webrtc_m130

Go to file

Autoroller b41b11bb99 Roll chromium_revision 5ce8880baa..ef3275516a (573458:573561)

Change log: 5ce8880baa..ef3275516a
Full diff: 5ce8880baa..ef3275516a

Changed dependencies:
* src/base: 19fde59a04..a13625cac6
* src/ios: f3eb8ce122..90c50414df
* src/testing: 90c261a3c9..1165763e7e
* src/third_party: 4779a4c250..21280f5bf9
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/f6618d3f74..21acb2a822
* src/third_party/depot_tools: 92745a5b0e..c3975e55e1
* src/tools: c04857d6e1..81f121bedf
DEPS diff: 5ce8880baa..ef3275516a/DEPS

No update to Clang.

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

Change-Id: I13ce4ab3c36bbaa6180e53bd777c081881212b0e
Reviewed-on: https://webrtc-review.googlesource.com/87801
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23893}

2018-07-10 01:18:38 +00:00

api

AEC3: Slower adaptation of main filter

2018-07-05 14:37:27 +00:00

audio

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

build_overrides

…

call

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

common_audio

Removing warning suppression flags in common_audio/.

2018-07-04 11:13:09 +00:00

common_video

Add bit depth information to PlanarYuvBuffer

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

data

…

examples

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

infra

Flip luci.webrtc.try to production

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

logging

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

media

Lower TL0 bitrate to 40% for lowest stream when using short 3tl

2018-07-08 12:17:58 +00:00

modules

Remove a legacy DCHEC in FakeDecodeFromFile.

2018-07-09 19:56:58 +00:00

ortc

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

p2p

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

Use absl::make_unique and absl::WrapUnique directly

2018-07-05 10:59:49 +00:00

resources

AGC2 RNN VAD: Polishing.

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

rtc_base

Delegate WIN32_LEAN_AND_MEAN definition to the toolchain.

2018-07-09 19:54:28 +00:00

rtc_tools

Break out Agc code from audio_processing.

2018-07-06 13:29:43 +00:00

sdk

Adds PeerConnectionDependencies to Java API.

2018-07-09 23:57:05 +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

Removing /wd4334 from system_wrappers.

2018-07-05 09:18:34 +00:00

test

Limit input size to iSAC and iLBC decoder fuzzers

2018-07-05 14:30:12 +00:00

tools_webrtc

roll_deps: Accept any prefix (like 'git_revision:'), not only 'version:' for CIPD

2018-07-03 09:41:53 +00:00

video

Stop using VideoCodec.targetBitrate for vp8 screenshare config

2018-07-06 15:13:18 +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

…

codereview.settings

…

common_types.h

Reformat the WebRTC code base

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

DEPS

Roll chromium_revision 5ce8880baa..ef3275516a (573458:573561)

2018-07-10 01:18:38 +00:00

LICENSE

…

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

…

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

Adding CheckNoWarningSuppressionFlagsAreAdded.

2018-07-04 09:01:32 +00:00

pylintrc

…

README.chromium

…

README.md

…

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%