| commit | 59ade0172fdb7d521bca8ca4e837abf7e90bc604 | [log] [tgz] |
|---|---|---|
| author | Per Kjellander <perkj@webrtc.org> | Fri Dec 02 08:09:37 2022 +0000 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Fri Dec 02 12:03:25 2022 +0000 |
| tree | cf797638238868ba87195dfbc2784d5466d5cfb7 | |
| parent | 3fcd49e97237db3668f03991129126b6781b936d [diff] |
Reland "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream" This reverts commit 75170be4acc90fece7c65f1a5b9bef03a5cc3880. Reason for revert: Perf regression not affecting open source. Original change's description: > Revert "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream" > > This reverts commit d8c4de71722c9de38f942932be21d4015f32a3bc. > > Reason for revert: Tentative revert due to possible perf regression. b/260123362 > > Original change's description: > > Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream > > > > VideoSendStreamImpl::Start and VideoSendStream::Start are not used by PeerConnections, only StartPerRtpStream. > > Therefore this cl: > > - Change implementation of VideoSendStream::Start to use VideoSendStream::StartPerRtpStream. VideoSendstream::Start is kept for convenience. > > - Remove VideoSendStreamImpl::Start() since it was only used by tests that use call and is confusing. > > - RtpVideoSender::SetActive is removed/changed to RtpVideoSender::Stop(). For normal operations RtpVideoSender::SetActiveModules is used. > > > > Bug: none > > Change-Id: I43b153250b07c02fe63c84e3c4cec18d4ec0d47a > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283660 > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Commit-Queue: Per Kjellander <perkj@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#38698} > > Bug: none > Change-Id: I4f0d27679e51361b9ec54d2ae8e4d972527875d1 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284940 > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Auto-Submit: Per Kjellander <perkj@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38725} Bug: b/260400659 Change-Id: Ie8e545edcad85284a7d612183a8e4201672d0b5e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285900 Auto-Submit: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#38794}
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.