This CL changed the behavior of the logging APIs: https://webrtc-review.googlesource.com/c/src/+/4160 They now no longer automatically load the native library. So the comment needs to be updated. This change broke at least one application that was trying to enable logging before anything else. NOTRY=True TBR=sakal@webrtc.org Bug: None Change-Id: I845aa8de2eebf3687884fbf69668f56e11a3d93c Reviewed-on: https://webrtc-review.googlesource.com/6200 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20127}
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
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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