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102 Commits

Author SHA1 Message Date
jiayl@webrtc.org
7d4891d3f1 Fixes two issues in how we handle OfferToReceiveX for CreateOffer: 
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.

2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.

BUG=2108
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7068

Review URL: https://webrtc-codereview.appspot.com/16309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7124 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 21:43:15 +00:00
jiayl@webrtc.org
c172320bd2 Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android. 
This reverts commit r7068.

TBR=kjellander@webrtc.org
BUG=2108

Review URL: https://webrtc-codereview.appspot.com/23539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7108 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 20:44:36 +00:00
buildbot@webrtc.org
992febb997 (Auto)update libjingle 74873066-> 74873164 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 16:39:08 +00:00
buildbot@webrtc.org
818b7b3ac9 (Auto)update libjingle 74825084-> 74825992 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7074 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 00:14:03 +00:00
jiayl@webrtc.org
52055a276d Fixes two issues in how we handle OfferToReceiveX for CreateOffer: 
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.

2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.

BUG=2108
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 17:12:25 +00:00
buildbot@webrtc.org
fa4535b270 (Auto)update libjingle 74694022-> 74696326 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 16:49:04 +00:00
buildbot@webrtc.org
3740d74106 (Auto)update libjingle 73927658-> 73927775 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6958 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-22 22:27:04 +00:00
henrike@webrtc.org
0481f15f02 (Auto)update libjingle 73399579-> 73626167 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6928 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-19 14:56:59 +00:00
buildbot@webrtc.org
a09a99950e (Auto)update libjingle 73222930-> 73226398 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 17:26:08 +00:00
buildbot@webrtc.org
65b98d12c3 (Auto)update libjingle 72839629-> 72847605 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6854 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 22:09:08 +00:00
buildbot@webrtc.org
5b1ebacca2 (Auto)update libjingle 72820109-> 72822008 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6850 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 17:18:00 +00:00
buildbot@webrtc.org
d509678a4e (Auto)update libjingle 72819313-> 72820109 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6849 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 16:57:07 +00:00
buildbot@webrtc.org
94b996cc18 (Auto)update libjingle 72785516-> 72819313 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 16:47:14 +00:00
buildbot@webrtc.org
476efa2031 (Auto)update libjingle 72785180-> 72785516 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6842 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 04:55:21 +00:00
jiayl@webrtc.org
56d8e05238 A followup to r6828 to fix a condition check in mediasession.cc. 
BUG=2395
R=juberti@chromium.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6832 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 23:52:36 +00:00
jiayl@webrtc.org
e7d47a1473 Maintain the order of the m-lines in CreateOffer and CreateAnswer. 
The order in the offer follows the order in the current local description.
The order in the answer follows the order in the current offer.

BUG=2395
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6828 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 19:19:05 +00:00
buildbot@webrtc.org
e0d03f13e4 (Auto)update libjingle 72443101-> 72446860 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6815 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-02 03:04:01 +00:00
buildbot@webrtc.org
6e203d50a3 (Auto)update libjingle 72442050-> 72443101 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-02 01:13:04 +00:00
buildbot@webrtc.org
52148c2f74 (Auto)update libjingle 72430895-> 72442050 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-02 00:56:56 +00:00
buildbot@webrtc.org
7cb60ccae1 (Auto)update libjingle 72407428-> 72430895 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6812 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-01 22:03:36 +00:00
buildbot@webrtc.org
d4e598d57a (Auto)update libjingle 72097588-> 72159069 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-29 17:36:52 +00:00
tkchin@webrtc.org
42fe4350fe Remove Thread::RunningForChannelManager(). 
I haven't heard of this failing, so it should be safe to remove. Let me know if this isn't the case.

BUG=3388
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6695 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 17:52:43 +00:00
tommi@webrtc.org
b5348c64bb Minor refactoring of the session classes. 
Make member variables that never change and are touched on multiple threads, const.
Move implementations of setters/getters of variables that can change, into the cc file in preparation of adding thread correctness checks.

This is a relanding of a cl already reviewed but got reverted by mistake.

TBR=xians@google.com

Review URL: https://webrtc-codereview.appspot.com/12979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6676 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-14 20:11:49 +00:00
wu@webrtc.org
ff1b1bf094 When creating an answer, takes the codec preference from the offer. 
This change is based on RFC3264:

"Although the answerer MAY list the formats in their desired order of preference, it is RECOMMENDED that unless there is a specific reason, the answerer list formats in the same relative order they were present in the offer."

BUG=2868
TEST=unit tests and manually with munge-sdp test
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/14589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6514 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-20 20:57:42 +00:00
buildbot@webrtc.org
bb2d65895b (Auto)update libjingle 69617317-> 69623266 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6508 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-20 14:58:56 +00:00
buildbot@webrtc.org
75ce92086c (Auto)update libjingle 69600065-> 69617317 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6507 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-20 12:30:24 +00:00
buildbot@webrtc.org
58e7c8660c (Auto)update libjingle 69588980-> 69589535 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6503 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-20 00:26:50 +00:00
buildbot@webrtc.org
1ef789d455 (Auto)update libjingle 69568113-> 69587333 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6500 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-19 23:54:12 +00:00
buildbot@webrtc.org
88d9fa63df (Auto)update libjingle 69291002-> 69292418 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 14:11:32 +00:00
buildbot@webrtc.org
117afeec91 (Auto)update libjingle 69188577-> 69260070 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 07:11:01 +00:00
jiayl@webrtc.org
e3cdd9959e Revert "Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio." 
This reverts commit 56631a14bdae24aa0bfaceeb2b57df729fee1227.

TBR=henrike@webrtc.org
BUG=3235

Review URL: https://webrtc-codereview.appspot.com/19669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 22:32:57 +00:00
jiayl@webrtc.org
745a39cced Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio. 
BUG=3235
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6356 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 19:24:02 +00:00
pbos@webrtc.org
910473b31a Fix C++11 -Wnarrowing in channel_unittest.cc. 
Implicit conversion from int to unsigned char inside {} initializers is
ill-formed C++11 and triggers a warning in clang when building it as
such.

BUG=
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6351 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-06 15:44:00 +00:00
buildbot@webrtc.org
1d66be22c8 (Auto)update libjingle 68203780-> 68206793 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 22:54:24 +00:00
jiayl@webrtc.org
8dcd43c4f7 Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF. 
This is the first step toward switching completely to UDP/TLS/RTP/SAVPF.

BUG=2796
R=juberti@webrtc.org, pthatcher@google.com

Review URL: https://webrtc-codereview.appspot.com/13439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6276 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 22:07:59 +00:00
fischman@webrtc.org
39eccefbde Disable ChannelManagerTest.StartupShutdownOnUnstartedThread 
The test is testing a scenario that shouldn't happen.

BUG=3388
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6238 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 17:50:38 +00:00
fischman@webrtc.org
e5063b1733 Thread: delete racy API (Release()) and fix racy code (started()). 
- Thread::Release() wrote a local variable on the calling thread but read it on
  another thread, with no synchronization.  Happily it has no non-test callers
  so deleting it instead of trying to fix it (see bug for details).
- Thread::started_ similarly was racily being written to; replaced with a
  running_ Event, and hid the accessor except for tests & legacy callers,
  with a note about why it's a bad idea.

webrtc/base patched with:
git diff origin --relative=talk/base | patch -p1 -dwebrtc/base
followed by manual merge of 3 thunks that ran afoul of naming differences
between talk/base and webrtc/base.

BUG=3388
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14589005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6236 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-23 17:28:50 +00:00
buildbot@webrtc.org
49a6a27bf0 (Auto)update libjingle 67555838-> 67643194 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6206 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 00:24:54 +00:00
buildbot@webrtc.org
6bfd6196ff (Auto)update libjingle 67052073-> 67134648 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6174 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 16:15:59 +00:00
buildbot@webrtc.org
3e924683d4 (Auto)update libjingle 67037200-> 67043374 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-14 21:29:04 +00:00
buildbot@webrtc.org
ca27236272 (Auto)update libjingle 66541346-> 66556498 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6088 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-08 23:10:23 +00:00
buildbot@webrtc.org
5ee0f05d5f (Auto)update libjingle 66138442-> 66236292 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-05 20:18:08 +00:00
jiayl@webrtc.org
9c16c39e61 Sets the SCTP port codec in the native SessionDescription. 
Previously it's only set when a SDP string is parsed into SessionDescription, causing failuring for native client.

BUG=3141
R=juberti@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 18:30:30 +00:00
buildbot@webrtc.org
af6640fce7 (Auto)update libjingle 65729829-> 65752960 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6004 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 21:31:51 +00:00
henrike@webrtc.org
f5bebd40f3 (Auto)update libjingle 64247466-> 64326665 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5845 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-04 18:39:07 +00:00
sergeyu@chromium.org
e42b8ab129 Cleanups in libjingle to make it compile with chromium_code=1 
Fixed all warnings that show up when compiling libjingle
in chromium with compiling with chromium_code=1.
chromium_code=1 enables various warnings that are off by
default. Most changes are for unused variables and consts.

R=pthatcher@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5769 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 00:31:35 +00:00
henrike@webrtc.org
8b61011b6f (Auto)update libjingle 63293120-> 63352036 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5720 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-18 21:39:10 +00:00
mallinath@webrtc.org
827faae0ec Fixing incorrect memset. 
Found when ENABLE_EXTERNAL_AUTH is enabled in chrome.

TBR=ronghuawu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5691 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-13 02:15:12 +00:00
henrike@webrtc.org
0537634154 (Auto)update libjingle 62713454-> 62865357 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5670 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-10 15:53:12 +00:00
pbos@webrtc.org
371243dfa3 Remove std:: prefixes from C functions in talk/. 
std::memcpy -> memcpy for instance. This change was motivated by a
compile report complaining that std::rand() was used instead of rand(),
probably with a stdlib.h include instead of cstdlib. Use of C functions
without the std:: prefix is a lot more common, so removing std:: to
address this.

BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5657 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-07 15:22:04 +00:00