| commit | cb44343006c1fcf282ae3ce55640ba297be2ad74 | [log] [tgz] |
|---|---|---|
| author | deadbeef <deadbeef@webrtc.org> | Mon Dec 12 19:12:36 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Dec 12 19:12:42 2016 |
| tree | 9a73f775ab736103a1ab0960a2bd1507f34508d5 | |
| parent | ccecdd456097f3dd82d6bdcb116692ebb6891731 [diff] |
Add SSRC to RtpEncodingParameters for audio. Was added for video initially, but not for audio. BUG=webrtc:6862 Review-Url: https://codereview.webrtc.org/2568553002 Cr-Commit-Position: refs/heads/master@{#15552}
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.