platform-external-webrtc

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Autoroller 38eac97df0 Roll chromium_revision 6989818705..f2d1e453de (563760:563863)

Change log: 6989818705..f2d1e453de
Full diff: 6989818705..f2d1e453de

Roll chromium third_party 77080c8cca..02d962eb0e
Change log: 77080c8cca..02d962eb0e

Changed dependencies:
* src/base: a6941a2497..0ecab11ffa
* src/build: adb59b644b..3c4efebb96
* src/ios: 2570306e97..8e0b21f69f
* src/testing: b44cad8154..3ae9d12c9d
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b48f5b4915..b26b30dc08
* src/third_party/libyuv: a9626b9daf..196e2e72a3
* src/tools: 189162d71b..c50cb3b65c
DEPS diff: 6989818705..f2d1e453de/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: Ib3458c9efd883f2bcc1e25f522827cf7beb0ce88
Reviewed-on: https://webrtc-review.googlesource.com/80740
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23499}

2018-06-02 00:05:45 +00:00

api

Remove the audio codec factory methods that don't take AudioCodecPairId

2018-06-01 11:04:07 +00:00

audio

Delete RtpFeedback. The ssrc for a receive stream should be known at

2018-05-28 11:05:19 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Wire up VAAPI VP8 experimental support in WebRTC.

2018-05-28 12:30:19 +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

WebRTC: Replace ProjectRootPath by ResourcePath

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

examples

ObjC: Notify local video track

2018-05-30 22:36:14 +00:00

infra

Flip luci.webrtc.try to production

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

logging

Split IceCandidatePairEventType enum.

2018-05-31 08:42:10 +00:00

media

Implement proper SCTP data channel closing procedure.

2018-05-31 21:01:53 +00:00

modules

Unstable BWE due to improper bit rate padding for VP9.

2018-06-01 20:07:06 +00:00

ortc

New class FakePeriodicVideoTrackSource, simplifying shutdown logic.

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

p2p

Fixing bug with PseudoTcp that corrupts data if initial packet is lost.

2018-05-31 18:54:58 +00:00

Create PeerConnectionFactoryDependencies to prevent new function overloads.

2018-05-31 23:41:12 +00:00

resources

AGC2 RNN VAD: Polishing.

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

rtc_base

Revert "Reland "Injectable logging""

2018-06-01 20:10:26 +00:00

rtc_tools

Split LoggedBweProbeResult into -Success and -Failure.

2018-05-29 13:41:04 +00:00

sdk

Revert "Reland "Injectable logging""

2018-06-01 20:10:26 +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

Add JoinFilename to testsupport code, replacing use of rtc::Pathname.

2018-06-01 11:36:51 +00:00

third_party

Roll chromium_revision 29e2805f88..3a0333ff4e (563524:563625)

2018-06-01 15:07:13 +00:00

tools_webrtc

Skip Mac-10.13.4 for iOS 10 tests, because the machines don't have the SDK

2018-05-30 13:40:39 +00:00

video

Unstable BWE due to improper bit rate padding for VP9.

2018-06-01 20:07:06 +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

ObjC: Notify local video track

2018-05-30 22:36:14 +00:00

BUILD.gn

Rely on use_fuzzing_engine && optimize_for_fuzzing to define WEBRTC_UNSAFE_FUZZER_MODE.

2018-05-30 04:28:28 +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

Delete definitions of NULL.

2018-05-31 07:55:15 +00:00

DEPS

Roll chromium_revision 6989818705..f2d1e453de (563760:563863)

2018-06-02 00:05:45 +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

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

2018-05-29 15:17:04 +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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