platform-external-webrtc

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Autoroller b044d3a28f Roll chromium_revision e5132b72fb..63c80f2e33 (556255:556358)

Change log: e5132b72fb..63c80f2e33
Full diff: e5132b72fb..63c80f2e33

Changed dependencies:
* src/base: 9e6250b93e..6772872e6c
* src/build: 0a40aa9e7a..1fd2d08a6f
* src/ios: 65414e754c..2a4cc722a7
* src/testing: 6c6b879b06..9c2c516d0f
* src/third_party: 569ccecd65..b29bfab599
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/39e3bc6ed3..3059fd7dba
* src/third_party/depot_tools: 142a92ce8b..5e5f2d6035
* src/tools: 1c56a94eb1..1293f814ef
DEPS diff: e5132b72fb..63c80f2e33/DEPS

No update to Clang.

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

Change-Id: Ic61f2643bce860c7ec9d965568be7345aaa5f078
Reviewed-on: https://webrtc-review.googlesource.com/74695
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23133}

2018-05-06 20:22:56 +00:00

api

Implement RtpParameters.transaction_id for PC RtpSenderInterface

2018-05-04 13:07:25 +00:00

audio

Uses config struct with bitrate allocator.

2018-04-19 09:15:33 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Moving task queue from Call to transport controller.

2018-05-04 12:44:44 +00:00

common_audio

Disabling openmax_dl

2018-04-24 13:08:56 +00:00

common_video

Move BitrateAllocation to api/ and rename it VideoBitrateAllocation

2018-04-23 15:31:27 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

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

examples

Fix: make sure to call UI APIs from main thread.

2018-05-03 14:14:11 +00:00

infra

Revert "Disable internal lite bot."

2018-04-11 06:45:07 +00:00

logging

Log probing bitrate and cluster id as int32 in event log.

2018-05-04 14:40:44 +00:00

media

Fixed init of H.264 profile/level support table on iOS

2018-05-04 14:56:34 +00:00

modules

AEC3: External delay - Fix mismatch in time units

2018-05-04 16:52:24 +00:00

ortc

Disable flaky OrtcFactoryIntegrationTests on debug iOS 64 builds.

2018-04-24 13:17:06 +00:00

p2p

Change the receiving state update rule.

2018-04-17 01:00:02 +00:00

Integrated PeerConnectionDependencies into PeerConnection::Initialize

2018-05-04 19:18:05 +00:00

resources

Revert "Reland "AGC2 RNN VAD: Recurrent Neural Network impl""

2018-05-04 11:52:26 +00:00

rtc_base

Fix VerifyPeerCertMatchesHost tests.

2018-05-04 17:14:14 +00:00

rtc_tools

Split NetEq simulation and jitter buffer plot to be able to plot other metrics in the simulation.

2018-05-04 15:47:24 +00:00

sdk

Fixed init of H.264 profile/level support table on iOS

2018-05-04 14:56:34 +00:00

stats

PeerConnectionInterface::GetStats() with selector argument added.

2018-03-26 12:08:20 +00:00

style-guide

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

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

system_wrappers

Clean up posix-only files.

2018-04-18 00:18:04 +00:00

test

Moving demux from FakeNetworkPipe to DirectTransport.

2018-04-25 10:13:03 +00:00

tools_webrtc

Revert "Reland "AGC2 RNN VAD: Recurrent Neural Network impl""

2018-05-04 11:52:26 +00:00

video

Break out the part of the iSAC codec that's used for Voice Activity Detection

2018-05-04 08:53:34 +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

Opt out of "Migrate the Android Support Lib to android_deps".

2018-04-05 13:40:53 +00:00

.vpython

vpython: Specify dependency on pywin32

2018-02-14 13:56:39 +00:00

AUTHORS

Use IFA_LOCAL instead of IFA_ADDRESS over IPv4 network on ANDROID

2018-04-27 14:16:01 +00:00

BUILD.gn

Mark built-in software video codecs as poisonous.

2018-04-25 11:34:33 +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

Don't expose resilience mode in VP8 and VP9 configuration.

2018-04-25 13:54:33 +00:00

common_types.h

Reland "Reland "Remove our stream << overloads from non-test build targets.""

2018-05-03 10:41:41 +00:00

DEPS

Roll chromium_revision e5132b72fb..63c80f2e33 (556255:556358)

2018-05-06 20:22:56 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

…

LICENSE_THIRD_PARTY

Remove custom MD5 / SHA-1 implementations.

2018-02-19 15:03:35 +00:00

native-api.md

Remove legacy VoiceEngine.

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

OWNERS

Add back OWNERS entris that went missing

2018-03-06 10:31:11 +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

Add ToString() methods to classes with << operators, preparing for deprecations.

2018-04-03 11:21:30 +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

Add style guidance about forward declarations.

2018-03-28 20:58:27 +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

Fixing root_files WATCHLIST regex.

2018-04-19 06:52:18 +00:00

webrtc.gni

Mark built-in software video codecs as poisonous.

2018-04-25 11:34:33 +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

Languages

C++ 88.6%

C 3.3%

Java 3%

Objective-C++ 1.9%

Python 1.9%

Other 1%