Commit Graph

80 Commits

Author SHA1 Message Date
900996290c Add methods to set the ICE connection receiving_timeout values.
BUG=

Review URL: https://codereview.webrtc.org/1231913003

Cr-Commit-Position: refs/heads/master@{#9572}
2015-07-13 19:19:42 +00:00
0f620f4e31 Make sure we process all pending offer/answer requests before terminating.
This fixes a bug in the WebRtcSessionDescriptionFactory where messages would be dropped or worse yet processed after the factory was deleted.

BUG=chromium:507307

Review URL: https://codereview.webrtc.org/1231823002

Cr-Commit-Position: refs/heads/master@{#9557}
2015-07-09 10:25:04 +00:00
54360510ff Add flakyness check based on the recently received packets.
BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1207563002 .

Cr-Commit-Position: refs/heads/master@{#9553}
2015-07-08 18:08:39 +00:00
d935f912b1 Don't try to parse empty Ice urls.
https://crrev.com/7c4e7458b5ce99c13a75d5be7d718ef94e2f8f9f added support
to pass a list of urls for IceServer configurations. This CL fixes a
potential crash when empty urls are passed.

BUG=2096
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51969004

Cr-Commit-Position: refs/heads/master@{#9334}
2015-05-29 20:14:28 +00:00
7c4e7458b5 Support multiple URLs in PeerConnectionInterface::IceServer
This adds support for multiple URLs in a IceServer configuration as
defined in http://w3c.github.io/webrtc-pc/#idl-def-RTCIceServer.

BUG=2096
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/57489004

Cr-Commit-Position: refs/heads/master@{#9320}
2015-05-28 21:06:44 +00:00
64dad838e6 Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
The original change was reverted due to a breakage in the chrome build.
This change includes a fix for this.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49329004

Cr-Commit-Position: refs/heads/master@{#9169}
2015-05-11 10:44:20 +00:00
1f629232d5 Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit fd32f35aff8fc28ec084bddc274de284e0422a57.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55369004

Cr-Commit-Position: refs/heads/master@{#9165}
2015-05-10 09:06:20 +00:00
fd32f35aff Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit cdb47a4533b7b1e29e803ed6591a68bb1a4f1692.

Contains a tentative fix to the chrome build breakage caused by the
original change.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47139004

Cr-Commit-Position: refs/heads/master@{#9164}
2015-05-10 09:03:00 +00:00
cdb47a4533 Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit 208a2294cde839025318f1b3d57559cb0611a4e7.
Breaks the Chrome build.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53399004

Cr-Commit-Position: refs/heads/master@{#9161}
2015-05-08 12:03:46 +00:00
208a2294cd Adding a new constraint to set NetEq buffer capacity from peerconnection
This change makes it possible to set a custom value for the maximum
capacity of the packet buffer in NetEq (the audio jitter buffer). The
default value is 50 packets, but any value can be set with the new
functionality.

R=jmarusic@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50869004

Cr-Commit-Position: refs/heads/master@{#9159}
2015-05-08 10:58:51 +00:00
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