Commit Graph

628 Commits

Author SHA1 Message Date
Alex Glaznev
faa6d076b7 Remove a few verbose log messages from webrtcvideoengine2.
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49189004

Cr-Commit-Position: refs/heads/master@{#9105}
2015-04-28 16:40:51 +00:00
Erik Språng
143cec1cc6 Set correct encoder-specific settings for vpx in the new API.
Also, make VideoEncoderConfig::ContentType an enum class.

BUG=4569
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46069004

Cr-Commit-Position: refs/heads/master@{#9093}
2015-04-28 08:01:14 +00:00
Peter Boström
c4188fd3c7 Use IncomingVideoStream in VideoReceiveStream.
Decouples VideoReceiveStream further from webrtc/video_engine/ as well
as most of webrtc/modules/video_render/ resulting in a simpler setup.

BUG=1695
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50749004

Cr-Commit-Position: refs/heads/master@{#9080}
2015-04-24 13:15:40 +00:00
Henrik Kjellander
24d4485614 Enable -Wunused-private-field warning for talk/
BUG=4242
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49139004

Cr-Commit-Position: refs/heads/master@{#9069}
2015-04-23 12:50:59 +00:00
Peter Boström
ee0b00e8a9 Prevent recv-stream reconfig on identical codecs.
Receive streams seem to be reconfigured with identical codecs when
another stream is removed. Preventing this reconfiguration makes sure
that existing streams don't report stats during teardown when the stream
is still supposed to be running.

BUG=1788
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44249004

Cr-Commit-Position: refs/heads/master@{#9059}
2015-04-22 16:40:58 +00:00
Fredrik Solenberg
b67288283a Move cricket::FakeCall and associates to a separate file.
BUG=4574
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49129004

Cr-Commit-Position: refs/heads/master@{#9057}
2015-04-22 13:34:57 +00:00
Peter Boström
393347ff98 Report receive-side packet loss.
BUG=4558
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48219004

Cr-Commit-Position: refs/heads/master@{#9054}
2015-04-22 12:52:31 +00:00
Henrik Kjellander
7c027b64ae Enable more Clang warnings for talk/
BUG=4242
R=andresp@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46999004

Cr-Commit-Position: refs/heads/master@{#9053}
2015-04-22 11:21:10 +00:00
Shao Changbin
e62202fedf Support handling multiple RTX but only generate SDP with RTX associated with VP8.
This implementation registers RTX-APT map inside RTP sender and receiver.
While it only generates SDP with RTX associated with VP8 to make it
compatible with previous Chrome versions.

Should add following changes after reaches stable,
* Use RTX-APT map for building and restoring RTP packets.
* Add RTX support for RED or VP9 in Video engine.
* Set RTX payload type for RED inside FecConfig in EndToEndTest.

BUG=4024
R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36889004

Cr-Commit-Position: refs/heads/master@{#9040}
2015-04-21 12:25:42 +00:00
Karl Wiberg
9478437fde rtc::Buffer improvements
1. Constructors, SetData(), and AppendData() now accept uint8_t*,
     int8_t*, and char*. Previously, they accepted void*, meaning that
     any kind of pointer was accepted. I think requiring an explicit
     cast in cases where the input array isn't already of a byte-sized
     type is a better compromise between convenience and safety.

  2. data() can now return a uint8_t* instead of a char*, which seems
     more appropriate for a byte array, and is harder to mix up with
     zero-terminated C strings. data<int8_t>() is also available so
     that callers that want that type instead won't have to cast, as
     is data<char>() (which remains the default until all existing
     callers have been fixed).

  3. Constructors, SetData(), and AppendData() now accept arrays
     natively, not just decayed to pointers. The advantage of this is
     that callers don't have to pass the size separately.

  4. There are new constructors that allow setting size and capacity
     without initializing the array. Previously, this had to be done
     separately after construction.

  5. Instead of TransferTo(), Buffer now supports swap(), and move
     construction and assignment, and has a Pass() method that works
     just like std::move(). (The Pass method is modeled after
     scoped_ptr::Pass().)

R=jmarusic@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42989004

Cr-Commit-Position: refs/heads/master@{#9033}
2015-04-20 12:03:00 +00:00
Åsa Persson
352b2d7a19 Fix for sent/received RTCP packet counters returned by GetRtcpPacketTypeCounters. The returned counters are incorrect: sent_packets returns stats from a sent stream (and received_packets returns stats from a receive stream).
Add separate functions for returning stats from send/receive stream and updated how functions are used.

Add test implementation for histogram methods in system_wrappers/interface/metrics.h.

BUG=4519
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49639004

Cr-Commit-Position: refs/heads/master@{#9009}
2015-04-15 16:00:37 +00:00
Magnus Jedvert
4b76c02362 Roll chromium_revision 8af41b3..dcb0929 (324854:325030)
This is a major libyuv update (almost 200 revisions):
d204db6..32ad6e0

Relevant changes:
* src/third_party/libyuv: d204db6..32ad6e0
* src/third_party/nss: d1edb68..9506806
Details: 8af41b3..dcb0929/DEPS

Since bayer and Q420 format support have been removed from libyuv, all tests related to those format are removed.

Clang version was not updated in this roll.

R=kjellander@webrtc.org
TBR=tommi

Review URL: https://webrtc-codereview.appspot.com/48989004

Cr-Commit-Position: refs/heads/master@{#9008}
2015-04-15 15:22:19 +00:00
Peter Boström
3c3f646064 Prevent null-stream reconfigs on RTP extensions.
If a codec fails to set (e.g. there's no codec configured), this
prevents a stream reconfigure with an invalid config. Reconfiguring a
stream without correct codec settings causes a CHECK failure.

BUG=chromium:475116
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44219004

Cr-Commit-Position: refs/heads/master@{#9007}
2015-04-15 14:27:39 +00:00
Peter Boström
e432800aeb Enable CPU adaptation by default.
WebRtcVideoEngine2 doesn't support CPU-monitor-based adaptation and as
such requires encoder-time-based CPU adaptation to perform any
adaptation at all.

BUG=4536
R=asapersson@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49679004

Cr-Commit-Position: refs/heads/master@{#9001}
2015-04-14 20:45:23 +00:00
Peter Thatcher
56d50288e0 Remove SignalCaptureStateChange from MediaEngine.
It's no longer used by anything.

R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/48069004

Cr-Commit-Position: refs/heads/master@{#8994}
2015-04-14 00:17:36 +00:00
Peter Thatcher
77f0e3f7b6 Remove GetStartCaptureFormat and some related code.
It is no longer used by anything.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48039004

Cr-Commit-Position: refs/heads/master@{#8990}
2015-04-13 17:44:56 +00:00
Peter Boström
e7b221f476 Remove deadlock in WebRtcVideoEngine2.
Acquiring stream_lock_ in WebRtcVideoChannel2 in a callback from Call
forms a lock-order inversion between process-thread locks and libjingle
locks, manifesting as CPU adaptation requests blocking on stream
creation that is blocked on the CPU adaptation request finishing.

R=asapersson@webrtc.org, mflodman@webrtc.org
BUG=4535,chromium:475065

Review URL: https://webrtc-codereview.appspot.com/50679004

Cr-Commit-Position: refs/heads/master@{#8985}
2015-04-13 13:34:32 +00:00
Noah Richards
99c2fe5d2b Fix NullVideoEngine's CreateChannel implementation.
BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44149004

Cr-Commit-Position: refs/heads/master@{#8980}
2015-04-10 21:32:42 +00:00
Thiago Farina
9bfe3daf73 Cleanup: Remove i420_video_frame.h header.
It is just a pass through to webrtc/video_frame.h. Updated the callers
to include webrtc/video_frame.h instead and removed i420_video_frame.h.

This should fix pbos' TODO in i420_video_frame.h.

Tested on Linux with the following command lines:

$ rm -rf out/
$ ./webrtc/build/gyp_webrtc
$ ninja -C out/Debug

BUG=None
TEST=see above
R=magjed@webrtc.org, pbos@webrtc.org, tommi@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46819004

Patch from Thiago Farina <tfarina@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#8973}
2015-04-10 10:52:15 +00:00
Magnus Jedvert
f6c003eda5 cricket::VideoFrameFactory: Handle if created frame is null
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46869004

Cr-Commit-Position: refs/heads/master@{#8972}
2015-04-10 10:44:51 +00:00
Magnus Jedvert
0184057d54 VideoAdapterTest: Replace FileVideoCapturer with FakeVideoCapturer
The unittests are currently flaky due to the use of FileVideoCapturer.

BUG=4317
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49649004

Cr-Commit-Position: refs/heads/master@{#8969}
2015-04-10 09:18:39 +00:00
Peter Boström
76c53d36bc Remove ViE interface usage from VideoReceiveStream.
References channels and underlying objects directly instead of using
interfaces referenced with channel id. Channel creation is still done as
before for now.

BUG=1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46849004

Cr-Commit-Position: refs/heads/master@{#8958}
2015-04-09 12:35:46 +00:00
Peter Boström
15cf019a00 Add field-trial flag to disable WebRtcVideoEngine2.
BUG=chromium:475164
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45059004

Cr-Commit-Position: refs/heads/master@{#8957}
2015-04-09 11:55:47 +00:00
Per
9b3f56ea05 Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.""
This reverts commit e41d774c4d.

Original code review: https://webrtc-codereview.appspot.com/43999004/
Reason for reland: There was nothing wrong with this cl as is, but it breaks chrome compatibility. We will now reland this and fix Chrome during roll.

Patset 1: Original cl.
Patchset 2: Removed more code that is no longer needed.

R=magjed@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org

BUG=1128

Review URL: https://webrtc-codereview.appspot.com/45049004

Cr-Commit-Position: refs/heads/master@{#8956}
2015-04-09 11:44:19 +00:00
Peter Boström
ad1f9b61a3 Remove warning on input frames before config.
Removes log spam for AppRTC when only one client is connected.

BUG=4512
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48019005

Cr-Commit-Position: refs/heads/master@{#8947}
2015-04-08 12:04:06 +00:00
Per
e41d774c4d Revert "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection."
This reverts commit 75db861258.

Revert "Fix build breakage in WrappedI420Buffer::native_handle()"

This reverts commit 3211934ebf.

Reason for revert: Breaks chrome build and tests on clank, See https://codereview.chromium.org/1067803002/

BUG=1128
TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43079004

Cr-Commit-Position: refs/heads/master@{#8940}
2015-04-07 15:20:56 +00:00
Bjorn Volcker
1d83f1e89f talk/media/webrtc/webrtcvoiceengine: Delay Agnostic AEC should not override HW-AEC
In https://webrtc-codereview.appspot.com/48699004/ I made the audio option delay_agnostic_aec override HW-AEC if such exists. That is not an expected behavior and is fixed in this CL.

In addition we now check if EnableBuiltInAEC() was successful before disabling the SW-AEC. This revealed a bug in that return value, also fixed here.

BUG=4472
R=henrika@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47969004

Cr-Commit-Position: refs/heads/master@{#8936}
2015-04-07 13:25:52 +00:00
Per
75db861258 Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.
BUG=1128
R=magjed@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43999004

Cr-Commit-Position: refs/heads/master@{#8932}
2015-04-07 10:50:49 +00:00
Thiago Farina
ef88309a6e Cleanup: Forward declare AudioFrame type in voiceprocess.h
No need to include this header since the API is just taking a pointer to
it.

BUG=1092
TEST=./webrtc/build/gyp_webrtc && ninja -C out/Debug
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44059004

Patch from Thiago Farina <tfarina@chromium.org>.

Cr-Commit-Position: refs/heads/master@{#8928}
2015-04-06 10:36:41 +00:00
Henrik Boström
037bad7497 ~CaptureManager: DCHECK(capture_states_.empty()) instead of CHECK until we fix not empty bug.
BUG=chromium:320200
R=perkj@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49579004

Cr-Commit-Position: refs/heads/master@{#8922}
2015-04-02 10:10:18 +00:00
Guo-wei Shieh
64c1e8cda5 Enable CVO by default through webrtc pipeline.
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.

Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome

BUG=4145
R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org

Committed: https://crrev.com/1b1c15cad16de57053bb6aa8a916079e0534bdae
Cr-Commit-Position: refs/heads/master@{#8905}

Review URL: https://webrtc-codereview.appspot.com/47399004

Cr-Commit-Position: refs/heads/master@{#8917}
2015-04-01 22:33:15 +00:00
Minyue
31331cfd2d Revert "Enable CVO by default through webrtc pipeline."
This reverts commit 1b1c15cad1.

Due to failure on
http://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/4092
and following builds (the test hangs and never finishes).
R=kjellander@webrtc.org
TBR=guoweis@chromium.org
TESTED=Local revert + execution of libjingle_peerconnection_java_unittest show that this is the culprit.

Review URL: https://webrtc-codereview.appspot.com/47909004

Cr-Commit-Position: refs/heads/master@{#8911}
2015-04-01 14:20:11 +00:00
Guo-wei Shieh
1b1c15cad1 Enable CVO by default through webrtc pipeline.
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined.

Tests completed:
1. android standalone to android standalone
2. android standalone to chrome (with and without this change)
3. android on chrome

BUG=4145
R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47399004

Cr-Commit-Position: refs/heads/master@{#8905}
2015-04-01 02:42:50 +00:00
Magnus Jedvert
379069f676 VideoRenderCallback::RenderFrame: Make I420VideoFrame& ref const.
RenderFrame should not modify the I420VideoFrame (and we don't).

This CL changes the declaration of RenderFrame from:
int32_t RenderFrame(const uint32_t streamId, I420VideoFrame& videoFrame)
to:
int32_t RenderFrame(const uint32_t streamId, const I420VideoFrame& videoFrame)

BUG=1128
R=mflodman@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46689005

Cr-Commit-Position: refs/heads/master@{#8902}
2015-03-31 17:52:37 +00:00
Peter Boström
23914fe756 Reject RTP one-byte extension ID 0.
Only accept local identifiers in the range 1-14 inclusive.

BUG=1788, chromium:471328
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50549004

Cr-Commit-Position: refs/heads/master@{#8900}
2015-03-31 13:08:13 +00:00
Magnus Jedvert
1ecfd55044 videoadapter_unittest.cc: Revert removal of '#if defined(HAVE_WEBRTC_VIDEO)'
This CL reverts some parts of "Delete VideoAdapter::AdaptFrame" https://webrtc-codereview.appspot.com/44769004/.

Reason for revert: Should not touch HAVE_WEBRTC_VIDEO since libjingle_media_unittests does not compile without anyway.

BUG=4317
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48699005

Cr-Commit-Position: refs/heads/master@{#8888}
2015-03-30 09:25:04 +00:00
Peter Boström
dfd53fe26b Raise streams for SetMaxSendBitrates above 2000k.
Fixes b=AS effectively not setting bitrates above 2000k.

BUG=1788,4469
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47839004

Cr-Commit-Position: refs/heads/master@{#8882}
2015-03-27 14:58:17 +00:00
Peter Boström
53eda3dbd0 Add tests for r8811.
All these tests crashed before r8811. These tests should've been with
that change but r8811 was pushed in before to make bots green.

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

Review URL: https://webrtc-codereview.appspot.com/48669004

Cr-Commit-Position: refs/heads/master@{#8881}
2015-03-27 14:53:30 +00:00
Peter Boström
8ed6a4bba4 Remove unused non-standard capture stats.
Removes 'googCaptureJitterMs' and 'googCaptureQueueDelayMsPerS' from
talk/. The overuse-detection method used is based on encoding time,
so these stats aren't useful enough to warrant having them showing up in
GetStats().

BUG=
R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50469004

Cr-Commit-Position: refs/heads/master@{#8874}
2015-03-27 09:01:11 +00:00
Magnus Jedvert
3954e1dfe1 Remove unused implementations in cricket::VideoFrame
This CL moves dummy implementations from cricket::VideoFrame to NullVideoFrame instead.

R=guoweis@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50409004

Cr-Commit-Position: refs/heads/master@{#8873}
2015-03-27 08:48:45 +00:00
Minyue Li
7100dcd317 Adding "usedtx" as Opus codec parameter.
This is according to https://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03

Specifically,

usedtx: specifies if the decoder prefers the use of DTX. values are 1 and 0. If no value is specified, usedtx is assumed to be 0.

BUG=1014
R=juberti@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48499004

Cr-Commit-Position: refs/heads/master@{#8872}
2015-03-27 04:06:35 +00:00
Peter Boström
74d9ed7d85 Report send codec name in GetStats().
BUG=4461
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51439004

Cr-Commit-Position: refs/heads/master@{#8869}
2015-03-26 15:28:43 +00:00
Peter Boström
d6f4c25eed Reject streams reusing simulcast or RTX SSRCs.
BUG=1788, chromium:470122, chromium:470856
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42919004

Cr-Commit-Position: refs/heads/master@{#8868}
2015-03-26 15:23:13 +00:00
Stefan Holmer
e590416722 Moving the pacer and the pacer thread to ChannelGroup.
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out.

BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45549004

Cr-Commit-Position: refs/heads/master@{#8864}
2015-03-26 10:11:22 +00:00
Brave Yao
5225dd8180 If audio ptime is negotiated in SDP, then we would set the audio codec with negotiated packet size if it's allowed. If the negotiated packet size is not supported by the working codec, then we would use the next smallest size.
BUG=4289
TEST=Manual/Auto Test
R=juberti@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44629004

Cr-Commit-Position: refs/heads/master@{#8863}
2015-03-25 23:39:33 +00:00
Bjorn Volcker
bf395c1fc0 Add WebRTC Media Constraint to force using Delay Agnostic AEC on Android
If built-in Echo Cancellation is available on a device it is automatically enabled. The reason is that it in most cases performs better than the WebRTC software echo control for mobile. The drawback is that we can not develop, test and rollout the delay agnostic AEC (DA-AEC) on Android as for desktops.

This CL includes
- adding a media constraint to enable/disable DA-AEC.
- automatically turning on echo cancellation if DA-AEC is enabled.
- a fix in the AEC that enables delay estimation when DA-AEC is enabled, but delay metrics is disabled.
- sets the Config struct ReportedDelay, which controls DA-AEC internally in the AEC.

The test code to verify that it works in AppRTCDemo can be found here:
https://webrtc-codereview.appspot.com/50479004/

BUG=4472
TESTED=locally on N7, N6, Android One
R=glaznev@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48699004

Cr-Commit-Position: refs/heads/master@{#8861}
2015-03-25 21:46:10 +00:00
Peter Boström
d4362cd336 Reject StreamParams with RTX SSRCs not in ssrcs.
BUG=1788, chromium:470122
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44859004

Cr-Commit-Position: refs/heads/master@{#8855}
2015-03-25 13:17:33 +00:00
Magnus Jedvert
e61c64dbb1 Delete NullVideoRenderer
NullVideoRenderer is not used.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51419004

Cr-Commit-Position: refs/heads/master@{#8850}
2015-03-24 15:11:24 +00:00
Niklas Enbom
07a4ba5d1a Simulcast settings for 1080p. Using same bit rates for all 3 modes since only one is used in reality, and the plan is to unify them.
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45779004

Cr-Commit-Position: refs/heads/master@{#8849}
2015-03-24 14:48:03 +00:00
Magnus Jedvert
ac27e20477 Delete VideoAdapter::AdaptFrame
This CL deletes VideoAdapter::AdaptFrame and replaces the remaining calls with AdaptFrameResolution instead.

I do not expect this CL to fix the flaky VideoAdapterTests yet. I intend to replace FileVideoCapturer with a deterministic FakeVideoCapturer in a follow-up CL.

BUG=4317
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44769004

Cr-Commit-Position: refs/heads/master@{#8848}
2015-03-24 14:18:52 +00:00