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891 Commits

Author SHA1 Message Date
buildbot@webrtc.org
3c1970f9f3 (Auto)update libjingle 79414100-> 79428003 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7664 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 17:58:41 +00:00
andresp@webrtc.org
188d3b2245 Enable VP9 video codec support on webrtcvideoengine behind a field trial. 
BUG=chromium:431285
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 13:21:04 +00:00
henrik.lundin@webrtc.org
f85dbce041 Reapply "Advertise G722 as 8 kHz rather than 16 kHz"" 
This reverts r7653 and relands r7645. The reason for the original revert was that G722 disappeared from the SDP offer. This is now fixed. Also, a unit test was updated compared with the original change.

BUG=3951
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7662 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 12:25:00 +00:00
perkj@webrtc.org
d105cc81dc Change dummy address to use 0.0.0.0 instead of :: 
This is to not break compatiblity with FF.

https://code.google.com/p/chromium/issues/detail?id=430333

TBR=pthatcher@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7661 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 11:22:06 +00:00
pbos@webrtc.org
a2ef4fe9c3 Prevent a lot of VideoSendStream reconfigures. 
Checking whether we're setting the same configuration or not.
Experimentally this brings down underlying reconfigures from ~20 to
about 4-5.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/30909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7659 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 10:54:43 +00:00
andresp@webrtc.org
82775b1396 Refactor webrtcvideoengines to have the default list of supported codecs being generated in runtime. 
This will allow to plugin VP9 based on a field trial.

R=pbos@webrtc.org, pbos, pthatcher

Review URL: https://webrtc-codereview.appspot.com/27949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7658 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 09:37:54 +00:00
henrika@webrtc.org
5e160660a6 Reland Volume buttons in AppRTCDemo should affect output audio volume (part I). 
Second attempt to land https://webrtc-codereview.appspot.com/32399004/

TBR=perkj@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/30919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7657 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 20:35:13 +00:00
henrik.lundin@webrtc.org
dced5d7835 Revert "Advertise G722 as 8 kHz rather than 16 kHz" 
This reverts r7645.

TBR=pthatcher@webrtc.org
BUG=3951

Review URL: https://webrtc-codereview.appspot.com/24199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7653 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 15:27:43 +00:00
buildbot@webrtc.org
34bda43fa6 (Auto)update libjingle 79326895-> 79329222 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7652 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 12:44:55 +00:00
henrika@webrtc.org
e5421e9602 Volume buttons in AppRTCDemo should affect output audio volume. 
BUG=3279
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7651 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 12:19:19 +00:00
perkj@webrtc.org
fd0efb694a Remove deprecated PeerConnection APIs. 
Removes PeerConnectionObserver::OnError.
Removes MediaConstraints argument to PeerConnection::AddStream.
None of these have ever been implemented and have been removed from the spec.

R=tommi@chromium.org

Review URL: https://webrtc-codereview.appspot.com/24189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7650 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 12:16:36 +00:00
andresp@webrtc.org
19b4741004 Removing unused method GetDefaultVideoEncoderConfig. 
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 11:16:32 +00:00
buildbot@webrtc.org
0ef890a4ba (Auto)update libjingle 79285346-> 79320771 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7647 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 09:22:08 +00:00
mcasas@webrtc.org
6340acde68 AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. 
Also removed some unused "summary" ListPreference
fields.

The looks of it can be found in [1] (lowest row).

[1] https://drive.google.com/file/d/0By6DR2QIwc_ZQm9TMW5YVEpsMWc/view?usp=sharing

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7646 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 09:05:48 +00:00
henrik.lundin@webrtc.org
1dcca4028f Advertise G722 as 8 kHz rather than 16 kHz 
G722 is a 16 kHz (wideband) speech codec, but a "bug" in the RFC
has it listed as 8 kHz. This means that the codec should be
advertised as 8 kHz in SDP messages. This change fixes that.

R=juberti@google.com
TBR=pthatcher@webrtc.org
BUG=3951
TEST=Verify that the G722 is advertised as a=rtpmap:9 G722/8000, not /16000.

Review URL: https://webrtc-codereview.appspot.com/27879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7645 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 08:55:01 +00:00
tkchin@webrtc.org
ee9d61ce45 This fixes a small memory leak (found using Xcode/Instruments on iOS) in 
the ObjC bindings of PeerConnection. The generated session description has
to be released by the recipient

BUG=3985
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28959004

Patch from Matthias Liebig <matthias.gcode@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7636 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-05 22:01:53 +00:00
stefan@webrtc.org
0bae1fab4a Wire up bandwidth stats to the new API and webrtcvideoengine2. 
Adds stats to verify bandwidth and pacer stats.

BUG=1788
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-05 14:05:29 +00:00
buildbot@webrtc.org
a22a628356 (Auto)update libjingle 79205306-> 79244016 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7633 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-05 13:25:48 +00:00
buildbot@webrtc.org
795d003770 (Auto)update libjingle 79200114-> 79205306 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7627 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-05 00:14:02 +00:00
tkchin@webrtc.org
8125744a5f Cleanup RTCVideoRenderer interface. 
RTCVideoRenderer should be a protocol not a class. This change includes
an adapter for use with the C++ apis. The video views have been refactored
to implement that protocol.

BUG=3795
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7626 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 23:06:15 +00:00
buildbot@webrtc.org
45ecf4c092 (Auto)update libjingle 79169148-> 79192489 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7624 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 21:48:54 +00:00
mcasas@webrtc.org
8944c9d08b AppRTCDemoActivity: use differnet Themes for different API levels 
The current AndroidManifest.xml hardcodes a Theme that
is only available in Android L or later (Material). To
maintain backwards compatibility, and for better App
style, create a single Theme/Style and define it for
different APIs.

I tested this in two Nexus %, one with prerelease L
and another with a KK 4.4.2 and the Theme is indeed
automagically updated :)

Note that this is compatible with
https://webrtc-codereview.appspot.com/26979004/

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7619 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 17:26:22 +00:00
pbos@webrtc.org
fad9aecbf5 Remove protected files from talk/PRESUBMIT.py. 
All files may now be committed to.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7616 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 16:06:35 +00:00
pbos@webrtc.org
88ef632286 Falling back on single-stream on multiple SSRC. 
Instead of failing, use one stream. Also clamp video min bitrate.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/31949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7615 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 15:29:29 +00:00
perkj@webrtc.org
b5d045e94d ReAdd PeerConnectionInterface::AddStream to fix Chrome build. 
AddStream(MediaStreamInterface* stream, const MediaConstraintsInterface* constraints);
This will be removed once Chrome has been updated.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7608 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 13:01:33 +00:00
tommi@webrtc.org
18de6f9622 Change the PeerConnection proxy templates to use blocking method calls instead of using Thread::Send. 
The problem with Thread::Send is that it processes incoming pending messages and for the proxies,
this can mean that multiple incoming calls can concurrently run on the same thread, resulting in unexpected behavior.

See e.g. crbug.com/429740 (and more)

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7607 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 12:08:48 +00:00
perkj@webrtc.org
c2dd5ee2c0 Prepare for removal of PeerConnectionObserver::OnError. 
Prepare for removal of constraints to PeerConnection::AddStream.

OnError has never been implemented and has been removed from the spec.
Also, constraints to PeerConnection::AddStream has also been removed from the spec and have never been implemented.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7605 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 11:31:29 +00:00
buildbot@webrtc.org
a663d90ae3 (Auto)update libjingle 79104430-> 79104922 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7602 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 22:29:18 +00:00
glaznev@webrtc.org
5f38c8d1b8 Android AppRTCDemo improvements: 
- Add a room list to ConnectActivity with buttons to add/remove rooms.
- Add loopback call button.
- Add option to toggle full screen / letterbox video.
- Add camera fps settings.
- Fix device to landscape orientation for HD video until issue 3936
will be fixed.
- Fix a few crashes by avoiding calling peer connection and
GAE signaling function while connection is closing.
- Better handling GAE channel error - catch channel exceptions and
display dialog with error messages.

BUG=3939, 3935
R=kjellander@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7601 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 22:18:52 +00:00
pbos@webrtc.org
96a93259b3 Implement external decoder support in WebRtcVideoEngine2. 
R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/30839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7594 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 14:46:44 +00:00
henrik.lundin@webrtc.org
2236267b5e Disable PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate under MSan 
This test is flaky on MSan bots.

BUG=3980
TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7591 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 13:38:50 +00:00
kjellander@webrtc.org
5072e0f6cd Update Android projects to API level 21. 
The update in https://webrtc-codereview.appspot.com/23309004
was not enough, so this updates to 21 instead.

This is required in order to roll chromium_revision to
keep up with Chrome, as third_party/android_tools have now
dropped support for API level 20.

Commands used:
third_party/android_tools/sdk/tools/android update project --name OpenSlDemo --target android-21 --path webrtc/examples/android/opensl_loopback
third_party/android_tools/sdk/tools/android update project --name WebRTCDemo --target android-21 --path webrtc/examples/android/media_demo/
third_party/android_tools/sdk/tools/android update project --name AppRTCDemo --target android-21 --path talk/examples/android/
Then I restored the changes of the ANDROID_SDK_ROOT -> ANDROID_HOME since it seems the Chromium build toolchain doesn't set it properly when
build/android/envsetup.sh is sourced.

BUG=
R=glaznev@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7587 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 23:26:10 +00:00
kjellander@webrtc.org
c2c94a9a9f Change default JVM location to /usr/lib/jvm/java-7-openjdk-amd64 
Given that OpenJDK 1.7 is the recommended Java SDK for
Chromium these days, we should get rid of linking to the old
non-standardized link referring to a Sun Java 1.6 SDK.

Instead of requiring all users to set JAVA_HOME, I prefer
have the most common path as default and and close webrtc:2113
as won't fix after this is submitted.

BUG=2113
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7584 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 19:01:41 +00:00
kjellander@webrtc.org
78c222bfae Update all .isolate files for the new format. 
R=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27809004

Patch from Marc-Antoine Ruel <maruel@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 18:08:09 +00:00
kjellander@webrtc.org
8a130c1084 Update Android projects to API level 20. 
This is required in order to roll chromium_revision to
keep up with Chrome, as third_party/android_tools have now
dropped support for API level 19.

Commands used:
third_party/android_tools/sdk/tools/android update project --name OpenSlDemo --target android-20 --path webrtc/examples/android/opensl_loopback
third_party/android_tools/sdk/tools/android update project --name WebRTCDemo --target android-20 --path webrtc/examples/android/media_demo/
third_party/android_tools/sdk/tools/android update project --name AppRTCDemo --target android-20 --path talk/examples/android/
Then I restored the changes of the ANDROID_SDK_ROOT -> ANDROID_HOME since it seems the Chromium build toolchain doesn't set it properly when
build/android/envsetup.sh is sourced.

BUG=
R=glaznev@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7582 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 17:13:37 +00:00
pbos@webrtc.org
b7ed7799e7 Implement conference-mode temporal-layer screencast. 
Renames VideoStream::temporal_layers to temporal_layer_thresholds_bps to
convey that it contains thresholds needed to ramp up between them (1
threshold -> 2 temporal layers, etc.).

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788,1667

Review URL: https://webrtc-codereview.appspot.com/23269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7578 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 13:08:10 +00:00
pbos@webrtc.org
3bf3d238c8 Configure A/V sync in WebRtcVideoEngine2. 
Sets up A/V sync for the first video receive channel with the default
voice channel. This is only done when conference mode is disabled to
preserve existing behavior. Ideally we'd know which voice channel to
sync with here.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/23249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7577 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 12:59:34 +00:00
minyue@webrtc.org
2dc6f3154d Adapting bitrate according to maxplaybackrate for Opus. 
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 05:33:10 +00:00
tkchin@webrtc.org
14146e40aa arm64 iOS build. 
Allows successful build of arm64 libraries using
GYP_DEFINES="OS=ios target_arch=arm64 target_subarch=arm64".
Note that not all libraries will be NEON optimized (eg common_audio),
however most importantly libvpx will be. WEBRTC_ARCH_ARM needs to be
defined so that libvpx doesn't post-process, which is significantly
detrimental to performance.

BUG=3898
R=kjellander@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7573 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 00:14:39 +00:00
jiayl@webrtc.org
50ca986bc1 Improve the logging when a TCP connection is deleted. 
BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7572 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 23:50:54 +00:00
minyue@webrtc.org
8219529b98 Cleaning up r7562-7567. 
Wrongly used git svn dcommit for committing a CL.

Then two reverts were applied.

Still something needs to be cleaned.

BUG=

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7568 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 08:23:54 +00:00
buildbot@webrtc.org
879fac81d1 (Auto)update libjingle 78822708-> 78823675 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7567 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:50:13 +00:00
minyue@webrtc.org
5f73a37597 Revert 7563 "before rebase" due to wrong submission 
> before rebase

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7566 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:49:58 +00:00
minyue@webrtc.org
c11cc8d947 Revert 7564 "to submit" due to wrong submission 
> to submit

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7565 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:46:47 +00:00
minyue@webrtc.org
de386bf67b to submit 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7564 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:20:09 +00:00
minyue@webrtc.org
c673bb9f29 before rebase 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7563 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:19:57 +00:00
minyue@webrtc.org
0b62672576 adding default rates 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7562 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:19:49 +00:00
pbos@webrtc.org
776e6f289c Use external VideoDecoders in VideoReceiveStream. 
Removes direct VideoCodec use from the new API, exposes VideoDecoders
through webrtc/video_decoder.h similar to VideoEncoders.

Also includes some preparation for wiring up external decoders in
WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they
were allocated internally or externally.

Additionally addresses a data race in VideoReceiver that was exposed with this change.

R=mflodman@webrtc.org, stefan@webrtc.org
TBR=pthatcher@webrtc.org
BUG=2854,1667

Review URL: https://webrtc-codereview.appspot.com/27829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7560 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 15:28:39 +00:00
buildbot@webrtc.org
1abc146aa5 (Auto)update libjingle 78738075-> 78738103 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7554 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 08:14:14 +00:00
perkj@webrtc.org
7998089789 ApprtDemo Android: Switch between front and back camera. 
This adds a UI icon for switching between the front and back camera.
This cl adds the possibility to change between the front and back camera while in a call
or before the other end have connected.

BUG=3786
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7553 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 08:10:03 +00:00