Limit size of a video frame when packetizing it To avoid various corner cases when frame is too large. Such large frames are impractical for RTP. Add a throttled log when a video frame is dropped by a packetizer. Bug: chromium:503013378 Change-Id: Ie8c55d7c3976f0cf7420163e30e4f95f21a8b3d2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/490600 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#48214}
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.