commit | 5a98049f6abeb985172628938f034bf08b4220d6 | [log] [tgz] |
---|---|---|
author | Oleh Prypin <oprypin@webrtc.org> | Fri Mar 09 12:27:24 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Mar 09 12:28:39 2018 |
tree | 6105d4291fbe5bd0e8c70a563ca815dc972db124 | |
parent | 9486b117daac09c9f7ac8450ccda835938cf3150 [diff] |
Revert "Reland "Rework rtp packet history"" This reverts commit 7bb37b884b197ea22e2830b043c09018c186bad5. Reason for revert: Breaks downstream projects Original change's description: > Reland "Rework rtp packet history" > > This is a reland of 6328d7cbbc8a72fdc81a766c0bf4039e1e2e7887 > > Original change's description: > > Rework rtp packet history > > > > This CL rewrites the history from the ground up, but keeps the logic > > (mostly) intact. It does however lay the groundwork for adding a new > > mode where TransportFeedback messages can be used to remove packets > > from the history as we know the remote end has received them. > > > > This should both reduce memory usage and make the payload based padding > > a little more likely to be useful. > > > > My tests show a reduction of ca 500-800kB reduction in memory usage per > > rtp module. So with simulcast and/or fec this will increase. Lossy > > links and long RTT will use more memory. > > > > I've also slightly update the interface to make usage with/without > > pacer less unintuitive, and avoid making a copy of the entire RTP > > packet just to find the ssrc and sequence number to put into the pacer. > > > > The more aggressive culling is not enabled by default. I will > > wire that up in a follow-up CL, as there's some interface refactoring > > required. > > > > Bug: webrtc:8975 > > Change-Id: I0c1bb528f32eeed0fb276b4ae77ae3235656980f > > Reviewed-on: https://webrtc-review.googlesource.com/59441 > > Commit-Queue: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22347} > > Bug: webrtc:8975 > Change-Id: Ibbdbcc3c13bd58d994ad66f789a95ef9bd9bc19b > Reviewed-on: https://webrtc-review.googlesource.com/60900 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22356} TBR=danilchap@webrtc.org,sprang@webrtc.org Change-Id: Id698f5dbba6f9f871f37501d056e2b8463ebae50 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8975 Reviewed-on: https://webrtc-review.googlesource.com/61020 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22358}
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.