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Johannes Kron 10aeb7403f Rename index.md to README.md to make it automatically show up 
Also add a heading to each file.

No-Try: True
Bug: webrtc:11335
Change-Id: I5e935741662558e72e417fa80a48c5ecda66c5f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169854
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30702}
2020-03-06 10:43:51 +00:00
api Adds field trial to separate audio and video packets for delay-based overuse detection. 2020-03-05 16:29:55 +00:00
audio Only update the current time of a played out frame if a new frame is played out. 2020-02-25 12:48:32 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Transform encoded frame in RTPSenderVideo. 2020-03-03 08:17:49 +00:00
common_audio Optimizations and refactoring of the APM 3-band split filter 2020-02-24 13:19:14 +00:00
common_video Remove wildcard ownership for build files. 2020-02-19 14:05:46 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Rename index.md to README.md to make it automatically show up 2020-03-06 10:43:51 +00:00
examples Remove wildcard ownership for build files. 2020-02-19 14:05:46 +00:00
logging Replace DataSize and DataRate factories with newer versions 2020-02-18 16:09:50 +00:00
media Implement new specification for degradation preference 2020-03-05 14:24:25 +00:00
modules Move EventWrapper class to target video_coding_legacy. 2020-03-06 08:39:35 +00:00
p2p Make Connection::id() const 2020-03-05 09:30:18 +00:00
pc Adds field trial to separate audio and video packets for delay-based overuse detection. 2020-03-05 16:29:55 +00:00
resources iSAC API wrapper unit test fix 2020-02-27 14:27:23 +00:00
rtc_base Update RTC_CHECK and RTC_LOG macros so they work when called from xxxxx::rtc namespaces 2020-03-04 22:53:34 +00:00
rtc_tools Add YUV to IVF video converter util 2020-02-19 14:44:21 +00:00
sdk Implement new specification for degradation preference 2020-03-05 14:24:25 +00:00
stats Remove wildcard ownership for build files. 2020-02-19 14:05:46 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Move EventWrapper class to target video_coding_legacy. 2020-03-06 08:39:35 +00:00
test Add printout of supported codecs in PC test framework 2020-03-05 18:05:26 +00:00
tools_webrtc Whitespace change to kick bots. 2020-02-28 06:44:59 +00:00
video Replace AdaptCount with a single counter. 2020-03-06 08:43:47 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore new format cleanup. 2019-07-11 16:18:51 +00:00
.gitignore Add .clangd to .gitignore 2019-10-28 12:27:50 +00:00
.gn Switch to compiling WebRTC -std=c++14 by default 2019-09-09 19:24:16 +00:00
.vpython Add source-side perf upload script for WebRTC. 2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md Fix typo in abseil-in-webrtc.md. 2019-12-18 14:27:34 +00:00
AUTHORS Reland "Reland "Reland "Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5.""""" 2020-02-26 20:35:54 +00:00
BUILD.gn Move video_replay under rtc_tools/. 2020-02-07 17:57:30 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +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
common_types.h Reland "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery" 2020-02-07 08:23:58 +00:00
DEPS Roll chromium_revision 70eb5f7c71..4dc8a31053 (747482:747587) 2020-03-06 06:33:23 +00:00
ENG_REVIEW_OWNERS Enforce LGTM from owners of depends-on paths in DEPS via presubmit. 2018-09-28 12:49:54 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt
native-api.md Delete unused I420 "codec" 2018-12-18 12:30:58 +00:00
OWNERS Add #COMPONENT to WebRTC. 2019-10-08 12:20:39 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Fix public_deps presubmit and gn format fighting each other. 2020-01-30 11:22:46 +00:00
presubmit_test.py Use source_sets in component builds and static_library in release builds. 2019-10-17 21:17:18 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
pylintrc Fixing py lint errors 2018-07-23 15:28:48 +00:00
README.chromium Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
README.md Tell users where they can find the native API headers 2017-11-14 10:36:46 +00:00
style-guide.md Add guidance to style guide how to reference a bug in a TODO 2019-12-11 11:55:52 +00:00
WATCHLISTS Remove benwright@webrtc.org from WATCHLISTS 2020-01-31 18:46:52 +00:00
webrtc.gni Reformat GN files. 2020-01-21 12:13:11 +00:00
webrtc_lib_link_test.cc Rewrite the lib link test to just be a binary. 2019-10-18 07:42:20 +00:00
whitespace.txt Whitespace change 2020-01-28 07:53:15 +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 http://www.webrtc.org/native-code/development 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