commit | 7bb37b884b197ea22e2830b043c09018c186bad5 | [log] [tgz] |
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author | Erik Språng <sprang@webrtc.org> | Fri Mar 09 08:52:59 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Mar 09 11:42:34 2018 |
tree | 47e078e6310764ed6fea3d25f0294b2dc80b0880 | |
parent | 881f16891bb8ebca588b0dfda9ee177896ad1eef [diff] |
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}
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.