| commit | 883eefc59e96da2dcb23a36f178610a92587cd73 | [log] [tgz] |
|---|---|---|
| author | Henrik Boström <hbos@webrtc.org> | Mon May 27 11:40:25 2019 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon May 27 12:45:22 2019 |
| tree | ddc6f6b179878a5ec2fc5efc4566df64c651eff1 | |
| parent | 6e436d1cc0d54ed39bc28d4dac2e9439de9a5b65 [diff] |
Implement RTCRemoteInboundRtpStreamStats for both audio and video. This implements the essentials of RTCRemoteInboundRtpStreamStats. This includes: - ssrc - transportId - codecId - packetsLost - jitter - localId - roundTripTime https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict* The following members are not implemented because they require more work... - From RTCReceivedRtpStreamStats: packetsReceived, packetsDiscarded, packetsRepaired, burstPacketsLost, burstPacketsDiscarded, burstLossCount, burstDiscardCount, burstLossRate, burstDiscardRate, gapLossRate and gapDiscardRate. - From RTCRemoteInboundRtpStreamStats: fractionLost. Bug: webrtc:10455, webrtc:10456 Change-Id: If2ab0da7105d8c93bba58e14aa93bd22ffe57f1d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138067 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28073}
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 http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.