| commit | baa2059a2a70a24dca4309bb9780d60ca804e7b9 | [log] [tgz] |
|---|---|---|
| author | Tomas Gunnarsson <tommi@webrtc.org> | Fri Nov 28 18:09:56 2025 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Fri Nov 28 21:07:39 2025 |
| tree | 85cce6781b1528edd6007ffadb57ee36f61ea699 | |
| parent | 633fe880a7f56d5f48370dbaa42e191182c94c79 [diff] |
Revert "Fix Loss Based BWE V2 blocked by missing RR RTCP feedback" This reverts commit 00b7d8fdcd8e5f4712176ec6be797d358a698926. Reason for revert: This seems to have broken a test: test/scenario/video_stream_unittest.cc:267 Expected: (num_vga_frames_) > (0u), actual: 0 vs 0 Stack trace: 0x7fa730f7b92e: webrtc::test::VideoStreamTest_ResolutionAdaptsToAvailableBandwidth_Test::TestBody() @ ??:?? 0x7fa6777bfb26: testing::Test::Run() @ ??:?? 0x7fa6777c0b6b: testing::TestInfo::Run() @ ??:?? ... Google Test internal frames ... Original change's description: > Fix Loss Based BWE V2 blocked by missing RR RTCP feedback > > Loss Based BWE V2 was being blocked by an early return when RR RTCP > feedback was missing, even though V2 doesn't depend on RR RTCP packets. > > This change reorders the checks in UpdateEstimate() to: > - Check Loss Based BWE V2 readiness first > - Only check RR RTCP feedback for traditional loss-based BWE > > This allows V2 to work independently of RR RTCP while preserving > existing behavior for traditional BWE. > > Bug: webrtc:463715720 > Change-Id: Ia9ed189298c73b200f8bb41a3a9c105f4ca8095e > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/428481 > Reviewed-by: Diep Bui <diepbp@webrtc.org> > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Commit-Queue: Per Kjellander <perkj@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#46337} Bug: webrtc:463715720 Change-Id: I8471ea3db1a205276feb8fd80caf6a7fc530b562 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/429460 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Diep Bui <diepbp@webrtc.org> Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#46347}
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.