webrtc_m130

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Autoroller 024eeff421 Roll chromium_revision 9df92afb16..9d565db4c0 (566630:568343)

Change log: 9df92afb16..9d565db4c0
Full diff: 9df92afb16..9d565db4c0

Roll chromium third_party 01aaf419f6..5e63803713
Change log: 01aaf419f6..5e63803713

Changed dependencies:
* src/base: 6b48dbc0d2..4fcbb21d49
* src/build: 169887d089..f29a733cc2
* src/buildtools: 6f4dae280c..5941c1b3df
* src/ios: b0428063aa..84fac4e727
* src/testing: 5951b2830b..04314205f8
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/5267ef7b4a..6ff2ba80b7
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/fdacd1639e..f13bae2df0
* src/third_party/depot_tools: e09b6845cf..79d42dfb11
* src/third_party/googletest/src: 9077ec7efe..ce468a17c4
* src/third_party/libvpx/source/libvpx: 37a0283b18..8648a64c83
* src/third_party/libyuv: 196e2e72a3..780cdfed4e
* src/tools: e61dbb7de4..59c08146e7
* src/tools/swarming_client: 281c390193..9a518d097d
DEPS diff: 9df92afb16..9d565db4c0/DEPS

No update to Clang.

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

Change-Id: Ia79e1d33a4a422c50a1802ca6abcd7c60121dfa5
Reviewed-on: https://webrtc-review.googlesource.com/84104
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23648}

2018-06-19 05:30:50 +00:00

api

Improves buffer time calculation in network control tester.

2018-06-15 12:39:59 +00:00

audio

Replace rtc::Optional with absl::optional in audio, call and video

2018-06-15 12:09:49 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Replace rtc::Optional with absl::optional in audio, call and video

2018-06-15 12:09:49 +00:00

common_audio

Remove unused WavFile::FormatAsString method.

2018-06-14 09:05:20 +00:00

common_video

Delete unused stats for preferred_bitrate.

2018-06-07 08:11:07 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

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

examples

Release ADM after passing it to PCF in AppRTC

2018-06-12 11:42:41 +00:00

infra

Flip luci.webrtc.try to production

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

logging

Pass the RtcEventLog instance to ICE via JsepTransportController.

2018-06-12 05:04:35 +00:00

media

Extract rtc_base/base64.h and rtc_base/base64.cc into separate target.

2018-06-18 16:44:47 +00:00

modules

[desktopCapture Mac]reorder execution order in start/release processing

2018-06-18 23:10:17 +00:00

ortc

Replace rtc::Optional with absl::optional in media, ortc, p2p

2018-06-16 07:09:59 +00:00

p2p

Removing unused cricket::Port constructor.

2018-06-18 21:55:04 +00:00

Extract rtc_base/base64.h and rtc_base/base64.cc into separate target.

2018-06-18 16:44:47 +00:00

resources

AGC2 RNN VAD: Polishing.

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

rtc_base

Extract rtc_base/base64.h and rtc_base/base64.cc into separate target.

2018-06-18 16:44:47 +00:00

rtc_tools

Replace rtc::Optional with absl::optional in test and rtc_tools

2018-06-18 13:15:23 +00:00

sdk

Refactor checks to use a copy of the new logging backend.

2018-06-15 13:48:25 +00:00

stats

PeerConnectionInterface::GetStats() with selector argument added.

2018-03-26 12:08:20 +00:00

style-guide

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

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

system_wrappers

Clean up posix-only files.

2018-04-18 00:18:04 +00:00

test

Replace rtc::Optional with absl::optional in test and rtc_tools

2018-06-18 13:15:23 +00:00

third_party

Roll chromium_revision 9df92afb16..9d565db4c0 (566630:568343)

2018-06-19 05:30:50 +00:00

tools_webrtc

Make instructions for checkin_chrome_dep a bit clearer.

2018-06-15 17:08:56 +00:00

video

Replace rtc::Optional with absl::optional in audio, call and video

2018-06-15 12:09:49 +00:00

.clang-format

Tune ObjC clang-format configuration

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

.git-blame-ignore-revs

…

.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

Fix: NetEq PacketBuffer logs discarded packet with wrong codec level when new packet replaces the lower level packet

2018-06-08 14:58:18 +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

Reland "Move class VideoCodec from common_types.h to its own api header file."

2018-06-11 19:23:20 +00:00

DEPS

Roll chromium_revision 9df92afb16..9d565db4c0 (566630:568343)

2018-06-19 05:30:50 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

…

LICENSE_THIRD_PARTY

Add base64 webrtc owned third_party dep

2018-06-15 18:54:26 +00:00

native-api.md

Remove legacy VoiceEngine.

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

OWNERS

Add phoglund as root owner.

2018-05-18 15:57:56 +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 presubmit error if third_party/.git exists.

2018-05-23 09:17:30 +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

Fixing root_files WATCHLIST regex.

2018-04-19 06:52:18 +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

C++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%