pwestin@webrtc.org 1de01354e6 Adding playout buffer status to the voe video sync ...

Review URL: https://webrtc-codereview.appspot.com/1311004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3835 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-04-11 20:23:35 +00:00

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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 UDP transport API from VoE

2013-04-03 15:43:57 +00:00