commit | f349e53ca54eefbb252b198cfd5f1a06fef43a40 | [log] [tgz] |
---|---|---|
author | Jianhui Dai <jianhui.j.dai@intel.com> | Wed Dec 01 11:23:31 2021 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Tue Dec 07 16:15:47 2021 |
tree | 7e610653cac192f55c08050d30819a69324533c5 | |
parent | 5c198e100dd901ade7609411d9e4e2712d985b93 [diff] |
Reland "Call: Deduplicate SentPacket notifications" This is a reland of Ib121d5af07abe208bd7d36715a234f48cdabb032 In order to be backward compatible with bandwidth estimation behavior, pass all packets without a |packet_id| to downstream. Original change's description: > Call: Deduplicate SentPacket notifications > > When bundling is in effect, multiple senders may be sharing the same > transport. It means every |sent_packet| will be multiply notified from > different channels, WebRtcVoiceMediaChannel or WebRtcVideoChannel. > Record |last_sent_packet_| to deduplicate redundant notifications to > downstream objects. > > This CL reduces 50% PostTask/Wakeup of Dynamic Mode Pacer. > > [1] https://datatracker.ietf.org/doc/html/rfc8829#section-4.1.1 > [2] https://datatracker.ietf.org/doc/html/rfc8843 > > Bug: webrtc:13417 > Change-Id: Ib121d5af07abe208bd7d36715a234f48cdabb032 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238720 > Reviewed-by: Markus Handell <handellm@webrtc.org> > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Tommi <tommi@webrtc.org> > Commit-Queue: Markus Handell <handellm@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35417} Bug: webrtc:13417, webrtc:13437 Change-Id: Ia5e9fbe5e4f47fe851935ca2484125411114cb68 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239437 Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35492}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.