platform-external-webrtc

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Autoroller 26ef7cde11 Roll chromium_revision b0784bef91..86f8273e2f (581665:581886)

Change log: b0784bef91..86f8273e2f
Full diff: b0784bef91..86f8273e2f

Changed dependencies:
* src/base: 76793f5417..6abc23a627
* src/build: 03ce38cdd9..1f2ff68a92
* src/ios: 7c0de39a86..a128bdd3fc
* src/testing: 72f3763b1f..dcefe6ada5
* src/third_party: 28c1ea1319..05e0f49727
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9aa552b157..bca7d20ad7
* src/tools: 93447e2d7b..da25d064ae
DEPS diff: b0784bef91..86f8273e2f/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None

Change-Id: Ic467e12304b3093b40297ce57421e25b04eba0e1
Reviewed-on: https://webrtc-review.googlesource.com/93340
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24255}

2018-08-10 09:33:05 +00:00

api

Remove rtc::{Make,Wrap}Unique and their header file + unit tests

2018-08-08 10:45:52 +00:00

audio

Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver.

2018-08-08 11:45:21 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Reland "Refactor RtpVideoStreamReceiver without RtpReceiver."

2018-08-10 08:56:49 +00:00

common_audio

Move spl_sqrt_floor dep to proper third_party directory

2018-08-07 10:33:13 +00:00

common_video

Delete root header file typedef.h.

2018-07-25 14:59:26 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

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

examples

Remove unnecessary //base:base_java dependencies.

2018-08-06 12:04:35 +00:00

infra

cq_name is no longer used and can and should be removed

2018-08-09 12:12:35 +00:00

logging

Remove memcheck.

2018-08-07 07:40:08 +00:00

media

Experimental improvements for simulcast screenshare

2018-08-09 15:10:55 +00:00

modules

Reland "Refactor RtpVideoStreamReceiver without RtpReceiver."

2018-08-10 08:56:49 +00:00

ortc

Delete root header file typedef.h.

2018-07-25 14:59:26 +00:00

p2p

Remove definition and usage of macro GTEST_RELATIVE_PATH.

2018-08-08 11:00:11 +00:00

Delete struct webrtc::PacketTime.

2018-08-07 10:07:15 +00:00

resources

Fix gitignore pattern to never exclude sha1 files

2018-08-03 12:45:45 +00:00

rtc_base

Remove old base64 header

2018-08-09 10:06:41 +00:00

rtc_tools

Remove definition and usage of macro GTEST_RELATIVE_PATH.

2018-08-08 11:00:11 +00:00

sdk

Android: Allow YuvConverter to be reused

2018-08-09 12:36:37 +00:00

stats

Removing clang:find_bad_constructs from stats/.

2018-07-16 14:27:07 +00:00

style-guide

Add style guide rule about paired .h and .cc files

2018-03-14 13:02:35 +00:00

system_wrappers

Delete root header file typedef.h.

2018-07-25 14:59:26 +00:00

test

Remove RTPVideoHeader::vp9() accessors.

2018-08-09 10:53:28 +00:00

tools_webrtc

Add post-submit builders without dcheck_always_on

2018-08-09 11:29:15 +00:00

video

Reland "Refactor RtpVideoStreamReceiver without RtpReceiver."

2018-08-10 08:56:49 +00:00

.clang-format

Tune ObjC clang-format configuration

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

.git-blame-ignore-revs

Let git-hyper-blame ignore format commit.

2018-06-20 09:26:44 +00:00

.gitignore

Roll back checking in the third_party directory

2018-06-27 13:04:08 +00:00

.gn

Roll chromium_revision 474eca0589..b0784bef91 (581204:581665)

2018-08-09 12:11:24 +00:00

.vpython

Add vpython dependencies needed to run presubmit tests on LUCI

2018-05-18 08:10:25 +00:00

AUTHORS

Fixed issue with BGRA RTCCVPixelBuffer scale and crop

2018-07-24 08:23:26 +00:00

BUILD.gn

Remove definition and usage of macro GTEST_RELATIVE_PATH.

2018-08-08 11:00:11 +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

Delete struct webrtc::PacketTime.

2018-08-07 10:07:15 +00:00

DEPS

Roll chromium_revision b0784bef91..86f8273e2f (581665:581886)

2018-08-10 09:33:05 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

…

native-api.md

Remove legacy VoiceEngine.

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

OWNERS

Add mbonadei@ to build configs OWNERS.

2018-06-20 12:39:11 +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

Fixing py lint errors

2018-07-23 15:28:48 +00:00

PRESUBMIT.py

Remove memcheck.

2018-08-07 07:40:08 +00:00

pylintrc

Fixing py lint errors

2018-07-23 15:28:48 +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

WATCHLISTS

Adding mbonadei@ to build_files WATCHLIST.

2018-06-20 12:38:06 +00:00

webrtc.gni

Remove memcheck.

2018-08-07 07:40:08 +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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Java 3%

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Python 1.9%

Other 1%