bjornv@webrtc.org
6a94734d4d
Adds back set_sample_rate_hz() when Init is called in recordings.
...
Recordings that had a AnalyzeReverseStream() call prior to ProcessStream() where aborted due to sample rates being set upon call by ProcessStream(). That change was done in r5346.
Before we have a smarter handling on how to set sample rate automatically, this CL adds back that setting.
BUG=
TESTED=trybots, modules_unittests
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5394 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 08:41:09 +00:00
andrew@webrtc.org
ea9392d5eb
MIPS optimizations for NS audio processing module
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4139006
Patch from Ljubomir Papuga <lpapuga@mips.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5393 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 07:22:01 +00:00
sergeyu@chromium.org
fb4e256d49
Fix crash in MouseCursor::CopyOf()
...
This issue was causing test failures with the latest webrtc roll.
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7249005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5392 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 04:45:35 +00:00
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