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chromium-webrtc-autoroll 954f7274ac Roll chromium_revision afd5c7343a..97fa68dda6 (950813:950914) 
Change log: afd5c7343a..97fa68dda6
Full diff: afd5c7343a..97fa68dda6

Changed dependencies
* src/build: c244c09e3a..47daaa3cef
* src/ios: d1494f30f6..3a66471639
* src/testing: 0639a45bc9..7a93fae3be
* src/third_party: bf51619340..873b8799ff
* src/third_party/androidx: 5hNh49V3iJuALiRHZPIr3VRcsp8mtCj-ATiBKF6Y4iIC..ZwN1FJVBOl4Uffy4fDLOpuNkz2AqPE9BKdRfZAw6IW0C
* src/tools: 73bae9dab8..76ee54275a
DEPS diff: afd5c7343a..97fa68dda6/DEPS

No update to Clang.

BUG=None

Change-Id: I46f692f614ab03c7ccad38a50912014d99e09568
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241023
Bot-Commit: 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/main@{#35524}
2021-12-13 04:13:24 +00:00
api
Declare Plan B DEPRECATED
2021-12-08 21:10:16 +00:00
audio
Fix mb.py presubmit issues.
2021-12-08 08:53:00 +00:00
build_overrides
Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions
2021-05-27 16:16:01 +00:00
call
Update WebRTC code version (2021-12-12T04:04:39).
2021-12-12 05:35:43 +00:00
common_audio
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
2021-11-15 21:44:59 +00:00
common_video
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
2021-11-15 21:44:59 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Add contributing.md
2021-11-03 14:59:46 +00:00
examples
Deprecate BasicNetworkManager default constructor
2021-12-09 12:10:20 +00:00
g3doc
Add static AsString functions for PeerConnectionInterface enums
2021-11-02 12:29:50 +00:00
logging
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
2021-11-15 21:44:59 +00:00
media
Allow extremely low resolution for simulcast path
2021-11-25 19:07:57 +00:00
modules
Optimize CascadedBiQuadFilter.
2021-12-10 14:02:43 +00:00
net/dcsctp
Revert "dcsctp: Use rtc::CopyOnWriteBuffer"
2021-12-02 12:33:46 +00:00
p2p
Fix TestTurnServer to not depend on rtc::Thread::socketserver() accessor
2021-11-29 21:10:31 +00:00
pc
Tolerate overlong MID in channel.cc
2021-12-09 18:13:30 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Deprecate BasicNetworkManager default constructor
2021-12-09 12:10:20 +00:00
rtc_tools
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
2021-11-15 21:44:59 +00:00
sdk
remove last traces of WebRTC-Audio-SendSideBwe
2021-12-09 15:14:14 +00:00
stats
Implement missing candidate pair packets/bytes sent/received stats.
2021-09-28 23:27:05 +00:00
system_wrappers
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
2021-11-15 21:44:59 +00:00
test
New FrameBuffer3.
2021-12-07 18:31:37 +00:00
tools_webrtc
Relax gn check test.
2021-12-10 09:33:01 +00:00
video
Disable VideoSendStreamTest.RelayToDirectRoute due to flakiness
2021-12-09 17:46:00 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +00:00
.gitignore
Add .cache to .gitignore.
2021-01-20 15:01:07 +00:00
.gn
Roll chromium_revision 867b87eb29..b56d8868f3 (933713:941610)
2021-11-15 17:34:08 +00:00
.style.yapf
Fix mb.py presubmit issues.
2021-12-08 08:53:00 +00:00
.vpython
Update links to point at main branch
2021-07-22 16:41:26 +00:00
.vpython3
DEPS: use python3
2021-11-29 07:27:52 +00:00
AUTHORS
Added main profile to supported H264 codecs
2021-11-10 10:56:13 +00:00
BUILD.gn
Reland "Add dav1d decoder to WebRTC."
2021-11-23 15:59:58 +00:00
CODE_OF_CONDUCT.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +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 afd5c7343a..97fa68dda6 (950813:950914)
2021-12-13 04:13:24 +00:00
DIR_METADATA
Move metadata in OWNERS files to DIR_METADATA files.
2021-02-08 19:09:33 +00:00
ENG_REVIEW_OWNERS
Remove phoglund from ENG_REVIEW_OWNERS
2021-10-08 08:29:42 +00:00
g3doc.lua
Improve webrtc documentation infra. Preview at:
2021-03-30 10:29:30 +00:00
LICENSE
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
license_template.txt
…
native-api.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
OWNERS
Fix mb.py presubmit issues.
2021-12-08 08:53:00 +00:00
PATENTS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py
Add presubmit check to guard against assert() usage.
2021-07-22 17:08:26 +00:00
presubmit_test.py
Add presubmit check to guard against assert() usage.
2021-07-22 17:08:26 +00:00
PRESUBMIT.py
fix some typos
2021-08-12 18:37:10 +00:00
pylintrc
Fix mb.py presubmit issues.
2021-12-08 08:53:00 +00:00
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
doc: add g3doc sitemap to toplevel readme
2021-07-23 07:55:17 +00:00
WATCHLISTS
Update WATCHLISTS
2021-08-23 13:37:55 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
Reland "Add dav1d decoder to WebRTC."
2021-11-23 15:59:58 +00:00
whitespace.txt
Test LKGR finder.
2021-09-24 20:09:34 +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
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