ebafdc8484
This moves some GN check configurations out of .gn to individual targets. This commit also removes the source file 'mocks/mock_rtp_rtcp.h' from the static_library 'rtp_rtcp' because it depends on a 'testonly = true' target. After a check this seems only included in the unitest code: $ grep -Rn "mocks/mock_rtp_rtcp.h" webrtc/modules/rtp_rtcp/ webrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc:18:#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc:17:#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" This commit also removes the dependency on '//webrt/modules/video_coding' because it seems that the following include can be removed: #include "webrtc/modules/video_coding/include/video_coding_defines.h" The now checked target is: "//webrtc/modules/rtp_rtcp/*" BUG=webrtc:6828 NOTRY=True Review-Url: https://codereview.webrtc.org/2598963002 Cr-Commit-Position: refs/heads/master@{#15760}
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. This page is maintained by the Google Chrome team.
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://chromium.googlesource.com/external/webrtc
- 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