| commit | 9af2b6012a782c113c1f5793ca13d5b08210d2f0 | [log] [tgz] |
|---|---|---|
| author | denicija <denicija@webrtc.org> | Thu Nov 17 08:43:43 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Nov 17 08:44:09 2016 |
| tree | 368cb457a1d515a759e2d5dae342780ed3a457c8 | |
| parent | a62f5826d7c318b9ff2fe4b408758ac968b24b07 [diff] |
Propagate bitrate setting to RTCRtpSender. This CL wires everything up and enables actual setting of the max bitrate encoding parameter on the video RTP sender. The following changes were made * Add maxbitrate property to the settings model and settings store. Make sure to store and read the maxbitrate from storage (to persist between app launches and make testing easier) * Fix setup of encoding parameters for the rtp sender as previous timing was not right. * Fix header of RTCRtpSender to expose needed parameter BUG=webrtc:6654 Review-Url: https://codereview.webrtc.org/2492693003 Cr-Commit-Position: refs/heads/master@{#15120}
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.