dec73a8164
The extracts and refactors some test code from rtp_sender_unittest.cc and puts it in a new target intended to only test RtpSenderEgress, and do it as pure unit test, rather than the unholy not-quite-unit-not-quite-integration-test thingy we have today. Only a first test case is actually ported with this CL, but it's a start... Bug: webrtc:11340 Change-Id: Ie2cdde63a00a6ff6eba7b8d443eeb76ce2a527c9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216180 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33841}
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