webrtc_m130

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Autoroller d48b5f5a41 Roll chromium_revision a0e2753f75..3668886840 (575203:575305)

Change log: a0e2753f75..3668886840
Full diff: a0e2753f75..3668886840

Changed dependencies:
* src/base: 821dd575a1..24eccdc599
* src/build: c0771e95b8..59d0512879
* src/ios: cb26b2b3bc..71fbd9970f
* src/third_party: d089c514a2..c052d35886
* src/tools: 30ba26f846..21fc68bfed
DEPS diff: a0e2753f75..3668886840/DEPS

No update to Clang.

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

Change-Id: I99e7f9e905ad4d2b32bae24cf5e4de7c2d184361
Reviewed-on: https://webrtc-review.googlesource.com/88801
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23987}

2018-07-16 17:08:27 +00:00

api

Move allocation and rtp conversion logic out of payload router.

2018-07-16 13:34:37 +00:00

audio

Better handle target audio bitrate allocation.

2018-07-13 09:06:06 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

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

call

Move allocation and rtp conversion logic out of payload router.

2018-07-16 13:34:37 +00:00

common_audio

Remove useless import of arm.gni

2018-07-12 14:39:00 +00:00

common_video

Add bit depth information to PlanarYuvBuffer

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

data

WebRTC: Replace ProjectRootPath by ResourcePath

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

examples

Android: Remove use of deprecated functions related to legacy video codecs

2018-07-16 09:28:07 +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

Reland "Reland "Add Profile 2 configuration to VP9 Encoder and Decoder""

2018-07-14 06:51:20 +00:00

modules

AEC3: Adding explicit handling of microphone gain changes

2018-07-16 16:02:07 +00:00

ortc

RtpReceiverInterface::stream_ids() added.

2018-07-11 10:14:56 +00:00

p2p

Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""

2018-07-12 21:35:47 +00:00

Re-enable skipped test.

2018-07-16 07:31:07 +00:00

resources

AGC2 RNN VAD: Polishing.

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

rtc_base

Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""

2018-07-12 21:35:47 +00:00

rtc_tools

Break out Agc code from audio_processing.

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

sdk

Fixed crash when PCF is destroyed before MediaSource/Track in ObjC

2018-07-16 12:03:16 +00:00

stats

Removing clang:find_bad_constructs from stats/.

2018-07-16 14:27:07 +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 find_bad_constructs from system_wrappers.

2018-07-16 14:44:47 +00:00

test

AEC3: Adding explicit handling of microphone gain changes

2018-07-16 16:02:07 +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

Reland "Reland "Add Profile 2 configuration to VP9 Encoder and Decoder""

2018-07-14 06:51:20 +00:00

.clang-format

Tune ObjC clang-format configuration

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

.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

Move allocation and rtp conversion logic out of payload router.

2018-07-16 13:34:37 +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.h

Reland "Remove rtc::Optional alias and api:optional target"

2018-07-11 19:02:51 +00:00

DEPS

Roll chromium_revision a0e2753f75..3668886840 (575203:575305)

2018-07-16 17:08:27 +00:00

LICENSE

Moving src/webrtc into src/.

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

license_template.txt

Update template to follow chromium copyright style

2013-04-24 01:01:28 +00:00

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

Adding CheckNoWarningSuppressionFlagsAreAdded.

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