webrtc_m130

Go to file

Artem Titov cb76c70534 Revert "Roll chromium_revision e1ef7d4b6b..b47e7752c6 (568794:569173)"

This reverts commit fd2457b10fc2713bfd5d86952216cf89b6283a90.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Roll chromium_revision e1ef7d4b6b..b47e7752c6 (568794:569173)
> 
> Change log: e1ef7d4b6b..b47e7752c6
> Full diff: e1ef7d4b6b..b47e7752c6
> 
> Roll chromium third_party ab9fbe29c9..630af48a96
> Change log: ab9fbe29c9..630af48a96
> 
> Changed dependencies:
> * src/base: 03e1bc561f..6070b24b9f
> * src/build: bb306be407..511e258eee
> * src/ios: 48697bf3a1..ce1db7deb1
> * src/testing: 6440c4ea3a..80a4cfaab7
> * src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/f153b902be..f3c454475a
> * src/third_party/depot_tools: 1cabdc4643..a28b14f122
> * src/tools: 66f1089d0c..f6bb2a6fb4
> DEPS diff: e1ef7d4b6b..b47e7752c6/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: I7553549111225f2c5bc769dec114cd058ae699d5
> Reviewed-on: https://webrtc-review.googlesource.com/84600
> Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
> Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23692}

TBR=buildbot@webrtc.org

Change-Id: I7d136553622fe4c4e49dbd9c60406811a8b11bd8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Cq-Include-Trybots: master.internal.tryserver.corp.webrtc:linux_internal
Reviewed-on: https://webrtc-review.googlesource.com/84720
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23708}

2018-06-21 12:53:44 +00:00

api

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

2018-06-21 12:50:03 +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

Delete pre_decode_callback.

2018-06-20 07:04:09 +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

Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.

2018-06-21 12:23:03 +00:00

modules

Revert "NetEq: Deprecate playout modes Fax, Off and Streaming"

2018-06-21 12:36:44 +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

Add support for content-hint value "text"

2018-06-21 11:29:20 +00:00

resources

AGC2 RNN VAD: Polishing.

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

rtc_base

Remove usage of rtc_base/checks.h in 3pp base64.cc

2018-06-21 11:30:41 +00:00

rtc_tools

Removing usage of //build/config/compiler:no_size_t_to_int_warning.

2018-06-20 13:44:26 +00:00

sdk

Fixed crash when PCF is destroyed before MediaStream in ObjC

2018-06-21 11:12:30 +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

Revert "NetEq: Deprecate playout modes Fax, Off and Streaming"

2018-06-21 12:36:44 +00:00

third_party

Revert "Roll chromium_revision e1ef7d4b6b..b47e7752c6 (568794:569173)"

2018-06-21 12:53:44 +00:00

tools_webrtc

Removing some TSan suppressions around Thread class.

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

video

Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.

2018-06-21 12:23:03 +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

Implement H264 simulcast support and generalize SimulcastEncoderAdapter use for H264 & VP8.

2018-06-21 12:23:03 +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

Revert "Roll chromium_revision e1ef7d4b6b..b47e7752c6 (568794:569173)"

2018-06-21 12:53:44 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

…

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

C++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%