| commit | 4b4dc86c611e8db9b75de40d2b6ddd5f215e95ab | [log] [tgz] |
|---|---|---|
| author | nisse <nisse@webrtc.org> | Wed Feb 17 13:25:36 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Feb 17 13:25:40 2016 |
| tree | e5f2708b127f1a9014da8a8c068a92d50479cafd | |
| parent | 22785c709956365ac51bc3b79135e6debc6418ae [diff] |
Remove conference_mode flag from AudioOptions and VideoOptions. For audio, the flag is apparently unused. For video, the flag is moved to VideoSendParameters, with the intention to keep only per-stream flags in VideoOptions. The flag is used for the webrtcvideoengine2 logic commented like // Conference mode screencast uses 2 temporal layers split at 100kbit. // For screenshare in conference mode, tl0 and tl1 bitrates are piggybacked // on the VideoCodec struct as target and max bitrates, respectively. // See eg. webrtc::VP8EncoderImpl::SetRates(). BUG=webrtc:5426 Review URL: https://codereview.webrtc.org/1697163002 Cr-Commit-Position: refs/heads/master@{#11651}
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.