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chromium-webrtc-autoroll a92051a5f1 Roll chromium_revision 97ecd0e93c..1891773fa9 (1008934:1009039) 
Change log: 97ecd0e93c..1891773fa9
Full diff: 97ecd0e93c..1891773fa9

Changed dependencies
* src/base: 84bbfd82bf..f902b7a9a2
* src/buildtools: c12bf52fc4..8b16338d17
* src/buildtools/linux64: git_revision:c547ca1497e3ff0dcbc0b2cb036b3d40380cbeeb..git_revision:37baefb026b199605affa7bcb24810d1724ce373
* src/buildtools/mac: git_revision:c547ca1497e3ff0dcbc0b2cb036b3d40380cbeeb..git_revision:37baefb026b199605affa7bcb24810d1724ce373
* src/buildtools/win: git_revision:c547ca1497e3ff0dcbc0b2cb036b3d40380cbeeb..git_revision:37baefb026b199605affa7bcb24810d1724ce373
* src/ios: 1782819473..051b5c47a8
* src/testing: 917341f8be..61885dc88b
* src/third_party: 262b2b3c55..c940de3868
* src/third_party/androidx: rub6aXBz5r6WR9pBd1H-bRnRkT8jX123XYM5pQG_xhwC..aaLNF-LbuC3COrA7CxH-p6Kk9DXknyL3CJbRKKLARSoC
* src/third_party/r8: ovozeRSDDfERnEFpDo_WS6OYOcEF7oT1JzGxCSf-g0kC..L2ZGhyJ-Hc8AnXHo5dhtNdI2F2bC_9yi2lqTbJ98sLcC
* src/tools: aa7c757af3..3bb820e5ed
DEPS diff: 97ecd0e93c..1891773fa9/DEPS

No update to Clang.

BUG=None

Change-Id: I89f06e839308c20b593eba51f8a6db828e80be96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264481
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@{#37058}
2022-05-31 11:05:14 +00:00
api
WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 18/inf
2022-05-31 10:50:58 +00:00
audio
Add SetTransportCc to ReceiveStreamInterface.
2022-05-30 14:07:04 +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 (2022-05-31T04:04:22).
2022-05-31 05:05:24 +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
Remove old data files.
2018-10-05 14:40:21 +00:00
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
Use ios_perf_test for M1 arm64 perf tester
2022-05-24 14:59:12 +00:00
logging
Improve RTC event log parser.
2022-05-18 09:14:40 +00:00
media
Add SetTransportCc to ReceiveStreamInterface.
2022-05-30 14:07:04 +00:00
modules
Fix compile error with -Werror
2022-05-31 09:59:34 +00:00
net/dcsctp
dcsctp: Add priority to dcsctp handover state
2022-05-30 15:52:16 +00:00
p2p
Revert "Don't create PacketSocketFactory inside BasicPortAllocatorSession"
2022-05-31 09:22:24 +00:00
pc
stats: expose minPlayoutDelay as nonstandard stat
2022-05-31 10:05:35 +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
Rewrite "Reset all maps in AndroidNetworkMonitor Start()/Stop()"
2022-05-31 10:49:55 +00:00
rtc_tools
event log analyzer: log rtp timestamps in question
2022-05-30 14:46:14 +00:00
sdk
Rewrite "Reset all maps in AndroidNetworkMonitor Start()/Stop()"
2022-05-31 10:49:55 +00:00
stats
stats: expose minPlayoutDelay as nonstandard stat
2022-05-31 10:05:35 +00:00
system_wrappers
Adopt absl::string_view in system_wrappers/
2022-05-17 08:23:24 +00:00
test
Revert "Don't create PacketSocketFactory inside BasicPortAllocatorSession"
2022-05-31 09:22:24 +00:00
tools_webrtc
Remove crosscompile for Perf Mac M1 Arm64 12
2022-05-23 09:54:48 +00:00
video
Add AV1 to WebRTC.Video.Encoder.CodecType histogram.
2022-05-30 15:43:25 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +00:00
.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
Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit
2018-10-25 08:19:53 +00:00
DEPS
Roll chromium_revision 97ecd0e93c..1891773fa9 (1008934:1009039)
2022-05-31 11:05:14 +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
No description provided
Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%