ad0f3baf90
The following targets have been merged into audio_coding_unittests: * cng_unittests * g711_unittests * g722_unittests * isacfix_unittests * pcm16b_unittests Some of them were empty and were created with the assumption they were needed in order to get code coverage (which was actually not needed). The following test has been removed since it was empty: * audio_conference_mixer_unittests BUG=none TEST=trybots passing (well, except for the unittests that are removed, they're failing until the try server configuration is updated) Review URL: https://webrtc-codereview.appspot.com/971008 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3222 4adac7df-926f-26a2-2b94-8c16560cd09d
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.