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chromium-webrtc-autoroll d942282c44 Roll chromium_revision 02833e653c..2e1ac8de05 (705539:705654)

Change log: 02833e653c..2e1ac8de05
Full diff: 02833e653c..2e1ac8de05

Changed dependencies
* src/base: b5865c85b3..0fc56f3318
* src/build: 0548583d8f..eff336ca2e
* src/ios: 555b81336f..17c89cddc2
* src/testing: 16b957e154..7c5e65993f
* src/third_party: 517c42c71a..942ed47fe8
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/6a2609dae2..bc4c09df64
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/c7af59115e..0b3653782e
* src/third_party/depot_tools: 5abb9b7c5d..033004652f
* src/tools: 1759fbce87..9299893f6d
DEPS diff: 02833e653c..2e1ac8de05/DEPS

No update to Clang.

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

Change-Id: I29130b2e1ebd4090768dbdde87c7cb5f445c9367
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157000
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@{#29471}

2019-10-14 21:03:18 +00:00

api

Add a turn port prune policy to keep the first ready turn port.

2019-10-14 19:08:23 +00:00

audio

Fix GetStats bytesSent/Received, wireup headerBytesSent/Received

2019-10-14 13:07:13 +00:00

build_overrides

Remove crbug.com/904400 workaround.

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

call

Fix GetStats bytesSent/Received, wireup headerBytesSent/Received

2019-10-14 13:07:13 +00:00

common_audio

Use std::make_unique instead of absl::make_unique.

2019-09-17 15:47:29 +00:00

common_video

Always pass arguments to INSTANTIATE_TEST_SUITE_P.

2019-09-30 12:52:07 +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

New build target api:media_interface

2019-09-19 09:32:27 +00:00

logging

Cleanup includes in modules/include/module_common_types.h

2019-10-07 16:06:26 +00:00

media

Fix GetStats bytesSent/Received, wireup headerBytesSent/Received

2019-10-14 13:07:13 +00:00

modules

Adds logging of NetworkStateEstimator estimates.

2019-10-14 15:42:17 +00:00

p2p

Add a turn port prune policy to keep the first ready turn port.

2019-10-14 19:08:23 +00:00

Add a turn port prune policy to keep the first ready turn port.

2019-10-14 19:08:23 +00:00

resources

Delete voice_detection() pointer to submodule

2019-10-07 13:06:05 +00:00

rtc_base

Use GlobalLock to protect logging

2019-10-14 16:43:19 +00:00

rtc_tools

Plot delay feedback in RTCP arrival order.

2019-10-11 13:52:10 +00:00

sdk

Add a turn port prune policy to keep the first ready turn port.

2019-10-14 19:08:23 +00:00

stats

Fix GetStats bytesSent/Received, wireup headerBytesSent/Received

2019-10-14 13:07:13 +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

Add a turn port prune policy to keep the first ready turn port.

2019-10-14 19:08:23 +00:00

tools_webrtc

Stop using goma for MSVC bots.

2019-10-08 15:19:17 +00:00

video

Use bitrate limits provided by encoder.

2019-10-14 12:57:24 +00:00

.clang-format

…

.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

Update style guide for absl::make_unique.

2019-09-18 06:10:58 +00:00

AUTHORS

SetStreams API of RtpSender wrapped for iOS and Android

2019-10-08 13:51:19 +00:00

BUILD.gn

Add missing dependencies to the static library.

2019-10-10 08:22:59 +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 02833e653c..2e1ac8de05 (705539:705654)

2019-10-14 21:03:18 +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

…

native-api.md

Delete unused I420 "codec"

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

OWNERS

Add #COMPONENT to WebRTC.

2019-10-08 12:20:39 +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

absl::make_unique presubmit check.

2019-09-17 17:47:31 +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++ 88.6%

C 3.3%

Java 3%

Objective-C++ 1.9%

Python 1.9%

Other 1%