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chromium-webrtc-autoroll eec1d04aa9 Roll chromium_revision 8b5b046f02..022da7ca60 (715044:716991) 
Change log: 8b5b046f02..022da7ca60
Full diff: 8b5b046f02..022da7ca60

Changed dependencies
* src/base: 20e7dd7480..29c24a8b46
* src/build: ed0d663913..c60d0d9d66
* src/buildtools: 140e4d7c45..6b3e658d6f
* src/buildtools/third_party/libc++/trunk: 5938e0582b..78d6a7767e
* src/ios: 899a8a4bc9..01eee3d70d
* src/testing: d4d918486a..a6c25af3b7
* src/third_party: aad8135d1e..24a1541bd5
* src/third_party/bazel: tQPvsIj1Gtw5iXssKy7OREE-S02u7zItrw42l3DHUroC..VjMsf48QUWw8n7XtJP2AuSjIGmbQeYdWdwyxVvIRLmAC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/d709b0d892..6ba98ff601
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/3992f65642..55394ddc90
* src/third_party/depot_tools: 9577daf667..639872c8ef
* src/third_party/libyuv: 53b529e362..6afd9becdf
* src/third_party/lss: https://chromium.googlesource.com/linux-syscall-support.git/+log/8048ece6c1..726d71ec08
* src/tools: 224c37eeae..f2ed5a7b38
DEPS diff: 8b5b046f02..022da7ca60/DEPS

No update to Clang.

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

Change-Id: I44e824d8cfee4c3f4536be4442b93ed30fefe56b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160060
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@{#29847}
2019-11-20 14:42:26 +00:00
api
RTC_EXPORT RTCStatsMember's specialized members.
2019-11-20 13:53:37 +00:00
audio
Add missing assert.h for win no-test build
2019-11-17 07:52:32 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Using unit types in TransportFeedbackAdapter.
2019-11-06 12:25:00 +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
Add scale and compare methods to VideoFrame::UpdateRect
2019-11-19 15:42:42 +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
Change VideoAdapter::OnResolutionFramerateRequest to VideoAdapter::OnSinkWants
2019-11-19 10:14:31 +00:00
modules
Makes RoundRobinPacketQueue use same field trials as PacingController.
2019-11-20 13:36:46 +00:00
p2p
Move stun.h to api/.
2019-11-18 16:11:27 +00:00
pc
Make SCTPtransport enter "closed" state when DTLStransport does.
2019-11-16 14:56:01 +00:00
resources
Reland "Refactoring of the noise suppressor and adding true multichannel support"
2019-10-31 11:56:01 +00:00
rtc_base
In rtc::ByteBuffer drop support for ORDER_HOST as unused
2019-11-15 11:48:42 +00:00
rtc_tools
EventLogVisualizer pacer delay plot fix.
2019-11-19 10:23:31 +00:00
sdk
Add @Nullable annotations to quiet errorprone.
2019-11-19 12:50:30 +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
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
test
Change VideoAdapter::OnResolutionFramerateRequest to VideoAdapter::OnSinkWants
2019-11-19 10:14:31 +00:00
tools_webrtc
Manual chromium roll: Compile using JDK 11
2019-11-20 12:13:43 +00:00
video
Stop using DEPRECATED_SingleThreadedTaskQueueForTesting in MultiStreamTester
2019-11-19 10:52:12 +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
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
Revert "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5.""
2019-10-30 18:14:52 +00:00
BUILD.gn
Move stun.h to api/.
2019-11-18 16:11:27 +00:00
CODE_OF_CONDUCT.md
…
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 8b5b046f02..022da7ca60 (715044:716991)
2019-11-20 14:42:26 +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
…
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
…
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
…
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
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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