5cf8c2c501
`RTCInboundRtpStreamStats.lastPacketReceivedTimestamp` must be a time value in milliseconds with Unix epoch as time origin (see bugs.webrtc.org/12605#c4). This change fixes both audio and video `RTCInboundRtpStreamStats` stats. Tested: verified from chrome://webrtc-internals during an appr.tc call Bug: webrtc:12605 Change-Id: I68157fcf01a5933f3d4e5d3918b4a9d3fbd64f16 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212865 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33547}
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