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Author SHA1 Message Date
Donald Curtis
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
kwiberg@webrtc.org
eebcab5ce9 rtc::Buffer: Rename length to size, for conformance with the STL 
And add a constructor for creating an uninitialized Buffer of a
specified size.

(I intend to follow up with more Buffer changes, but since it's rather
widely used, the rename is quite noisy and works better as a separate
CL.)

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48579004

Cr-Commit-Position: refs/heads/master@{#8841}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8841 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-24 09:20:19 +00:00
henrik.lundin@webrtc.org
0b44b58a3c Limit disabling of PeerConnectionEndToEndTest.Call to Windows 
The test seems to be flaky only on Windows.

BUG=4464
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44829004

Cr-Commit-Position: refs/heads/master@{#8835}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 19:48:19 +00:00
henrik.lundin@webrtc.org
82e8ae4ee8 Disable PeerConnectionEndToEndTest.Call in libjingle_peerconnection_unittest 
The test has been flaky recently.

BUG=4464
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46689004

Cr-Commit-Position: refs/heads/master@{#8832}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8832 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-23 14:25:50 +00:00
glaznev@webrtc.org
4ddc9387bd Support VP8 hardware encoding and decoding on IA devices. 
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42829004

Cr-Commit-Position: refs/heads/master@{#8812}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8812 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-20 21:21:17 +00:00
guoweis@webrtc.org
840da7b755 Implement Rotation in Android Renderer. 
Make use of rotation information from the frame and rotate it accordingly when we render the frame.

BUG=4145
R=glaznev@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8770

Review URL: https://webrtc-codereview.appspot.com/50369004

Cr-Commit-Position: refs/heads/master@{#8781}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 16:58:49 +00:00
guoweis@webrtc.org
3fffd66dfa Revert "Implement Rotation in Android Renderer." 
This reverts commit 835ec63d8a64bbc8a573a5e0b7a09659188122d2.

TBR=guoweis@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/51399004

Cr-Commit-Position: refs/heads/master@{#8771}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8771 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 04:20:47 +00:00
guoweis@webrtc.org
835ec63d8a Implement Rotation in Android Renderer. 
Make use of rotation information from the frame and rotate it accordingly when we render the frame.

BUG=4145
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50369004

Cr-Commit-Position: refs/heads/master@{#8770}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-18 02:44:39 +00:00
pthatcher@webrtc.org
592470b4ff Remove a dependency of BaseChannel on WebRtcSession by having WebRtcSession push down new media descriptions to BaseChannel rather than having BaseChannel listen to the description changes from WebRtcSession. 
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47599004

Cr-Commit-Position: refs/heads/master@{#8743}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8743 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 21:16:23 +00:00
pthatcher@webrtc.org
4eeef584a7 Remove a hacky dependency of BaseChannel on BaseSession by moving the handling of DTLS setup failure into a signal on BaseChannel rather than a method call on BaseSession. 
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=decurtis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47589004

Cr-Commit-Position: refs/heads/master@{#8740}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8740 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 19:34:40 +00:00
pthatcher@webrtc.org
c04a97f054 Move from BaseSession::GetStats to WebRtcSession::GetTransportStats 
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

Review URL: https://webrtc-codereview.appspot.com/45639004

Cr-Commit-Position: refs/heads/master@{#8739}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 19:32:23 +00:00
magjed@webrtc.org
2056ee3e3c Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*." 
This reverts commit r8731.

Reason for revert: Breakes Chromium FYI bots.

TBR=hbos, tommi

Review URL: https://webrtc-codereview.appspot.com/40359004

Cr-Commit-Position: refs/heads/master@{#8733}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 13:48:18 +00:00
hbos@webrtc.org
93d9d6503e I420VideoFrame.CreateFrame: Removed unnecessary buffer size arguments. 
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45629004

Cr-Commit-Position: refs/heads/master@{#8732}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 13:26:41 +00:00
hbos@webrtc.org
2dc5fa69b2 Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*. 
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40299004

Cr-Commit-Position: refs/heads/master@{#8731}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 13:02:19 +00:00
tommi@webrtc.org
4b89aa03bb Change StatsCollector to use DCHECK instead of ASSERT. 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46579004

Cr-Commit-Position: refs/heads/master@{#8729}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8729 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 09:52:41 +00:00
kjellander@webrtc.org
eed2fcaa76 Roll chromium_revision 00e438c..8d51d96 (320241:320682) 
Relevant changes:
* src/third_party/android_tools: fd5a8ec..98a4345
Details: 00e438c..8d51d96/DEPS

This required updating our Android projects to API level 22,
as third_party/android_tools dropped support for API level 21.

Command used:
perl -pi -e "s/android-21/android-22/g" `find . -name project.properties`
Using 'android update project' would also work but that changes the
ANDROID_SDK_ROOT -> ANDROID_HOME, which the Chromium build toolchain
doesn't set properly when build/android/envsetup.sh is sourced.

Clang version was not updated in this roll.

R=henrika@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42779004

Cr-Commit-Position: refs/heads/master@{#8728}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8728 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 09:00:41 +00:00
changbin.shao@webrtc.org
2d25b44f47 Check associated payload type when negotiate RTX codecs. 
At the moment, only payload name is checked when match two RTX codecs.
This will cause wrong behavior of codec negotiation if multiple RTX codecs
are added.

BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34189004

Cr-Commit-Position: refs/heads/master@{#8727}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8727 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-16 04:15:23 +00:00
tommi@webrtc.org
c29f7f3a5f Disable assert for nr of threads in PeerConnectionTest.java. 
This test is flaky so we need to figure out a better way to do it.
I've documented what we've observed and added a todo for myself to figure out a solution.

R=kjellander@webrtc.org
BUG=4424

Review URL: https://webrtc-codereview.appspot.com/46599004

Cr-Commit-Position: refs/heads/master@{#8725}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8725 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-14 18:15:47 +00:00
jiayl@webrtc.org
d83f4eff84 Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns. 
BUG=crbug/464995
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8689

Committed: https://code.google.com/p/webrtc/source/detail?r=8701

Committed: https://code.google.com/p/webrtc/source/detail?r=8706

Review URL: https://webrtc-codereview.appspot.com/42659004

Cr-Commit-Position: refs/heads/master@{#8722}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8722 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 21:26:44 +00:00
pthatcher@webrtc.org
b01c707209 Use a NULL session in unit tests that don't actually use the session. 
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=decurtis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49379004

Cr-Commit-Position: refs/heads/master@{#8721}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8721 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 20:05:46 +00:00
pthatcher@webrtc.org
b4aac13810 Cleanup SocketMonitor a little so that it can handle a change in transport channel. And cleanup some names and style and such as well. 
This is a part of the big BUNDLE implementation at https://webrtc-codereview.appspot.com/45519004/

R=guoweis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49399004

Cr-Commit-Position: refs/heads/master@{#8720}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8720 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 18:25:54 +00:00
jiayl@webrtc.org
8372888b07 Revert "Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns." 
This reverts commit 45bc01a7172402aa4bb8d457474300533c273413.
TBR=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/47559004

Cr-Commit-Position: refs/heads/master@{#8711}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8711 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 00:32:43 +00:00
jiayl@webrtc.org
fde1de93f9 Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns. 
BUG=crbug/464995
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8689

Committed: https://code.google.com/p/webrtc/source/detail?r=8701

Review URL: https://webrtc-codereview.appspot.com/42659004

Cr-Commit-Position: refs/heads/master@{#8706}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 23:02:13 +00:00
guoweis@webrtc.org
00c509ad1c Add concept of whether video renderer supports rotation. 
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.

Tested with peerconnection_client on windows, AppRTCDemo on Mac.

BUG=4145
R=glaznev@webrtc.org, pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8660

Committed: https://code.google.com/p/webrtc/source/detail?r=8661

Review URL: https://webrtc-codereview.appspot.com/43569004

Cr-Commit-Position: refs/heads/master@{#8705}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8705 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 21:38:19 +00:00
jiayl@webrtc.org
04cd69887d Revert "Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns." 
This reverts commit 93604daf0ea1fea1148ce07793bfa538d177c876.

TBR=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/40329004

Cr-Commit-Position: refs/heads/master@{#8704}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8704 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 21:36:42 +00:00
guoweis@webrtc.org
4f85288e71 Socket options are only applied when first setting TransportChannelImpl. 
Also fixed the issue when we have an TransportChannelImpl, the socket
option is not preserved.

Since this is a code path that will be modified by bundle (which Peter also has a test case already), we don't need a test case here.

BUG=4374
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42699004

Cr-Commit-Position: refs/heads/master@{#8702}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8702 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 20:10:22 +00:00
jiayl@webrtc.org
93604daf0e Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns. 
BUG=crbug/464995
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8689

Review URL: https://webrtc-codereview.appspot.com/42659004

Cr-Commit-Position: refs/heads/master@{#8701}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8701 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 19:05:22 +00:00
tommi@webrtc.org
d3900296ae Use a variant for storing stats values in StatsCollector code. 
This cuts down on the amount of string copying we currently do and paves the way for separating the code that fetches the stats from the code that populates the stats reports.  As is, that code is intertwined, so we populate the stats on both signaling and worker thread.

I'm also adding some documentation and TODOs for further improvements.

BUG=2822
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47459004

Cr-Commit-Position: refs/heads/master@{#8700}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8700 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 16:36:15 +00:00
tommi@webrtc.org
75b7f17c29 Temporary interface change to StatsReport::Id. 
This change is just to allow rolling into Chromium, update Chromium and then commit the actual change in WebRTC that requires the interface change. It allows using a StatsReport::Id object as a pointer (foo->Bar()), since in an upcoming change, Id objects will be pointers.

R=magjed@webrtc.org
BUG=2822

Review URL: https://webrtc-codereview.appspot.com/43689004

Cr-Commit-Position: refs/heads/master@{#8697}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8697 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 13:31:24 +00:00
mflodman@webrtc.org
e9413c686e Revert 8689 "Fix an issue in DtlsIdentityStore when the store is..." 
> Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns.
> 
> BUG=crbug/464995
> R=pthatcher@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/42659004

TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42729004

Cr-Commit-Position: refs/heads/master@{#8690}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8690 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-12 08:42:18 +00:00
jiayl@webrtc.org
2a3942adc6 Fix an issue in DtlsIdentityStore when the store is destroyed before the worker thread task returns. 
BUG=crbug/464995
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42659004

Cr-Commit-Position: refs/heads/master@{#8689}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8689 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-11 23:33:42 +00:00
glaznev@webrtc.org
fc516077ed Fix Android AppRTCDemo failure on devices with one or no camera. 
- Disable video call on devices with no camera.
- Open default camera and disable camera switch on
devices with one camera.

BUG=4373
R=braveyao@webrtc.org, wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46539004

Cr-Commit-Position: refs/heads/master@{#8674}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-10 18:21:19 +00:00
guoweis@webrtc.org
f9a75d99b9 Revert "Add concept of whether video renderer supports rotation." 
This reverts commit 0ad48935fc5b92be6e10924a9ee3b0dc39c79104.

TBR=guoweis@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/41199004

Cr-Commit-Position: refs/heads/master@{#8663}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-10 06:37:41 +00:00
guoweis@webrtc.org
0ad48935fc Add concept of whether video renderer supports rotation. 
Rotation is best done when rendered in GPU, added the shader code which rotates the frame. For renderers which don't support rotation, the rotation will be done before sending down the frame to render. By default, assume renderer can't do rotation.

BUG=4145
R=glaznev@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43569004

Cr-Commit-Position: refs/heads/master@{#8660}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8660 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-10 02:43:40 +00:00
henrika@webrtc.org
474d1eb223 Adds C++/JNI/Java unit test for audio device module on Android. 
This CL adds support for unittests of the AudioDeviceModule on Android using both Java and C++. The new framework uses ::testing::TesWithParam to support both Java-based audio and OpenSL ES based audio. However, given existing issues in our OpenSL ES implementation, the list of test parameters only contains Java in this first version. Open SL ES will be enabled as soon as the backend has been refactored.

It also:

- Removes the redundant JNIEnv* argument in webrtc::VoiceEngine::SetAndroidObjects().
- Modifies usage of enable_android_opensl and the WEBRTC_ANDROID_OPENSLES define.
- Adds kAndroidJavaAudio and kAndroidOpenSLESAudio to AudioLayer enumerator.
- Fixes some bugs which were discovered when running the tests.

BUG=NONE
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40069004

Cr-Commit-Position: refs/heads/master@{#8651}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8651 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-09 12:40:43 +00:00
glaznev@webrtc.org
dc08a230da Fix H.264 start code position search. 
This will address incorrect start code search
in a sequence like 00 00 00 00 00 01.
Thanks Noah.

R=noahric@chromium.org, wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41159004

Cr-Commit-Position: refs/heads/master@{#8639}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8639 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-06 23:32:42 +00:00
magjed@webrtc.org
c2008a0e8c RTCOpenGLVideoRenderer: Add support for padded frames 
This CL allows RTCOpenGLVideoRenderer to handle frames with pitch > width by making an intermediate frame copy.

BUG=4381,1128
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46509004

Cr-Commit-Position: refs/heads/master@{#8637}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8637 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-06 16:59:23 +00:00
jiayl@webrtc.org
b4cd093f41 Change the unintentioal CHECK to DCHECK in DtlsIdentityStore. 
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41139004

Cr-Commit-Position: refs/heads/master@{#8636}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8636 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-06 16:32:57 +00:00
magjed@webrtc.org
45cdcce5f5 Remove TextureVideoFrame 
TextureVideoFrame is currently an empty shell that only provides a convenience constructor of I420VideoFrame with a texture buffer. This CL moves that constructor, and all unittests, of TextureVideoFrame into the base class. Then it's possible to completely remove TextureVideoFrame and all its files. Also, there is no point in having I420VideoFrame virtual anymore.

R=pbos@webrtc.org, perkj@webrtc.org, stefan@webrtc.org
TBR=mflodman

Review URL: https://webrtc-codereview.appspot.com/40229004

Cr-Commit-Position: refs/heads/master@{#8629}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8629 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-06 10:41:47 +00:00
tommi@webrtc.org
5af41aabae Fix uninitialized variable. If FindConstraint() returns false, we check |value| in two places and at that point, it can hold an uninitialized value. Caught by Linux Memcheck builder. 
http://chromegw.corp.google.com/i/client.webrtc/builders/Linux%20Memcheck/builds/3351/steps/libjingle_peerconnection_unittest/logs/0A34BA777AB03D08

TBR=perkj@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/43579004

Cr-Commit-Position: refs/heads/master@{#8611}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8611 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-05 08:42:13 +00:00
guoweis@webrtc.org
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
guoweis@webrtc.org
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
guoweis@webrtc.org
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
jiayl@webrtc.org
61e00b0bca Create a in-memory DTLS identity store that keeps a free identity generated in the background. 
BUG=4241
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8576

Committed: https://code.google.com/p/webrtc/source/detail?r=8581

Review URL: https://webrtc-codereview.appspot.com/37889004

Cr-Commit-Position: refs/heads/master@{#8605}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8605 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-04 22:18:18 +00:00
tommi@webrtc.org
92f4018d80 Start using std::map for Values in the statscollector. This is in preparaton for more work which will cut down on the string copying work we do. 
Rename "AddValue" methods to AddXxx where Xxx is the type being added. Moving forward, we'll support those types natively without conversion to string.

Normalizing the extraction code to have fewer places that add the same stats and data driven additions to reports instead of multiple call sites.

BUG=2822
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47369004

Cr-Commit-Position: refs/heads/master@{#8597}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8597 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-04 15:25:44 +00:00
kjellander@webrtc.org
14665ff7d4 Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro 
Clang version changed 223108:230914
Details: e144d30..6fdb142/tools/clang/scripts/update.sh

Removes the OVERRIDE macro defined in:
* webrtc/base/common.h
* webrtc/typedefs.h

The majority of the source changes were done by running this in src/:
perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h"  -o -name "*.cc*" -o -name "*.mm*"`

which converted all:
virtual Foo() OVERRIDE
functions to:
Foo() override

Then I manually edited:
* talk/media/webrtc/fakewebrtccommon.h
* webrtc/test/fake_common.h

Remaining uses of OVERRIDE was fixed by search+replace.

Manual edits were done to fix virtual destructors that were
overriding inherited ones.

Finally a build error related to the pure virtual definitions of
Read, Write and Rewind in common_types.h required a bit of
refactoring in:
* webrtc/common_types.cc
* webrtc/common_types.h
* webrtc/system_wrappers/interface/file_wrapper.h
* webrtc/system_wrappers/source/file_impl.cc

This roll should make it possible for us to finally re-enable deadlock
detection for TSan on the buildbots.

BUG=4106
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41069004

Cr-Commit-Position: refs/heads/master@{#8596}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-04 13:04:54 +00:00
pbos@webrtc.org
058b1f17ac Remove GetReceiveBandwidthEstimatorStats. 
Removes unnecessary non-standard stats that we don't really make use of.

BUG=
R=pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47379004

Cr-Commit-Position: refs/heads/master@{#8588}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8588 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-04 08:55:16 +00:00
guoweis@webrtc.org
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
pthatcher@webrtc.org
7bea1ffe77 Expose negotiated ciphers through stats API. 
Use the new internal API to expose the negotiated SRTP/SSL ciphers
through the stats API.
This is a follow-up to https://webrtc-codereview.appspot.com/37209004.

BUG=3976
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35169004

Cr-Commit-Position: refs/heads/master@{#8584}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8584 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-04 01:38:49 +00:00
jiayl@webrtc.org
be77872d2c Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background." 
Breaking Chromium FYI.

TBR=pthatcher@webrtc.org

This reverts commit 369f68255ffd3d6f3e449e0defeae820cefd4f29.

BUG=4241
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8576

Review URL: https://webrtc-codereview.appspot.com/37889004


Review URL: https://webrtc-codereview.appspot.com/47389004

Cr-Commit-Position: refs/heads/master@{#8583}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-04 00:19:16 +00:00