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Author SHA1 Message Date
buildbot@webrtc.org
af81b9bffd (Auto)update libjingle 68499439-> 68501302 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6320 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-04 00:08:54 +00:00
buildbot@webrtc.org
251fdf64cb (Auto)update libjingle 68495561-> 68499439 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6319 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 23:43:48 +00:00
henrike@webrtc.org
09a71cd9ce talk/ios: Fixes source after corrupt sync in r6305 (which corrupted r6291). 
BUG=N/A
R=tkchin@webrtc.org
TBR=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6318 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 22:46:23 +00:00
buildbot@webrtc.org
53217848b2 (Auto)update libjingle 68465410-> 68487517 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6317 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 21:09:11 +00:00
marpan@webrtc.org
4ef254f781 Enable videoprocessor_integrationtest tests on android. 
R=kjellander@webrtc.org, stefan@webrtc.org
TBR=holmer@google.com

Review URL: https://webrtc-codereview.appspot.com/15599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6316 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 16:42:03 +00:00
fischman@webrtc.org
83eb7dff5c PeerConnection(java): disable wait for flaky ICEConnection.COMPLETED. 
This should be reverted when COMPLETED is delivered reliably.

BUG=3021
TESTED=without this patch the test fails in Debug mode after a handful of runs.  With this patch 100 runs passed in a row on my desktop.
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6315 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 16:38:08 +00:00
turaj@webrtc.org
ddc6bc9347 Revert 6312 "Re-enable AudioCodingModuleMtTest" 
An example of botbreakage is http://chromegw.corp.google.com/i/client.webrtc/builders/Linux%20Memcheck/builds/1807

> Re-enable AudioCodingModuleMtTest
> 
> Increase timeout and decrease test length. Also fixing a bug in the
> test, and make sure the test aborts if fatal failure occurrs.
> 
> BUG=3426
> R=kwiberg@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/13579005

TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6314 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 15:25:34 +00:00
pbos@webrtc.org
289a35c56d Add empty webrtcmediaengine.cc. 
Should contain CreateWebRtcMediaEngine as soon as
libjingle/libjingle.gyp in Chromium builds this file. This file is added
ahead of time to get a smoother rolling process.

BUG=1788
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6313 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 14:51:34 +00:00
henrik.lundin@webrtc.org
8d13cd1956 Re-enable AudioCodingModuleMtTest 
Increase timeout and decrease test length. Also fixing a bug in the
test, and make sure the test aborts if fatal failure occurrs.

BUG=3426
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6312 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 12:53:21 +00:00
kwiberg@webrtc.org
8e4401b5a0 Reformat integer accessors to look like their float counterparts 
The new format is at least as easy to read, and takes less space.

BUG=
R=aluebs@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6311 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 10:04:13 +00:00
kwiberg@webrtc.org
f2e4a99a39 Add kwiberg@webrtc.org to watchlist for audio_coding and audio_processing 
BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6310 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 10:01:39 +00:00
buildbot@webrtc.org
b525a9d790 (Auto)update libjingle 68379861-> 68445177 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6309 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 09:42:15 +00:00
pbos@webrtc.org
044bdacfef Remove kMaxWaitForStatsMs from tsanv2 compilation. 
As some tests are #ifdef'd out on THREAD_SANITIZER this constant
triggers an unused-const-variable warning which breaks the build.

BUG=1205,3220
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 09:40:01 +00:00
kwiberg@webrtc.org
c0035a67a1 Remove an optimization that's no longer worth the extra complexity it causes 
The data_ optimization was a way to operate on the data directly
instead of copying it, applicable in the mono, non-float case. Since a
few audio_processing steps are already using floats (with more
hopefully to come), we don't end up benefiting from the optimization
anyway, so we might as well remove it.

BUG=
R=aluebs@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6307 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-03 09:10:06 +00:00
buildbot@webrtc.org
34a08b4fb8 (Auto)update libjingle 68275107-> 68379861 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6305 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 15:48:10 +00:00
solenberg@webrtc.org
a28c697d93 - Get rid of 'using' from .h 
- Add parenthesis to make order of evaluation clearer.

BUG=
R=minyue@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6304 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 15:22:33 +00:00
kjellander@webrtc.org
2f7c7ce020 Remove old perf_expectations no longer used. 
This has been replaced with the Chromium Perf
Dashboard web application a long time ago.

BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6303 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 15:03:21 +00:00
henrik.lundin@webrtc.org
2bd032e11c Disable MouseCursorMonitorTest 
Last attempt reverted. Trying again in a different way.

This CL effectively reverts r6300.

BUG=3245
TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/20549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6301 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 14:52:34 +00:00
henrik.lundin@webrtc.org
4ecae6e753 Disable MouseCursorMonitorTest.FromScreen 
The test is flaky.

BUG=3245
TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6300 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 14:17:06 +00:00
henrik.lundin@webrtc.org
fe41a8f68d Adding thread annotations to parts of Audio Coding Module 
Picking some low-hanging fruit. Add annotations for acm_crit_sect_ that
do not require lock changes. Also adding annotations for callbacks.

BUG=3401
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6299 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 11:45:26 +00:00
bjornv@webrtc.org
2812b59acd Re-enables CommonFormats test for Android. 
It seems like this was a one time only and not a flaky test.

BUG=3376
TESTED=trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15649005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6298 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 11:27:29 +00:00
pbos@webrtc.org
174a67439b Enable -Wall, -Wextra and -Wunused-variable for talk/ on clang. 
Also removes one case of unused-variable.

BUG=3220
R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15619005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6297 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-02 07:58:30 +00:00
jiayl@webrtc.org
8a09af3f67 Fix the build error from OpenSSLStreamAdapter::SSLVerifyCallback 
TBR=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/17639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 23:24:08 +00:00
fischman@webrtc.org
360507b12b VideoCaptureAndroid: don't synchronized on camera thread. 
BUG=3421
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6295 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 23:17:38 +00:00
jiayl@webrtc.org
0163674f99 Make OpenSSLStreamAdapter verify the leaf certificate digest for chained certificates. 
It used to compre a parent certificate's digest against the SDP fingerprint and caused connection failure.

BUG=3383
R=bemasc@webrtc.org, juberti@webrtc.org, rsleevi@chromium.org

Review URL: https://webrtc-codereview.appspot.com/17589005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 23:14:08 +00:00
andrew@webrtc.org
222d8d3b1f Add a TSAN suppression for a benign TRACE_EVENT race. 
TBR=hclam
BUG=3409

Review URL: https://webrtc-codereview.appspot.com/15639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 23:14:00 +00:00
tkchin@webrtc.org
56d114627b Fix AppRTC target configuration in libjingle_examples.gyp. 
libjingle_peerconnection_objc doesn't exist as a target in 32bit, so AppRTCDemo
needs that guard as well.

R=andrew@webrtc.org

BUG=

Review URL: https://webrtc-codereview.appspot.com/18489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6292 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 23:04:39 +00:00
tkchin@webrtc.org
acca675bcf Implement mac version of AppRTCDemo. 
- Refactored and moved AppRTCDemo to support sharing AppRTC connection code between iOS and mac counterparts.
- Refactored OpenGL rendering code to be shared between iOS and mac counterparts.
- iOS AppRTCDemo now respects video aspect ratio.

BUG=2168
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6291 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 22:26:06 +00:00
jiayl@webrtc.org
9f8164c060 Fix two bugs in DataChannel state transition. 
1. OnStateChange should not be fired if state is not changed.
2. RemotePeerRequestClose should be a no-op if it's already closed.

TBR=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/21559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 21:53:17 +00:00
andrew@webrtc.org
1fddd6185d Add a Reset() method to AudioFrame. 
This method is introduced to try to avoid inconsistent resetting of
AudioFrame members to default/uninitialized values.

Use it at the call points of DownConvertToCodecFormat(). Results in the
following minor functional changes:
- speech_activity_ is set to its uninitialized value. AFAICT, this
member isn't used at all in the capture path.
- timestamp_ is switched from -1 to 0. This member doesn't appear to be
used either in the capture path, but left a TODO for wu to change the
default value to better represent the uninitialized state.

Bonus: Don't copy the frame on error in RemixAndResample(). An error
indicates a logical fault (as pointed out by the asserts) that we should
not attempt to recover from.
BUG=3111
R=turaj@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21519007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6289 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 17:28:50 +00:00
andrew@webrtc.org
af48aaadf4 Disable AudioCodingModuleMtTest due to memcheck and tsan failures. 
This is a new test; the failures are not due to a change in underlying code.

TBR=henrik.lundin
BUG=3426

Review URL: https://webrtc-codereview.appspot.com/19589005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6288 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 17:11:15 +00:00
buildbot@webrtc.org
1678db9df6 (Auto)update libjingle 68230113-> 68244456 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6287 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 14:02:09 +00:00
henrik.lundin@webrtc.org
288bd15db8 Multi-threaded test for Audio Coding Module 
This CL adds a basic multi-threaded extention of the ACM unit test.
The test has three threads. One thread adds raw audio to the sender
side and encodes it. The next thread adds encoded RTP packets to the
receiver. The last thread pulls decoded audio out of the receiver.

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6286 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 13:00:35 +00:00
pbos@webrtc.org
b4e3c254ee Add native_test dependency to webrtc_perf_tests. 
Required to run the binary on Android bots.

BUG=3423
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6285 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 12:42:10 +00:00
stefan@webrtc.org
420b2567f3 Fix bug where RTP headers in the packet history were replaced with the RTX wrapped headers. 
This caused only the first retransmission to be successful.
Introduced with https://code.google.com/p/webrtc/source/detail?r=5728.

BUG=1811
R=asapersson@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 12:17:15 +00:00
minyue@webrtc.org
a816180f93 Fixing a bug regarding VOE packet loss rate feedback to ACM 
Phenomenon:

When packet loss rate was fed to a codec that does not implement packet loss adaptive encoding, VoE logs an error.

Reason:

The ACM function SetPacketLossRate(int rate) return -1 unnecessarily too often. It was intended for more severe errors like
1. codec is not ready
2. input rate is out of range

BUG=webrtc:3413
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 09:28:07 +00:00
sprang@webrtc.org
6e732c6765 Revert 6272 "Update generated asm offsets scripts." 
Revert since it fails webrtc-in-chromium Android bots.

> Update generated asm offsets scripts.
>
> Libvpx updated the unpack scripts to fix building dependencies.
>
> Roll libvpx 269083:273304
> See https://codereview.chromium.org/295313002/
> https://codereview.chromium.org/298063002/
> https://codereview.chromium.org/305533008/
> https://codereview.chromium.org/305703002/
> https://codereview.chromium.org/298383003/
> https://codereview.chromium.org/302863004/
> for the libvpx changes.
>
> BUG=377062
> R=andrew@webrtc.org, michaelbai@chromium.org
>
> Review URL: https://webrtc-codereview.appspot.com/12579008

TBR=fgalligan@google.com

Review URL: https://webrtc-codereview.appspot.com/12649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6282 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 09:19:03 +00:00
buildbot@webrtc.org
540a2251aa (Auto)update libjingle 68230011-> 68230113 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:40:35 +00:00
pbos@webrtc.org
35efb839ed Implement new-API test RecvStreamWithoutRtx. 
R=pthatcher@google.com, pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/20449005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:40:04 +00:00
pbos@webrtc.org
c34bb3a886 Log default receive stream creation. 
Log when receiving a packet that doesn't have a receiver, this way you
can tell from logs where the AddRecvStream call came from.

R=pthatcher@google.com, pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/17459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6279 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:38:43 +00:00
pbos@webrtc.org
198647473b Implement and fix new-API NackIsEnabled test. 
Required enabling NACK on receiver side which was apparently missed.

BUG=1788
R=pthatcher@google.com, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16499007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6278 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 07:35:47 +00:00
buildbot@webrtc.org
1d66be22c8 (Auto)update libjingle 68203780-> 68206793 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 22:54:24 +00:00
jiayl@webrtc.org
8dcd43c4f7 Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF. 
This is the first step toward switching completely to UDP/TLS/RTP/SAVPF.

BUG=2796
R=juberti@webrtc.org, pthatcher@google.com

Review URL: https://webrtc-codereview.appspot.com/13439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6276 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 22:07:59 +00:00
fischman@webrtc.org
abe01dd634 AppRTCDemo(android): run in full-screen & immersive mode. 
Also:
- Only show stats HUD on demand
- Only collect stats when HUD is showing
- Don't render solid green frame when video is not present in either direction

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6275 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 21:46:52 +00:00
wu@webrtc.org
21a5d449b7 Increase VPMVideoDecimator's initial max_frame_rate_ to 60, which allow us potentially do 60fps. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21499006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6274 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 19:43:26 +00:00
wu@webrtc.org
7a9a3b70b3 * Revert clock.cc changes made in 6178, but keep the changes to the test. 
* Use the new appoach proposed by jib in https://review.webrtc.org/10439004/ to fix the windows clock issue.

BUg=3325

R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6273 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 19:40:28 +00:00
fgalligan@google.com
2a8efa8971 Update generated asm offsets scripts. 
Libvpx updated the unpack scripts to fix building dependencies.

Roll libvpx 269083:273304
See https://codereview.chromium.org/295313002/
https://codereview.chromium.org/298063002/
https://codereview.chromium.org/305533008/
https://codereview.chromium.org/305703002/
https://codereview.chromium.org/298383003/
https://codereview.chromium.org/302863004/
for the libvpx changes.

BUG=377062
R=andrew@webrtc.org, michaelbai@chromium.org

Review URL: https://webrtc-codereview.appspot.com/12579008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6272 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 17:08:34 +00:00
henrike@webrtc.org
caa01b172e Rebase webrtc/base with r6250: 
cd webrtc/base
svn diff -r 6249:6250 http://webrtc.googlecode.com/svn/trunk/talk/base >
6250.diff
patch -p0 -i 6250.diff

BUG=3379
TBR=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6271 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 15:53:39 +00:00
jiayl@webrtc.org
5dc51fbe50 Closes the DataChannel when the send buffer is full or on transport errors. 
As stated in the spec.

BUG=2645
R=pthatcher@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6270 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 15:33:54 +00:00
jiayl@webrtc.org
001fd2d503 Fire OnRenegotiationNeeded only for the first SCTP DataChannel. 
Subsequent DataChannels do not need renegotiation since SCTP data streams are not negotiated through SDP.

BUG=2431
R=pthatcher@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6268 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-29 15:31:11 +00:00