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chromium-webrtc-autoroll 1a7a31588d Roll chromium_revision 20a633fc5a..3a3815b9ca (827865:827982) 
Change log: 20a633fc5a..3a3815b9ca
Full diff: 20a633fc5a..3a3815b9ca

Changed dependencies
* src/base: 0cdde2d003..7f50568af5
* src/build: cb25cdf8bb..089f0350ca
* src/ios: 9edef1cedf..f5a3fd5fc0
* src/testing: 207cb1c178..a16a050138
* src/third_party: 478e1c8fc9..d00c966f59
* src/third_party/depot_tools: fd5c198347..da73999f79
* src/tools: a1813a7956..3a9fc33e28
DEPS diff: 20a633fc5a..3a3815b9ca/DEPS

No update to Clang.

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

Change-Id: I6c9d3e14ca7e858247aa20c442fc8c69e5fa6a7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193781
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@{#32616}
2020-11-17 00:52:30 +00:00
api
VoipVolumeControl subAPI for VoIP API
2020-11-13 19:27:12 +00:00
audio
VoipVolumeControl subAPI for VoIP API
2020-11-13 19:27:12 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
In Av1 packetizer set marker bit with respect of end_of_picture flag
2020-11-16 11:08:48 +00:00
common_audio
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
2020-11-09 10:47:55 +00:00
common_video
Reland "Add scaling interface to VideoFrameBuffer"
2020-10-09 08:30:50 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
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
Make SEA to be codec agnostic.
2020-11-11 18:34:35 +00:00
modules
In Av1 packetizer set marker bit with respect of end_of_picture flag
2020-11-16 11:08:48 +00:00
p2p
Check for oversized TURN usernames
2020-11-02 13:46:16 +00:00
pc
[getStats] Reduce blocking thread-invokes from 2 to 1.
2020-11-13 09:36:18 +00:00
resources
Increased high frequency transparency
2020-11-02 10:42:10 +00:00
rtc_base
CallbackList: Don't allow reentrancy
2020-11-11 12:26:14 +00:00
rtc_tools
Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED()
2020-11-09 10:47:55 +00:00
sdk
in Av1 encoder wrapper communicate end_of_picture flag similar to VP9
2020-11-11 14:00:52 +00:00
stats
Fix -Wrange-loop-analysis.
2020-11-05 15:55:33 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Delete use of RWLockWrapper from SimulatedClock
2020-11-04 08:01:08 +00:00
test
In Av1 packetizer set marker bit with respect of end_of_picture flag
2020-11-16 11:08:48 +00:00
tools_webrtc
Introduce length checking of all STUN byte string attributes
2020-11-13 12:31:37 +00:00
video
Ensure EncoderSink::OnBitrateAllocationUpdated not called after Stop
2020-11-13 16:26:41 +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
Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
2020-09-07 08:37:14 +00:00
.vpython
Add 'requests' to .vpython.
2020-09-09 14:36:03 +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
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
DEPS
Roll chromium_revision 20a633fc5a..3a3815b9ca (827865:827982)
2020-11-17 00:52:30 +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
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
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
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
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +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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C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%