platform-external-webrtc

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Autoroller 0b9de80bca Roll chromium_revision 24887ed1ab..80343dd58e (535697:536688)

Change log: 24887ed1ab..80343dd58e
Full diff: 24887ed1ab..80343dd58e

Changed dependencies:
* src/base: f1f7eec82e..ac3d2b8118
* src/build: c7a79acbd2..39738e75b2
* src/buildtools: 2637e7e911..a09e064635
* src/ios: 0959011404..299ef76e84
* src/testing: 81576293e6..5f7e36cad6
* src/third_party: 50ccc08052..40927da822
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e653c4b823..e7298f36f7
* src/third_party/depot_tools: 71236c3af6..46541b4996
* src/third_party/ffmpeg: f5964c36e1..58a80d1556
* src/third_party/googletest/src: ea31cb15f0..0062e4869f
* src/third_party/libvpx/source/libvpx: ac54d233b6..edc9a46876
* src/third_party/libyuv: b792e0dbc1..439fc3ce3d
* src/tools: 4ee4992a32..d4fa619090
DEPS diff: 24887ed1ab..80343dd58e/DEPS

No update to Clang.

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

Change-Id: I04640b220654ddd2c79af55bc49058c65bd98184
Reviewed-on: https://webrtc-review.googlesource.com/53040
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22013}

2018-02-14 10:19:03 +00:00

api

Add UMA counters for type of SDP semantic in use.

2018-02-14 00:18:20 +00:00

audio

Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY.

2018-02-07 10:07:28 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Disable a couple flaky RampUpTest tests on Mac.

2018-02-12 23:17:51 +00:00

common_audio

Stop using public_deps in common_audio.

2018-02-06 09:44:20 +00:00

common_video

Prevent potential integer overflow in sps parser

2018-02-09 13:52:48 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

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

examples

Add RtcEventLog to AppRTCMobile with preference setting.

2018-02-12 19:12:05 +00:00

infra

Remove win_chromium_webrtc_compile_rel_ng from CQ

2018-02-01 19:26:24 +00:00

logging

Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY.

2018-02-07 10:07:28 +00:00

media

Assorted logging pedantry

2018-02-13 10:47:24 +00:00

modules

Revert "Moved congestion controller to goog_cc folder."

2018-02-14 10:09:44 +00:00

ortc

Change FakePeriodicVideoCapturer to use a TaskQueue instead of Thread.

2018-02-07 20:51:51 +00:00

p2p

Assorted logging pedantry

2018-02-13 10:47:24 +00:00

Prepare StatsCollector to work with RtpTransceivers

2018-02-14 03:42:04 +00:00

resources

Adding FourPeople_1280x720_30.yuv.

2018-02-12 15:55:00 +00:00

rtc_base

Assorted logging pedantry

2018-02-13 10:47:24 +00:00

rtc_tools

Revert "Moved congestion controller to goog_cc folder."

2018-02-14 10:09:44 +00:00

sdk

Fix rendering on an iPhone X's tall screen.

2018-02-14 09:17:55 +00:00

stats

Removing useless dependencies on //testing/gmock.

2018-01-26 13:34:12 +00:00

system_wrappers

Reland Use runtime enabled features API to enable dual stream mode

2018-01-18 12:22:49 +00:00

test

Fix infinite loop in rtp packet parsing

2018-02-13 14:42:45 +00:00

tools_webrtc

Add .vpython to WebRTC bot isolates.

2018-02-14 08:28:14 +00:00

video

Move creation of OveruseFrameDetector to VideoSendStream.

2018-02-13 15:45:03 +00:00

.clang-format

Tune ObjC clang-format configuration

2017-05-11 09:14:18 +00:00

.git-blame-ignore-revs

…

.gitignore

Reland "Make it possible to run video_quality_loopback_test in swarming."

2018-01-23 13:03:17 +00:00

.gn

Re-enabling libyuv 'gn check'.

2018-02-09 10:31:04 +00:00

.vpython

Add psutil to vpython dependencies (used on builder bots)

2017-09-04 08:04:16 +00:00

AUTHORS

Expose a link-local network interfaces enumeration option

2018-02-06 19:12:04 +00:00

BUILD.gn

Implements JavaToNativeStringMap and adds tests for native API.

2018-02-09 10:34:44 +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

Add flexfec payload name to string-type conversions

2018-01-31 08:58:39 +00:00

common_types.h

Remove unused fields from VideoCodecVP8.

2018-02-09 15:55:59 +00:00

DEPS

Roll chromium_revision 24887ed1ab..80343dd58e (535697:536688)

2018-02-14 10:19:03 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

…

LICENSE_THIRD_PARTY

Moving src/webrtc into src/.

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

native-api.md

Remove legacy VoiceEngine.

2018-01-12 11:31:52 +00:00

OWNERS

Add phoglund@ to various OWNERS and remove kjellander@

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

PATENTS

Moving src/webrtc into src/.

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

presubmit_test_mocks.py

Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks.

2017-10-13 03:48:26 +00:00

presubmit_test.py

Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks.

2017-10-13 03:48:26 +00:00

PRESUBMIT.py

Delete unused MediaFile module.

2018-01-29 11:18:18 +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

Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"""

2018-01-10 15:55:04 +00:00

typedefs.h

Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix

2018-02-05 11:24:59 +00:00

WATCHLISTS

Move remaining traces of VoiceEngine

2018-01-17 13:27:47 +00:00

webrtc.gni

Disable use of libevent under Fuchsia.

2018-02-13 08:23:48 +00:00

whitespace.txt

Moving src/webrtc into src/.

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