webrtc_m130

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Autoroller 5864639a46 Roll chromium_revision bb3f839eb8..b2c5a55fc1 (530955:531202)

Change log: bb3f839eb8..b2c5a55fc1
Full diff: bb3f839eb8..b2c5a55fc1

Changed dependencies:
* src/base: 310ce2c960..f3eff4526a
* src/build: ea310ceb2f..73b05cab8b
* src/ios: aa417958a5..1f32969ab5
* src/testing: 1c94cd6747..2e1c1c949f
* src/third_party: bbed5fe807..44b9ae20e3
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/0a20f3ce6c..b176429567
* src/tools: 3ffd21067e..3b711e7752
DEPS diff: bb3f839eb8..b2c5a55fc1/DEPS

No update to Clang.

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

Change-Id: Ibf4934229774d5f23055db59b3b606736c62c261
Reviewed-on: https://webrtc-review.googlesource.com/43100
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21732}

2018-01-23 11:40:16 +00:00

api

Move MediaConfig to its own header file and target.

2018-01-23 11:02:16 +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

Make fec controller plug-able.

2018-01-22 11:48:16 +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

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

2018-01-22 16:12:10 +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 MediaConfig to its own header file and target.

2018-01-23 11:02:16 +00:00

modules

Make screenshare_layers frame dropper less aggressive

2018-01-22 21:15:29 +00:00

ortc

Configure media flow correctly with Unified Plan

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

p2p

Replace bind2nd with lambdas in turnport.cc for C++ 17 compatibility.

2018-01-19 20:34:22 +00:00

Move MediaConfig to its own header file and target.

2018-01-23 11:02:16 +00:00

resources

Add a new NetEq decoding unit test for Opus with DTX

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

rtc_base

Add functions to securely fill memory with zeros.

2018-01-20 16:57:27 +00:00

rtc_tools

Making rtc_tools:unpack_aecdump publicly visible.

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

sdk

Forward fix jni_generator_helper.h.

2018-01-23 10:33:26 +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

Get rid of packet loss related stuff from videoprocessor.

2018-01-22 15:45:58 +00:00

tools_webrtc

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

2018-01-22 16:12:10 +00:00

video

Avoid infinite recursion if a RED packet encapsulate a RED packet.

2018-01-22 14:51:34 +00:00

.clang-format

Tune ObjC clang-format configuration

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

.git-blame-ignore-revs

…

.gitignore

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

2018-01-22 16:12:10 +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 bb3f839eb8..b2c5a55fc1 (530955:531202)

2018-01-23 11:40:16 +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%