webrtc_m130

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Autoroller 74d0ee56f5 Roll chromium_revision df2aa84fb3..8cfb6c74d5 (530479:530584)

Change log: df2aa84fb3..8cfb6c74d5
Full diff: df2aa84fb3..8cfb6c74d5

Changed dependencies:
* src/base: 4fa4f2e7b6..067c7f2d28
* src/build: 972ab23cfd..7a820cd77e
* src/ios: e2f335a5df..36c7edea5b
* src/testing: f8b6da6158..17532dc4aa
* src/third_party: 0859e50e39..83c585ea19
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b4706e7320..3318544c51
* src/tools: cbb25e893f..959729d3d8
DEPS diff: df2aa84fb3..8cfb6c74d5/DEPS

No update to Clang.

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

Change-Id: I52e6fa3df416a88a9ffc38333faf99b74483e48d
Reviewed-on: https://webrtc-review.googlesource.com/41360
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21700}

2018-01-19 20:12:03 +00:00

api

Move AudioOptions to its own header file and target.

2018-01-19 13:00:32 +00:00

audio

Move remaining traces of VoiceEngine

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

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Reland Use runtime enabled features API to enable dual stream mode

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

common_audio

Replace left shift with equivalent multiplication.

2018-01-12 15:29:59 +00:00

common_video

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

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

data

WebRTC: Replace ProjectRootPath by ResourcePath

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

examples

Reland Use runtime enabled features API to enable dual stream mode

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

infra

Make new Chromium compile trybots mandatory.

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

logging

Remove dependency on system_wrappers from rtc_event_log.

2018-01-19 18:04:03 +00:00

media

Move AudioOptions to its own header file and target.

2018-01-19 13:00:32 +00:00

modules

APM-QA anntator for sound level measurement

2018-01-19 17:26:22 +00:00

ortc

Configure media flow correctly with Unified Plan

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

p2p

Delete ConnectionMonitor.

2018-01-18 08:03:27 +00:00

Make PeerConnection take reference to UMA observer.

2018-01-18 20:07:58 +00:00

resources

Add a new NetEq decoding unit test for Opus with DTX

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

rtc_base

Remove improper build_with_mozilla clauses in rtc_base BUILD.gn

2018-01-17 16:44:47 +00:00

rtc_tools

Making rtc_tools:unpack_aecdump publicly visible.

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

sdk

Add MediaTek H264 and VP8 HW Codec Support with field trial

2018-01-19 19:58:03 +00:00

stats

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

2018-01-10 15:55:04 +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 warning 4373.

2018-01-19 10:37:44 +00:00

tools_webrtc

Remove unused manifest.

2018-01-18 14:24:48 +00:00

video

Remove dependency on system_wrappers from rtc_event_log.

2018-01-19 18:04:03 +00:00

.clang-format

Tune ObjC clang-format configuration

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

.git-blame-ignore-revs

Create .git-blame-ignore-revs and add Java format CL to it.

2016-10-20 09:20:39 +00:00

.gitignore

Roll chromium_revision f93b8b19f2..adf969a7cb (513366:514871) and more.

2017-11-08 21:42:48 +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

RemoteBitrateEstimatorAbsSendTime: check clock is a valid ref

2018-01-16 01:27:11 +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

Adds the active field to SimulcastStream and VideoCodec structs.

2018-01-17 23:36:47 +00:00

common_types.h

Adds the active field to SimulcastStream and VideoCodec structs.

2018-01-17 23:36:47 +00:00

DEPS

Roll chromium_revision df2aa84fb3..8cfb6c74d5 (530479:530584)

2018-01-19 20:12:03 +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

Remove legacy VoiceEngine.

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