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Artem Titov 0f2ce5cc1c Revert "Migrate WebRTC documentation to new renderer" 
This reverts commit 3eceaf4669.

Reason for revert: 

Original change's description:
> Migrate WebRTC documentation to new renderer
>
> Bug: b/258408932
> Change-Id: Ib96f39fe0c3912f9746bcc09d079097a145d6115
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290987
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39205}

Bug: b/258408932
Change-Id: I16cb4088bee3fc15c2bb88bd692c592b3a7db9fe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291560
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39209}
2023-01-26 20:19:12 +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 Roll chromium_revision a484be4b74..e5191e93ab (1096680:1096792) 2023-01-25 16:41:49 +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.py Update portaudio to the latest 2022-05-13 09:01:34 +00:00
presubmit_test.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
presubmit_test_mocks.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +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.gni Verify field trials looked up through field_trial::FindFullName 2022-10-20 10:46:01 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +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