| commit | a48ddb763695dd6742fdcb6f0e5fa4f9818595e7 | [log] [tgz] |
|---|---|---|
| author | Per <perkj@chromium.org> | Thu Sep 29 09:48:50 2016 |
| committer | Per <perkj@chromium.org> | Thu Sep 29 09:49:01 2016 |
| tree | 65fe633f31af4dec5b1fac2660e96fb700dc1b5c | |
| parent | fd0d42669204e6dd92a60736bca7ae0196663024 [diff] |
Add VideoSendStream::Stats::prefered_media_bitrate_bps This cl move calculation of stats for prefered_media_bitrate_bps from webrtcvideoengine2.GetStats to SendStatisticsProxy::OnEncoderReconfigured. This aligns better with how other send stats are reported and is needed as a prerequisite for moving video encoder configuration due to video resolution change from WebRtcVideoEngine2 to ViEEncoder. BUG=webrtc:6371 R=mflodman@webrtc.org, sprang@webrtc.org Review URL: https://codereview.webrtc.org/2368223002 . Cr-Commit-Position: refs/heads/master@{#14431}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.