Commit graph

31 commits

Author SHA1 Message Date
Elad Alon
7d6a4c045c Connect LossNotificationController to RtpRtcp
* LossNotificationController is the class that decides when to issue
  LossNotification RTCP messages.
* RtpRtcp handles the technicalities of producing RTCP messages.

Bug: webrtc:10336
Change-Id: I292536257a984ca85d21d9cfa38e7ff2569cbb39
Reviewed-on: https://webrtc-review.googlesource.com/c/124123
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26840}
2019-02-25 16:08:35 +00:00
Elad Alon
74f0a51f97 Move kFeedbackMessageType from Remb to Psfb
The FMT 15 is not specific only to REMB or loss notification messages.
Rather, it is the Application Layer FB (AFB) of Psfb (Payload Specific
Feedback Messages).
See https://tools.ietf.org/html/rfc4585#section-6.3

TBR=terelius@webrtc.org

Bug: webrtc:10336
Change-Id: I8cd27ef9ee044bf7b7e7c1bd1a53c1dae2d95006
Reviewed-on: https://webrtc-review.googlesource.com/c/123886
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26827}
2019-02-24 00:11:33 +00:00
Johannes Kron
8e847ee96a Make recv_deltas optional in TransportFeedback packets
Bug: webrtc:10263
Change-Id: I49c4a4710a5c34a62b53080e708c310a8484831b
Reviewed-on: https://webrtc-review.googlesource.com/c/122543
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26687}
2019-02-14 13:34:57 +00:00
Bjorn Terelius
dfd5c4b15e Parse XR, FIR and PLI in rtc_event_log_parser.cc
Bug: webrtc:10312
Change-Id: I1b24e23f8002feef8a2ef928130ac6da19c3cd81
Reviewed-on: https://webrtc-review.googlesource.com/c/122580
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26670}
2019-02-13 19:11:42 +00:00
Johannes Kron
99b9149cee Enable padding bit in TransportFeedback packets
Set padding bit if the TransportFeedback packet contains zero padding.
Also write number of padding elements at the last position of the packet.

Bug: webrtc:10263
Change-Id: I8d17bc0e889f658ac383dec64ddcb95d438c9702
Reviewed-on: https://webrtc-review.googlesource.com/c/122240
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26646}
2019-02-12 11:55:34 +00:00
Elad Alon
f8e7ccb967 Create new RTCP feedback message - LossIndication
Create a new RTCP feedback message for reporting the loss and/or non-decodability of video frames, to be used by the upcoming injectable VideoFrameBufferController. The new feedback message should report:
1. The sequence number of the last decoded non-discardable video frame. (TBD: If a multi-packet frame, should it be the sequence number of the first, last, or any of the packets?)
2. The sequence number of the last received RTP packet in the stream.
3. A decodability flag, whose specific meaning depends on the last-received
   RTP sequence number. The decodability flag is true if and only if all of
   the frame's dependencies are known to be decodable, and the frame itself
   is not yet known to be unassemblable.
   * Clarification #1: In a multi-packet frame, the first packet's
     dependencies are known, but it is not yet known whether all parts
     of the current frame will be received.
   * Clarification #2: In a multi-packet frame, the dependencies would be
     unknown if the first packet was not received. Then, the packet will
     be known-unassemblable.

Bug: webrtc:10226
Change-Id: I1563c944477e3ed40235e82ab99a439414632aff
Reviewed-on: https://webrtc-review.googlesource.com/c/118931
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26387}
2019-01-24 12:21:00 +00:00
Elad Alon
3fdf90d621 PSFB without REMB magic word is not an error
Several PSFB messages might be supported, distinguished using
the unique identifier. If the unique identifier is not REMB, it's
not an error, and so a warning should not be issued.

Bug: webrtc:10226
Change-Id: I5e79b473bd54cf0964f19329efb33354f63f5d5e
Reviewed-on: https://webrtc-review.googlesource.com/c/118686
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26359}
2019-01-22 12:29:47 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Yves Gerey
988cc0870b [Cleanup] Add missing #include. Remove useless ones.
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 11:32:56 +00:00
Niels Möller
44b384d013 Delete support for VoIP metrics (RFC 3611 4.7)
Bug: None
Change-Id: I2f3cd622d3863fa88a9e1971894eced8eeb777e6
Reviewed-on: https://webrtc-review.googlesource.com/c/103805
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25007}
2018-10-05 10:07:57 +00:00
Jonas Olsson
366a50c4ef Remove simple stringstream usages.
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.

The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.

Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
2018-09-06 12:53:19 +00:00
Mirko Bonadei
55d5ef0048 Remove clang:find_bad_constructs suppression from logging (part 1).
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.

Bug: webrtc:9251, webrtc:163
Change-Id: I443c0c77cef30a3f5712f72ce88db80d1fb29874
Reviewed-on: https://webrtc-review.googlesource.com/96642
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24567}
2018-09-05 08:33:17 +00:00
Qingsi Wang
b2db53998d Parse the number of packets lost in RTCP SR as a signed integer.
The cumulative number of packets lost in a RTCP sender report can be
negative if there are duplicates. This CL fixes a bug that the parser of
RTCP reports treats the field as an unsigned integer, and incorrectly
reports large packet losses when a negative loss is reported.

Bug: webrtc:9601
Change-Id: I1109ac0741614d61bda743e13a390b7d3e147a9c
Reviewed-on: https://webrtc-review.googlesource.com/92942
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#24234}
2018-08-08 16:44:11 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
Danil Chapovalov
d264df587f Replace rtc::Optional with absl::optional in modules/rtp_rtcp
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated using script:
#!/bin/bash
dir=modules/rtp_rtcp
find $dir -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $dir -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: Ife720849709959046329c1c9faa3f31aa13274dc
Reviewed-on: https://webrtc-review.googlesource.com/83584
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23624}
2018-06-15 09:53:35 +00:00
Niels Möller
65ec0fc81e Delete unneeded includes of basictypes.h.
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.

Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.

Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
2018-05-21 19:35:08 +00:00
Mirta Dvornicic
b1f063db32 Handle Receiver Reference Time Report from multiple receivers.
Bug: webrtc:9122
Change-Id: I996f02da26b11a4829fda740fdd452470daf4d24
Reviewed-on: https://webrtc-review.googlesource.com/66781
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22871}
2018-04-16 09:54:07 +00:00
Bjorn Terelius
b6a7fc0f03 Make rtcp::TransportFeedback copyable.
Bug: webrtc:8111
Change-Id: I2a71eb7ab5a913427adfab6f71703850a48fbd03
Reviewed-on: https://webrtc-review.googlesource.com/57181
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22218}
2018-02-28 08:06:20 +00:00
Danil Chapovalov
2a5ce2bcf8 Fix clang style errors in rtp_rtcp and dependant targets
Mark functions with override instead of virtual.
Add explicit non-trivial constructors/assign operators/destructors.
Define them in .cc files instead of inlining
use auto* instead of auto when deduced type is raw pointer

Bug: webrtc:163
Change-Id: I4d8a05d6a64fcc2ca16d02c5fcf9488fda832a6d
Reviewed-on: https://webrtc-review.googlesource.com/48781
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21927}
2018-02-07 09:48:28 +00:00
Harald Alvestrand
70206d6608 Reland "Make RTCP cumulative_lost be a signed value"
Instead of modifying the API, we'll add a new function to return
the true value, and have a shim that returns what other code expects.

> This reverts commit 4c34f435db.
>
> Reason for revert: Broke internal projects. Type mismatch.
>
> Original change's description:
> > Make RTCP cumulative_lost be a signed value
> >
> > This is formally defined as a signed 24-bit value in RFC 3550 section 6.4.1.
> > See RFC 3550 Appendix A.3 for the reason why it may turn negative.
> >
> > Noticed on discuss-webrtc mailing list.
> >
> > BUG=webrtc:8626
> >
> > Change-Id: I7317f73e9490a876e8445bd3d6b66095ce53ca0a
> > Reviewed-on: https://webrtc-review.googlesource.com/30901
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21142}
>
> TBR=stefan@webrtc.org,hta@webrtc.org
>
> Change-Id: I544f7979d584cfb72a2d0d526f4fef84aebeecb3
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8626
> Reviewed-on: https://webrtc-review.googlesource.com/31040
> Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21144}

Change-Id: I95c8c248f4f85c4d1aa2a47424d8c4d954d4ae7a
Bug: webrtc:8626
Reviewed-on: https://webrtc-review.googlesource.com/31220
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21154}
2017-12-08 08:47:09 +00:00
Zhi Huang
062a8ead3b Revert "Make RTCP cumulative_lost be a signed value"
This reverts commit 4c34f435db.

Reason for revert: Broke internal projects. Type mismatch.

Original change's description:
> Make RTCP cumulative_lost be a signed value
> 
> This is formally defined as a signed 24-bit value in RFC 3550 section 6.4.1.
> See RFC 3550 Appendix A.3 for the reason why it may turn negative.
> 
> Noticed on discuss-webrtc mailing list.
> 
> BUG=webrtc:8626
> 
> Change-Id: I7317f73e9490a876e8445bd3d6b66095ce53ca0a
> Reviewed-on: https://webrtc-review.googlesource.com/30901
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21142}

TBR=stefan@webrtc.org,hta@webrtc.org

Change-Id: I544f7979d584cfb72a2d0d526f4fef84aebeecb3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8626
Reviewed-on: https://webrtc-review.googlesource.com/31040
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21144}
2017-12-07 18:12:48 +00:00
Harald Alvestrand
4c34f435db Make RTCP cumulative_lost be a signed value
This is formally defined as a signed 24-bit value in RFC 3550 section 6.4.1.
See RFC 3550 Appendix A.3 for the reason why it may turn negative.

Noticed on discuss-webrtc mailing list.

BUG=webrtc:8626

Change-Id: I7317f73e9490a876e8445bd3d6b66095ce53ca0a
Reviewed-on: https://webrtc-review.googlesource.com/30901
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21142}
2017-12-07 17:00:57 +00:00
Danil Chapovalov
5c3cc41cef Change RtcpPacket::PacketReadyCallback to rtc::FunctionView
from interface


Bug: webrtc:5565
Change-Id: I2df5d7a0554b938888581f1c73dbdb8b85c387cc
Reviewed-on: https://webrtc-review.googlesource.com/8680
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21136}
2017-12-07 11:20:08 +00:00
Danil Chapovalov
327c43c29b Add sending Nack to RtcpTransceiver
Bug: webrtc:8239
Change-Id: Idf27bb05958d9eceaf601078019f05444232581f
Reviewed-on: https://webrtc-review.googlesource.com/26260
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20907}
2017-11-28 11:57:58 +00:00
Oskar Sundbom
3419cf909e Optional: Use nullopt and implicit construction in /modules/rtp_rtcp
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=danilchap@webrtc.org

Bug: None
Change-Id: Ib4694d183f04d675f2ea66d39661fdffb2a984f1
Reviewed-on: https://webrtc-review.googlesource.com/23604
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20846}
2017-11-23 11:36:08 +00:00
Karl Wiberg
e40468ba3d Move some numeric utility code from rtc_base/ to rtc_base/numerics/
Specifically, I'm moving

  safe_compare.h
  safe_conversions.h
  safe_minmax.h

They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.

BUG=webrtc:8445

Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
2017-11-22 11:21:47 +00:00
Mirko Bonadei
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros.
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
2017-11-09 11:56:32 +00:00
Danil Chapovalov
eb0ed28b90 Add rtcp::CommonHeader::packet_size function
as an alternative to NextPacket function
to allow cleaner iterating over stacked rtcp packets.

Bug: webrtc:5565
Change-Id: I261afe2684e5fcb5fa3e7bcce272fbefeebd0b66
Reviewed-on: https://webrtc-review.googlesource.com/21360
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20615}
2017-11-09 10:50:15 +00:00
Danil Chapovalov
6c170578e6 Move rtcp packet classes from rtp_rtcp to rtp_rtcp_format target
Bug: None
Change-Id: I353228fd5b75bd4fceeaee1bb6fd07b01dac56a1
Reviewed-on: https://webrtc-review.googlesource.com/1480
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19867}
2017-09-15 17:36:30 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

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


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

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

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00