signalapp/webrtc
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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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Karl Wiberg 02f4e32b08 Make some new rtc_base targets publicly visible 
Bug: webrtc:9987
Change-Id: I207514c8790d2f3a043ed083790261b1c4b7ba33
Reviewed-on: https://webrtc-review.googlesource.com/c/122084
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26627}
2019-02-11 08:58:58 +00:00
api Fix PeerConnectionInterface::StartRtcEventLog documentation. 2019-02-11 08:00:39 +00:00
audio Propagate base minimum delay to audio_receiver_stream 2019-02-06 11:07:42 +00:00
build_overrides Disable hermetic toolchain when building on macOS 10.14. 2018-12-07 13:57:28 +00:00
call H.264 temporal layers w/frame marking (PART 2/3) 2019-02-09 16:47:09 +00:00
common_audio Stop using Googletest legacy APIs. 2019-01-31 13:23:33 +00:00
common_video Always add/rewrite VUI and set max_num_reorder_frames to 0. 2019-02-04 09:18:46 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
examples Remove unused file. 2019-02-06 10:24:07 +00:00
logging Stop hard-coding default IDs for RTP extensions 2019-02-09 01:04:35 +00:00
media H.264 temporal layers w/frame marking (PART 2/3) 2019-02-09 16:47:09 +00:00
modules H.264 temporal layers w/frame marking (PART 2/3) 2019-02-09 16:47:09 +00:00
p2p Remove TCPPort incoming_only_ member. 2019-02-04 19:46:30 +00:00
pc Stop hard-coding default IDs for RTP extensions 2019-02-09 01:04:35 +00:00
resources Reland of Opus multistream. 2019-01-29 12:16:19 +00:00
rtc_base Make some new rtc_base targets publicly visible 2019-02-11 08:58:58 +00:00
rtc_tools Do not build rtp_generator in Chromium builds. 2019-02-04 14:48:58 +00:00
sdk Delete unused media constraints 2019-02-08 14:45:00 +00:00
stats Expose video freeze metrics in GetStats. 2019-02-04 09:58:08 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Makes Clock interface fully mutable. 2019-01-31 08:12:05 +00:00
test Calculate next process time in simulated network. 2019-02-08 19:33:17 +00:00
tools_webrtc Resume rolling buildtools, now as chromium/src/buildtools 2019-02-08 10:36:39 +00:00
video H.264 temporal layers w/frame marking (PART 2/3) 2019-02-09 16:47:09 +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 format commit. 2018-06-20 09:26:44 +00:00
.gitignore Add Visual Studio Code project folder to gitignore file. 2019-01-21 18:42:33 +00:00
.gn Re-enable gtest absl pretty printers. 2018-08-13 13:54:05 +00:00
.vpython Add vpython dependencies needed to run presubmit tests on LUCI 2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md Allow use of functions in absl/algorithms 2019-01-28 18:42:16 +00:00
AUTHORS Add ',' between elements in RTCStatsReport::ToJson 2019-01-08 16:05:23 +00:00
BUILD.gn Set WEBRTC_USE_H264 in common_config 2019-02-08 10:17:04 +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 Delete constant RTP_PAYLOAD_NAME_SIZE 2019-01-25 07:59:52 +00:00
DEPS Roll chromium_revision eb2aa6ea6a..339f6a582b (630596:630696) 2019-02-11 08:30:32 +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 Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
native-api.md Delete unused I420 "codec" 2018-12-18 12:30:58 +00:00
OWNERS Clean up root OWNERS. 2018-11-09 14:23:59 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
presubmit_test.py Fixing py lint errors 2018-07-23 15:28:48 +00:00
PRESUBMIT.py Add PRESUBMIT to enforce usage of new Googletest APIs. 2019-02-06 11:06:02 +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 Remove rule that discourages passing optional by const reference 2019-02-05 11:58:05 +00:00
WATCHLISTS Modify pc/ WATCHLISTS definition 2019-01-31 22:09:16 +00:00
webrtc.gni Override default manifest from Chromium in WebRTC. 2019-01-24 13:00:49 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +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