d516b25852
This CL is part of the Call-Level Adaptation Processing design doc: https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing The "input state" of a VideoStream, needed for adaptation and decision-making, are: source resolution and frame rate, codec type and min pixels per frame (based on encoder scaling settings). These values are modified on the encoder queue of the VideoStreamEncoder. But in order to unblock call-level adaptation processing, where adaptation and decision making happens off the encoder queue, a snapshot of the input states need to be available at point of processing: introducing the VideoStreamInputState. In this CL, the VideoStreamInputStateProvider is added to provide input state snapshots across threads based on input from VideoStreamEncoder and VideoStreamEncoderObserver. The input state's HasInputFrameSizeAndFramesPerSecond() can now be DCHECKed inside the VideoStreamAdapter in favor of having less Adaptation::Status codes. Whether input is "sufficient" for adaptation is now the responsibility of the Processor. (Goal: adapter is purely a Adaptation generator and apply-er.) Somewhat tangental, this CL also deletes VideoStreamEncoder-specific methods from ResourceAdaptationProcessorInterface making them an implementation detail of ResourceAdaptationProcessor. In a future CL, the "processor" will be split up into a "processor" part and a "video stream encoder resource manager" part - more on that later. Bug: webrtc:11172 Change-Id: Id9b158f569db0140b75360aaf0f7e2e28fb924f4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172928 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Cr-Commit-Position: refs/heads/master@{#31098}
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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct