| commit | a8fc9d940f64604cc45a64eb536435b9f24d0523 | [log] [tgz] |
|---|---|---|
| author | Tony Herre <herre@google.com> | Thu Sep 25 05:46:08 2025 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Thu Sep 25 07:59:33 2025 |
| tree | 53dca89ae1b15877ae23cb38bb0e83b0e65c8ea9 | |
| parent | 73282ed44968c79fbd6efdc0cfb7067092c2ec1b [diff] |
Reland "Store a raw_packetization bool in RtpSenderVideo rather than inferring" This reverts commit 37ba143ad7d89961d109c91448dd284d16397cbd. Reason for revert: Reland without the CHECK to allow changing downstream to set RTPSenderVideo::Config::raw_packetization rather than calling SendVideo with an absent codec_type. Original change's description: > Revert "Store a raw_packetization bool in RtpSenderVideo rather than inferring" > > This reverts commit fc1cbcf05221bf26b6fa94ecb018a5225060ddb4. > > Reason for revert: Reverting while we figure out why the check is triggering downstream > > Original change's description: > > Store a raw_packetization bool in RtpSenderVideo rather than inferring > > > > Ensure all calls to SendVideo() use a raw packetizer when they should > > rather than inferring it from the absence of |codec_type| - an issue > > when the caller doesn't know the correct packetization (eg > > RTPSenderVideoFrameTransformerDelegate). > > > > Bug: b/446768451 > > Change-Id: Ib628d69ac1697de63cc293c5f4c681d6450f72d9 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/411560 > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Auto-Submit: Tony Herre <herre@google.com> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Commit-Queue: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Guido Urdaneta <guidou@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#45724} > > Bug: b/446768451 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Change-Id: I456daecded27f9e52e126b8e5760238704f67300 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/411940 > Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org> > Bot-Commit: Rubber Stamper <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#45731} Bug: b/446768451 Change-Id: I828c13259c99a6b973c510d20b8646258454d25e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/412000 Commit-Queue: Tony Herre <herre@google.com> Reviewed-by: Henrik Boström <hbos@webrtc.org> Auto-Submit: Tony Herre <herre@google.com> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#45736}
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.