ce4aef16ee
This is a re-land of https://codereview.webrtc.org/1353263005/ which was reverted because of perf-regressions. Changes since that CL: * Change LayerFilteringTransport to send a padding packet instead of dropping it for data that should be filtered out. This prevents confusion due to changed sequence numbers. * Changed timing of stats poller thread in VideoAnalyzer. Startup was racy wrt initializion of send_stream_. * Minor formatting issues. PERF NOTE: This change will affect some performance numbers slightly. In particular, {encode_frame_rate, encode_time_ms, encode_usage_percent, media_bitrate_bps} will change due to timing of the measurements. BUG= R=pbos@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1412233003 Cr-Commit-Position: refs/heads/master@{#10483}
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