platform-external-webrtc

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Autoroller 57dc9e367c Roll chromium_revision ffaf1e2ba6..ef538e3112 (565764:566490)

Change log: ffaf1e2ba6..ef538e3112
Full diff: ffaf1e2ba6..ef538e3112

Roll chromium third_party fb3dc2a0aa..cbc2a20101
Change log: fb3dc2a0aa..cbc2a20101

Changed dependencies:
* src/base: 2743076235..02a6c4cdd0
* src/build: 459adce3eb..3c4d6b6d24
* src/ios: 71b35d6ee6..0c6d3816a0
* src/testing: f624b1f4b7..e0597e0b5d
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/eca23c365a..fdacd1639e
* src/third_party/depot_tools: e05f18d477..e09b6845cf
* src/third_party/freetype/src: 0589f6e6ee..8f1ed54877
* src/third_party/googletest/src: 145d05750b..9077ec7efe
* src/tools: 76e5757c8f..c7862334ce
DEPS diff: ffaf1e2ba6..ef538e3112/DEPS

Clang version changed 332838:334100
Details: ffaf1e2ba6..ef538e3112/tools/clang/scripts/update.py

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal
NO_AUTOIMPORT_DEPS_CHECK=true

Change-Id: Iac45f00b35127f886b32aac8b79578fbe528fb00
Reviewed-on: https://webrtc-review.googlesource.com/83220
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23585}

2018-06-12 18:26:40 +00:00

api

Add HeaderExtensions to RtpParameters

2018-06-12 17:01:40 +00:00

audio

Enable send side audio TWCC only if WebRTC-Audio-ForceNoTWCC is not enabled.

2018-06-12 16:01:20 +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

Release ADM after passing it to PCF in AppRTC

2018-06-12 11:42:41 +00:00

infra

Flip luci.webrtc.try to production

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

logging

Pass the RtcEventLog instance to ICE via JsepTransportController.

2018-06-12 05:04:35 +00:00

media

Add HeaderExtensions to RtpParameters

2018-06-12 17:01:40 +00:00

modules

Make rtc_software_fallback_wrappers target visible.

2018-06-12 12:51:34 +00:00

ortc

New class FakePeriodicVideoTrackSource, simplifying shutdown logic.

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

p2p

Remove stringstreams from p2p/

2018-06-12 08:08:57 +00:00

Add HeaderExtensions to RtpParameters

2018-06-12 17:01:40 +00:00

resources

AGC2 RNN VAD: Polishing.

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

rtc_base

LogMessage::UpdateMinLogSeverity: Don't ignore all but the last stream

2018-06-12 10:56:21 +00:00

rtc_tools

Remove manual references to exe_and_shlib_deps

2018-06-12 06:07:16 +00:00

sdk

Add HeaderExtensions to RtpParameters

2018-06-12 17:01:40 +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

Document that preferred VideoFrame constructor takes no RTP timestamp.

2018-06-11 18:42:40 +00:00

third_party

Roll chromium_revision ffaf1e2ba6..ef538e3112 (565764:566490)

2018-06-12 18:26:40 +00:00

tools_webrtc

Explicitly setting use_lld=false on MSVC bots.

2018-06-12 13:49:38 +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

Fix: NetEq PacketBuffer logs discarded packet with wrong codec level when new packet replaces the lower level packet

2018-06-08 14:58:18 +00:00

BUILD.gn

Make rtc_software_fallback_wrappers target visible.

2018-06-12 12:51:34 +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

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

2018-06-11 19:23:20 +00:00

DEPS

Roll chromium_revision ffaf1e2ba6..ef538e3112 (565764:566490)

2018-06-12 18:26:40 +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

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

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C 3.3%

Java 3%

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Python 1.9%

Other 1%