| commit | 529f83c521c6726017bed9da16664063b0041eaf | [log] [tgz] |
|---|---|---|
| author | ehmaldonado <ehmaldonado@webrtc.org> | Wed Jul 27 15:14:32 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Jul 27 15:14:42 2016 |
| tree | 4041db5c4e2fb6575f68b38066de9b3bff619417 | |
| parent | 81031d6dca23a0c3c390f3fff24054de3651426c [diff] |
Add webrtc_perf_tests to BUILD.gn Updated the sources in audio_processing:audioproc_test_utils to match the configuration on "webrtc/modules/audio_processing/audio_processing_tests.gypi" Removed audio_buffer_tools from modules_unittests to match the gyp file. BUG=webrtc:6041 Review-Url: https://codereview.webrtc.org/2178963002 Cr-Commit-Position: refs/heads/master@{#13541}
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.