| commit | 1acfbd22cc30bc72af53187f25f3c389792968d3 | [log] [tgz] |
|---|---|---|
| author | hbos <hbos@webrtc.org> | Thu Nov 17 23:43:29 2016 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Fri Nov 18 07:43:39 2016 +0000 |
| tree | e30b7965d7f1fb12cc5b2209313b7cfecafdf50e | |
| parent | 7b9feeeaad18a436d19f6c5172c5ad8468362814 [diff] |
Expose RtpCodecParameters to VoiceMediaInfo stats. Payload type -> RtpCodecParameters maps added for sender and receiver. This is a follow-up to https://codereview.webrtc.org/2484193002/ which did the same thing for VideoMediaInfo. This information will be used to produce RTCCodecStats[1]. Voice[Sender/Receiver]Info is updated with current codec payload type for every stream which can be used to look up the codec in VoiceMediaInfo. [1] https://w3c.github.io/webrtc-stats/#codec-dict* BUG=chromium:659117 Review-Url: https://codereview.webrtc.org/2503383002 Cr-Commit-Position: refs/heads/master@{#15144}
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.