Refactor: Move ssrc, media_channel, SetFrameDecryptor, and SetFrameTransformer to RtpReceiverBase - Add media_channel() pure virtual getter to RtpReceiverInternal - Move ssrc() implementation and signaled_ssrc_ member to RtpReceiverBase - Move SetFrameTransformer/frame_transformer_ to RtpReceiverBase - Move SetFrameDecryptor/GetFrameDecryptor/frame_decryptor_ to RtpReceiverBase - Add media_channel() override to AudioRtpReceiver and VideoRtpReceiver Both concrete receivers had identical implementations of these methods. Consolidating them in the base class reduces duplication and provides a foundation for wiring SFrame decryption at the base class level. Bug: webrtc:479862368 Change-Id: I8c2769006820e683d06491522b31ba321e975d16 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/466421 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#47550}
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.