stefan@webrtc.org
fdcb42dac4
Fix potential crash when depacketizing VP8.
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Caused by a missing check for H264 when reading the RTPVideoTypeHeader union.
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6825 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-05 13:21:18 +00:00
buildbot@webrtc.org
8e885990ae
(Auto)update libjingle 72566057-> 72591796
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6824 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04 23:56:14 +00:00
henrike@webrtc.org
d6542852f3
Unbreaks linux.cc in Chromium.
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BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6823 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04 21:51:14 +00:00
jiayl@webrtc.org
b18bf5e47d
Adds the support of RTCOfferOptions for PeerConnectionInterface::CreateOffer.
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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
fbarchard@google.com
b01ce14b13
add some comments about DEPS lkgr for chromium
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BUG=none
TESTED=none
R=harryjin@google.com
Review URL: https://webrtc-codereview.appspot.com/16209005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6821 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04 18:07:19 +00:00
henrike@webrtc.org
c9b507253f
DrMemory suppression due to r6811.
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BUG=3655
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6820 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04 16:48:24 +00:00
henrike@webrtc.org
ee135f78b7
Memcheck suppression. Re-suppress 3478 suppression after namespace change from talk_base to rtc.
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BUG=3478
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6819 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04 15:35:14 +00:00
buildbot@webrtc.org
a27342b7af
(Auto)update libjingle 72446860-> 72550257
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6818 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04 15:22:32 +00:00
minyue@webrtc.org
0040a6ef97
This is a setup to solve
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https://code.google.com/p/webrtc/issues/detail?id=1906
In particular, we add an API to call Opus's set maximum bandwidth to prevent the encoder from coding audio content beyond this bandwidth so as to increase computation and transmission efficiency (without affecting sampling rate).
BUG=
R=henrik.lundin@webrtc.org , turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6817 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04 14:41:57 +00:00
asapersson@webrtc.org
84b9e1e9d9
Fix for retransmission. Base layer packets were not retransmitted.
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Issue introduced in r6669.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6816 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04 11:59:42 +00:00
buildbot@webrtc.org
e0d03f13e4
(Auto)update libjingle 72443101-> 72446860
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6815 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-02 03:04:01 +00:00
buildbot@webrtc.org
6e203d50a3
(Auto)update libjingle 72442050-> 72443101
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6814 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-02 01:13:04 +00:00
buildbot@webrtc.org
52148c2f74
(Auto)update libjingle 72430895-> 72442050
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6813 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-02 00:56:56 +00:00
buildbot@webrtc.org
7cb60ccae1
(Auto)update libjingle 72407428-> 72430895
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6812 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-01 22:03:36 +00:00
buildbot@webrtc.org
3bc48247b7
(Auto)update libjingle 72403605-> 72407428
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6811 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-01 16:53:32 +00:00
buildbot@webrtc.org
6955213eca
(Auto)update libjingle 72389720-> 72403605
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6810 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-01 15:52:45 +00:00
solenberg@webrtc.org
42d65ce8d7
Fix memory leak in FakeSSLCertificate::GetChain(), discovered by Linux Memcheck build/try bots.
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TBR=hellner
BUG=
Review URL: https://webrtc-codereview.appspot.com/18969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6809 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-01 10:07:46 +00:00
buildbot@webrtc.org
1a678c61f1
(Auto)update libjingle 72320533-> 72380285
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6808 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-01 06:21:50 +00:00
buildbot@webrtc.org
6b21b71068
(Auto)update libjingle 72205295-> 72320533
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6806 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-31 15:08:53 +00:00
stefan@webrtc.org
e1c9caf6ee
Fix mistake in rtp/rtcp/BUILD.gn introduced with r6804.
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TEST=buildtools/linux64/gn gen out/Default
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6805 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-31 15:07:59 +00:00
stefan@webrtc.org
2ec560606b
Add H.264 packetization.
...
This also includes:
- Creating new packetizer and depacketizer interfaces.
- Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition.
- Created a Create() factory method for packetizers and depacketizers.
R=niklas.enbom@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-31 14:59:24 +00:00
stefan@webrtc.org
bfe6e08195
Add simulation of network effects to video_loopback tool.
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Also add support for uniform random packet loss to the fake network pipe.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14039004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6803 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-31 12:30:18 +00:00
henrike@webrtc.org
d9843da9ee
libjingle: stop building files in talk/base as they are no longer used as of r6799
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BUG=3379
R=thorcarpenter@google.com
Review URL: https://webrtc-codereview.appspot.com/16189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6802 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-30 04:00:52 +00:00
fbarchard@google.com
48305f5f4c
Disable warning 4702 which affects map, xlist and others on vs2012 and vs2013.
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BUG=3584
TESTED=python webrtc\build\gyp_webrtc -G msvs_version=2013 & ninja -C out\Release
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6801 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-30 00:16:20 +00:00
fbarchard@google.com
901debdad3
roll libyuv to r1038 from r1035 to add gyp define that makes jpeg optional.
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BUG=libyuv:346
TESTED=set GYP_DEFINES=target_arch=ia32 libyuv_disable_jpeg=1
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6800 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 18:07:07 +00:00
buildbot@webrtc.org
d4e598d57a
(Auto)update libjingle 72097588-> 72159069
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 17:36:52 +00:00
solenberg@webrtc.org
fc8b0871d9
Remove dependency on openssl for android, add dependency on boringssl. Should make Android bots green again.
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TBR=hellner
Review URL: https://webrtc-codereview.appspot.com/21079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6798 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 15:23:59 +00:00
andrew@webrtc.org
fdbe1442c5
Use C functions in aec for MIPS
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With GCC 4.9, the MIPS NDK toolchain has been changed to only support 16 spregs by default - the even-numbered ones. This has been changed to support the R6 MIPS architecture. While the old behaviour could be restored by adding "-modd-spreg", this would come with a performance hit because the kernel would emulate odd-numbered spregs and missing R2 instructions.
As a result of this change, the functions removed in this CL no longer compile as there are no longer enough spregs for the compiler to assign. So we are removing these functions and they will use the C implementation until the MIPS code is rewritten.
R=andrew@webrtc.org , ljubomir.papuga@gmail.com , pasko@chromium.org
Review URL: https://webrtc-codereview.appspot.com/16159005
Patch from Fabrice de Gans-Riberi <fdegans@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6797 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 14:39:10 +00:00
asapersson@webrtc.org
e75d78d32d
Integrate rtcp packet class to rtcp receiver tests.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6795 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 08:21:50 +00:00
henrike@webrtc.org
1e7d60e451
merge_libs.py: fixes Windows breakage: there should be no space after "lib /OUT:".
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TBR=andrew@webrtc.org
BUG=b/15773179
Review URL: https://webrtc-codereview.appspot.com/16999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6793 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 04:45:23 +00:00
buildbot@webrtc.org
51c5508bf1
(Auto)update libjingle 72016417-> 72097588
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6792 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-28 22:26:15 +00:00
pbos@webrtc.org
8aed945842
Remove a disabled test.
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ConstrainsSetCodecsAccordingToEncoderConfig has been removed from
webrtcvideoengine_unittest.cc, removing this one as well.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6789 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-26 10:16:49 +00:00
pbos@webrtc.org
4fe98a9124
Remove clang-format rm_binaries.py DEPS entry.
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Breaks runhooks.
BUG=
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6788 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-25 23:26:09 +00:00
henrike@webrtc.org
961293d469
webrtc/base: FileModifyTime -> OlderThan as that's what it was ever used as. Needed for cl/70828325.
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BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6787 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-25 21:58:50 +00:00
sergeyu@chromium.org
af9e7943d1
Fix compilation on windows with clang, indentation cleanups
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R=henrike@webrtc.org , thakis@chromium.org
TBR=hellner@chromium.org
Committed: https://code.google.com/p/webrtc/source/detail?r=6779
Review URL: https://webrtc-codereview.appspot.com/18849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6786 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-25 19:42:19 +00:00
pbos@webrtc.org
257e130a16
Set NACK/REMB when setting receive codecs.
...
Enabling an additional test to ensure that REMB can be both enabled and
disabled by setting VideoCodecs.
BUG=1788
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6785 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-25 19:01:32 +00:00
fgalligan@google.com
3155f2b65d
Roll chromium 282879:285412.
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Pick up the libvpx roll:
https://codereview.chromium.org/401983003/
R=marpan@google.com
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21959005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6784 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-25 18:58:26 +00:00
henrike@webrtc.org
185636cf70
Revert of 6778 "Refactor StatsCollector and associated types."
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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
henrike@webrtc.org
c7b8f39e2e
Fixes "argument list too long" problem on Linux by using the "find" command instead of re-implementing one in python.
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R=pbos@webrtc.org
TBR=andrew@webrtc.org
BUG=b/15773179
Review URL: https://webrtc-codereview.appspot.com/18929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6782 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-25 18:36:55 +00:00
turaj@webrtc.org
1ebd2e96df
Remove timestamp retreival warning/error.
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An error reported while retreiving playout timestamp if no RTP packet received, yet. This causes an overflow of errors/warnings in applications where few channel are created but only one is actively engaged in a conversation. Therefore, we don't find such logging informative (there is no check upon correctness of timestamp computaion only if a packet already received).
BUG=3545
TEST=manual with voe_cmd_test,try bots
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6781 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-25 17:50:10 +00:00
sergeyu@chromium.org
2386882266
Revert "Fix compilation on windows with clang, indentation cleanups"
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This reverts commit f628eaedfeea97e13c63c78dd42f2b1c76723619.
TBR=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/13069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6780 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-25 17:37:12 +00:00
sergeyu@chromium.org
a44fce5920
Fix compilation on windows with clang, indentation cleanups
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R=henrike@webrtc.org , thakis@chromium.org
TBR=hellner@chromium.org
Review URL: https://webrtc-codereview.appspot.com/18849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6779 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-25 17:28:25 +00:00
tommi@webrtc.org
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
jiayl@webrtc.org
06b04ec4ab
Fix a crash in statscollector.cc caused by invoking methods on the worker thread which destroys the Transport.
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BUG=3579
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6776 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-24 20:41:20 +00:00
mflodman@webrtc.org
f9460688a6
Make sure padding is sent on the first sending RTP module.
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R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6774 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-24 16:41:25 +00:00
buildbot@webrtc.org
45304ff0a7
(Auto)update libjingle 71829282-> 71834788
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6773 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-24 16:06:35 +00:00
henrike@webrtc.org
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
buildbot@webrtc.org
437d57db5b
(Auto)update libjingle 71775619-> 71778545
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6771 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23 21:40:28 +00:00
henrike@webrtc.org
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
henrike@webrtc.org
8721f989bf
Revert 6766 "Temporarily add a default ctor to StatsReport and make |id| non const. As soon as I've updated the chrome side, I'll revert this cl."
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BUG=N/A
TBR=ajm@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6769 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-23 21:38:09 +00:00