920b653c31d0b0f3535b1bdcb0091250adbdf382
This runs `gn gen --check` with default args to detect mismatches between #includes and dependencies in the BUILD.gn files, as well as general build errors. Run this before uploading a CL for early detection, otherwise such errors will cause per-platform try jobs to fail. Bug: webrtc:8279 Change-Id: Ib87e2e3f40b8d1146ea5c1202fb113508a3f05e3 Reviewed-on: https://webrtc-review.googlesource.com/5482 Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20208}
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
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