commit | 68e7b00928f99a80c65f1d15c438f46bedd839e9 | [log] [tgz] |
---|---|---|
author | Shigemasa Watanabe <shigemasa7watanabe@gmail.com> | Tue Feb 25 08:38:50 2025 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Thu Feb 27 13:03:04 2025 |
tree | 2b3952b37d3c0ed77fd1b388c99ff9f4556e3772 | |
parent | b85686c8bf1aca57b0bb0183798235f134ed1aa2 [diff] |
Reland "Revert "Reland "Allow sending to separate payload types for each simulcast index.""" This is a reland of commit Ife7d43471c85fdea9bd26cc982bce410c0d75527 In the previous implementation, there was an issue where switching codecs with VideoEncoderSelector would cause a crash, so I have fixed it. Specifically, if there are no changes to params.send_codecs, I now remove all parameters related to mixed-codec simulcast. This should ensure that when an unintended codec switch occurs, the behavior remains the same as before. Additionally, I have added a test to reproduce this crash issue. I confirmed that the issue occurred in the previous implementation and that it does not occur in the current implementation. Original change's description: > Revert "Reland "Allow sending to separate payload types for each simulcast index."" > > This reverts commit 49ac6b758cc3c28be2fc13028a829f016b453d39. > > Reason for revert: Break codec switch in singlecast. > > Original change's description: > > Reland "Allow sending to separate payload types for each simulcast index." > > > > This is a reland of commit bcb19c00ba8ab1788ba3c08f28ee1b23e0cc77b9 > > > > Original change's description: > > > Allow sending to separate payload types for each simulcast index. > > > > > > This change is for mixed-codec simulcast. > > > > > > By obtaining the payload type via RtpConfig::GetStreamConfig(), > > > the correct payload type can be retrieved regardless of whether > > > RtpConfig::stream_configs is initialized or not. > > > > > > Bug: webrtc:362277533 > > > Change-Id: I6b2a1ae66356b20a832565ce6729c3ce9e73a161 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364760 > > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > > Commit-Queue: Florent Castelli <orphis@webrtc.org> > > > Reviewed-by: Florent Castelli <orphis@webrtc.org> > > > Cr-Commit-Position: refs/heads/main@{#43197} > > > > Bug: webrtc:362277533 > > Change-Id: Ia82c3390cceb9f68315c2fd9ba5114693669af32 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374780 > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#43787} > > Bug: webrtc:362277533 > Change-Id: Ife7d43471c85fdea9bd26cc982bce410c0d75527 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376040 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Reviewed-by: Jonas Oreland <jonaso@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Commit-Queue: Jonas Oreland <jonaso@webrtc.org> > Reviewed-by: Per Kjellander <perkj@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Evan Shrubsole <eshr@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#43830} Bug: webrtc:362277533 Change-Id: I1772fa478a4fd4d5096d9bf727ed4de045861c4e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378100 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#44007}
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.