Propagate the Sframe encryptor to the video RTP send path Carries the Sframe encryptor created by RtpSenderBase down through the media channel and call/ layers until it reaches RTPSenderVideo, which now holds it as a member. Frames are still not encrypted; this CL only makes the encryptor reachable from the place that will use it. Flow: - RtpSenderBase::CreateSframeEncryptorOrError() posts the encryptor to the worker thread and hands it to the media channel. - WebRtcVideoSendChannel::SetSframeEncryptor() stores it in the channel's sframe_options_ and pushes the updated options to every existing send stream. - Streams created later pick the encryptor up on their own, because AddSendStream copies sframe_options_ into config.rtp. - RtpVideoSender forwards rtp_config.sframe_options.encryptor into RTPSenderVideo::Config, which stores it in sframe_encryptor_. Details: - SframeSendOptions gains the `encryptor` handle that was left as a TODO when the struct was introduced. - call/DEPS gets a specific_include_rules entry allowing sframe_options.h to include modules/sframe/, which checkdeps otherwise rejects for the call/ target. SetSframeEncryptor no longer takes an ssrc: - Sframe is Unified Plan only, so a send channel has a single send stream and the ssrc carried no routing information - the channel never used it to look anything up. - Storing the encryptor in the channel's sframe_options_ removes that ordering dependency entirely: streams that already exist are updated through UpdateSframeOptions(), and streams created afterwards read the options when they are built. Neither direction needs to know an ssrc. The fail-closed drop in SendEncodedImage() is left unconditional on purpose. The encryptor is now reachable but unused, so enabling the `!sframe_encryptor_` half of the check would let plaintext onto the wire for a stream that negotiated Sframe. The condition stays commented out, with a TODO, until the following CL adds the encryption itself. Bug: webrtc:479862368 Change-Id: I6e34c89d086199023dee407edb82641ec95de0f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/507440 Commit-Queue: Kacper Wasniowski <kwasniow@cisco.com> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#48801}
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.