commit | de1735058bc2e170af452e18b293ba44bb2b86a7 | [log] [tgz] |
---|---|---|
author | Jonas Oreland <jonaso@webrtc.org> | Fri Jan 31 07:55:05 2025 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Fri Jan 31 09:47:56 2025 |
tree | 7313b55fabe0ef5a50e5c6a12b9cc6d87b1d35bf | |
parent | 9a407346fd6abc4557ab1b0e8f776be1cc27aa08 [diff] |
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}
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.