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Author SHA1 Message Date
henrik.lundin@webrtc.org
def1e97ed2 Implement AudioEncoderPcmU/A classes and convert AudioDecoder tests 
BUG=3926
R=kjellander@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7481 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 12:48:29 +00:00
bjornv@webrtc.org
78ea06dd34 audio_coding/codecs/ilbc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
Removed usage of trivial macro.

BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7480 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 07:17:24 +00:00
henrik.lundin@webrtc.org
913f7b8d5e Fix for glitches in ACM when switching desired output sample rate 
The problem was that if the output sample rate is changed such from one
where no resampling is needed to a rate that requires resampling, the
first output from the resampler will contain an onset period. The
solution provided in this CL is to keep a copy of the last output frame
in ACM, and if the resampler is engaged, it will be primed with this
old frame before resampling the current frame.

BUG=3919
R=bjornv@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 06:54:23 +00:00
glaznev@webrtc.org
a8c0edd29f Avoid using EGLContext class for Android 4.1 and below. 
Support for this class was added in Android 4.2, so
disable surface decoding for lower Android versions.

BUG=3901
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7478 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 19:08:05 +00:00
bjornv@webrtc.org
b69ea9a35a common_audio: Replaced invalid operand in min_max_operations_neon.S" 
Vector Move immediate can not load #0x7FFF. Changed to us vdup from already loaded register.

BUG=N/A
TESTED=ios and android trybots
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7477 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 14:08:35 +00:00
pbos@webrtc.org
fa553ef605 Set up start bitrate in WebRtcVideoEngine2. 
R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/27789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7476 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 11:07:07 +00:00
pbos@webrtc.org
b35b136480 Make avg_{psnr,ssim}_threshold_ const. 
Triggered warning on next clang version being rolled as these variables
are annotated to be protected by crit_.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/24949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7475 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 09:14:38 +00:00
bjornv@webrtc.org
2abebe7baf audio_coding/codecs/isac/main: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7474 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 08:26:41 +00:00
bjornv@webrtc.org
a5ce7bbe17 audio_coding/neteq: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7473 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 08:24:54 +00:00
henrike@webrtc.org
28100cb388 Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." 
BUG=N/A
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 22:03:39 +00:00
buildbot@webrtc.org
7992b40994 (Auto)update libjingle 77953038-> 77970462 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7471 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 21:20:28 +00:00
henrike@webrtc.org
b1dac33cac Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..." 
BUG=3932
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/27779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7470 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 18:54:46 +00:00
glaznev@webrtc.org
58202946a7 Cleaning up Android AppRTCDemo. 
- Move signaling code from Activity to a separate class
and add interface for AppRTC signaling. For now
only pure GAE signaling implements this interface.
- Move peer connection, video source and peer connection
and SDP observer code from Activity to a separate class.
- Main Activity class will do only high level calls and
event handling for peer connection and signaling classes.
- Also add video renderer position update and use full
screen for local preview until the connection is established.

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

Review URL: https://webrtc-codereview.appspot.com/24019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7469 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 17:42:38 +00:00
houssainy@google.com
0371a37f85 Moving creating TURN configration to the host machine instead of the bots - rtcBot 
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7468 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 16:43:50 +00:00
glaznev@webrtc.org
f7030d4ed7 Query Android device orientation on every camera frame received. 
Remove orientation listener from Android camera, since device
orientation change events are not well synchronized with actual
device display orientation. Plus these event may not be delivered
at all if device is in stationary position causing initial camera
frames appear rotated.

BUG=
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 16:25:06 +00:00
henrike@webrtc.org
9c58ea8d56 rtc_unittest: copied gtest excludes from libjingle_p2p_unittest since its tests have move to rtc_unittests. 
BUG=3925
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/28739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7466 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 16:12:33 +00:00
houssainy@google.com
c221db6165 Test names changed from e.g) testOneWayVideo/chrome=>chrome to testOneWayVideo/chrome-chrome. 
Because the symbol ">"  is interpreted as special command for output to file in bash commands.

TBR= andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7465 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 09:13:43 +00:00
henrik.lundin@webrtc.org
264e66f7a5 Add encoded_timestamp to AudioEncoder base class 
BUG=3926
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-16 21:16:07 +00:00
henrik.lundin@webrtc.org
9ea6f8a84d New interface class AudioEncoder 
This class will be the base for new C++ wrapper classes for all
encoders.

BUG=3926
TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7463 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-16 11:26:24 +00:00
stefan@webrtc.org
8efaa270d8 Disable a bunch of Nat and Ice tests when running under DrMemory. 
BUG=3925
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7462 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-16 11:21:42 +00:00
andresp@webrtc.org
458c2c3b06 Improve rtcbot to load all test files at start and allow them to registerTests 
via: registerBotTest. After loading all tests main.js starts running the
requested one on the command arguments.

R=houssainy@google.com

Review URL: https://webrtc-codereview.appspot.com/29779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7461 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-16 07:36:37 +00:00
asapersson@webrtc.org
9aed002090 Add ability to include a larger time span (in addition to encode time) for measuring the processing time of a frame. 
Controlled by setting enable_extended_processing_usage. Enabled by default.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7460 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-16 06:57:12 +00:00
henrike@webrtc.org
d1ba6d9cbf Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. 
BUG=3379
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27709005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 17:30:28 +00:00
houssainy@google.com
3e2f8ff36c Selecting bot_type changed to be specified in the test file 
Selecting bot_type changed to be specified in the test file instead of
specify it in the running command.

Now we can write test for rtcBot that run one bot on chrome for android
and the other bot on chrome for desktop.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7458 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 15:01:11 +00:00
pbos@webrtc.org
e93cbd13d5 Fix data races in ThreadTest.ThreeThreadsInvoke. 
R=henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/26819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7457 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 14:54:56 +00:00
bjornv@webrtc.org
f87c0aff7f audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
Also includes a typo in a comment.
Affects
* aecm
* hpf

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7456 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 12:51:23 +00:00
bjornv@webrtc.org
f02ba9be54 audio_processing/agc: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
Affects AGC only.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 11:16:48 +00:00
bjornv@webrtc.org
8dc00d76af audio_processing/ns: Replaced macro WEBRTC_SPL_RSHIFT_W32 with >> 
Affects fixed point version of Noise Suppression.

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7454 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 09:31:40 +00:00
henrik.lundin@webrtc.org
99e561f6a6 Extend AcmSwitchingOutputFrequencyOldApi with more frequencies 
Also reducing test duration, since the issue is triggered anyway.
The tests that are not failing are now enabled.

BUG=3919
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7453 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 08:50:00 +00:00
kjellander@webrtc.org
64f5611b3d Roll chromium_revision 2d714fa..de13cf4 (298667:299488) 
Mainly to pick up https://codereview.chromium.org/648613007
to fix some MSan issues.

Summary of changes (2d714fa..de13cf4/DEPS):
* third_party/boringssl 51fcd87..7ea8481
* third_party/icu d2abf6c..8ac906f
* third_party/usrsctp/usrsctplib dfd687b..a11b3c5

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7452 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 05:59:42 +00:00
bjornv@webrtc.org
fab5439112 common_audio: Removed version API from signal_processing 
The Signal Processing version API is not used anymore.

BUG=3353
R=kwiberg@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7451 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 04:38:42 +00:00
buildbot@webrtc.org
81ddc78536 (Auto)update libjingle 77701902-> 77709729 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 22:39:24 +00:00
buildbot@webrtc.org
1ecbe45c7e (Auto)update libjingle 77689511-> 77696841 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7449 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 20:29:28 +00:00
pbos@webrtc.org
43336b6b9f Remove unused (no-op) VideoOptions. 
Removing VideoOptions: adapt_input_to_encoder, adapt_view_switch,
video_one_layer_screencast and video_high_bitrate.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7448 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 19:12:06 +00:00
henrike@webrtc.org
a4351a045d libjingle: use _stricmp instead of deprecated stricmp. 
BUG=N/A
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7447 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 17:07:41 +00:00
pbos@webrtc.org
a73a678e25 Remove -1 from Call::Config::start_bitrate_bps. 
Instead initialize it to a good default value. The code does the same,
but we don't have to check explicitly for -1.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7445 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 11:52:10 +00:00
stefan@webrtc.org
eb24b04f16 Add periodic logging of received RTP headers and estimated clock offsets for e2e delay. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7444 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 11:40:13 +00:00
henrik.lundin@webrtc.org
81a78930ee New ACM test to trigger audio glitch when switching output sample rate 
This CL implements a new unit test. The test is designed to trigger
a problem in ACM where switching the desired output frequency creates
a short discontinuity in the output audio. The problem itself is not
solved in this CL, but the failing test is disabled for now.

BUG=3919
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7443 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 10:49:58 +00:00
stefan@webrtc.org
c216b9aeaf Add a packet loss full stack test to the new API. 
Remove all full stack tests for the old API.

BUG=3750
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7442 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 10:38:49 +00:00
kwiberg@webrtc.org
a57678a70e Workarounds for a bug in VS2013.3 linker when PGO is turned on. 
See crbug.com/421607 for more details about this. This CL solve a linker bug when the PGO is turned on, without changing the behaviour or the performances.

BUG=crbug.com/421607
R=kwiberg@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26789005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7441 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 09:40:04 +00:00
pbos@webrtc.org
7fe1e03dd6 Wire up external encoders. 
R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/30649005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7440 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 04:25:33 +00:00
buildbot@webrtc.org
f68cc0b0c3 (Auto)update libjingle 77554188-> 77629208 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7439 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 01:17:42 +00:00
marpan@webrtc.org
82e6fa533c Move exlusion of VP9 integration tests for DrMemory 
from modules_unittests to modules_tests file.

Also rename and move ProcessNoLossChangeBitRate, 
and move TestVp8Impl.BaseUnitTest to proper file.

The previous commit r7435 disabled it in the wrong file.

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7438 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 00:34:19 +00:00
aluebs@webrtc.org
b6af4283ca Adjust speech probability in NS when echo 
The average speech probability for the higher band is multiplied by the quotient of the process and analyze powers, to avoid thinking that suppressed echo is speech. In order to do this both magnitudes, alanyze and process, needed to be stored. This also was used to calculate different previous STSA estimates for analyze and process.
This CL was tested on two long team member recordings (bjornv and kwiberg) and the noisiest (5) recordings from the QA set.

BUG=webrtc:3763
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7437 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 20:48:05 +00:00
henrike@webrtc.org
1e6a5dd14e Removes xmllite from talk/xmllite since webrtc/xmllite is used instead. 
BUG=3379
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/23039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7436 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 18:27:11 +00:00
marpan@webrtc.org
8bee130fa0 Disable VP9 integration tests on DrMemory. 
Will try re-enabling them on next libvpx roll using faster codec speed setting.

BUG=3917

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7435 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 17:10:40 +00:00
bjornv@webrtc.org
bc1a4578e0 common_audio: Removed macro WEBRTC_SPL_RSHIFT_W16 
Replaced the trivial right shift macro at remaining 4 places and removed from signal_processing.

Affected components:
* vad
* aecm

BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7434 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 14:00:43 +00:00
kwiberg@webrtc.org
a3722b643d iSAC tests: Type buffers as uint8_t[] to avoid casts 
The iSAC interface functions now expect uint8_t arrays, so change some
arrays to be of that type instead of casting at each point of use.

R=bjornv@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7433 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 13:29:04 +00:00
bjornv@webrtc.org
d4fe824862 audio_processing: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >> 
The implementation of WEBRTC_SPL_RSHIFT_W16 is simply >>. This CL removes the macro usage in audio_processing and signal_processing.

Affected components:
* aecm
* agc
* nsx

Indirectly affecting (through signal_processing changes)
* codecs/cng
* codecs/isac/fix
* codecs/isac/main

BUG=3348,3353
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28699005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7432 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 13:01:13 +00:00
kwiberg@webrtc.org
396a5e0001 WebRtcIsac_Decode et al.: Type encoded data as uint8[], not uint16[] 
This patch changes WebRtcIsac_Decode, WebRtcIsac_DecodeRcu, and
WebRtcIsacfix_Decode so that they read the encoded data from a uint8
array instead of a uint16 array.

BUG=909
R=aluebs@webrtc.org, bjornv@webrtc.org, henrik.lundin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7431 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 11:23:24 +00:00