webrtc_m130

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chromium-webrtc-autoroll 3462793296 Roll chromium_revision 1d12ff693d..3cf04dec00 (696696:696812)

Change log: 1d12ff693d..3cf04dec00
Full diff: 1d12ff693d..3cf04dec00

Changed dependencies
* src/base: 4e24f6c092..03bceb0723
* src/build: e7f81b6504..36c63090ad
* src/ios: d8a0bae322..06351f1c5f
* src/testing: 15e0bc2f47..13f4bfd01e
* src/third_party: 3355b26c6e..a624039fb7
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ecd10922ee..914862e8ec
* src/third_party/freetype/src: 3de1b8d0b0..cc17f852d5
* src/tools: 3692d5fe84..1133315201
DEPS diff: 1d12ff693d..3cf04dec00/DEPS

Clang version changed 8455294f2ac13d587b13d728038a9bffa7185f2b:b4160cb94c54f0b31d0ce14694950dac7b6cd83f
Details: 1d12ff693d..3cf04dec00/tools/clang/scripts/update.py

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ia39d25cc4bb0666ae08bb94763f79e07de2849e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153030
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#29199}

2019-09-16 16:34:50 +00:00

api

Encoder switching based on network and/or resolution conditions.

2019-09-16 13:43:29 +00:00

audio

Introduce api/crypto/BUILD.gn.

2019-09-13 17:21:47 +00:00

build_overrides

Remove crbug.com/904400 workaround.

2019-03-15 18:36:23 +00:00

call

Delete flag VideoReceiveStream::Config::Rtp::remb

2019-09-16 11:20:55 +00:00

common_audio

Remove rtc_use_lto GN arg.

2019-08-20 14:00:49 +00:00

common_video

Implement Dependency Descriptor reader

2019-09-16 15:55:54 +00:00

data

Remove old data files.

2018-10-05 14:40:21 +00:00

docs

Fixing some typos.

2019-09-10 10:03:50 +00:00

examples

Adds setAudio[Track/Record]StateCallback interfaces to the Java ADM

2019-09-09 08:10:41 +00:00

logging

Delete deprecated rtc_event.h file

2019-09-16 15:59:54 +00:00

media

Encoder switching based on network and/or resolution conditions.

2019-09-16 13:43:29 +00:00

modules

Implement Dependency Descriptor reader

2019-09-16 15:55:54 +00:00

p2p

Introduce api/crypto/BUILD.gn.

2019-09-13 17:21:47 +00:00

Introduce api/crypto/BUILD.gn.

2019-09-13 17:21:47 +00:00

resources

Use the AEC3 high-pass filter for the whole APM

2019-08-23 20:04:10 +00:00

rtc_base

Store timestamp for each sample to be able to plot them in future

2019-09-11 11:13:49 +00:00

rtc_tools

Fix time units in plotted charts

2019-09-13 13:04:32 +00:00

sdk

Introduce api/crypto/BUILD.gn.

2019-09-13 17:21:47 +00:00

stats

Add qualityLimitationResolutionChanges stat

2019-09-09 15:22:57 +00:00

style-guide

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

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

system_wrappers

2019-09-10 12:34:38 +00:00

test

Delete flag VideoReceiveStream::Config::Rtp::remb

2019-09-16 11:20:55 +00:00

tools_webrtc

Stop explicitly setting use_prebuilt_instrumented_libraries on msan bots.

2019-09-12 18:21:38 +00:00

video

Encoder switching based on network and/or resolution conditions.

2019-09-16 13:43:29 +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 new format cleanup.

2019-07-11 16:18:51 +00:00

.gitignore

Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH.

2019-09-04 18:49:28 +00:00

.gn

Switch to compiling WebRTC -std=c++14 by default

2019-09-09 19:24:16 +00:00

.vpython

Add vpython dependencies needed to run presubmit tests on LUCI

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

abseil-in-webrtc.md

Allowing buffering a LNTF (loss notification) feedback message in RTCPSender

2019-06-03 16:28:34 +00:00

AUTHORS

Revert "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5."

2019-09-06 05:36:23 +00:00

BUILD.gn

Introduce api/crypto/BUILD.gn.

2019-09-13 17:21:47 +00:00

CODE_OF_CONDUCT.md

Add code of conduct to WebRTC repo

2017-05-16 12:09:13 +00:00

codereview.settings

Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit

2018-10-25 08:19:53 +00:00

common_types.h

Format almost everything.

2019-07-08 13:45:15 +00:00

DEPS

Roll chromium_revision 1d12ff693d..3cf04dec00 (696696:696812)

2019-09-16 16:34:50 +00:00

ENG_REVIEW_OWNERS

Enforce LGTM from owners of depends-on paths in DEPS via presubmit.

2018-09-28 12:49:54 +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

native-api.md

Delete unused I420 "codec"

2018-12-18 12:30:58 +00:00

OWNERS

Add juberti@ to webrtc root owners

2019-05-17 18:11:58 +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

Fixing py lint errors

2018-07-23 15:28:48 +00:00

PRESUBMIT.py

Move docs about native code development into a repo directory.

2019-09-10 09:42:26 +00:00

pylintrc

Fixing py lint errors

2018-07-23 15:28:48 +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

Update WebRTC's C++ style guide to reflect the switch to C++14.

2019-09-16 11:45:35 +00:00

WATCHLISTS

Add saza to audio watchlists

2019-09-03 14:55:43 +00:00

webrtc.gni

Remove rtc_use_lto GN arg.

2019-08-20 14:00:49 +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%