943ad970f4
The RTCReceivedRtpStreamStats hierarchy, which inherit from RTCRtpStreamStats, already contain members ssrc, kind, codec_id and transport_id so there's no need to list them inside RTCRemoteInboundrtpStreamStats. This CL removes duplicates so that we don't DCHECK-crash on Android, and adds a unit test ensuring we never accidentally list the same member twice. Bug: webrtc:12658 Change-Id: I27925eadddc6224bf6d6a91784ed7cafd7a4cfb3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214343 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33649}
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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
- Reporting bugs