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}
…
…
…
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.
Development
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Languages
C++
90.3%
Java
2.9%
C
2.2%
Objective-C++
2%
Python
1.3%
Other
1%