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chromium-webrtc-autoroll eec1d04aa9 Roll chromium_revision 8b5b046f02..022da7ca60 (715044:716991)

Change log: 8b5b046f02..022da7ca60
Full diff: 8b5b046f02..022da7ca60

Changed dependencies
* src/base: 20e7dd7480..29c24a8b46
* src/build: ed0d663913..c60d0d9d66
* src/buildtools: 140e4d7c45..6b3e658d6f
* src/buildtools/third_party/libc++/trunk: 5938e0582b..78d6a7767e
* src/ios: 899a8a4bc9..01eee3d70d
* src/testing: d4d918486a..a6c25af3b7
* src/third_party: aad8135d1e..24a1541bd5
* src/third_party/bazel: tQPvsIj1Gtw5iXssKy7OREE-S02u7zItrw42l3DHUroC..VjMsf48QUWw8n7XtJP2AuSjIGmbQeYdWdwyxVvIRLmAC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/d709b0d892..6ba98ff601
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/3992f65642..55394ddc90
* src/third_party/depot_tools: 9577daf667..639872c8ef
* src/third_party/libyuv: 53b529e362..6afd9becdf
* src/third_party/lss: https://chromium.googlesource.com/linux-syscall-support.git/+log/8048ece6c1..726d71ec08
* src/tools: 224c37eeae..f2ed5a7b38
DEPS diff: 8b5b046f02..022da7ca60/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I44e824d8cfee4c3f4536be4442b93ed30fefe56b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160060
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#29847}

2019-11-20 14:42:26 +00:00

api

RTC_EXPORT RTCStatsMember's specialized members.

2019-11-20 13:53:37 +00:00

audio

Add missing assert.h for win no-test build

2019-11-17 07:52:32 +00:00

build_overrides

Remove crbug.com/904400 workaround.

2019-03-15 18:36:23 +00:00

call

Using unit types in TransportFeedbackAdapter.

2019-11-06 12:25:00 +00:00

common_audio

Use source_sets in component builds and static_library in release builds.

2019-10-17 21:17:18 +00:00

common_video

Add scale and compare methods to VideoFrame::UpdateRect

2019-11-19 15:42:42 +00:00

data

Remove old data files.

2018-10-05 14:40:21 +00:00

docs

Fixing some typos.

2019-09-10 10:03:50 +00:00

examples

Add @Nullable annotations to quiet errorprone.

2019-11-19 12:50:30 +00:00

logging

Detach thread checker in RtcEventLogImpl destructor.

2019-11-13 15:42:19 +00:00

media

Change VideoAdapter::OnResolutionFramerateRequest to VideoAdapter::OnSinkWants

2019-11-19 10:14:31 +00:00

modules

Makes RoundRobinPacketQueue use same field trials as PacingController.

2019-11-20 13:36:46 +00:00

p2p

Move stun.h to api/.

2019-11-18 16:11:27 +00:00

Make SCTPtransport enter "closed" state when DTLStransport does.

2019-11-16 14:56:01 +00:00

resources

Reland "Refactoring of the noise suppressor and adding true multichannel support"

2019-10-31 11:56:01 +00:00

rtc_base

In rtc::ByteBuffer drop support for ORDER_HOST as unused

2019-11-15 11:48:42 +00:00

rtc_tools

EventLogVisualizer pacer delay plot fix.

2019-11-19 10:23:31 +00:00

sdk

Add @Nullable annotations to quiet errorprone.

2019-11-19 12:50:30 +00:00

stats

Use explicit template decl/def for RTCNonStandardStatsMember.

2019-11-13 11:12:40 +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 source_sets in component builds and static_library in release builds.

2019-10-17 21:17:18 +00:00

test

Change VideoAdapter::OnResolutionFramerateRequest to VideoAdapter::OnSinkWants

2019-11-19 10:14:31 +00:00

tools_webrtc

Manual chromium roll: Compile using JDK 11

2019-11-20 12:13:43 +00:00

video

Stop using DEPRECATED_SingleThreadedTaskQueueForTesting in MultiStreamTester

2019-11-19 10:52:12 +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 new format cleanup.

2019-07-11 16:18:51 +00:00

.gitignore

Add .clangd to .gitignore

2019-10-28 12:27:50 +00:00

.gn

Switch to compiling WebRTC -std=c++14 by default

2019-09-09 19:24:16 +00:00

.vpython

Add source-side perf upload script for WebRTC.

2019-11-18 14:37:01 +00:00

abseil-in-webrtc.md

Update style guide for absl::make_unique.

2019-09-18 06:10:58 +00:00

AUTHORS

Revert "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5.""

2019-10-30 18:14:52 +00:00

BUILD.gn

Move stun.h to api/.

2019-11-18 16:11:27 +00:00

CODE_OF_CONDUCT.md

Add code of conduct to WebRTC repo

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

codereview.settings

Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit

2018-10-25 08:19:53 +00:00

common_types.h

Format almost everything.

2019-07-08 13:45:15 +00:00

DEPS

Roll chromium_revision 8b5b046f02..022da7ca60 (715044:716991)

2019-11-20 14:42:26 +00:00

ENG_REVIEW_OWNERS

Enforce LGTM from owners of depends-on paths in DEPS via presubmit.

2018-09-28 12:49:54 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

…

native-api.md

Delete unused I420 "codec"

2018-12-18 12:30:58 +00:00

OWNERS

Add #COMPONENT to WebRTC.

2019-10-08 12:20:39 +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

Use source_sets in component builds and static_library in release builds.

2019-10-17 21:17:18 +00:00

PRESUBMIT.py

Add possibility to skip check_includes presubmit check.

2019-10-22 19:35:31 +00:00

pylintrc

Fixing py lint errors

2018-07-23 15:28:48 +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

Update WebRTC's C++ style guide to reflect the switch to C++14.

2019-09-16 11:45:35 +00:00

WATCHLISTS

Add saza to audio watchlists

2019-09-03 14:55:43 +00:00

webrtc_lib_link_test.cc

Rewrite the lib link test to just be a binary.

2019-10-18 07:42:20 +00:00

webrtc.gni

Manual chromium roll: Compile using JDK 11

2019-11-20 12:13:43 +00:00

whitespace.txt

Revert "Whitespace change"

2019-11-11 14:58:20 +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++ 88.6%

C 3.3%

Java 3%

Objective-C++ 1.9%

Python 1.9%

Other 1%