commit | 632a03c0cd712be0d89f968d668d43c527e1e836 | [log] [tgz] |
---|---|---|
author | Erik Språng <sprang@webrtc.org> | Thu Feb 06 16:04:48 2020 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Feb 06 16:05:02 2020 |
tree | 631390a1e5ea5d6366528a4972c5e59d3515fcf4 | |
parent | 67dba30178b16a1727f8727df084c82eabb06013 [diff] |
Revert "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery" This reverts commit 4f68f5398d7fa3d47c449e99893c9bea07bf7ca2. Reason for revert: Breaks downstream project 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=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org Change-Id: Ide922e680ae36bb69b95e58002482cf5ed57e254 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11340 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168304 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30475}
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.