| commit | a65704b5c905c97970f28eb3d1df7369a700efdf | [log] [tgz] |
|---|---|---|
| author | hbos <hbos@webrtc.org> | Mon Nov 14 02:28:16 2016 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Nov 14 10:28:20 2016 +0000 |
| tree | 6fb4aa2bc14968183498f86f959085517c3931ba | |
| parent | 82ebe02491d066697717ae386f886b752729e013 [diff] |
Expose RtpCodecParameters to VideoMediaInfo stats. Payload type -> RtpCodecParameters maps added for sender and receiver side. It contains information that will be needed for RTCCodecStats[1] dictionaries. Video[Sender/Receiver]Info is updated with current codec payload type for every stream which can be used to look up the codec in VideoMediaInfo. A similar change should be made for VoiceMediaInfo and Voice[Sender/Receiver]Info. [1] https://w3c.github.io/webrtc-stats/#codec-dict* BUG=chromium:659117 Review-Url: https://codereview.webrtc.org/2484193002 Cr-Commit-Position: refs/heads/master@{#15060}
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.