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path: root/webrtc/modules/audio_conference_mixer
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2013-01-22Replace AudioFrame's operator= with CopyFrom().andrew@webrtc.org
Enforce DISALLOW_COPY_AND_ASSIGN to catch offenders. Review URL: https://webrtc-codereview.appspot.com/1031007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3395 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-18Remove '<(library)' in gyp files.wjia@webrtc.org
This will remove all usage of '<(library)' in all webrtc gyp files. Review URL: https://webrtc-codereview.appspot.com/1049005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3392 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-07Properly zero out unmixed frames.andrew@webrtc.org
BUG=6770157 Review URL: https://webrtc-codereview.appspot.com/933037 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3248 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03Removing redundant codec unittest targets.kjellander@webrtc.org
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
2012-10-22Move src/ -> webrtc/andrew@webrtc.org
TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
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