platform-external-webrtc

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Autoroller 0428d37305 Roll chromium_revision f9a24d5d6f..e9fcce491e (564886:565609)

Change log: f9a24d5d6f..e9fcce491e
Full diff: f9a24d5d6f..e9fcce491e

Roll chromium third_party addc496eeb..b1a95b2d51
Change log: addc496eeb..b1a95b2d51

Changed dependencies:
* src/base: 769dcfb902..6d59c6b0f4
* src/build: 2574ebeaaf..472d5df73f
* src/ios: 044af8282c..2c466eefd8
* src/testing: 5ab6ab6e1c..4162fb6122
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/5601bdac1a..5267ef7b4a
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/01f608bbcf..c9718952e5
* src/third_party/depot_tools: e89dcf7dee..8e6f58c7e6
* src/third_party/libvpx/source/libvpx: 3f7e6cc020..87386826a9
* src/tools: 8a16887980..1702b5a977
DEPS diff: f9a24d5d6f..e9fcce491e/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: I28be447ede8150e4030611bdf3d1e4dda49ea32a
Reviewed-on: https://webrtc-review.googlesource.com/82120
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23549}

2018-06-08 12:11:51 +00:00

api

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

2018-06-08 11:04:23 +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

Delete unused stats for preferred_bitrate.

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

common_audio

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

2018-05-23 12:07:10 +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

Broadcast extension for AppRTCMobile on iOS

2018-06-04 08:49:21 +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

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

2018-06-08 11:04:23 +00:00

modules

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

2018-06-08 11:04:23 +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

Delete deprecated CreateAudioSource method, with constraints.

2018-06-04 08:19:30 +00:00

resources

AGC2 RNN VAD: Polishing.

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

rtc_base

Re-enabling SanitizerTest.MsanUninitialized.

2018-06-07 11:39:15 +00:00

rtc_tools

Split LoggedBweProbeResult into -Success and -Failure.

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

sdk

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

2018-06-08 11:04:23 +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

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

2018-06-08 11:04:23 +00:00

third_party

Roll chromium_revision f9a24d5d6f..e9fcce491e (564886:565609)

2018-06-08 12:11:51 +00:00

tools_webrtc

Remove MIPS MB config since we don't need it anymore.

2018-06-05 12:35:20 +00:00

video

Add full stack test with link capacity of 150 kbit/s.

2018-06-08 11:33:30 +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

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

2018-06-08 11:04:23 +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

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

2018-06-08 11:04:23 +00:00

common_types.h

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

2018-06-08 11:04:23 +00:00

DEPS

Roll chromium_revision f9a24d5d6f..e9fcce491e (564886:565609)

2018-06-08 12:11:51 +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

Add R8 to DEPS.

2018-06-08 10:31:38 +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

Broadcast extension for AppRTCMobile on iOS

2018-06-04 08:49:21 +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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