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chromium-webrtc-autoroll e72e9ef59e Roll chromium_revision 667142f9aa..d1cee5d400 (1017367:1017491) 
Change log: 667142f9aa..d1cee5d400
Full diff: 667142f9aa..d1cee5d400

Changed dependencies
* src/base: a024c83cb9..03ecc15b73
* src/build: c017063154..ae20ef26a5
* src/ios: 2ee40e01ec..5a04ed5904
* src/testing: 075169d9db..b8e361d4ee
* src/third_party: 1132092121..b8388687c8
* src/third_party/androidx: ffRu7ou3A-tWC0cp9INlqplJN090p9v3saKCut6KJZ8C..4P3G5yy2a9JRgJERG8h3ApAoreMO-OjDG4RMhEcnZDYC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d44f2956da..3236751e83
* src/third_party/depot_tools: 1d5d7803fa..d9a9c40738
* src/third_party/fuchsia-sdk/sdk: version:8.20220623.1.1..version:8.20220623.3.1
* src/third_party/perfetto: d3300e34ed..2d2d50afec
* src/tools: b119e7ab63..9c7a2fc349
DEPS diff: 667142f9aa..d1cee5d400/DEPS

No update to Clang.

BUG=None

Change-Id: Id54e88d1443032a47412374e3d5373ca8ee1f73d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266800
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@{#37319}
2022-06-24 04:50:09 +00:00
api
Delete ProcessThread creation from test TimeController as unused
2022-06-23 09:41:43 +00:00
audio
Move to_queued_task.h and pending_task_safety_flag.h into public API
2022-06-17 09:20:39 +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 (2022-06-23T04:02:47).
2022-06-23 05:42:42 +00:00
common_audio
Adopt absl::string_view in common_audio/
2022-05-13 15:00:14 +00:00
common_video
Add 420 and 422 10 bit h264 decoding.
2022-06-17 11:12:10 +00:00
data
…
docs
Add contributing.md
2021-11-03 14:59:46 +00:00
examples
Revert "Reland "Delete old Android ADM.""
2022-06-16 16:07:49 +00:00
g3doc
Add documentation for field trials
2022-06-23 08:40:41 +00:00
infra
Temporarily disable video_capture_tests on linux
2022-06-23 10:13:52 +00:00
logging
Remove the last build cycle in WebRTC
2022-06-22 10:44:51 +00:00
media
add 422 8 and 10 bit decoding support
2022-06-22 15:08:44 +00:00
modules
add 422 8 and 10 bit decoding support
2022-06-22 15:08:44 +00:00
net/dcsctp
Reland "dcsctp: Use stream scheduler in send queue"
2022-06-22 08:41:12 +00:00
p2p
Remove the last build cycle in WebRTC
2022-06-22 10:44:51 +00:00
pc
Delete some unused sigslot dependencies
2022-06-23 12:30:22 +00:00
resources
AEC3: Changing the default for the use_conservative_tail_frequency_response flag.
2021-12-21 17:35:26 +00:00
rtc_base
tls: add logging of expected/received hashes on failures
2022-06-19 22:33:10 +00:00
rtc_tools
Move rtc::make_ref_counted to api/
2022-06-15 09:47:38 +00:00
sdk
Remove the last build cycle in WebRTC
2022-06-22 10:44:51 +00:00
stats
Implement Outbound/InboundRtpStreamStats.mid.
2022-06-17 08:44:09 +00:00
system_wrappers
Delete variant of rtc::split that copies the output fields
2022-06-09 08:29:33 +00:00
test
Delete ProcessThread creation from test TimeController as unused
2022-06-23 09:41:43 +00:00
tools_webrtc
Use the condensed library name when fixing "foo:foo" dependencies
2022-06-23 12:57:39 +00:00
video
Remove WebRTC.Video.DecodeTimePerFrameInMs. histograms
2022-06-23 13:01:42 +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
Prevent jsoncpp from hiding deprecated declarations in WebRTC
2022-04-11 12:33:47 +00:00
.mailmap
Add .mailmap for git.
2022-02-20 14:22:13 +00:00
.style.yapf
Fix mb.py presubmit issues.
2021-12-08 08:53:00 +00:00
.vpython
Remove unused script webrtc_dashboard_upload.py
2022-03-21 12:54:42 +00:00
.vpython3
Remove unused script webrtc_dashboard_upload.py
2022-03-21 12:54:42 +00:00
AUTHORS
GCC: Avoid symbol clash in RenderBuffer
2022-05-24 10:47:56 +00:00
BUILD.gn
SVC: Add end to end tests for VP8 and VP9
2022-06-22 11:07:01 +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 667142f9aa..d1cee5d400 (1017367:1017491)
2022-06-24 04:50:09 +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
Fix add some eng prod owners to PRESUBMIT.py.
2022-03-18 13:19:07 +00:00
PATENTS
…
presubmit_test_mocks.py
tools_webrtc dir converted to py3 + top level PRESUBMIT script
2022-02-08 14:42:26 +00:00
presubmit_test.py
tools_webrtc dir converted to py3 + top level PRESUBMIT script
2022-02-08 14:42:26 +00:00
PRESUBMIT.py
Update portaudio to the latest
2022-05-13 09:01:34 +00:00
pylintrc
tools_webrtc dir converted to py3 + top level PRESUBMIT script
2022-02-08 14:42:26 +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
[Cast Convergence] Replace is_chromecast with new args
2022-06-16 00:50:08 +00:00
whitespace.txt
Trigger CI bots
2021-12-16 17:45:31 +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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