:00
Wink Saville
d9c7fde961 am a00d89e8: am 2b858cae: Merge "Telephony: Add support to read 3GPP2 sms from CSIM/RUIM"
* commit 'a00d89e824c523c51955176b61e50472828fd8b1':
  Telephony: Add support to read 3GPP2 sms from CSIM/RUIM
2010-12-07 22:22:01 -08:00
Wink Saville
a00d89e824 am 2b858cae: Merge "Telephony: Add support to read 3GPP2 sms from CSIM/RUIM"
* commit '2b858caecb3c293c47b48eed12a55a49e3039874':
  Telephony: Add support to read 3GPP2 sms from CSIM/RUIM
2010-12-07 22:18:14 -08:00
Wink Saville
2b858caecb Merge "Telephony: Add support to read 3GPP2 sms from CSIM/RUIM" 2010-12-07 21:57:53 -08:00
Chung-yih Wang
c9cc9ab590 am 5f86d7f5: Merge "Fix SIP bug of different transport/port used for requests." into gingerbread
* commit '5f86d7f50beba9f6327b8a04defe4e989a153d4a':
  Fix SIP bug of different transport/port used for requests.
2010-12-06 22:10:29 -08:00
Chung-yih Wang
f053292d7a Fix SIP bug of different transport/port used for requests.
bug: http://b/3156148
Change-Id: I4fa5b274d2e90ebde12d9e99822dc193a65bad32
2010-12-07 10:36:19 +08:00
John Wang
ae2a441911 am 4567847d: Add "canDial" check.
* commit '4567847d461afac08a80518637a0e48eff3c5247':
  Add "canDial" check.
2010-12-01 11:10:42 -08:00
John Wang
4567847d46 Add "canDial" check.
For bug #3164802.

CallManager allow a new phone call only if ALL of the following are true:

- Phone is not powered off
- There's no incoming or waiting call
- There's available call slot in either foreground or background
- The foreground call is ACTIVE or IDLE or DISCONNECTED.

Change-Id: I0124d600fd8c63b8c608301f3889b3faec47f1db
2010-12-01 10:26:49 -08:00
Hung-ying Tyan
ed34b244f1 am d7116ff1: Merge "Do not suppress error feedback during a SIP call." into gingerbread
* commit 'd7116ff1f0d1a3c14992273d0b899c3b71ba6d3f':
  Do not suppress error feedback during a SIP call.
2010-11-30 22:55:37 -08:00
Hung-ying Tyan
d7116ff1f0 Merge "Do not suppress error feedback during a SIP call." into gingerbread 2010-11-30 22:53:26 -08:00
David Brown
4c11eee7ec am 04639ba0: Reduce the outrageous verbosity of CallerInfo.toString().
* commit '04639ba0a939988d00131e61458807dac650f9c3':
  Reduce the outrageous verbosity of CallerInfo.toString().
2010-11-30 18:11:21 -08:00
David Brown
04639ba0a9 Reduce the outrageous verbosity of CallerInfo.toString().
Bug: 3121292
Change-Id: Ia8383891ef29a003acbd627b25ce87a187ef95c0
2010-11-30 15:49:48 -08:00
David Brown
b9c19be7c1 am 91abcb62: Merge "Fix bug 3121292: Contact photo not shown correctly for SIP calls" into gingerbread
* commit '91abcb624a6a873a2becbbf0f8186d6533daeb89':
  Fix bug 3121292: Contact photo not shown correctly for SIP calls
2010-11-30 15:24:42 -08:00
David Brown
91abcb624a Merge "Fix bug 3121292: Contact photo not shown correctly for SIP calls" into gingerbread 2010-11-30 15:20:45 -08:00
Wink Saville
2ebb3a2d9b am f3166799: Merge "Fix GSM permanent failure handling, DO NOT MERGE." into gingerbread
* commit 'f316679971be356dbb01f991e95742bc5f2a8383':
  Fix GSM permanent failure handling, DO NOT MERGE.
2010-11-30 11:46:34 -08:00
Hung-ying Tyan
0e58a95298 am 0bba9535: Merge "Throw proper exceptions in SipManager" into gingerbread
* commit '0bba9535413f9ceefe03f1cef9ddaddccd05cae5':
  Throw proper exceptions in SipManager
2010-11-30 11:46:01 -08:00
Wink Saville
f316679971 Merge "Fix GSM permanent failure handling, DO NOT MERGE." into gingerbread 2010-11-30 08:15:41 -08:00
Hung-ying Tyan
4189d99b6e Do not suppress error feedback during a SIP call.
Bug: 3124788
Change-Id: Ia0a06f72336d1795515428eba0c9f875c32d13d1
2010-11-30 17:00:45 +08:00
Hung-ying Tyan
0bba953541 Merge "Throw proper exceptions in SipManager" into gingerbread 2010-11-30 00:51:22 -08:00
Jean-Baptiste Queru
8484e57f30 resolved conflicts for merge of e4ae7fc3 to gingerbread-plus-aosp
Change-Id: I2e0a0ed622bc4c32d79936b30ebbf9068b3bdee8
2010-11-22 16:32:34 -08:00
Henrik Hall
95bc625e29 Enabling cell broadcast (SMS-CB) support in the platform.
Adding a simple API enabling applications to control SMS-CB reception.
Implementing parsing, assembly and dispatching of SMS-CB messages over GSM.

Change-Id: Iee841605a45a3af60c7602af175056afb03a38da
2010-11-19 15:00:00 +01:00
Jean-Baptiste Queru
977d01f392 am 27c06bab: Merge "Release reference when putting RILRequest back into the pool."
* commit '27c06bab513a1893444d50bb5dedbad5c0100029':
  Release reference when putting RILRequest back into the pool.
2010-11-17 18:09:08 -08:00
Wink Saville
a20d02c2e1 Fix GSM permanent failure handling, DO NOT MERGE.
Wait until all APN's have been tried before checking for permanent errors
and then, don't do retires only if all of the APN's had permanent errors.

Also, don't disable the requested apn type because if we do we won't
be able to setup data because there won't be an apn type.

This was tested by creating a new non existent APN, I chose:
  Name="badapn1"
  APN="badapn1"
  Server="noapn.com"

Then selecting "badapn1" will cause a permanent error.

bug: 3202729
Change-Id: I182c7197456c849176ce08d7d1459359f8c3b30e
2010-11-17 15:33:36 -08:00
John Wang
1ed7d65b5d am fac4a689: Merge "Fix the audio mode glitch during hangup." into gingerbread
* commit 'fac4a689f86b0d46a2c76cec0a6ce2f4bac2a22a':
  Fix the audio mode glitch during hangup.
2010-11-10 19:28:31 -08:00
John Wang
d19f44f3e3 Fix the audio mode glitch during hangup.
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
2010-11-10 15:35:51 -08:00
Hung-ying Tyan
8d1b2a17d9 Throw proper exceptions in SipManager
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
2010-11-03 18:09:31 +08:00