commit | a12b1d625cc992de704b9947c4d11957ebfbc0f8 | [log] [tgz] |
---|---|---|
author | Erik Språng <sprang@webrtc.org> | Wed Mar 14 12:39:24 2018 +0100 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Mar 15 09:54:56 2018 +0000 |
tree | 7bdc20b818395faa7b890aef16fb2cec6b700fe1 | |
parent | a11005ae3ffa63ef84a1e02cf98f91b0c572fc16 [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: I162cb9a1eccddf567bdda7285f8296dc2f005503 Reviewed-on: https://webrtc-review.googlesource.com/60900 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Original-Commit-Position: refs/heads/master@{#22356} Reviewed-on: https://webrtc-review.googlesource.com/61661 Cr-Commit-Position: refs/heads/master@{#22438}
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.