5fe85d23a2
This is a reland of commit 09f03be54804e81f626c26e8fde8c86cc952545f Use max_num_layers instead of encoder_config.number_of_streams when calculation stream resolutions in EncoderStreamFactory::GetStreamResolutions(). Original change's description: > Pass true stream resolutions to GetSimulcastConfig() > > Before this change GetSimulcastConfig() received only maximum resolution as an input parameter and derived resolutions for low quality simulcast streams assuming 1/2 scaling factor [1]. These days applications can configure resolution scaling factors via RtpEncodingParameters. If the configured resolution scaling factors were different from 1/2 then we got wrong bitrate limits from GetSimulcastConfig(). Now resolutions are calculated using scaling factor from RtpEncodingParameters (or default 1/2) for all streams in EncoderStreamFactory::CreateEncoderStreams() and then passed to GetSimulcastConfig(). > > Moved tests from simulcast_unittest.cc to encoder_stream_factory_unittest.cc. Mapping of old to new tests: > * GetConfigWithLimitedMaxLayersForResolution -> ReducesStreamCountWhenResolutionIsLow > * GetConfigWithLowResolutionScreenshare -> ReducesLegacyScreencastStreamCountWhenResolutionIsLow > * GetConfigWithNotLimitedMaxLayersForResolution -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled > * GetConfigWithNormalizedResolution -> AdjustsResolutionWhenUnaligned > * GetConfigWithNormalizedResolutionDivisibleBy4 -> MakesResolutionDivisibleBy4 > * GetConfigWithNormalizedResolutionDivisibleBy8 -> not needed (MakesResolutionDivisibleBy4 should be enough). > * GetConfigForLegacyLayerLimit -> KeepsStreamCountUnchangedWhenResolutionIsHigh and ReducesStreamCountWhenResolutionIsLow > * GetConfigForLegacyLayerLimitWithRequiredHD -> KeepsStreamCountUnchangedWhenLegacyLimitIsDisabled > > [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/config/simulcast.cc;l=297-298;drc=1b78a7eb3f418460da03672b1d1af1d9488bb544 > > Bug: webrtc:351644568, b/352504711 > Change-Id: I0028904ab0bb1e27b9c1b7cd3fb9a8ccf447fa35 > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357280 > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#42651} Bug: webrtc:351644568, b/352504711 Change-Id: Ib3fd859257b61c2a5d695b8b8f45c95495117c0e No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357520 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42654}
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.
Development
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation