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chromium-webrtc-autoroll 43fc10fc40 Roll chromium_revision b50c75ae41..fbbdfd477b (947439:947567) 
Change log: b50c75ae41..fbbdfd477b
Full diff: b50c75ae41..fbbdfd477b

Changed dependencies
* src/base: 0175fce238..a82eea6df3
* src/build: 7d4410e011..2c47f6b719
* src/ios: 4e83e0173c..e89fc3f392
* src/third_party: 1a819afbae..842ebd1e14
* src/third_party/androidx: kOGHSX2ia0dsRAm8DcdBlff2C07mTWt9YaieS6a5K74C..8psIPDigEQ4L6gV80_7FCGGuyGy_ExaUlGBytDFeHX8C
* src/third_party/perfetto: 4a9b683ccb..aeff6922a9
* src/tools: f79e695e27..903f362c7c
DEPS diff: b50c75ae41..fbbdfd477b/DEPS

No update to Clang.

BUG=None

Change-Id: Ib70757a33654c5034c28703a3ca490e923715fd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239700
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@{#35464}
2021-12-02 18:40:47 +00:00
api
Remove NetEq::Create.
2021-11-30 10:31:16 +00:00
audio
New struct PeerNetworkDependencies
2021-11-23 08:37:36 +00:00
build_overrides
Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions
2021-05-27 16:16:01 +00:00
call
Revert "Call: Deduplicate SentPacket notifications"
2021-11-30 11:49:51 +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
…
docs
Add contributing.md
2021-11-03 14:59:46 +00:00
examples
Deprecate and remove usage for WARNING log level
2021-11-27 22:21:54 +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
Delete legacy RtpPacketSendInfo::ssrc field
2021-12-02 11:59:37 +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
sdp: temporarily raise mid limit to 32
2021-11-30 21:05:51 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Fix rate statistic when time window running out of samples
2021-11-30 23:57:40 +00:00
rtc_tools
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
2021-11-15 21:44:59 +00:00
sdk
Deprecate and remove usage for WARNING log level
2021-11-27 22:21:54 +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
Deprecate and remove usage for WARNING log level
2021-11-27 22:21:54 +00:00
tools_webrtc
Revert "Use gtest_parallel with 1 worker for webrtc_perf_tests."
2021-12-02 13:16:00 +00:00
video
Fix potential use after move in RtpVideoStreamReceiver
2021-11-30 11:20:21 +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
.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
…
DEPS
Roll chromium_revision b50c75ae41..fbbdfd477b (947439:947567)
2021-12-02 18:40:47 +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
…
license_template.txt
…
native-api.md
Reference "main" branches instead of "master" branches.
2021-07-15 11:07:44 +00:00
OWNERS
Add .vpython3 file to webrtc
2021-09-15 16:56:30 +00:00
PATENTS
…
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
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: 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
No description provided
Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%