commit | 3c2359c663a6935e47e08977204c346baf44f8df | [log] [tgz] |
---|---|---|
author | Artem Titarenko <artit@webrtc.org> | Thu May 05 10:29:51 2022 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Thu May 05 10:41:13 2022 |
tree | 4d12571b27676b16fc1e1923e1a6d3a3824eaeaa | |
parent | 45361f78ed18c350b3edcaef19ae4c7cf167e95b [diff] |
Revert "RTP video stream receivers: By default consider frames decryptable." This reverts commit 658dfb74e563295b7ed4961d06c68afbd566ef8d. Reason for revert: Breaks downstream tests. Original change's description: > RTP video stream receivers: By default consider frames decryptable. > > Looks like the original code [0] that should limit the amount of keyframe requests behaves a bit strange in a situation when the first keyframe is missed. Effectively in the encrypted session the receiver can't enforce getting the keyframe until it receives at least one frame which is decryptable [1]. And with dependency descriptors it can't do that until it receives a keyframe which contains proper DD header [2]. This leads to unnecessary delays until the sender sends a keyframe itself. > > In this CL we "trust" that the stream is decryptable from the beginning unless proven the opposite [3]. > > [0]: https://webrtc-review.googlesource.com/c/src/+/123414/ > [1]: https://webrtc.googlesource.com/src/+/9432768024b0397f2dccfec0cab30f33dde87b93/video/video_receive_stream2.cc#950 > [2]: https://webrtc.googlesource.com/src/+/9432768024b0397f2dccfec0cab30f33dde87b93/video/rtp_video_stream_receiver2.cc#415 > [3]: https://webrtc.googlesource.com/src/+/9432768024b0397f2dccfec0cab30f33dde87b93/video/rtp_video_stream_receiver2.cc#882 > > Bug: webrtc:10330 > Change-Id: I167d728ddc7cde74a5c5e3327bce7364ed97b7ea > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260326 > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Commit-Queue: Artem Titarenko <artit@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#36775} Bug: webrtc:10330 Change-Id: I1e390c938502048a678a9c3a9a88a44f08dc058f No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261261 Reviewed-by: Artem Titarenko <artit@webrtc.org> Auto-Submit: Artem Titarenko <artit@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36777}
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.