http://crbug.com/732224 should be able to be covered by FallbackDesktopCapturerWrapper if its SelectSource() behaves like this. This change also adds histogram to track the failure of SelectSource(). This change was reviewed at https://chromium-review.googlesource.com/c/external/webrtc/+/662906. Bug: chromium:732224 Change-Id: I6cffe745a48d77acb703bfb0a0d042b170d8a937 TBR: jamiewalch@chromium.org Reviewed-on: https://webrtc-review.googlesource.com/1380 Reviewed-by: Zijie He <zijiehe@google.com> Commit-Queue: Zijie He <zijiehe@google.com> Cr-Commit-Position: refs/heads/master@{#19839}
Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
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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.
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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