da9e284308
This is a pure rename+move of PeerConnectionSimulcastMediaFlowTests. The reason for renaming is to reflect that a) this is an integration test, not a unit test, and b) not all of the tests use simulcast (some use a single encoding, i.e. singlecast or SVC). Shared helper functions between PeerConnectionEncodingsIntegrationTest and PeerConnectionSimulcastTests are placed in a test-only util file. # Already pass, no need to wait for chromium bots for webrtc testonly CL NOTRY=True Bug: webrtc:15063 Change-Id: Iec90d1a7ab712be1395c7644723422c8c6179974 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300540 Reviewed-by: Jeremy Leconte <jleconte@google.com> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39799}
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
- Documentation