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
This is a copy of http://review.webrtc.org/864014/
This adds a FuncDurationEst to each codec instance which estimates
the duration of a packet given the packet contents and the duration
of the previous packet. By default, this simply returns the
duration of the previous packet (which is what is currently assumed
to be the duration of all future packets). This patch also provides
an initial implementation of this function for G.711 which returns
the actual number of samples in the packet.
BUG=issue1015
Review URL: https://webrtc-codereview.appspot.com/935016
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3129 4adac7df-926f-26a2-2b94-8c16560cd09d
