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chromium-webrtc-autoroll edd2786a79 Roll chromium_revision 598b93304b..93c32ed01f (1002121:1002259) 
Change log: 598b93304b..93c32ed01f
Full diff: 598b93304b..93c32ed01f

Changed dependencies
* src/base: 1fb3d1d730..a58e181148
* src/build: 1c34f36484..8c1118a6ba
* src/buildtools/third_party/libc++abi/trunk: ecff200716..bb1e46cb72
* src/ios: 78de2e463b..ff224ae111
* src/testing: 1295d94c88..663fe5ea6e
* src/third_party: 655473b8ac..13f944fa3f
* src/third_party/androidx: 9EARqDo1HNj2IE6F5pbKWOUSN2acBpBu04AW3SP7qAEC..2V7ZDLIKJDlBnGSSJXQDu9ZmBIHGqJzUGaNLc9A8nkoC
* src/third_party/perfetto: 1de2530cbb..4786303513
* src/tools: e3a84ba077..06c7fa81ae
DEPS diff: 598b93304b..93c32ed01f/DEPS

No update to Clang.

BUG=None

Change-Id: I4e76abb55cd4627e20bb143bcd74780b7c55e5cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262141
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@{#36857}
2022-05-11 20:25:32 +00:00
api
stats: implement transport iceState
2022-05-10 13:55:21 +00:00
audio
Add GetRtpExtensionMap to ReceiveStream and remove GetRtpExtensions.
2022-05-10 13:50:31 +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 (2022-05-11T04:01:49).
2022-05-11 06:30:13 +00:00
common_audio
Delete rtc_base/format_macros.h
2022-05-09 12:03:21 +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
Disable ARDAppClientTest testSession on simulator.
2022-04-28 07:39:23 +00:00
g3doc
Remove mention of MSVC support from docs
2022-05-02 09:26:32 +00:00
infra
Add MacArm64 builder and perf config
2022-05-10 08:52:00 +00:00
logging
Adopt absl::string_view in CopyOnWriteBuffer
2022-05-10 10:32:41 +00:00
media
Add GetRtpExtensionMap to ReceiveStream and remove GetRtpExtensions.
2022-05-10 13:50:31 +00:00
modules
Switch SSE2 architecture check to x86 platforms only
2022-05-11 18:12:54 +00:00
net/dcsctp
dcsctp: Fix typo for handing I-FORWARD-TSN.
2022-05-11 10:26:06 +00:00
p2p
stats: implement transport iceState
2022-05-10 13:55:21 +00:00
pc
Safeguard SctpDataChannel against detached controller
2022-05-11 10:54:13 +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 rtc::join
2022-05-11 14:37:04 +00:00
rtc_tools
Improved RobustThroughputEstimator
2022-05-11 08:45:53 +00:00
sdk
Delete rtc_base/format_macros.h
2022-05-09 12:03:21 +00:00
stats
stats: implement transport iceState
2022-05-10 13:55:21 +00:00
system_wrappers
Fixes to support building in -std=c++20 mode.
2022-05-05 17:15:58 +00:00
test
Delete rtc_base/format_macros.h
2022-05-09 12:03:21 +00:00
tools_webrtc
Add specific mac arm64 mb config
2022-05-10 13:08:31 +00:00
video
Add GetRtpExtensionMap to ReceiveStream and remove GetRtpExtensions.
2022-05-10 13:50:31 +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
Switch SSE2 architecture check to x86 platforms only
2022-05-11 18:12:54 +00:00
BUILD.gn
Create a "slow peerconnection unittests" target
2022-05-05 14:40:38 +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 598b93304b..93c32ed01f (1002121:1002259)
2022-05-11 20:25:32 +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
Remove usrsctp, dcSCTP is now the unique SCTP implementation
2022-04-04 10:30:46 +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
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C 2.2%
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Python 1.3%
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