commit | ebafdc84843256bac7f125ff637d7e006f67d595 | [log] [tgz] |
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author | mbonadei <mbonadei@webrtc.org> | Thu Dec 22 15:35:39 2016 |
committer | Commit bot <commit-bot@chromium.org> | Thu Dec 22 15:35:39 2016 |
tree | f04a8c8c630e3c808a3a0715efd12eec8a66d216 | |
parent | 000d16396e748c7593ac41b24a876a6c20f0d277 [diff] |
Refactor webrtc/modules/rtp_rtcp for GN check 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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.