webrtc_m130

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Autoroller 87e4479924 Roll chromium_revision 72ef4e4784..ecf8a6133e (569500:569618)

Change log: 72ef4e4784..ecf8a6133e
Full diff: 72ef4e4784..ecf8a6133e

Roll chromium third_party 1703674da3..46683344d7
Change log: 1703674da3..46683344d7

Changed dependencies:
* src/base: 9559d3ec4d..f7595e419a
* src/build: 379377c701..69593eb8fa
* src/ios: a0396d2bff..181b18c878
* src/testing: 6de780425c..8354b28f74
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/7a75f465f5..87eefd4f11
* src/third_party/depot_tools: 4099daa97b..c5a26a769e
* src/tools: 2e2b8c6c4c..592ddd1d14
DEPS diff: 72ef4e4784..ecf8a6133e/DEPS

No update to Clang.

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

Change-Id: Id2fa947be652922f0ebfb2cbcab8252c8dc9eae6
Reviewed-on: https://webrtc-review.googlesource.com/85022
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23719}

2018-06-22 15:08:06 +00:00

api

Generalize SimulcastEncoderAdapter, use for H264 & VP8.

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

audio

Reformat the WebRTC code base

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

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Convert video quality test from a TEST_F to a TEST fixture.

2018-06-21 15:49:43 +00:00

common_audio

Replace rtc::Optional with absl::optional

2018-06-21 09:32:56 +00:00

common_video

Replace rtc::Optional with absl::optional

2018-06-21 09:32:56 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

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

examples

Replace rtc::Optional with absl::optional

2018-06-21 09:32:56 +00:00

infra

Flip luci.webrtc.try to production

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

logging

Replace rtc::Optional with absl::optional

2018-06-21 09:32:56 +00:00

media

Generalize SimulcastEncoderAdapter, use for H264 & VP8.

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

modules

Fix for VP9 K-SVC video freeze frame when send bandwidth is restricted.

2018-06-21 17:53:35 +00:00

ortc

Reformat the WebRTC code base

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

p2p

Replace rtc::Optional with absl::optional in rtc_base

2018-06-21 11:23:40 +00:00

Convert video quality test from a TEST_F to a TEST fixture.

2018-06-21 15:49:43 +00:00

resources

AGC2 RNN VAD: Polishing.

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

rtc_base

Reland "Reland "Injectable logging""

2018-06-21 13:44:53 +00:00

rtc_tools

Aligning time in audio jitter buffer plot to other plots in rtc event log visualizer.

2018-06-21 14:23:53 +00:00

sdk

Reland "Reland "Injectable logging""

2018-06-21 13:44:53 +00:00

stats

Reformat the WebRTC code base

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

style-guide

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

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

system_wrappers

Replace rtc::Optional with absl::optional

2018-06-21 09:32:56 +00:00

test

Fix for VP9 K-SVC video freeze frame when send bandwidth is restricted.

2018-06-21 17:53:35 +00:00

third_party

Roll chromium_revision cb8b61b491..72ef4e4784 (569376:569500)

2018-06-22 02:09:00 +00:00

tools_webrtc

Removing some TSan suppressions around Thread class.

2018-06-20 19:42:48 +00:00

video

Generalize SimulcastEncoderAdapter, use for H264 & VP8.

2018-06-21 15:57:43 +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

Remove third_party from DEPS file to prepare to check it into webrtc.

2018-05-11 09:30:12 +00:00

.gn

Opt out of "Migrate the Android Support Lib to android_deps".

2018-04-05 13:40:53 +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

Android: Add helper class for generating OpenGL shaders

2018-06-15 09:06:45 +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 72ef4e4784..ecf8a6133e (569500:569618)

2018-06-22 15:08:06 +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

Add format check to git cl presubmit

2018-06-20 14:46:46 +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

THIRD_PARTY_CHROMIUM_DEPS.json

Add R8 to DEPS.

2018-06-08 10:31:38 +00:00

THIRD_PARTY_WEBRTC_DEPS.json

Add base64 webrtc owned third_party dep

2018-06-15 18:54:26 +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

Remove manual references to exe_and_shlib_deps

2018-06-12 06:07:16 +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

Languages

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Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%