webrtc_m130

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Autoroller cf5de1da07 Roll chromium_revision a88423acf9..a714568fbe (571410:571512)

Change log: a88423acf9..a714568fbe
Full diff: a88423acf9..a714568fbe

Changed dependencies:
* src/base: 311c937b26..8c01d4ef25
* src/build: c9333f9faf..06960ce32c
* src/buildtools: 9c9fd97928..aec56e2607
* src/ios: 34302909a8..dc2780aff3
* src/testing: b47e929d27..5e63b2909b
* src/third_party: b77d94a9b3..42df0ae52f
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e2d6bbca62..f76f0b4406
* src/third_party/depot_tools: ae1f03388f..406de133ef
* src/tools: 6ff0d88db8..0597286f57
DEPS diff: a88423acf9..a714568fbe/DEPS

Clang version changed 335608:335864
Details: a88423acf9..a714568fbe/tools/clang/scripts/update.py

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

Change-Id: I85ce46b41d6cd2302fef17a86025e4b5eb913a8b
Reviewed-on: https://webrtc-review.googlesource.com/86380
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23792}

2018-06-29 17:20:14 +00:00

api

Add PeerConnection GetRtpSender/ReceiverCapabilities

2018-06-28 12:40:07 +00:00

audio

Remove StreamStatistician::IsPacketInOrder

2018-06-28 08:44:40 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html ) for audio stream.

2018-06-27 10:33:40 +00:00

common_audio

Extract fft4g into separate build target

2018-06-26 13:39:25 +00:00

common_video

Add bit depth information to PlanarYuvBuffer

2018-06-26 20:23:37 +00:00

data

…

examples

Replace rtc::Optional with absl::optional

2018-06-21 09:32:56 +00:00

infra

Flip luci.webrtc.try to production

2018-05-30 08:30:00 +00:00

logging

Port RtcEventLog encoder unittests to the new parser API.

2018-06-28 15:31:23 +00:00

media

2018-06-27 10:33:40 +00:00

modules

Removes redundant AlrDetector.

2018-06-29 16:28:04 +00:00

ortc

Add PeerConnection GetRtpSender/ReceiverCapabilities

2018-06-28 12:40:07 +00:00

p2p

Remove the flag PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS.

2018-06-27 00:43:09 +00:00

Enable PeerConnectionEndToEndTest.CallWithLegacySdp on ASan.

2018-06-29 05:00:46 +00:00

resources

AGC2 RNN VAD: Polishing.

2018-05-15 16:41:02 +00:00

rtc_base

Turn rtc::{Make,Wrap}Unique into aliases for their Abseil counterparts

2018-06-29 13:48:33 +00:00

rtc_tools

Adding NetEq lifetime stats to event log visualizer.

2018-06-26 11:27:09 +00:00

sdk

Detach audio devices from thread on terminate.

2018-06-29 12:00:17 +00:00

stats

Adding "is_standardized" flag to RTCStatsMember.

2018-06-28 00:43:46 +00:00

style-guide

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

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

system_wrappers

Use the sparse histogram in RTC_HISTOGRAM_ENUMERATION_SPARSE.

2018-06-27 00:41:29 +00:00

test

Add Generic frame descritpor header extension

2018-06-29 15:02:44 +00:00

tools_webrtc

Roll back checking in the third_party directory

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

video

Remove StreamStatistician::IsPacketInOrder

2018-06-28 08:44:40 +00:00

.clang-format

…

.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

Set gtest_enable_absl_printers to true.

2018-06-29 09:36:17 +00:00

.vpython

Add vpython dependencies needed to run presubmit tests on LUCI

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

AUTHORS

Generalize SimulcastEncoderAdapter, use for H264 & VP8.

2018-06-21 15:57:43 +00:00

BUILD.gn

A new PeerConnection level perf test.

2018-06-27 23:19:05 +00:00

CODE_OF_CONDUCT.md

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codereview.settings

Make Gerrit the default for WebRTC changes

2017-09-29 01:38:07 +00:00

common_types.h

Reformat the WebRTC code base

2018-06-19 14:00:39 +00:00

DEPS

Roll chromium_revision a88423acf9..a714568fbe (571410:571512)

2018-06-29 17:20:14 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

…

LICENSE_THIRD_PARTY

Remove third party dependecies that are not more in the source code

2018-06-21 11:33:41 +00:00

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

Roll chromium_revision 95336cb92b..191d55580e (557816:557824)

2018-05-11 11:17:05 +00:00

PRESUBMIT.py

Roll back checking in the third_party directory

2018-06-27 13:04:08 +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

Remove typedefs.h from webrtc/ root (part 1)

2018-05-23 12:07:10 +00:00

WATCHLISTS

Adding mbonadei@ to build_files WATCHLIST.

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

webrtc.gni

Don't call deprecated FFmpeg API.

2018-06-26 13:57:35 +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++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%