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chromium-webrtc-autoroll 9063ea7700 Roll chromium_revision 0526cc47b2..fbd658a96e (823879:824119) 
Change log: 0526cc47b2..fbd658a96e
Full diff: 0526cc47b2..fbd658a96e

Changed dependencies
* src/base: 4efa53828a..fbe91eb467
* src/build: ef5fe70ce0..21d4eee6d1
* src/buildtools: 95fa84e35d..b6526d9ad3
* src/buildtools/linux64: git_revision:f5f465b52e5df6d4de9fbe67cc79f1914ba4dbfc..git_revision:e002e68a48d1c82648eadde2f6aafa20d08c36f2
* src/buildtools/mac: git_revision:f5f465b52e5df6d4de9fbe67cc79f1914ba4dbfc..git_revision:e002e68a48d1c82648eadde2f6aafa20d08c36f2
* src/buildtools/win: git_revision:f5f465b52e5df6d4de9fbe67cc79f1914ba4dbfc..git_revision:e002e68a48d1c82648eadde2f6aafa20d08c36f2
* src/ios: ab1845d590..c742561187
* src/testing: 616adef0a3..e3fb440e81
* src/third_party: 0ae41f05ca..7cc258641c
* src/third_party/freetype/src: b049cadeb1..0bd50c9043
* src/third_party/icu: c7c91f829d..c2a4cae149
* src/third_party/libyuv: 19d71f6b35..f014dbd87a
* src/tools: ca70468ec0..d6572acaed
DEPS diff: 0526cc47b2..fbd658a96e/DEPS

Clang version changed llvmorg-12-init-5627-gf086e85e:llvmorg-12-init-10666-gc9f69ee7
Details: 0526cc47b2..fbd658a96e/tools/clang/scripts/update.py

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

Change-Id: If72553e1a1b5c2a7061bc830fdef03bde66c0409
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191740
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@{#32554}
2020-11-04 20:46:44 +00:00
api
Remove remaining variables related to incomplete frames.
2020-11-04 16:07:43 +00:00
audio
Enable continuous audio polling from ADM after StopPlay in VoIP API
2020-11-04 19:18:03 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
Reland "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
2020-11-02 11:05:56 +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
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
doc: mention video_replay tool for reporting video bugs
2020-10-30 11:22:01 +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
Free data if SCTP packet is delivered after transport destruction.
2020-10-29 23:04:14 +00:00
modules
Delete class RWLockWrapper, and its build target.
2020-11-04 14:53:13 +00:00
p2p
Check for oversized TURN usernames
2020-11-02 13:46:16 +00:00
pc
Move "options" from ConnectionContext to PeerConnectionFactory
2020-11-04 11:10:53 +00:00
resources
Increased high frequency transparency
2020-11-02 10:42:10 +00:00
rtc_base
Delete class RWLockWrapper, and its build target.
2020-11-04 14:53:13 +00:00
rtc_tools
Remove simulated neteq packet loss plot.
2020-11-03 17:25:50 +00:00
sdk
Remove remaining variables related to incomplete frames.
2020-11-04 16:07:43 +00:00
stats
stats: s/victim/other
2020-10-09 15:40:13 +00:00
style-guide
…
system_wrappers
Delete use of RWLockWrapper from SimulatedClock
2020-11-04 08:01:08 +00:00
test
Analyze quality of dropped frames in VideoProcessor.
2020-10-29 08:23:49 +00:00
tools_webrtc
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
video
Reland "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
2020-11-02 11:05:56 +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 "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 0526cc47b2..fbd658a96e (823879:824119)
2020-11-04 20:46:44 +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
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +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
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 bots.
2020-10-22 19:13:15 +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
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%