| commit | f7b1b95f11c74cb5369fdd528b73c70a50f2e206 | [log] [tgz] |
|---|---|---|
| author | Alessio Bazzica <alessiob@webrtc.org> | Tue Mar 23 16:23:04 2021 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Mar 23 18:44:12 2021 |
| tree | 902e44e90786224a2b227c492582e10659056e15 | |
| parent | 26abdaf478e6663a69443a5a271a0ec623b941d1 [diff] |
Add `RTCRemoteOutboundRtpStreamStats` for audio streams Changes: - adding the `RTCRemoteOutboundRtpStreamStats` dictionary (see [1]) - collection of remote outbound stats (only for audio streams) - adding `remote_id` to the inbound stats and set with the ID of the corresponding remote outbound stats only if the latter are available - unit tests [1] https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats Tested: verified from chrome://webrtc-internals during an appr.tc call Bug: webrtc:12529 Change-Id: Ide91dc04a3c387ba439618a9c6b64a95994a1940 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211042 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33545}
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.
See here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.