commit | 03bce3f49d84f4f88d8eca3450837cb26a272dd1 | [log] [tgz] |
---|---|---|
author | Etienne Pierre-doray <etiennep@chromium.org> | Mon Mar 29 17:36:15 2021 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Apr 06 16:59:12 2021 |
tree | f2e47fed524f5ec8a3b199a8147a7c53bc2a19d5 | |
parent | b9fa3195867408b8b97a27e9464c583edbf60276 [diff] |
Reland "[Battery]: Delay start of TaskQueuePacedSender." Take 3 This is a reland of 89cb65ed663a9000b9f7c90a78039bd85731e9ae ... and f28aade91dcc2cb8f590dc1379ac7ab5c1981909 ... and 2072b87261a6505a88561bdeab3e7405d7038eaa Reason for revert: Failing DuoGroupsMediaQualityTest due to missing TaskQueuePacedSender::EnsureStarted() in google3. Fix: This CL adds the logic behind TaskQueuePacedSender::EnsureStarted, but initializes with |is_started| = true. Once the caller in google3 is updated, |is_started| can be switched to false by default. > Original change's description: > Reason for revert: crashes due to uninitialized pacing_bitrate_ > crbug.com/1190547 > Apparently pacer() is sometimes being used before EnsureStarted() > Fix: Instead of delaying first call to SetPacingRates(), > this CL no-ops MaybeProcessPackets() until EnsureStarted() > is called for the first time. > Original change's description: > > [Battery]: Delay start of TaskQueuePacedSender. > > > > To avoid unnecessary repeating tasks, TaskQueuePacedSender is started > > only upon RtpTransportControllerSend::EnsureStarted(). > > > > More specifically, the repeating task happens in > > TaskQueuePacedSender::MaybeProcessPackets() every 500ms, using a self > > task_queue_.PostDelayedTask(). > > > > Bug: chromium:1152887 > > Change-Id: I72c96d2c4b491d5edb45a30b210b3797165cbf48 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208560 > > Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33421} > > Bug: chromium:1152887 > Change-Id: I9aba4882a64bbee7d97ace9059dea8a24c144f93 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212880 > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org> > Cr-Commit-Position: refs/heads/master@{#33554} Bug: chromium:1152887 Change-Id: Ie365562bd83aefdb2757a65e20a4cf3eece678b9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213000 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Etienne Pierre-Doray <etiennep@chromium.org> Cr-Commit-Position: refs/heads/master@{#33629}
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.