webrtc_m130

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Artem Titov c34e381d61 Roll chromium_revision 039110971b..52f78b1683 (560284:561464)

Change log: 039110971b..52f78b1683
Full diff: 039110971b..52f78b1683

Roll chromium third_party cc1af82934..008fb7071c
Change log: cc1af82934..008fb7071c

Changed dependencies:
* src/base: 8e89780685..40343e3fbc
* src/build: 66897e4d72..bd04ef7233
* src/ios: 02a22b3900..de97874e25
* src/testing: 671c6a4522..b4c21a01c2
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/7ca7a59f02..1986f5a957
* src/third_party/depot_tools: 083eb25f9a..4e9b50ab86
* src/third_party/googletest/src: 08d5b1f33a..145d05750b
* src/third_party/libvpx/source/libvpx: d99abe9a9a..e27a331778
* src/tools: ff5c71196b..c923d1173c
DEPS diff: 039110971b..52f78b1683/DEPS

Clang version changed 332335:332838
Details: 039110971b..52f78b1683/tools/clang/scripts/update.py

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: I0aa2e7087bc0871ddafffb9eae424c6c76cc5b47
Reviewed-on: https://webrtc-review.googlesource.com/78762
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23385}

2018-05-24 13:48:31 +00:00

api

Enables comparison with infinite timestamps.

2018-05-24 12:45:21 +00:00

audio

Drop the RTT as input to IsRetransmitOfOldPacket.

2018-05-23 13:14:40 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Removing references to webrtc::VideoSendStream::DegradationPreference.

2018-05-21 20:20:57 +00:00

common_audio

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

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

common_video

Directly include VideoBitrateAllocation in common_video/ targets

2018-05-23 17:57:14 +00:00

data

…

examples

Move socklen_t definition for windows to win32.h.

2018-05-24 11:17:30 +00:00

infra

Switch one try builder (win_x64_clang_dbg) to LUCI

2018-05-24 11:50:45 +00:00

logging

Delete unneeded includes of call/video_config.h.

2018-05-18 09:00:56 +00:00

media

Directly include VideoBitrateAllocation in media/ targets

2018-05-23 13:56:40 +00:00

modules

Replaces redundant congestion controller components

2018-05-24 13:35:31 +00:00

ortc

New class FakePeriodicVideoTrackSource, simplifying shutdown logic.

2018-05-21 10:27:55 +00:00

p2p

Don't assume that RTC_LOG's << operator is std::ostream

2018-05-23 10:07:20 +00:00

Refactor VideoTrackSource, without raw pointer injection.

2018-05-23 15:42:10 +00:00

resources

AGC2 RNN VAD: Polishing.

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

rtc_base

Move socklen_t definition for windows to win32.h.

2018-05-24 11:17:30 +00:00

rtc_tools

Replaces redundant congestion controller components

2018-05-24 13:35:31 +00:00

sdk

Add getSupportedCodecs to VideoDecoderFactory interface.

2018-05-21 13:21:45 +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

Move socklen_t definition for windows to win32.h.

2018-05-24 11:17:30 +00:00

third_party

Roll chromium_revision 039110971b..52f78b1683 (560284:561464)

2018-05-24 13:48:31 +00:00

tools_webrtc

Disable rolling of ffmpeg.

2018-05-24 09:01:15 +00:00

video

Add tests where the incoming stream changes codec type.

2018-05-24 08:15:15 +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

Use range-based-for instead of std::for_each and std::mem_fun

2018-05-17 13:51:02 +00:00

BUILD.gn

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

2018-05-23 12:07:10 +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.cc

Don't expose resilience mode in VP8 and VP9 configuration.

2018-04-25 13:54:33 +00:00

common_types.h

Reland "Reland "Remove our stream << overloads from non-test build targets.""

2018-05-03 10:41:41 +00:00

DEPS

Roll chromium_revision 039110971b..52f78b1683 (560284:561464)

2018-05-24 13:48:31 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

…

LICENSE_THIRD_PARTY

Remove custom MD5 / SHA-1 implementations.

2018-02-19 15:03:35 +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

Change structure of deps file and tool for adding chromium dep.

2018-05-17 08:52:31 +00:00

THIRD_PARTY_WEBRTC_DEPS.json

Reland: Add presubmit check for changes in 3pp

2018-05-22 13:11:18 +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

Revert "Reland "Adding absl includes and defines to rtc_* templates.""

2018-05-21 14:38:36 +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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