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chromium-webrtc-autoroll 0680c4871c Roll chromium_revision 22cbef27a6..a3e750c9b0 (946167:946315) 
Change log: 22cbef27a6..a3e750c9b0
Full diff: 22cbef27a6..a3e750c9b0

Changed dependencies
* src/base: f8009dc41e..e6e0ee5bae
* src/build: 2f9e5b5867..e0d5dbffaa
* src/ios: d11bb31289..af6e3bb63e
* src/testing: 42caab826d..69383604c0
* src/third_party: 17c7be145e..92767b0feb
* src/third_party/androidx: iVGn06-lNVsnXSXyo3Xe7rE4VVi_XIqRPKaXxKiUqp8C..XSHrYFNYVHl0vna-Qi6iVbg-_eXSr31_Z04UJ-zVc1YC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/75423c310e..8cf0922154
* src/third_party/depot_tools: 51d99b2f3a..804165b445
* src/tools: 25c7e11129..0e7a87dbf5
DEPS diff: 22cbef27a6..a3e750c9b0/DEPS

No update to Clang.

BUG=None

Change-Id: I3a7d136d51ddf23cb7ed7cc856272684fc152daa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239425
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@{#35438}
2021-11-30 04:30:11 +00:00
api
Always call IsOk() to ensure audio codec configuration is valid when negotiating.
2021-11-26 10:11:21 +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
Remove unnecessary copy of suspended_ssrcs.
2021-11-29 10:54:30 +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
Roll chromium_revision 1ce7d592c3..fe555f2e69 (944520:945923)
2021-11-29 16:23:30 +00:00
net/dcsctp
dcsctp: Use rtc::CopyOnWriteBuffer
2021-11-29 11:53:19 +00:00
p2p
Fix TestTurnServer to not depend on rtc::Thread::socketserver() accessor
2021-11-29 21:10:31 +00:00
pc
Fix TestTurnServer to not depend on rtc::Thread::socketserver() accessor
2021-11-29 21:10:31 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Deprecate and remove usage for WARNING log level
2021-11-27 22:21:54 +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
Avoid TBR in DEPS autoroller (use CC instead).
2021-11-29 12:54:01 +00:00
video
VideoSendStreamTest: Fix flaky UAF.
2021-11-29 15:58:40 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
…
.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 22cbef27a6..a3e750c9b0 (946167:946315)
2021-11-30 04:30:11 +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
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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