Commit Graph

6455 Commits

Author SHA1 Message Date
0803c03f9a Fix build/isolate.gypi path in webrtc_tests.gypi.
BUG=2535
R=kjellander@webrtc.org
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3039005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5045 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 18:10:29 +00:00
b7a171825b Drop ViEDecoderObserver::DecoderTiming impl now that WebRtcDecoderObserver rolled in r5038.
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5044 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 17:36:59 +00:00
16e03b7bd8 Separate Call API/build files from video_engine/.
BUG=2535
R=andrew@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5042 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 16:32:01 +00:00
1a3a6e5340 Removing the threshold from the auto-mute APIs
The threshold is now set equal to the minimum bitrate of the
encoder. The test is also changed to have the REMB values
depend on the minimum bitrate from the encoder.

BUG=2436
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5040 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 10:16:14 +00:00
fe5d36b6fe Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.
We will do some refactoring of video engine and would like to use the
same rtcp stats struct there. Both video and audio seem to use 8bit
fraction lost, so that is changed in the struct as well.

BUG=
R=henrik.lundin@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5039 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 09:21:07 +00:00
c94abd313e Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5036 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-25 18:15:09 +00:00
0729460acb Added a "interleaved_" flag to webrtc::AudioFrame.
And also did some format refactoring on the AudioFrame class, no change on the functionalities on those format refactoring code.

BUG=
TEST=compile
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5032 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-25 12:50:46 +00:00
b3731da68f Prefix MOVE_ONLY_TYPE_FOR_CPP_03 with WEBRTC_.
Will fix a redefinition error in Chromium against webrtc head.

TESTED=trybots
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5029 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-24 15:16:53 +00:00
b56d0e383e Change the low-bitrate handling in BitrateControllerImpl
Changing to using strategy classes rather than having two different
derived classes of BitrateControllerImpl. This enables run-time switching
of the strategy, which is now possible through a new API. The reason is
that it must fit the current design of ViE.

BUG=2436
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5028 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-24 09:24:06 +00:00
37bb4974e7 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals.
R=juberti@google.com, mikhal@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5027 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23 23:59:45 +00:00
d1bcf1180a Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined.
Works around a multiple definition error from webrtc and libjingle.

Corresponds to the libjingle change here:
https://critique.corp.google.com/#review/55489575-p10

TESTED=trybots
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5025 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23 19:11:32 +00:00
22858d4785 Add an extended filter option to audioproc.
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5024 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23 14:07:17 +00:00
042e91c2b2 Fix for incorrect RTT estimation. A too low RTT value could be estimated.
R=andrew@webrtc.org, holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5023 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23 13:58:31 +00:00
ba975e2078 Porting auto mute to new ViE API
This CL also includes tests for the auto mute function. A few minor lint
warnings were fixed too. Note that the auto mute function is still work
in progress.

The callback ViEEncoderObserver::VideoAutoMuted was not ported from the
old API. This is TBD; see issue 2457.

BUG=2436
R=holmer@google.com, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2340004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5021 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23 11:04:57 +00:00
886aef09a8 Fixing broken tests in voe_auto_test extended
This CL fixes the problem with voe_auto_test extended-codec test, as well as
extended-file test. First problem was that Opus was not added as a special case, like the other codecs, and the second problem was that the tests were not updated when test files were moved to the resources catalogue.

There are still some tests that fails. Here is a list of all extended tests and their status:

Base: fails - the reason seem to be that external transport has been removed.
CallReport: passes
Codec: passes (with this CL)
DTMF: passes
Encryption: fails or is dissabled?
VoEExternalMedia: passes
File: passes (with this CL)
Hardware: passes
NetEqStats: empty?
Network: passes
RTP_RTCP: fails
VideoSync: fails
VolumeControl: passes

BUG=issue2234
R=andrew@webrtc.org, henrika@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2023004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5020 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-23 10:39:56 +00:00
31628aae7e Upgrade scoped_ptr to Chromium's latest version.
Analogous to the recent libjingle change: http://cl/54929753-p10.
This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather
than scoped_array and scoped_ptr_malloc respectively.

- Add Chromium's template-based COMPILE_ASSERT. We didn't have this
previously in order to support the macro in C. Instead, move the
existing macro to compile_assert_c.h.
- Additionally copy the move.h and template_util.h depedencies and add
the WARN_UNUSED_RESULT macro.
- Leave scoped_array and scoped_ptr_malloc for now, but mark as
deprecated.
- Remove scoped_ptr foo(NULL) use. The default constructor handles it.
- Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc.
- Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove
some repeated code.

TESTED=trybots
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2449005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-22 12:50:00 +00:00
06b60c07b7 Roll chromium_revision 228675:229708
This will pick up the -Wunused-const-variable
Clang warning being enabled by default (chromium:307668).

BUG=none
TEST=trybots passing.
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5014 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-22 12:09:48 +00:00
621df678c8 WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN.
Mostly to remove a long-standing TODO...

TESTED=trybots
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5013 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-22 10:27:23 +00:00
943e3b95a6 Add CurrentLayerId() to temporal layers.
same patch as: https://webrtc-codereview.appspot.com/2427004/

TBR=holmer@google.com

Review URL: https://webrtc-codereview.appspot.com/2729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5012 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-22 01:55:07 +00:00
9c735c4e25 Updated WebRTC version to 3.45
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5009 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21 16:34:50 +00:00
8215106371 Framework for testing bandwidth estimation.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2317004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5008 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21 14:23:26 +00:00
29dd0de5b3 Changing the bitrate clamping in BitrateControllerImpl
This CL implements an alternative to the bitrate clamping that is done
in BitrateControllerImpl. The default behavior is unchanged, but if
the new algorithm is enabled the behavior is as follows:
When the new bitrate is lower than the sum of min bitrates, the
algorithm will give each observer up to its min bitrate, one
observer at a time, until the bitrate budget is depleted. Thus,
with this change, some observers may get less than their min bitrate,
or even zero.

Unit tests are implemented.

Also fixing two old lint warnings in the affected files.

This change is related to the auto-muter feature.

BUG=2436
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2439005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5007 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21 14:00:01 +00:00
0d19ed9a06 AutoMute: Adding channel_id parameter to callback.
BUG=2436
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2390004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5006 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21 12:37:13 +00:00
fe1ef935e7 Implement I420FrameCallbacks in Call.
BUG=2425
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2393004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5005 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21 10:34:43 +00:00
e05362916c Make sure the first frame isn't dropped.
If frames were delivered within the same millisecond as VideoCaptureImpl
was created, or the timestamp weren't granular enough then the first
frame would be mistakenly dropped because of having the same timestamp
as a previous one, even though there was no previous one.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5004 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21 09:02:30 +00:00
eb61a851d5 Move audio_e2e_harness into include_tests==1 condition.
To avoid compile errors when WebRTC is built as a part of
Chromium.

TEST=ran gclient runhooks locally.
BUG=none
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5003 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-21 08:40:56 +00:00
88a310886e Add audio_e2e_test target to tools.gyp
The moving this GYP target out of webrtc.gyp in
https://code.google.com/p/webrtc/source/detail?r=4949
this should have been added into tools.gyp.

TEST=trybots passing
BUG=none
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5002 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-19 18:10:04 +00:00
fb648da2b9 Protect _transportPtr, which can be accessed by different threads in the case of external transport. This change avoid the potential use-after-free, e.g. the case in the reported bug.
BUG=2508
RISK=P1
TEST=try bots
R=henrika@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2425004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5000 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-18 21:10:51 +00:00
3e00505e9a Have padding decay to zero if no frames are being captured.
BUG=1837
TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4998 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-18 15:05:29 +00:00
893c07f209 Disable the -Wno-unused-const-variable Clang warning on Mac
In r4966 this was disabled on Linux for WebRTC code in order to detect any new unused const variables.
This CL does the same for Mac.

BUG=none
TEST=added an unused const and verified compilation fails
when this patch is applied. Mac trybots passing as well.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4997 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-18 14:42:27 +00:00
89b1e688ca Minor comment fix after clang reformat.
TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2479005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4996 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-18 14:23:29 +00:00
2df89c0c8b MouseCursorMonitor implementation for OSX and Windows.
BUG=crbug.com/173265
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2388004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4994 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-17 19:47:18 +00:00
6342066974 Fix tsan failures in channel.cc regarding to the volume settings.
BUG=2461
TEST=try bots
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2377004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4992 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-17 18:28:55 +00:00
675e260ad1 Check the number of playout channels instead of the send channels in StopPlayout()
BUG=2467
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2420004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4989 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-17 16:15:34 +00:00
c11148b352 Compound/reduced-size RTCP in VideoReceiveStream.
BUG=2424
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2413004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4987 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-17 14:14:42 +00:00
d030972139 Remove unused kPowTableFrac which causes anroid clang build failure.
http://build.chromium.org/p/tryserver.chromium/builders/android_clang_dbg/builds/84322/steps/compile/logs/stdio

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2417004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4981 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 20:32:09 +00:00
25fce9adc5 Fixed issue with how MTU is calculated.
BUG=
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2410004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4976 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 13:29:14 +00:00
b400aa7cd4 Don't pad if only one stream is sent, except if auto muted.
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2406004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4975 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 13:03:10 +00:00
e7009f3b0d Revert "Disable tests for TSan v2"
This reverts r4787 since r4966 got us a newer Clang
version with these issues fixed.

BUG=2259,2334
TEST=test_support_unittests and system_wrappers_unittests passed execution under TSan v2 for 100 iterations:
GYP_DEFINES='tsan=1 linux_use_tcmalloc=0 release_extra_cflags="-gline-tables-only"' gclient runhooks
ninja -C out/Release test_support_unittests system_wrappers_unittests

TSAN_OPTIONS="suppressions=tools/valgrind-webrtc/tsan_v2/suppressions.txt print_suppressions=1 report_signal_unsafe=0 report_thread_leaks=0 history_size=7" out/Release/system_wrappers_unittests --gtest_repeat=100 --gtest_break_on_failure

TSAN_OPTIONS="suppressions=tools/valgrind-webrtc/tsan_v2/suppressions.txt print_suppressions=1 report_signal_unsafe=0 report_thread_leaks=0 history_size=7" out/Release/test_support_unittests --gtest_repeat=100 --gtest_break_on_failure
R=hta@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2407004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4974 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 12:26:48 +00:00
5d957e29f7 Wired up max packet size and added simple test.
BUG=2428
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2384004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4973 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 11:37:54 +00:00
9401524211 Run FullStack tests without render windows.
Also disables test on valgrind platforms, it has no chance to keep up.

BUG=2278
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2159008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4972 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 11:05:37 +00:00
5ed4f46af1 Remove TSan v2 disabled test in condition_variable_unittest.cc
When we rolled our chromium_revision 226126:228675 in r4966
we picked up Clang r191856, which fixes the problem we've
seen earlier in condition_variable_unittest.cc.
Because of this, I'm now re-enabling this test.

TEST=trybots passing
BUG=2259
R=hta@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2404004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4971 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 10:50:41 +00:00
b44c2a3193 Open file in binary in CreateFromYuvFile().
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2399005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4969 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 08:46:06 +00:00
e6e749da38 Add MouseCursorRenderer.
The new class acts as a wrapper for DesktopCapturer interface. It takes
mouse shape and position from MouseCursorCapturer and renders it on the
frames produced by underlying DesktopCapturer.

BUG=crbug.com/173265
R=wez@chromium.org
TBR=andrew@webrtc.org (modules.gyp)

Review URL: https://webrtc-codereview.appspot.com/2387004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4968 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 02:48:41 +00:00
2767b53f66 Add MouseCursorCapturer interface with implementation for X11.
The new interface will be used to capture cursor shape and position and
blend it into the image captured with desktop capturers.

BUG=crbug.com/173265
R=wez@chromium.org
TBR=andrew@webrtc.org (modules.gyp)

Review URL: https://webrtc-codereview.appspot.com/2386005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4967 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 02:42:38 +00:00
3555303cb0 Roll chromium_revision 226126:228675 and fix clang warnings
By request from thakis@chromium.org, I disabled the
-Wno-unused-const-variable setting that is set in Chromium's
common.gypi so we can prepare our code for it's removal.

This required some cleanup in order to get the code to compile
with Clang having the -Wunused-const-variable warning enabled.

TEST=all trybots passing
BUG=none
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2400004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4966 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-15 20:10:17 +00:00
e5021fe590 Make RtpData and RtpFeedback destructors public.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4965 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-15 10:38:40 +00:00
266c7b330a Move ChromaGenerator to common_video/.
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2394004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4964 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-15 09:15:47 +00:00
c2e471d8b3 Compile out unused kMinTrustedDelayMs.
TBR=niklas.enbom@webrtc.org
TESTED=trybots

Review URL: https://webrtc-codereview.appspot.com/2398004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4963 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-15 02:11:21 +00:00
901ae77618 Android: Fixes WebRTCDemo build (missing Java code).
TBR=ajm@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/2395005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4961 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-14 21:46:53 +00:00