gecko-dev/third_party/libwebrtc
stransky 6d59f634f7 Bug 1693849 [PipeWire] Always check allocated video buffer size, r=ng
Differential Revision: https://phabricator.services.mozilla.com/D105802
2021-02-20 17:12:56 +00:00
..
build/internal
chromium_deps
google_apis/build
net
third_party/gflags
tools/clang
webrtc Bug 1693849 [PipeWire] Always check allocated video buffer size, r=ng 2021-02-20 17:12:56 +00:00
AUTHORS
DEPS
LICENSE
Makefile.old
OWNERS
README.md
dummy_file.txt
peerconnection.Makefile
peerconnection_client.target.mk

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. This page is maintained by the Google Chrome team.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

More info