ac287ee8b5
render_time time field (means capture time for sender side) is used by rtcp SenderReport to calculate offset since last frame and to estimate rtp timestamp for the time SenderReport should be send at. mapping between rtp timestamp and ntp time in SenderReport is used for stream synchronization. calculation of rtp_timestamp (using ntp_time of incoming video frame) for rtp packets is unchanged. BUG=webrtc:5433, webrtc:5504, webrtc:5505 Review URL: https://codereview.webrtc.org/1693443002 Cr-Commit-Position: refs/heads/master@{#11820}
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. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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