| commit | dc1c62cd30e42d5bc31db347126476ca8e1b6c58 | [log] [tgz] |
|---|---|---|
| author | skvlad <skvlad@webrtc.org> | Thu Mar 17 02:07:43 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Mar 17 02:07:49 2016 |
| tree | 2b52872047633fa01a266312fcc096fd7195ee3b | |
| parent | df6416aa502d5a9694875a22fcdf286c10f836ea [diff] |
Enable setting the maximum bitrate limit in RtpSender. This change allows the application to limit the bitrate of the outgoing audio and video streams at runtime. The API roughly follows the WebRTC API draft, defining the RTCRtpParameters structure witn exactly one encoding (simulcast streams are not exposed in the API for now). (https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters) BUG= Review URL: https://codereview.webrtc.org/1788583004 Cr-Commit-Position: refs/heads/master@{#12025}
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.