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webrtc_m130/webrtc/voice_engine/test
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pbos@webrtc.org 6e788df19e Remove vim/emacs modelines from .gypi files
BUG=1655

Review URL: https://webrtc-codereview.appspot.com/1326005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3857 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-16 12:40:34 +00:00
..
android
Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested.
2013-03-28 09:14:36 +00:00
auto_test
Adding playout buffer status to the voe video sync
2013-04-11 20:23:35 +00:00
cmd_test
Fix no received audio in tests.
2013-04-04 17:24:15 +00:00
win_test
WebRtc_Word32 -> int32_t in voice_engine/
2013-04-09 10:09:10 +00:00
voice_engine_tests.gypi
Remove vim/emacs modelines from .gypi files
2013-04-16 12:40:34 +00:00
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