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chromium-webrtc-autoroll 39c172d2a5 Roll chromium_revision 98d2d3586b..7ffc87db5f (829346:829478) 
Change log: 98d2d3586b..7ffc87db5f
Full diff: 98d2d3586b..7ffc87db5f

Changed dependencies
* src/base: 74286ae676..c5ee3d3a66
* src/build: 13721f7533..ee1c0017e6
* src/ios: 95cc5f773d..13ac7e8c0c
* src/testing: 6c6bbeca5d..43906c65e3
* src/third_party: 29c5de97fb..30ba6c99db
* src/third_party/android_deps/libs/androidx_appcompat_appcompat_resources: version:1.2.0-beta01-cr0..version:1.2.0-cr0
* src/third_party/android_deps/libs/androidx_media_media: version:1.0.0-cr0..version:1.2.0-cr0
* src/third_party/android_deps/libs/androidx_mediarouter_mediarouter: version:1.0.0-cr0..version:1.3.0-SNAPSHOT-cr0
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/0c019226a9..2ed6fc040f
* src/third_party/depot_tools: 5589652282..2f8e0fa49d
* src/tools: 0bda20101d..b0b70752e4
DEPS diff: 98d2d3586b..7ffc87db5f/DEPS

Clang version changed llvmorg-12-init-11780-g6ef07111:llvmorg-12-init-12083-g94e4ec64
Details: 98d2d3586b..7ffc87db5f/tools/clang/scripts/update.py

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

Change-Id: Ic40f6f24056c49b6c8574043ceedbdbc60537963
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194220
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@{#32646}
2020-11-20 03:09:24 +00:00
api
Make video scalability mode configurable from peerconnection level.
2020-11-18 12:06:03 +00:00
audio
Reland "Rename FATAL() into RTC_FATAL()."
2020-11-18 20:49:08 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
Disable flaky test.
2020-11-18 18:22:58 +00:00
common_audio
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
2020-11-09 10:47:55 +00:00
common_video
Reland "Rename FATAL() into RTC_FATAL()."
2020-11-18 20:49:08 +00:00
data
…
docs
doc: port release notes
2020-11-12 19:18:59 +00:00
examples
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
logging
Fix incorrect ToUnsigned in RTC event log.
2020-11-02 16:50:36 +00:00
media
Remove dependency on AsyncInvoker in SctpTransport
2020-11-19 23:25:42 +00:00
modules
Removed superfluous check in FrameBuffer2.
2020-11-19 17:06:14 +00:00
p2p
Check for oversized TURN usernames
2020-11-02 13:46:16 +00:00
pc
Remove dependency on AsyncInvoker in SctpTransport
2020-11-19 23:25:42 +00:00
resources
Increased high frequency transparency
2020-11-02 10:42:10 +00:00
rtc_base
Reland "Rename FATAL() into RTC_FATAL()."
2020-11-18 20:49:08 +00:00
rtc_tools
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
2020-11-09 10:47:55 +00:00
sdk
Delete objc RTCRtpFragmentationHeader
2020-11-19 14:05:48 +00:00
stats
stats: add transportId to codec stats
2020-11-17 12:34:39 +00:00
style-guide
…
system_wrappers
Delete use of RWLockWrapper from SimulatedClock
2020-11-04 08:01:08 +00:00
test
Reland "Rename FATAL() into RTC_FATAL()."
2020-11-18 20:49:08 +00:00
tools_webrtc
Reduce test flakiness because of upload issues
2020-11-18 22:41:35 +00:00
video
Make video scalability mode configurable from peerconnection level.
2020-11-18 12:06:03 +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
Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
2020-09-07 08:37:14 +00:00
.vpython
Add protobuf-py2_py3 3.13.0 to .vpython.
2020-11-19 16:00:58 +00:00
abseil-in-webrtc.md
Polish the "Using Abseil in WebRTC" docs
2020-10-16 13:42:00 +00:00
AUTHORS
Fix "control reaches end of non-void function" warnings
2020-10-27 10:22:23 +00:00
BUILD.gn
Rename RoboCaller to CallbackList.
2020-10-23 15:14:22 +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
DEPS
Roll chromium_revision 98d2d3586b..7ffc87db5f (829346:829478)
2020-11-20 03:09:24 +00:00
ENG_REVIEW_OWNERS
…
LICENSE
…
license_template.txt
…
native-api.md
Make transient suppression optionally excludable via defines
2020-04-02 11:44:07 +00:00
OWNERS
Remove phoglund as root owner.
2020-03-30 12:15:56 +00:00
PATENTS
…
presubmit_test_mocks.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
presubmit_test.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
PRESUBMIT.py
Add transition define RTC_OBJC_HAVE_LEGACY_RTC_RTP_FRAGMENTATION_HEADER
2020-11-17 13:51:29 +00:00
pylintrc
Undo enforcing of PEP-8 pylint changes for method and function names.
2020-11-10 18:26:25 +00:00
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
doc: move bug reporting instructions to the repository
2020-10-21 14:47:49 +00:00
style-guide.md
C++ style: We don't allow designated initializers
2020-06-03 09:11:09 +00:00
WATCHLISTS
Remove benwright@webrtc.org from WATCHLISTS
2020-01-31 18:46:52 +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
Reland "Remove placeholder Obj-C headers and use angle-bracketed headers."
2020-10-22 11:29:48 +00:00
whitespace.txt
Trigger tests.
2020-11-16 19:06:26 +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 here 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
Languages
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Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%