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chromium-webrtc-autoroll c4834a58d7 Roll chromium_revision 0a07e9993a..cbcae3bb9f (818869:818998) 
Change log: 0a07e9993a..cbcae3bb9f
Full diff: 0a07e9993a..cbcae3bb9f

Changed dependencies
* src/base: de96eb635d..5efb55bb55
* src/build: 2c9bdffb45..d709f77b70
* src/ios: 3b372067c9..586e9572e6
* src/testing: 72ce87626a..a06f7d1223
* src/third_party: df2e73da4e..5e895d424f
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/749daf9498..edbb060595
* src/third_party/perfetto: e6f794f1dc..be6747613d
* src/tools: 5a8ae6d50c..5c94bee7c1
DEPS diff: 0a07e9993a..cbcae3bb9f/DEPS

No update to Clang.

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

Change-Id: I915a3df71e900f6f3bcd55a663073ff865871a0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189900
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@{#32454}
2020-10-20 19:58:30 +00:00
api
Delete leftover mention of AsyncInvoker
2020-10-19 13:10:42 +00:00
audio
Revert "Introduce RTC_NO_UNIQUE_ADDRESS."
2020-10-07 07:37:01 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
Hookup VideoSendStreamImpl::OnVideoLayersAllocationUpdate to RtpVideoSender.
2020-10-19 11:37:23 +00:00
common_audio
Reland "Activating AVX2 support by default"
2020-10-03 22:37:13 +00:00
common_video
Reland "Add scaling interface to VideoFrameBuffer"
2020-10-09 08:30:50 +00:00
data
…
docs
Adding documentation to become a WebRTC committer.
2020-09-02 07:49:04 +00:00
examples
Switch AppRTCMobile #import headers.
2020-10-20 07:23:47 +00:00
logging
Add test for logging of large compound RTCP packets.
2020-10-19 21:52:38 +00:00
media
Allow to set av1 scalability mode after encoder is constructed
2020-10-19 10:42:23 +00:00
modules
In Vp9 wrapper allow to use ScalableVideoController with simulcast structures
2020-10-20 14:16:50 +00:00
p2p
Revert "Reland "Replace sigslot usages with robocaller library.""
2020-10-09 18:07:56 +00:00
pc
Delete the "initialize" method of PeerConnectionFactory
2020-10-20 16:52:39 +00:00
resources
Reland "Activating AVX2 support by default"
2020-10-03 22:37:13 +00:00
rtc_base
Revert "Reland "Replace sigslot usages with robocaller library.""
2020-10-09 18:07:56 +00:00
rtc_tools
Reland "Add scaling interface to VideoFrameBuffer"
2020-10-09 08:30:50 +00:00
sdk
Remove usage of "use_ios_simulator".
2020-10-20 10:48:08 +00:00
stats
stats: s/victim/other
2020-10-09 15:40:13 +00:00
style-guide
…
system_wrappers
Reland "Activating AVX2 support by default"
2020-10-03 22:37:13 +00:00
test
Allow to set av1 scalability mode after encoder is constructed
2020-10-19 10:42:23 +00:00
tools_webrtc
IWYU: Don't filter for already-included header files.
2020-10-19 10:12:03 +00:00
video
Hookup VideoSendStreamImpl::OnVideoLayersAllocationUpdate to RtpVideoSender.
2020-10-19 11:37:23 +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 '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 bug with the sps-pps-idr-in-keyframe fmtp parameter.
2020-09-19 09:23:39 +00:00
BUILD.gn
Rename CancerStickCastle to RoboCaller.
2020-09-21 19:50:39 +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 0a07e9993a..cbcae3bb9f (818869:818998)
2020-10-20 19:58:30 +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
Inclusive language in PRESUBMIT.py.
2020-07-22 10:01:23 +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
Fix PRESUBMIT.py warnings.
2020-09-07 08:34:44 +00:00
pylintrc
Update pylintrc to match Chromium style guide.
2020-10-07 10:55:06 +00:00
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
Fix link in documentation. (take 2)
2020-04-16 11:08:43 +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
Revert "Reland "Remove placeholder Obj-C headers and use angle-bracketed headers.""
2020-10-19 08:32:12 +00:00
whitespace.txt
Triggering all the bots.
2020-10-12 16:23:53 +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%
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Python 1.3%
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