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Author SHA1 Message Date
kma@webrtc.org
f87177a757 To fix a bug in InverseFFTAndWindow() function in AECM. 
It's a bufer overwritting issue, and thus Android AppRTCDemo app was broken (reported by Ami).
Tested with audioproc offline test. Bit-exact.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4415 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 23:43:33 +00:00
kma@webrtc.org
b6a6a24fda Updated WebRtcNsx_PrepareSpectrumNeon() in accordance with the new real FFT interface in APM. For reference, you can check https://webrtc-codereview.appspot.com/1830004/diff/92001/webrtc/modules/audio_processing/ns/nsx_core.c, line 594 "static void PrepareSpectrumC()". 
Tested with audioproc. Bit exact.

R=andrew@webrtc.org, johannkoenig@google.com

Review URL: https://webrtc-codereview.appspot.com/1859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4411 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 16:24:34 +00:00
braveyao@webrtc.org
b6433b7a1e Access receiving_ under receive_cs critical section 
Note: InsertRTPPacket/InsertRTCPPacket could be merged into 
ReceivedRTPPacket, as there are no other callers.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1869005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4410 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 09:02:46 +00:00
sergeyu@chromium.org
abab1d8456 Don't set clang_use_chrome_plugins in common.gypi 
This caused a failure on chrome os ASAN bots (where that flag is disabled):
http://build.chromium.org/p/chromium.memory/builders/Chromium%20OS%20%28x86%29%20ASAN/builds/5491

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1882004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4408 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-26 00:55:46 +00:00
henrike@webrtc.org
14c966c706 Fixes resources and data path in modules_unittests.isolate. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1859005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4407 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 22:44:04 +00:00
andrew@webrtc.org
b86fbaf1d4 Downstream latest Chromium SincResampler changes. 
Replace the BlockSize() workaround we were using previously to support
the push wrapper with the upstream request_frames interface. This
requires a bit of a trick to ensure we don't add more delay than
necessary. On the first pass we use a dummy Resample() call in order to
prime the buffer such that all later calls only require a single input
request through Run().

Notably, this brings in an optimized loop condition, improving
performance by ~2% - 3% on tested platforms and avoids a 20% performance
hit with clang. This addresses issue2041.

Only negligible changes to the PushSincResamplerTest SNR thresholds, due
to a fractional sample adjustment in output delay.

This still retains the per-instance CPU detection, as webrtc lacks a
LazyInstance helper for static initialization.

Ideally, we would adopt SetRatio() in PushSincResampler's
InitializeIfNeeded() for on-the-fly changes, but this will require a way
to update request_frames.

The diff against Chromium upstream is available here:
https://codereview.chromium.org/19470003

BUG=2041
TESTED=unit tests, voe_cmd_test in loopback running through all codecs
with 44.1 kHz and 48 kHz device formats using a stereo mic.

R=dalecurtis@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1838004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 22:04:30 +00:00
sergeyu@chromium.org
099b8c9e8e Update include paths in device_info_external.cc 
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1875004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 18:41:43 +00:00
andrew@webrtc.org
61e596fc49 Add a Config class interface to AudioProcessing for passing options. 
Pass the Config down to all AudioProcessing components.

Also add an EchoCancellationImplWrapper to optionally create different
EchoCancellationImpls.

BUG=2117
TBR=turaj@webrtc.org
TESTED=git try

Review URL: https://webrtc-codereview.appspot.com/1843004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4400 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 18:28:29 +00:00
niklas.enbom@webrtc.org
8e3bbedacd Fix include path in video_capture_external.cc 
Fix build error introduced in r4337

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1873004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4397 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-25 16:55:58 +00:00
kma@webrtc.org
fc8aaf02e1 Formalized Real 16-bit FFT for APM. 
It also prepares for introducing Real 16-bit FFT Neon code from Openmax to SPL. CL https://webrtc-codereview.appspot.com/1819004/ takes care of that, but this CL is a prerequisite of that one.
Tested audioproc with an offline file. Bit exact.

R=andrew@webrtc.org, rtoy@google.com

Review URL: https://webrtc-codereview.appspot.com/1830004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-24 17:38:23 +00:00
sergeyu@chromium.org
d102e66ef9 Fix ScreenCapturerLinux not to use XDamage when requested. 
When moving this code to webrtc I added line "use_x_damage=true" for
debugging and forgot to remove it when landing this code, so the
capturer always tries to use XDamage.

BUG=crbug.com/263003
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/1854004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-23 20:05:42 +00:00
fischman@webrtc.org
678cf29d8b webrtc/common_types.h: Document bitrate fields' units. 
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1847004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-23 18:32:10 +00:00
henrike@webrtc.org
8d27a1c723 Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle 
BUG=1932
TESTED=git try
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1851004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-23 18:15:11 +00:00
mflodman@webrtc.org
6879c8adad Hooking up first simple CPU adaptation version. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1767004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-23 11:35:00 +00:00
henrike@webrtc.org
5c280ecd57 Revert 4382 "Makes webrtc and libjingle build from the same gyp-..." 
Failures: breaks build bots. Will have to disable Android NDK build for libjingle. The TSAN issues are in webrtc which should be unaffected. Flakey? Here are the failing tests:
 http://chromegw/i/internal.client.webrtc/builders/Android%20NDK/builds/303 and http://chromegw/i/internal.client.webrtc/builders/Linux%20Tsan/builds/284

> Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle
> 
> BUG=1932
> TESTED=git try
> R=andrew@webrtc.org, fischman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1836004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1834005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-23 03:30:32 +00:00
henrike@webrtc.org
5fcddf2334 Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle 
BUG=1932
TESTED=git try
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1836004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4382 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-23 00:27:43 +00:00
henrike@webrtc.org
390fcb7a20 Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder. 
BUG=2091
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1833004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4381 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-22 22:32:50 +00:00
yujie.mao@webrtc.org
129afc29fb Correctly rebuild WebRTCDemo after jni/ source file changes 
BUG=1980
TEST=Modify source file under jni/ and WebRTCDemo will rebuild
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1831004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4377 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-20 04:43:08 +00:00
henrike@webrtc.org
0df5b8dfa6 Revert 4372 "Makes webrtc and libjingle build from the same gyp-..." 
> Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches.
> 
> TESTED=git try
> BUG=1932
> R=andrew@webrtc.org, fischman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1804004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1835004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-18 18:58:29 +00:00
henrike@webrtc.org
4e4bf4db8b Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. 
TESTED=git try
BUG=1932
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1804004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-18 18:33:55 +00:00
fischman@webrtc.org
c6d5b50b41 AppRTCDemo: build fixes for iOS build in webrtc 
BUG=1421,1450,1451
TESTED=git try, also the same patch (along with a bunch of other, non-webrtc changes) in a libjingle checkout allows building iOS AppRTCDemo
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4371 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-18 02:02:07 +00:00
tnakamura@webrtc.org
d2102afa2a Undo libvpx include changes in r4348 to fix build. 
A longer term fix is needed, but this at least quickly unblocks the build.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1816005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-17 18:48:24 +00:00
pbos@webrtc.org
a3f30143b7 Default constructor for RtcpAppHandler. 
Whenever this test (RtcpApplicationDefinedPacketsCanBeSentAndReceived) fails
because it's being run on a slower system (such as one running under valgrind),
valgrind reports a lot of undefined-value errors. Initializing the data
makes sure that, while the EXPECT_EQs trigger, they don't cause any errors in
valgrind.

BUG=
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1822004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-17 14:25:45 +00:00
tnakamura@webrtc.org
64e2cbf184 clean up incomplete revert in r4357 
Also revert r4319, will follow up with pbos

Reason for recent series of reverts: video freezes when testing with packet loss

R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1817004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 21:52:59 +00:00
tnakamura@webrtc.org
aa4d96a134 Revert r4301 
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 19:25:04 +00:00
henrike@webrtc.org
42581545eb Fixes: Resolves conflict that will happen when merging libjingle's and WebRTC's supplemental.gyp. By separating build_with_chromium and build_with_libjingle one can now just define build_with_libjingle in libjingle's supplemental.gyp. Once that is done it will be possible to merge the two supplemental.gyp-files. I.e. in WebRTC the supplemental.gyp would only set build_with_chromium to 0 since there is no longer any reason to disable logging and tests as they will be accessible in the same repository as libjingle. 
Libjingle sets the variables here: https://code.google.com/p/libjingle/source/browse/trunk/talk/supplement.gypi

BUG=N/A
R=andrew@webrtc.org, fischman@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1787005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4354 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 16:37:22 +00:00
pbos@webrtc.org
3d8647f17d Include files from webrtc/.. paths in signal_processing/. 
BUG=1662
R=andrew@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1784004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4352 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 13:32:03 +00:00
pbos@webrtc.org
0c4e05afbb Include files from webrtc/.. paths in media_file/. 
BUG=1662
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1784005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4351 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 13:05:40 +00:00
pbos@webrtc.org
9b82dced8d Make sure first RTP packet counts as in-order. 
BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1811004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4350 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 13:03:35 +00:00
pbos@webrtc.org
2e10b8e4a0 Include files from webrtc/.. paths in bitrate_controller/. 
BUG=1662
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1787004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 12:54:53 +00:00
pbos@webrtc.org
a4407329d4 Include files from webrtc/.. paths in video_coding/. 
BUG=1662
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1783006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 12:32:05 +00:00
elham@webrtc.org
4a44ea21d7 Revert r4320 "Fix three uninitialized members in rtp_receiver_impl.cc" 
TBR=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1803004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 21:46:06 +00:00
elham@webrtc.org
4888fd4827 Revert r4321 "Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered" 
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1790006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4345 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 21:21:48 +00:00
elham@webrtc.org
b7eda43810 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on 
several SSRCs"

R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1774006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 21:08:27 +00:00
elham@webrtc.org
6f5707e184 Revert r4328 
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1774005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4343 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 20:59:52 +00:00
elham@webrtc.org
8543c1c77c Updated WebRTC version to 3.36 
TBR=tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1780005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4341 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 17:19:45 +00:00
pbos@webrtc.org
df119c9a45 Remove dead video_capture for QuickTime. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4339 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-12 18:08:13 +00:00
pbos@webrtc.org
a9b74ad716 Include files from webrtc/.. paths in video_capture/. 
BUG=1662
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1788004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4337 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-12 10:03:52 +00:00
pbos@webrtc.org
8b06200802 Include files from webrtc/.. paths in utility/. 
BUG=1662
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1786004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4336 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-12 08:28:10 +00:00
pbos@webrtc.org
0ed57c51a3 Remove dead code testAPI.cc. 
BUG=
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1783005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4335 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-12 08:23:05 +00:00
pbos@webrtc.org
5aa3f1b4c0 Include files from webrtc/.. paths in video_render/. 
BUG=1662
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1782006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4334 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-12 08:12:08 +00:00
pbos@webrtc.org
5b10d8fb18 Fix some voe_auto_test uninitialised-value errors. 
BUG=
R=tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1783004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-11 15:50:07 +00:00
pbos@webrtc.org
811269df40 Include files from webrtc/.. paths in audio_device/. 
BUG=1662
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1785005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-11 13:24:38 +00:00
pbos@webrtc.org
db6e3f8bc5 Fix root-relative includes for pacing/. 
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4329 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-11 09:50:05 +00:00
stefan@webrtc.org
e4736eee20 Fixes a crash when sending SR reports from a sender only module. 
BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1790004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-11 08:28:35 +00:00
braveyao@webrtc.org
aeba6e8740 ModuleRTPRTCP call rtcp_sender_.TMMBR() directly instead of calling its own API. 
BUG=2051
TEST=autotest
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1790005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4327 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-11 08:06:37 +00:00
pbos@webrtc.org
96edd56170 Sorted headers under rtp_rtcp/. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1781005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 15:40:42 +00:00
pbos@webrtc.org
69215d8432 Include files from webrtc/.. paths in video_engine/. 
BUG=1662
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1759005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4324 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 15:02:02 +00:00
pbos@webrtc.org
adf23a55f8 Direct3D renderer for new VideoEngine API tests. 
TEST=Rendered video in video_loopback test.
BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1573004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4323 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 14:07:56 +00:00
stefan@webrtc.org
717d147ebb Support sending multiple report blocks and keeping track of statistics on several SSRCs. 
BUG=1811
TEST=vie_auto_test --automated, voe_auto_test --automated, trybots
R=andresp@webrtc.org, tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1768004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 13:39:27 +00:00