webrtc_m130

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Autoroller 2036da9cd8 Roll chromium_revision da763f1ccd..34ad909848 (532638:532738)

Change log: da763f1ccd..34ad909848
Full diff: da763f1ccd..34ad909848

Changed dependencies:
* src/base: 2a3cdc3126..352cd788c1
* src/build: 33dd9b84fd..6491c4c2c8
* src/ios: e18677caa3..9deb3fd568
* src/testing: 1d4c820fb9..cd2b6a1191
* src/third_party: 131ce7168b..3694484c69
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/94cd196a80..a62dbf88d8
* src/third_party/depot_tools: 7a4ced2773..6fe29419be
* src/tools: 6a5902dcd8..f3b128409c
DEPS diff: da763f1ccd..34ad909848/DEPS

No update to Clang.

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

Change-Id: Id25046592f26325892cbe77380f49d64dc31382d
Reviewed-on: https://webrtc-review.googlesource.com/45921
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21796}

2018-01-30 03:34:13 +00:00

api

Removing useless dependencies on //testing/gmock.

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

audio

Removing useless dependencies on //testing/gmock.

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

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Use rtc::ToString instead of std::to_string

2018-01-29 19:26:09 +00:00

common_audio

Removing useless dependencies on //testing/gmock.

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

common_video

Reland "Rename stereo video codec to multiplex"

2018-01-29 20:37:59 +00:00

data

…

examples

Increasing "SERVER_WAIT" for TCPChannelClient tests.

2018-01-29 19:39:49 +00:00

infra

Make new Chromium compile trybots mandatory.

2018-01-12 09:41:51 +00:00

logging

Removing useless dependencies on //testing/gmock.

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

media

Reland "Rename stereo video codec to multiplex"

2018-01-29 20:37:59 +00:00

modules

Reland "Rename stereo video codec to multiplex"

2018-01-29 20:37:59 +00:00

ortc

Configure media flow correctly with Unified Plan

2018-01-18 19:01:38 +00:00

p2p

Implemented the GetRemoteAudioSSLCertificate method.

2018-01-27 23:48:36 +00:00

Implemented the GetRemoteAudioSSLCertificate method.

2018-01-27 23:48:36 +00:00

resources

Add a new NetEq decoding unit test for Opus with DTX

2017-11-28 10:45:38 +00:00

rtc_base

Propagate jsoncpp include path to depenent targets.

2018-01-29 09:50:18 +00:00

rtc_tools

Making rtc_tools:unpack_aecdump publicly visible.

2018-01-11 09:04:30 +00:00

sdk

Never use surface mode in MediaCodecVideoEncoder if egl_context_ is null.

2018-01-29 11:03:37 +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

Reland "Rename stereo video codec to multiplex"

2018-01-29 20:37:59 +00:00

tools_webrtc

Parameterize PeerConnection end to end tests for Unified Plan

2018-01-25 04:03:18 +00:00

video

Reland "Rename stereo video codec to multiplex"

2018-01-29 20:37:59 +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

Pinning NDK api levels.

2018-01-18 16:55:58 +00:00

.vpython

Add psutil to vpython dependencies (used on builder bots)

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

AUTHORS

Add missing stdio.h header in files using scanf/sscanf function.

2018-01-26 13:15:52 +00:00

BUILD.gn

Reland Use runtime enabled features API to enable dual stream mode

2018-01-18 12:22:49 +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

Reland "Rename stereo video codec to multiplex"

2018-01-29 20:37:59 +00:00

common_types.h

Reland "Rename stereo video codec to multiplex"

2018-01-29 20:37:59 +00:00

DEPS

Roll chromium_revision da763f1ccd..34ad909848 (532638:532738)

2018-01-30 03:34:13 +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

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

WATCHLISTS

Move remaining traces of VoiceEngine

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

webrtc.gni

Update gn files to support Mozilla build

2018-01-16 07:51:23 +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++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%