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Author SHA1 Message Date
turaj@webrtc.org
d6a7a5f385 Small fixes to run ACM2 tests. 
BUG=
R=minyue@google.com

Review URL: https://webrtc-codereview.appspot.com/2238004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-25 01:09:23 +00:00
turaj@webrtc.org
ff43c85ef1 API add to set background noise mode. 
Background noise mode.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2194005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-25 00:07:27 +00:00
sergeyu@chromium.org
8d757ac0a2 Fix window capturer not to leak HDC. 
Previously Windows window capturer kept DC handles between captures. As
result it was leaking DC handles in SelectWindow(). Fixed it so that it
calls GetWindowDC() for each capture.

R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4834 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-24 23:13:51 +00:00
sergeyu@chromium.org
958cdf68f3 Fix window capturer to stop capturing when the target is minimized. 
BUG=crbug.com/288205
R=alexeypa@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2273005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4833 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-24 22:10:13 +00:00
andrew@webrtc.org
f832a551cc Disable flaky TestPartialFrameHeader. 
Example failure:
[http://chromegw/i/internal.client.webrtc/builders/Linux%20Asan/builds/657]

TBR=wu@webrtc.org
BUG=2409

Review URL: https://webrtc-codereview.appspot.com/2286004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4832 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-24 20:09:30 +00:00
andrew@webrtc.org
c2ac5ed588 Add more TSAN suppressions for libjingle_media_unittest. 
BUG=2078
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2282004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4830 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-24 18:45:19 +00:00
andrew@webrtc.org
641587f938 Disable some VP8 tests on Android. 
DecodeWithACompleteKeyFrame and FixedTemporalLayersStrategy.

TBR=andresp
BUG=2037

Review URL: https://webrtc-codereview.appspot.com/2283004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4829 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-24 18:43:28 +00:00
andrew@webrtc.org
f0f92fae12 Disable flaky SendDataMultipleClocks. 
Example failure:
[http://chromegw/i/internal.client.webrtc/builders/Linux32%20Debug/builds/719]

TBR=mallinath
BUG=2409

Review URL: https://webrtc-codereview.appspot.com/2270005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4828 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-24 16:26:41 +00:00
henrika@webrtc.org
9b6eefcedf Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket 
Ensures that we always call DeRegisterExternalTransport() even if a fuzz test calls DeRegisterExternalTransport in an unintialized state.

TBR=tommi
BUG=296804 in crbug.com

Review URL: https://webrtc-codereview.appspot.com/2275008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4827 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-24 15:42:49 +00:00
kjellander@webrtc.org
ebd3ab0ce1 Add libjingle_peerconnection_objc_test to buildbot_tests.py 
To run this test, the executable in
libjingle_peerconnection_objc_test.app/Contents/MacOS/libjingle_peerconnection_objc_test
must be run. Running executables like this is not supported by
the buildbot scripts, which makes the paths to the stdio invalid.
Adding it to our command-line wrapping script and executing it as
'libjingle_peerconnection_objc_test' on the bots should work around
this limitation.

BUG=2120
TEST=locally building and executing the test on Mac using:
GYP_DEFINES="host_arch=x64 target_arch=x64" gclient runhooks
ninja -C out/Release
out/Release/buildbot_tests.py -t libjingle_peerconnection_objc_test

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2275007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4826 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-24 13:41:16 +00:00
kjellander@webrtc.org
6e86349273 Disable tests that crash the OS X kernel when run under memcheck. 
These libjingle tests crashes the OS X kernel when run under
memcheck on Mac 10.6. I didn't file bugs for them since it's unlikely
we can fix this anyway. There are several other tests disabled
in the libjingle code with similar comments, without bugs
assigned to them.
See talk/base/physicalsocketserver_unittest.cc for examples.

Affected waterfall: http://build.chromium.org/p/client.webrtc.fyi/waterfall

TEST=none
BUG=none
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2278006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4825 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-24 12:34:51 +00:00
stefan@webrtc.org
b0e6eb50b5 Revert r4823 "Reenable test and remove flaky expects." 
TBR=mflodman@webrtc.org

BUG=2415

Review URL: https://webrtc-codereview.appspot.com/2277005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4824 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-24 10:38:57 +00:00
stefan@webrtc.org
01aad09a01 Reenable test and remove flaky expects. 
BUG=2415
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2278005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4823 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-24 10:16:52 +00:00
henrik.lundin@webrtc.org
b426c469b9 MediaOptimization: Converting a few members to scoped_ptrs 
For consistency with other parts of the code.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2275006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4822 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-24 07:41:53 +00:00
andrew@webrtc.org
6ffc74ee0e Disable flaky RunsRtpRtcpTestWithoutErrors. 
TBR=mflodman
BUG=2415

Review URL: https://webrtc-codereview.appspot.com/2270006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4821 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 23:25:39 +00:00
andrew@webrtc.org
eb524d997b Remove deprecated AudioCodingModule::Destroy. 
Have Channel hold a pointer rather than reference, and shorten the name.

TESTED=trybots
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2256004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4820 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 23:02:24 +00:00
mallinath@webrtc.org
1112c30e1e Update libjingle to 53057474. 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2274004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4818 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 20:34:45 +00:00
asapersson@webrtc.org
e2af622edf - Reset capture deltas at resolution change. 
- Applied smoothing of capture jitter.
- Adjusted thresholds.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2070005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4817 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 20:05:39 +00:00
henrik.lundin@webrtc.org
bec11ef632 Reformatting media_optimization.cc and .h 
Ran both tools/refactoring/webrtc_reformat.py and clang-format.
Changing VCMMediaOptimization -> MediaOptimization and
VCMEncodedFrameSample -> EncodedFrameSample.
Aligning the order of methods in .h and .cc files and fixing comments.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2265007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4816 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 19:54:25 +00:00
asapersson@webrtc.org
b533a82bf9 Disabled flaky tests. 
BUG=2409
R=andrew@webrtc.org, mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2267005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4815 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 19:47:49 +00:00
vikasmarwaha@webrtc.org
7a7b929882 Updated dc1.html to support SCTP transport. 
R=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/2058004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 18:03:33 +00:00
fischman@webrtc.org
334865e2a1 Re-enable VideoCaptureTest.CreateDelete 
Previously the test insisted on non-zero delay, but 0 is not a crazy delay value
(esp. on a fake camera device!).  Instead we now test for delay>=0 being set at
all.

BUG=2405
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2267004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 17:58:45 +00:00
elham@webrtc.org
038e8e64ef Updated WebRTC version to 3.42 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2271004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4811 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 16:45:25 +00:00
andresp@webrtc.org
98fcd2d4c3 Adding unit tests for default temporal layer strategy. 
R=mflodman@webrtc.org, stefan@webrtc.org, stefan

Review URL: https://webrtc-codereview.appspot.com/2235005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4810 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 11:12:59 +00:00
stefan@webrtc.org
cdd3d4d139 Revert test change in r4808. 
This was supposed to be an EXPECT_GT, I just misunderstood it in the previous CL. Added a sleep after the EXPECT_GT and before bytes_received_after = bytes_received_before.

BUG=1790
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2265006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4809 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 09:43:07 +00:00
stefan@webrtc.org
269dd4264f Reduce flakiness in network down test. 
The encoder is in the process of encoding when the network goes down, so we need to wait until it has finished before we expect no more packets to be sent.

Also fixed a test which was testing the wrong thing.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2258008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4808 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 08:42:39 +00:00
dwkang@webrtc.org
63fe8e1f38 Enable SetInitialPlayoutDelay on Android. 
Background:
In Chrome mirroring which uses 500ms buffering mode,
audio video mismatch happens in the begining because of the lack of the api.

BUG=b/10538425
TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release  -f *InitialPlayoutDelayTest*'
R=henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2177004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-23 05:42:22 +00:00
sergeyu@chromium.org
2edb642810 Fix bugs in DesktopRegion::Subtract(). 
Fixed a bug that caused the crash in the linked bug and also couple of
other issues in the same code. Also added more tests.

BUG=crbug.com/295057
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-20 21:29:18 +00:00
vikasmarwaha@webrtc.org
cee0dfb57a Made sure that DTLS/SRTP is set to false in apprtc demo when testing loopback. See crbug/294881 for details. 
R=juberti@google.com, mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2268004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-20 21:26:07 +00:00
turaj@webrtc.org
10e6cc7e23 VAD changes ported to ACM2. 
This CL ports the relevant parts of  https://code.google.com/p/webrtc/source/detail?r=4625 to ACM2.

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2264004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4804 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-20 16:38:26 +00:00
turaj@webrtc.org
362a55e7b0 Address Windows 64-bits warnings. 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2203004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4803 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-20 16:25:28 +00:00
pbos@webrtc.org
0e63e76781 Enable FEC for VideoSendStream. 
Test only checks for FEC without NACK. Test for FEC with NACK postponed
until later.

BUG=2230
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2246004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-20 11:56:26 +00:00
stefan@webrtc.org
9c74be7bd1 Disable flaky video capture test. 
BUG=2405
TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2265005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-20 08:52:31 +00:00
stefan@webrtc.org
4f3624d39e Avoid recursively taking critical section. 
TEST=trybots
BUG=2261
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2258006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4800 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-20 07:43:17 +00:00
jiayl@webrtc.org
dd57cd6ed5 Removing the tsan text exclusion since the tests should be passing now. 
BUG=2299, 2290, 2291
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2260004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-20 00:51:53 +00:00
fischman@webrtc.org
d29ab4e17c Suppress SSL error strings on mac_asan to unbreak that build 
Example borkedness: http://chromegw/i/client.webrtc/builders/Mac%20Asan/builds/642/steps/libjingle_p2p_unittest/logs/stdio

R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/2263004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4798 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 23:54:56 +00:00
fischman@webrtc.org
76fe9309b9 Use link_settings instead of all_dependent_settings to pacify xcode gyp generator 
Should unbreak e.g. http://chromegw/i/chromium.webrtc.fyi/builders/Mac%20%5Blatest%20WebRTC%20trunk%5D/builds/2396

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2261004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4796 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 21:11:08 +00:00
fischman@webrtc.org
ccddd0a941 Roll webrtc's chromium_revision 217707:224141 
Also adds -lm for executables depending on isac since the newer clang in the
newer chromium revision requires it, and -lstdc++ for dependencies of the objc lib because newer gyp links with gcc instead of g++ for non-C++-containing libs.

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2177007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 20:27:32 +00:00
pbos@webrtc.org
6917e19ad4 Rename EngineTest to CallTest. 
There's no real notion of VideoEngine left in these classes. They're
end-to-end tests built on Call, so CallTest makes more sense.

This also contains a modification to RtpRtcpObserver moving the
responsibility of creating the event that signals when the observation
is complete to RtpRtcpObserver. New tests are about to be introduced and
this will reduce code duplication.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2258005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4793 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 14:22:12 +00:00
tina.legrand@webrtc.org
a03e34e9ab Heap-use-after-free in WebRtcNetEQ_RecInRTPStruct 
Pointer to released memory was not set to NULL, which means
you could get a heap-us-after-free in the code. It happens if one of the slaves of NetEq is deleted, but we keep trying to decode packets.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4792 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 13:32:14 +00:00
andresp@webrtc.org
ab6549562b Refactor frame generation code so it can be used by multiple modules. 
R=pbos@webrtc.org, stefan@webrtc.org, pbos, stefan
BUG=

Review URL: https://webrtc-codereview.appspot.com/2240004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4791 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 12:14:03 +00:00
stefan@webrtc.org
7a30dfdc69 Disable NACK bandwidth statistics test due to being too flaky. 
Tests for new API currently provide partial coverage, and will soon
provide full coverage.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2151005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 12:08:55 +00:00
stefan@webrtc.org
b5a191bfe7 Fixes a flake in network down tests. 
And reduces the flakiness in NACK tests.

TESTS=trybots
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2258004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 11:14:35 +00:00
kjellander@webrtc.org
d8a9b86671 Disable tests for TSan v2 
These are tests that fail and that I haven't been able to suppress.
I assume they're caused by the same bug in TSan v2 as described in
webrtc:2259.

BUG=2259,2334
TEST=Ran the tests similar to the instructions in the bugs, passed 100 iterations.
R=andrew@webrtc.org, hta@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2250004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4787 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 07:15:36 +00:00
wu@webrtc.org
967bfff54d Update talk to 52534915. 
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2251004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4786 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 05:49:50 +00:00
turaj@webrtc.org
532f3dc548 Compile ACM2 and ACM1. 
First patch set is the same as patch set 3 of http://review.webrtc.org/2237004/

-Make ACM1 to depend on ACM2.
-Remove APIs to set and get background noise mode. There is no VoE call to these
APIs.
-Remove APIs to set and get receive side VAD mode. There is no VoE call to these
APIs, and NetEq 4, doesn't support them.
-Remove callback for in-band DTMF detection. ACM doesn't support in-band DTMF
detection.
-Use acm_common_defs.h everywhere required.
-Complete ACM factory method.
-Update ACMCodecDatabase of ACM2. CNG full-band need to be define-guarded.
Remove dynamic payload-type assignment.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4785 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-19 00:12:23 +00:00
andrew@webrtc.org
f3930e941c Small refactoring of AudioProcessing use in channel.cc. 
- Apply consistent naming.
- Use a scoped_ptr for rx_audioproc_.
- Remove now unnecessary AudioProcessing::Destroy().

R=bjornv@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2184007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4784 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-18 22:37:32 +00:00
henrik.lundin@webrtc.org
0d5da25e6c NetEq4: Making a few more members scoped_ptrs 
This CL converts a few members in NetEqImpl form regular pointers
to scoped_ptrs.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2245004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4783 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-18 21:12:38 +00:00
henrik.lundin@webrtc.org
5a43370cdb Dedicated speed test for NetEq3 
This is the same test as was aleready implemented for NetEq3 in r4763.

BUG=1363
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2234004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4782 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-18 20:58:33 +00:00
kjellander@webrtc.org
7a968a8f07 Add more TSan and Dr Memory suppressions for modules_unittests 
I'm trying to get these tests green on Windows in
http://build.chromium.org/p/client.webrtc.fyi

BUG=2319,2323
TEST=local runs passing
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2230004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-18 19:18:20 +00:00