82a94494b1
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.
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