Commit Graph

10120 Commits

Author SHA1 Message Date
4a41361f98 Android SurfaceViewRenderer: Never hold a pending frame indefinitely
The original purpose with keeping one pending frame in SurfaceViewRenderer was to reduce latency for the first rendered frame when we are waiting for the Surface to be created. However, it is very dangerous to hold a pending frame indefinitely when used with a SurfaceTexture, because the SurfaceTexture only has one frame and thus holding a frame in the renderer will freeze everything and typically cause timeout crashes.

Review URL: https://codereview.webrtc.org/1435413006

Cr-Commit-Position: refs/heads/master@{#10638}
2015-11-13 16:48:06 +00:00
Per
c01c25434b Revert of Android MediaCodecVideoDecoder: Manage lifetime of texture frames (patchset #12 id:320001 of https://codereview.webrtc.org/1422963003/ )
Reason for revert:
Causes fallback to SW decoder if a renderer is put in the background.

Original issue's description:
> Patchset 1 is a pure
> revert of "Revert of "Android MediaCodecVideoDecoder: Manage lifetime of texture frames" https://codereview.webrtc.org/1378033003/
>
> Following patchsets move the responsibility of calculating the decode time to Java.
>
> TESTED= Apprtc loopback using H264 and VP8 on N5, N6, N7, S5
>
> Committed: https://crrev.com/9cb8982e64f08d3d630bf7c3d2bcc78c10db88e2
> Cr-Commit-Position: refs/heads/master@{#10597}

TBR=magjed@webrtc.org,glaznev@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true

Review URL: https://codereview.webrtc.org/1441363002 .

Cr-Commit-Position: refs/heads/master@{#10637}
2015-11-13 15:58:37 +00:00
f8506cbdd8 rtcp::Ij renamed to rtcp::ExtendedJitterReport
to match name given in the RFC5450
  private member renamed to inter_arrival_jitters_ for the same reason.
rtcp::ExtendedJitterReport moved into own file
accessors and Parse function added
  to make class usable for parsing packet

Review URL: https://codereview.webrtc.org/1434213004

Cr-Commit-Position: refs/heads/master@{#10636}
2015-11-13 15:33:26 +00:00
cbe9f51cf8 Revert of Remove global list of SRTP sessions. (patchset #4 id:60001 of https://codereview.webrtc.org/1416093010/ )
Reason for revert:
Unfortunately this breaks an internal downstream project since we have an ancient libsrtp. Reverting until we can figure out how to update our libsrtp.

Original issue's description:
> Remove global list of SRTP sessions.
> Instead save a reference to the SrtpSession inside the srtp_ctx_t.
>
> BUG=webrtc:5133
>
> Committed: https://crrev.com/9cafd972779ed7b25886ab276e0ede7b7a8b76a1
> Cr-Commit-Position: refs/heads/master@{#10591}

TBR=juberti@google.com,juberti@webrtc.org,jbauch@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5133

Review URL: https://codereview.webrtc.org/1442863003

Cr-Commit-Position: refs/heads/master@{#10635}
2015-11-13 14:55:17 +00:00
0fa9b22789 Remove scoped_ptrs for VCM sender_ and receiver_.
Put VideoSender/VideoReceiver flat within the object, not as
scoped_ptrs, giving fewer allocations and looking a bit nicer.

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1443613002

Cr-Commit-Position: refs/heads/master@{#10634}
2015-11-13 13:59:59 +00:00
df948f03b3 rtcp::ReportBlock refactored to contain parsing
Review URL: https://codereview.webrtc.org/1420283022

Cr-Commit-Position: refs/heads/master@{#10633}
2015-11-13 11:03:18 +00:00
0a41893e36 Remove BitrateController dependency fromVideoReceiveStream.
I have another CL moving REMB from CongestonController to Call, then
I'll remove CongestoinController from this class too.

R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1442003002 .

Cr-Commit-Position: refs/heads/master@{#10632}
2015-11-13 10:12:16 +00:00
464c0878d9 Rename screenshare test.
Renamed the test to reflect what is actually tested. What the old test
did I don't know since there has never been possible to use screenshare
with two temporal layers in VP9.

BUG=chromium:554515

Review URL: https://codereview.webrtc.org/1441693002

Cr-Commit-Position: refs/heads/master@{#10631}
2015-11-13 10:08:11 +00:00
0e7e259ebd Move BitrateAllocator from BitrateController logic to Call.
This is a step on the way to have variable bitrate for audio and is
intended to be as much of a no-op as possible.

BUG=webrtc:5079

Review URL: https://codereview.webrtc.org/1441673002

Cr-Commit-Position: refs/heads/master@{#10630}
2015-11-13 05:02:46 +00:00
69191ed845 Roll chromium_revision 4771dd5..3b7968d (359351:359482)
Change log: 4771dd5..3b7968d
Full diff: 4771dd5..3b7968d

Changed dependencies:
* src/buildtools: 4a95614..3ba3ca2
* src/third_party/libvpx_new/source/libvpx: 9ecb99a..0941ff7
DEPS diff: 4771dd5..3b7968d/DEPS

No update to Clang.

TBR=marpan@webrtc.org, stefan@webrtc.org,
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1442883002

Cr-Commit-Position: refs/heads/master@{#10629}
2015-11-13 03:57:08 +00:00
faac497af5 Fix for scenario where m-line is revived after being set to port 0.
When this is detected, we'll now "reconfigure" the senders and
receivers, which will reconnect the capturers/renderers to the
underlying streams which have been recreated.

BUG=webrtc:2136

Review URL: https://codereview.webrtc.org/1428243005

Cr-Commit-Position: refs/heads/master@{#10628}
2015-11-12 23:33:14 +00:00
69d0d46c25 Roll chromium_revision e658ee0..4771dd5 (359300:359351)
Change log: e658ee0..4771dd5
Full diff: e658ee0..4771dd5

No dependencies changed.
No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1438993004

Cr-Commit-Position: refs/heads/master@{#10627}
2015-11-12 21:01:19 +00:00
2cd7afe7e2 Do not delete a connection until it has not received anything for 30 seconds.
BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1422623015 .

Cr-Commit-Position: refs/heads/master@{#10626}
2015-11-12 19:14:38 +00:00
8597543ae8 Schedule a CreatePermissionRequest after the success of a previous request
unless a channel binding request is already scheduled.

BUG=5178
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1434603002 .

Cr-Commit-Position: refs/heads/master@{#10625}
2015-11-12 19:07:25 +00:00
68876f990e Introduces Android API level linting, fixes all current API lint errors.
This CL attempts to annotate accesses on >16 API levels using as
small scopes as possible. The TargetApi notations mean "yes, I know
I'm accessing a higher API and I take responsibility for gating the
call on Android API level". The Encoder/Decoder classes are annotated
on the whole class, but they're only accessed through JNI; we should
annotate on method level otherwise and preferably on private methods.

This patch also fixes some compiler-level deprecation warnings (i.e.
-Xlint:deprecation), but probably not all of them.

BUG=webrtc:5063
R=henrika@webrtc.org, kjellander@webrtc.org, magjed@webrtc.org

Review URL: https://codereview.webrtc.org/1412673008 .

Cr-Commit-Position: refs/heads/master@{#10624}
2015-11-12 16:37:01 +00:00
56a34df928 Re-add a thread check in Call::Call that was removed by mistake in a rebase.
BUG=

Review URL: https://codereview.webrtc.org/1434263002

Cr-Commit-Position: refs/heads/master@{#10623}
2015-11-12 16:24:50 +00:00
9576e54836 Reland "Prepare MediaCodecVideoEncoder for surface textures.""
This reverts commit 12f680214e28dc5f0a13ac8afc0d1445f89e67e6.
Original cl in https://codereview.webrtc.org/1396073003/
Prepare MediaCodecVideoEncoder for surface textures.
This refactors MediaVideoEncoder to prepare for adding support to encode from textures. The C++ layer does not have any functional changes.
- Moves ResetEncoder to always work on the codec thread
- Adds use of ThreadChecker.
- Change Java MediaEncoder.Init to return true or false and introduce method getInputBuffers.
- Add simple unit test for Java MediaCodecVideoEncoder.

The pure revert of the revert is in patchset 1.
Patchset 2, moves getting the input buffer to before storing pending timestamps etc to fix b/24984012.

BUG=webrtc:4993 b/24984012

Review URL: https://codereview.webrtc.org/1406203002

Cr-Commit-Position: refs/heads/master@{#10622}
2015-11-12 14:43:22 +00:00
8093d5442e Change default SSRC for RTCP receiver reports to not collide with video.
BUG=chromium:547661
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1438183002

Cr-Commit-Position: refs/heads/master@{#10621}
2015-11-12 14:02:35 +00:00
dfe434e20e Roll chromium_revision b0415d9..e658ee0 (359214:359300)
Change log: b0415d9..e658ee0
Full diff: b0415d9..e658ee0

Changed dependencies:
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/ad38dc7..d7421eb
DEPS diff: b0415d9..e658ee0/DEPS

No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1442603002

Cr-Commit-Position: refs/heads/master@{#10620}
2015-11-12 12:52:42 +00:00
5dda80abea Remove webrtc/modules/video_{capture,render}/include
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=pbos@webrtc.org, perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1439823002 .

Cr-Commit-Position: refs/heads/master@{#10619}
2015-11-12 11:47:02 +00:00
e71b24e421 OpenSL ES stability improvements.
This CL does two things:

1) Improves stability in the existing OpenSL ES implementation for devices that
supports OpenSL ES. The cost is a slight increase in latency since the focus here
has been on avoiding audio glitches.

2) Adds a new Java API to exclude usage of OpenSL ES to enable comparisons between
OpenSL ES and Java based audio backends.

BUG=b/22452539

Review URL: https://codereview.webrtc.org/1440623002

Cr-Commit-Position: refs/heads/master@{#10618}
2015-11-12 09:48:36 +00:00
fc6affc60d Android SurfaceViewRenderer: Call glClear() for every frame to avoid bad GL state
BUG=webrtc:5147

Review URL: https://codereview.webrtc.org/1436883002

Cr-Commit-Position: refs/heads/master@{#10617}
2015-11-12 09:08:37 +00:00
96839648a0 Trivial initialization fix in AudioDeviceIOS
NOTRY=TRUE
TBR=tkchin
BUG=webrtc:5058

Review URL: https://codereview.webrtc.org/1435003002

Cr-Commit-Position: refs/heads/master@{#10616}
2015-11-12 09:01:24 +00:00
31c8167fa8 Roll chromium_revision 7e059f9..b0415d9 (359143:359214)
Change log: 7e059f9..b0415d9
Full diff: 7e059f9..b0415d9

Changed dependencies:
* src/third_party/libvpx_new/source/libvpx: eba14dd..9ecb99a
DEPS diff: 7e059f9..b0415d9/DEPS

No update to Clang.

TBR=marpan@webrtc.org, stefan@webrtc.org,
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1440853003

Cr-Commit-Position: refs/heads/master@{#10615}
2015-11-12 03:57:20 +00:00
a8e9f5e4e4 A little cleanup in p2ptransportchannel and transportchannel.
No functional changes.

BUG=

Review URL: https://codereview.webrtc.org/1411103012

Cr-Commit-Position: refs/heads/master@{#10614}
2015-11-12 00:15:12 +00:00
066ded99cb Relax the stun ping check on valid result.
1. allow situation where all ping is lost
2. use the raw count to calculate the interval.

Since we now send 1 request per IP, the chance of losing all of them is higher and we shouldn't just quit if we don't have any response.

BUG=

Review URL: https://codereview.webrtc.org/1406223011

Cr-Commit-Position: refs/heads/master@{#10613}
2015-11-11 23:04:10 +00:00
33daa7ef85 Roll chromium_revision 4a38519..7e059f9 (359080:359143)
Change log: 4a38519..7e059f9
Full diff: 4a38519..7e059f9

No dependencies changed.
No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1432363002

Cr-Commit-Position: refs/heads/master@{#10612}
2015-11-11 21:53:56 +00:00
6b14f9377d Adjust parameter for VP9 resize unittest.
Needed for upcoming libvpx roll.

TBR=stefan@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1432773005 .

Cr-Commit-Position: refs/heads/master@{#10611}
2015-11-11 21:42:13 +00:00
9b5ee9c0d9 Send back ping response if the ping comes from an unknown address.
BUG=webrtc:5171

Review URL: https://codereview.webrtc.org/1424703012

Cr-Commit-Position: refs/heads/master@{#10610}
2015-11-11 21:19:25 +00:00
653b8e02f2 Reland of Adding the ability to change ICE servers through SetConfiguration. (patchset #1 id:1 of https://codereview.webrtc.org/1424803004/ )
Reason for revert:
Relanding with compile warning fixed.

Original issue's description:
> Revert of Adding the ability to change ICE servers through SetConfiguration. (patchset #7 id:120001 of https://codereview.webrtc.org/1391013007/ )
>
> Reason for revert:
> Caused compiler warning, breaking Chrome FYI bots.
>
> Original issue's description:
> > Adding the ability to change ICE servers through SetConfiguration.
> >
> > Added a SetIceServers method to PortAllocator. Also added a new
> > PeerConnection Initialize method that takes a PortAllocator, in the
> > hope that we can get rid of PortAllocatorFactoryInterface, since the
> > only substantial thing a factory does is convert the webrtc:: ICE
> > servers to cricket:: versions.
> >
> > Committed: https://crrev.com/d3b26d94399ff539db375a9b84010ee75479d4cf
> > Cr-Commit-Position: refs/heads/master@{#10420}
>
> TBR=pthatcher@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/18a944bf0ac9eed872dc009bd58e6bc12c946303
> Cr-Commit-Position: refs/heads/master@{#10421}

TBR=pthatcher@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1414313003

Cr-Commit-Position: refs/heads/master@{#10609}
2015-11-11 20:55:18 +00:00
9b72af94cd Remove webrtc/modules/audio_processing/{aec,aecm,ns}/include
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1440523002 .

Cr-Commit-Position: refs/heads/master@{#10608}
2015-11-11 19:16:28 +00:00
e03cab94c1 When running this code in chromium on a machine with IPv6 disabled, the RTC_DCHECK fails and in release build, it could leak to further crash in chromium's rtc_peer_connection_hanlder.cc.
Here is the right fix.

BUG=webrtc:5061
R=pthatcher@google.com
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1437933002 .

Cr-Commit-Position: refs/heads/master@{#10607}
2015-11-11 19:11:28 +00:00
ee2bac26dd AcmReceiver::InsertPacket and NetEq::InsertPacket: Take ArrayView arguments
Instead of separate pointer and size arguments.

Review URL: https://codereview.webrtc.org/1429943004

Cr-Commit-Position: refs/heads/master@{#10606}
2015-11-11 18:34:07 +00:00
91d926038f Add receive bitrate UMA stats.
Review URL: https://codereview.webrtc.org/1440603002

Cr-Commit-Position: refs/heads/master@{#10605}
2015-11-11 18:13:07 +00:00
4dc941128f CodecManager::RegisterEncoder: Call SetFec on new encoder, not old
BUG=webrtc:5028

Review URL: https://codereview.webrtc.org/1416633011

Cr-Commit-Position: refs/heads/master@{#10604}
2015-11-11 16:34:28 +00:00
718b6c72ae Add waiting to SetSendSsrc tests.
These tests were flaky since a paced packet could arrive between
consecutive calls to NumRtpPackets or NumRtpBytes.

BUG=webrtc:5193
R=stefan@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1436753003 .

Cr-Commit-Position: refs/heads/master@{#10603}
2015-11-11 16:19:40 +00:00
4b56904b70 Fix race in VideoSendStreamTest.RtcpSenderReportContainsMediaBytesSent.
BUG=webrtc:5194

Review URL: https://codereview.webrtc.org/1434963002

Cr-Commit-Position: refs/heads/master@{#10602}
2015-11-11 14:40:03 +00:00
00ac85e2e3 Update temporal up switch field for non-flexible mode according to updates in the RTP payload profile.
The U bit is no longer obtained from the SS data.

https://tools.ietf.org/id/draft-ietf-payload-vp9-01.txt

BUG=chromium:500602

Review URL: https://codereview.webrtc.org/1433273002

Cr-Commit-Position: refs/heads/master@{#10601}
2015-11-11 13:30:55 +00:00
f616a35203 Roll chromium_revision 5a2ae99..4a38519 (359027:359080)
Change log: 5a2ae99..4a38519
Full diff: 5a2ae99..4a38519

No dependencies changed.
No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1433253002

Cr-Commit-Position: refs/heads/master@{#10600}
2015-11-11 12:58:31 +00:00
fa566d610f Remove webrtc/examples/android/media_demo.
The JNI code for VoiceEngine is not maintained and VoiceEngine is being
refactored. This is not a supported Java interface, use AppRTCDemo as a
starting point instead.

Also renames webrtc/libjingle_examples.gyp webrtc/webrtc_examples.gyp to
replace the previous file (that only contained media_demo).

BUG=
R=henrika@webrtc.org, kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1439593002 .

Cr-Commit-Position: refs/heads/master@{#10599}
2015-11-11 12:11:21 +00:00
cbfabbf818 Fix potential tearing issue in VideoRendererGui.
This make sure that the texture copy is syncronized.

To reproduce the problem I:
Reverted "Revert of "Android MediaCodecVideoDecoder: Manage lifetime of texture frames" https://codereview.webrtc.org/1378033003/"
commit 543b6ca30a43eeb069c699291460ce6bacc7959d.
Reverted "Enable SurfaceViewRenderer for AppRTCDemo"
commit 7076729c57c27aa813760d2038be02c36f4d7649.
and ran ApprtDemo in loopback and changed the orientation a couple of times.

TBR=glaznev@webrtc.org

Review URL: https://codereview.webrtc.org/1437823002

Cr-Commit-Position: refs/heads/master@{#10598}
2015-11-11 11:38:39 +00:00
9cb8982e64 Patchset 1 is a pure
revert of "Revert of "Android MediaCodecVideoDecoder: Manage lifetime of texture frames" https://codereview.webrtc.org/1378033003/

Following patchsets move the responsibility of calculating the decode time to Java.

TESTED= Apprtc loopback using H264 and VP8 on N5, N6, N7, S5

Review URL: https://codereview.webrtc.org/1422963003

Cr-Commit-Position: refs/heads/master@{#10597}
2015-11-11 11:27:05 +00:00
b2d1c5026d SurfaceViewRenderer: Add resource name to log outputs and exceptions
Add resource name to log outputs to distinguish local renderer from remote renderer.

This Cl also adds some thread checks and factors out a small helper function makeBlack().

Review URL: https://codereview.webrtc.org/1420203003

Cr-Commit-Position: refs/heads/master@{#10596}
2015-11-11 11:06:49 +00:00
1323fc39ba Remove webrtc/test/channel_transport/include
Move the header file into webrtc/test/channel_transport instead.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel -m tryserver.webrtc --bot=ios_rel
R=henrika@webrtc.org, henrikg@webrtc.org

Review URL: https://codereview.webrtc.org/1431983006 .

Cr-Commit-Position: refs/heads/master@{#10595}
2015-11-11 09:34:35 +00:00
5237aaf243 Convert usage of ARRAY_SIZE to arraysize.
ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.

BUG=None
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1405023016

Cr-Commit-Position: refs/heads/master@{#10594}
2015-11-11 07:44:39 +00:00
e134a534cd Roll chromium_revision 6f156f7..5a2ae99 (358880:359027)
Change log: 6f156f7..5a2ae99
Full diff: 6f156f7..5a2ae99

Changed dependencies:
* src/buildtools: c2f2598..4a95614
* src/third_party/libyuv: 98eb102..6100f50
DEPS diff: 6f156f7..5a2ae99/DEPS

No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1440483002

Cr-Commit-Position: refs/heads/master@{#10593}
2015-11-11 03:58:51 +00:00
ad13d2f817 Round Rate computations from RateTracker.
BUG=534221
R=asapersson@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1410533004 .

Cr-Commit-Position: refs/heads/master@{#10592}
2015-11-11 00:34:58 +00:00
9cafd97277 Remove global list of SRTP sessions.
Instead save a reference to the SrtpSession inside the srtp_ctx_t.

BUG=webrtc:5133

Review URL: https://codereview.webrtc.org/1416093010

Cr-Commit-Position: refs/heads/master@{#10591}
2015-11-10 22:48:46 +00:00
9af97f8910 WebRTC should generate default private address even when adapter enumeration is disabled.
Introduce a DefaultAddressProvider such that rtc::Network can't access other part of NetworkManager.

This also removes the hack of generating the loopback address. The dependency has been removed by https://codereview.chromium.org/1417023003/

BUG=webrtc:5061
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1411253008 .

Cr-Commit-Position: refs/heads/master@{#10590}
2015-11-10 22:47:49 +00:00
542059ef10 Roll chromium_revision bff0bbb..6f156f7 (358822:358880)
Change log: bff0bbb..6f156f7
Full diff: bff0bbb..6f156f7

No dependencies changed.
No update to Clang.

TBR=
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1437633004

Cr-Commit-Position: refs/heads/master@{#10589}
2015-11-10 22:10:24 +00:00