pthatcher@webrtc.org
4cb3856a4d
Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."
...
This reverts r7939 because it broke Chromium and other depedent projects that rely on certain logic remaining in p2p/base/session.cc and not in webrtc/libjingle/session.cc.
BUG=
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7940 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 02:28:25 +00:00
pthatcher@webrtc.org
536f999e58
Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
...
This is an un-revert of r7992 and r7993.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7939 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 01:22:02 +00:00
pbos@webrtc.org
e728ee03ba
Remove or rename typedefs with _t prefixes.
...
_t prefixes are reserved for additional typenames in POSIX.
R=henrik.lundin@webrtc.org , hta@webrtc.org , stefan@webrtc.org
BUG=162
Review URL: https://webrtc-codereview.appspot.com/36559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7931 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-17 13:43:55 +00:00
pthatcher@webrtc.org
f050791ba0
Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository."
...
This reverts r7992.
It broke the Chromium build because the Chroumium build relies on the logic in webtc/libjingle/session.cc, but Chromium doesn't compile that file.
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7923 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 22:28:03 +00:00
pthatcher@webrtc.org
4afb59903c
Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository.
...
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7922 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 21:37:37 +00:00
pthatcher@webrtc.org
e2b7585bc2
Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository.
...
R=juberti@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7921 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 21:09:08 +00:00
pbos@webrtc.org
18a3896bd2
Revert r7886:7887.
...
Broke build steps in other code that uses securetunnelsessionclient.cc
and others.
TBR=tommi@webrtc.org ,pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/36439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7890 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 07:03:04 +00:00
pthatcher@webrtc.org
dee76f3b89
Move the obvious/easy Jingle-specific code into webrtc/libjingle.
...
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7886 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 21:04:42 +00:00
pthatcher@webrtc.org
40b276ea7b
Cleanup little things found when refactoring.
...
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/33519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7880 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 02:44:30 +00:00
magjed@webrtc.org
35c1ace185
Revert 7698 "WebRtcVideoMediaChannel::SetSendParams: Don't cap r..."
...
Reason for revert is failed testcases:
WebRtcVideoEngineExtendedTestFake.ResetSimulcastSendCodecOnNewFrameSize
WebRtcVideoEngineExtendedTestFake.MultipleSendStreamsDifferentFormats
> WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution
>
> BUG=3936
> R=pthatcher@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/30039004
TBR=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7700 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 16:21:49 +00:00
magjed@webrtc.org
52da44b7e6
WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution
...
BUG=3936
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7698 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 15:43:11 +00:00
henrik.lundin@webrtc.org
f85dbce041
Reapply "Advertise G722 as 8 kHz rather than 16 kHz""
...
This reverts r7653 and relands r7645. The reason for the original revert was that G722 disappeared from the SDP offer. This is now fixed. Also, a unit test was updated compared with the original change.
BUG=3951
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7662 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-07 12:25:00 +00:00
henrike@webrtc.org
269fb4bc90
move xmpp and p2p to webrtc
...
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.
BUG=3379
Review URL: https://webrtc-codereview.appspot.com/26999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 22:20:11 +00:00
tommi@webrtc.org
f15dee6980
Check if a datachannel in the current local description is an sctp channel before assuming rtp.
...
When generating an offer from a local description when 'sctp' is not explicitly set in the
media session options, we were generating an offer with an RTP datachannel even though the
channel in the local description was already sctp.
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7539 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 22:15:04 +00:00
pthatcher@webrtc.org
2e7ee4b28b
Fix the SrtpFilter crash caused by two local offers.
...
BUG=http://crbug.com/421774
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7530 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 16:10:29 +00:00
henrike@webrtc.org
28100cb388
Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
...
BUG=N/A
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 22:03:39 +00:00
henrike@webrtc.org
d1ba6d9cbf
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
...
BUG=3379
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27709005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 17:30:28 +00:00
buildbot@webrtc.org
81ddc78536
(Auto)update libjingle 77701902-> 77709729
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7450 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 22:39:24 +00:00
buildbot@webrtc.org
1ecbe45c7e
(Auto)update libjingle 77689511-> 77696841
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7449 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 20:29:28 +00:00
jiayl@webrtc.org
742922b313
Make the media content send only if offerToReceive is false while local streams exist.
...
We previously do not add the media content if offerToReceive is false.
BUG=3833
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7390 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-07 21:32:43 +00:00
jiayl@webrtc.org
7dfb7fa189
Reland disallowing blocking calls on the worker thread.
...
This fixed the issue that invoking the call when the thread is not started.
BUG=3559
R=juberti@webrtc.org , thorcarpenter@google.com
Review URL: https://webrtc-codereview.appspot.com/24769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7325 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-29 22:45:55 +00:00
pbos@webrtc.org
34f2a9ea72
Initialize SSL in unittest_main.cc.
...
Instead of having each test individually initialize and tear down SSL
move this to unittest_main.cc so that all tests are properly
initialized and new tests "don't have to think about it".
R=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/30549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7316 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-28 11:36:45 +00:00
thorcarpenter@google.com
a21d071607
Reverting part of
...
https://webrtc-codereview.appspot.com/15089004/diff/140001/talk/session/media/channelmanager.cc?context=10&column_width=80
because of a major regression hanging the executable on start.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7309 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-26 17:19:14 +00:00
pbos@webrtc.org
05305116d6
Explicitly initialize SSL for tests.
...
Adding missing SSL initialization/cleanups in
TransportDescriptionFactoryTest and MediaSessionTest.
These being missing prevent these tests from being run individually
without other tests preceding them that initialize SSL.
BUG=3860
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7300 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25 15:50:26 +00:00
jiayl@webrtc.org
3987b6de50
Fix a problem in Thread::Send.
...
Previously if thread A->Send is called on thread B, B->ReceiveSends will be called, which enables an arbitrary thread to invoke calls on B while B is wait for A->Send to return. This caused mutliple problems like issue 3559, 3579.
The fix is to limit B->ReceiveSends to only process requests from A.
Also disallow the worker thread invoking other threads.
BUG=3559
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7290 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 17:14:05 +00:00
pbos@webrtc.org
d60d79a145
Thread annotation of rtc::CriticalSection.
...
Effectively re-lands r5516 which was reverted because talk/-only
checkouts existed. This now resides in webrtc/base/, so no talk/-only
checkouts should be possible.
This change also enables -Wthread-safety for talk/ and fixes a bug in
talk/media/webrtc/webrtcvideoengine2.cc where a guarded variable was
read without taking the corresponding lock.
R=andresp@webrtc.org , mflodman@webrtc.org , pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7284 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 07:10:57 +00:00
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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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.
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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
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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
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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
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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
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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
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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