| commit | 56e611bbdab19400f454520b6a6a1bfd3da1e989 | [log] [tgz] |
|---|---|---|
| author | Erik Språng <sprang@webrtc.org> | Thu Feb 06 17:10:08 2020 +0100 |
| committer | Commit Bot <commit-bot@chromium.org> | Fri Feb 07 08:23:58 2020 +0000 |
| tree | 9577d69742b26ddb8036cacc5050786690fa3173 | |
| parent | ecd6fc84cf070865eaa131a35b77ef9cdff4cbfc [diff] |
Reland "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery" This is a reland of 4f68f5398d7fa3d47c449e99893c9bea07bf7ca2 Original change's description: > Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery > > The PlayoutDelayOracle was responsible for making sure the PlayoutDelay > header extension was successfully propagated to the receiving side. Once > it was determined that the receiver had received a frame with the new > delay tag, it's no longer necessary to propagate. > > The issue with this implementation is that it is based on max > extended sequence number reported via RTCP, which makes it often slow > to react, could theoretically fail to produce desired outcome (max > received > X does not guarantee X was fully received and decoded), and > added a lot of code complexity. > > The guarantee of delivery can in fact be accomplished more reliably and > with less code by making sure to tag each frame until an undiscardable > frame is sent. > > This allows containing the logic fully within RTPSenderVideo. > > Bug: webrtc:11340 > Change-Id: I2d1d2b6b67f4f07b8b33336f8fcfcde724243eef > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168221 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30473} TBR=stefan@webrtc.org Bug: webrtc:11340 Change-Id: I2fdd0004121b13b96497b21e052359e31d0c477a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168305 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30479}
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.