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Author SHA1 Message Date
andrew@webrtc.org
8f35afab8c Exclude protoc objects from merge_libs.py. 
BUG=b/12567343
R=wjia@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5391 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-16 00:31:57 +00:00
sergeyu@chromium.org
4b26e2eee3 Update libjingle to 59676287 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-15 23:15:54 +00:00
mallinath@webrtc.org
7a2ca7c621 Update needed to MockScreenCapturer after new methods addition to webrtc::ScreenCapturer. 
This change is also must for rolling webrtc in chrome.

R=jiayl@webrtc.org
TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/7219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5389 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-15 19:00:13 +00:00
wu@webrtc.org
8f19cb9fbc Revert 5387 "Re-enable webrtcvoice/videoengine unittests." 
Missed the result from the last try bot.

> Re-enable webrtcvoice/videoengine unittests.
> 
> TEST=try bots
> BUG=
> R=mallinath@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/7149004

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5388 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 22:31:11 +00:00
wu@webrtc.org
eda6823397 Re-enable webrtcvoice/videoengine unittests. 
TEST=try bots
BUG=
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 22:15:09 +00:00
jiayl@webrtc.org
017b619010 Extends the ScreenCapturer interface for individual display screen cast. 
Real implementations for each platform will be added in future CLs.

BUG=2787
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/6819005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 18:26:37 +00:00
wjia@webrtc.org
03cfde2d10 Roll Chromium 238260 -> 243863 
R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5385 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 17:48:34 +00:00
andresp@webrtc.org
8c5b27de9a Allow to skip turn by passing ts=false to apprtc. 
R=braveyao@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5384 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 17:00:23 +00:00
pbos@webrtc.org
39fcfd78ae Remove empty VideoCodecGeneric struct. 
Struct was added prematurely and triggers a warning with
-Wextern-c-compat in latest clang.

R=henrika@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/7119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 12:55:59 +00:00
henrik.lundin@webrtc.org
d9faa46d57 Changing to using factory methods for some classes in NetEq 
In this CL, the Expand, Accelerate and PreemptiveExpand objects are
created using factory methods. The factory methods are injected into
NetEqImpl on creation. This is a step towards implementing a no-decode
operation.

BUG=2776
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6999005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5382 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 10:18:45 +00:00
henrika@webrtc.org
aebb1ade9d pRevert 5371 "Revert 5367 "Update talk to 59410372."" 
> Revert 5367 "Update talk to 59410372."
> 
> > Update talk to 59410372.
> > 
> > R=jiayl@webrtc.org, wu@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/6929004
> 
> TBR=mallinath@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/6999004

TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5381 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 10:00:58 +00:00
aluebs@webrtc.org
4371d4650a Temporarily disabling some more audio processing tests. 
R=andrew@webrtc.org, bjornv@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7019005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5380 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-14 08:57:22 +00:00
sergeyu@chromium.org
eb31b45aaf Fix MouseCursorMonitorMac to return correct hotspot position. 
Previusly (0, 0) was always return as mouse cursor hotspot.

BUG=2779
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5379 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 23:25:17 +00:00
henrike@webrtc.org
3907c2e7e5 Removes the remaining uses of the list wrapper class and the list wrapper class. 
BUG=2164
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7019007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5378 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 22:41:34 +00:00
fischman@webrtc.org
dde7aee40f WebRTCDemo: fix out-of-bounds array read. 
Also removed the WebRtcCamera class, which has become an empty wrapper around
CameraInfo in the post-rewrite world.

First pointed out by Jeremy Mao <yujie.mao@webrtc.org> in
http://review.webrtc.org/6869004/

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5377 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 22:15:38 +00:00
fischman@webrtc.org
d7568a08c3 PeerConnection(java): Add OnRenegotiationNeeded support 
Also:
- Make PeerConnectionObserver::OnRenegotiationNeeded() pure virtual to avoid
  this sort of mistake in the future.
- Sprinkle @Override annotations on some callback definitions that were missing
  them.
- Fix a JNI method-signature-lookup typo (s/(V)V/()V/) in PCOJava::OnError()
- Add an explicit ScopedLocalFrameRef to PCOJava::OnError() to match all other
  C++-fired callbacks, for consistency.

BUG=2771
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5376 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 22:04:12 +00:00
elham@webrtc.org
ad1863de74 Updated Webrtc version to 3.49 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5374 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 17:49:49 +00:00
henrike@webrtc.org
79cf3acc79 Removes usage of ListWrapper from several files. 
BUG=2164
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5373 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 15:21:30 +00:00
andresp@webrtc.org
d0b436a935 Revert "Activate ACM test for Android in modules_tests." (rev5364). 
TBR=turaj@webrtc.org,tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6999006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5372 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 13:15:59 +00:00
henrika@webrtc.org
44461fa5cb Revert 5367 "Update talk to 59410372." 
> Update talk to 59410372.
> 
> R=jiayl@webrtc.org, wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/6929004

TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5371 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 09:35:02 +00:00
aluebs@webrtc.org
8bc4fcfeb6 Temporarily disabling audio processing tests. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6889005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5370 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 09:14:47 +00:00
henrik.lundin@webrtc.org
2c03bf1641 Increasing simulation time for NetEqPerformanceTest 
This is to get better "signal-to-noise ratio" in the performance bots.
The neteq4-runtime metric is expected to increase by a factor of 10.

BUG=2397
TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6989005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5369 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 09:04:23 +00:00
bjornv@webrtc.org
bbd47fc5b5 Enables robust delay validation in AEC delay logging. 
* Explicitly disabled robust validation in AECM.
* Updated audio_processing_unittests for using robust delay validation in AEC.
* Updated output_data_float.pb (not needed for Android nor fixed point, since AECM is untouched).

BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-13 08:54:34 +00:00
mallinath@webrtc.org
0f3356e20b Update talk to 59410372. 
R=jiayl@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5367 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-11 01:26:23 +00:00
andrew@webrtc.org
023cc5abc7 Minor voice engine improvements around AGC. 
- Remove one unneeded lock in CaptureLevel(), as the call to this
method should always come on the same thread as PrepareDemux().
- Remove check on analog AGC before doing volume calculations. Saves a
bit of code. Instead check if the incoming volume is set to zero, which
is a potentially common occurrence as it indicates no volume is
available.

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

Review URL: https://webrtc-codereview.appspot.com/6859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5366 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-11 01:25:53 +00:00
henrike@webrtc.org
573a1b45b5 Android: Fixes crash when exiting WebRTCDemo. 
BUG=2738
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5365 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 22:58:06 +00:00
turaj@webrtc.org
7cc64b3747 Activate ACM test for Android in modules_tests. 
TEST=local on Nexus 7.
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5364 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 22:35:09 +00:00
pbos@webrtc.org
f777cf2547 Permitting double start/stopping of streams. 
It doesn't make too much sense to hard enforce that the user keeps track
of which streams are started and which are not.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5363 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 18:47:32 +00:00
henrik.lundin@webrtc.org
a366e810a9 Adding NetEq performance test to webrtc_perf_tests 
The performance test is based on the neteq4_speed_test application. The
bulk of the test code is extracted into a test class, and included into
the neteq_unittest_tools target. The actual gtest that runs the
performance test is implemented in neteq_performance_unittest.cc, and
built as a part of webrtc_perf_tests.

The old stand-alone test application is now made dependent on the new
test class, to avoid code duplication.

BUG=2397
R=andrew@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5362 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 08:24:04 +00:00
bjornv@webrtc.org
fa8d534e09 Delay Estimator: Adds unittests for robust validation. 
In addition to unittests a cast losing constness was corrected.
The tests added are:
1. Adjusting allowed_offset when robust validation is disabled should have no impact.
2. For noise free signals there should be no difference between robust validation or not.
3. Robust validation acts faster during startup.

BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5361 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-10 07:42:07 +00:00
sergeyu@chromium.org
4625df3e3e Fix NaCl compilation 
nethelpers.cc was using LOG() but didn't include logging.h

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6829005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5360 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 21:26:50 +00:00
henrik.lundin@webrtc.org
e7ce437333 Fixing lint errors in NetEq4 
Just taking care of a few old lint errors.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 14:01:55 +00:00
andresp@webrtc.org
c5aeb2aa15 Make code simpler on VCMEncodedCallback. 
R=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5358 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 08:04:32 +00:00
andresp@webrtc.org
1df9dc3957 Isolate register post encode callback in video coding module to simplify code and critical sections. 
R=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 08:01:57 +00:00
vikasmarwaha@webrtc.org
bb0de3ca9f Updated Demos so they work on FF, changed the third argument in CreateOffer to null as it doesnot really require sdpConstraints. 
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/6769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5356 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 00:51:19 +00:00
fischman@webrtc.org
4177615e87 PeerConnection(java): replace ScopedLocalRef with ScopedLocalRefFrame and fix a local reference leak in OnMessage. 
Hopefully the approach of pushing/popping frames will be easier to avoid messing up than remembering to annotate every single local reference with a ScopedLocalRef.

BUG=2761
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5355 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-09 00:31:17 +00:00
fischman@webrtc.org
1794693ec8 AppRTCDemo(android): close() the throw-away DataChannel. 
Otherwise, the PeerConnection remembers the channel enough to include an
m=application line in its offer SDP, causing connection to chrome to fail, since
apprtc.appspot.com doesn't include an RtpDataChannels:true constraint in its
RTCPeerConnection constructor call.

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6729005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5354 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-08 18:29:34 +00:00
andresp@webrtc.org
b08a12d6e8 Isolate debug recording from video sender into a thread safe small class. 
R=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5353 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-08 12:38:22 +00:00
solenberg@webrtc.org
ab2405164a Add another test case for AST/TOF switching. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5899005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5352 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-08 08:59:44 +00:00
bjornv@webrtc.org
bccd53de57 Delay Estimator: Converts a constant into a configurable parameter. 
The parameter is used in the robust validation scheme, which will be turned on in a separate CL.

* Setter and getter for allowed delay offset.
* Updated unittests.

BUG=None
TESTED=modules_unittests, trybots
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5351 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-08 08:18:15 +00:00
wu@webrtc.org
e00265ed49 Fix a compile error on Android on sctpdataengine.cc. 
TEST=try bots
BUG=
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5350 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 19:32:40 +00:00
andrew@webrtc.org
d335094852 Init to 16 kHz in the fixed-point profile. 
Fixes modules_unittests for fixed-point builds (Android).

TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/6709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5349 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 18:57:10 +00:00
andrew@webrtc.org
b6541ca3a1 Ensure capture_levels_ is sized correctly at init time. 
Fixes failing voe_auto_test and audioproc_perf.

TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/6699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5348 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 18:36:10 +00:00
phoglund@webrtc.org
cf9d364063 Now printing less output from compare_videos.py. 
Alternative solution to the one in
https://codereview.chromium.org/114003006/.

I considered adding a verbose flag, but it needs to be passed through
like 5 functions, so I didn't think it was worth it for a function of
such speculative use.

BUG=chromium:327990
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5347 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 17:59:30 +00:00
andrew@webrtc.org
60730cfe3c Remove the requirement to call set_sample_rate_hz and friends. 
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)

Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.

TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.

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

Review URL: https://webrtc-codereview.appspot.com/3949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 17:45:09 +00:00
pbos@webrtc.org
39669c5c8f Remove outdated DestroyVideoSendStream comment. 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5345 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 12:27:22 +00:00
sprang@webrtc.org
ccd42840bc Wire up statistics in video send stream of new video engine api 
Note, this CL does not contain any tests. Those are implemeted as call
tests and will be submitted when the receive stream is wired up as well.

BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5559006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-07 09:54:34 +00:00
fischman@webrtc.org
0b7d8e6fcb AppRTC: Alert the user to failure to acquire TURN server. 
Hopefully will result in quicker turnaround time for CEOD/turnserver fixes.
Might trigger undesirable levels of bogus/spammy/unhelpful/PEBCAK reports to
discuss-webrtc, in which case I'll remove the second part of the message.

R=juberti@google.com, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4779005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5343 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-06 23:46:53 +00:00
marpan@webrtc.org
acc05ac7d1 Roll libvpx 232686:241571 
TBR=ajm@google.com

Review URL: https://webrtc-codereview.appspot.com/6599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5342 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-06 21:04:22 +00:00
andrew@webrtc.org
a9bdee6690 Add Christophe Dumez to AUTHORS. 
Copied from Chromium's AUTHORS.

R=ch.dumez@samsung.com

Review URL: https://webrtc-codereview.appspot.com/5559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5341 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-06 19:43:21 +00:00