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Danil Chapovalov d0321c5e5a Deduplicate set of the rtp header extension uri constants 
Bug: webrtc:7472
Change-Id: Ic0b4f2cc3374ba70a043310b5046d8bf91f0acbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231949
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34990}
2021-09-14 13:38:44 +00:00
api Deduplicate set of the rtp header extension uri constants 2021-09-14 13:38:44 +00:00
audio Deduplicate set of the rtp header extension uri constants 2021-09-14 13:38:44 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call Deduplicate set of the rtp header extension uri constants 2021-09-14 13:38:44 +00:00
common_audio Use backticks not vertical bars to denote variables in comments 2021-08-10 10:40:03 +00:00
common_video SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated. 2021-08-30 10:20:55 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Update Mac prerequisites 2021-08-23 19:52:17 +00:00
examples Replace AV1X with AV1 2021-09-14 08:29:02 +00:00
g3doc Use absl instead of self-made function for low-level bit counting 2021-08-26 08:56:37 +00:00
logging Migrate rtc event log from rtc::BitBuffer to BitstreamReader 2021-09-07 14:03:27 +00:00
media Fix potential crash during SimulcastEncoderAdapter tear down. 2021-09-14 09:15:22 +00:00
modules Deduplicate set of the rtp header extension uri constants 2021-09-14 13:38:44 +00:00
net/dcsctp dcsctp: Use integer math in RTO calculations 2021-09-08 22:06:04 +00:00
p2p Delete BasicPacketSocketFactory default constructor 2021-09-03 10:46:29 +00:00
pc Replace AV1X with AV1 2021-09-14 08:29:02 +00:00
resources Disable high-pass filtering of the AEC reference 2021-02-23 07:06:11 +00:00
rtc_base QualityRampupExperiment: SetMaxBitrate may not be set correctly. 2021-09-11 10:28:43 +00:00
rtc_tools Improve points visualization in metrics_plotter. 2021-09-10 10:59:58 +00:00
sdk Replace AV1X with AV1 2021-09-14 08:29:02 +00:00
stats Reland "Wire up non-sender RTT for audio, and implement related standardized stats." 2021-09-06 14:26:55 +00:00
system_wrappers Use GTEST_SKIP() instead of early return. 2021-08-12 15:24:13 +00:00
test Add danilchap@webrtc.org as owner of test/fuzzers/ 2021-09-14 08:48:03 +00:00
tools_webrtc Add TSAN suppression for usrsctp 2021-09-03 12:18:16 +00:00
video VideoSendStreamTest: Add tests for encoder reconfiguration. 2021-09-13 13:14:22 +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 Increase iOS deployment target from 10 to 12. 2021-07-02 17:02:27 +00:00
.vpython Update links to point at main branch 2021-07-22 16:41:26 +00:00
AUTHORS Fix potential crash during SimulcastEncoderAdapter tear down. 2021-09-14 09:15:22 +00:00
BUILD.gn Allow export of Obj-C symbols without C++ ones. 2021-07-30 22:54:59 +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 58d05cfd21..48501b3f18 (917624:917742) 2021-09-02 18:38:08 +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 kwiberg@webrtc.org from OWNERS files 2020-12-04 15:11:26 +00:00
g3doc.lua Improve webrtc documentation infra. Preview at: 2021-03-30 10:29:30 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt
native-api.md Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +00:00
OWNERS Remove inactive owners. 2021-08-31 14:27:49 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py fix some typos 2021-08-12 18:37:10 +00:00
presubmit_test.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +00:00
presubmit_test_mocks.py Add presubmit check to guard against assert() usage. 2021-07-22 17:08:26 +00:00
pylintrc Undo enforcing of PEP-8 pylint changes for method and function names. 2020-11-10 18:26:25 +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.gni Revert "Reland "PipeWire capturer: implement proper DMA-BUFs support""" 2021-09-03 11:28:26 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Trigger bots 2021-09-08 18:30:03 +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