| commit | 75ea06f0faee397777badaf3568cd2824a8bd983 | [log] [tgz] |
|---|---|---|
| author | Henrik Boström <hbos@webrtc.org> | Mon Mar 20 12:36:29 2023 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Mon Mar 20 12:47:21 2023 |
| tree | 7c39421275d2117bf5cf1c18e0bc45ae39a07eb3 | |
| parent | 934a88a460f320d54addd3830a132c8adb935b7c [diff] |
Revert "Ship ability to opt-in to VP9/AV1 simulcast." This reverts commit 75990b9a8f98ea2d597a31472fb778ec4d55f698. Reason for revert: Breaks downstream, a use case of having three VP9 encodings, scalability mode only specified on the first layer (L2T2_KEY) and the other two layers not having a scalability mode but also being active=false appears to trigger a DCHECK in call/rtp_video_sender.cc:501. More investigation needed Original change's description: > Ship ability to opt-in to VP9/AV1 simulcast. > > With this unflagging, an app can opt-in to simulcast when using multiple > encodings by specifying RTCRtpEncodingParameters.scalabilityMode. This > ensures backwards-compat with apps relying on 3 encodings to mean SVC > who traditionally have not specified scalabilityMode. > > It fixes the spec/API bug of asking for simulcast and not getting > simulcast. The field trial exists only as a kill-switch with a TODO to > remove it. > > This ships initial support, however note that the VP9/AV1 simulcast uses > SimulcastRateAllocator (just like VP8/H264 simulcast). This rate > allocator uses more kbps than SvcRateAllocator. This should be revisited > to avoid significant higher bitrates, for example when comparing VP9 > simulcast to VP9 SVC. > > Shipping the ability for apps to opt-in makes it easier to exercise > these new code paths and allows initial feedback from developers, but > due to the high bitrate (= same bitrate as VP8/H264 simulcast today) > many apps may find that VP9 SVC is still more beneficial for BW reasons. > > Bug: webrtc:14884, webrtc:15005 > Change-Id: I748aae1adb47acc8a6b79b5852cff6aa47a46f5d > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298046 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Evan Shrubsole <eshr@google.com> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39601} Bug: webrtc:14884, webrtc:15005 Change-Id: Ic8f77e6a2971f493d6cd8c23faecd435058a8847 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298440 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39605}
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.