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chromium-webrtc-autoroll 6b0bedb05a Roll chromium_revision b24fc2ea76..5dee08f539 (1012289:1012417) 
Change log: b24fc2ea76..5dee08f539
Full diff: b24fc2ea76..5dee08f539

Changed dependencies
* src/base: 901d29a224..e3df4ab156
* src/build: a789781b39..33d06bd2d5
* src/buildtools/linux64: git_revision:fd6cae41bd7d5d255dc2fb96004a8bf74ac9d972..git_revision:2ecd43a10266bd091c98e6dcde507c64f6a0dad3
* src/buildtools/mac: git_revision:fd6cae41bd7d5d255dc2fb96004a8bf74ac9d972..git_revision:2ecd43a10266bd091c98e6dcde507c64f6a0dad3
* src/buildtools/third_party/libc++abi/trunk: 11395e54b4..013bcd820a
* src/buildtools/win: git_revision:fd6cae41bd7d5d255dc2fb96004a8bf74ac9d972..git_revision:2ecd43a10266bd091c98e6dcde507c64f6a0dad3
* src/ios: 1ac4226228..32b0e6a40f
* src/testing: 4d0a162419..c2041fe223
* src/third_party: c0cb113105..35b5a7576f
* src/third_party/androidx: JPzPwtVT1B0wxFtrYWysxKnlZrx9hVhFuT00NvADQ64C..h_P2inqiGpL43nlN62apySX-dW_9fOY90d7Iuh8ayj8C
* src/third_party/lss: https://chromium.googlesource.com/linux-syscall-support.git/+log/92a65a8f5d..1d387f43f3
* src/tools: 389744913f..49e63d1324
DEPS diff: b24fc2ea76..5dee08f539/DEPS

No update to Clang.

BUG=None

Change-Id: Ide4dc0789532a4b78a259854039e43615165cd7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265540
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#37161}
2022-06-09 10:28:43 +00:00
api
Delete variant of rtc::split that copies the output fields
2022-06-09 08:29:33 +00:00
audio
Migrate some scripts to python3
2022-06-01 10:09:36 +00:00
build_overrides
Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions
2021-05-27 16:16:01 +00:00
call
For VP9 assume max number of spatial layers to simulate generic descriptor
2022-06-08 11:36:54 +00:00
common_audio
Adopt absl::string_view in common_audio/
2022-05-13 15:00:14 +00:00
common_video
Remove dependency on rtc_base_approved from most targets
2022-04-25 12:15:30 +00:00
data
…
docs
Add contributing.md
2021-11-03 14:59:46 +00:00
examples
Update android example gradle and target sdk version.
2022-05-31 07:14:24 +00:00
g3doc
Remove mention of MSVC support from docs
2022-05-02 09:26:32 +00:00
infra
Rename 'iOS64 Sim Debug (iOS 14.0)' bot.
2022-06-03 07:17:27 +00:00
logging
In RtcEventLogParser use strict time types to access rtcp::TransportFeedback
2022-06-08 11:52:25 +00:00
media
Delete variant of rtc::split that copies the output fields
2022-06-09 08:29:33 +00:00
modules
Delete variant of rtc::split that copies the output fields
2022-06-09 08:29:33 +00:00
net/dcsctp
dcsctp: Add virtual time stream scheduler
2022-06-08 23:33:50 +00:00
p2p
Delete variant of rtc::split that copies the output fields
2022-06-09 08:29:33 +00:00
pc
Delete variant of rtc::split that copies the output fields
2022-06-09 08:29:33 +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
Delete variant of rtc::split that copies the output fields
2022-06-09 08:29:33 +00:00
rtc_tools
Disable tests depending on libyuv upscale filtering
2022-06-02 13:24:46 +00:00
sdk
Make the libaom AV1 encoder optional in RTCDefaultVideoEncoderFactory.
2022-06-01 10:54:55 +00:00
stats
stats: expose minPlayoutDelay as nonstandard stat
2022-05-31 10:05:35 +00:00
system_wrappers
Delete variant of rtc::split that copies the output fields
2022-06-09 08:29:33 +00:00
test
Reland "Don't create PacketSocketFactory inside BasicPortAllocatorSession"
2022-06-07 11:24:16 +00:00
tools_webrtc
Rename 'iOS64 Sim Debug (iOS 14.0)' bot.
2022-06-03 07:17:27 +00:00
video
RtpSenderInterface::SetEncoderSelector
2022-06-08 11:06:52 +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
Add support for stand-alone Fuchsia build.
2022-05-30 09:24:43 +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 b24fc2ea76..5dee08f539 (1012289:1012417)
2022-06-09 10:28:43 +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
Fork test/ios/google_test_runner.mm from Chromium's //base
2022-05-04 11:26:07 +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
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%