Currently, when input fps is higher than the configured target fps in screenshare_layers, we drop frames with the help of a rate tracker using a one second sliding window. This is not optimal. For instance, given a 5fps limit and a 30fps capturer, the window will not be saturated until we have added the first 5 frames. Then we will drop all frames until the oldest one drops out, at which point we can immediately fill it's place. This results in quick 5 frame bursts every second. In addition to rate tracker, also set a limit on minimum interval required between input frames, based on target frame rate. Bug: webrtc:4172 Change-Id: I49f0abf4d549673cc6b3fafe580b529ea3feaf57 Reviewed-on: https://webrtc-review.googlesource.com/34380 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21325}
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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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