| commit | 82a94494b1fc211a9b9e0256fc207b012357f191 | [log] [tgz] |
|---|---|---|
| author | kjellander <kjellander@webrtc.org> | Mon Jun 13 05:12:01 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Jun 13 05:12:10 2016 |
| tree | 1953902d2c212e3791831f42765515c07fa29dcb | |
| parent | 979c268830794316dc1d05ea7242eb7310f6bc23 [diff] |
GN: Add rtc_media_unittests Changes: * Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global scope to match GYP. * Enable sctpdataengine_unittest.cc for iOS, which should have been done in https://codereview.webrtc.org/1587193006 * Renamed GN target rtc_base_test_utils -> rtc_base_tests_utils to match GYP. * Added dependencies on call, modules/video_coding and video for rtc_media. * Added dependency on audio for rtc_media_unitttests (couldn't be added to rtc_media due to circular dependency problem). BUG=webrtc:5949 TESTED=Built and ran the tests on Mac. NOTRY=True Review-Url: https://codereview.webrtc.org/2050313002 Cr-Commit-Position: refs/heads/master@{#13106}
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.