Commit Graph

70 Commits

Author SHA1 Message Date
cac1b38135 Expose RTCConfiguration to java JNI and add an option to disable TCP
BUG=4585, 4589
R=glaznev@webrtc.org, juberti@google.com, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49809004

Cr-Commit-Position: refs/heads/master@{#9125}
2015-04-30 19:35:32 +00:00
77d444a433 Handle the case when hoststring is empty.
BUG=chromium:480536
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46109004

Cr-Commit-Position: refs/heads/master@{#9081}
2015-04-24 13:38:17 +00:00
0e209b03bf Update bundle behavior to match BundlePolicy spec in http://rtcweb-wg.github.io/jsep/.
BUG=1574
R=juberti@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36659004

Cr-Commit-Position: refs/heads/master@{#8851}
2015-03-24 16:30:02 +00:00
bbce5efaa6 Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8582

Committed: https://code.google.com/p/webrtc/source/detail?r=8607

Review URL: https://webrtc-codereview.appspot.com/43529004

Cr-Commit-Position: refs/heads/master@{#8609}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8609 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 04:39:07 +00:00
d43b2c098d Revert "Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%."
This reverts commit 86c33e3a94f51f8e4b4f305708ec327786ad3794.

TBR=guoweis@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/47409004

Cr-Commit-Position: refs/heads/master@{#8608}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8608 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 04:03:44 +00:00
86c33e3a94 Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8582

Review URL: https://webrtc-codereview.appspot.com/43529004

Cr-Commit-Position: refs/heads/master@{#8607}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8607 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-05 03:40:48 +00:00
fc2f146af2 Revert "Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%."
This reverts commit bbbdeed2bff31777ca7d298d17336fe94626f5b3.

TBR=juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/41109004

Cr-Commit-Position: refs/heads/master@{#8585}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8585 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 04:50:41 +00:00
bbbdeed2bf Turn on IPv6 for WebRTC as default as required before ramping the experiment to 30%.
BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43529004

Cr-Commit-Position: refs/heads/master@{#8582}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8582 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-03 23:27:41 +00:00
877ac765ad Cleanup and prepare for bundling.
- Add a GetOptions function. Needed for eventual bundle testing to
  confirm that channel options are preserved.
- Simplify unit tests and cleanup unused code.

This is a re-roll of 8237 (https://webrtc-codereview.appspot.com/39699004) with a default GetOption implementation.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38909004

Cr-Commit-Position: refs/heads/master@{#8245}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8245 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-04 22:03:41 +00:00
c5f697135e Revert 8237 "Cleanup and prepare for bundling."
libjingle_peerconnection_objc_test consistently failing on Mac64 Debug.

> Cleanup and prepare for bundling.
> 
> - Add a GetOptions function. Needed for eventual bundle testing to
>   confirm that channel options are preserved.
> - Simplify unit tests and cleanup unused code.
> 
> BUG=1574
> R=pthatcher@webrtc.org, tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/39699004

TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34959004

Cr-Commit-Position: refs/heads/master@{#8241}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8241 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-04 10:22:43 +00:00
af01d93aa2 Cleanup and prepare for bundling.
- Add a GetOptions function. Needed for eventual bundle testing to
  confirm that channel options are preserved.
- Simplify unit tests and cleanup unused code.

BUG=1574
R=pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39699004

Cr-Commit-Position: refs/heads/master@{#8237}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8237 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-03 23:14:18 +00:00
5f93d0a140 Update libjingle license statements at top of talk files for consistency
BUG=2133
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 21:36:13 +00:00
7169afd9d5 With IPv6 enabled, it's important to know whether IPv6 is really used or not. BestConnection is tracked for this purpose. Also added a test case to verify the end to end behavior.
BUG=411086
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30919005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7814 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 17:59:29 +00:00
fd0efb694a Remove deprecated PeerConnection APIs.
Removes PeerConnectionObserver::OnError.
Removes MediaConstraints argument to PeerConnection::AddStream.
None of these have ever been implemented and have been removed from the spec.

R=tommi@chromium.org

Review URL: https://webrtc-codereview.appspot.com/24189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7650 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 12:16:36 +00:00
b5d045e94d ReAdd PeerConnectionInterface::AddStream to fix Chrome build.
AddStream(MediaStreamInterface* stream, const MediaConstraintsInterface* constraints);
This will be removed once Chrome has been updated.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7608 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 13:01:33 +00:00
c2dd5ee2c0 Prepare for removal of PeerConnectionObserver::OnError.
Prepare for removal of constraints to PeerConnection::AddStream.

OnError has never been implemented and has been removed from the spec.
Also, constraints to PeerConnection::AddStream has also been removed from the spec and have never been implemented.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7605 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 11:31:29 +00:00
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
1732df6129 Use flags set by the port allocator.
Currently, port allocator flags are ignored. This is inconvenient if you
want to have your own PortAllocatorFactory subclass.

BUG=webrtc:3958
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7524 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-27 03:01:37 +00:00
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
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
2c1bcea1bc Enable ipv6 by default for webrtc under a Finch experiment.
Reapply 23529005 after fixing the build break issue (Chromium:582133002)

Committed: https://code.google.com/p/webrtc/source/detail?r=7253

Review URL: https://webrtc-codereview.appspot.com/23529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7278 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 16:23:02 +00:00
97ed39344a Reapply 23529005 after fixing the build break issue (Chromium:582133002)
Review URL: https://webrtc-codereview.appspot.com/23529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7253 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-19 21:06:12 +00:00
595b23c66f Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..."
Breaks Chrome build and prevents rolling WebRTC into Chrome DEPS.

> Enable ipv6 by default for webrtc under a Finch experiment.
> 
> BUG=413437 (chromium)
> https://code.google.com/p/chromium/issues/detail?id=413437
> 
> Review URL: https://webrtc-codereview.appspot.com/23529005

TBR=guoweis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7190 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16 08:58:22 +00:00
cd309e3168 Enable ipv6 by default for webrtc under a Finch experiment.
BUG=413437 (chromium)
https://code.google.com/p/chromium/issues/detail?id=413437

Review URL: https://webrtc-codereview.appspot.com/23529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7184 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-15 16:31:13 +00:00
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
3d81b1b22a Relanding https://code.google.com/p/webrtc/source/detail?r=7093, after it got
reverted due to some internal compile failures.

In this CL changes are done in portallocator_unittest.cc, in particular to EXPECT_EQ checking in new tests.

Original patch committed in https://code.google.com/p/webrtc/source/detail?r=7093

TBR=juberti@webrtc.org
BUG=1179

Review URL: https://webrtc-codereview.appspot.com/22329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7118 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-09 14:38:10 +00:00
8b0b21161a Revert 7093: "Implementing ICE Transports type handling in libjingle transport."
TBR=mallinath@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/28419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7112 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 22:46:28 +00:00
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
7256d31d28 Implementing ICE Transports type handling in libjingle transport.
BUG=1179
R=juberti@webrtc.org, bemasc@webrtc.org, jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7093 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 04:08:44 +00:00
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
5b06b06cc0 Revert 6897 (i.e. Reland 6863) - "Revert 6863 "Refactor StatsCollector and associated..."
The bot that had the problem was using an old version of STL, so relanding.

> Revert 6863 "Refactor StatsCollector and associated types."
> 
> Breaks chrome compilation on Mac:
> 
> /Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/vector.tcc:252:8:
> error: no matching constructor for initialization of
> 'webrtc::StatsReport'
>           _Tp __x_copy = __x;
>               ^          ~~~
> /Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/stl_vector.h:608:4:
> note: in instantiation of member function
> 'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
> >::_M_insert_aux' requested here
>           _M_insert_aux(end(), __x);
>           ^
> ../../content/renderer/media/mock_peer_connection_impl.cc:282:11:
> note: in instantiation of member function
> 'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
> >::push_back' requested here
>   reports.push_back(report1);
>           ^
> ../../third_party/libjingle/source/talk/app/webrtc/statstypes.h:49:3:
> note: candidate constructor not viable: requires 0 arguments, but 1
> was provided
>   StatsReport() : timestamp(0) {}
> 
> 
> 
> > Refactor StatsCollector and associated types.
> > * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> > * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> > * Report ids are now const.
> > * Copying of data has been greatly reduced.
> > * This change includes preparation work for making GetStats fully async.
> > 
> > This is a reland of r6778 which was reverted due to fyi bots failing.
> > I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
> > 
> > R=xians@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/15119004
> 
> TBR=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/21169004

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6908 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-15 08:38:30 +00:00
22fa032f22 Revert 6863 "Refactor StatsCollector and associated types."
Breaks chrome compilation on Mac:

/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/vector.tcc:252:8:
error: no matching constructor for initialization of
'webrtc::StatsReport'
          _Tp __x_copy = __x;
              ^          ~~~
/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/stl_vector.h:608:4:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::_M_insert_aux' requested here
          _M_insert_aux(end(), __x);
          ^
../../content/renderer/media/mock_peer_connection_impl.cc:282:11:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::push_back' requested here
  reports.push_back(report1);
          ^
../../third_party/libjingle/source/talk/app/webrtc/statstypes.h:49:3:
note: candidate constructor not viable: requires 0 arguments, but 1
was provided
  StatsReport() : timestamp(0) {}



> Refactor StatsCollector and associated types.
> * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> * Report ids are now const.
> * Copying of data has been greatly reduced.
> * This change includes preparation work for making GetStats fully async.
> 
> This is a reland of r6778 which was reverted due to fyi bots failing.
> I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
> 
> R=xians@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/15119004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6897 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13 23:11:04 +00:00
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
730bf30da7 Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.

This is a reland of r6778 which was reverted due to fyi bots failing.
I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6863 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-11 14:08:33 +00:00
b18bf5e47d Adds the support of RTCOfferOptions for PeerConnectionInterface::CreateOffer.
Constraints are still supported for CreateOffer, but converted to RTCOfferOptions internally.

BUG=3282
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6822 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04 18:34:16 +00:00
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
185636cf70 Revert of 6778 "Refactor StatsCollector and associated types."
Breakes FYI bots.

BUG=N/A
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6783 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-25 18:44:42 +00:00
190d269c0f Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.

(This is a reland of the original attempt in r6747)

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6778 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-25 10:32:30 +00:00
39f831fbb0 Re-revert of 6747 "Refactor StatsCollector and associated types."
Breakes FYI bots.

BUG=N/A
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6772 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-24 14:20:52 +00:00
437d57db5b (Auto)update libjingle 71775619-> 71778545
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6771 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23 21:40:28 +00:00
8c7e3291a9 Revert 6747 "Refactor StatsCollector and associated types."
Breakes FYI bots.

BUG=N/A
TBR=ajm@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6770 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23 21:38:58 +00:00
756b8462eb Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome.  This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.

R=xians@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=6745

Review URL: https://webrtc-codereview.appspot.com/18819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6747 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 11:24:17 +00:00
fd61a1d693 Revert 6745 "Refactor StatsCollector and associated types."
Broke build on android.

> Refactor StatsCollector and associated types.
> * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome.  This is the first phase.
> * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> * Report ids are now const.
> * Copying of data has been greatly reduced.
> * This change includes preparation work for making GetStats fully async.
> 
> R=xians@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/18819004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6746 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 11:05:28 +00:00
647e05cfcd Refactor StatsCollector and associated types.
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome.  This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6745 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-21 10:55:11 +00:00
03505bcb7a Make StatsCollector depend on always having a valid session pointer.
This is required since the session pointer is currently used on multiple threads but there's no synchronization code to guard it.
I'm removing the set_session() method and session() getter since they would cause problems if used without synchronization.

This is a reland of an already reviewed cl that got reverted by mistake.

TBR=xians@google.com

Review URL: https://webrtc-codereview.appspot.com/13959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6677 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-14 20:15:26 +00:00
001fd2d503 Fire OnRenegotiationNeeded only for the first SCTP DataChannel.
Subsequent DataChannels do not need renegotiation since SCTP data streams are not negotiated through SDP.

BUG=2431
R=pthatcher@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6268 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-29 15:31:11 +00:00
b445f26f24 Fixing correct UMA metric for PeerConnection enabled with IPv4 Vs IPv6.
BUG=N/A
TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21499007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6242 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 22:19:37 +00:00
d37bcfa882 Changed enums to less generic names.
IPv4/IPv6 will be sent when RegisterUMAObserver is called. This is done
as Initialize is not called through interface.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14469006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6112 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-12 23:10:18 +00:00
0f2a22b3fa Removed sending metrics from PeerConnection about IPv4 and IPv6.
Reasons: 1: There is memcheck failure.
         2: DoInitialize is called before RegisterUMAObserver,
            which means this will be never triggered in real cases.

BUG=3326
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6097 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-09 21:15:06 +00:00
1567b8cf8c (Auto)update libjingle 66540208-> 66541346
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6085 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-08 19:54:16 +00:00