Commit Graph

134 Commits

Author SHA1 Message Date
80ac24dd36 Allow max 1 block per type in RTCP Extended Reports
Design of individual block in ExtendedReports packet suggest there is
no point to have more than one block per type.
This CL reduce complexity of having several blocks of the same type in
same report.

BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2378113002
Cr-Commit-Position: refs/heads/master@{#14855}
2016-10-31 15:40:55 +00:00
cc91d284e4 Moved RtcEventLog files from call/ to logging/
The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.

After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).

The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).

This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.

BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/2380683005 .

Cr-Commit-Position: refs/heads/master@{#14485}
2016-10-04 01:31:32 +00:00
b19d288c94 Remove chain of methods in RtpRtcp module to get current payload frequency for RTCP SRs, which anyway was stuck to defaults for video/audio.
BUG=webrtc:2795,webrtc:6458

Review-Url: https://codereview.webrtc.org/2362373002
Cr-Commit-Position: refs/heads/master@{#14476}
2016-10-03 13:22:32 +00:00
798896a4aa Replace RtcpReceiveTimeInfo with rtcp::ReceiveTimeInfo
structs are exactly the same but last one follow naming style.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2368983002
Cr-Commit-Position: refs/heads/master@{#14415}
2016-09-28 09:54:30 +00:00
822a16f64c Reland of Unify rtcp packet setters (patchset #1 id:1 of https://codereview.webrtc.org/2372713005/ )
Reason for revert:
Fix backward compatibility support

Original issue's description:
> Revert of Unify rtcp packet setters (patchset #8 id:130001 of https://codereview.webrtc.org/2348623003/ )
>
> Reason for revert:
> Breaks compilation of internal downstream project.
>
> Original issue's description:
> > Unify rtcp packet setters
> > Renamed setters in rtcp classes
> > from WithField to SetField
> > from WithItem to AddItem or SetItems
> > from From to SetSenderSsrc
> > from To to SetMediaSsrc
> > Some redundant or unsued setters removed.
> > Pass-by-const& replaced with pass-by-value when appropriate.
> >
> > BUG=webrtc:5260
> >
> > Committed: https://crrev.com/20e77c7b8a9f19942ef3c3c4f1fa3888b2cd54ea
> > Cr-Commit-Position: refs/heads/master@{#14393}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5260
>
> Committed: https://crrev.com/efc6e41866662e0922858fbce1d9ee3bdd0637ed
> Cr-Commit-Position: refs/heads/master@{#14400}

TBR=sprang@webrtc.org,stefan@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2370313002
Cr-Commit-Position: refs/heads/master@{#14402}
2016-09-27 16:27:52 +00:00
efc6e41866 Revert of Unify rtcp packet setters (patchset #8 id:130001 of https://codereview.webrtc.org/2348623003/ )
Reason for revert:
Breaks compilation of internal downstream project.

Original issue's description:
> Unify rtcp packet setters
> Renamed setters in rtcp classes
> from WithField to SetField
> from WithItem to AddItem or SetItems
> from From to SetSenderSsrc
> from To to SetMediaSsrc
> Some redundant or unsued setters removed.
> Pass-by-const& replaced with pass-by-value when appropriate.
>
> BUG=webrtc:5260
>
> Committed: https://crrev.com/20e77c7b8a9f19942ef3c3c4f1fa3888b2cd54ea
> Cr-Commit-Position: refs/heads/master@{#14393}

TBR=sprang@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2372713005
Cr-Commit-Position: refs/heads/master@{#14400}
2016-09-27 15:39:39 +00:00
20e77c7b8a Unify rtcp packet setters
Renamed setters in rtcp classes
from WithField to SetField
from WithItem to AddItem or SetItems
from From to SetSenderSsrc
from To to SetMediaSsrc
Some redundant or unsued setters removed.
Pass-by-const& replaced with pass-by-value when appropriate.

BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2348623003
Cr-Commit-Position: refs/heads/master@{#14393}
2016-09-27 08:37:51 +00:00
853ecb21f7 Style cleanup in UpdateTmmbr:
function names style updated,
unused return type removed.
Comment style fixed, redundant comments removed.
pass-by-pointer parameter changed to pass-by-value because can't be nullptr any more.

NOTRY=true
BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2258523005
Cr-Commit-Position: refs/heads/master@{#13848}
2016-08-22 15:26:22 +00:00
2b616397de Remove TMMBRSet class
by cleaning RTCPReceiveInfo class
and following cleaning of RTCPReceiver::BoundingSet function.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2254703003
Cr-Commit-Position: refs/heads/master@{#13817}
2016-08-18 13:17:48 +00:00
71fead2146 Reland of StartTimestamp generated randomly in RtpSender constructor (patchset #1 id:1 of https://codereview.webrtc.org/2248413002/ )
Reason for revert:
Reland: downstream code expectation about rtp_sender timestamp adjusted.

Original issue's description:
> Revert of StartTimestamp generated randomly in RtpSender constructor (patchset #4 id:60001 of https://codereview.webrtc.org/2241193002/ )
>
> Reason for revert:
> Breaks downstream code.
>
> Original issue's description:
> > StartTimestamp generated randomly in RtpSender constructor
> > instead of not-randomly at SetSendingState(true)
> > Renamed to timestamp_offset_ to better match meaning of the variable.
> >
> > R=asapersson@webrtc.org, terelius@webrtc.org
> >
> > Committed: https://crrev.com/4466782ae43e1b1125a55ee7e18abd10dd37cede
> > Cr-Commit-Position: refs/heads/master@{#13796}
>
> TBR=asapersson@webrtc.org,terelius@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/86c96948e340cf8b879bddb0c7293f3b5ad4dad4
> Cr-Commit-Position: refs/heads/master@{#13798}

TBR=asapersson@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2257083002
Cr-Commit-Position: refs/heads/master@{#13811}
2016-08-18 09:02:16 +00:00
86c96948e3 Revert of StartTimestamp generated randomly in RtpSender constructor (patchset #4 id:60001 of https://codereview.webrtc.org/2241193002/ )
Reason for revert:
Breaks downstream code.

Original issue's description:
> StartTimestamp generated randomly in RtpSender constructor
> instead of not-randomly at SetSendingState(true)
> Renamed to timestamp_offset_ to better match meaning of the variable.
>
> R=asapersson@webrtc.org, terelius@webrtc.org
>
> Committed: https://crrev.com/4466782ae43e1b1125a55ee7e18abd10dd37cede
> Cr-Commit-Position: refs/heads/master@{#13796}

TBR=asapersson@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review-Url: https://codereview.webrtc.org/2248413002
Cr-Commit-Position: refs/heads/master@{#13798}
2016-08-17 15:12:27 +00:00
4466782ae4 StartTimestamp generated randomly in RtpSender constructor
instead of not-randomly at SetSendingState(true)
Renamed to timestamp_offset_ to better match meaning of the variable.

R=asapersson@webrtc.org, terelius@webrtc.org

Review URL: https://codereview.webrtc.org/2241193002 .

Cr-Commit-Position: refs/heads/master@{#13796}
2016-08-17 13:07:49 +00:00
2f69ce9498 Cleaned out candidateSet member from TMMBRHelp class
leaving that class memberless.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2234783002
Cr-Commit-Position: refs/heads/master@{#13776}
2016-08-16 10:21:44 +00:00
daa90a7e35 Cleaned out boundingSet member from TMMBRHelp class
BUG=webrtc:5565
R=philipel@webrtc.org

Review URL: https://codereview.webrtc.org/2224933002 .

Cr-Commit-Position: refs/heads/master@{#13702}
2016-08-10 09:30:08 +00:00
70ffead256 Reimplemented fix for bogus RTP timestamp in RTCP packet created before RTP packet.
Now it check if rtp timestamp can be calculating instead of checking number of rtp packets. This way it works for reconfigured streams too.

It also moved deeper into rtcp_sender class to prevent SR no matter the reason it need to be genereated. This way it prevents creating compound rtcp packets that have to start with Sender Report and Sender Reports as response to (mostly theoretical) sr-request rtcp packet.

BUG=webrtc:1600
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1639253007 .

Cr-Commit-Position: refs/heads/master@{#13503}
2016-07-20 13:27:09 +00:00
13deaad1bd TMMBRHelp moved from member object/base class to stack object,
indicating the usage of this helper is local.
With local usage critical section become obvisously useless and removed.

BUG=webrtc:5565
R=åsapersson

Review-Url: https://codereview.webrtc.org/1959013003
Cr-Commit-Position: refs/heads/master@{#12881}
2016-05-24 20:25:35 +00:00
6eaa3a41ce _boundingSetToSend moved out of tmmbr_help_ into tmmbn_to_send_
because in the TMMBRHelp class it is independent of other members.

BUG=webrtc:5565
R=philipel

Review-Url: https://codereview.webrtc.org/1746773002
Cr-Commit-Position: refs/heads/master@{#12669}
2016-05-09 17:59:55 +00:00
4485ffb58d #include "webrtc/base/constructormagic.h" where appropriate
Any file that uses the RTC_DISALLOW_* macros should #include
"webrtc/base/constructormagic.h", but a shocking number of them don't.
This causes trouble when we try to wean files off of #including
scoped_ptr.h, since a bunch of files get their constructormagic macros
only from there.

Rather than fixing these errors one by one as they turn up, this CL
simply ensures that every file in the WebRTC tree that uses the
RTC_DISALLOW_* macros #includes "webrtc/base/constructormagic.h".

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1917043005

Cr-Commit-Position: refs/heads/master@{#12509}
2016-04-26 15:14:48 +00:00
Per
83d0910694 Move Ownership of RtpModules to VideoSendStream from VieChannel and remove use of vie_channel and vie_receiver from video_send_stream.
The purpose of this refactoring is a first step of separating the encoder parts from the RTP transport.

BUG=webrtc:5687
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1864313003 .

Cr-Commit-Position: refs/heads/master@{#12377}
2016-04-15 12:59:21 +00:00
1c392cc5cf Avoid rescheduling the next RTCP packet if the RTCP sender status doesn't change.
The change made in https://codereview.webrtc.org/1757683002 introduced an extra call to RTCPSender::SetRTCPStatus after the video receive stream is created. The SetRTCPStatus call results in no state change, as the RTCP sender is already enabled, however, it reschedules the next RTCP packet to be RTCP_INTERVAL_VIDEO_MS/2 (500) ms in the future.
Before the change, the next packet time was only set by the previous call to RTCPSender::SetSSRC, which placed it 100 ms in the future. The change, therefore, changed the timing of multiple performance tests - as it now takes a different length of time to ramp up to the same bandwidth.

BUG=chromium:597332

Review URL: https://codereview.webrtc.org/1826093004

Cr-Commit-Position: refs/heads/master@{#12203}
2016-04-01 21:46:54 +00:00
41befcee7d Make rtcp sender use max transfer unit.
Remove packet overhead from rtp sender as unused.

R=philipel, åsapersson

Review URL: https://codereview.webrtc.org/1827953002

Cr-Commit-Position: refs/heads/master@{#12165}
2016-03-30 18:11:55 +00:00
56036ffc45 cleanup RTCPSender
rtc::scoped_ptr -> std::unique_ptr.
CrticalSectionWrapper -> CriticalSection.
assert -> DCHECK.
removed unused headers.
removed unused using.
removed unused member field.

BUG=webrtc:5520, webrtc:5565
R=åsapersson

Review URL: https://codereview.webrtc.org/1806603002

Cr-Commit-Position: refs/heads/master@{#12091}
2016-03-22 18:14:16 +00:00
a094fd1550 RTT intermediate calculation use ntp time instead of milliseconds.
Compact NTP representation was designed exactly for that purpose: calculate RTT. No need to map to ms before doing arithmetic on this values.
  Because of this change there is no need to keep mapping between compact ntp presentation and milliseconds in the RTCPSender.

BUG=webrtc:5565
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1491843004 .

Cr-Commit-Position: refs/heads/master@{#11710}
2016-02-22 17:59:47 +00:00
0e2e50ca1c Always append the BYE packet type at the end
When composing a RTCP packet, if there is a BYE
to be appended, preserve it and append it at the
end after all other packet types are added.

BUG=webrtc:5498
NOTRY=true

Review URL: https://codereview.webrtc.org/1674963004

Cr-Commit-Position: refs/heads/master@{#11672}
2016-02-18 16:33:33 +00:00
9d0c43242b Remove video-codec max bitrate from TMMBN.
TMMBN was capped by configured max bitrate for no apparent reason.
Removing this to not require payload-type reconfiguration on new
video-codec settings. Actual removal of payload-type reconfiguration
will happen in a pending CL.

BUG=webrtc:5494
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1702043002 .

Cr-Commit-Position: refs/heads/master@{#11639}
2016-02-16 16:59:36 +00:00
f174e3a260 [rtp_rtcp] rtcp::Tmmbr cleaned and got Parse function
Added accessor and Parse function
removed dependencies on structures from rtcp_utility.h (except RtcpCommonHeader)

BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1675583002

Cr-Commit-Position: refs/heads/master@{#11502}
2016-02-05 12:56:40 +00:00
de13882d94 rtcp::ExtenededReports packet class got Parse function
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1557593002

Cr-Commit-Position: refs/heads/master@{#11428}
2016-01-29 19:26:20 +00:00
32e590ec13 class doesn't rely on structures in RTCPUtility to store data.
supports several fci items in same packet.
got accessors to read data

BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1544403002 .

Cr-Commit-Position: refs/heads/master@{#11354}
2016-01-22 10:05:10 +00:00
429c345b02 Fixes a bug which incorrectly logs incoming RTCP as outgoing.
Adds logging to RTPSender and RTCPSender, pushing an event log pointer from Channel through ModuleRtpRtcpImpl to the Sender objects.

BUG=webrtc:4741

Review URL: https://codereview.webrtc.org/1571283002

Cr-Commit-Position: refs/heads/master@{#11336}
2016-01-21 13:42:10 +00:00
34ed2b95a5 [rtp_rtcp] rtcp::SenderReport moved into own file and got Parse function
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1544983002

Cr-Commit-Position: refs/heads/master@{#11288}
2016-01-18 10:43:38 +00:00
1567d0bd98 [rtp_rtcp] rtcp::Sdes moved into own file
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1439553003/

BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1592763002 .

Cr-Commit-Position: refs/heads/master@{#11274}
2016-01-15 16:34:32 +00:00
2c13297bf5 [rtp_rtcp] rtcp::Rpsi moved into own file
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1550293003/

BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1583233007 .

Cr-Commit-Position: refs/heads/master@{#11272}
2016-01-15 14:21:34 +00:00
256e5b23f8 Cleaning/Parsing will be done in the https://codereview.webrtc.org/1557593002/
BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1579213005 .

Cr-Commit-Position: refs/heads/master@{#11271}
2016-01-15 13:16:36 +00:00
5679da1291 [rtp_rtcp] rtcp::Fir moved into own file
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1544403002

BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1581983003 .

Cr-Commit-Position: refs/heads/master@{#11269}
2016-01-15 12:19:59 +00:00
a5eba6c98b [rtp_rtcp] rtcp::Remb moved into own file
Cleaning/Parsing will be done in the https://codereview.webrtc.org/1552773002/

BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1590883002 .

Cr-Commit-Position: refs/heads/master@{#11268}
2016-01-15 11:40:27 +00:00
2f7dea164d [rtp_rtcp] rtcp::Empty moved into own file and renamed to CompoundPacket on the way
Class renamed to indicated use of the rtcp::Empty class: it can only be used as container for other rtcp packets.
All tests that use Append function moved from rtcp_packet_unittest, even if they did not use Empty class.
This is because there is plan to make RtcpPacket class lighter by moving Append functionality to CompoundPacket class.

BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1582503002

Cr-Commit-Position: refs/heads/master@{#11234}
2016-01-13 10:03:09 +00:00
92e677a1f8 [rtp_rtcp] rtcp::Sli packet moved into own file and got Parse function
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1551893002

Cr-Commit-Position: refs/heads/master@{#11228}
2016-01-12 18:05:00 +00:00
7e8145f05d [rtp_rtcp] rtcp::Tmmbr moved into own file
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1575023002

Cr-Commit-Position: refs/heads/master@{#11206}
2016-01-11 19:49:24 +00:00
ef3d805f6e [rtp_rtcp] rtcp::Tmmbn moved into own file
explicetly unchanged.

BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1578713002

Cr-Commit-Position: refs/heads/master@{#11201}
2016-01-11 11:31:17 +00:00
a72e7349d5 [rtp_rtcp] cleanup in RTCPSender class internals.
PrepareReportBlock and AddReportBlock private functions merged:
  PrepareReportBlock moved report block from statistic to temporary structure
  AddReportBlock copied that temporary structure into temporary map right after.
  Thanks to rtcp packet classes that temporary structure is now unneccesary.

BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1538833002

Cr-Commit-Position: refs/heads/master@{#11112}
2015-12-22 16:07:48 +00:00
a8890a57a5 rtcp::Nack packet moved into own file and got Parse function
Review URL: https://codereview.webrtc.org/1461623003

Cr-Commit-Position: refs/heads/master@{#11111}
2015-12-22 11:43:10 +00:00
91941ae493 rtcp::VoipMetric block moved into own file and got Parse function
Review URL: https://codereview.webrtc.org/1452733002

Cr-Commit-Position: refs/heads/master@{#11030}
2015-12-15 15:06:44 +00:00
6db6cdc604 [rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
BUG=webrtc:5277
R=mflodman

Review URL: https://codereview.webrtc.org/1513303003

Cr-Commit-Position: refs/heads/master@{#11025}
2015-12-15 10:54:50 +00:00
47a740bc5e [rtp_rtcp] lint errors about rand() usage fixed.
rand() usage replaced with new Random class, which also makes it clearer what interval random number is in.

BUG=webrtc:5277
R=mflodman

Review URL: https://codereview.webrtc.org/1519503002

Cr-Commit-Position: refs/heads/master@{#11019}
2015-12-15 08:30:12 +00:00
b8b6fbb7a5 lint build/include errors fixed in rtp_rtcp module
BUG=webrtc:5277
R=mflodman

Review URL: https://codereview.webrtc.org/1505993003

Cr-Commit-Position: refs/heads/master@{#10971}
2015-12-10 13:05:35 +00:00
162abd3562 lint whitespace warning removed from most rtp_rtcp/source/ files
rtcp_utility, rtp_utility, tmmbr_help, rtcp_receiver, rtcp_receiver_help are explicetly excluded from the cleanup becaues there are short plans (or cls) to do a deeper cleaning there.

BUG=webrtc:5277
R=pbos@webrtc.org, mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1512493002

Cr-Commit-Position: refs/heads/master@{#10966}
2015-12-10 10:39:45 +00:00
fc47ed6c05 rtcp::Rrtr block moved into own file and got Parse function
BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson

Review URL: https://codereview.webrtc.org/1496883002 .

Cr-Commit-Position: refs/heads/master@{#10912}
2015-12-07 13:46:42 +00:00
97f7e13c23 rtcp::ReceiverReport moved into own file and got Parse function
BUG=webrtc:5260
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1453083002 .

Cr-Commit-Position: refs/heads/master@{#10897}
2015-12-04 15:13:40 +00:00
f7c5776d42 Refactorings to send RTCP packets directly via the RtcpPacket callback, with some simplifications enabled by this. NACK now also sent via RtcpPacket.
BUG=webrtc:2450
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1309833002 .

Cr-Commit-Position: refs/heads/master@{#10888}
2015-12-04 09:40:54 +00:00
f8385aded0 rtcp::Pli moved into own file and got a Parse function
Created rtcp::Psfb abstract class between rtcp::Pli and rtcp::RtcpPacket to hold common data for Feedback Message.

BUG=webrtc:5260

Review URL: https://codereview.webrtc.org/1446513002

Cr-Commit-Position: refs/heads/master@{#10823}
2015-11-27 13:36:17 +00:00