Always use gn.py in depot_tools instead of just gn. The https://cs.chromium.org/chromium/src/build/find_depot_tools.py is looking up the DEPS-pinned copy in third_party/depot_tools and adds it to the path when add_depot_tools_to_path() is called. Similar use: https: //cs.chromium.org/search/?q=%22find_depot_tools.add_depot_tools_to_path()%22&sq=package:chromium&type=cs Bug: webrtc:8393 Change-Id: I3cfa3d96b4d0f60e8099e556876bc94340b1bbb5 Reviewed-on: https://webrtc-review.googlesource.com/12540 Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@google.com> Commit-Queue: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20333}
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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