bd09a46aa1
Moves OnSendSideDelayUpdated and OnSendPacketUpdated out from rtp_sender_unittest and into rtp_sender_egress_unittest and rtp_rtcp_impl2_unittest. The former test now only tests the logic for updating send-side-delay stats. The latter is now on a proper RtpRtcp-level and also verifies that frame timestamps makes it to the egress (as assumed by the first test). Bug: webrtc:11340 Change-Id: I784042ad91eb66a4d1eebdbbc625f9522528bfb5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218502 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33996}
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
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs