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landrey ca1cfd45de Add missing src/third_party/jdk11 dependency and roll chromium into webrtc 
Roll chromium_revision e5191e93ab..1f33300013 (1096792:1097480)

Change log: e5191e93ab..1f33300013
Full diff: e5191e93ab..1f33300013

Changed dependencies
* src/base: 451505f0f1..211595b9e1
* src/build: 1a2bf3bafa..8e59fa7328
* src/buildtools: 0cc02fb849..3c7e3f1b8b
* src/buildtools/third_party/libc++/trunk: 1dfd0022c4..1127c78cf9
* src/ios: 5c6f9398b9..c44e3b09fa
* src/testing: 3bec4d3b1c..89218c737f
* src/third_party: c16ccdd4ad..97f83ef911
* src/third_party/androidx: 3nXDIlFQ-oAo4egDVeC0GbueTl4mOfPrEDzT5jgDEK8C..8KUWsjmvRQJlzdaro14SgP8nok3-kHY2h00BEjXLJqQC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/45986b07fa..abaad53f0c
* src/third_party/depot_tools: c710e2d56a..44e9bee34f
* src/third_party/jdk: egbcSHbmF1XZQbKxp_PQiGLFWlQK65krTGqQE-Bj4j8C..-FR8HTNcMfxy7J2HUaWVa0QmEE4f68iotzvFbqOj2LEC
* src/third_party/perfetto: ee03e391a1..7706133c5b
* src/third_party/r8: SCuPfdKTpEJfVbWzckpmWx125Zf3pO-4RitqcfXX-RQC..lhnuNLpWpWBVM6efX0iIg5i9ztfW8VKpMvkyfWCxfr0C
* src/tools: 7c53822d54..8a2826d35f
* src/tools/luci-go: git_revision:36cf4b54527c49aca7ed3023995a081e14f1852a..git_revision:221383f749a2c5b8587449d3d2e4982857daa9e7
* src/tools/luci-go: git_revision:36cf4b54527c49aca7ed3023995a081e14f1852a..git_revision:221383f749a2c5b8587449d3d2e4982857daa9e7
DEPS diff: e5191e93ab..1f33300013/DEPS

No update to Clang.

BUG=b/266775646

Change-Id: I2597b77b4c3d29d88674825aa7e241f2be0a8820
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291436
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39210}
2023-01-27 08:52:48 +00:00
api
Revert "Migrate WebRTC documentation to new renderer"
2023-01-26 20:19:12 +00:00
audio
Reset encoder when audio send stream is stopped.
2023-01-26 15:20:02 +00:00
build_overrides
Use default values provided by PartitionAlloc instead of hard-coded ones
2022-12-07 09:11:35 +00:00
call
Ensure CallTest derived tests per default set min/max audio bitrate.
2023-01-26 17:36:01 +00:00
common_audio
Make header files self contained.
2022-10-08 08:38:36 +00:00
common_video
Add 444 10 bits support for H264 and VP9
2023-01-17 12:32:26 +00:00
data
…
docs
Update WebRTC doc related to webrtc.org accounts.
2023-01-16 09:34:28 +00:00
examples
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
experiments
Add tool for generating field trial registry header
2022-10-18 07:25:43 +00:00
g3doc
Revert "Migrate WebRTC documentation to new renderer"
2023-01-26 20:19:12 +00:00
infra
Fix gtest-output and resultdb for fuchsia
2023-01-25 14:27:38 +00:00
logging
Revert "Migrate WebRTC documentation to new renderer"
2023-01-26 20:19:12 +00:00
media
Reland "Delete PacketReceiver::DeliverPacket from all implementations"
2023-01-25 18:18:29 +00:00
modules
Revert "Migrate WebRTC documentation to new renderer"
2023-01-26 20:19:12 +00:00
net/dcsctp
Remove all usage of //rtc_base target
2023-01-16 14:36:06 +00:00
p2p
Revert "Migrate WebRTC documentation to new renderer"
2023-01-26 20:19:12 +00:00
pc
Revert "Migrate WebRTC documentation to new renderer"
2023-01-26 20:19:12 +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
Remove the //rtc_base target
2023-01-23 14:00:21 +00:00
rtc_tools
Ensure VideoRtpReplayer use new PacketReceiver::DeliverRtp packet.
2023-01-18 12:47:41 +00:00
sdk
Disable RTCCameraVideoCapturerTestsWithMockedCaptureSession.
2023-01-18 09:45:06 +00:00
stats
Revert "Migrate WebRTC documentation to new renderer"
2023-01-26 20:19:12 +00:00
system_wrappers
[Unwrap] Migrate RtpToNtpEstimator to use RtpTimestampUnwrapper
2023-01-11 17:14:41 +00:00
test
Revert "Migrate WebRTC documentation to new renderer"
2023-01-26 20:19:12 +00:00
tools_webrtc
[infra] Remove CQ shadow builders with reclient
2023-01-24 06:59:13 +00:00
video
Revert "Migrate WebRTC documentation to new renderer"
2023-01-26 20:19:12 +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
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
Remove dimension check in SimulcastUtility::ValidSimulcastParameters
2023-01-11 13:41:55 +00:00
BUILD.gn
Delete unused Audio Bwe integration test.
2023-01-26 09:31:44 +00:00
CODE_OF_CONDUCT.md
Revert "Migrate WebRTC documentation to new renderer"
2023-01-26 20:19:12 +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
Add missing src/third_party/jdk11 dependency and roll chromium into webrtc
2023-01-27 08:52:48 +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
Revert "Migrate WebRTC documentation to new renderer"
2023-01-26 20:19:12 +00:00
LICENSE
…
license_template.txt
…
native-api.md
Revert "Migrate WebRTC documentation to new renderer"
2023-01-26 20:19:12 +00:00
OWNERS
Add infra owners file
2022-12-02 09:21:47 +00:00
OWNERS_INFRA
Add infra owners file
2022-12-02 09:21:47 +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
Remove xooglers from WATCHLISTS and OWNERS
2022-11-30 15:33:25 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
Verify field trials looked up through field_trial::FindFullName
2022-10-20 10:46:01 +00:00
whitespace.txt
Trigger bots
2022-11-17 21:29:53 +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%