drgreen/android_frameworks_base
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
87,827 Commits 1 Branch 0 Tags
Commit Graph

284 Commits

Author SHA1 Message Date
Chia-chi Yeh
5a7c6d298e am 1c2eab2d: am 955ab37c: Merge "RTP: Enable GSM-EFR codec." into gingerbread 
Merge commit '1c2eab2d86faa9c647a9893f761a50cfa28d9d57'

* commit '1c2eab2d86faa9c647a9893f761a50cfa28d9d57':
  RTP: Enable GSM-EFR codec.
 2010-09-30 10:59:48 -07:00
Chia-chi Yeh
21ae1ad6a6 RTP: Minor fixes with polishing. 
Change-Id: I50641373989e512fb489b5017edbcfd7848fe8b9
 2010-09-30 16:07:44 +08:00
Hung-ying Tyan
d29e075418 Merge "Add uri field to SipManager.ListenerRelay" into gingerbread 2010-09-30 00:05:58 -07:00
Hung-ying Tyan
9e1d308e99 Add uri field to SipManager.ListenerRelay 
in case mSession is not available.

Change-Id: Ifee2c129e48aa1177f648f176413ab6aa5606770
 2010-09-30 15:00:34 +08:00
Chia-chi Yeh
dfd1484e3b Merge "RTP: Adjust the jitter buffer to 512ms." into gingerbread 2010-09-29 23:24:32 -07:00
Chia-chi Yeh
3520bd4313 RTP: Adjust the jitter buffer to 512ms. 
Change-Id: Ia91c1aa1a03b65dbd329ea98383f370844e2b0c0
 2010-09-30 13:51:26 +08:00
Chia-chi Yeh
8ff0722453 am 1254b9c5: am cd386649: Merge "RTP: Revise the workaround of private addresses and fix bugs." into gingerbread 
Merge commit '1254b9c534c5f027f8928fbb3e743e57d55bd13d'

* commit '1254b9c534c5f027f8928fbb3e743e57d55bd13d':
  RTP: Revise the workaround of private addresses and fix bugs.
 2010-09-29 22:13:44 -07:00
Hung-ying Tyan
6a53489ae5 SipService: add UID check. 
Only allow creator or radio user to access profiles.

Change-Id: I548938f117926bcc878419142d1b5d818a4e70df
 2010-09-30 12:40:11 +08:00
Chia-chi Yeh
0a537b78d3 Merge "RTP: Enable AMR codec." into gingerbread 2010-09-29 18:32:24 -07:00
Hung-ying Tyan
2365b78e64 Merge "SIP: misc fixes." into gingerbread 2010-09-29 18:12:12 -07:00
Chia-chi Yeh
f88fc1fa90 RTP: Enable AMR codec. 
Change-Id: I49e6bdc1b67306b44173f2f346f8372a50264870
 2010-09-30 08:55:12 +08:00
Hung-ying Tyan
fb3a98b1d8 SIP: misc fixes. 
+ Fix keepalive timer event leak due to the race between stopping timer and
  the async'ed timeout handler
+ SipSessionImpl: set state before handling an event to ensure we get correct
  state when some error occurs during handling the event.
+ Fix potential NPE in SipManager.ListenerRelay.getUri().

Change-Id: I021ee34f83059fd4fbb64b30bea427a5462aa51b
 2010-09-30 08:10:17 +08:00
Chia-chi Yeh
f4ae94229d RTP: Enable GSM-EFR codec. 
Change-Id: I9d84009e4557a0a82c1f9d7d543922741be97c77
 2010-09-30 03:07:57 +08:00
Chia-chi Yeh
fe5298992a RTP: Revise the workaround of private addresses and fix bugs. 
Change-Id: Ie654b569f47049aa452eca8d3e6d4a98ac18469c
 2010-09-30 02:43:48 +08:00
Chia-chi Yeh
dcf2be6cf6 am ebfe5632: am e006e4d2: Merge changes Iae1913fb,I38dbefef into gingerbread 
Merge commit 'ebfe5632db275a89b49ab828064ba90db59702cf'

* commit 'ebfe5632db275a89b49ab828064ba90db59702cf':
  RTP: Enable GSM codec.
  RTP: Refactor out G711 codecs into another file.
 2010-09-28 19:47:07 -07:00
Chia-chi Yeh
e006e4d2c9 Merge changes Iae1913fb,I38dbefef into gingerbread 
* changes:
  RTP: Enable GSM codec.
  RTP: Refactor out G711 codecs into another file.
 2010-09-28 19:40:59 -07:00
Chia-chi Yeh
a6f950c968 RTP: Enable GSM codec. 
Change-Id: Iae1913fb0643f1c66b5d16f24d51924d363e5ef5
 2010-09-29 10:36:52 +08:00
Chia-chi Yeh
9783052ec1 am df31e03c: am 320cdcb1: Merge "RTP: Delay the initialization of AudioTrack and AudioRecord." into gingerbread 
Merge commit 'df31e03c47cd4caf45d8a58cf1fe5893da48ec6c'

* commit 'df31e03c47cd4caf45d8a58cf1fe5893da48ec6c':
  RTP: Delay the initialization of AudioTrack and AudioRecord.
 2010-09-28 17:38:09 -07:00
Hung-ying Tyan
0b3968ae53 am 0d447760: am 6d028dd2: Merge "SIP: Feedback any provisional responses in addition to RING" into gingerbread 
Merge commit '0d44776016cecf1c7c826c4784f8f867a56235f0'

* commit '0d44776016cecf1c7c826c4784f8f867a56235f0':
  SIP: Feedback any provisional responses in addition to RING
 2010-09-28 17:37:04 -07:00
Chia-chi Yeh
78c11b3cf1 RTP: Refactor out G711 codecs into another file. 
Change-Id: I38dbefef2315a28d44683e86a51e69f38e3f20ec
 2010-09-29 05:46:19 +08:00
Chia-chi Yeh
320cdcb122 Merge "RTP: Delay the initialization of AudioTrack and AudioRecord." into gingerbread 2010-09-28 14:41:18 -07:00
Chia-chi Yeh
9083c84af1 RTP: Delay the initialization of AudioTrack and AudioRecord. 
Related to http://b/3043844.

Change-Id: I2c4fd9f64e6eba597d68b2ea1ceeff83103697db
 2010-09-29 05:24:34 +08:00
Hung-ying Tyan
6c6eacda80 am f7e13400: am 624d5b4e: SIP: add DisconnectCause.SERVER_ERROR 
Merge commit 'f7e13400b24bdc5461e640cfb2c9cb2bbb2b6236'

* commit 'f7e13400b24bdc5461e640cfb2c9cb2bbb2b6236':
  SIP: add DisconnectCause.SERVER_ERROR
 2010-09-28 12:47:49 -07:00
Hung-ying Tyan
6057cd00d9 SIP: Feedback any provisional responses in addition to RING 
The only exception is TRYING.
Also remove an unused import in SipSessionGroup.

http://b/issue?id=3021865

Change-Id: I160982b0c4b417362f1fb961217db90c3a585ce5
 2010-09-29 02:26:47 +08:00
Hung-ying Tyan
624d5b4e8c SIP: add DisconnectCause.SERVER_ERROR 
and fix how SipErrorCode.SERVER_ERROR is determinted from server response, not
from local exceptions.

http://b/issue?id=3041332

Change-Id: Idce67e29858d5c7573b98b7fa1fac074913d71d6
 2010-09-28 14:54:13 +08:00
Hung-ying Tyan
a57afb6a6c resolved conflicts for merge of 2a36a778 to master 
Change-Id: Ia70adeef06afddd29c827405fb5657bf9f5a29a3
 2010-09-28 12:17:44 +08:00
Hung-ying Tyan
7e54ef71db Move SipService out of SystemServer to phone process. 
Companion CL: https://android-git/g/#change,70187
http://b/issue?id=2998069

Change-Id: I90923ac522ef363a4e04292f652d413c5a1526ad
 2010-09-28 05:19:35 +08:00
Hung-ying Tyan
5a474a2bb8 am 44669d31: am fd144d76: Merge "SipAudioCall: remove SipManager dependency." into gingerbread 
Merge commit '44669d31d1d5b094d7b7d3e393281440ea0c9536'

* commit '44669d31d1d5b094d7b7d3e393281440ea0c9536':
  SipAudioCall: remove SipManager dependency.
 2010-09-27 11:47:42 -07:00
Hung-ying Tyan
031d878682 am fe2d279c: am 00a22064: SipService: handle cross-domain authentication error 
Merge commit 'fe2d279c5ef571340f20d433badd9f68072299af'

* commit 'fe2d279c5ef571340f20d433badd9f68072299af':
  SipService: handle cross-domain authentication error
 2010-09-27 11:47:32 -07:00
Hung-ying Tyan
fd144d7667 Merge "SipAudioCall: remove SipManager dependency." into gingerbread 2010-09-27 10:54:27 -07:00
Hung-ying Tyan
00a22064ef SipService: handle cross-domain authentication error 
and add new CROSS_DOMAIN_AUTHENTICATION error code and OUT_OF_NETWORK
DisconnectCause.

http://b/issue?id=3020185

Change-Id: Icc0a341599d5a72b7cb2d43675fbddc516544978
 2010-09-27 10:45:24 -07:00
Chung-yih Wang
5e18ad0c53 am 4a04a312: am bd229420: Fix the unhold issue especially if one is behind NAT. 
Merge commit '4a04a3129bd30a996dd302b982aeca8f228f57e8'

* commit '4a04a3129bd30a996dd302b982aeca8f228f57e8':
  Fix the unhold issue especially if one is behind NAT.
 2010-09-27 09:07:43 -07:00
Chung-yih Wang
bd2294204e Fix the unhold issue especially if one is behind NAT. 
+call startAudio() when call is established.

Change-Id: Ib6a1e34017fb83007ce275da1991058e8b803833
 2010-09-27 23:53:39 +08:00
Hung-ying Tyan
3a4197e642 SipAudioCall: remove SipManager dependency. 
Change-Id: I2dc8bf427e52f64529ee0e0261362b975a8917c6
 2010-09-24 23:27:40 +08:00
Hung-ying Tyan
a97c5f7779 Merge "fix build" 2010-09-23 23:24:35 -07:00
Hung-ying Tyan
fb0264096e fix build 
Change-Id: Iff05b5ea7f535f532eec2af1edf78fdf8acfa21c
 2010-09-24 14:23:31 +08:00
Chia-chi Yeh
658bec9567 SDP: remove dead code. 
Change-Id: I2a5764a2b9cabc54b0ac18666e494c1cb39c4e9b
 2010-09-24 10:17:42 +08:00
Hung-ying Tyan
84a357bb6a Refactoring SIP classes to get ready for API review. 
+ replace SipAudioCall and its Listener interfaces with real implementations,
  + remove SipAudioCallImpl.java, most of it is has become part of SipAudioCall,
+ add SipSession and its Listener classes to wrap ISipSession and ISipSessionListener,
+ move SipSessionState to SipSession.State,
+ make SipManager keep context and remove the context argument from many methods of its,
+ rename SipManager.getInstance() to newInstance(),
+ rename constant names for action strings and extra keys to follow conventions,
+ set thread names for debugging purpose.

Change-Id: Ie1790dc0e8f49c06c7fc80d33fec0f673a9c3044
 2010-09-24 10:06:59 +08:00
repo sync
0b7d6de155 Fix the build. 
Change-Id: I82210cb2d41f532583f83ea17e6f2d8d49280a30
 2010-09-23 14:52:24 +08:00
repo sync
84f7f6ba39 SIP: Make SipAudioCallImpl use SimpleSessionDescription instead of javax.sdp. 
Change-Id: I7efff4f29ca84c3e7c17ef066b7186b514a777b2
 2010-09-23 14:07:45 +08:00
Chia-chi Yeh
e6c0c10958 SDP: Add a simple class to help manipulate session descriptions. 
Change-Id: I1631ee20e8b4a9ad8e2184356b5d13de66e03db1
 2010-09-23 13:31:01 +08:00
repo sync
7a69aeffda RTP: Add log throttle for "no data". 
Change-Id: I14d9886a40fa780514cbc6c5bac6fb2a670f55f4
 2010-09-23 05:46:01 +08:00
Chia-chi Yeh
4033a67d0e RTP: Update native part to reflect the API change. 
Change-Id: Ic2858920ad77d7312f2429f89ca509a481363431
 2010-09-23 03:36:43 +08:00
Chia-chi Yeh
37adc522f6 RTP: Add two getters to retrieve the current configuration from AudioStream. 
Change-Id: Iff588130653242f6ddd6a6b663df775ecb276768
 2010-09-23 03:34:14 +08:00
Chia-chi Yeh
32e106b7bd RTP: Extend codec capability and update the APIs. 
Change-Id: I37ba9d83c2de3c5dae2bfc1b7513df7f6fee3c5c
 2010-09-23 03:32:04 +08:00
Hung-ying Tyan
8544560ccc SipPhone: fix missing-call DisconnectCause feedback 
also fix delivering bad news before closing a SipAudioCallImpl object so that
apps can get the current audio-call object state before it's closed:

http://b/issue?id=3009262

Change-Id: I94c19dae8b4f252de869e614ec462b19b4ff2077
 2010-09-20 13:06:30 +08:00
Hung-ying Tyan
97963794af SIP: convert enum to static final int. 
Converts SipErrorCode and SipSessionState.

Change-Id: Iee3a465649ea89d395b2336bbd673c25113e5f93
 2010-09-20 09:51:31 +08:00
Hung-ying Tyan
c4b87477c0 SIP: add config flag for wifi-only configuration. 
http://b/issue?id=2994029

Change-Id: I328da9b0f8b70d660dbcefffdac8250341792101
 2010-09-20 08:03:20 +08:00
Hung-ying Tyan
afa583e655 SipAudioCall: expose startAudio() 
so that apps can start audio when time is right.

Change-Id: I7ae96689d3a8006b34097533bc2434bc3814b82a
 2010-09-17 15:58:18 +08:00
Hung-ying Tyan
9352cf1a4d Add timer to SIP session creation process. 
+ add timer parameter to ISipSession.make/changeCall(),
+ add timer paramter to SipAudioCall.make/answer/hold/continueCall()'s,
+ add timer parameter to SipManager.makeAudioCall(),
+ modify implementation in SipSessionGroup, SipAudioCallImpl accordingly,
+ make SipPhone to use it with 8-second timeout.

http://b/issue?id=2994748

Change-Id: I661a887e5810087ddc5e2318335e2fa427f80ec6
 2010-09-17 13:31:05 +08:00