This defines the following methods: - OnFrame(), replaces SetLastFramePixelCount(). - OnFrameDroppedDueToSize(), a rename of FrameDroppedDueToSize() to match the other methods. - OnEncodeStarted(), a rename of the incorrectly named FrameCaptured(). - OnEncodeCompleted(), a rename of the poorly named FrameSent(). In order to get rid of SetLastFramePixelCount(), the "we don't know the frame size" use case - which was previously implicitly avoided by invoking SetLastFramePixelCount() with a made-up value for last_frame_info_ - is now avoided using ".value_or()" in LastInputFrameSizeOrDefault(). This does mean that a constant 144p resolution value is referenced in two places, but the fact that this is a magic value is at least made explicit. This may help future improvements. Bug: webrtc:11222 Change-Id: I3b28daa8c5ecf57c6537957d4759f15e24bb2234 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166961 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30352}
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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 http://www.webrtc.org/native-code/development 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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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