platform-external-webrtc

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Autoroller 57900cb933 Roll chromium_revision 6df2efa531..79cbcdf6fb (572277:572378)

Change log: 6df2efa531..79cbcdf6fb
Full diff: 6df2efa531..79cbcdf6fb

Changed dependencies:
* src/base: 7ffd231167..b321921624
* src/build: 798d88a968..91b88ae14f
* src/ios: bbb1a1380e..9615de8dfb
* src/testing: c2d62722b5..b3ea40231d
* src/third_party: 2ed97d2760..3b7f89fc9f
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9f46241c6c..10e0a3798e
* src/third_party/libvpx/source/libvpx: 583859d739..03abd2c8f3
* src/tools: 62a39dba9a..fec28f646d
DEPS diff: 6df2efa531..79cbcdf6fb/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,marpan@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I7042e8b3b43cec65b7d5626a7945102d7407fb32
Reviewed-on: https://webrtc-review.googlesource.com/87040
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23827}

2018-07-03 23:09:20 +00:00

api

Adding shampson (me) as an owner to pc/ & api/.

2018-07-03 20:39:17 +00:00

audio

Remove StreamStatistician::IsPacketInOrder

2018-06-28 08:44:40 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

2017-10-19 09:21:12 +00:00

call

Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html ) for audio stream.

2018-06-27 10:33:40 +00:00

common_audio

Extract fft4g into separate build target

2018-06-26 13:39:25 +00:00

common_video

Add bit depth information to PlanarYuvBuffer

2018-06-26 20:23:37 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

2016-11-22 18:43:05 +00:00

examples

Remove usage of //build/config/clang:extra_warnings.

2018-07-03 12:35:49 +00:00

infra

Flip luci.webrtc.try to production

2018-05-30 08:30:00 +00:00

logging

Port RtcEventLog encoder unittests to the new parser API.

2018-06-28 15:31:23 +00:00

media

Add sprang@ as owner for simulcast.cc/h

2018-07-03 15:00:33 +00:00

modules

NetEq: Handle nested RED packets

2018-07-03 20:27:57 +00:00

ortc

Removing deadbeef@ from OWNERS files.

2018-07-02 00:40:38 +00:00

p2p

Fix a bug in TurnServer that causes flakiness in webrtc_perf_tests.

2018-07-02 18:51:23 +00:00

Adding shampson (me) as an owner to pc/ & api/.

2018-07-03 20:39:17 +00:00

resources

AGC2 RNN VAD: Polishing.

2018-05-15 16:41:02 +00:00

rtc_base

Move RTC_CHECK_OP error message construction out of header file.

2018-07-03 15:21:13 +00:00

rtc_tools

Removes redundant delay based bwe.

2018-07-02 09:11:33 +00:00

sdk

Remove usage of //build/config/clang:extra_warnings.

2018-07-03 12:35:49 +00:00

stats

Adding "is_standardized" flag to RTCStatsMember.

2018-06-28 00:43:46 +00:00

style-guide

Add style guide rule about paired .h and .cc files

2018-03-14 13:02:35 +00:00

system_wrappers

Use the sparse histogram in RTC_HISTOGRAM_ENUMERATION_SPARSE.

2018-06-27 00:41:29 +00:00

test

Discard frame self-dependency when parsing genric frame descriptor

2018-07-03 10:28:05 +00:00

tools_webrtc

roll_deps: Accept any prefix (like 'git_revision:'), not only 'version:' for CIPD

2018-07-03 09:41:53 +00:00

video

Remove StreamStatistician::IsPacketInOrder

2018-06-28 08:44:40 +00:00

.clang-format

Tune ObjC clang-format configuration

2017-05-11 09:14:18 +00:00

.git-blame-ignore-revs

Let git-hyper-blame ignore format commit.

2018-06-20 09:26:44 +00:00

.gitignore

Roll back checking in the third_party directory

2018-06-27 13:04:08 +00:00

.gn

Set gtest_enable_absl_printers to true.

2018-06-29 09:36:17 +00:00

.vpython

Add vpython dependencies needed to run presubmit tests on LUCI

2018-05-18 08:10:25 +00:00

AUTHORS

Generalize SimulcastEncoderAdapter, use for H264 & VP8.

2018-06-21 15:57:43 +00:00

BUILD.gn

A new PeerConnection level perf test.

2018-06-27 23:19:05 +00:00

CODE_OF_CONDUCT.md

Add code of conduct to WebRTC repo

2017-05-16 12:09:13 +00:00

codereview.settings

Make Gerrit the default for WebRTC changes

2017-09-29 01:38:07 +00:00

common_types.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

DEPS

Roll chromium_revision 6df2efa531..79cbcdf6fb (572277:572378)

2018-07-03 23:09:20 +00:00

LICENSE

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

license_template.txt

Update template to follow chromium copyright style

2013-04-24 01:01:28 +00:00

LICENSE_THIRD_PARTY

Remove third party dependecies that are not more in the source code

2018-06-21 11:33:41 +00:00

native-api.md

Remove legacy VoiceEngine.

2018-01-12 11:31:52 +00:00

OWNERS

Add mbonadei@ to build configs OWNERS.

2018-06-20 12:39:11 +00:00

PATENTS

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

presubmit_test_mocks.py

Reland: Add presubmit check for changes in 3pp

2018-05-22 13:11:18 +00:00

presubmit_test.py

Roll chromium_revision 95336cb92b..191d55580e (557816:557824)

2018-05-11 11:17:05 +00:00

PRESUBMIT.py

Roll back checking in the third_party directory

2018-06-27 13:04:08 +00:00

pylintrc

Removing invalid-name from disabled pylint checks.

2017-10-11 08:06:49 +00:00

README.chromium

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

README.md

Tell users where they can find the native API headers

2017-11-14 10:36:46 +00:00

style-guide.md

Add style guidance about forward declarations.

2018-03-28 20:58:27 +00:00

typedefs.h

Remove typedefs.h from webrtc/ root (part 1)

2018-05-23 12:07:10 +00:00

WATCHLISTS

Adding mbonadei@ to build_files WATCHLIST.

2018-06-20 12:38:06 +00:00

webrtc.gni

Don't call deprecated FFmpeg API.

2018-06-26 13:57:35 +00:00

whitespace.txt

Whitespace change

2018-02-23 10:34:16 +00:00

README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info

Official web site: http://www.webrtc.org
Master source code repo: https://webrtc.googlesource.com/src
Samples and reference apps: https://github.com/webrtc
Mailing list: http://groups.google.com/group/discuss-webrtc
Continuous build: http://build.chromium.org/p/client.webrtc
Coding style guide
Code of conduct

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C 3.3%

Java 3%

Objective-C++ 1.9%

Python 1.9%

Other 1%