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chromium-webrtc-autoroll aa8f2771b4 Roll chromium_revision 2fe18c5940..ae499b5a6c (1050141:1051662) 
Change log: 2fe18c5940..ae499b5a6c
Full diff: 2fe18c5940..ae499b5a6c

Changed dependencies
* src/base: 0efc92e419..29eb97497d
* src/build: 7fcdda4d2b..237be0ab6c
* src/buildtools: b79692f320..7fca26a6d1
* src/buildtools/third_party/libc++/trunk: a4dc7f17ca..c23f69f322
* src/ios: 17d8594fb9..c798350a34
* src/testing: 9df4b2c558..cddecff250
* src/third_party: 3deedbc463..73e87bc5c2
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/5b90d5e756..5eb0675d9c
* src/third_party/depot_tools: 08bb5c487f..9ef048cd50
* src/third_party/freetype/src: 33ceac2afc..c9c326312f
* src/third_party/libaom/source/libaom: https://aomedia.googlesource.com/aom.git/+log/fd35001068..6ed0c7a325
* src/third_party/libunwindstack: 3c86843ae0..e184bd94bf
* src/third_party/perfetto: 768b3f2265..e78894f91b
* src/third_party/r8/d8: Qn31g4m2cofkyvGgm46Uzkzds5DKdNYrdPePwRkVnv4C..xSIcikL1PDal0LfVuHAK-q1NOalVeNp3x9KaFZ8Yju4C
* src/tools: 35efe9a8f5..1b7f9d69b7
DEPS diff: 2fe18c5940..ae499b5a6c/DEPS

Clang version changed llvmorg-16-init-4949-g8ace9ea258c9:llvmorg-16-init-5189-gbfcd536a
Details: 2fe18c5940..ae499b5a6c/tools/clang/scripts/update.py

BUG=None

Change-Id: I105a30b32f796acc60fb931c4e142357dd76d2ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276827
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@{#38218}
2022-09-27 09:03:21 +00:00
api
Migrate CallSimulator to the new perf metrics logging API
2022-09-26 19:37:51 +00:00
audio
Cleanup: Deduplicate static functions that create network links
2022-09-26 16:45:30 +00:00
build_overrides
Add stub for build_overrides/partition_alloc.gni
2022-08-29 12:17:02 +00:00
call
Make DegradedCall work without DCHECK failure
2022-09-27 08:02:01 +00:00
common_audio
Fix math involving enums in C++20
2022-09-27 06:55:31 +00:00
common_video
Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable
2022-07-20 08:15:08 +00:00
data
…
docs
doc: backfill M90-M105 release notes
2022-08-29 21:04:32 +00:00
examples
Replace Thread::Invoke with Thread::BlockingCall
2022-09-09 10:44:17 +00:00
g3doc
Clarify how to reference WebRTC bugs in TODOs
2022-07-01 08:03:34 +00:00
infra
Add reclient properties to perf builders
2022-09-09 09:04:35 +00:00
logging
Encode remote link capacity estimates in legacy RTC event log format
2022-08-11 12:57:02 +00:00
media
Reland "RtpEncodingParameters::request_resolution patch 1"
2022-09-23 11:48:19 +00:00
modules
Migrate VideoCodecTestFixture on new perf metrics logging API
2022-09-27 08:31:12 +00:00
net/dcsctp
Fix several UBsan issues with memcpy
2022-09-14 09:35:39 +00:00
p2p
Add test coverage for P2PTransportChannel refactor (#8/8)
2022-09-21 16:49:02 +00:00
pc
Make DegradedCall work without DCHECK failure
2022-09-27 08:02:01 +00:00
resources
Clarify and extend test support for certain sample rates in audio processing
2022-08-03 14:26:36 +00:00
rtc_base
Make ExpectationToString generate detailed logs in more cases.
2022-09-26 16:24:04 +00:00
rtc_tools
Migrate video_quality_analysis on new perf metrics logger API
2022-09-26 10:44:00 +00:00
sdk
Renamed methods.
2022-09-15 14:08:22 +00:00
stats
[Stats] Avoid DCHECK crashing if SSRCs are not unique.
2022-09-26 10:28:01 +00:00
system_wrappers
Delete nisse@webrtc.org from OWNERS files
2022-07-28 08:47:38 +00:00
test
PCLF: Ignore discarded frames in the DefaultVideoQualityAnalyzer
2022-09-26 13:42:01 +00:00
tools_webrtc
Adapt generate_sslroots.py to work with different openssl versions
2022-09-22 15:15:20 +00:00
video
Cleanup: Deduplicate static functions that create network links
2022-09-26 16:45:30 +00:00
.clang-format
…
.git-blame-ignore-revs
…
.gitignore
…
.gn
Set Fuchsia Api level + update SDK version
2022-09-14 08:49:56 +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
Add python grpc to .vpython3 for ios test runner
2022-09-16 12:26:48 +00:00
AUTHORS
loongarch: Add loongarch architecture support.
2022-09-13 10:44:59 +00:00
BUILD.gn
Introduce MetricsExporter API with stdout implementation
2022-09-17 00:56:29 +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 2fe18c5940..ae499b5a6c (1050141:1051662)
2022-09-27 09:03:21 +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
…
README.md
doc: add g3doc sitemap to toplevel readme
2021-07-23 07:55:17 +00:00
WATCHLISTS
Remove fhernqvist from watchlists
2022-08-11 14:44:52 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
Add Fuchsia platform deps
2022-09-23 07:49:47 +00:00
whitespace.txt
Trigger CI.
2022-08-12 11:03:03 +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%