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chromium-webrtc-autoroll befa435375 Roll chromium_revision 022da7ca60..5395db8bba (716991:718200) 
Change log: 022da7ca60..5395db8bba
Full diff: 022da7ca60..5395db8bba

Changed dependencies
* src/base: 29c24a8b46..7f873bc1ba
* src/build: c60d0d9d66..9b5f0ee593
* src/ios: 01eee3d70d..b3575ac7bb
* src/testing: a6c25af3b7..2ef35136de
* src/third_party: 24a1541bd5..044439b650
* src/third_party/android_build_tools/bundletool: D5lTGqnC49aEB2WwySxcHjPzhSMmkyVTB-vEupzVvXsC..VVve-571EEzx-giwEOU0vCrIe9D9a_TjP6ka1GgeVSUC
* src/third_party/android_ndk: https://chromium.googlesource.com/android_ndk.git/+log/89e8db0cdf..27c0a8d090
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/55394ddc90..258f61029f
* src/third_party/depot_tools: 639872c8ef..c50b0963e3
* src/third_party/r8: 7iz_2pdTN2RZRzgoVnxCi1Ro0iUSsEsvXGgmBgXG6z4C..-oXGY8FjY2ZuIBHoGAByn8N6Vn2b0wB2QO8Ct_169XoC
* src/tools: f2ed5a7b38..19ba6a2c54
DEPS diff: 022da7ca60..5395db8bba/DEPS

Clang version changed 64a362e7216a43e3ad44e50a89265e72aeb14294:e84b7a5fe230e42b8e6fe451369874a773bf1867
Details: 022da7ca60..5395db8bba/tools/clang/scripts/update.py

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

Change-Id: Icb5d06669ab3e4bac5d973883691498def0e6d17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160323
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@{#29881}
2019-11-22 18:49:12 +00:00
api
Add TimeController to api/test/ and add a CreateTimeController API.
2019-11-22 17:07:23 +00:00
audio
Adding VoIP specific channel adjustments
2019-11-22 15:25:05 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Adds injectable trials from peerconnection down to transport controller.
2019-11-21 12:41:45 +00:00
common_audio
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
common_video
Revert "VideoFrame: Store a reference to an encoded frame"
2019-11-21 14:55:21 +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
Add @Nullable annotations to quiet errorprone.
2019-11-19 12:50:30 +00:00
logging
Detach thread checker in RtcEventLogImpl destructor.
2019-11-13 15:42:19 +00:00
media
Nuke p2p/base/stun.h
2019-11-22 14:02:10 +00:00
modules
Add multi-channel support to AECM
2019-11-22 17:16:53 +00:00
p2p
Move IceFieldTrials into own .h-file
2019-11-22 15:01:15 +00:00
pc
VideoRtpReceiver: factor out VideoRtpTrackSource.
2019-11-22 10:13:59 +00:00
resources
Reland "Refactoring of the noise suppressor and adding true multichannel support"
2019-10-31 11:56:01 +00:00
rtc_base
Revert lock for logging to CriticalSection
2019-11-21 12:09:25 +00:00
rtc_tools
EventLogVisualizer pacer delay plot fix.
2019-11-19 10:23:31 +00:00
sdk
Make base classes for native video encoder/decoder public
2019-11-21 17:04:50 +00:00
stats
Use explicit template decl/def for RTCNonStandardStatsMember.
2019-11-13 11:12:40 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Remove dependencies on Chromium's webrtc_overrides.
2019-11-21 09:42:07 +00:00
test
Add TimeController to api/test/ and add a CreateTimeController API.
2019-11-22 17:07:23 +00:00
tools_webrtc
[UBSan] Add suppressions for vp8, opus, absl.
2019-11-21 13:40:47 +00:00
video
Stop using DEPRECATED_SingleThreadedTaskQueueForTesting in MultiStreamTester
2019-11-19 10:52:12 +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
Add .clangd to .gitignore
2019-10-28 12:27:50 +00:00
.gn
Switch to compiling WebRTC -std=c++14 by default
2019-09-09 19:24:16 +00:00
.vpython
Add source-side perf upload script for WebRTC.
2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md
Update style guide for absl::make_unique.
2019-09-18 06:10:58 +00:00
AUTHORS
Force alignment of generated JVM called functions.
2019-11-21 12:34:35 +00:00
BUILD.gn
Move stun.h to api/.
2019-11-18 16:11:27 +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 022da7ca60..5395db8bba (716991:718200)
2019-11-22 18:49:12 +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
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
PRESUBMIT.py
Add possibility to skip check_includes presubmit check.
2019-10-22 19:35:31 +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_lib_link_test.cc
Rewrite the lib link test to just be a binary.
2019-10-18 07:42:20 +00:00
webrtc.gni
Manual chromium roll: Compile using JDK 11
2019-11-20 12:13:43 +00:00
whitespace.txt
Revert "Whitespace change"
2019-11-11 14:58:20 +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

Description
No description provided
Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%